From 4f9c16ccfa26691dbb9a5d9e7d5098eb934ccdbe Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 30 Apr 2008 16:20:19 +0200 Subject: [PATCH 001/179] [ALSA] soc - tlv320aic3x - revisit clock setup This patch cleans up the clocking setup for aic3x codecs. It drops the dividers table and determines the PLL control values programatically. Under certain conditions, the PLL is disabled entirely which could save some power. Signed-off-by: Daniel Mack Acked-by: Jarkko Nikula Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 231 ++++++++++++++------------------- sound/soc/codecs/tlv320aic3x.h | 4 + 2 files changed, 98 insertions(+), 137 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 09b1661b8a3a..738b3b634d74 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -49,7 +49,7 @@ #include "tlv320aic3x.h" #define AUDIO_NAME "aic3x" -#define AIC3X_VERSION "0.1" +#define AIC3X_VERSION "0.2" /* codec private data */ struct aic3x_priv { @@ -648,81 +648,6 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) return 0; } -struct aic3x_rate_divs { - u32 mclk; - u32 rate; - u32 fsref_reg; - u8 sr_reg:4; - u8 pllj_reg; - u16 plld_reg; -}; - -/* AIC3X codec mclk clock divider coefficients */ -static const struct aic3x_rate_divs aic3x_divs[] = { - /* 8k */ - {12000000, 8000, 48000, 0xa, 16, 3840}, - {19200000, 8000, 48000, 0xa, 10, 2400}, - {22579200, 8000, 48000, 0xa, 8, 7075}, - {33868800, 8000, 48000, 0xa, 5, 8049}, - /* 11.025k */ - {12000000, 11025, 44100, 0x6, 15, 528}, - {19200000, 11025, 44100, 0x6, 9, 4080}, - {22579200, 11025, 44100, 0x6, 8, 0}, - {33868800, 11025, 44100, 0x6, 5, 3333}, - /* 16k */ - {12000000, 16000, 48000, 0x4, 16, 3840}, - {19200000, 16000, 48000, 0x4, 10, 2400}, - {22579200, 16000, 48000, 0x4, 8, 7075}, - {33868800, 16000, 48000, 0x4, 5, 8049}, - /* 22.05k */ - {12000000, 22050, 44100, 0x2, 15, 528}, - {19200000, 22050, 44100, 0x2, 9, 4080}, - {22579200, 22050, 44100, 0x2, 8, 0}, - {33868800, 22050, 44100, 0x2, 5, 3333}, - /* 32k */ - {12000000, 32000, 48000, 0x1, 16, 3840}, - {19200000, 32000, 48000, 0x1, 10, 2400}, - {22579200, 32000, 48000, 0x1, 8, 7075}, - {33868800, 32000, 48000, 0x1, 5, 8049}, - /* 44.1k */ - {12000000, 44100, 44100, 0x0, 15, 528}, - {19200000, 44100, 44100, 0x0, 9, 4080}, - {22579200, 44100, 44100, 0x0, 8, 0}, - {33868800, 44100, 44100, 0x0, 5, 3333}, - /* 48k */ - {12000000, 48000, 48000, 0x0, 16, 3840}, - {19200000, 48000, 48000, 0x0, 10, 2400}, - {22579200, 48000, 48000, 0x0, 8, 7075}, - {33868800, 48000, 48000, 0x0, 5, 8049}, - /* 64k */ - {12000000, 64000, 96000, 0x1, 16, 3840}, - {19200000, 64000, 96000, 0x1, 10, 2400}, - {22579200, 64000, 96000, 0x1, 8, 7075}, - {33868800, 64000, 96000, 0x1, 5, 8049}, - /* 88.2k */ - {12000000, 88200, 88200, 0x0, 15, 528}, - {19200000, 88200, 88200, 0x0, 9, 4080}, - {22579200, 88200, 88200, 0x0, 8, 0}, - {33868800, 88200, 88200, 0x0, 5, 3333}, - /* 96k */ - {12000000, 96000, 96000, 0x0, 16, 3840}, - {19200000, 96000, 96000, 0x0, 10, 2400}, - {22579200, 96000, 96000, 0x0, 8, 7075}, - {33868800, 96000, 96000, 0x0, 5, 8049}, -}; - -static inline int aic3x_get_divs(int mclk, int rate) -{ - int i; - - for (i = 0; i < ARRAY_SIZE(aic3x_divs); i++) { - if (aic3x_divs[i].rate == rate && aic3x_divs[i].mclk == mclk) - return i; - } - - return 0; -} - static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -730,57 +655,9 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct aic3x_priv *aic3x = codec->private_data; - int i; - u8 data, pll_p, pll_r, pll_j; - u16 pll_d; - - i = aic3x_get_divs(aic3x->sysclk, params_rate(params)); - - /* Route Left DAC to left channel input and - * right DAC to right channel input */ - data = (LDAC2LCH | RDAC2RCH); - switch (aic3x_divs[i].fsref_reg) { - case 44100: - data |= FSREF_44100; - break; - case 48000: - data |= FSREF_48000; - break; - case 88200: - data |= FSREF_44100 | DUAL_RATE_MODE; - break; - case 96000: - data |= FSREF_48000 | DUAL_RATE_MODE; - break; - } - aic3x_write(codec, AIC3X_CODEC_DATAPATH_REG, data); - - /* codec sample rate select */ - data = aic3x_divs[i].sr_reg; - data |= (data << 4); - aic3x_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data); - - /* Use PLL for generation Fsref by equation: - * Fsref = (MCLK * K * R)/(2048 * P); - * Fix P = 2 and R = 1 and calculate K, if - * K = J.D, i.e. J - an interger portion of K and D is the fractional - * one with 4 digits of precision; - * Example: - * For MCLK = 22.5792 MHz and Fsref = 48kHz: - * Select P = 2, R= 1, K = 8.7074, which results in J = 8, D = 7074 - */ - pll_p = 2; - pll_r = 1; - pll_j = aic3x_divs[i].pllj_reg; - pll_d = aic3x_divs[i].plld_reg; - - data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); - aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); - aic3x_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); - aic3x_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); - aic3x_write(codec, AIC3X_PLL_PROGC_REG, (pll_d >> 6) << PLLD_MSB_SHIFT); - aic3x_write(codec, AIC3X_PLL_PROGD_REG, - (pll_d & 0x3F) << PLLD_LSB_SHIFT); + int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; + u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; + u16 pll_d = 1; /* select data word length */ data = @@ -800,6 +677,94 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data); + /* Fsref can be 44100 or 48000 */ + fsref = (params_rate(params) % 11025 == 0) ? 44100 : 48000; + + /* Try to find a value for Q which allows us to bypass the PLL and + * generate CODEC_CLK directly. */ + for (pll_q = 2; pll_q < 18; pll_q++) + if (aic3x->sysclk / (128 * pll_q) == fsref) { + bypass_pll = 1; + break; + } + + if (bypass_pll) { + pll_q &= 0xf; + aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); + aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); + } else + aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); + + /* Route Left DAC to left channel input and + * right DAC to right channel input */ + data = (LDAC2LCH | RDAC2RCH); + data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000; + if (params_rate(params) >= 64000) + data |= DUAL_RATE_MODE; + aic3x_write(codec, AIC3X_CODEC_DATAPATH_REG, data); + + /* codec sample rate select */ + data = (fsref * 20) / params_rate(params); + if (params_rate(params) < 64000) + data /= 2; + data /= 5; + data -= 2; + data |= (data << 4); + aic3x_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data); + + if (bypass_pll) + return 0; + + /* Use PLL + * find an apropriate setup for j, d, r and p by iterating over + * p and r - j and d are calculated for each fraction. + * Up to 128 values are probed, the closest one wins the game. + * The sysclk is divided by 1000 to prevent integer overflows. + */ + codec_clk = (2048 * fsref) / (aic3x->sysclk / 1000); + + for (r = 1; r <= 16; r++) + for (p = 1; p <= 8; p++) { + int clk, tmp = (codec_clk * pll_r * 10) / pll_p; + u8 j = tmp / 10000; + u16 d = tmp % 10000; + + if (j > 63) + continue; + + if (d != 0 && aic3x->sysclk < 10000000) + continue; + + /* This is actually 1000 * ((j + (d/10000)) * r) / p + * The term had to be converted to get rid of the + * division by 10000 */ + clk = ((10000 * j * r) + (d * r)) / (10 * p); + + /* check whether this values get closer than the best + * ones we had before */ + if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) { + pll_j = j; pll_d = d; pll_r = r; pll_p = p; + last_clk = clk; + } + + /* Early exit for exact matches */ + if (clk == codec_clk) + break; + } + + if (last_clk == 0) { + printk(KERN_ERR "%s(): unable to setup PLL\n", __func__); + return -EINVAL; + } + + data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); + aic3x_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); + aic3x_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); + aic3x_write(codec, AIC3X_PLL_PROGC_REG, (pll_d >> 6) << PLLD_MSB_SHIFT); + aic3x_write(codec, AIC3X_PLL_PROGD_REG, + (pll_d & 0x3F) << PLLD_LSB_SHIFT); + return 0; } @@ -826,16 +791,8 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = codec->private_data; - switch (freq) { - case 12000000: - case 19200000: - case 22579200: - case 33868800: - aic3x->sysclk = freq; - return 0; - } - - return -EINVAL; + aic3x->sysclk = freq; + return 0; } static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index d0cdeeb629de..d49d001e6e4c 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -109,6 +109,7 @@ #define LLOPM_CTRL 86 #define RLOPM_CTRL 93 /* Clock generation control register */ +#define AIC3X_GPIOB_REG 101 #define AIC3X_CLKGEN_CTRL_REG 102 /* Page select register bits */ @@ -128,12 +129,15 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 #define PLLD_MSB_SHIFT 0 #define PLLD_LSB_SHIFT 2 /* Clock generation register bits */ +#define CODEC_CLKIN_PLLDIV 0 +#define CODEC_CLKIN_CLKDIV 1 #define PLL_CLKIN_SHIFT 4 #define MCLK_SOURCE 0x0 #define PLL_CLKDIV_SHIFT 0 From 54e7e6167d29a4a98207884b2fbd28b0b3fe91f6 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 30 Apr 2008 16:20:52 +0200 Subject: [PATCH 002/179] [ALSA] soc - tlv320aic3x - add GPIO support This patch adds support for AIC3x GPIO lines. They can be configured for many possible functions as well as be driven manually. I also introduced i2c read functionality since the GPIO state register has to be read from hardware every time and can not be served from cache. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 53 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic3x.h | 49 ++++++++++++++++++++++++++++++- 2 files changed, 101 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 738b3b634d74..957996e0eba2 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -138,6 +138,20 @@ static int aic3x_write(struct snd_soc_codec *codec, unsigned int reg, return -EIO; } +/* + * read from the aic3x register space + */ +static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg, + u8 *value) +{ + *value = reg & 0xff; + if (codec->hw_read(codec->control_data, value, 1) != 1) + return -EIO; + + aic3x_write_reg_cache(codec, reg, *value); + return 0; +} + #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ @@ -911,6 +925,33 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) return 0; } +void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state) +{ + u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG; + u8 bit = gpio ? 3: 0; + u8 val = aic3x_read_reg_cache(codec, reg) & ~(1 << bit); + aic3x_write(codec, reg, val | (!!state << bit)); +} +EXPORT_SYMBOL_GPL(aic3x_set_gpio); + +int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio) +{ + u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG; + u8 val, bit = gpio ? 2: 1; + + aic3x_read(codec, reg, &val); + return (val >> bit) & 1; +} +EXPORT_SYMBOL_GPL(aic3x_get_gpio); + +int aic3x_headset_detected(struct snd_soc_codec *codec) +{ + u8 val; + aic3x_read(codec, AIC3X_RT_IRQ_FLAGS_REG, &val); + return (val >> 2) & 1; +} +EXPORT_SYMBOL_GPL(aic3x_headset_detected); + #define AIC3X_RATES SNDRV_PCM_RATE_8000_96000 #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -977,6 +1018,7 @@ static int aic3x_resume(struct platform_device *pdev) static int aic3x_init(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->codec; + struct aic3x_setup_data *setup = socdev->codec_data; int reg, ret = 0; codec->name = "aic3x"; @@ -1067,6 +1109,10 @@ static int aic3x_init(struct snd_soc_device *socdev) /* off, with power on */ aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + /* setup GPIO functions */ + aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4); + aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4); + aic3x_add_controls(codec); aic3x_add_widgets(codec); ret = snd_soc_register_card(socdev); @@ -1174,6 +1220,12 @@ static struct i2c_client client_template = { .name = "AIC3X", .driver = &aic3x_i2c_driver, }; + +static int aic3x_i2c_read(struct i2c_client *client, u8 *value, int len) +{ + value[0] = i2c_smbus_read_byte_data(client, value[0]); + return (len == 1); +} #endif static int aic3x_probe(struct platform_device *pdev) @@ -1208,6 +1260,7 @@ static int aic3x_probe(struct platform_device *pdev) if (setup->i2c_address) { normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t) i2c_master_send; + codec->hw_read = (hw_read_t) aic3x_i2c_read; ret = i2c_add_driver(&aic3x_i2c_driver); if (ret != 0) printk(KERN_ERR "can't add i2c driver"); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index d49d001e6e4c..c1dd1ac0ceac 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -108,8 +108,14 @@ #define DACR1_2_RLOPM_VOL 92 #define LLOPM_CTRL 86 #define RLOPM_CTRL 93 -/* Clock generation control register */ +/* GPIO/IRQ registers */ +#define AIC3X_STICKY_IRQ_FLAGS_REG 96 +#define AIC3X_RT_IRQ_FLAGS_REG 97 +#define AIC3X_GPIO1_REG 98 +#define AIC3X_GPIO2_REG 99 +#define AIC3X_GPIOA_REG 100 #define AIC3X_GPIOB_REG 101 +/* Clock generation control register */ #define AIC3X_CLKGEN_CTRL_REG 102 /* Page select register bits */ @@ -175,8 +181,49 @@ /* Default input volume */ #define DEFAULT_GAIN 0x20 +/* GPIO API */ +enum { + AIC3X_GPIO1_FUNC_DISABLED = 0, + AIC3X_GPIO1_FUNC_AUDIO_WORDCLK_ADC = 1, + AIC3X_GPIO1_FUNC_CLOCK_MUX = 2, + AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV2 = 3, + AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV4 = 4, + AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV8 = 5, + AIC3X_GPIO1_FUNC_SHORT_CIRCUIT_IRQ = 6, + AIC3X_GPIO1_FUNC_AGC_NOISE_IRQ = 7, + AIC3X_GPIO1_FUNC_INPUT = 8, + AIC3X_GPIO1_FUNC_OUTPUT = 9, + AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK = 10, + AIC3X_GPIO1_FUNC_AUDIO_WORDCLK = 11, + AIC3X_GPIO1_FUNC_BUTTON_IRQ = 12, + AIC3X_GPIO1_FUNC_HEADSET_DETECT_IRQ = 13, + AIC3X_GPIO1_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 14, + AIC3X_GPIO1_FUNC_ALL_IRQ = 16 +}; + +enum { + AIC3X_GPIO2_FUNC_DISABLED = 0, + AIC3X_GPIO2_FUNC_HEADSET_DETECT_IRQ = 2, + AIC3X_GPIO2_FUNC_INPUT = 3, + AIC3X_GPIO2_FUNC_OUTPUT = 4, + AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT = 5, + AIC3X_GPIO2_FUNC_AUDIO_BITCLK = 8, + AIC3X_GPIO2_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 9, + AIC3X_GPIO2_FUNC_ALL_IRQ = 10, + AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_OR_AGC_IRQ = 11, + AIC3X_GPIO2_FUNC_HEADSET_OR_BUTTON_PRESS_OR_SHORT_CIRCUIT_IRQ = 12, + AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_IRQ = 13, + AIC3X_GPIO2_FUNC_AGC_NOISE_IRQ = 14, + AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ = 15 +}; + +void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state); +int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio); +int aic3x_headset_detected(struct snd_soc_codec *codec); + struct aic3x_setup_data { unsigned short i2c_address; + unsigned int gpio_func[2]; }; extern struct snd_soc_codec_dai aic3x_dai; From eeec12bf7b7d80d1c9cae5aae0dff7e2f928c64b Mon Sep 17 00:00:00 2001 From: Graeme Gregory Date: Wed, 30 Apr 2008 19:27:40 +0200 Subject: [PATCH 003/179] [ALSA] soc - DAPM - add hook to read state of DAPM widget This adds a hook to read the power state of a DAPM widget, I use this in the gta02 driver to expose certain DAPM widgets in the mixer for ease of audio routing. Signed-off-by: Graeme Gregory Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc-dapm.h | 2 ++ sound/soc/soc-dapm.c | 23 +++++++++++++++++++++++ 2 files changed, 25 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a105b01e06d5..40cc695b69b6 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -223,6 +223,8 @@ int snd_soc_dapm_sys_add(struct device *dev); /* dapm audio endpoint control */ int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, char *pin, int status); +int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec, + char *pin); int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec); /* dapm widget types */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index af3326c63504..9fd5ee818e89 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1342,6 +1342,29 @@ int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint); +/** + * snd_soc_dapm_get_endpoint_status - get audio endpoint status + * @codec: audio codec + * @endpoint: audio signal endpoint (or start point) + * + * Get audio endpoint status - connected or disconnected. + * + * Returns status + */ +int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec, + char *endpoint) +{ + struct snd_soc_dapm_widget *w; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (!strcmp(w->name, endpoint)) + return w->connected; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_endpoint); + /** * snd_soc_dapm_free - free dapm resources * @socdev: SoC device From 650f6b13318bce6a8d59bfa48fe15b5832623cbc Mon Sep 17 00:00:00 2001 From: Stephen Rothwell Date: Thu, 1 May 2008 10:49:18 +0200 Subject: [PATCH 004/179] [ALSA] sound: fix export symbol typo Signed-off-by: Stephen Rothwell Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9fd5ee818e89..c60200ccde60 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1363,7 +1363,7 @@ int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec, return 0; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_get_endpoint); +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_endpoint_status); /** * snd_soc_dapm_free - free dapm resources From 1894c59fdb63692f5ba2576875cc558b856935ca Mon Sep 17 00:00:00 2001 From: Tim Niemeyer Date: Mon, 5 May 2008 14:16:12 +0200 Subject: [PATCH 005/179] [ALSA] soc - Patch to add debug messages to the neo1973_wm8753 (GTA01) sound driver Signed-off-by: Tim Niemeyer Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 46 ++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 0e9d1c5f2484..e469186a407d 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -43,6 +43,14 @@ #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" +/* Debugging stuff */ +#define S3C24XX_SOC_NEO1973_WM8753_DEBUG 0 +#if S3C24XX_SOC_NEO1973_WM8753_DEBUG +#define DBG(x...) printk(KERN_DEBUG "s3c24xx-soc-neo1973-wm8753: " x) +#else +#define DBG(x...) +#endif + /* define the scenarios */ #define NEO_AUDIO_OFF 0 #define NEO_GSM_CALL_AUDIO_HANDSET 1 @@ -67,6 +75,8 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, int ret = 0; unsigned long iis_clkrate; + DBG("Entered %s\n", __func__); + iis_clkrate = s3c24xx_i2s_get_clockrate(); switch (params_rate(params)) { @@ -151,6 +161,8 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + DBG("Entered %s\n", __func__); + /* disable the PLL */ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0); } @@ -172,6 +184,8 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, int ret = 0; unsigned long iis_clkrate; + DBG("Entered %s\n", __func__); + iis_clkrate = s3c24xx_i2s_get_clockrate(); if (params_rate(params) != 8000) @@ -213,6 +227,8 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + DBG("Entered %s\n", __func__); + /* disable the PLL */ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0); } @@ -233,6 +249,8 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) { + DBG("Entered %s\n", __func__); + switch (neo1973_scenario) { case NEO_AUDIO_OFF: snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); @@ -315,6 +333,8 @@ static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + DBG("Entered %s\n", __func__); + if (neo1973_scenario == ucontrol->value.integer.value[0]) return 0; @@ -327,6 +347,8 @@ static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; static void lm4857_write_regs(void) { + DBG("Entered %s\n", __func__); + if (i2c_master_send(i2c, lm4857_regs, 4) != 4) printk(KERN_ERR "lm4857: i2c write failed\n"); } @@ -338,6 +360,8 @@ static int lm4857_get_reg(struct snd_kcontrol *kcontrol, int shift = (kcontrol->private_value >> 8) & 0x0F; int mask = (kcontrol->private_value >> 16) & 0xFF; + DBG("Entered %s\n", __func__); + ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; return 0; } @@ -364,6 +388,8 @@ static int lm4857_get_mode(struct snd_kcontrol *kcontrol, { u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; + DBG("Entered %s\n", __func__); + if (value) value -= 5; @@ -376,6 +402,8 @@ static int lm4857_set_mode(struct snd_kcontrol *kcontrol, { u8 value = ucontrol->value.integer.value[0]; + DBG("Entered %s\n", __func__); + if (value) value += 5; @@ -483,6 +511,8 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) { int i, err; + DBG("Entered %s\n", __func__); + /* set up NC codec pins */ snd_soc_dapm_set_endpoint(codec, "LOUT2", 0); snd_soc_dapm_set_endpoint(codec, "ROUT2", 0); @@ -583,6 +613,8 @@ static int lm4857_amp_probe(struct i2c_adapter *adap, int addr, int kind) { int ret; + DBG("Entered %s\n", __func__); + client_template.adapter = adap; client_template.addr = addr; @@ -606,6 +638,8 @@ exit_err: static int lm4857_i2c_detach(struct i2c_client *client) { + DBG("Entered %s\n", __func__); + i2c_detach_client(client); kfree(client); return 0; @@ -613,6 +647,8 @@ static int lm4857_i2c_detach(struct i2c_client *client) static int lm4857_i2c_attach(struct i2c_adapter *adap) { + DBG("Entered %s\n", __func__); + return i2c_probe(adap, &addr_data, lm4857_amp_probe); } @@ -620,6 +656,8 @@ static u8 lm4857_state; static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) { + DBG("Entered %s\n", __func__); + dev_dbg(&dev->dev, "lm4857_suspend\n"); lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; if (lm4857_state) { @@ -631,6 +669,8 @@ static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) static int lm4857_resume(struct i2c_client *dev) { + DBG("Entered %s\n", __func__); + if (lm4857_state) { lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); lm4857_write_regs(); @@ -640,6 +680,8 @@ static int lm4857_resume(struct i2c_client *dev) static void lm4857_shutdown(struct i2c_client *dev) { + DBG("Entered %s\n", __func__); + dev_dbg(&dev->dev, "lm4857_shutdown\n"); lm4857_regs[LM4857_CTRL] &= 0xf0; lm4857_write_regs(); @@ -671,6 +713,8 @@ static int __init neo1973_init(void) { int ret; + DBG("Entered %s\n", __func__); + neo1973_snd_device = platform_device_alloc("soc-audio", -1); if (!neo1973_snd_device) return -ENOMEM; @@ -691,6 +735,8 @@ static int __init neo1973_init(void) static void __exit neo1973_exit(void) { + DBG("Entered %s\n", __func__); + i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); } From 87af38dafe4f930921b9217c21ec6d72cad56bcc Mon Sep 17 00:00:00 2001 From: Daniel Jacobowitz Date: Wed, 7 May 2008 12:05:10 +0200 Subject: [PATCH 006/179] [ALSA] ac97 - Add virtual master control to VT1616/VT1617A codec. Enable VMASTER for VT1616 / VT1617A codec. Signed-off-by: Daniel Jacobowitz Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/drivers/Kconfig | 1 + sound/pci/ac97/ac97_patch.c | 76 +++++++++++++++++++++++++++++++++++++ 2 files changed, 77 insertions(+) diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 602b58e3b55d..159137bf4c11 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -58,6 +58,7 @@ config SND_AC97_CODEC tristate select SND_PCM select AC97_BUS + select SND_VMASTER config SND_DUMMY tristate "Dummy (/dev/null) soundcard" diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 1292dcee072d..92817f7d46d2 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3352,8 +3352,66 @@ AC97_SINGLE("Downmix LFE and Center to Front", 0x5a, 12, 1, 0), AC97_SINGLE("Downmix Surround to Front", 0x5a, 11, 1, 0), }; +static const char *slave_vols_vt1616[] = { + "Front Playback Volume", + "Surround Playback Volume", + "Center Playback Volume", + "LFE Playback Volume", + NULL +}; + +static const char *slave_sws_vt1616[] = { + "Front Playback Switch", + "Surround Playback Switch", + "Center Playback Switch", + "LFE Playback Switch", + NULL +}; + +/* find a mixer control element with the given name */ +static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97, + const char *name) +{ + struct snd_ctl_elem_id id; + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(id.name, name); + return snd_ctl_find_id(ac97->bus->card, &id); +} + +/* create a virtual master control and add slaves */ +int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, + const unsigned int *tlv, const char **slaves) +{ + struct snd_kcontrol *kctl; + const char **s; + int err; + + kctl = snd_ctl_make_virtual_master(name, tlv); + if (!kctl) + return -ENOMEM; + err = snd_ctl_add(ac97->bus->card, kctl); + if (err < 0) + return err; + + for (s = slaves; *s; s++) { + struct snd_kcontrol *sctl; + + sctl = snd_ac97_find_mixer_ctl(ac97, *s); + if (!sctl) { + snd_printdd("Cannot find slave %s, skipped\n", *s); + continue; + } + err = snd_ctl_add_slave(kctl, sctl); + if (err < 0) + return err; + } + return 0; +} + static int patch_vt1616_specific(struct snd_ac97 * ac97) { + struct snd_kcontrol *kctl; int err; if (snd_ac97_try_bit(ac97, 0x5a, 9)) @@ -3361,6 +3419,24 @@ static int patch_vt1616_specific(struct snd_ac97 * ac97) return err; if ((err = patch_build_controls(ac97, &snd_ac97_controls_vt1616[1], ARRAY_SIZE(snd_ac97_controls_vt1616) - 1)) < 0) return err; + + /* There is already a misnamed master switch. Rename it. */ + kctl = snd_ac97_find_mixer_ctl(ac97, "Master Playback Volume"); + if (!kctl) + return -EINVAL; + + snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Front Playback"); + + err = snd_ac97_add_vmaster(ac97, "Master Playback Volume", + kctl->tlv.p, slave_vols_vt1616); + if (err < 0) + return err; + + err = snd_ac97_add_vmaster(ac97, "Master Playback Switch", + NULL, slave_sws_vt1616); + if (err < 0) + return err; + return 0; } From 48008b598bb9cfffbd871512f49d84eb5b885a00 Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Thu, 8 May 2008 13:01:32 +0200 Subject: [PATCH 007/179] [ALSA] i2c: cs8427.c use put_unaligned helper Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/i2c/cs8427.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index e57e9cbe6a0f..9c3d361accfb 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include @@ -264,10 +265,7 @@ int snd_cs8427_create(struct snd_i2c_bus *bus, goto __fail; } /* write default channel status bytes */ - buf[0] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 0)); - buf[1] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 8)); - buf[2] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 16)); - buf[3] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 24)); + put_unaligned_le32(SNDRV_PCM_DEFAULT_CON_SPDIF, buf); memset(buf + 4, 0, 24 - 4); if (snd_cs8427_send_corudata(device, 0, buf, 24) < 0) goto __fail; From f3e9d5d1fd6a164611043c053de585a35d76d6a9 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 8 May 2008 15:42:15 +0200 Subject: [PATCH 008/179] [ALSA] snd_usb_caiaq: add support for 'Session I/O' interface This patch adds suport for Native Instruments new 'Guitar Rig Session I/O' audio hardware. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/usb/caiaq/caiaq-audio.c | 1 + sound/usb/caiaq/caiaq-device.c | 12 +++++++++--- sound/usb/caiaq/caiaq-device.h | 1 + 3 files changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index 24970a5c888f..b3a603325835 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -637,6 +637,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_SESSIONIO): dev->samplerates |= SNDRV_PCM_RATE_88200; dev->samplerates |= SNDRV_PCM_RATE_192000; break; diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index a972f77bd785..83175083e50f 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,14 +42,15 @@ #endif MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.6"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.8"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," "{Native Instruments, Kore Controller}," "{Native Instruments, Kore Controller 2}," - "{Native Instruments, Audio Kontrol 1}" - "{Native Instruments, Audio 8 DJ}}"); + "{Native Instruments, Audio Kontrol 1}," + "{Native Instruments, Audio 8 DJ}," + "{Native Instruments, Session I/O}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ @@ -110,6 +111,11 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_AUDIO8DJ }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_SESSIONIO + }, { /* terminator */ } }; diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h index 96a491379c60..f9fbdbae269d 100644 --- a/sound/usb/caiaq/caiaq-device.h +++ b/sound/usb/caiaq/caiaq-device.h @@ -11,6 +11,7 @@ #define USB_PID_KORECONTROLLER2 0x4712 #define USB_PID_AK1 0x0815 #define USB_PID_AUDIO8DJ 0x1978 +#define USB_PID_SESSIONIO 0x1915 #define EP1_BUFSIZE 64 #define CAIAQ_USB_STR_LEN 0xff From c17cf06bfc4417a79d452c266e819c510f6a8344 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 May 2008 12:45:56 +0200 Subject: [PATCH 009/179] [ALSA] Remove unneeded ugly hack for i386 in memalloc.c The hack for dma_alloc_coherent() is no longer needed on 2.6.26 since the base code was improved. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/core/memalloc.c | 62 ------------------------------------------- 1 file changed, 62 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 23b7bc02728b..f5d6d8d12979 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -79,68 +79,6 @@ struct snd_mem_list { #define snd_assert(expr, args...) /**/ #endif -/* - * Hacks - */ - -#if defined(__i386__) -/* - * A hack to allocate large buffers via dma_alloc_coherent() - * - * since dma_alloc_coherent always tries GFP_DMA when the requested - * pci memory region is below 32bit, it happens quite often that even - * 2 order of pages cannot be allocated. - * - * so in the following, we allocate at first without dma_mask, so that - * allocation will be done without GFP_DMA. if the area doesn't match - * with the requested region, then realloate with the original dma_mask - * again. - * - * Really, we want to move this type of thing into dma_alloc_coherent() - * so dma_mask doesn't have to be messed with. - */ - -static void *snd_dma_hack_alloc_coherent(struct device *dev, size_t size, - dma_addr_t *dma_handle, - gfp_t flags) -{ - void *ret; - u64 dma_mask, coherent_dma_mask; - - if (dev == NULL || !dev->dma_mask) - return dma_alloc_coherent(dev, size, dma_handle, flags); - dma_mask = *dev->dma_mask; - coherent_dma_mask = dev->coherent_dma_mask; - *dev->dma_mask = 0xffffffff; /* do without masking */ - dev->coherent_dma_mask = 0xffffffff; /* do without masking */ - ret = dma_alloc_coherent(dev, size, dma_handle, flags); - *dev->dma_mask = dma_mask; /* restore */ - dev->coherent_dma_mask = coherent_dma_mask; /* restore */ - if (ret) { - /* obtained address is out of range? */ - if (((unsigned long)*dma_handle + size - 1) & ~dma_mask) { - /* reallocate with the proper mask */ - dma_free_coherent(dev, size, ret, *dma_handle); - ret = dma_alloc_coherent(dev, size, dma_handle, flags); - } - } else { - /* wish to success now with the proper mask... */ - if (dma_mask != 0xffffffffUL) { - /* allocation with GFP_ATOMIC to avoid the long stall */ - flags &= ~GFP_KERNEL; - flags |= GFP_ATOMIC; - ret = dma_alloc_coherent(dev, size, dma_handle, flags); - } - } - return ret; -} - -/* redefine dma_alloc_coherent for some architectures */ -#undef dma_alloc_coherent -#define dma_alloc_coherent snd_dma_hack_alloc_coherent - -#endif /* arch */ - /* * * Generic memory allocators From d023dc0aa25d6a4f7dd0d109179a2077d22a7ad2 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:18:27 +0200 Subject: [PATCH 010/179] [ALSA] oxygen: fix version in MODULE_LICENSE Adjust the MODULE_LICENSE strings to properly reflect the actual license. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/hifier.c | 2 +- sound/pci/oxygen/oxygen.c | 2 +- sound/pci/oxygen/oxygen_lib.c | 2 +- sound/pci/oxygen/virtuoso.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 090dd4354a28..c5d2c86cacb2 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -28,7 +28,7 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("TempoTec HiFier driver"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 63f185c1ed1e..c1b30cc75935 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -43,7 +43,7 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("C-Media CMI8788 driver"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8788}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 897697d43506..6e3879527dcd 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -32,7 +32,7 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("C-Media CMI8788 helper library"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); static irqreturn_t oxygen_interrupt(int dummy, void *dev_id) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 7f84fa5deca2..4618a622a296 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -79,7 +79,7 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("Asus AVx00 driver"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{Asus,AV100},{Asus,AV200}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; From c13650079ba3bed1c0bdd9bf4a13274be7676ff6 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:19:53 +0200 Subject: [PATCH 011/179] [ALSA] oxygen: add symbol for I/O space size Remove another magic number - add a symbol for the size of the PCI I/O range. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/oxygen.h | 2 ++ sound/pci/oxygen/oxygen_lib.c | 4 ++-- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index a71c6e059260..6c6efede4d2f 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -16,6 +16,8 @@ #define PCM_AC97 5 #define PCM_COUNT 6 +#define OXYGEN_IO_SIZE 0x100 + /* model-specific configuration of outputs/inputs */ #define PLAYBACK_0_TO_I2S 0x001 #define PLAYBACK_1_TO_SPDIF 0x004 diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 6e3879527dcd..60bc15917349 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -173,7 +173,7 @@ static void oxygen_proc_read(struct snd_info_entry *entry, int i, j; snd_iprintf(buffer, "CMI8788\n\n"); - for (i = 0; i < 0x100; i += 0x10) { + for (i = 0; i < OXYGEN_IO_SIZE; i += 0x10) { snd_iprintf(buffer, "%02x:", i); for (j = 0; j < 0x10; ++j) snd_iprintf(buffer, " %02x", oxygen_read8(chip, i + j)); @@ -455,7 +455,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, } if (!(pci_resource_flags(pci, 0) & IORESOURCE_IO) || - pci_resource_len(pci, 0) < 0x100) { + pci_resource_len(pci, 0) < OXYGEN_IO_SIZE) { snd_printk(KERN_ERR "invalid PCI I/O range\n"); err = -ENXIO; goto err_pci_regions; From e58aee95806c9d2bbcfc84cb85ce958e360165ef Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:20:51 +0200 Subject: [PATCH 012/179] [ALSA] oxygen: save register writes Save the written values of all CMI8788 and AC97 registers and of some of the DAC/ADC registers so that it is possible to restore the register state later. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/oxygen.c | 17 ++++++++++-- sound/pci/oxygen/oxygen.h | 6 ++++ sound/pci/oxygen/oxygen_io.c | 22 ++++++++++++--- sound/pci/oxygen/virtuoso.c | 54 +++++++++++++++++++----------------- 4 files changed, 67 insertions(+), 32 deletions(-) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index c1b30cc75935..8287ac28fe78 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -80,6 +80,7 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); struct generic_data { u8 ak4396_ctl2; + u16 saved_wm8785_registers[2]; }; static void ak4396_write(struct oxygen *chip, unsigned int codec, @@ -99,12 +100,16 @@ static void ak4396_write(struct oxygen *chip, unsigned int codec, static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value) { + struct generic_data *data = chip->model_data; + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | OXYGEN_SPI_CLOCK_160 | (3 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); + if (reg < ARRAY_SIZE(data->saved_wm8785_registers)) + data->saved_wm8785_registers[reg] = value; } static void ak4396_init(struct oxygen *chip) @@ -135,10 +140,16 @@ static void ak5385_init(struct oxygen *chip) static void wm8785_init(struct oxygen *chip) { + struct generic_data *data = chip->model_data; + + data->saved_wm8785_registers[0] = WM8785_MCR_SLAVE | + WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST; + data->saved_wm8785_registers[1] = WM8785_WL_24; + wm8785_write(chip, WM8785_R7, 0); - wm8785_write(chip, WM8785_R0, WM8785_MCR_SLAVE | - WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST); - wm8785_write(chip, WM8785_R1, WM8785_WL_24); + wm8785_write(chip, WM8785_R0, data->saved_wm8785_registers[0]); + wm8785_write(chip, WM8785_R1, data->saved_wm8785_registers[1]); + snd_component_add(chip->card, "WM8785"); } diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 6c6efede4d2f..29c9fa4964b2 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -80,6 +80,12 @@ struct oxygen { struct work_struct spdif_input_bits_work; struct work_struct gpio_work; wait_queue_head_t ac97_waitqueue; + union { + u8 _8[OXYGEN_IO_SIZE]; + __le16 _16[OXYGEN_IO_SIZE / 2]; + __le32 _32[OXYGEN_IO_SIZE / 4]; + } saved_registers; + u16 saved_ac97_registers[2][0x40]; }; struct oxygen_model { diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 5569606ee87f..83f135f80df4 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -44,18 +44,21 @@ EXPORT_SYMBOL(oxygen_read32); void oxygen_write8(struct oxygen *chip, unsigned int reg, u8 value) { outb(value, chip->addr + reg); + chip->saved_registers._8[reg] = value; } EXPORT_SYMBOL(oxygen_write8); void oxygen_write16(struct oxygen *chip, unsigned int reg, u16 value) { outw(value, chip->addr + reg); + chip->saved_registers._16[reg / 2] = cpu_to_le16(value); } EXPORT_SYMBOL(oxygen_write16); void oxygen_write32(struct oxygen *chip, unsigned int reg, u32 value) { outl(value, chip->addr + reg); + chip->saved_registers._32[reg / 4] = cpu_to_le32(value); } EXPORT_SYMBOL(oxygen_write32); @@ -63,7 +66,10 @@ void oxygen_write8_masked(struct oxygen *chip, unsigned int reg, u8 value, u8 mask) { u8 tmp = inb(chip->addr + reg); - outb((tmp & ~mask) | (value & mask), chip->addr + reg); + tmp &= ~mask; + tmp |= value & mask; + outb(tmp, chip->addr + reg); + chip->saved_registers._8[reg] = tmp; } EXPORT_SYMBOL(oxygen_write8_masked); @@ -71,7 +77,10 @@ void oxygen_write16_masked(struct oxygen *chip, unsigned int reg, u16 value, u16 mask) { u16 tmp = inw(chip->addr + reg); - outw((tmp & ~mask) | (value & mask), chip->addr + reg); + tmp &= ~mask; + tmp |= value & mask; + outw(tmp, chip->addr + reg); + chip->saved_registers._16[reg / 2] = cpu_to_le16(tmp); } EXPORT_SYMBOL(oxygen_write16_masked); @@ -79,7 +88,10 @@ void oxygen_write32_masked(struct oxygen *chip, unsigned int reg, u32 value, u32 mask) { u32 tmp = inl(chip->addr + reg); - outl((tmp & ~mask) | (value & mask), chip->addr + reg); + tmp &= ~mask; + tmp |= value & mask; + outl(tmp, chip->addr + reg); + chip->saved_registers._32[reg / 4] = cpu_to_le32(tmp); } EXPORT_SYMBOL(oxygen_write32_masked); @@ -128,8 +140,10 @@ void oxygen_write_ac97(struct oxygen *chip, unsigned int codec, oxygen_write32(chip, OXYGEN_AC97_REGS, reg); /* require two "completed" writes, just to be sure */ if (oxygen_ac97_wait(chip, OXYGEN_AC97_INT_WRITE_DONE) >= 0 && - ++succeeded >= 2) + ++succeeded >= 2) { + chip->saved_ac97_registers[codec][index / 2] = data; return; + } } snd_printk(KERN_ERR "AC'97 write timeout\n"); } diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 4618a622a296..110786b2e3b8 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -132,6 +132,9 @@ struct xonar_data { u8 ext_power_int_reg; u8 ext_power_bit; u8 has_power; + u8 pcm1796_oversampling; + u8 cs4398_fm; + u8 cs4362a_fm; }; static void pcm1796_write(struct oxygen *chip, unsigned int codec, @@ -186,12 +189,13 @@ static void xonar_d2_init(struct oxygen *chip) data->anti_pop_delay = 300; data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE; + data->pcm1796_oversampling = PCM1796_OS_64; for (i = 0; i < 4; ++i) { pcm1796_write(chip, i, 18, PCM1796_MUTE | PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD); pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); - pcm1796_write(chip, i, 20, PCM1796_OS_64); + pcm1796_write(chip, i, 20, data->pcm1796_oversampling); pcm1796_write(chip, i, 21, 0); pcm1796_write(chip, i, 16, 0x0f); /* set ATL/ATR after ATLD */ pcm1796_write(chip, i, 17, 0x0f); @@ -226,6 +230,9 @@ static void xonar_dx_init(struct oxygen *chip) data->ext_power_reg = OXYGEN_GPI_DATA; data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; data->ext_power_bit = GPI_DX_EXT_POWER; + data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; + data->cs4362a_fm = CS4362A_FM_SINGLE | + CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, OXYGEN_2WIRE_LENGTH_8 | @@ -236,8 +243,7 @@ static void xonar_dx_init(struct oxygen *chip) cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN); cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); /* configure */ - cs4398_write(chip, 2, CS4398_FM_SINGLE | - CS4398_DEM_NONE | CS4398_DIF_LJUST); + cs4398_write(chip, 2, data->cs4398_fm); cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L); cs4398_write(chip, 4, CS4398_MUTEP_LOW | CS4398_PAMUTE); cs4398_write(chip, 5, 0xfe); @@ -249,16 +255,13 @@ static void xonar_dx_init(struct oxygen *chip) CS4362A_RMP_UP | CS4362A_ZERO_CROSS | CS4362A_SOFT_RAMP); cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE); cs4362a_write(chip, 0x05, 0); - cs4362a_write(chip, 0x06, CS4362A_FM_SINGLE | - CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); + cs4362a_write(chip, 0x06, data->cs4362a_fm); cs4362a_write(chip, 0x07, 0x7f | CS4362A_MUTE); cs4362a_write(chip, 0x08, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x09, CS4362A_FM_SINGLE | - CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); + cs4362a_write(chip, 0x09, data->cs4362a_fm); cs4362a_write(chip, 0x0a, 0x7f | CS4362A_MUTE); cs4362a_write(chip, 0x0b, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x0c, CS4362A_FM_SINGLE | - CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); + cs4362a_write(chip, 0x0c, data->cs4362a_fm); cs4362a_write(chip, 0x0d, 0x7f | CS4362A_MUTE); cs4362a_write(chip, 0x0e, 0x7f | CS4362A_MUTE); /* clear power down */ @@ -294,12 +297,13 @@ static void xonar_dx_cleanup(struct oxygen *chip) static void set_pcm1796_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { + struct xonar_data *data = chip->model_data; unsigned int i; - u8 value; - value = params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; + data->pcm1796_oversampling = + params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; for (i = 0; i < 4; ++i) - pcm1796_write(chip, i, 20, value); + pcm1796_write(chip, i, 20, data->pcm1796_oversampling); } static void update_pcm1796_volume(struct oxygen *chip) @@ -342,24 +346,24 @@ static void set_cs53x1_params(struct oxygen *chip, static void set_cs43xx_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { - u8 fm_cs4398, fm_cs4362a; + struct xonar_data *data = chip->model_data; - fm_cs4398 = CS4398_DEM_NONE | CS4398_DIF_LJUST; - fm_cs4362a = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; + data->cs4398_fm = CS4398_DEM_NONE | CS4398_DIF_LJUST; + data->cs4362a_fm = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; if (params_rate(params) <= 50000) { - fm_cs4398 |= CS4398_FM_SINGLE; - fm_cs4362a |= CS4362A_FM_SINGLE; + data->cs4398_fm |= CS4398_FM_SINGLE; + data->cs4362a_fm |= CS4362A_FM_SINGLE; } else if (params_rate(params) <= 100000) { - fm_cs4398 |= CS4398_FM_DOUBLE; - fm_cs4362a |= CS4362A_FM_DOUBLE; + data->cs4398_fm |= CS4398_FM_DOUBLE; + data->cs4362a_fm |= CS4362A_FM_DOUBLE; } else { - fm_cs4398 |= CS4398_FM_QUAD; - fm_cs4362a |= CS4362A_FM_QUAD; + data->cs4398_fm |= CS4398_FM_QUAD; + data->cs4362a_fm |= CS4362A_FM_QUAD; } - cs4398_write(chip, 2, fm_cs4398); - cs4362a_write(chip, 0x06, fm_cs4362a); - cs4362a_write(chip, 0x09, fm_cs4362a); - cs4362a_write(chip, 0x0c, fm_cs4362a); + cs4398_write(chip, 2, data->cs4398_fm); + cs4362a_write(chip, 0x06, data->cs4362a_fm); + cs4362a_write(chip, 0x09, data->cs4362a_fm); + cs4362a_write(chip, 0x0c, data->cs4362a_fm); } static void update_cs4362a_volumes(struct oxygen *chip) From bbbfb5526650cb9d01c5c230a4e3cbbc18474c41 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:21:48 +0200 Subject: [PATCH 013/179] [ALSA] oxygen: simplify DAC volume initialization When initializing the DAC volume registers, we can just use the generic volume update functions instead of setting the registers manually. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/hifier.c | 15 ++--- sound/pci/oxygen/oxygen.c | 27 ++++---- sound/pci/oxygen/virtuoso.c | 123 +++++++++++++++++------------------- 3 files changed, 77 insertions(+), 88 deletions(-) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index c5d2c86cacb2..0e425d554505 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -62,6 +62,12 @@ static void ak4396_write(struct oxygen *chip, u8 reg, u8 value) AK4396_WRITE | (reg << 8) | value); } +static void update_ak4396_volume(struct oxygen *chip) +{ + ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]); + ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); +} + static void hifier_init(struct oxygen *chip) { struct hifier_data *data = chip->model_data; @@ -70,8 +76,7 @@ static void hifier_init(struct oxygen *chip) ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); ak4396_write(chip, AK4396_CONTROL_2, data->ak4396_ctl2); ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM); - ak4396_write(chip, AK4396_LCH_ATT, 0); - ak4396_write(chip, AK4396_RCH_ATT, 0); + update_ak4396_volume(chip); snd_component_add(chip->card, "AK4396"); snd_component_add(chip->card, "CS5340"); @@ -100,12 +105,6 @@ static void set_ak4396_params(struct oxygen *chip, ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); } -static void update_ak4396_volume(struct oxygen *chip) -{ - ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]); - ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); -} - static void update_ak4396_mute(struct oxygen *chip) { struct hifier_data *data = chip->model_data; diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 8287ac28fe78..ad8d950fddc6 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -112,6 +112,18 @@ static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value) data->saved_wm8785_registers[reg] = value; } +static void update_ak4396_volume(struct oxygen *chip) +{ + unsigned int i; + + for (i = 0; i < 4; ++i) { + ak4396_write(chip, i, + AK4396_LCH_ATT, chip->dac_volume[i * 2]); + ak4396_write(chip, i, + AK4396_RCH_ATT, chip->dac_volume[i * 2 + 1]); + } +} + static void ak4396_init(struct oxygen *chip) { struct generic_data *data = chip->model_data; @@ -125,9 +137,8 @@ static void ak4396_init(struct oxygen *chip) AK4396_CONTROL_2, data->ak4396_ctl2); ak4396_write(chip, i, AK4396_CONTROL_3, AK4396_PCM); - ak4396_write(chip, i, AK4396_LCH_ATT, 0); - ak4396_write(chip, i, AK4396_RCH_ATT, 0); } + update_ak4396_volume(chip); snd_component_add(chip->card, "AK4396"); } @@ -194,18 +205,6 @@ static void set_ak4396_params(struct oxygen *chip, } } -static void update_ak4396_volume(struct oxygen *chip) -{ - unsigned int i; - - for (i = 0; i < 4; ++i) { - ak4396_write(chip, i, - AK4396_LCH_ATT, chip->dac_volume[i * 2]); - ak4396_write(chip, i, - AK4396_RCH_ATT, chip->dac_volume[i * 2 + 1]); - } -} - static void update_ak4396_mute(struct oxygen *chip) { struct generic_data *data = chip->model_data; diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 110786b2e3b8..bd9e28583518 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -182,6 +182,28 @@ static void xonar_common_init(struct oxygen *chip) oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); } +static void update_pcm1796_volume(struct oxygen *chip) +{ + unsigned int i; + + for (i = 0; i < 4; ++i) { + pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]); + pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]); + } +} + +static void update_pcm1796_mute(struct oxygen *chip) +{ + unsigned int i; + u8 value; + + value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; + if (chip->dac_mute) + value |= PCM1796_MUTE; + for (i = 0; i < 4; ++i) + pcm1796_write(chip, i, 18, value); +} + static void xonar_d2_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -192,14 +214,12 @@ static void xonar_d2_init(struct oxygen *chip) data->pcm1796_oversampling = PCM1796_OS_64; for (i = 0; i < 4; ++i) { - pcm1796_write(chip, i, 18, PCM1796_MUTE | PCM1796_DMF_DISABLED | - PCM1796_FMT_24_LJUST | PCM1796_ATLD); pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); pcm1796_write(chip, i, 20, data->pcm1796_oversampling); pcm1796_write(chip, i, 21, 0); - pcm1796_write(chip, i, 16, 0x0f); /* set ATL/ATR after ATLD */ - pcm1796_write(chip, i, 17, 0x0f); } + update_pcm1796_mute(chip); /* set ATLD before ATL/ATR */ + update_pcm1796_volume(chip); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT); oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT); @@ -221,6 +241,37 @@ static void xonar_d2x_init(struct oxygen *chip) xonar_d2_init(chip); } +static void update_cs4362a_volumes(struct oxygen *chip) +{ + u8 mute; + + mute = chip->dac_mute ? CS4362A_MUTE : 0; + cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute); + cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute); + cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute); + cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute); + cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute); + cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute); +} + +static void update_cs43xx_volume(struct oxygen *chip) +{ + cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2); + cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2); + update_cs4362a_volumes(chip); +} + +static void update_cs43xx_mute(struct oxygen *chip) +{ + u8 reg; + + reg = CS4398_MUTEP_LOW | CS4398_PAMUTE; + if (chip->dac_mute) + reg |= CS4398_MUTE_B | CS4398_MUTE_A; + cs4398_write(chip, 4, reg); + update_cs4362a_volumes(chip); +} + static void xonar_dx_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -245,9 +296,6 @@ static void xonar_dx_init(struct oxygen *chip) /* configure */ cs4398_write(chip, 2, data->cs4398_fm); cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L); - cs4398_write(chip, 4, CS4398_MUTEP_LOW | CS4398_PAMUTE); - cs4398_write(chip, 5, 0xfe); - cs4398_write(chip, 6, 0xfe); cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP | CS4398_ZERO_CROSS | CS4398_SOFT_RAMP); cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST); @@ -256,14 +304,10 @@ static void xonar_dx_init(struct oxygen *chip) cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE); cs4362a_write(chip, 0x05, 0); cs4362a_write(chip, 0x06, data->cs4362a_fm); - cs4362a_write(chip, 0x07, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x08, 0x7f | CS4362A_MUTE); cs4362a_write(chip, 0x09, data->cs4362a_fm); - cs4362a_write(chip, 0x0a, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x0b, 0x7f | CS4362A_MUTE); cs4362a_write(chip, 0x0c, data->cs4362a_fm); - cs4362a_write(chip, 0x0d, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x0e, 0x7f | CS4362A_MUTE); + update_cs43xx_volume(chip); + update_cs43xx_mute(chip); /* clear power down */ cs4398_write(chip, 8, CS4398_CPEN); cs4362a_write(chip, 0x01, CS4362A_CPEN); @@ -306,28 +350,6 @@ static void set_pcm1796_params(struct oxygen *chip, pcm1796_write(chip, i, 20, data->pcm1796_oversampling); } -static void update_pcm1796_volume(struct oxygen *chip) -{ - unsigned int i; - - for (i = 0; i < 4; ++i) { - pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]); - pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]); - } -} - -static void update_pcm1796_mute(struct oxygen *chip) -{ - unsigned int i; - u8 value; - - value = PCM1796_FMT_24_LJUST | PCM1796_ATLD; - if (chip->dac_mute) - value |= PCM1796_MUTE; - for (i = 0; i < 4; ++i) - pcm1796_write(chip, i, 18, value); -} - static void set_cs53x1_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -366,37 +388,6 @@ static void set_cs43xx_params(struct oxygen *chip, cs4362a_write(chip, 0x0c, data->cs4362a_fm); } -static void update_cs4362a_volumes(struct oxygen *chip) -{ - u8 mute; - - mute = chip->dac_mute ? CS4362A_MUTE : 0; - cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute); - cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute); - cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute); - cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute); - cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute); - cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute); -} - -static void update_cs43xx_volume(struct oxygen *chip) -{ - cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2); - cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2); - update_cs4362a_volumes(chip); -} - -static void update_cs43xx_mute(struct oxygen *chip) -{ - u8 reg; - - reg = CS4398_MUTEP_LOW | CS4398_PAMUTE; - if (chip->dac_mute) - reg |= CS4398_MUTE_B | CS4398_MUTE_A; - cs4398_write(chip, 4, reg); - update_cs4362a_volumes(chip); -} - static void xonar_gpio_changed(struct oxygen *chip) { struct xonar_data *data = chip->model_data; From 75146fc0f9368ea41419792ac8bfdd19273b4473 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:22:43 +0200 Subject: [PATCH 014/179] [ALSA] oxygen: separate out hardware initialization code Create separate functions for the code that initializes the hardware, as opposed to initializing internal driver state, so that they can be reused for resume support. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/hifier.c | 15 ++++++++--- sound/pci/oxygen/oxygen.c | 26 +++++++++++++----- sound/pci/oxygen/virtuoso.c | 54 +++++++++++++++++++++++-------------- 3 files changed, 64 insertions(+), 31 deletions(-) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 0e425d554505..6e45a58ad14b 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -68,15 +68,22 @@ static void update_ak4396_volume(struct oxygen *chip) ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); } +static void hifier_registers_init(struct oxygen *chip) +{ + struct hifier_data *data = chip->model_data; + + ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); + ak4396_write(chip, AK4396_CONTROL_2, data->ak4396_ctl2); + ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM); + update_ak4396_volume(chip); +} + static void hifier_init(struct oxygen *chip) { struct hifier_data *data = chip->model_data; data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; - ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); - ak4396_write(chip, AK4396_CONTROL_2, data->ak4396_ctl2); - ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM); - update_ak4396_volume(chip); + hifier_registers_init(chip); snd_component_add(chip->card, "AK4396"); snd_component_add(chip->card, "CS5340"); diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index ad8d950fddc6..800ae304a247 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -124,12 +124,11 @@ static void update_ak4396_volume(struct oxygen *chip) } } -static void ak4396_init(struct oxygen *chip) +static void ak4396_registers_init(struct oxygen *chip) { struct generic_data *data = chip->model_data; unsigned int i; - data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; for (i = 0; i < 4; ++i) { ak4396_write(chip, i, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); @@ -139,6 +138,14 @@ static void ak4396_init(struct oxygen *chip) AK4396_CONTROL_3, AK4396_PCM); } update_ak4396_volume(chip); +} + +static void ak4396_init(struct oxygen *chip) +{ + struct generic_data *data = chip->model_data; + + data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; + ak4396_registers_init(chip); snd_component_add(chip->card, "AK4396"); } @@ -149,6 +156,15 @@ static void ak5385_init(struct oxygen *chip) snd_component_add(chip->card, "AK5385"); } +static void wm8785_registers_init(struct oxygen *chip) +{ + struct generic_data *data = chip->model_data; + + wm8785_write(chip, WM8785_R7, 0); + wm8785_write(chip, WM8785_R0, data->saved_wm8785_registers[0]); + wm8785_write(chip, WM8785_R1, data->saved_wm8785_registers[1]); +} + static void wm8785_init(struct oxygen *chip) { struct generic_data *data = chip->model_data; @@ -156,11 +172,7 @@ static void wm8785_init(struct oxygen *chip) data->saved_wm8785_registers[0] = WM8785_MCR_SLAVE | WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST; data->saved_wm8785_registers[1] = WM8785_WL_24; - - wm8785_write(chip, WM8785_R7, 0); - wm8785_write(chip, WM8785_R0, data->saved_wm8785_registers[0]); - wm8785_write(chip, WM8785_R1, data->saved_wm8785_registers[1]); - + wm8785_registers_init(chip); snd_component_add(chip->card, "WM8785"); } diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index bd9e28583518..e95dc5717ed7 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -204,15 +204,11 @@ static void update_pcm1796_mute(struct oxygen *chip) pcm1796_write(chip, i, 18, value); } -static void xonar_d2_init(struct oxygen *chip) +static void pcm1796_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; unsigned int i; - data->anti_pop_delay = 300; - data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE; - data->pcm1796_oversampling = PCM1796_OS_64; - for (i = 0; i < 4; ++i) { pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); pcm1796_write(chip, i, 20, data->pcm1796_oversampling); @@ -220,6 +216,17 @@ static void xonar_d2_init(struct oxygen *chip) } update_pcm1796_mute(chip); /* set ATLD before ATL/ATR */ update_pcm1796_volume(chip); +} + +static void xonar_d2_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + data->anti_pop_delay = 300; + data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE; + data->pcm1796_oversampling = PCM1796_OS_64; + + pcm1796_init(chip); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT); oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT); @@ -272,24 +279,10 @@ static void update_cs43xx_mute(struct oxygen *chip) update_cs4362a_volumes(chip); } -static void xonar_dx_init(struct oxygen *chip) +static void cs43xx_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; - data->anti_pop_delay = 800; - data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; - data->ext_power_reg = OXYGEN_GPI_DATA; - data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; - data->ext_power_bit = GPI_DX_EXT_POWER; - data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; - data->cs4362a_fm = CS4362A_FM_SINGLE | - CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; - - oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, - OXYGEN_2WIRE_LENGTH_8 | - OXYGEN_2WIRE_INTERRUPT_MASK | - OXYGEN_2WIRE_SPEED_FAST); - /* set CPEN (control port mode) and power down */ cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN); cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); @@ -311,6 +304,27 @@ static void xonar_dx_init(struct oxygen *chip) /* clear power down */ cs4398_write(chip, 8, CS4398_CPEN); cs4362a_write(chip, 0x01, CS4362A_CPEN); +} + +static void xonar_dx_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + data->anti_pop_delay = 800; + data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; + data->ext_power_reg = OXYGEN_GPI_DATA; + data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->ext_power_bit = GPI_DX_EXT_POWER; + data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; + data->cs4362a_fm = CS4362A_FM_SINGLE | + CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; + + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_FAST); + + cs43xx_init(chip); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE); From 92215f3a178080bd9d7c65879499e9474e54d55c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:23:02 +0200 Subject: [PATCH 015/179] [ALSA] virtuoso: add xonar_enable_output() Move the setting of the output enable GPIO bit to a separate function. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/virtuoso.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index e95dc5717ed7..abd5313d19b5 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -162,6 +162,14 @@ static void cs4362a_write(struct oxygen *chip, u8 reg, u8 value) oxygen_write_i2c(chip, I2C_DEVICE_CS4362A, reg, value); } +static void xonar_enable_output(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + msleep(data->anti_pop_delay); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); +} + static void xonar_common_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -173,13 +181,12 @@ static void xonar_common_init(struct oxygen *chip) data->has_power = !!(oxygen_read8(chip, data->ext_power_reg) & data->ext_power_bit); } - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CS53x1_M_MASK); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_CS53x1_M_MASK | data->output_enable_bit); oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, GPIO_CS53x1_M_SINGLE, GPIO_CS53x1_M_MASK); oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); - msleep(data->anti_pop_delay); - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, data->output_enable_bit); - oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); + xonar_enable_output(chip); } static void update_pcm1796_volume(struct oxygen *chip) From 4a4bc53bc52978dd6c918531921da925fd047d95 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:24:39 +0200 Subject: [PATCH 016/179] [ALSA] oxygen: add PM support Add suspend/resume support. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/hifier.c | 5 ++ sound/pci/oxygen/oxygen.c | 12 ++++ sound/pci/oxygen/oxygen.h | 6 ++ sound/pci/oxygen/oxygen_lib.c | 100 ++++++++++++++++++++++++++++++++++ sound/pci/oxygen/oxygen_pcm.c | 1 + sound/pci/oxygen/virtuoso.c | 22 ++++++++ 6 files changed, 146 insertions(+) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 6e45a58ad14b..7442460583dd 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -146,6 +146,7 @@ static const struct oxygen_model model_hifier = { .init = hifier_init, .control_filter = hifier_control_filter, .cleanup = hifier_cleanup, + .resume = hifier_registers_init, .set_dac_params = set_ak4396_params, .set_adc_params = set_cs5340_params, .update_dac_volume = update_ak4396_volume, @@ -186,6 +187,10 @@ static struct pci_driver hifier_driver = { .id_table = hifier_ids, .probe = hifier_probe, .remove = __devexit_p(oxygen_pci_remove), +#ifdef CONFIG_PM + .suspend = oxygen_pci_suspend, + .resume = oxygen_pci_resume, +#endif }; static int __init alsa_card_hifier_init(void) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 800ae304a247..7c8ae31eb468 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -192,6 +192,12 @@ static void generic_cleanup(struct oxygen *chip) { } +static void generic_resume(struct oxygen *chip) +{ + ak4396_registers_init(chip); + wm8785_registers_init(chip); +} + static void set_ak4396_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -278,6 +284,7 @@ static const struct oxygen_model model_generic = { .owner = THIS_MODULE, .init = generic_init, .cleanup = generic_cleanup, + .resume = generic_resume, .set_dac_params = set_ak4396_params, .set_adc_params = set_wm8785_params, .update_dac_volume = update_ak4396_volume, @@ -305,6 +312,7 @@ static const struct oxygen_model model_meridian = { .owner = THIS_MODULE, .init = meridian_init, .cleanup = generic_cleanup, + .resume = ak4396_registers_init, .set_dac_params = set_ak4396_params, .set_adc_params = set_ak5385_params, .update_dac_volume = update_ak4396_volume, @@ -353,6 +361,10 @@ static struct pci_driver oxygen_driver = { .id_table = oxygen_ids, .probe = generic_oxygen_probe, .remove = __devexit_p(oxygen_pci_remove), +#ifdef CONFIG_PM + .suspend = oxygen_pci_suspend, + .resume = oxygen_pci_resume, +#endif }; static int __init alsa_card_oxygen_init(void) diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 29c9fa4964b2..74a644880074 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -97,6 +97,8 @@ struct oxygen_model { int (*control_filter)(struct snd_kcontrol_new *template); int (*mixer_init)(struct oxygen *chip); void (*cleanup)(struct oxygen *chip); + void (*suspend)(struct oxygen *chip); + void (*resume)(struct oxygen *chip); void (*pcm_hardware_filter)(unsigned int channel, struct snd_pcm_hardware *hardware); void (*set_dac_params)(struct oxygen *chip, @@ -125,6 +127,10 @@ struct oxygen_model { int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, const struct oxygen_model *model); void oxygen_pci_remove(struct pci_dev *pci); +#ifdef CONFIG_PM +int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state); +int oxygen_pci_resume(struct pci_dev *pci); +#endif /* oxygen_mixer.c */ diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 60bc15917349..22f37851045e 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -314,6 +314,10 @@ static void oxygen_init(struct oxygen *chip) OXYGEN_SPDIF_LOCK_MASK | OXYGEN_SPDIF_RATE_MASK); oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits); + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_STANDARD); oxygen_clear_bits8(chip, OXYGEN_MPU401_CONTROL, OXYGEN_MPU401_LOOPBACK); oxygen_write8(chip, OXYGEN_GPI_INTERRUPT_MASK, 0); oxygen_write16(chip, OXYGEN_GPIO_INTERRUPT_MASK, 0); @@ -534,3 +538,99 @@ void oxygen_pci_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } EXPORT_SYMBOL(oxygen_pci_remove); + +#ifdef CONFIG_PM +int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct oxygen *chip = card->private_data; + unsigned int i, saved_interrupt_mask; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + + for (i = 0; i < PCM_COUNT; ++i) + if (chip->streams[i]) + snd_pcm_suspend(chip->streams[i]); + + if (chip->model->suspend) + chip->model->suspend(chip); + + spin_lock_irq(&chip->reg_lock); + saved_interrupt_mask = chip->interrupt_mask; + chip->interrupt_mask = 0; + oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); + spin_unlock_irq(&chip->reg_lock); + + synchronize_irq(chip->irq); + flush_scheduled_work(); + chip->interrupt_mask = saved_interrupt_mask; + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} +EXPORT_SYMBOL(oxygen_pci_suspend); + +static const u32 registers_to_restore[OXYGEN_IO_SIZE / 32] = { + 0xffffffff, 0x00ff077f, 0x00011d08, 0x007f00ff, + 0x00300000, 0x00000fe4, 0x0ff7001f, 0x00000000 +}; +static const u32 ac97_registers_to_restore[2][0x40 / 32] = { + { 0x18284fa2, 0x03060000 }, + { 0x00007fa6, 0x00200000 } +}; + +static inline int is_bit_set(const u32 *bitmap, unsigned int bit) +{ + return bitmap[bit / 32] & (1 << (bit & 31)); +} + +static void oxygen_restore_ac97(struct oxygen *chip, unsigned int codec) +{ + unsigned int i; + + oxygen_write_ac97(chip, codec, AC97_RESET, 0); + msleep(1); + for (i = 1; i < 0x40; ++i) + if (is_bit_set(ac97_registers_to_restore[codec], i)) + oxygen_write_ac97(chip, codec, i * 2, + chip->saved_ac97_registers[codec][i]); +} + +int oxygen_pci_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct oxygen *chip = card->private_data; + unsigned int i; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + if (pci_enable_device(pci) < 0) { + snd_printk(KERN_ERR "cannot reenable device"); + snd_card_disconnect(card); + return -EIO; + } + pci_set_master(pci); + + oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); + for (i = 0; i < OXYGEN_IO_SIZE; ++i) + if (is_bit_set(registers_to_restore, i)) + oxygen_write8(chip, i, chip->saved_registers._8[i]); + if (chip->has_ac97_0) + oxygen_restore_ac97(chip, 0); + if (chip->has_ac97_1) + oxygen_restore_ac97(chip, 1); + + if (chip->model->resume) + chip->model->resume(chip); + + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +EXPORT_SYMBOL(oxygen_pci_resume); +#endif /* CONFIG_PM */ diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index b17c405e069d..9680aa35f81f 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -517,6 +517,7 @@ static int oxygen_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_SUSPEND: pausing = 0; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index abd5313d19b5..9a2c16bf94e0 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -359,6 +359,18 @@ static void xonar_dx_cleanup(struct oxygen *chip) oxygen_clear_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); } +static void xonar_d2_resume(struct oxygen *chip) +{ + pcm1796_init(chip); + xonar_enable_output(chip); +} + +static void xonar_dx_resume(struct oxygen *chip) +{ + cs43xx_init(chip); + xonar_enable_output(chip); +} + static void set_pcm1796_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -551,6 +563,8 @@ static const struct oxygen_model xonar_models[] = { .control_filter = xonar_d2_control_filter, .mixer_init = xonar_mixer_init, .cleanup = xonar_cleanup, + .suspend = xonar_cleanup, + .resume = xonar_d2_resume, .set_dac_params = set_pcm1796_params, .set_adc_params = set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, @@ -579,6 +593,8 @@ static const struct oxygen_model xonar_models[] = { .control_filter = xonar_d2_control_filter, .mixer_init = xonar_mixer_init, .cleanup = xonar_cleanup, + .suspend = xonar_cleanup, + .resume = xonar_d2_resume, .set_dac_params = set_pcm1796_params, .set_adc_params = set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, @@ -608,6 +624,8 @@ static const struct oxygen_model xonar_models[] = { .control_filter = xonar_dx_control_filter, .mixer_init = xonar_dx_mixer_init, .cleanup = xonar_dx_cleanup, + .suspend = xonar_dx_cleanup, + .resume = xonar_dx_resume, .set_dac_params = set_cs43xx_params, .set_adc_params = set_cs53x1_params, .update_dac_volume = update_cs43xx_volume, @@ -652,6 +670,10 @@ static struct pci_driver xonar_driver = { .id_table = xonar_ids, .probe = xonar_probe, .remove = __devexit_p(oxygen_pci_remove), +#ifdef CONFIG_PM + .suspend = oxygen_pci_suspend, + .resume = oxygen_pci_resume, +#endif }; static int __init alsa_card_xonar_init(void) From d55d7a1cbbd069f8368ec5c67480d319e7b227b9 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:25:39 +0200 Subject: [PATCH 017/179] [ALSA] oxygen: add symbols for buffer/period size constraints Introduce symbols for the buffer/period size constraints so that their limits and relationships become clearer. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/oxygen_pcm.c | 46 +++++++++++++++++++++++------------ 1 file changed, 30 insertions(+), 16 deletions(-) diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 9680aa35f81f..09a16e459de9 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -24,6 +24,16 @@ #include #include "oxygen.h" +/* most DMA channels have a 16-bit counter for 32-bit words */ +#define BUFFER_BYTES_MAX ((1 << 16) * 4) +/* the multichannel DMA channel has a 24-bit counter */ +#define BUFFER_BYTES_MAX_MULTICH ((1 << 24) * 4) + +#define PERIOD_BYTES_MIN 64 + +#define DEFAULT_BUFFER_BYTES (BUFFER_BYTES_MAX / 2) +#define DEFAULT_BUFFER_BYTES_MULTICH (1024 * 1024) + static const struct snd_pcm_hardware oxygen_stereo_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -44,11 +54,11 @@ static const struct snd_pcm_hardware oxygen_stereo_hardware = { .rate_max = 192000, .channels_min = 2, .channels_max = 2, - .buffer_bytes_max = 256 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 128 * 1024, + .buffer_bytes_max = BUFFER_BYTES_MAX, + .period_bytes_min = PERIOD_BYTES_MIN, + .period_bytes_max = BUFFER_BYTES_MAX / 2, .periods_min = 2, - .periods_max = 2048, + .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN, }; static const struct snd_pcm_hardware oxygen_multichannel_hardware = { .info = SNDRV_PCM_INFO_MMAP | @@ -70,11 +80,11 @@ static const struct snd_pcm_hardware oxygen_multichannel_hardware = { .rate_max = 192000, .channels_min = 2, .channels_max = 8, - .buffer_bytes_max = 2048 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 256 * 1024, + .buffer_bytes_max = BUFFER_BYTES_MAX_MULTICH, + .period_bytes_min = PERIOD_BYTES_MIN, + .period_bytes_max = BUFFER_BYTES_MAX_MULTICH / 2, .periods_min = 2, - .periods_max = 16384, + .periods_max = BUFFER_BYTES_MAX_MULTICH / PERIOD_BYTES_MIN, }; static const struct snd_pcm_hardware oxygen_ac97_hardware = { .info = SNDRV_PCM_INFO_MMAP | @@ -88,11 +98,11 @@ static const struct snd_pcm_hardware oxygen_ac97_hardware = { .rate_max = 48000, .channels_min = 2, .channels_max = 2, - .buffer_bytes_max = 256 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 128 * 1024, + .buffer_bytes_max = BUFFER_BYTES_MAX, + .period_bytes_min = PERIOD_BYTES_MIN, + .period_bytes_max = BUFFER_BYTES_MAX / 2, .periods_min = 2, - .periods_max = 2048, + .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN, }; static const struct snd_pcm_hardware *const oxygen_hardware[PCM_COUNT] = { @@ -664,12 +674,14 @@ int oxygen_pcm_init(struct oxygen *chip) snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - 512 * 1024, 2048 * 1024); + DEFAULT_BUFFER_BYTES_MULTICH, + BUFFER_BYTES_MAX_MULTICH); if (ins) snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - 128 * 1024, 256 * 1024); + DEFAULT_BUFFER_BYTES, + BUFFER_BYTES_MAX); } outs = !!(chip->model->pcm_dev_cfg & PLAYBACK_1_TO_SPDIF); @@ -689,7 +701,8 @@ int oxygen_pcm_init(struct oxygen *chip) strcpy(pcm->name, "Digital"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - 128 * 1024, 256 * 1024); + DEFAULT_BUFFER_BYTES, + BUFFER_BYTES_MAX); } if (chip->has_ac97_1) { @@ -719,7 +732,8 @@ int oxygen_pcm_init(struct oxygen *chip) strcpy(pcm->name, outs ? "Front Panel" : "Analog 2"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - 128 * 1024, 256 * 1024); + DEFAULT_BUFFER_BYTES, + BUFFER_BYTES_MAX); } return 0; } From ca1f30ad6c3f002d1d9b9355a53b8bbf2fe70588 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:26:01 +0200 Subject: [PATCH 018/179] [ALSA] virtuoso: restrict period time to less than 10 s Add a constraint for the period time so that there are less than ten seconds between interrupts so that ALSA does not assume that the device is dead. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/oxygen_pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 09a16e459de9..c4ad65a3406f 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -165,6 +165,12 @@ static int oxygen_open(struct snd_pcm_substream *substream, if (err < 0) return err; } + if (channel == PCM_MULTICH) { + err = snd_pcm_hw_constraint_minmax + (runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 0, 8192000); + if (err < 0) + return err; + } snd_pcm_set_sync(substream); chip->streams[channel] = substream; From 4ba1327ab8ce179c40862f3dedb4ebaaa491d737 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:51:19 +0200 Subject: [PATCH 019/179] [ALSA] soc - DAPM - Add bulk control registration Most SoC drivers cut'n'paste a loop iterating over an array to register their DAPM controls. Provide a function they can call instead. Signed-off-by: Mark Brown Cc: Graeme Gregory Cc: Frank Mandarino Cc: Jarkko Nikula Cc: Richard Purdie Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc-dapm.h | 3 +++ sound/soc/soc-dapm.c | 27 +++++++++++++++++++++++++++ 2 files changed, 30 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 40cc695b69b6..1f30616afe71 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -205,6 +205,9 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_new_control(struct snd_soc_codec *codec, const struct snd_soc_dapm_widget *widget); +int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, + const struct snd_soc_dapm_widget *widget, + int num); /* dapm path setup */ int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c60200ccde60..811d65271012 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1233,6 +1233,33 @@ int snd_soc_dapm_new_control(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); +/** + * snd_soc_dapm_new_controls - create new dapm controls + * @codec: audio codec + * @widget: widget array + * @num: number of widgets + * + * Creates new DAPM controls based upon the templates. + * + * Returns 0 for success else error. + */ +int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, + const struct snd_soc_dapm_widget *widget, + int num) +{ + int i, ret; + + for (i = 0; i < num; i++) { + ret = snd_soc_dapm_new_control(codec, widget); + if (ret < 0) + return ret; + widget++; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); + + /** * snd_soc_dapm_stream_event - send a stream event to the dapm core * @codec: audio codec From 105f1c28442301237d20b05a3d52d9987614016f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:52:19 +0200 Subject: [PATCH 020/179] [ALSA] soc - DAPM - Bulk route registration ASoC codecs and machine drivers that use DAPM routes all cut'n'paste a loop iterating over a null terminated array of routes. Factor out this into a bulk registration function, improving the error reporting for most users, and deprecate the old API to help out of tree users pick up the changes. Signed-off-by: Mark Brown Cc: Graeme Gregory Cc: Frank Mandarino Cc: Jarkko Nikula Cc: Richard Purdie Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc-dapm.h | 17 +++++++++- sound/soc/soc-dapm.c | 72 +++++++++++++++++++++++++++++++--------- 2 files changed, 73 insertions(+), 16 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 1f30616afe71..bf4cf0c1d37f 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -193,6 +193,7 @@ struct snd_soc_dapm_widget; enum snd_soc_dapm_type; struct snd_soc_dapm_path; struct snd_soc_dapm_pin; +struct snd_soc_dapm_route; /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, @@ -210,10 +211,12 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, int num); /* dapm path setup */ -int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, +int __deprecated snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink_name, const char *control_name, const char *src_name); int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec); void snd_soc_dapm_free(struct snd_soc_device *socdev); +int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, + const struct snd_soc_dapm_route *route, int num); /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, @@ -250,6 +253,18 @@ enum snd_soc_dapm_type { snd_soc_dapm_post, /* machine specific post widget - exec last */ }; +/* + * DAPM audio route definition. + * + * Defines an audio route originating at source via control and finishing + * at sink. + */ +struct snd_soc_dapm_route { + const char *sink; + const char *control; + const char *source; +}; + /* dapm audio path between two widgets */ struct snd_soc_dapm_path { char *name; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 811d65271012..1ef6d94b8357 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -841,21 +841,8 @@ int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints); -/** - * snd_soc_dapm_connect_input - connect dapm widgets - * @codec: audio codec - * @sink: name of target widget - * @control: mixer control name - * @source: name of source name - * - * Connects 2 dapm widgets together via a named audio path. The sink is - * the widget receiving the audio signal, whilst the source is the sender - * of the audio signal. - * - * Returns 0 for success else error. - */ -int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink, - const char * control, const char *source) +static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, + const char *sink, const char *control, const char *source) { struct snd_soc_dapm_path *path; struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; @@ -957,8 +944,63 @@ err: kfree(path); return ret; } + +/** + * snd_soc_dapm_connect_input - connect dapm widgets + * @codec: audio codec + * @sink: name of target widget + * @control: mixer control name + * @source: name of source name + * + * Connects 2 dapm widgets together via a named audio path. The sink is + * the widget receiving the audio signal, whilst the source is the sender + * of the audio signal. + * + * This function has been deprecated in favour of snd_soc_dapm_add_routes(). + * + * Returns 0 for success else error. + */ +int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink, + const char *control, const char *source) +{ + return snd_soc_dapm_add_route(codec, sink, control, source); +} EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input); +/** + * snd_soc_dapm_add_routes - Add routes between DAPM widgets + * @codec: codec + * @route: audio routes + * @num: number of routes + * + * Connects 2 dapm widgets together via a named audio path. The sink is + * the widget receiving the audio signal, whilst the source is the sender + * of the audio signal. + * + * Returns 0 for success else error. On error all resources can be freed + * with a call to snd_soc_card_free(). + */ +int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, + const struct snd_soc_dapm_route *route, int num) +{ + int i, ret; + + for (i = 0; i < num; i++) { + ret = snd_soc_dapm_add_route(codec, route->sink, + route->control, route->source); + if (ret < 0) { + printk(KERN_ERR "Failed to add route %s->%s\n", + route->source, + route->sink); + return ret; + } + route++; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); + /** * snd_soc_dapm_new_widgets - add new dapm widgets * @codec: audio codec From d0cc0d3a95cc3c022ee118072d243d3670ec1663 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:55:22 +0200 Subject: [PATCH 021/179] [ALSA] soc - tlv320aic3x - Convert to use bulk registration APIs Signed-off-by: Mark Brown Cc: Jarkko Nikula Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 957996e0eba2..cb8365ac0c02 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -497,7 +497,7 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_INPUT("LINE2R"), }; -static const char *intercon[][3] = { +static const struct snd_soc_dapm_route intercon[] = { /* Left Output */ {"Left DAC Mux", "DAC_L1", "Left DAC"}, {"Left DAC Mux", "DAC_L2", "Left DAC"}, @@ -641,22 +641,15 @@ static const char *intercon[][3] = { {"Right Line Out", NULL, "Right Line2 Bypass Mixer"}, {"Mono Out", NULL, "Right Line2 Bypass Mixer"}, {"Right HP Out", NULL, "Right Line2 Bypass Mixer"}, - - /* terminator */ - {NULL, NULL, NULL}, }; static int aic3x_add_widgets(struct snd_soc_codec *codec) { - int i; - - for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets, + ARRAY_SIZE(aic3x_dapm_widgets)); /* set up audio path interconnects */ - for (i = 0; intercon[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, intercon[i][0], - intercon[i][1], intercon[i][2]); + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); snd_soc_dapm_new_widgets(codec); return 0; From 25191c45aec54cd01b53391bb0b0e1e60377a5fc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:55:48 +0200 Subject: [PATCH 022/179] [ALSA] soc - Zaurus - Convert to bulk DAPM registration APIs Signed-off-by: Mark Brown Cc: Richard Purdie Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/pxa/corgi.c | 12 ++++-------- sound/soc/pxa/poodle.c | 14 +++++--------- sound/soc/pxa/spitz.c | 14 +++++--------- sound/soc/pxa/tosa.c | 14 ++++---------- 4 files changed, 18 insertions(+), 36 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 7f32a1167572..852f16d963a1 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -247,7 +247,7 @@ SND_SOC_DAPM_HP("Headset Jack", NULL), }; /* Corgi machine audio map (connections to the codec pins) */ -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* headset Jack - in = micin, out = LHPOUT*/ {"Headset Jack", NULL, "LHPOUT"}, @@ -265,8 +265,6 @@ static const char *audio_map[][3] = { /* Same as the above but no mic bias for line signals */ {"MICIN", NULL, "Line Jack"}, - - {NULL, NULL, NULL}, }; static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", @@ -303,13 +301,11 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) } /* Add corgi specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, + ARRAY_SIZE(wm8731_dapm_widgets)); /* Set up corgi specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_sync_endpoints(codec); return 0; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 7e830b218943..810f1fe158ab 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -215,8 +215,8 @@ SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event), }; -/* Corgi machine audio_mapnections to the codec pins */ -static const char *audio_map[][3] = { +/* Corgi machine connections to the codec pins */ +static const struct snd_soc_dapm_route audio_map[] = { /* headphone connected to LHPOUT1, RHPOUT1 */ {"Headphone Jack", NULL, "LHPOUT"}, @@ -225,8 +225,6 @@ static const char *audio_map[][3] = { /* speaker connected to LOUT, ROUT */ {"Ext Spk", NULL, "ROUT"}, {"Ext Spk", NULL, "LOUT"}, - - {NULL, NULL, NULL}, }; static const char *jack_function[] = {"Off", "Headphone"}; @@ -263,13 +261,11 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) } /* Add poodle specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, + ARRAY_SIZE(wm8731_dapm_widgets)); /* Set up poodle specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_sync_endpoints(codec); return 0; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index d8b8372db00e..73a6df8709e1 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -250,7 +250,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { }; /* Spitz machine audio_map */ -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* headphone connected to LOUT1, ROUT1 */ {"Headphone Jack", NULL, "LOUT1"}, @@ -269,8 +269,6 @@ static const char *audio_map[][3] = { /* line is connected to input 1 - no bias */ {"LINPUT1", NULL, "Line Jack"}, - - {NULL, NULL, NULL}, }; static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", @@ -313,13 +311,11 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) } /* Add spitz specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, + ARRAY_SIZE(wm8750_dapm_widgets)); - /* Set up spitz specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + /* Set up spitz specific audio paths */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_sync_endpoints(codec); return 0; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 7346d7e5d066..fda2aa056d53 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -154,7 +154,7 @@ SND_SOC_DAPM_SPK("Speaker", NULL), }; /* tosa audio map */ -static const char *audio_map[][3] = { +static const snd_soc_dapm_route audio_map[] = { /* headphone connected to HPOUTL, HPOUTR */ {"Headphone Jack", NULL, "HPOUTL"}, @@ -173,8 +173,6 @@ static const char *audio_map[][3] = { {"Headset Jack", NULL, "HPOUTR"}, {"LINEINR", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Headset Jack"}, - - {NULL, NULL, NULL}, }; static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", @@ -208,15 +206,11 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) } /* add tosa specific widgets */ - for (i = 0; i < ARRAY_SIZE(tosa_dapm_widgets); i++) { - snd_soc_dapm_new_control(codec, &tosa_dapm_widgets[i]); - } + snd_soc_dapm_new_controls(codec, &tosa_dapm_widgets, + ARRAY_SIZE(tosa_dapm_widgets)); /* set up tosa specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); - } + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_sync_endpoints(codec); return 0; From 51e6a8411a9440f0fdba6cdd7d779e74f89debc4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:57:37 +0200 Subject: [PATCH 023/179] [ALSA] soc - eti_b1_wm8731 - Convert to use bulk DAPM control registration Signed-off-by: Mark Brown Cc: Frank Mandarino Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/at91/eti_b1_wm8731.c | 17 ++++------------- 1 file changed, 4 insertions(+), 13 deletions(-) diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 1347dcf3f80b..4a383a4a0ff1 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -191,7 +191,7 @@ static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", NULL), }; -static const char *intercon[][3] = { +static const struct snd_soc_dapm_route intercon[] = { /* speaker connected to LHPOUT */ {"Ext Spk", NULL, "LHPOUT"}, @@ -199,9 +199,6 @@ static const char *intercon[][3] = { /* mic is connected to Mic Jack, with WM8731 Mic Bias */ {"MICIN", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Int Mic"}, - - /* terminator */ - {NULL, NULL, NULL}, }; /* @@ -209,20 +206,14 @@ static const char *intercon[][3] = { */ static int eti_b1_wm8731_init(struct snd_soc_codec *codec) { - int i; - DBG("eti_b1_wm8731_init() called\n"); /* Add specific widgets */ - for(i = 0; i < ARRAY_SIZE(eti_b1_dapm_widgets); i++) { - snd_soc_dapm_new_control(codec, &eti_b1_dapm_widgets[i]); - } + snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets, + ARRAY_SIZE(eti_b1_dapm_widgets)); /* Set up specific audio path interconnects */ - for(i = 0; intercon[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, intercon[i][0], - intercon[i][1], intercon[i][2]); - } + snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); From 8f3112d7a847c2933a42ce29f17899f585d09106 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:58:03 +0200 Subject: [PATCH 024/179] [ALSA] soc - neo1973_wm8753 - Convert to bulk DAPM registration APIs Signed-off-by: Mark Brown Cc: Graeme Gregory Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index e469186a407d..79c5027273cb 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -425,8 +425,7 @@ static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { }; -/* example machine audio_mapnections */ -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route dapm_routes[] = { /* Connections to the lm4857 amp */ {"Audio Out", NULL, "LOUT1"}, @@ -449,8 +448,6 @@ static const char *audio_map[][3] = { /* Connect the ALC pins */ {"ACIN", NULL, "ACOP"}, - - {NULL, NULL, NULL}, }; static const char *lm4857_mode[] = { @@ -526,8 +523,8 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) set_scenario_endpoints(codec, NEO_AUDIO_OFF); /* Add neo1973 specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); /* add neo1973 specific controls */ for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) { @@ -538,11 +535,9 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) return err; } - /* set up neo1973 specific audio path audio_mapnects */ - for (i = 0; audio_map[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); - } + /* set up neo1973 specific audio routes */ + err = snd_soc_dapm_add_routes(codec, dapm_routes, + ARRAY_SIZE(dapm_routes)); snd_soc_dapm_sync_endpoints(codec); return 0; From acf497f996aa08f03c62a3150abd7939ae23de4c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:58:30 +0200 Subject: [PATCH 025/179] [ALSA] soc - davinci-evm - Update for bulk DAPM registration APIs Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/davinci/davinci-evm.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index fcd165240333..4c70a0ed3397 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -71,7 +71,7 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { }; /* davinci-evm machine audio_mapnections to the codec pins */ -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* Headphone connected to HPLOUT, HPROUT */ {"Headphone Jack", NULL, "HPLOUT"}, {"Headphone Jack", NULL, "HPROUT"}, @@ -90,23 +90,17 @@ static const char *audio_map[][3] = { {"LINE2L", NULL, "Line In"}, {"LINE1R", NULL, "Line In"}, {"LINE2R", NULL, "Line In"}, - - {NULL, NULL, NULL}, }; /* Logic for a aic3x as connected on a davinci-evm */ static int evm_aic3x_init(struct snd_soc_codec *codec) { - int i; - /* Add davinci-evm specific widgets */ - for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets, + ARRAY_SIZE(aic3x_dapm_widgets)); /* Set up davinci-evm specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); /* not connected */ snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); From 1a2505988ea650b61bd07722e99080a40ff27653 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:58:57 +0200 Subject: [PATCH 026/179] [ALSA] soc - n810 - Update for bulk DAPM registration APIs Signed-off-by: Mark Brown Cc: Jarkko Nikula Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/omap/n810.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 6533563a6011..c2afffbe8da6 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -177,7 +177,7 @@ static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { {"Headphone Jack", NULL, "HPLOUT"}, {"Headphone Jack", NULL, "HPROUT"}, @@ -217,13 +217,11 @@ static int n810_aic33_init(struct snd_soc_codec *codec) } /* Add N810 specific widgets */ - for (i = 0; i < ARRAY_SIZE(aic33_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &aic33_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, aic33_dapm_widgets, + ARRAY_SIZE(aic33_dapm_widgets)); /* Set up N810 specific audio path audio_map */ - for (i = 0; i < ARRAY_SIZE(audio_map); i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_sync_endpoints(codec); From f99a633a151686a599413bef758253dfd04887d1 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 15 May 2008 11:01:36 +0200 Subject: [PATCH 027/179] [ALSA] ASoC: Convert N810 machine driver to use gpiolib Use gpiolib since it is now available for OMAPs. Change also references to HW version RX44 to product name N810. Signed-off-by: Jarkko Nikula Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/omap/n810.c | 25 ++++++++++++++----------- 1 file changed, 14 insertions(+), 11 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c2afffbe8da6..c32487b6c7b0 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -30,15 +30,15 @@ #include #include -#include +#include #include #include "omap-mcbsp.h" #include "omap-pcm.h" #include "../codecs/tlv320aic3x.h" -#define RX44_HEADSET_AMP_GPIO 10 -#define RX44_SPEAKER_AMP_GPIO 101 +#define N810_HEADSET_AMP_GPIO 10 +#define N810_SPEAKER_AMP_GPIO 101 static struct clk *sys_clkout2; static struct clk *sys_clkout2_src; @@ -154,9 +154,9 @@ static int n810_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 1); + gpio_set_value(N810_SPEAKER_AMP_GPIO, 1); else - omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 0); + gpio_set_value(N810_SPEAKER_AMP_GPIO, 0); return 0; } @@ -165,9 +165,9 @@ static int n810_jack_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 1); + gpio_set_value(N810_HEADSET_AMP_GPIO, 1); else - omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 0); + gpio_set_value(N810_HEADSET_AMP_GPIO, 0); return 0; } @@ -303,12 +303,12 @@ static int __init n810_soc_init(void) clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); - if (omap_request_gpio(RX44_HEADSET_AMP_GPIO) < 0) + if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) BUG(); - if (omap_request_gpio(RX44_SPEAKER_AMP_GPIO) < 0) + if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0) BUG(); - omap_set_gpio_direction(RX44_HEADSET_AMP_GPIO, 0); - omap_set_gpio_direction(RX44_SPEAKER_AMP_GPIO, 0); + gpio_direction_output(N810_HEADSET_AMP_GPIO, 0); + gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0); return 0; err2: @@ -323,6 +323,9 @@ err1: static void __exit n810_soc_exit(void) { + gpio_free(N810_SPEAKER_AMP_GPIO); + gpio_free(N810_HEADSET_AMP_GPIO); + platform_device_unregister(n810_snd_device); } From 02330fbaaded5b603cba112e4bbf62cdadec159a Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Fri, 16 May 2008 12:18:29 +0200 Subject: [PATCH 028/179] [ALSA] PCI168 snd-azt3328 Linux driver: another huge update - figured out 'Digital(ly) Enhanced Game Port' functionality, implemented support for it (eliminating gameport polling overhead) - removed optional joystick activation, gameport now enabled unconditionally, since we now support it via the PCI I/O space, not via conflict-prone legacy I/O (which I was thus able to DISABLE now)! - fix playback bug (a muted wave output would get unmuted upon start of playback, of course this is not what we want, thus remember mute state) - implement partial power management: when idle, lower clock rate and disable codec (reduced noise!), and disable gameport circuit when unused - instantiate OPL3 timer, too - much better implementation of snd_azf3328_mixer_write_volume_gradually() - slightly optimized interrupt handling - lots of cleanup This time, I also found a way to verify proper OPL3 operation via MIDI file playback (emulation via synth hardware). Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/azt3328.c | 1200 ++++++++++++++++++++++++++++--------------- sound/pci/azt3328.h | 197 ++++++- 2 files changed, 972 insertions(+), 425 deletions(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 5f63af6b88a2..b832333c3023 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1,6 +1,6 @@ /* * azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168). - * Copyright (C) 2002, 2005, 2006, 2007 by Andreas Mohr + * Copyright (C) 2002, 2005 - 2008 by Andreas Mohr * * Framework borrowed from Bart Hartgers's als4000.c. * Driver developed on PCI168 AP(W) version (PCI rev. 10, subsystem ID 1801), @@ -35,9 +35,20 @@ * (3 weeks' worth of evenings filled with driver work). * (and no, I did NOT go the easy way: to pick up a SB PCI128 for 9 Euros) * - * The AZF3328 chip (note: AZF3328, *not* AZT3328, that's just the driver name - * for compatibility reasons) has the following features: + * It is quite likely that the AZF3328 chip is the PCI cousin of the + * AZF3318 ("azt1020 pnp", "MM Pro 16") ISA chip, given very similar specs. * + * The AZF3328 chip (note: AZF3328, *not* AZT3328, that's just the driver name + * for compatibility reasons) from Azfin (joint-venture of Aztech and Fincitec, + * Fincitec acquired by National Semiconductor in 2002, together with the + * Fincitec-related company ARSmikro) has the following features: + * + * - compatibility & compliance: + * - Microsoft PC 97 ("PC 97 Hardware Design Guide", + * http://www.microsoft.com/whdc/archive/pcguides.mspx) + * - Microsoft PC 98 Baseline Audio + * - MPU401 UART + * - Sound Blaster Emulation (DOS Box) * - builtin AC97 conformant codec (SNR over 80dB) * Note that "conformant" != "compliant"!! this chip's mixer register layout * *differs* from the standard AC97 layout: @@ -48,21 +59,28 @@ * addresses illegally. So far unfortunately it looks like the very flexible * ALSA AC97 support is still not enough to easily compensate for such a * grave layout violation despite all tweaks and quirks mechanisms it offers. - * - builtin genuine OPL3 + * - builtin genuine OPL3 - verified to work fine, 20080506 * - full duplex 16bit playback/record at independent sampling rate - * - MPU401 (+ legacy address support) FIXME: how to enable legacy addr?? + * - MPU401 (+ legacy address support, claimed by one official spec sheet) + * FIXME: how to enable legacy addr?? * - game port (legacy address support) - * - builtin 3D enhancement (said to be YAMAHA Ymersion) * - builtin DirectInput support, helps reduce CPU overhead (interrupt-driven - * features supported) + * features supported). - See common term "Digital Enhanced Game Port"... + * (probably DirectInput 3.0 spec - confirm) + * - builtin 3D enhancement (said to be YAMAHA Ymersion) * - built-in General DirectX timer having a 20 bits counter * with 1us resolution (see below!) - * - I2S serial port for external DAC + * - I2S serial output port for external DAC * - supports 33MHz PCI spec 2.1, PCI power management 1.0, compliant with ACPI * - supports hardware volume control * - single chip low cost solution (128 pin QFP) * - supports programmable Sub-vendor and Sub-system ID * required for Microsoft's logo compliance (FIXME: where?) + * At least the Trident 4D Wave DX has one bit somewhere + * to enable writes to PCI subsystem VID registers, that should be it. + * This might easily be in extended PCI reg space, since PCI168 also has + * some custom data starting at 0x80. What kind of config settings + * are located in our extended PCI space anyway?? * - PCI168 AP(W) card: power amplifier with 4 Watts/channel at 4 Ohms * * Note that this driver now is actually *better* than the Windows driver, @@ -74,6 +92,24 @@ * - "timidity -iAv -B2,8 -Os -EFreverb=0" * - "pmidi -p 128:0 jazz.mid" * + * OPL3 hardware playback testing, try something like: + * cat /proc/asound/hwdep + * and + * aconnect -o + * Then use + * sbiload -Dhw:x,y --opl3 /usr/share/sounds/opl3/std.o3 ......./drums.o3 + * where x,y is the xx-yy number as given in hwdep. + * Then try + * pmidi -p a:b jazz.mid + * where a:b is the client number plus 0 usually, as given by aconnect above. + * Oh, and make sure to unmute the FM mixer control (doh!) + * NOTE: power use during OPL3 playback is _VERY_ high (70W --> 90W!) + * despite no CPU activity, possibly due to hindering ACPI idling somehow. + * Shouldn't be a problem of the AZF3328 chip itself, I'd hope. + * Higher PCM / FM mixer levels seem to conflict (causes crackling), + * at least sometimes. Maybe even use with hardware sequencer timer above :) + * adplay/adplug-utils might soon offer hardware-based OPL3 playback, too. + * * Certain PCI versions of this card are susceptible to DMA traffic underruns * in some systems (resulting in sound crackling/clicking/popping), * probably because they don't have a DMA FIFO buffer or so. @@ -87,6 +123,8 @@ * better than a VIA, yet ironically I still get crackling, like many other * people with the same chipset. * Possible remedies: + * - use speaker (amplifier) output instead of headphone output + * (in case crackling is due to overloaded output clipping) * - plug card into a different PCI slot, preferrably one that isn't shared * too much (this helps a lot, but not completely!) * - get rid of PCI VGA card, use AGP instead @@ -94,18 +132,23 @@ * - fiddle with PCI latency settings (setpci -v -s BUSID latency_timer=XX) * Not too helpful. * - Disable ACPI/power management/"Auto Detect RAM/PCI Clk" in BIOS - * + * * BUGS - * - full-duplex might *still* be problematic, not fully tested recently + * - full-duplex might *still* be problematic, however a recent test was fine * - (non-bug) "Bass/Treble or 3D settings don't work" - they do get evaluated * if you set PCM output switch to "pre 3D" instead of "post 3D". * If this can't be set, then get a mixer application that Isn't Stupid (tm) * (e.g. kmix, gamix) - unfortunately several are!! - * + * - locking is not entirely clean, especially the audio stream activity + * ints --> may be racy + * - an _unconnected_ secondary joystick at the gameport will be reported + * to be "active" (floating values, not precisely -1) due to the way we need + * to read the Digital Enhanced Game Port. Not sure whether it is fixable. + * * TODO * - test MPU401 MIDI playback etc. - * - add some power micro-management (disable various units of the card - * as long as they're unused). However this requires I/O ports which I + * - add more power micro-management (disable various units of the card + * as long as they're unused). However this requires more I/O ports which I * haven't figured out yet and which thus might not even exist... * The standard suspend/resume functionality could probably make use of * some improvement, too... @@ -113,6 +156,7 @@ * - figure out some cleverly evil scheme to possibly make ALSA AC97 code * fully accept our quite incompatible ""AC97"" mixer and thus save some * code (but I'm not too optimistic that doing this is possible at all) + * - use MMIO (memory-mapped I/O)? Slightly faster access, e.g. for gameport. */ #include @@ -138,7 +182,7 @@ MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE)) -#define SUPPORT_JOYSTICK 1 +#define SUPPORT_GAMEPORT 1 #endif #define DEBUG_MISC 0 @@ -147,13 +191,14 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #define DEBUG_PLAY_REC 0 #define DEBUG_IO 0 #define DEBUG_TIMER 0 +#define DEBUG_GAME 0 #define MIXER_TESTING 0 #if DEBUG_MISC #define snd_azf3328_dbgmisc(format, args...) printk(KERN_ERR format, ##args) #else #define snd_azf3328_dbgmisc(format, args...) -#endif +#endif #if DEBUG_CALLS #define snd_azf3328_dbgcalls(format, args...) printk(format, ##args) @@ -163,25 +208,31 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #define snd_azf3328_dbgcalls(format, args...) #define snd_azf3328_dbgcallenter() #define snd_azf3328_dbgcallleave() -#endif +#endif #if DEBUG_MIXER #define snd_azf3328_dbgmixer(format, args...) printk(format, ##args) #else #define snd_azf3328_dbgmixer(format, args...) -#endif +#endif #if DEBUG_PLAY_REC #define snd_azf3328_dbgplay(format, args...) printk(KERN_ERR format, ##args) #else #define snd_azf3328_dbgplay(format, args...) -#endif +#endif #if DEBUG_MISC #define snd_azf3328_dbgtimer(format, args...) printk(KERN_ERR format, ##args) #else #define snd_azf3328_dbgtimer(format, args...) -#endif +#endif + +#if DEBUG_GAME +#define snd_azf3328_dbggame(format, args...) printk(KERN_ERR format, ##args) +#else +#define snd_azf3328_dbggame(format, args...) +#endif static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ module_param_array(index, int, NULL, 0444); @@ -195,39 +246,44 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable AZF3328 soundcard."); -#ifdef SUPPORT_JOYSTICK -static int joystick[SNDRV_CARDS]; -module_param_array(joystick, bool, NULL, 0444); -MODULE_PARM_DESC(joystick, "Enable joystick for AZF3328 soundcard."); -#endif - static int seqtimer_scaling = 128; module_param(seqtimer_scaling, int, 0444); MODULE_PARM_DESC(seqtimer_scaling, "Set 1024000Hz sequencer timer scale factor (lockup danger!). Default 128."); +struct snd_azf3328_audio_stream { + struct snd_pcm_substream *substream; + int enabled; + int running; + unsigned long portbase; +}; + +enum snd_azf3328_stream_index { + AZF_PLAYBACK = 0, + AZF_CAPTURE = 1, +}; + struct snd_azf3328 { /* often-used fields towards beginning, then grouped */ - unsigned long codec_port; - unsigned long io2_port; - unsigned long mpu_port; - unsigned long synth_port; - unsigned long mixer_port; + + unsigned long codec_io; /* usually 0xb000, size 128 */ + unsigned long game_io; /* usually 0xb400, size 8 */ + unsigned long mpu_io; /* usually 0xb800, size 4 */ + unsigned long opl3_io; /* usually 0xbc00, size 8 */ + unsigned long mixer_io; /* usually 0xc000, size 64 */ spinlock_t reg_lock; struct snd_timer *timer; - + struct snd_pcm *pcm; - struct snd_pcm_substream *playback_substream; - struct snd_pcm_substream *capture_substream; - unsigned int is_playing; - unsigned int is_recording; + struct snd_azf3328_audio_stream audio_stream[2]; struct snd_card *card; struct snd_rawmidi *rmidi; -#ifdef SUPPORT_JOYSTICK +#ifdef SUPPORT_GAMEPORT struct gameport *gameport; + int axes[4]; #endif struct pci_dev *pci; @@ -236,10 +292,10 @@ struct snd_azf3328 { #ifdef CONFIG_PM /* register value containers for power management * Note: not always full I/O range preserved (just like Win driver!) */ - u16 saved_regs_codec [AZF_IO_SIZE_CODEC_PM / 2]; - u16 saved_regs_io2 [AZF_IO_SIZE_IO2_PM / 2]; - u16 saved_regs_mpu [AZF_IO_SIZE_MPU_PM / 2]; - u16 saved_regs_synth[AZF_IO_SIZE_SYNTH_PM / 2]; + u16 saved_regs_codec[AZF_IO_SIZE_CODEC_PM / 2]; + u16 saved_regs_game [AZF_IO_SIZE_GAME_PM / 2]; + u16 saved_regs_mpu [AZF_IO_SIZE_MPU_PM / 2]; + u16 saved_regs_opl3 [AZF_IO_SIZE_OPL3_PM / 2]; u16 saved_regs_mixer[AZF_IO_SIZE_MIXER_PM / 2]; #endif }; @@ -252,126 +308,181 @@ static const struct pci_device_id snd_azf3328_ids[] = { MODULE_DEVICE_TABLE(pci, snd_azf3328_ids); -static inline void -snd_azf3328_codec_outb(const struct snd_azf3328 *chip, int reg, u8 value) + +static int +snd_azf3328_io_reg_setb(unsigned reg, u8 mask, int do_set) { - outb(value, chip->codec_port + reg); + u8 prev = inb(reg), new; + + new = (do_set) ? (prev|mask) : (prev & ~mask); + /* we need to always write the new value no matter whether it differs + * or not, since some register bits don't indicate their setting */ + outb(new, reg); + if (new != prev) + return 1; + + return 0; +} + +static int +snd_azf3328_io_reg_setw(unsigned reg, u16 mask, int do_set) +{ + u16 prev = inw(reg), new; + + new = (do_set) ? (prev|mask) : (prev & ~mask); + /* we need to always write the new value no matter whether it differs + * or not, since some register bits don't indicate their setting */ + outw(new, reg); + if (new != prev) + return 1; + + return 0; +} + +static inline void +snd_azf3328_codec_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) +{ + outb(value, chip->codec_io + reg); } static inline u8 -snd_azf3328_codec_inb(const struct snd_azf3328 *chip, int reg) +snd_azf3328_codec_inb(const struct snd_azf3328 *chip, unsigned reg) { - return inb(chip->codec_port + reg); + return inb(chip->codec_io + reg); } static inline void -snd_azf3328_codec_outw(const struct snd_azf3328 *chip, int reg, u16 value) +snd_azf3328_codec_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) { - outw(value, chip->codec_port + reg); + outw(value, chip->codec_io + reg); } static inline u16 -snd_azf3328_codec_inw(const struct snd_azf3328 *chip, int reg) +snd_azf3328_codec_inw(const struct snd_azf3328 *chip, unsigned reg) { - return inw(chip->codec_port + reg); + return inw(chip->codec_io + reg); } static inline void -snd_azf3328_codec_outl(const struct snd_azf3328 *chip, int reg, u32 value) +snd_azf3328_codec_outl(const struct snd_azf3328 *chip, unsigned reg, u32 value) { - outl(value, chip->codec_port + reg); + outl(value, chip->codec_io + reg); +} + +static inline u32 +snd_azf3328_codec_inl(const struct snd_azf3328 *chip, unsigned reg) +{ + return inl(chip->codec_io + reg); } static inline void -snd_azf3328_io2_outb(const struct snd_azf3328 *chip, int reg, u8 value) +snd_azf3328_game_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) { - outb(value, chip->io2_port + reg); + outb(value, chip->game_io + reg); +} + +static inline void +snd_azf3328_game_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) +{ + outw(value, chip->game_io + reg); } static inline u8 -snd_azf3328_io2_inb(const struct snd_azf3328 *chip, int reg) +snd_azf3328_game_inb(const struct snd_azf3328 *chip, unsigned reg) { - return inb(chip->io2_port + reg); -} - -static inline void -snd_azf3328_mixer_outw(const struct snd_azf3328 *chip, int reg, u16 value) -{ - outw(value, chip->mixer_port + reg); + return inb(chip->game_io + reg); } static inline u16 -snd_azf3328_mixer_inw(const struct snd_azf3328 *chip, int reg) +snd_azf3328_game_inw(const struct snd_azf3328 *chip, unsigned reg) { - return inw(chip->mixer_port + reg); + return inw(chip->game_io + reg); } -static void -snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip, int reg, int do_mute) +static inline void +snd_azf3328_mixer_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) { - unsigned long portbase = chip->mixer_port + reg + 1; - unsigned char oldval; + outw(value, chip->mixer_io + reg); +} + +static inline u16 +snd_azf3328_mixer_inw(const struct snd_azf3328 *chip, unsigned reg) +{ + return inw(chip->mixer_io + reg); +} + +#define AZF_MUTE_BIT 0x80 + +static int +snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip, + unsigned reg, int do_mute +) +{ + unsigned long portbase = chip->mixer_io + reg + 1; + int updated; /* the mute bit is on the *second* (i.e. right) register of a * left/right channel setting */ - oldval = inb(portbase); - if (do_mute) - oldval |= 0x80; - else - oldval &= ~0x80; - outb(oldval, portbase); + updated = snd_azf3328_io_reg_setb(portbase, AZF_MUTE_BIT, do_mute); + + /* indicate whether it was muted before */ + return (do_mute) ? !updated : updated; } static void -snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg, unsigned char dst_vol_left, unsigned char dst_vol_right, int chan_sel, int delay) +snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, + unsigned reg, + unsigned char dst_vol_left, + unsigned char dst_vol_right, + int chan_sel, int delay +) { - unsigned long portbase = chip->mixer_port + reg; + unsigned long portbase = chip->mixer_io + reg; unsigned char curr_vol_left = 0, curr_vol_right = 0; - int left_done = 0, right_done = 0; - + int left_change = 0, right_change = 0; + snd_azf3328_dbgcallenter(); - if (chan_sel & SET_CHAN_LEFT) + + if (chan_sel & SET_CHAN_LEFT) { curr_vol_left = inb(portbase + 1); - else - left_done = 1; - if (chan_sel & SET_CHAN_RIGHT) + + /* take care of muting flag contained in left channel */ + if (curr_vol_left & AZF_MUTE_BIT) + dst_vol_left |= AZF_MUTE_BIT; + else + dst_vol_left &= ~AZF_MUTE_BIT; + + left_change = (curr_vol_left > dst_vol_left) ? -1 : 1; + } + + if (chan_sel & SET_CHAN_RIGHT) { curr_vol_right = inb(portbase + 0); - else - right_done = 1; - - /* take care of muting flag (0x80) contained in left channel */ - if (curr_vol_left & 0x80) - dst_vol_left |= 0x80; - else - dst_vol_left &= ~0x80; + + right_change = (curr_vol_right > dst_vol_right) ? -1 : 1; + } do { - if (!left_done) { - if (curr_vol_left > dst_vol_left) - curr_vol_left--; - else - if (curr_vol_left < dst_vol_left) - curr_vol_left++; - else - left_done = 1; - outb(curr_vol_left, portbase + 1); + if (left_change) { + if (curr_vol_left != dst_vol_left) { + curr_vol_left += left_change; + outb(curr_vol_left, portbase + 1); + } else + left_change = 0; } - if (!right_done) { - if (curr_vol_right > dst_vol_right) - curr_vol_right--; - else - if (curr_vol_right < dst_vol_right) - curr_vol_right++; - else - right_done = 1; + if (right_change) { + if (curr_vol_right != dst_vol_right) { + curr_vol_right += right_change; + /* during volume change, the right channel is crackling * somewhat more than the left channel, unfortunately. * This seems to be a hardware issue. */ - outb(curr_vol_right, portbase + 0); + outb(curr_vol_right, portbase + 0); + } else + right_change = 0; } if (delay) mdelay(delay); - } while ((!left_done) || (!right_done)); + } while ((left_change) || (right_change)); snd_azf3328_dbgcallleave(); } @@ -379,7 +490,7 @@ snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg * general mixer element */ struct azf3328_mixer_reg { - unsigned int reg; + unsigned reg; unsigned int lchan_shift, rchan_shift; unsigned int mask; unsigned int invert: 1; @@ -544,13 +655,14 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, "Mix", "Mic" }; static const char * const texts3[] = { - "Mic", "CD", "Video", "Aux", + "Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone" }; static const char * const texts4[] = { "pre 3D", "post 3D" }; struct azf3328_mixer_reg reg; + const char *p = NULL; snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -561,18 +673,20 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, if (reg.reg == IDX_MIXER_ADVCTL2) { switch(reg.lchan_shift) { case 8: /* modem out sel */ - strcpy(uinfo->value.enumerated.name, texts1[uinfo->value.enumerated.item]); + p = texts1[uinfo->value.enumerated.item]; break; case 9: /* mono sel source */ - strcpy(uinfo->value.enumerated.name, texts2[uinfo->value.enumerated.item]); + p = texts2[uinfo->value.enumerated.item]; break; case 15: /* PCM Out Path */ - strcpy(uinfo->value.enumerated.name, texts4[uinfo->value.enumerated.item]); + p = texts4[uinfo->value.enumerated.item]; break; } } else - strcpy(uinfo->value.enumerated.name, texts3[uinfo->value.enumerated.item] -); + if (reg.reg == IDX_MIXER_REC_SELECT) + p = texts3[uinfo->value.enumerated.item]; + + strcpy(uinfo->value.enumerated.name, p); return 0; } @@ -583,7 +697,7 @@ snd_azf3328_get_mixer_enum(struct snd_kcontrol *kcontrol, struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol); struct azf3328_mixer_reg reg; unsigned short val; - + snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); val = snd_azf3328_mixer_inw(chip, reg.reg); if (reg.reg == IDX_MIXER_REC_SELECT) { @@ -605,7 +719,7 @@ snd_azf3328_put_mixer_enum(struct snd_kcontrol *kcontrol, struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol); struct azf3328_mixer_reg reg; unsigned int oreg, nreg, val; - + snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); oreg = snd_azf3328_mixer_inw(chip, reg.reg); val = oreg; @@ -717,15 +831,16 @@ snd_azf3328_mixer_new(struct snd_azf3328 *chip) snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000); /* mute and zero volume channels */ - for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_init_values); idx++) { + for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_init_values); ++idx) { snd_azf3328_mixer_outw(chip, snd_azf3328_init_values[idx][0], snd_azf3328_init_values[idx][1]); } - + /* add mixer controls */ sw = snd_azf3328_mixer_controls; - for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_mixer_controls); idx++, sw++) { + for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_mixer_controls); + ++idx, ++sw) { if ((err = snd_ctl_add(chip->card, snd_ctl_new1(sw, chip))) < 0) return err; } @@ -757,8 +872,8 @@ snd_azf3328_hw_free(struct snd_pcm_substream *substream) } static void -snd_azf3328_setfmt(struct snd_azf3328 *chip, - unsigned int reg, +snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, + unsigned reg, unsigned int bitrate, unsigned int format_width, unsigned int channels @@ -769,24 +884,25 @@ snd_azf3328_setfmt(struct snd_azf3328 *chip, snd_azf3328_dbgcallenter(); switch (bitrate) { - case 4000: val |= SOUNDFORMAT_FREQ_SUSPECTED_4000; break; - case 4800: val |= SOUNDFORMAT_FREQ_SUSPECTED_4800; break; - case 5512: val |= SOUNDFORMAT_FREQ_5510; break; /* the AZF3328 names it "5510" for some strange reason */ - case 6620: val |= SOUNDFORMAT_FREQ_6620; break; - case 8000: val |= SOUNDFORMAT_FREQ_8000; break; - case 9600: val |= SOUNDFORMAT_FREQ_9600; break; - case 11025: val |= SOUNDFORMAT_FREQ_11025; break; - case 13240: val |= SOUNDFORMAT_FREQ_SUSPECTED_13240; break; - case 16000: val |= SOUNDFORMAT_FREQ_16000; break; - case 22050: val |= SOUNDFORMAT_FREQ_22050; break; - case 32000: val |= SOUNDFORMAT_FREQ_32000; break; - case 44100: val |= SOUNDFORMAT_FREQ_44100; break; - case 48000: val |= SOUNDFORMAT_FREQ_48000; break; - case 66200: val |= SOUNDFORMAT_FREQ_SUSPECTED_66200; break; + case AZF_FREQ_4000: val |= SOUNDFORMAT_FREQ_SUSPECTED_4000; break; + case AZF_FREQ_4800: val |= SOUNDFORMAT_FREQ_SUSPECTED_4800; break; + case AZF_FREQ_5512: + /* the AZF3328 names it "5510" for some strange reason */ + val |= SOUNDFORMAT_FREQ_5510; break; + case AZF_FREQ_6620: val |= SOUNDFORMAT_FREQ_6620; break; + case AZF_FREQ_8000: val |= SOUNDFORMAT_FREQ_8000; break; + case AZF_FREQ_9600: val |= SOUNDFORMAT_FREQ_9600; break; + case AZF_FREQ_11025: val |= SOUNDFORMAT_FREQ_11025; break; + case AZF_FREQ_13240: val |= SOUNDFORMAT_FREQ_SUSPECTED_13240; break; + case AZF_FREQ_16000: val |= SOUNDFORMAT_FREQ_16000; break; + case AZF_FREQ_22050: val |= SOUNDFORMAT_FREQ_22050; break; + case AZF_FREQ_32000: val |= SOUNDFORMAT_FREQ_32000; break; default: snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate); - val |= SOUNDFORMAT_FREQ_44100; - break; + /* fall-through */ + case AZF_FREQ_44100: val |= SOUNDFORMAT_FREQ_44100; break; + case AZF_FREQ_48000: val |= SOUNDFORMAT_FREQ_48000; break; + case AZF_FREQ_66200: val |= SOUNDFORMAT_FREQ_SUSPECTED_66200; break; } /* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */ /* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */ @@ -805,10 +921,10 @@ snd_azf3328_setfmt(struct snd_azf3328 *chip, val |= SOUNDFORMAT_FLAG_16BIT; spin_lock_irqsave(&chip->reg_lock, flags); - + /* set bitrate/format */ snd_azf3328_codec_outw(chip, reg, val); - + /* changing the bitrate/format settings switches off the * audio output with an annoying click in case of 8/16bit format change * (maybe shutting down DAC/ADC?), thus immediately @@ -830,31 +946,81 @@ snd_azf3328_setfmt(struct snd_azf3328 *chip, snd_azf3328_dbgcallleave(); } +static inline void +snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328 *chip, + unsigned reg +) +{ + /* choose lowest frequency for low power consumption. + * While this will cause louder noise due to rather coarse frequency, + * it should never matter since output should always + * get disabled properly when idle anyway. */ + snd_azf3328_codec_setfmt(chip, reg, AZF_FREQ_4000, 8, 1); +} + +static inline void +snd_azf3328_codec_enable(struct snd_azf3328 *chip, int enable) +{ + /* no idea what exactly is being done here, but I strongly assume it's + * PM related */ + snd_azf3328_io_reg_setw( + chip->codec_io+IDX_IO_6AH, + IO_6A_PAUSE_PLAYBACK_BIT8, + !enable + ); +} + +static void +snd_azf3328_codec_activity(struct snd_azf3328 *chip, + enum snd_azf3328_stream_index stream_type, + int enable +) +{ + int need_change = (chip->audio_stream[stream_type].running != enable); + + snd_azf3328_dbgplay( + "codec_activity: type %d, enable %d, need_change %d\n", + stream_type, enable, need_change + ); + if (need_change) { + enum snd_azf3328_stream_index other = + (stream_type == AZF_PLAYBACK) ? + AZF_CAPTURE : AZF_PLAYBACK; + /* small check to prevent shutting down the other party + * in case it's active */ + if ((enable) || !(chip->audio_stream[other].running)) + snd_azf3328_codec_enable(chip, enable); + + /* ...and adjust clock, too + * (reduce noise and power consumption) */ + if (!enable) + snd_azf3328_codec_setfmt_lowpower( + chip, + chip->audio_stream[stream_type].portbase + + IDX_IO_PLAY_SOUNDFORMAT + ); + } + chip->audio_stream[stream_type].running = enable; +} + static void snd_azf3328_setdmaa(struct snd_azf3328 *chip, long unsigned int addr, unsigned int count, unsigned int size, - int do_recording) + enum snd_azf3328_stream_index stream_type +) { - unsigned long flags, portbase; - unsigned int is_running; - snd_azf3328_dbgcallenter(); - if (do_recording) { - /* access capture registers, i.e. skip playback reg section */ - portbase = chip->codec_port + 0x20; - is_running = chip->is_recording; - } else { - /* access the playback register section */ - portbase = chip->codec_port + 0x00; - is_running = chip->is_playing; - } + if (!chip->audio_stream[stream_type].running) { + /* AZF3328 uses a two buffer pointer DMA playback approach */ - /* AZF3328 uses a two buffer pointer DMA playback approach */ - if (!is_running) { - unsigned long addr_area2; - unsigned long count_areas, count_tmp; /* width 32bit -- overflow!! */ + unsigned long flags, portbase, addr_area2; + + /* width 32bit (prevent overflow): */ + unsigned long count_areas, count_tmp; + + portbase = chip->audio_stream[stream_type].portbase; count_areas = size/2; addr_area2 = addr+count_areas; count_areas--; /* max. index */ @@ -884,11 +1050,11 @@ snd_azf3328_playback_prepare(struct snd_pcm_substream *substream) snd_azf3328_dbgcallenter(); #if 0 - snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); - snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, 0); + snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_PLAYBACK); #endif snd_azf3328_dbgcallleave(); return 0; @@ -906,11 +1072,11 @@ snd_azf3328_capture_prepare(struct snd_pcm_substream *substream) snd_azf3328_dbgcallenter(); #if 0 - snd_azf3328_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); - snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, 1); + snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_CAPTURE); #endif snd_azf3328_dbgcallleave(); return 0; @@ -923,6 +1089,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_pcm_runtime *runtime = substream->runtime; int result = 0; unsigned int status1; + int previously_muted; snd_azf3328_dbgcalls("snd_azf3328_playback_trigger cmd %d\n", cmd); @@ -930,20 +1097,23 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: snd_azf3328_dbgplay("START PLAYBACK\n"); - /* mute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + /* mute WaveOut (avoid clicking during setup) */ + previously_muted = + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); - snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); spin_lock(&chip->reg_lock); - /* stop playback */ + /* first, remember current value: */ status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS); + + /* stop playback */ status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); - + /* FIXME: clear interrupts or what??? */ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_IRQTYPE, 0xffff); spin_unlock(&chip->reg_lock); @@ -951,7 +1121,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) snd_azf3328_setdmaa(chip, runtime->dma_addr, snd_pcm_lib_period_bytes(substream), snd_pcm_lib_buffer_bytes(substream), - 0); + AZF_PLAYBACK); spin_lock(&chip->reg_lock); #ifdef WIN9X @@ -978,30 +1148,35 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) DMA_SOMETHING_ELSE); #endif spin_unlock(&chip->reg_lock); + snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 1); /* now unmute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); + if (!previously_muted) + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); - chip->is_playing = 1; snd_azf3328_dbgplay("STARTED PLAYBACK\n"); break; case SNDRV_PCM_TRIGGER_RESUME: snd_azf3328_dbgplay("RESUME PLAYBACK\n"); /* resume playback if we were active */ - if (chip->is_playing) + spin_lock(&chip->reg_lock); + if (chip->audio_stream[AZF_PLAYBACK].running) snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) | DMA_RESUME); + spin_unlock(&chip->reg_lock); break; case SNDRV_PCM_TRIGGER_STOP: snd_azf3328_dbgplay("STOP PLAYBACK\n"); - /* mute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + /* mute WaveOut (avoid clicking during setup) */ + previously_muted = + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); spin_lock(&chip->reg_lock); - /* stop playback */ + /* first, remember current value: */ status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS); + /* stop playback */ status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); @@ -1013,10 +1188,12 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) status1 &= ~DMA_PLAY_SOMETHING1; snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); spin_unlock(&chip->reg_lock); - + snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0); + /* now unmute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); - chip->is_playing = 0; + if (!previously_muted) + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); + snd_azf3328_dbgplay("STOPPED PLAYBACK\n"); break; case SNDRV_PCM_TRIGGER_SUSPEND: @@ -1035,7 +1212,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) printk(KERN_ERR "FIXME: unknown trigger mode!\n"); return -EINVAL; } - + snd_azf3328_dbgcallleave(); return result; } @@ -1057,17 +1234,19 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) snd_azf3328_dbgplay("START CAPTURE\n"); - snd_azf3328_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); spin_lock(&chip->reg_lock); - /* stop recording */ + /* first, remember current value: */ status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS); + + /* stop recording */ status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); - + /* FIXME: clear interrupts or what??? */ snd_azf3328_codec_outw(chip, IDX_IO_REC_IRQTYPE, 0xffff); spin_unlock(&chip->reg_lock); @@ -1075,7 +1254,7 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) snd_azf3328_setdmaa(chip, runtime->dma_addr, snd_pcm_lib_period_bytes(substream), snd_pcm_lib_buffer_bytes(substream), - 1); + AZF_CAPTURE); spin_lock(&chip->reg_lock); #ifdef WIN9X @@ -1102,24 +1281,27 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) DMA_SOMETHING_ELSE); #endif spin_unlock(&chip->reg_lock); + snd_azf3328_codec_activity(chip, AZF_CAPTURE, 1); - chip->is_recording = 1; snd_azf3328_dbgplay("STARTED CAPTURE\n"); break; case SNDRV_PCM_TRIGGER_RESUME: snd_azf3328_dbgplay("RESUME CAPTURE\n"); /* resume recording if we were active */ - if (chip->is_recording) + spin_lock(&chip->reg_lock); + if (chip->audio_stream[AZF_CAPTURE].running) snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) | DMA_RESUME); + spin_unlock(&chip->reg_lock); break; case SNDRV_PCM_TRIGGER_STOP: snd_azf3328_dbgplay("STOP CAPTURE\n"); spin_lock(&chip->reg_lock); - /* stop recording */ + /* first, remember current value: */ status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS); + /* stop recording */ status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); @@ -1129,8 +1311,8 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) status1 &= ~DMA_PLAY_SOMETHING1; snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); spin_unlock(&chip->reg_lock); - - chip->is_recording = 0; + snd_azf3328_codec_activity(chip, AZF_CAPTURE, 0); + snd_azf3328_dbgplay("STOPPED CAPTURE\n"); break; case SNDRV_PCM_TRIGGER_SUSPEND: @@ -1149,7 +1331,7 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) printk(KERN_ERR "FIXME: unknown trigger mode!\n"); return -EINVAL; } - + snd_azf3328_dbgcallleave(); return result; } @@ -1162,11 +1344,11 @@ snd_azf3328_playback_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t frmres; #ifdef QUERY_HARDWARE - bufptr = inl(chip->codec_port+IDX_IO_PLAY_DMA_START_1); + bufptr = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_START_1); #else bufptr = substream->runtime->dma_addr; #endif - result = inl(chip->codec_port+IDX_IO_PLAY_DMA_CURRPOS); + result = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_CURRPOS); /* calculate offset */ result -= bufptr; @@ -1183,11 +1365,11 @@ snd_azf3328_capture_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t frmres; #ifdef QUERY_HARDWARE - bufptr = inl(chip->codec_port+IDX_IO_REC_DMA_START_1); + bufptr = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_START_1); #else bufptr = substream->runtime->dma_addr; #endif - result = inl(chip->codec_port+IDX_IO_REC_DMA_CURRPOS); + result = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_CURRPOS); /* calculate offset */ result -= bufptr; @@ -1196,27 +1378,233 @@ snd_azf3328_capture_pointer(struct snd_pcm_substream *substream) return frmres; } +/******************************************************************/ + +#ifdef SUPPORT_GAMEPORT +static inline void +snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip, int enable) +{ + snd_azf3328_io_reg_setb( + chip->game_io+IDX_GAME_HWCONFIG, + GAME_HWCFG_IRQ_ENABLE, + enable + ); +} + +static inline void +snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, int enable) +{ + snd_azf3328_io_reg_setb( + chip->game_io+IDX_GAME_HWCONFIG, + GAME_HWCFG_LEGACY_ADDRESS_ENABLE, + enable + ); +} + +static inline void +snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, int enable) +{ + snd_azf3328_io_reg_setw( + chip->codec_io+IDX_IO_6AH, + IO_6A_SOMETHING2_GAMEPORT, + !enable + ); +} + +static inline void +snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip) +{ + /* + * skeleton handler only + * (we do not want axis reading in interrupt handler - too much load!) + */ + snd_azf3328_dbggame("gameport irq\n"); + + /* this should ACK the gameport IRQ properly, hopefully. */ + snd_azf3328_game_inw(chip, IDX_GAME_AXIS_VALUE); +} + +static int +snd_azf3328_gameport_open(struct gameport *gameport, int mode) +{ + struct snd_azf3328 *chip = gameport_get_port_data(gameport); + int res; + + snd_azf3328_dbggame("gameport_open, mode %d\n", mode); + switch (mode) { + case GAMEPORT_MODE_COOKED: + case GAMEPORT_MODE_RAW: + res = 0; + break; + default: + res = -1; + break; + } + + snd_azf3328_gameport_axis_circuit_enable(chip, (res == 0)); + + return res; +} + +static void +snd_azf3328_gameport_close(struct gameport *gameport) +{ + struct snd_azf3328 *chip = gameport_get_port_data(gameport); + + snd_azf3328_dbggame("gameport_close\n"); + snd_azf3328_gameport_axis_circuit_enable(chip, 0); +} + +static int +snd_azf3328_gameport_cooked_read(struct gameport *gameport, + int *axes, + int *buttons +) +{ + struct snd_azf3328 *chip = gameport_get_port_data(gameport); + int i; + u8 val; + unsigned long flags; + + snd_assert(chip, return 0); + + spin_lock_irqsave(&chip->reg_lock, flags); + val = snd_azf3328_game_inb(chip, IDX_GAME_LEGACY_COMPATIBLE); + *buttons = (~(val) >> 4) & 0xf; + + /* ok, this one is a bit dirty: cooked_read is being polled by a timer, + * thus we're atomic and cannot actively wait in here + * (which would be useful for us since it probably would be better + * to trigger a measurement in here, then wait a short amount of + * time until it's finished, then read values of _this_ measurement). + * + * Thus we simply resort to reading values if they're available already + * and trigger the next measurement. + */ + + val = snd_azf3328_game_inb(chip, IDX_GAME_AXES_CONFIG); + if (val & GAME_AXES_SAMPLING_READY) { + for (i = 0; i < 4; ++i) { + /* configure the axis to read */ + val = (i << 4) | 0x0f; + snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val); + + chip->axes[i] = snd_azf3328_game_inw( + chip, IDX_GAME_AXIS_VALUE + ); + } + } + + /* trigger next axes sampling, to be evaluated the next time we + * enter this function */ + + /* for some very, very strange reason we cannot enable + * Measurement Ready monitoring for all axes here, + * at least not when only one joystick connected */ + val = 0x03; /* we're able to monitor axes 1 and 2 only */ + snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val); + + snd_azf3328_game_outw(chip, IDX_GAME_AXIS_VALUE, 0xffff); + spin_unlock_irqrestore(&chip->reg_lock, flags); + + for (i = 0; i < 4; i++) { + axes[i] = chip->axes[i]; + if (axes[i] == 0xffff) + axes[i] = -1; + } + + snd_azf3328_dbggame("cooked_read: axes %d %d %d %d buttons %d\n", + axes[0], axes[1], axes[2], axes[3], *buttons + ); + + return 0; +} + +static int __devinit +snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) +{ + struct gameport *gp; + + int io_port = chip->game_io; + + chip->gameport = gp = gameport_allocate_port(); + if (!gp) { + printk(KERN_ERR "azt3328: cannot alloc memory for gameport\n"); + return -ENOMEM; + } + + gameport_set_name(gp, "AZF3328 Gameport"); + gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci)); + gameport_set_dev_parent(gp, &chip->pci->dev); + gp->io = io_port; + gameport_set_port_data(gp, chip); + + gp->open = snd_azf3328_gameport_open; + gp->close = snd_azf3328_gameport_close; + gp->fuzz = 16; /* seems ok */ + gp->cooked_read = snd_azf3328_gameport_cooked_read; + + /* DISABLE legacy address: we don't need it! */ + snd_azf3328_gameport_legacy_address_enable(chip, 0); + + snd_azf3328_gameport_axis_circuit_enable(chip, 0); + + gameport_register_port(chip->gameport); + + return 0; +} + +static void +snd_azf3328_gameport_free(struct snd_azf3328 *chip) +{ + if (chip->gameport) { + gameport_unregister_port(chip->gameport); + chip->gameport = NULL; + } + snd_azf3328_gameport_irq_enable(chip, 0); +} +#else +static inline int +snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) { return -ENOSYS; } +static inline void +snd_azf3328_gameport_free(struct snd_azf3328 *chip) { } +static inline void +snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip) +{ + printk(KERN_WARNING "huh, game port IRQ occurred!?\n"); +} +#endif /* SUPPORT_GAMEPORT */ + +/******************************************************************/ + static irqreturn_t snd_azf3328_interrupt(int irq, void *dev_id) { struct snd_azf3328 *chip = dev_id; u8 status, which; +#if DEBUG_PLAY_REC static unsigned long irq_count; +#endif status = snd_azf3328_codec_inb(chip, IDX_IO_IRQSTATUS); /* fast path out, to ease interrupt sharing */ - if (!(status & (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_MPU401|IRQ_TIMER))) + if (!(status & + (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_GAMEPORT|IRQ_MPU401|IRQ_TIMER) + )) return IRQ_NONE; /* must be interrupt for another device */ snd_azf3328_dbgplay("Interrupt %ld!\nIDX_IO_PLAY_FLAGS %04x, IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", - irq_count, + irq_count++ /* debug-only */, snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), status); - + if (status & IRQ_TIMER) { - /* snd_azf3328_dbgplay("timer %ld\n", inl(chip->codec_port+IDX_IO_TIMER_VALUE) & TIMER_VALUE_MASK); */ + /* snd_azf3328_dbgplay("timer %ld\n", + snd_azf3328_codec_inl(chip, IDX_IO_TIMER_VALUE) + & TIMER_VALUE_MASK + ); */ if (chip->timer) snd_timer_interrupt(chip->timer, chip->timer->sticks); /* ACK timer */ @@ -1232,11 +1620,16 @@ snd_azf3328_interrupt(int irq, void *dev_id) snd_azf3328_codec_outb(chip, IDX_IO_PLAY_IRQTYPE, which); spin_unlock(&chip->reg_lock); - if (chip->pcm && chip->playback_substream) { - snd_pcm_period_elapsed(chip->playback_substream); + if (chip->pcm && chip->audio_stream[AZF_PLAYBACK].substream) { + snd_pcm_period_elapsed( + chip->audio_stream[AZF_PLAYBACK].substream + ); snd_azf3328_dbgplay("PLAY period done (#%x), @ %x\n", which, - inl(chip->codec_port+IDX_IO_PLAY_DMA_CURRPOS)); + snd_azf3328_codec_inl( + chip, IDX_IO_PLAY_DMA_CURRPOS + ) + ); } else snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); if (which & IRQ_PLAY_SOMETHING) @@ -1249,16 +1642,23 @@ snd_azf3328_interrupt(int irq, void *dev_id) snd_azf3328_codec_outb(chip, IDX_IO_REC_IRQTYPE, which); spin_unlock(&chip->reg_lock); - if (chip->pcm && chip->capture_substream) { - snd_pcm_period_elapsed(chip->capture_substream); + if (chip->pcm && chip->audio_stream[AZF_CAPTURE].substream) { + snd_pcm_period_elapsed( + chip->audio_stream[AZF_CAPTURE].substream + ); snd_azf3328_dbgplay("REC period done (#%x), @ %x\n", which, - inl(chip->codec_port+IDX_IO_REC_DMA_CURRPOS)); + snd_azf3328_codec_inl( + chip, IDX_IO_REC_DMA_CURRPOS + ) + ); } else snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); if (which & IRQ_REC_SOMETHING) snd_azf3328_dbgplay("azt3328: unknown rec IRQ type occurred, please report!\n"); } + if (status & IRQ_GAMEPORT) + snd_azf3328_gameport_interrupt(chip); /* MPU401 has less critical IRQ requirements * than timer and playback/recording, right? */ if (status & IRQ_MPU401) { @@ -1268,7 +1668,6 @@ snd_azf3328_interrupt(int irq, void *dev_id) * If so, then I don't know how... */ snd_azf3328_dbgplay("azt3328: MPU401 IRQ\n"); } - irq_count++; return IRQ_HANDLED; } @@ -1287,8 +1686,8 @@ static const struct snd_pcm_hardware snd_azf3328_playback = .rates = SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT, - .rate_min = 4000, - .rate_max = 66200, + .rate_min = AZF_FREQ_4000, + .rate_max = AZF_FREQ_66200, .channels_min = 1, .channels_max = 2, .buffer_bytes_max = 65536, @@ -1315,8 +1714,8 @@ static const struct snd_pcm_hardware snd_azf3328_capture = .rates = SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT, - .rate_min = 4000, - .rate_max = 66200, + .rate_min = AZF_FREQ_4000, + .rate_max = AZF_FREQ_66200, .channels_min = 1, .channels_max = 2, .buffer_bytes_max = 65536, @@ -1329,10 +1728,24 @@ static const struct snd_pcm_hardware snd_azf3328_capture = static unsigned int snd_azf3328_fixed_rates[] = { - 4000, 4800, 5512, 6620, 8000, 9600, 11025, 13240, 16000, 22050, 32000, - 44100, 48000, 66200 }; + AZF_FREQ_4000, + AZF_FREQ_4800, + AZF_FREQ_5512, + AZF_FREQ_6620, + AZF_FREQ_8000, + AZF_FREQ_9600, + AZF_FREQ_11025, + AZF_FREQ_13240, + AZF_FREQ_16000, + AZF_FREQ_22050, + AZF_FREQ_32000, + AZF_FREQ_44100, + AZF_FREQ_48000, + AZF_FREQ_66200 +}; + static struct snd_pcm_hw_constraint_list snd_azf3328_hw_constraints_rates = { - .count = ARRAY_SIZE(snd_azf3328_fixed_rates), + .count = ARRAY_SIZE(snd_azf3328_fixed_rates), .list = snd_azf3328_fixed_rates, .mask = 0, }; @@ -1346,7 +1759,7 @@ snd_azf3328_playback_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; snd_azf3328_dbgcallenter(); - chip->playback_substream = substream; + chip->audio_stream[AZF_PLAYBACK].substream = substream; runtime->hw = snd_azf3328_playback; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_azf3328_hw_constraints_rates); @@ -1361,7 +1774,7 @@ snd_azf3328_capture_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; snd_azf3328_dbgcallenter(); - chip->capture_substream = substream; + chip->audio_stream[AZF_CAPTURE].substream = substream; runtime->hw = snd_azf3328_capture; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_azf3328_hw_constraints_rates); @@ -1375,7 +1788,7 @@ snd_azf3328_playback_close(struct snd_pcm_substream *substream) struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); snd_azf3328_dbgcallenter(); - chip->playback_substream = NULL; + chip->audio_stream[AZF_PLAYBACK].substream = NULL; snd_azf3328_dbgcallleave(); return 0; } @@ -1386,7 +1799,7 @@ snd_azf3328_capture_close(struct snd_pcm_substream *substream) struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); snd_azf3328_dbgcallenter(); - chip->capture_substream = NULL; + chip->audio_stream[AZF_CAPTURE].substream = NULL; snd_azf3328_dbgcallleave(); return 0; } @@ -1441,102 +1854,8 @@ snd_azf3328_pcm(struct snd_azf3328 *chip, int device) /******************************************************************/ -#ifdef SUPPORT_JOYSTICK -static int __devinit -snd_azf3328_config_joystick(struct snd_azf3328 *chip, int dev) -{ - struct gameport *gp; - struct resource *r; - - if (!joystick[dev]) - return -ENODEV; - - if (!(r = request_region(0x200, 8, "AZF3328 gameport"))) { - printk(KERN_WARNING "azt3328: cannot reserve joystick ports\n"); - return -EBUSY; - } - - chip->gameport = gp = gameport_allocate_port(); - if (!gp) { - printk(KERN_ERR "azt3328: cannot allocate memory for gameport\n"); - release_and_free_resource(r); - return -ENOMEM; - } - - gameport_set_name(gp, "AZF3328 Gameport"); - gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci)); - gameport_set_dev_parent(gp, &chip->pci->dev); - gp->io = 0x200; - gameport_set_port_data(gp, r); - - snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR, - snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) | LEGACY_JOY); - - gameport_register_port(chip->gameport); - - return 0; -} - -static void -snd_azf3328_free_joystick(struct snd_azf3328 *chip) -{ - if (chip->gameport) { - struct resource *r = gameport_get_port_data(chip->gameport); - - gameport_unregister_port(chip->gameport); - chip->gameport = NULL; - /* disable gameport */ - snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR, - snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) & ~LEGACY_JOY); - release_and_free_resource(r); - } -} -#else -static inline int -snd_azf3328_config_joystick(struct snd_azf3328 *chip, int dev) { return -ENOSYS; } -static inline void -snd_azf3328_free_joystick(struct snd_azf3328 *chip) { } -#endif - -/******************************************************************/ - -static int -snd_azf3328_free(struct snd_azf3328 *chip) -{ - if (chip->irq < 0) - goto __end_hw; - - /* reset (close) mixer */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); /* first mute master volume */ - snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000); - - /* interrupt setup - mask everything (FIXME!) */ - /* well, at least we know how to disable the timer IRQ */ - snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x00); - - if (chip->irq >= 0) - synchronize_irq(chip->irq); -__end_hw: - snd_azf3328_free_joystick(chip); - if (chip->irq >= 0) - free_irq(chip->irq, chip); - pci_release_regions(chip->pci); - pci_disable_device(chip->pci); - - kfree(chip); - return 0; -} - -static int -snd_azf3328_dev_free(struct snd_device *device) -{ - struct snd_azf3328 *chip = device->device_data; - return snd_azf3328_free(chip); -} - -/******************************************************************/ - -/*** NOTE: the physical timer resolution actually is 1024000 ticks per second, +/*** NOTE: the physical timer resolution actually is 1024000 ticks per second + *** (probably derived from main crystal via a divider of 24), *** but announcing those attributes to user-space would make programs *** configure the timer to a 1 tick value, resulting in an absolutely fatal *** timer IRQ storm. @@ -1564,7 +1883,7 @@ snd_azf3328_timer_start(struct snd_timer *timer) delay = 49; /* minimum time is 49 ticks */ } snd_azf3328_dbgtimer("setting timer countdown value %d, add COUNTDOWN|IRQ\n", delay); - delay |= TIMER_ENABLE_COUNTDOWN | TIMER_ENABLE_IRQ; + delay |= TIMER_COUNTDOWN_ENABLE | TIMER_IRQ_ENABLE; spin_lock_irqsave(&chip->reg_lock, flags); snd_azf3328_codec_outl(chip, IDX_IO_TIMER_VALUE, delay); spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -1582,7 +1901,7 @@ snd_azf3328_timer_stop(struct snd_timer *timer) chip = snd_timer_chip(timer); spin_lock_irqsave(&chip->reg_lock, flags); /* disable timer countdown and interrupt */ - /* FIXME: should we write TIMER_ACK_IRQ here? */ + /* FIXME: should we write TIMER_IRQ_ACK here? */ snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_azf3328_dbgcallleave(); @@ -1626,9 +1945,10 @@ snd_azf3328_timer(struct snd_azf3328 *chip, int device) snd_azf3328_timer_hw.resolution *= seqtimer_scaling; snd_azf3328_timer_hw.ticks /= seqtimer_scaling; - if ((err = snd_timer_new(chip->card, "AZF3328", &tid, &timer)) < 0) { + + err = snd_timer_new(chip->card, "AZF3328", &tid, &timer); + if (err < 0) goto out; - } strcpy(timer->name, "AZF3328 timer"); timer->private_data = chip; @@ -1636,6 +1956,8 @@ snd_azf3328_timer(struct snd_azf3328 *chip, int device) chip->timer = timer; + snd_azf3328_timer_stop(timer); + err = 0; out: @@ -1645,10 +1967,44 @@ out: /******************************************************************/ +static int +snd_azf3328_free(struct snd_azf3328 *chip) +{ + if (chip->irq < 0) + goto __end_hw; + + /* reset (close) mixer: + * first mute master volume, then reset + */ + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); + snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000); + + snd_azf3328_timer_stop(chip->timer); + snd_azf3328_gameport_free(chip); + + if (chip->irq >= 0) + synchronize_irq(chip->irq); +__end_hw: + if (chip->irq >= 0) + free_irq(chip->irq, chip); + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + + kfree(chip); + return 0; +} + +static int +snd_azf3328_dev_free(struct snd_device *device) +{ + struct snd_azf3328 *chip = device->device_data; + return snd_azf3328_free(chip); +} + #if 0 /* check whether a bit can be modified */ static void -snd_azf3328_test_bit(unsigned int reg, int bit) +snd_azf3328_test_bit(unsigned unsigned reg, int bit) { unsigned char val, valoff, valon; @@ -1659,42 +2015,74 @@ snd_azf3328_test_bit(unsigned int reg, int bit) outb(val|(1 << bit), reg); valon = inb(reg); - + outb(val, reg); - printk(KERN_ERR "reg %04x bit %d: %02x %02x %02x\n", reg, bit, val, valoff, valon); + printk(KERN_ERR "reg %04x bit %d: %02x %02x %02x\n", + reg, bit, val, valoff, valon + ); } #endif -#if DEBUG_MISC -static void +static inline void snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip) { +#if DEBUG_MISC u16 tmp; - snd_azf3328_dbgmisc("codec_port 0x%lx, io2_port 0x%lx, mpu_port 0x%lx, synth_port 0x%lx, mixer_port 0x%lx, irq %d\n", chip->codec_port, chip->io2_port, chip->mpu_port, chip->synth_port, chip->mixer_port, chip->irq); + snd_azf3328_dbgmisc( + "codec_io 0x%lx, game_io 0x%lx, mpu_io 0x%lx, " + "opl3_io 0x%lx, mixer_io 0x%lx, irq %d\n", + chip->codec_io, chip->game_io, chip->mpu_io, + chip->opl3_io, chip->mixer_io, chip->irq + ); - snd_azf3328_dbgmisc("io2 %02x %02x %02x %02x %02x %02x\n", snd_azf3328_io2_inb(chip, 0), snd_azf3328_io2_inb(chip, 1), snd_azf3328_io2_inb(chip, 2), snd_azf3328_io2_inb(chip, 3), snd_azf3328_io2_inb(chip, 4), snd_azf3328_io2_inb(chip, 5)); + snd_azf3328_dbgmisc("game %02x %02x %02x %02x %02x %02x\n", + snd_azf3328_game_inb(chip, 0), + snd_azf3328_game_inb(chip, 1), + snd_azf3328_game_inb(chip, 2), + snd_azf3328_game_inb(chip, 3), + snd_azf3328_game_inb(chip, 4), + snd_azf3328_game_inb(chip, 5) + ); - for (tmp=0; tmp <= 0x01; tmp += 1) - snd_azf3328_dbgmisc("0x%02x: opl 0x%04x, mpu300 0x%04x, mpu310 0x%04x, mpu320 0x%04x, mpu330 0x%04x\n", tmp, inb(0x388 + tmp), inb(0x300 + tmp), inb(0x310 + tmp), inb(0x320 + tmp), inb(0x330 + tmp)); + for (tmp = 0; tmp < 0x07; tmp += 1) + snd_azf3328_dbgmisc("mpu_io 0x%04x\n", inb(chip->mpu_io + tmp)); + + for (tmp = 0; tmp <= 0x07; tmp += 1) + snd_azf3328_dbgmisc("0x%02x: game200 0x%04x, game208 0x%04x\n", + tmp, inb(0x200 + tmp), inb(0x208 + tmp)); + + for (tmp = 0; tmp <= 0x01; tmp += 1) + snd_azf3328_dbgmisc( + "0x%02x: mpu300 0x%04x, mpu310 0x%04x, mpu320 0x%04x, " + "mpu330 0x%04x opl388 0x%04x opl38c 0x%04x\n", + tmp, + inb(0x300 + tmp), + inb(0x310 + tmp), + inb(0x320 + tmp), + inb(0x330 + tmp), + inb(0x388 + tmp), + inb(0x38c + tmp) + ); for (tmp = 0; tmp < AZF_IO_SIZE_CODEC; tmp += 2) - snd_azf3328_dbgmisc("codec 0x%02x: 0x%04x\n", tmp, snd_azf3328_codec_inw(chip, tmp)); + snd_azf3328_dbgmisc("codec 0x%02x: 0x%04x\n", + tmp, snd_azf3328_codec_inw(chip, tmp) + ); for (tmp = 0; tmp < AZF_IO_SIZE_MIXER; tmp += 2) - snd_azf3328_dbgmisc("mixer 0x%02x: 0x%04x\n", tmp, snd_azf3328_mixer_inw(chip, tmp)); + snd_azf3328_dbgmisc("mixer 0x%02x: 0x%04x\n", + tmp, snd_azf3328_mixer_inw(chip, tmp) + ); +#endif /* DEBUG_MISC */ } -#else -static inline void -snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip) {} -#endif static int __devinit snd_azf3328_create(struct snd_card *card, - struct pci_dev *pci, - unsigned long device_type, - struct snd_azf3328 ** rchip) + struct pci_dev *pci, + unsigned long device_type, + struct snd_azf3328 **rchip) { struct snd_azf3328 *chip; int err; @@ -1705,7 +2093,8 @@ snd_azf3328_create(struct snd_card *card, *rchip = NULL; - if ((err = pci_enable_device(pci)) < 0) + err = pci_enable_device(pci); + if (err < 0) return err; chip = kzalloc(sizeof(*chip), GFP_KERNEL); @@ -1721,20 +2110,25 @@ snd_azf3328_create(struct snd_card *card, /* check if we can restrict PCI DMA transfers to 24 bits */ if (pci_set_dma_mask(pci, DMA_24BIT_MASK) < 0 || pci_set_consistent_dma_mask(pci, DMA_24BIT_MASK) < 0) { - snd_printk(KERN_ERR "architecture does not support 24bit PCI busmaster DMA\n"); + snd_printk(KERN_ERR "architecture does not support " + "24bit PCI busmaster DMA\n" + ); err = -ENXIO; goto out_err; } - if ((err = pci_request_regions(pci, "Aztech AZF3328")) < 0) { + err = pci_request_regions(pci, "Aztech AZF3328"); + if (err < 0) goto out_err; - } - chip->codec_port = pci_resource_start(pci, 0); - chip->io2_port = pci_resource_start(pci, 1); - chip->mpu_port = pci_resource_start(pci, 2); - chip->synth_port = pci_resource_start(pci, 3); - chip->mixer_port = pci_resource_start(pci, 4); + chip->codec_io = pci_resource_start(pci, 0); + chip->game_io = pci_resource_start(pci, 1); + chip->mpu_io = pci_resource_start(pci, 2); + chip->opl3_io = pci_resource_start(pci, 3); + chip->mixer_io = pci_resource_start(pci, 4); + + chip->audio_stream[AZF_PLAYBACK].portbase = chip->codec_io + 0x00; + chip->audio_stream[AZF_CAPTURE].portbase = chip->codec_io + 0x20; if (request_irq(pci->irq, snd_azf3328_interrupt, IRQF_SHARED, card->shortname, chip)) { @@ -1747,29 +2141,29 @@ snd_azf3328_create(struct snd_card *card, synchronize_irq(chip->irq); snd_azf3328_debug_show_ports(chip); - - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) goto out_err; - } /* create mixer interface & switches */ - if ((err = snd_azf3328_mixer_new(chip)) < 0) + err = snd_azf3328_mixer_new(chip); + if (err < 0) goto out_err; -#if 0 - /* set very low bitrate to reduce noise and power consumption? */ - snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, 5512, 8, 1); -#endif + /* shutdown codecs to save power */ + /* have snd_azf3328_codec_activity() act properly */ + chip->audio_stream[AZF_PLAYBACK].running = 1; + snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0); /* standard chip init stuff */ - /* default IRQ init value */ + /* default IRQ init value */ tmp = DMA_PLAY_SOMETHING2|DMA_EPILOGUE_SOMETHING|DMA_SOMETHING_ELSE; spin_lock_irq(&chip->reg_lock); snd_azf3328_codec_outb(chip, IDX_IO_PLAY_FLAGS, tmp); snd_azf3328_codec_outb(chip, IDX_IO_REC_FLAGS, tmp); snd_azf3328_codec_outb(chip, IDX_IO_SOMETHING_FLAGS, tmp); - snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x00); /* disable timer */ spin_unlock_irq(&chip->reg_lock); snd_card_set_dev(card, &pci->dev); @@ -1805,52 +2199,61 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0 ); + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); if (card == NULL) return -ENOMEM; strcpy(card->driver, "AZF3328"); strcpy(card->shortname, "Aztech AZF3328 (PCI168)"); - if ((err = snd_azf3328_create(card, pci, pci_id->driver_data, &chip)) < 0) { + err = snd_azf3328_create(card, pci, pci_id->driver_data, &chip); + if (err < 0) goto out_err; - } card->private_data = chip; - if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_MPU401, - chip->mpu_port, MPU401_INFO_INTEGRATED, - pci->irq, 0, &chip->rmidi)) < 0) { - snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", chip->mpu_port); + err = snd_mpu401_uart_new( + card, 0, MPU401_HW_MPU401, chip->mpu_io, MPU401_INFO_INTEGRATED, + pci->irq, 0, &chip->rmidi + ); + if (err < 0) { + snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", + chip->mpu_io + ); goto out_err; } - if ((err = snd_azf3328_timer(chip, 0)) < 0) { + err = snd_azf3328_timer(chip, 0); + if (err < 0) goto out_err; - } - if ((err = snd_azf3328_pcm(chip, 0)) < 0) { + err = snd_azf3328_pcm(chip, 0); + if (err < 0) goto out_err; - } - if (snd_opl3_create(card, chip->synth_port, chip->synth_port+2, + if (snd_opl3_create(card, chip->opl3_io, chip->opl3_io+2, OPL3_HW_AUTO, 1, &opl3) < 0) { snd_printk(KERN_ERR "azf3328: no OPL3 device at 0x%lx-0x%lx?\n", - chip->synth_port, chip->synth_port+2 ); + chip->opl3_io, chip->opl3_io+2 + ); } else { - if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) { + /* need to use IDs 1, 2 since ID 0 is snd_azf3328_timer above */ + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + goto out_err; + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) goto out_err; - } } opl3->private_data = chip; sprintf(card->longname, "%s at 0x%lx, irq %i", - card->shortname, chip->codec_port, chip->irq); + card->shortname, chip->codec_io, chip->irq); - if ((err = snd_card_register(card)) < 0) { + err = snd_card_register(card); + if (err < 0) goto out_err; - } #ifdef MODULE printk( @@ -1861,19 +2264,18 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) 1024000 / seqtimer_scaling, seqtimer_scaling); #endif - if (snd_azf3328_config_joystick(chip, dev) < 0) - snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR, - snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) & ~LEGACY_JOY); + snd_azf3328_gameport(chip, dev); pci_set_drvdata(pci, card); dev++; err = 0; goto out; - + out_err: + snd_printk(KERN_ERR "azf3328: something failed, exiting\n"); snd_card_free(card); - + out: snd_azf3328_dbgcallleave(); return err; @@ -1894,27 +2296,27 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); struct snd_azf3328 *chip = card->private_data; - int reg; + unsigned reg; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - + snd_pcm_suspend_all(chip->pcm); - for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++) - chip->saved_regs_mixer[reg] = inw(chip->mixer_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg) + chip->saved_regs_mixer[reg] = inw(chip->mixer_io + reg * 2); /* make sure to disable master volume etc. to prevent looping sound */ snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); - - for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++) - chip->saved_regs_codec[reg] = inw(chip->codec_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++) - chip->saved_regs_io2[reg] = inw(chip->io2_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++) - chip->saved_regs_mpu[reg] = inw(chip->mpu_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++) - chip->saved_regs_synth[reg] = inw(chip->synth_port + reg * 2); + + for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) + chip->saved_regs_codec[reg] = inw(chip->codec_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) + chip->saved_regs_game[reg] = inw(chip->game_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg) + chip->saved_regs_mpu[reg] = inw(chip->mpu_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg) + chip->saved_regs_opl3[reg] = inw(chip->opl3_io + reg * 2); pci_disable_device(pci); pci_save_state(pci); @@ -1927,7 +2329,7 @@ snd_azf3328_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct snd_azf3328 *chip = card->private_data; - int reg; + unsigned reg; pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); @@ -1939,23 +2341,21 @@ snd_azf3328_resume(struct pci_dev *pci) } pci_set_master(pci); - for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++) - outw(chip->saved_regs_io2[reg], chip->io2_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++) - outw(chip->saved_regs_mpu[reg], chip->mpu_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++) - outw(chip->saved_regs_synth[reg], chip->synth_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++) - outw(chip->saved_regs_mixer[reg], chip->mixer_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++) - outw(chip->saved_regs_codec[reg], chip->codec_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) + outw(chip->saved_regs_game[reg], chip->game_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg) + outw(chip->saved_regs_mpu[reg], chip->mpu_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg) + outw(chip->saved_regs_opl3[reg], chip->opl3_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg) + outw(chip->saved_regs_mixer[reg], chip->mixer_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) + outw(chip->saved_regs_codec[reg], chip->codec_io + reg * 2); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif - - +#endif /* CONFIG_PM */ static struct pci_driver driver = { diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h index 679fa992e2bc..3448fd626f80 100644 --- a/sound/pci/azt3328.h +++ b/sound/pci/azt3328.h @@ -54,7 +54,10 @@ #define SOUNDFORMAT_XTAL1 0x00 #define SOUNDFORMAT_XTAL2 0x01 /* all _SUSPECTED_ values are not used by Windows drivers, so we don't - * have any hard facts, only rough measurements */ + * have any hard facts, only rough measurements. + * All we know is that the crystal used on the board has 24.576MHz, + * like many soundcards (which results in the frequencies below when + * using certain divider values selected by the values below) */ #define SOUNDFORMAT_FREQ_SUSPECTED_4000 0x0c | SOUNDFORMAT_XTAL1 #define SOUNDFORMAT_FREQ_SUSPECTED_4800 0x0a | SOUNDFORMAT_XTAL1 #define SOUNDFORMAT_FREQ_5510 0x0c | SOUNDFORMAT_XTAL2 @@ -72,6 +75,26 @@ #define SOUNDFORMAT_FLAG_16BIT 0x0010 #define SOUNDFORMAT_FLAG_2CHANNELS 0x0020 +/* define frequency helpers, for maximum value safety */ +enum { +#define AZF_FREQ(rate) AZF_FREQ_##rate = rate + AZF_FREQ(4000), + AZF_FREQ(4800), + AZF_FREQ(5512), + AZF_FREQ(6620), + AZF_FREQ(8000), + AZF_FREQ(9600), + AZF_FREQ(11025), + AZF_FREQ(13240), + AZF_FREQ(16000), + AZF_FREQ(22050), + AZF_FREQ(32000), + AZF_FREQ(44100), + AZF_FREQ(48000), + AZF_FREQ(66200), +#undef AZF_FREQ +} AZF_FREQUENCIES; + /** recording area (see also: playback bit flag definitions) **/ #define IDX_IO_REC_FLAGS 0x20 /* ??, PU:0x0000 */ #define IDX_IO_REC_IRQTYPE 0x22 /* ??, PU:0x0000 */ @@ -97,40 +120,164 @@ /** DirectX timer, main interrupt area (FIXME: and something else?) **/ #define IDX_IO_TIMER_VALUE 0x60 /* found this timer area by pure luck :-) */ - #define TIMER_VALUE_MASK 0x000fffffUL /* timer countdown value; triggers IRQ when timer is finished */ - #define TIMER_ENABLE_COUNTDOWN 0x01000000UL /* activate the timer countdown */ - #define TIMER_ENABLE_IRQ 0x02000000UL /* trigger timer IRQ on zero transition */ - #define TIMER_ACK_IRQ 0x04000000UL /* being set in IRQ handler in case port 0x00 (hmm, not port 0x64!?!?) had 0x0020 set upon IRQ handler */ + /* timer countdown value; triggers IRQ when timer is finished */ + #define TIMER_VALUE_MASK 0x000fffffUL + /* activate timer countdown */ + #define TIMER_COUNTDOWN_ENABLE 0x01000000UL + /* trigger timer IRQ on zero transition */ + #define TIMER_IRQ_ENABLE 0x02000000UL + /* being set in IRQ handler in case port 0x00 (hmm, not port 0x64!?!?) + * had 0x0020 set upon IRQ handler */ + #define TIMER_IRQ_ACK 0x04000000UL #define IDX_IO_IRQSTATUS 0x64 - #define IRQ_PLAYBACK 0x0001 - #define IRQ_RECORDING 0x0002 - #define IRQ_MPU401 0x0010 - #define IRQ_TIMER 0x0020 /* DirectX timer */ - #define IRQ_UNKNOWN1 0x0040 /* probably unused, or possibly I2S port? or gameport IRQ? */ - #define IRQ_UNKNOWN2 0x0080 /* probably unused, or possibly I2S port? or gameport IRQ? */ + /* some IRQ bit in here might also be used to signal a power-management timer + * timeout, to request shutdown of the chip (e.g. AD1815JS has such a thing). + * Some OPL3 hardware (e.g. in LM4560) has some special timer hardware which + * can trigger an OPL3 timer IRQ, so maybe there's such a thing as well... */ + + #define IRQ_PLAYBACK 0x0001 + #define IRQ_RECORDING 0x0002 + #define IRQ_UNKNOWN1 0x0004 /* most probably I2S port */ + #define IRQ_GAMEPORT 0x0008 /* Interrupt of Digital(ly) Enhanced Game Port */ + #define IRQ_MPU401 0x0010 + #define IRQ_TIMER 0x0020 /* DirectX timer */ + #define IRQ_UNKNOWN2 0x0040 /* probably unused, or possibly I2S port? */ + #define IRQ_UNKNOWN3 0x0080 /* probably unused, or possibly I2S port? */ #define IDX_IO_66H 0x66 /* writing 0xffff returns 0x0000 */ -#define IDX_IO_SOME_VALUE 0x68 /* this is set to e.g. 0x3ff or 0x300, and writable; maybe some buffer limit, but I couldn't find out more, PU:0x00ff */ -#define IDX_IO_6AH 0x6A /* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); actually inhibits PCM playback!!! maybe power management?? */ - #define IO_6A_PAUSE_PLAYBACK 0x0200 /* bit 9; sure, this pauses playback, but what the heck is this really about?? */ -#define IDX_IO_6CH 0x6C -#define IDX_IO_6EH 0x6E /* writing 0xffff returns 0x83fe */ -/* further I/O indices not saved/restored, so probably not used */ + /* this is set to e.g. 0x3ff or 0x300, and writable; + * maybe some buffer limit, but I couldn't find out more, PU:0x00ff: */ +#define IDX_IO_SOME_VALUE 0x68 + #define IO_68_RANDOM_TOGGLE1 0x0100 /* toggles randomly */ + #define IO_68_RANDOM_TOGGLE2 0x0200 /* toggles randomly */ + /* umm, nope, behaviour of these bits changes depending on what we wrote + * to 0x6b!! */ + +/* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); + * actually inhibits PCM playback!!! maybe power management??: */ +#define IDX_IO_6AH 0x6A + /* bit 5: enabling this will activate permanent counting of bytes 2/3 + * at gameport I/O (0xb402/3) (equal values each) and cause + * gameport legacy I/O at 0x0200 to be _DISABLED_! + * Is this Digital Enhanced Game Port Enable??? Or maybe it's Testmode + * for Enhanced Digital Gameport (see 4D Wave DX card): */ + #define IO_6A_SOMETHING1_GAMEPORT 0x0020 + /* bit 8; sure, this _pauses_ playback (later resumes at same spot!), + * but what the heck is this really about??: */ + #define IO_6A_PAUSE_PLAYBACK_BIT8 0x0100 + /* bit 9; sure, this _pauses_ playback (later resumes at same spot!), + * but what the heck is this really about??: */ + #define IO_6A_PAUSE_PLAYBACK_BIT9 0x0200 + /* BIT8 and BIT9 are _NOT_ able to affect OPL3 MIDI playback, + * thus it suggests influence on PCM only!! + * However OTOH there seems to be no bit anywhere around here + * which is able to disable OPL3... */ + /* bit 10: enabling this actually changes values at legacy gameport + * I/O address (0x200); is this enabling of the Digital Enhanced Game Port??? + * Or maybe this simply switches off the NE558 circuit, since enabling this + * still lets us evaluate button states, but not axis states */ + #define IO_6A_SOMETHING2_GAMEPORT 0x0400 + /* writing 0x0300: causes quite some crackling during + * PC activity such as switching windows (PCI traffic?? + * --> FIFO/timing settings???) */ + /* writing 0x0100 plus/or 0x0200 inhibits playback */ + /* since the Windows .INF file has Flag_Enable_JoyStick and + * Flag_Enable_SB_DOS_Emulation directly together, it stands to reason + * that some other bit in this same register might be responsible + * for SB DOS Emulation activation (note that the file did NOT define + * a switch for OPL3!) */ +#define IDX_IO_6CH 0x6C /* unknown; fully read-writable */ +#define IDX_IO_6EH 0x6E + /* writing 0xffff returns 0x83fe (or 0x03fe only). + * writing 0x83 (and only 0x83!!) to 0x6f will cause 0x6c to switch + * from 0000 to ffff. */ + +/* further I/O indices not saved/restored and not readable after writing, + * so probably not used */ -/*** I/O 2 area port indices ***/ +/*** Gameport area port indices ***/ /* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */ -#define AZF_IO_SIZE_IO2 0x08 -#define AZF_IO_SIZE_IO2_PM 0x06 +#define AZF_IO_SIZE_GAME 0x08 +#define AZF_IO_SIZE_GAME_PM 0x06 -#define IDX_IO2_LEGACY_ADDR 0x04 - #define LEGACY_SOMETHING 0x01 /* OPL3?? */ - #define LEGACY_JOY 0x08 +enum { + AZF_GAME_LEGACY_IO_PORT = 0x200 +} AZF_GAME_CONFIGS; +#define IDX_GAME_LEGACY_COMPATIBLE 0x00 + /* in some operation mode, writing anything to this port + * triggers an interrupt: + * yup, that's in case IDX_GAME_01H has one of the + * axis measurement bits enabled + * (and of course one needs to have GAME_HWCFG_IRQ_ENABLE, too) */ + +#define IDX_GAME_AXES_CONFIG 0x01 + /* NOTE: layout of this register awfully similar (read: "identical??") + * to AD1815JS.pdf (p.29) */ + + /* enables axis 1 (X axis) measurement: */ + #define GAME_AXES_ENABLE_1 0x01 + /* enables axis 2 (Y axis) measurement: */ + #define GAME_AXES_ENABLE_2 0x02 + /* enables axis 3 (X axis) measurement: */ + #define GAME_AXES_ENABLE_3 0x04 + /* enables axis 4 (Y axis) measurement: */ + #define GAME_AXES_ENABLE_4 0x08 + /* selects the current axis to read the measured value of + * (at IDX_GAME_AXIS_VALUE): + * 00 = axis 1, 01 = axis 2, 10 = axis 3, 11 = axis 4: */ + #define GAME_AXES_READ_MASK 0x30 + /* enable to have the latch continuously accept ADC values + * (and continuously cause interrupts in case interrupts are enabled); + * AD1815JS.pdf says it's ~16ms interval there: */ + #define GAME_AXES_LATCH_ENABLE 0x40 + /* joystick data (measured axes) ready for reading: */ + #define GAME_AXES_SAMPLING_READY 0x80 + + /* NOTE: other card specs (SiS960 and others!) state that the + * game position latches should be frozen when reading and be freed + * (== reset?) after reading!!! + * Freezing most likely means disabling 0x40 (GAME_AXES_LATCH_ENABLE), + * but how to free the value? */ + /* An internet search for "gameport latch ADC" should provide some insight + * into how to program such a gameport system. */ + + /* writing 0xf0 to 01H once reset both counters to 0, in some special mode!? + * yup, in case 6AH 0x20 is not enabled + * (and 0x40 is sufficient, 0xf0 is not needed) */ + +#define IDX_GAME_AXIS_VALUE 0x02 + /* R: value of currently configured axis (word value!); + * W: trigger axis measurement */ + +#define IDX_GAME_HWCONFIG 0x04 + /* note: bits 4 to 7 are never set (== 0) when reading! + * --> reserved bits? */ + /* enables IRQ notification upon axes measurement ready: */ + #define GAME_HWCFG_IRQ_ENABLE 0x01 + /* these bits choose a different frequency for the + * internal ADC counter increment. + * hmm, seems to be a combo of bits: + * 00 --> standard frequency + * 10 --> 1/2 + * 01 --> 1/20 + * 11 --> 1/200: */ + #define GAME_HWCFG_ADC_COUNTER_FREQ_MASK 0x06 + + /* enable gameport legacy I/O address (0x200) + * I was unable to locate any configurability for a different address: */ + #define GAME_HWCFG_LEGACY_ADDRESS_ENABLE 0x08 + +/*** MPU401 ***/ #define AZF_IO_SIZE_MPU 0x04 #define AZF_IO_SIZE_MPU_PM 0x04 -#define AZF_IO_SIZE_SYNTH 0x08 -#define AZF_IO_SIZE_SYNTH_PM 0x06 +/*** OPL3 synth ***/ +#define AZF_IO_SIZE_OPL3 0x08 +#define AZF_IO_SIZE_OPL3_PM 0x06 +/* hmm, given that a standard OPL3 has 4 registers only, + * there might be some enhanced functionality lurking at the end + * (especially since register 0x04 has a "non-empty" value 0xfe) */ /*** mixer I/O area port indices ***/ /* (only 0x22 of 0x40 bytes saved/restored by Windows driver) From 9ad593f6d326e7a4664e3856520f6c042f82a37f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 16 May 2008 12:34:47 +0200 Subject: [PATCH 029/179] [ALSA] hda - Fix DMA position inaccuracy Many HD-audio controllers seem inaccurate about the IRQ timing of PCM period updates. This has caused problems on audio quality; e.g. JACK doesn't work with two periods. This patch fixes the problem by checking the current DMA position at IRQ handler and delays the period-update via a workq if it's inaccurate. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 124 ++++++++++++++++++++++++++++++++------ 1 file changed, 106 insertions(+), 18 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b3a618eb42cd..6ba7ac01d9f6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -285,6 +285,7 @@ struct azx_dev { u32 *posbuf; /* position buffer pointer */ unsigned int bufsize; /* size of the play buffer in bytes */ + unsigned int period_bytes; /* size of the period in bytes */ unsigned int frags; /* number for period in the play buffer */ unsigned int fifo_size; /* FIFO size */ @@ -301,11 +302,10 @@ struct azx_dev { */ unsigned char stream_tag; /* assigned stream */ unsigned char index; /* stream index */ - /* for sanity check of position buffer */ - unsigned int period_intr; unsigned int opened :1; unsigned int running :1; + unsigned int irq_pending: 1; }; /* CORB/RIRB */ @@ -369,6 +369,9 @@ struct azx { /* for debugging */ unsigned int last_cmd; /* last issued command (to sync) */ + + /* for pending irqs */ + struct work_struct irq_pending_work; }; /* driver types */ @@ -908,6 +911,8 @@ static void azx_init_pci(struct azx *chip) } +static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev); + /* * interrupt handler */ @@ -930,11 +935,18 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) azx_dev = &chip->azx_dev[i]; if (status & azx_dev->sd_int_sta_mask) { azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); - if (azx_dev->substream && azx_dev->running) { - azx_dev->period_intr++; + if (!azx_dev->substream || !azx_dev->running) + continue; + /* check whether this IRQ is really acceptable */ + if (azx_position_ok(chip, azx_dev)) { + azx_dev->irq_pending = 0; spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(azx_dev->substream); spin_lock(&chip->reg_lock); + } else { + /* bogus IRQ, process it later */ + azx_dev->irq_pending = 1; + schedule_work(&chip->irq_pending_work); } } } @@ -973,6 +985,7 @@ static int azx_setup_periods(struct snd_pcm_substream *substream, azx_sd_writel(azx_dev, SD_BDLPU, 0); period_bytes = snd_pcm_lib_period_bytes(substream); + azx_dev->period_bytes = period_bytes; periods = azx_dev->bufsize / period_bytes; /* program the initial BDL entries */ @@ -1421,27 +1434,16 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } -static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) +static unsigned int azx_get_position(struct azx *chip, + struct azx_dev *azx_dev) { - struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct azx *chip = apcm->chip; - struct azx_dev *azx_dev = get_azx_dev(substream); unsigned int pos; if (chip->position_fix == POS_FIX_POSBUF || chip->position_fix == POS_FIX_AUTO) { /* use the position buffer */ pos = le32_to_cpu(*azx_dev->posbuf); - if (chip->position_fix == POS_FIX_AUTO && - azx_dev->period_intr == 1 && !pos) { - printk(KERN_WARNING - "hda-intel: Invalid position buffer, " - "using LPIB read method instead.\n"); - chip->position_fix = POS_FIX_NONE; - goto read_lpib; - } } else { - read_lpib: /* read LPIB */ pos = azx_sd_readl(azx_dev, SD_LPIB); if (chip->position_fix == POS_FIX_FIFO) @@ -1449,7 +1451,90 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) } if (pos >= azx_dev->bufsize) pos = 0; - return bytes_to_frames(substream->runtime, pos); + return pos; +} + +static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + struct azx *chip = apcm->chip; + struct azx_dev *azx_dev = get_azx_dev(substream); + return bytes_to_frames(substream->runtime, + azx_get_position(chip, azx_dev)); +} + +/* + * Check whether the current DMA position is acceptable for updating + * periods. Returns non-zero if it's OK. + * + * Many HD-audio controllers appear pretty inaccurate about + * the update-IRQ timing. The IRQ is issued before actually the + * data is processed. So, we need to process it afterwords in a + * workqueue. + */ +static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) +{ + unsigned int pos; + + pos = azx_get_position(chip, azx_dev); + if (chip->position_fix == POS_FIX_AUTO) { + if (!pos) { + printk(KERN_WARNING + "hda-intel: Invalid position buffer, " + "using LPIB read method instead.\n"); + chip->position_fix = POS_FIX_NONE; + pos = azx_get_position(chip, azx_dev); + } else + chip->position_fix = POS_FIX_POSBUF; + } + + if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) + return 0; /* NG - it's below the period boundary */ + return 1; /* OK, it's fine */ +} + +/* + * The work for pending PCM period updates. + */ +static void azx_irq_pending_work(struct work_struct *work) +{ + struct azx *chip = container_of(work, struct azx, irq_pending_work); + int i, pending; + + for (;;) { + pending = 0; + spin_lock_irq(&chip->reg_lock); + for (i = 0; i < chip->num_streams; i++) { + struct azx_dev *azx_dev = &chip->azx_dev[i]; + if (!azx_dev->irq_pending || + !azx_dev->substream || + !azx_dev->running) + continue; + if (azx_position_ok(chip, azx_dev)) { + azx_dev->irq_pending = 0; + spin_unlock(&chip->reg_lock); + snd_pcm_period_elapsed(azx_dev->substream); + spin_lock(&chip->reg_lock); + } else + pending++; + } + spin_unlock_irq(&chip->reg_lock); + if (!pending) + return; + cond_resched(); + } +} + +/* clear irq_pending flags and assure no on-going workq */ +static void azx_clear_irq_pending(struct azx *chip) +{ + int i; + + spin_lock_irq(&chip->reg_lock); + for (i = 0; i < chip->num_streams; i++) + chip->azx_dev[i].irq_pending = 0; + spin_unlock_irq(&chip->reg_lock); + flush_scheduled_work(); } static struct snd_pcm_ops azx_pcm_ops = { @@ -1676,6 +1761,7 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) int i; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + azx_clear_irq_pending(chip); for (i = 0; i < AZX_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); if (chip->initialized) @@ -1732,6 +1818,7 @@ static int azx_free(struct azx *chip) int i; if (chip->initialized) { + azx_clear_irq_pending(chip); for (i = 0; i < chip->num_streams; i++) azx_stream_stop(chip, &chip->azx_dev[i]); azx_stop_chip(chip); @@ -1857,6 +1944,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->irq = -1; chip->driver_type = driver_type; chip->msi = enable_msi; + INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work); chip->position_fix = check_position_fix(chip, position_fix[dev]); check_probe_mask(chip, dev); From b2efbbfba24efc8456d68d5af42d082ab1c2febc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 May 2008 12:30:58 +0200 Subject: [PATCH 030/179] [ALSA] ASoC: Remove in-code changelogs The overwhelming majority just say 'initial version' anyway. Signed-off-by: Mark Brown Acked-by: Ben Dooks Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/ac97.c | 3 --- sound/soc/codecs/wm9712.c | 3 --- sound/soc/codecs/wm9713.c | 3 --- sound/soc/pxa/corgi.c | 4 ---- sound/soc/pxa/pxa2xx-i2s.c | 3 --- sound/soc/pxa/spitz.c | 3 --- sound/soc/pxa/tosa.c | 3 --- sound/soc/s3c24xx/neo1973_wm8753.c | 4 ---- sound/soc/s3c24xx/s3c2443-ac97.c | 3 --- sound/soc/s3c24xx/s3c24xx-i2s.c | 5 ----- sound/soc/s3c24xx/s3c24xx-pcm.c | 4 ---- sound/soc/s3c24xx/smdk2443_wm9710.c | 3 --- sound/soc/soc-core.c | 4 ---- sound/soc/soc-dapm.c | 5 ----- 14 files changed, 50 deletions(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 2a1ffe396908..7bf2081b46fb 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -10,9 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 17th Oct 2005 Initial version. - * * Generic AC97 support. */ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 76c1e2d33e7d..89efe40c7c33 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -9,9 +9,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * Revision history - * 4th Feb 2006 Initial version. */ #include diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 1f241161445c..9e6b2fd7262b 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -10,9 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 4th Feb 2006 Initial version. - * * Features:- * * o Support for AC97 Codec, Voice DAC and Aux DAC diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 852f16d963a1..edeea63e80e8 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -11,10 +11,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * Revision history - * 30th Nov 2005 Initial version. - * */ #include diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 425071030970..35090c2870ff 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -9,9 +9,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * Revision history - * 12th Aug 2005 Initial version. */ #include diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 73a6df8709e1..092b5c776b40 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -12,9 +12,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 30th Nov 2005 Initial version. - * */ #include diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index fda2aa056d53..465ff0f458ef 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -12,9 +12,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 30th Nov 2005 Initial version. - * * GPIO's * 1 - Jack Insertion * 5 - Hookswitch (headset answer/hang up switch) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 79c5027273cb..c1a0161bc72e 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -10,10 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 20th Jan 2007 Initial version. - * 05th Feb 2007 Rename all to Neo1973 - * */ #include diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index e81d9a6c83da..0eed140dcd9b 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -10,9 +10,6 @@ * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. - * - * Revision history - * 21st Mar 2007 Initial Version */ #include diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 1ed6afd45459..ddf87246c77b 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -12,11 +12,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * - * Revision history - * 11th Dec 2006 Merged with Simtec driver - * 10th Nov 2006 Initial version. */ #include diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 7806ae614617..ef599745159c 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -12,10 +12,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * Revision history - * 11th Dec 2006 Merged with Simtec driver - * 10th Nov 2006 Initial version. */ #include diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index b4a56302b9ab..8515d6ff03f2 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -10,9 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 8th Mar 2007 Initial version. - * */ #include diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e148db940cfc..0318d8abe3e8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -14,10 +14,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 12th Aug 2005 Initial version. - * 25th Oct 2005 Working Codec, Interface and Platform registration. - * * TODO: * o Add hw rules to enforce rates, etc. * o More testing with other codecs/machines. diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1ef6d94b8357..8a3192bcee78 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -10,11 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 12th Aug 2005 Initial version. - * 25th Oct 2005 Implemented path power domain. - * 18th Dec 2005 Implemented machine and stream level power domain. - * * Features: * o Changes power status of internal codec blocks depending on the * dynamic configuration of codec internal audio paths and active From 1ef6ab75c7deef931d6308af282ed7d8e480e77f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 May 2008 12:31:55 +0200 Subject: [PATCH 031/179] [ALSA] ASoC: Make CPU and codec DAI operations have same type The CPU and codec DAI operations differ only in the presence of the digital mute operation for the codec so they may as well be the same type. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 30 ++++++------------------------ 1 file changed, 6 insertions(+), 24 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d3c8c033dff8..73accbcfbd2d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -272,9 +272,9 @@ struct snd_soc_ops { int (*trigger)(struct snd_pcm_substream *, int); }; -/* ASoC codec DAI ops */ -struct snd_soc_codec_ops { - /* codec DAI clocking configuration */ +/* ASoC DAI ops */ +struct snd_soc_dai_ops { + /* DAI clocking configuration */ int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai, int clk_id, unsigned int freq, int dir); int (*set_pll)(struct snd_soc_codec_dai *codec_dai, @@ -282,7 +282,7 @@ struct snd_soc_codec_ops { int (*set_clkdiv)(struct snd_soc_codec_dai *codec_dai, int div_id, int div); - /* CPU DAI format configuration */ + /* DAI format configuration */ int (*set_fmt)(struct snd_soc_codec_dai *codec_dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_codec_dai *codec_dai, @@ -293,24 +293,6 @@ struct snd_soc_codec_ops { int (*digital_mute)(struct snd_soc_codec_dai *, int mute); }; -/* ASoC cpu DAI ops */ -struct snd_soc_cpu_ops { - /* CPU DAI clocking configuration */ - int (*set_sysclk)(struct snd_soc_cpu_dai *cpu_dai, - int clk_id, unsigned int freq, int dir); - int (*set_clkdiv)(struct snd_soc_cpu_dai *cpu_dai, - int div_id, int div); - int (*set_pll)(struct snd_soc_cpu_dai *cpu_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out); - - /* CPU DAI format configuration */ - int (*set_fmt)(struct snd_soc_cpu_dai *cpu_dai, - unsigned int fmt); - int (*set_tdm_slot)(struct snd_soc_cpu_dai *cpu_dai, - unsigned int mask, int slots); - int (*set_tristate)(struct snd_soc_cpu_dai *, int tristate); -}; - /* SoC Codec DAI */ struct snd_soc_codec_dai { char *name; @@ -328,7 +310,7 @@ struct snd_soc_codec_dai { /* ops */ struct snd_soc_ops ops; - struct snd_soc_codec_ops dai_ops; + struct snd_soc_dai_ops dai_ops; /* DAI private data */ void *private_data; @@ -352,7 +334,7 @@ struct snd_soc_cpu_dai { /* ops */ struct snd_soc_ops ops; - struct snd_soc_cpu_ops dai_ops; + struct snd_soc_dai_ops dai_ops; /* DAI capabilities */ struct snd_soc_pcm_stream capture; From 0be9898adb6f58fee44f0fec0bbc0eac997ea9eb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 May 2008 12:31:28 +0200 Subject: [PATCH 032/179] [ALSA] ASoC: Clarify API for bias configuration Currently the ASoC core configures the bias levels in the system using a callback on codecs and machines called 'dapm_event', passing it PCI style power levels as SNDRV_CTL_POWER_ constants. This is more obscure than it needs to be and has caused confusion to driver authors, especially given that DAPM is also performing power management. Address this by renaming the callback function to 'set_bias_level' and using constants explicitly representing the off, standby, pre-on and on states which DAPM transitions through. Also unexport the API for setting bias level: there are currently no in-tree users of this API other than the core itself and it is likely that the core would need to be extended to cater for any users. Signed-off-by: Mark Brown Cc: Jarkko Nikula Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc-dapm.h | 3 ++- include/sound/soc.h | 28 ++++++++++++++++++---- sound/soc/codecs/tlv320aic3x.c | 26 ++++++++++---------- sound/soc/codecs/wm8731.c | 28 +++++++++++----------- sound/soc/codecs/wm8750.c | 36 ++++++++++++++-------------- sound/soc/codecs/wm8753.c | 36 ++++++++++++++-------------- sound/soc/codecs/wm9712.c | 28 +++++++++++----------- sound/soc/codecs/wm9713.c | 26 ++++++++++---------- sound/soc/soc-core.c | 25 +++++++++---------- sound/soc/soc-dapm.c | 44 ++++++++++++++++------------------ 10 files changed, 150 insertions(+), 130 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index bf4cf0c1d37f..f8223fae5804 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -221,7 +221,8 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); -int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event); +int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, + enum snd_soc_bias_level level); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/include/sound/soc.h b/include/sound/soc.h index 73accbcfbd2d..bca9538d9e50 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -102,6 +102,24 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&xenum } +/* + * Bias levels + * + * @ON: Bias is fully on for audio playback and capture operations. + * @PREPARE: Prepare for audio operations. Called before DAPM switching for + * stream start and stop operations. + * @STANDBY: Low power standby state when no playback/capture operations are + * in progress. NOTE: The transition time between STANDBY and ON + * should be as fast as possible and no longer than 10ms. + * @OFF: Power Off. No restrictions on transition times. + */ +enum snd_soc_bias_level { + SND_SOC_BIAS_ON, + SND_SOC_BIAS_PREPARE, + SND_SOC_BIAS_STANDBY, + SND_SOC_BIAS_OFF, +}; + /* * Digital Audio Interface (DAI) types */ @@ -356,7 +374,8 @@ struct snd_soc_codec { struct mutex mutex; /* callbacks */ - int (*dapm_event)(struct snd_soc_codec *codec, int event); + int (*set_bias_level)(struct snd_soc_codec *, + enum snd_soc_bias_level level); /* runtime */ struct snd_card *card; @@ -378,8 +397,8 @@ struct snd_soc_codec { /* dapm */ struct list_head dapm_widgets; struct list_head dapm_paths; - unsigned int dapm_state; - unsigned int suspend_dapm_state; + enum snd_soc_bias_level bias_level; + enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; /* codec DAI's */ @@ -449,7 +468,8 @@ struct snd_soc_machine { int (*resume_post)(struct platform_device *pdev); /* callbacks */ - int (*dapm_event)(struct snd_soc_machine *, int event); + int (*set_bias_level)(struct snd_soc_machine *, + enum snd_soc_bias_level level); /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index cb8365ac0c02..dc8a38d9e53a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -847,13 +847,14 @@ static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) +static int aic3x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { struct aic3x_priv *aic3x = codec->private_data; u8 reg; - switch (event) { - case SNDRV_CTL_POWER_D0: + switch (level) { + case SND_SOC_BIAS_ON: /* all power is driven by DAPM system */ if (aic3x->master) { /* enable pll */ @@ -862,10 +863,9 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) reg | PLL_ENABLE); } break; - case SNDRV_CTL_POWER_D1: - case SNDRV_CTL_POWER_D2: + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: + case SND_SOC_BIAS_STANDBY: /* * all power is driven by DAPM system, * so output power is safe if bypass was set @@ -877,7 +877,7 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) reg & ~PLL_ENABLE); } break; - case SNDRV_CTL_POWER_D3cold: + case SND_SOC_BIAS_OFF: /* force all power off */ reg = aic3x_read_reg_cache(codec, LINE1L_2_LADC_CTRL); aic3x_write(codec, LINE1L_2_LADC_CTRL, reg & ~LADC_PWR_ON); @@ -913,7 +913,7 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) } break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -979,7 +979,7 @@ static int aic3x_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; - aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -999,7 +999,7 @@ static int aic3x_resume(struct platform_device *pdev) codec->hw_write(codec->control_data, data, 2); } - aic3x_dapm_event(codec, codec->suspend_dapm_state); + aic3x_set_bias_level(codec, codec->suspend_bias_level); return 0; } @@ -1018,7 +1018,7 @@ static int aic3x_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = aic3x_read_reg_cache; codec->write = aic3x_write; - codec->dapm_event = aic3x_dapm_event; + codec->set_bias_level = aic3x_set_bias_level; codec->dai = &aic3x_dai; codec->num_dai = 1; codec->reg_cache_size = sizeof(aic3x_reg); @@ -1100,7 +1100,7 @@ static int aic3x_init(struct snd_soc_device *socdev) aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); /* off, with power on */ - aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* setup GPIO functions */ aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4); @@ -1271,7 +1271,7 @@ static int aic3x_remove(struct platform_device *pdev) /* power down chip */ if (codec->control_data) - aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3); + aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0cf9265fca8f..0f28aa4bcccb 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -435,29 +435,29 @@ static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8731_dapm_event(struct snd_soc_codec *codec, int event) +static int wm8731_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* vref/mid, osc on, dac unmute */ wm8731_write(codec, WM8731_PWR, reg); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ wm8731_write(codec, WM8731_PWR, reg | 0x0040); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ wm8731_write(codec, WM8731_ACTIVE, 0x0); wm8731_write(codec, WM8731_PWR, 0xffff); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -503,7 +503,7 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_codec *codec = socdev->codec; wm8731_write(codec, WM8731_ACTIVE, 0x0); - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -521,8 +521,8 @@ static int wm8731_resume(struct platform_device *pdev) data[1] = cache[i] & 0x00ff; codec->hw_write(codec->control_data, data, 2); } - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot); - wm8731_dapm_event(codec, codec->suspend_dapm_state); + wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8731_set_bias_level(codec, codec->suspend_bias_level); return 0; } @@ -539,7 +539,7 @@ static int wm8731_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = wm8731_read_reg_cache; codec->write = wm8731_write; - codec->dapm_event = wm8731_dapm_event; + codec->set_bias_level = wm8731_set_bias_level; codec->dai = &wm8731_dai; codec->num_dai = 1; codec->reg_cache_size = sizeof(wm8731_reg); @@ -557,7 +557,7 @@ static int wm8731_init(struct snd_soc_device *socdev) } /* power on device */ - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* set the update bits */ reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V); @@ -730,7 +730,7 @@ static int wm8731_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 16cd5d4d5ad9..62423f4493b0 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -686,29 +686,29 @@ static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int wm8750_dapm_event(struct snd_soc_codec *codec, int event) +static int wm8750_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: /* set vmid to 5k for quick power up */ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* mute dac and set vmid to 500k, enable VREF */ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: wm8750_write(codec, WM8750_PWR1, 0x0001); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -748,7 +748,7 @@ static void wm8750_work(struct work_struct *work) { struct snd_soc_codec *codec = container_of(work, struct snd_soc_codec, delayed_work.work); - wm8750_dapm_event(codec, codec->dapm_state); + wm8750_set_bias_level(codec, codec->bias_level); } static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) @@ -756,7 +756,7 @@ static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -777,12 +777,12 @@ static int wm8750_resume(struct platform_device *pdev) codec->hw_write(codec->control_data, data, 2); } - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge wm8750 caps */ - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D0; + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { + wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_ON; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000)); } @@ -803,7 +803,7 @@ static int wm8750_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = wm8750_read_reg_cache; codec->write = wm8750_write; - codec->dapm_event = wm8750_dapm_event; + codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; codec->reg_cache_size = sizeof(wm8750_reg); @@ -821,8 +821,8 @@ static int wm8750_init(struct snd_soc_device *socdev) } /* charge output caps */ - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D3hot; + wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_STANDBY; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ @@ -1021,7 +1021,7 @@ static int wm8750_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); run_delayed_work(&codec->delayed_work); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index fb41826c4c4c..9032b0c07c86 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1274,29 +1274,29 @@ static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int wm8753_dapm_event(struct snd_soc_codec *codec, int event) +static int wm8753_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 pwr_reg = wm8753_read_reg_cache(codec, WM8753_PWR1) & 0xfe3e; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x00c0); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: /* set vmid to 5k for quick power up */ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x01c1); break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* mute dac and set vmid to 500k, enable VREF */ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x0141); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: wm8753_write(codec, WM8753_PWR1, 0x0001); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -1500,7 +1500,7 @@ static void wm8753_work(struct work_struct *work) { struct snd_soc_codec *codec = container_of(work, struct snd_soc_codec, delayed_work.work); - wm8753_dapm_event(codec, codec->dapm_state); + wm8753_set_bias_level(codec, codec->bias_level); } static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) @@ -1512,7 +1512,7 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) if (!codec->card) return 0; - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -1537,12 +1537,12 @@ static int wm8753_resume(struct platform_device *pdev) codec->hw_write(codec->control_data, data, 2); } - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge wm8753 caps */ - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D0; + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { + wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_ON; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(caps_charge)); } @@ -1563,7 +1563,7 @@ static int wm8753_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = wm8753_read_reg_cache; codec->write = wm8753_write; - codec->dapm_event = wm8753_dapm_event; + codec->set_bias_level = wm8753_set_bias_level; codec->dai = wm8753_dai; codec->num_dai = 2; codec->reg_cache_size = sizeof(wm8753_reg); @@ -1584,8 +1584,8 @@ static int wm8753_init(struct snd_soc_device *socdev) } /* charge output caps */ - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D3hot; + wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_STANDBY; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(caps_charge)); @@ -1792,7 +1792,7 @@ static int wm8753_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); run_delayed_work(&codec->delayed_work); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 89efe40c7c33..e26cfcf0b4fc 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -571,23 +571,23 @@ struct snd_soc_codec_dai wm9712_dai[] = { }; EXPORT_SYMBOL_GPL(wm9712_dai); -static int wm9712_dapm_event(struct snd_soc_codec *codec, int event) +static int wm9712_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: ac97_write(codec, AC97_POWERDOWN, 0x0000); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: /* disable everything including AC link */ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -615,7 +615,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev, struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -632,7 +632,7 @@ static int wm9712_soc_resume(struct platform_device *pdev) return ret; } - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret == 0) { /* Sync reg_cache with the hardware after cold reset */ @@ -644,8 +644,8 @@ static int wm9712_soc_resume(struct platform_device *pdev) } } - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D0); + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + wm9712_set_bias_level(codec, SND_SOC_BIAS_ON); return ret; } @@ -679,7 +679,7 @@ static int wm9712_soc_probe(struct platform_device *pdev) codec->num_dai = ARRAY_SIZE(wm9712_dai); codec->write = ac97_write; codec->read = ac97_read; - codec->dapm_event = wm9712_dapm_event; + codec->set_bias_level = wm9712_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -703,7 +703,7 @@ static int wm9712_soc_probe(struct platform_device *pdev) /* set alc mux to none */ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm9712_add_controls(codec); wm9712_add_widgets(codec); ret = snd_soc_register_card(socdev); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 9e6b2fd7262b..4863636e9d56 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1094,33 +1094,33 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) } EXPORT_SYMBOL_GPL(wm9713_reset); -static int wm9713_dapm_event(struct snd_soc_codec *codec, int event) +static int wm9713_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 reg; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* enable thermal shutdown */ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff; ac97_write(codec, AC97_EXTENDED_MID, reg); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* enable master bias and vmid */ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff; ac97_write(codec, AC97_EXTENDED_MID, reg); ac97_write(codec, AC97_POWERDOWN, 0x0000); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: /* disable everything including AC link */ ac97_write(codec, AC97_EXTENDED_MID, 0xffff); ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -1157,7 +1157,7 @@ static int wm9713_soc_resume(struct platform_device *pdev) return ret; } - wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* do we need to re-start the PLL ? */ if (wm9713->pll_out) @@ -1173,8 +1173,8 @@ static int wm9713_soc_resume(struct platform_device *pdev) } } - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) - wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0); + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + wm9713_set_bias_level(codec, SND_SOC_BIAS_ON); return ret; } @@ -1213,7 +1213,7 @@ static int wm9713_soc_probe(struct platform_device *pdev) codec->num_dai = ARRAY_SIZE(wm9713_dai); codec->write = ac97_write; codec->read = ac97_read; - codec->dapm_event = wm9713_dapm_event; + codec->set_bias_level = wm9713_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -1235,7 +1235,7 @@ static int wm9713_soc_probe(struct platform_device *pdev) goto reset_err; } - wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* unmute the adc - move to kcontrol */ reg = ac97_read(codec, AC97_CD) & 0x7fff; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0318d8abe3e8..a05b3450aee8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -283,12 +283,12 @@ static void close_delayed_work(struct work_struct *work) /* are we waiting on this codec DAI stream */ if (codec_dai->pop_wait == 1) { - /* power down the codec to D1 if no longer active */ + /* Reduce power if no longer active */ if (codec->active == 0) { dbg("pop wq D1 %s %s\n", codec->name, codec_dai->playback.stream_name); - snd_soc_dapm_device_event(socdev, - SNDRV_CTL_POWER_D1); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_PREPARE); } codec_dai->pop_wait = 0; @@ -296,12 +296,12 @@ static void close_delayed_work(struct work_struct *work) codec_dai->playback.stream_name, SND_SOC_DAPM_STREAM_STOP); - /* power down the codec power domain if no longer active */ + /* Fall into standby if no longer active */ if (codec->active == 0) { dbg("pop wq D3 %s %s\n", codec->name, codec_dai->playback.stream_name); - snd_soc_dapm_device_event(socdev, - SNDRV_CTL_POWER_D3hot); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); } } } @@ -361,8 +361,8 @@ static int soc_codec_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); if (codec->active == 0 && codec_dai->pop_wait == 0) - snd_soc_dapm_device_event(socdev, - SNDRV_CTL_POWER_D3hot); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); } mutex_unlock(&pcm_mutex); @@ -435,9 +435,10 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } else { /* no delayed work - do we need to power up codec */ - if (codec->dapm_state != SNDRV_CTL_POWER_D0) { + if (codec->bias_level != SND_SOC_BIAS_ON) { - snd_soc_dapm_device_event(socdev, SNDRV_CTL_POWER_D1); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_PREPARE); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dapm_stream_event(codec, @@ -448,7 +449,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - snd_soc_dapm_device_event(socdev, SNDRV_CTL_POWER_D0); + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); if (codec_dai->dai_ops.digital_mute) codec_dai->dai_ops.digital_mute(codec_dai, 0); @@ -658,7 +659,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* close any waiting streams and save state */ run_delayed_work(&socdev->delayed_work); - codec->suspend_dapm_state = codec->dapm_state; + codec->suspend_bias_level = codec->bias_level; for(i = 0; i < codec->num_dai; i++) { char *stream = codec->dai[i].playback.stream_name; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8a3192bcee78..728f3ac2f304 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -763,21 +763,18 @@ static ssize_t dapm_widget_show(struct device *dev, } } - switch(codec->dapm_state){ - case SNDRV_CTL_POWER_D0: - state = "D0"; + switch (codec->bias_level) { + case SND_SOC_BIAS_ON: + state = "On"; break; - case SNDRV_CTL_POWER_D1: - state = "D1"; + case SND_SOC_BIAS_PREPARE: + state = "Prepare"; break; - case SNDRV_CTL_POWER_D2: - state = "D2"; + case SND_SOC_BIAS_STANDBY: + state = "Standby"; break; - case SNDRV_CTL_POWER_D3hot: - state = "D3hot"; - break; - case SNDRV_CTL_POWER_D3cold: - state = "D3cold"; + case SND_SOC_BIAS_OFF: + state = "Off"; break; } count += sprintf(buf + count, "PM State: %s\n", state); @@ -1358,27 +1355,28 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); /** - * snd_soc_dapm_device_event - send a device event to the dapm core + * snd_soc_dapm_set_bias_level - set the bias level for the system * @socdev: audio device - * @event: device event + * @level: level to configure * - * Sends a device event to the dapm core. The core then makes any - * necessary machine or codec power changes.. + * Configure the bias (power) levels for the SoC audio device. * * Returns 0 for success else error. */ -int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event) +int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, + enum snd_soc_bias_level level) { struct snd_soc_codec *codec = socdev->codec; struct snd_soc_machine *machine = socdev->machine; + int ret = 0; - if (machine->dapm_event) - machine->dapm_event(machine, event); - if (codec->dapm_event) - codec->dapm_event(codec, event); - return 0; + if (machine->set_bias_level) + ret = machine->set_bias_level(machine, level); + if (ret == 0 && codec->set_bias_level) + ret = codec->set_bias_level(codec, level); + + return ret; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_device_event); /** * snd_soc_dapm_set_endpoint - set audio endpoint status From 3ff3f64ba04b3e5a86dce5239b10268798f68ad7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 May 2008 12:32:25 +0200 Subject: [PATCH 033/179] [ALSA] ASoC: core checkpatch cleanups Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-core.c | 100 +++++++++++++++++++++++-------------------- 1 file changed, 54 insertions(+), 46 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a05b3450aee8..a3f091e0843a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -108,9 +108,9 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif -static inline const char* get_dai_name(int type) +static inline const char *get_dai_name(int type) { - switch(type) { + switch (type) { case SND_SOC_DAI_AC97_BUS: case SND_SOC_DAI_AC97: return "AC97"; @@ -178,9 +178,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) /* Check that the codec and cpu DAI's are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { runtime->hw.rate_min = - max(codec_dai->playback.rate_min, cpu_dai->playback.rate_min); + max(codec_dai->playback.rate_min, + cpu_dai->playback.rate_min); runtime->hw.rate_max = - min(codec_dai->playback.rate_max, cpu_dai->playback.rate_max); + min(codec_dai->playback.rate_max, + cpu_dai->playback.rate_max); runtime->hw.channels_min = max(codec_dai->playback.channels_min, cpu_dai->playback.channels_min); @@ -193,9 +195,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->playback.rates & cpu_dai->playback.rates; } else { runtime->hw.rate_min = - max(codec_dai->capture.rate_min, cpu_dai->capture.rate_min); + max(codec_dai->capture.rate_min, + cpu_dai->capture.rate_min); runtime->hw.rate_max = - min(codec_dai->capture.rate_max, cpu_dai->capture.rate_max); + min(codec_dai->capture.rate_max, + cpu_dai->capture.rate_max); runtime->hw.channels_min = max(codec_dai->capture.channels_min, cpu_dai->capture.channels_min); @@ -225,7 +229,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto machine_err; } - dbg("asoc: %s <-> %s info:\n",codec_dai->name, cpu_dai->name); + dbg("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); dbg("asoc: rate mask 0x%x\n", runtime->hw.rates); dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, runtime->hw.channels_max); @@ -272,7 +276,7 @@ static void close_delayed_work(struct work_struct *work) int i; mutex_lock(&pcm_mutex); - for(i = 0; i < codec->num_dai; i++) { + for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; dbg("pop wq checking: %s status: %s waiting: %s\n", @@ -511,7 +515,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (cpu_dai->ops.hw_params) { ret = cpu_dai->ops.hw_params(substream, params); if (ret < 0) { - printk(KERN_ERR "asoc: can't set interface %s hw params\n", + printk(KERN_ERR "asoc: interface %s hw params failed\n", cpu_dai->name); goto interface_err; } @@ -520,7 +524,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (platform->pcm_ops->hw_params) { ret = platform->pcm_ops->hw_params(substream, params); if (ret < 0) { - printk(KERN_ERR "asoc: can't set platform %s hw params\n", + printk(KERN_ERR "asoc: platform %s hw params failed\n", platform->name); goto platform_err; } @@ -539,7 +543,7 @@ interface_err: codec_dai->ops.hw_free(substream); codec_err: - if(machine->ops && machine->ops->hw_free) + if (machine->ops && machine->ops->hw_free) machine->ops->hw_free(substream); mutex_unlock(&pcm_mutex); @@ -628,15 +632,15 @@ static struct snd_pcm_ops soc_pcm_ops = { /* powers down audio subsystem for suspend */ static int soc_suspend(struct platform_device *pdev, pm_message_t state) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_codec_device *codec_dev = socdev->codec_dev; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_codec_device *codec_dev = socdev->codec_dev; struct snd_soc_codec *codec = socdev->codec; int i; /* mute any active DAC's */ - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 1); @@ -649,7 +653,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) if (machine->suspend_pre) machine->suspend_pre(pdev, state); - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); @@ -661,7 +665,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) run_delayed_work(&socdev->delayed_work); codec->suspend_bias_level = codec->bias_level; - for(i = 0; i < codec->num_dai; i++) { + for (i = 0; i < codec->num_dai; i++) { char *stream = codec->dai[i].playback.stream_name; if (stream != NULL) snd_soc_dapm_stream_event(codec, stream, @@ -675,7 +679,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) if (codec_dev->suspend) codec_dev->suspend(pdev, state); - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); @@ -690,17 +694,17 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* powers up audio subsystem after a suspend */ static int soc_resume(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_codec_device *codec_dev = socdev->codec_dev; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_codec_device *codec_dev = socdev->codec_dev; struct snd_soc_codec *codec = socdev->codec; int i; if (machine->resume_pre) machine->resume_pre(pdev); - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); @@ -709,8 +713,8 @@ static int soc_resume(struct platform_device *pdev) if (codec_dev->resume) codec_dev->resume(pdev); - for(i = 0; i < codec->num_dai; i++) { - char* stream = codec->dai[i].playback.stream_name; + for (i = 0; i < codec->num_dai; i++) { + char *stream = codec->dai[i].playback.stream_name; if (stream != NULL) snd_soc_dapm_stream_event(codec, stream, SND_SOC_DAPM_STREAM_RESUME); @@ -720,14 +724,14 @@ static int soc_resume(struct platform_device *pdev) SND_SOC_DAPM_STREAM_RESUME); } - /* unmute any active DAC's */ - for(i = 0; i < machine->num_links; i++) { + /* unmute any active DACs */ + for (i = 0; i < machine->num_links; i++) { struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 0); } - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); @@ -757,7 +761,7 @@ static int soc_probe(struct platform_device *pdev) if (machine->probe) { ret = machine->probe(pdev); - if(ret < 0) + if (ret < 0) return ret; } @@ -765,20 +769,20 @@ static int soc_probe(struct platform_device *pdev) struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->probe) { ret = cpu_dai->probe(pdev); - if(ret < 0) + if (ret < 0) goto cpu_dai_err; } } if (codec_dev->probe) { ret = codec_dev->probe(pdev); - if(ret < 0) + if (ret < 0) goto cpu_dai_err; } if (platform->probe) { ret = platform->probe(pdev); - if(ret < 0) + if (ret < 0) goto platform_err; } @@ -865,7 +869,7 @@ static int soc_new_pcm(struct snd_soc_device *socdev, codec_dai->codec = socdev->codec; /* check client and interface hw capabilities */ - sprintf(new_name, "%s %s-%s-%d",dai_link->stream_name, codec_dai->name, + sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name, get_dai_name(cpu_dai->type), num); if (codec_dai->playback.channels_min) @@ -876,7 +880,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback, capture, &pcm); if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + printk(KERN_ERR "asoc: can't create pcm for codec %s\n", + codec->name); kfree(rtd); return ret; } @@ -925,8 +930,9 @@ static ssize_t codec_reg_show(struct device *dev, step = codec->reg_cache_step; count += sprintf(buf, "%s registers\n", codec->name); - for(i = 0; i < codec->reg_cache_size; i += step) - count += sprintf(buf + count, "%2x: %4x\n", i, codec->read(codec, i)); + for (i = 0; i < codec->reg_cache_size; i += step) + count += sprintf(buf + count, "%2x: %4x\n", i, + codec->read(codec, i)); return count; } @@ -1069,7 +1075,7 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver)); /* create the pcms */ - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { ret = soc_new_pcm(socdev, &machine->dai_link[i], i); if (ret < 0) { printk(KERN_ERR "asoc: can't create pcm %s\n", @@ -1099,7 +1105,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) struct snd_soc_machine *machine = socdev->machine; int ret = 0, i, ac97 = 0, err = 0; - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { if (socdev->machine->dai_link[i].init) { err = socdev->machine->dai_link[i].init(codec); if (err < 0) { @@ -1108,7 +1114,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) continue; } } - if (socdev->machine->dai_link[i].codec_dai->type == + if (socdev->machine->dai_link[i].codec_dai->type == SND_SOC_DAI_AC97_BUS) ac97 = 1; } @@ -1119,7 +1125,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) ret = snd_card_register(codec->card); if (ret < 0) { - printk(KERN_ERR "asoc: failed to register soundcard for codec %s\n", + printk(KERN_ERR "asoc: failed to register soundcard for %s\n", codec->name); goto out; } @@ -1143,7 +1149,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) err = device_create_file(socdev->dev, &dev_attr_codec_reg); if (err < 0) - printk(KERN_WARNING "asoc: failed to add codec sysfs entries\n"); + printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); mutex_unlock(&codec->mutex); @@ -1169,7 +1175,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS - for(i = 0; i < codec->num_dai; i++) { + for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) { soc_ac97_dev_unregister(codec); @@ -1279,7 +1285,8 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) ; val = snd_soc_read(codec, e->reg); - ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); + ucontrol->value.enumerated.item[0] + = (val >> e->shift_l) & (bitmask - 1); if (e->shift_l != e->shift_r) ucontrol->value.enumerated.item[1] = (val >> e->shift_r) & (bitmask - 1); @@ -1573,7 +1580,8 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, val = val << shift; val2 = val2 << shift; - if ((err = snd_soc_update_bits(codec, reg, val_mask, val)) < 0) + err = snd_soc_update_bits(codec, reg, val_mask, val); + if (err < 0) return err; err = snd_soc_update_bits(codec, reg2, val_mask, val2); @@ -1589,7 +1597,7 @@ static int __devinit snd_soc_init(void) static void snd_soc_exit(void) { - platform_driver_unregister(&soc_driver); + platform_driver_unregister(&soc_driver); } module_init(snd_soc_init); From a65f0568f6cc8433877fb71dd7d36b551854b0bc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:54:43 +0200 Subject: [PATCH 034/179] [ALSA] soc - Convert Wolfson codec drivers to use bulk DAPM registration Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm8731.c | 16 ++++------------ sound/soc/codecs/wm8750.c | 16 ++++------------ sound/soc/codecs/wm8753.c | 17 ++++------------- sound/soc/codecs/wm9712.c | 15 ++++----------- sound/soc/codecs/wm9713.c | 15 ++++----------- 5 files changed, 20 insertions(+), 59 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0f28aa4bcccb..5acf43ab104e 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -193,7 +193,7 @@ SND_SOC_DAPM_INPUT("RLINEIN"), SND_SOC_DAPM_INPUT("LLINEIN"), }; -static const char *intercon[][3] = { +static const struct snd_soc_dapm_route intercon[] = { /* output mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, {"Output Mixer", "HiFi Playback Switch", "DAC"}, @@ -214,22 +214,14 @@ static const char *intercon[][3] = { {"Line Input", NULL, "LLINEIN"}, {"Line Input", NULL, "RLINEIN"}, {"Mic Bias", NULL, "MICIN"}, - - /* terminator */ - {NULL, NULL, NULL}, }; static int wm8731_add_widgets(struct snd_soc_codec *codec) { - int i; + snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, + ARRAY_SIZE(wm8731_dapm_widgets)); - for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); - - /* set up audio path interconnects */ - for (i = 0; intercon[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, intercon[i][0], - intercon[i][1], intercon[i][2]); + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); snd_soc_dapm_new_widgets(codec); return 0; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 62423f4493b0..1f11ad24551a 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -378,7 +378,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RINPUT3"), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* left mixer */ {"Left Mixer", "Playback Switch", "Left DAC"}, {"Left Mixer", "Left Bypass Switch", "Left Line Mux"}, @@ -470,22 +470,14 @@ static const char *audio_map[][3] = { /* ADC */ {"Left ADC", NULL, "Left ADC Mux"}, {"Right ADC", NULL, "Right ADC Mux"}, - - /* terminator */ - {NULL, NULL, NULL}, }; static int wm8750_add_widgets(struct snd_soc_codec *codec) { - int i; + snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, + ARRAY_SIZE(wm8750_dapm_widgets)); - for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); - - /* set up audio path audio_mapnects */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 9032b0c07c86..c32e6326be6c 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -523,7 +523,7 @@ SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_VMID("VREF"), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* left mixer */ {"Left Mixer", "Left Playback Switch", "Left DAC"}, {"Left Mixer", "Voice Playback Switch", "Voice DAC"}, @@ -674,23 +674,14 @@ static const char *audio_map[][3] = { /* ACOP */ {"ACOP", NULL, "ALC Mixer"}, - - /* terminator */ - {NULL, NULL, NULL}, }; static int wm8753_add_widgets(struct snd_soc_codec *codec) { - int i; + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); - for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]); - - /* set up the WM8753 audio map */ - for (i = 0; audio_map[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); - } + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e26cfcf0b4fc..d9789f1c8903 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -348,7 +348,7 @@ SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* virtual mixer - mixes left & right channels for spk and mono */ {"AC97 Mixer", NULL, "Left DAC"}, {"AC97 Mixer", NULL, "Right DAC"}, @@ -443,21 +443,14 @@ static const char *audio_map[][3] = { {"Speaker PGA", NULL, "Speaker Mux"}, {"LOUT2", NULL, "Speaker PGA"}, {"ROUT2", NULL, "Speaker PGA"}, - - {NULL, NULL, NULL}, }; static int wm9712_add_widgets(struct snd_soc_codec *codec) { - int i; + snd_soc_dapm_new_controls(codec, wm9712_dapm_widgets, + ARRAY_SIZE(wm9712_dapm_widgets)); - for (i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]); - - /* set up audio path connects */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 4863636e9d56..4f516a5a5616 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -453,7 +453,7 @@ SND_SOC_DAPM_INPUT("MIC2B"), SND_SOC_DAPM_VMID("VMID"), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* left HP mixer */ {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"}, @@ -604,21 +604,14 @@ static const char *audio_map[][3] = { {"Capture Mono Mux", "Stereo", "Capture Mixer"}, {"Capture Mono Mux", "Left", "Left Capture Source"}, {"Capture Mono Mux", "Right", "Right Capture Source"}, - - {NULL, NULL, NULL}, }; static int wm9713_add_widgets(struct snd_soc_codec *codec) { - int i; + snd_soc_dapm_new_controls(codec, wm9713_dapm_widgets, + ARRAY_SIZE(wm9713_dapm_widgets)); - for (i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]); - - /* set up audio path audio_mapnects */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; From 33e5b22285f63ede858c00456f3ffbc2ea79d6cf Mon Sep 17 00:00:00 2001 From: Werner Almesberger Date: Mon, 14 Apr 2008 14:26:44 +0200 Subject: [PATCH 035/179] [ALSA] soc - Fix s3c24xx-i2s LR sync while timer ticks are disabled When timer ticks are disabled when calling sound/soc/s3c24xx/s3c24xx-i2s.c:s3c24xx_snd_lrsync and the LR signal never happens, we loop forever. This has been observed in the following call chain: snd_pcm_common_ioctl1 -> snd_pcm_action_lock_irq -> snd_pcm_action_single -> snd_pcm_do_resume -> soc_pcm_trigger -> s3c24xx_i2s_trigger The patch below changes the timeout mechanism to use udelay, which doesn't need timer ticks. Signed-off-by: Werner Almesberger Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/s3c24xx-i2s.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index ddf87246c77b..4c52f7946d9e 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -175,7 +175,7 @@ static void s3c24xx_snd_rxctrl(int on) static int s3c24xx_snd_lrsync(void) { u32 iiscon; - unsigned long timeout = jiffies + msecs_to_jiffies(5); + int timeout = 50; /* 5ms */ DBG("Entered %s\n", __func__); @@ -184,8 +184,9 @@ static int s3c24xx_snd_lrsync(void) if (iiscon & S3C2410_IISCON_LRINDEX) break; - if (time_after(jiffies, timeout)) + if (!timeout--) return -ETIMEDOUT; + udelay(100); } return 0; From aea3bfbcfb0453217c8da6cfdc1b2394d214bee5 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 20 May 2008 14:22:44 +0200 Subject: [PATCH 036/179] [ALSA] ice1724: fix MIDI The VT1724 MIDI port is not MPU-401 compatible; remove the hacks that try to make the MPU-401 library work with it, and just use some simple device-specific code. Signed-off-by: Clemens Ladisch Tested-by: Pavel Hofman --- sound/pci/Kconfig | 2 +- sound/pci/ice1712/envy24ht.h | 10 +- sound/pci/ice1712/ice1712.h | 2 + sound/pci/ice1712/ice1724.c | 215 +++++++++++++++++++++++++++-------- 4 files changed, 178 insertions(+), 51 deletions(-) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7e4742109572..d95fbb2b5b9f 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -692,7 +692,7 @@ config SND_ICE1712 config SND_ICE1724 tristate "ICE/VT1724/1720 (Envy24HT/PT)" depends on SND - select SND_MPU401_UART + select SND_RAWMIDI select SND_AC97_CODEC select SND_VMASTER help diff --git a/sound/pci/ice1712/envy24ht.h b/sound/pci/ice1712/envy24ht.h index 43b9e3e858be..a0c5e009bb4a 100644 --- a/sound/pci/ice1712/envy24ht.h +++ b/sound/pci/ice1712/envy24ht.h @@ -93,9 +93,13 @@ enum { #define VT1724_REG_MPU_TXFIFO 0x0a /*byte ro. number of bytes in TX fifo*/ #define VT1724_REG_MPU_RXFIFO 0x0b /*byte ro. number of bytes in RX fifo*/ -//are these 2 the wrong way around? they don't seem to be used yet anyway -#define VT1724_REG_MPU_CTRL 0x0c /* byte */ -#define VT1724_REG_MPU_DATA 0x0d /* byte */ +#define VT1724_REG_MPU_DATA 0x0c /* byte */ +#define VT1724_REG_MPU_CTRL 0x0d /* byte */ +#define VT1724_MPU_UART 0x01 +#define VT1724_MPU_TX_EMPTY 0x02 +#define VT1724_MPU_TX_FULL 0x04 +#define VT1724_MPU_RX_EMPTY 0x08 +#define VT1724_MPU_RX_FULL 0x10 #define VT1724_REG_MPU_FIFO_WM 0x0e /*byte set the high/low watermarks for RX/TX fifos*/ #define VT1724_MPU_RX_FIFO 0x20 //1=rx fifo watermark 0=tx fifo watermark diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 3208901c740e..762fbd7a7507 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -333,6 +333,8 @@ struct snd_ice1712 { unsigned int has_spdif: 1; /* VT1720/4 - has SPDIF I/O */ unsigned int force_pdma4: 1; /* VT1720/4 - PDMA4 as non-spdif */ unsigned int force_rdma1: 1; /* VT1720/4 - RDMA1 as non-spdif */ + unsigned int midi_output: 1; /* VT1720/4: MIDI output triggered */ + unsigned int midi_input: 1; /* VT1720/4: MIDI input triggered */ unsigned int num_total_dacs; /* total DACs */ unsigned int num_total_adcs; /* total ADCs */ unsigned int cur_rate; /* current rate */ diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 67350901772c..e596d777d9dd 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -32,7 +32,7 @@ #include #include #include -#include +#include #include #include @@ -223,30 +223,153 @@ static unsigned int snd_vt1724_get_gpio_data(struct snd_ice1712 *ice) } /* - * MPU401 accessor + * MIDI */ -static unsigned char snd_vt1724_mpu401_read(struct snd_mpu401 *mpu, - unsigned long addr) + +static void vt1724_midi_clear_rx(struct snd_ice1712 *ice) { - /* fix status bits to the standard position */ - /* only RX_EMPTY and TX_FULL are checked */ - if (addr == MPU401C(mpu)) - return (inb(addr) & 0x0c) << 4; - else - return inb(addr); + unsigned int count; + + for (count = inb(ICEREG1724(ice, MPU_RXFIFO)); count > 0; --count) + inb(ICEREG1724(ice, MPU_DATA)); } -static void snd_vt1724_mpu401_write(struct snd_mpu401 *mpu, - unsigned char data, unsigned long addr) +static inline struct snd_rawmidi_substream * +get_rawmidi_substream(struct snd_ice1712 *ice, unsigned int stream) { - if (addr == MPU401C(mpu)) { - if (data == MPU401_ENTER_UART) - outb(0x01, addr); - /* what else? */ - } else - outb(data, addr); + return list_first_entry(&ice->rmidi[0]->streams[stream].substreams, + struct snd_rawmidi_substream, list); } +static void vt1724_midi_write(struct snd_ice1712 *ice) +{ + struct snd_rawmidi_substream *s; + int count, i; + u8 buffer[32]; + + s = get_rawmidi_substream(ice, SNDRV_RAWMIDI_STREAM_OUTPUT); + count = 31 - inb(ICEREG1724(ice, MPU_TXFIFO)); + if (count > 0) { + count = snd_rawmidi_transmit(s, buffer, count); + for (i = 0; i < count; ++i) + outb(buffer[i], ICEREG1724(ice, MPU_DATA)); + } +} + +static void vt1724_midi_read(struct snd_ice1712 *ice) +{ + struct snd_rawmidi_substream *s; + int count, i; + u8 buffer[32]; + + s = get_rawmidi_substream(ice, SNDRV_RAWMIDI_STREAM_INPUT); + count = inb(ICEREG1724(ice, MPU_RXFIFO)); + if (count > 0) { + count = min(count, 32); + for (i = 0; i < count; ++i) + buffer[i] = inb(ICEREG1724(ice, MPU_DATA)); + snd_rawmidi_receive(s, buffer, count); + } +} + +static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream, + u8 flag, int enable) +{ + struct snd_ice1712 *ice = substream->rmidi->private_data; + u8 mask; + + spin_lock_irq(&ice->reg_lock); + mask = inb(ICEREG1724(ice, IRQMASK)); + if (enable) + mask &= ~flag; + else + mask |= flag; + outb(mask, ICEREG1724(ice, IRQMASK)); + spin_unlock_irq(&ice->reg_lock); +} + +static int vt1724_midi_output_open(struct snd_rawmidi_substream *s) +{ + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 1); + return 0; +} + +static int vt1724_midi_output_close(struct snd_rawmidi_substream *s) +{ + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0); + return 0; +} + +static void vt1724_midi_output_trigger(struct snd_rawmidi_substream *s, int up) +{ + struct snd_ice1712 *ice = s->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&ice->reg_lock, flags); + if (up) { + ice->midi_output = 1; + vt1724_midi_write(ice); + } else { + ice->midi_output = 0; + } + spin_unlock_irqrestore(&ice->reg_lock, flags); +} + +static void vt1724_midi_output_drain(struct snd_rawmidi_substream *s) +{ + struct snd_ice1712 *ice = s->rmidi->private_data; + unsigned long timeout; + + /* 32 bytes should be transmitted in less than about 12 ms */ + timeout = jiffies + msecs_to_jiffies(15); + do { + if (inb(ICEREG1724(ice, MPU_CTRL)) & VT1724_MPU_TX_EMPTY) + break; + schedule_timeout_uninterruptible(1); + } while (time_after(timeout, jiffies)); +} + +static struct snd_rawmidi_ops vt1724_midi_output_ops = { + .open = vt1724_midi_output_open, + .close = vt1724_midi_output_close, + .trigger = vt1724_midi_output_trigger, + .drain = vt1724_midi_output_drain, +}; + +static int vt1724_midi_input_open(struct snd_rawmidi_substream *s) +{ + vt1724_midi_clear_rx(s->rmidi->private_data); + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_RX, 1); + return 0; +} + +static int vt1724_midi_input_close(struct snd_rawmidi_substream *s) +{ + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_RX, 0); + return 0; +} + +static void vt1724_midi_input_trigger(struct snd_rawmidi_substream *s, int up) +{ + struct snd_ice1712 *ice = s->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&ice->reg_lock, flags); + if (up) { + ice->midi_input = 1; + vt1724_midi_read(ice); + } else { + ice->midi_input = 0; + } + spin_unlock_irqrestore(&ice->reg_lock, flags); +} + +static struct snd_rawmidi_ops vt1724_midi_input_ops = { + .open = vt1724_midi_input_open, + .close = vt1724_midi_input_close, + .trigger = vt1724_midi_input_trigger, +}; + /* * Interrupt handler @@ -278,13 +401,10 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) #endif handled = 1; if (status & VT1724_IRQ_MPU_TX) { - if (ice->rmidi[0]) - snd_mpu401_uart_interrupt_tx(irq, - ice->rmidi[0]->private_data); - else /* disable TX to be sure */ - outb(inb(ICEREG1724(ice, IRQMASK)) | - VT1724_IRQ_MPU_TX, - ICEREG1724(ice, IRQMASK)); + spin_lock(&ice->reg_lock); + if (ice->midi_output) + vt1724_midi_write(ice); + spin_unlock(&ice->reg_lock); /* Due to mysterical reasons, MPU_TX is always * generated (and can't be cleared) when a PCM * playback is going. So let's ignore at the @@ -293,13 +413,12 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) status_mask &= ~VT1724_IRQ_MPU_TX; } if (status & VT1724_IRQ_MPU_RX) { - if (ice->rmidi[0]) - snd_mpu401_uart_interrupt(irq, - ice->rmidi[0]->private_data); - else /* disable RX to be sure */ - outb(inb(ICEREG1724(ice, IRQMASK)) | - VT1724_IRQ_MPU_RX, - ICEREG1724(ice, IRQMASK)); + spin_lock(&ice->reg_lock); + if (ice->midi_input) + vt1724_midi_read(ice); + else + vt1724_midi_clear_rx(ice); + spin_unlock(&ice->reg_lock); } /* ack MPU irq */ outb(status, ICEREG1724(ice, IRQSTAT)); @@ -2425,28 +2544,30 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, if (! c->no_mpu401) { if (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401) { - struct snd_mpu401 *mpu; - if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, - ICEREG1724(ice, MPU_CTRL), - (MPU401_INFO_INTEGRATED | - MPU401_INFO_NO_ACK | - MPU401_INFO_TX_IRQ), - ice->irq, 0, - &ice->rmidi[0])) < 0) { + struct snd_rawmidi *rmidi; + + err = snd_rawmidi_new(card, "MIDI", 0, 1, 1, &rmidi); + if (err < 0) { snd_card_free(card); return err; } - mpu = ice->rmidi[0]->private_data; - mpu->read = snd_vt1724_mpu401_read; - mpu->write = snd_vt1724_mpu401_write; - /* unmask MPU RX/TX irqs */ - outb(inb(ICEREG1724(ice, IRQMASK)) & - ~(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX), - ICEREG1724(ice, IRQMASK)); + ice->rmidi[0] = rmidi; + rmidi->private_data = ice; + strcpy(rmidi->name, "ICE1724 MIDI"); + rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | + SNDRV_RAWMIDI_INFO_INPUT | + SNDRV_RAWMIDI_INFO_DUPLEX; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &vt1724_midi_output_ops); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &vt1724_midi_input_ops); + /* set watermarks */ outb(VT1724_MPU_RX_FIFO | 0x1, ICEREG1724(ice, MPU_FIFO_WM)); outb(0x1, ICEREG1724(ice, MPU_FIFO_WM)); + /* set UART mode */ + outb(VT1724_MPU_UART, ICEREG1724(ice, MPU_CTRL)); } } From 62b126315369d52e40ad604e1798ff5b6265287e Mon Sep 17 00:00:00 2001 From: Chris Mennie Date: Mon, 19 May 2008 16:21:33 +0200 Subject: [PATCH 037/179] [ALSA] usb-audio - Support for Roland SonicCell sound module Added entry into usbquirks.h to recognize Roland SonicCell sound module by mostly duplicating the entry for the Roland SH-201. USB MIDI works just fine, though the USB audio is a little unreliable (but still works well enough). Signed-off-by: Chris Mennie Signed-off-by: Takashi Iwai --- sound/usb/usbquirks.h | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 82a8d14c26af..b7ab3ee7647e 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1379,6 +1379,39 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + /* Roland SonicCell */ + USB_DEVICE(0x0582, 0x00c2), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "SonicCell", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, + + /* Guillemot devices */ { /* From a72e72469a166c825196c3f20dabd352877fec2b Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Tue, 20 May 2008 01:06:55 +0300 Subject: [PATCH 038/179] [ALSA] remove CVS keywords This patch removes CVS keywords that weren't updated for a long time from comments. Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai --- include/sound/uda1341.h | 2 -- sound/arm/sa11xx-uda1341.c | 2 -- sound/i2c/l3/uda1341.c | 2 -- sound/pci/au88x0/au88x0_game.c | 2 -- 4 files changed, 8 deletions(-) diff --git a/include/sound/uda1341.h b/include/sound/uda1341.h index 2e564bfb37fe..110d5dc3a2be 100644 --- a/include/sound/uda1341.h +++ b/include/sound/uda1341.h @@ -15,8 +15,6 @@ * features support */ -/* $Id: uda1341.h,v 1.8 2005/11/17 14:17:21 tiwai Exp $ */ - #define UDA1341_ALSA_NAME "snd-uda1341" /* diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c index 0eff33ca0f79..faeddf3ecedb 100644 --- a/sound/arm/sa11xx-uda1341.c +++ b/sound/arm/sa11xx-uda1341.c @@ -21,8 +21,6 @@ * merged HAL layer (patches from Brian) */ -/* $Id: sa11xx-uda1341.c,v 1.27 2005/12/07 09:13:42 cladisch Exp $ */ - /*************************************************************************************************** * * To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai diff --git a/sound/i2c/l3/uda1341.c b/sound/i2c/l3/uda1341.c index bfa5d2c3608b..1f4942ea1414 100644 --- a/sound/i2c/l3/uda1341.c +++ b/sound/i2c/l3/uda1341.c @@ -17,8 +17,6 @@ * 2002-05-12 Tomas Kasparek another code cleanup */ -/* $Id: uda1341.c,v 1.18 2005/11/17 14:17:21 tiwai Exp $ */ - #include #include #include diff --git a/sound/pci/au88x0/au88x0_game.c b/sound/pci/au88x0/au88x0_game.c index bc212f41a38a..e291aa59742e 100644 --- a/sound/pci/au88x0/au88x0_game.c +++ b/sound/pci/au88x0/au88x0_game.c @@ -1,6 +1,4 @@ /* - * $Id: au88x0_game.c,v 1.9 2003/09/22 03:51:28 mjander Exp $ - * * Manuel Jander. * * Based on the work of: From 06b5fb97cec1a3ca61d10164118b00fe98a6a866 Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Tue, 20 May 2008 00:59:35 +0300 Subject: [PATCH 039/179] sound: sound/oss/: remove CVS keywords This patch removes CVS keywords that weren't updated for a long time from comments. Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai --- sound/oss/msnd.c | 2 -- sound/oss/msnd.h | 2 -- sound/oss/msnd_classic.h | 2 -- sound/oss/msnd_pinnacle.c | 5 ----- sound/oss/msnd_pinnacle.h | 2 -- 5 files changed, 13 deletions(-) diff --git a/sound/oss/msnd.c b/sound/oss/msnd.c index ba38d6200099..e4282d93a1aa 100644 --- a/sound/oss/msnd.c +++ b/sound/oss/msnd.c @@ -20,8 +20,6 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. * - * $Id: msnd.c,v 1.17 1999/03/21 16:50:09 andrewtv Exp $ - * ********************************************************************/ #include diff --git a/sound/oss/msnd.h b/sound/oss/msnd.h index d0ca582c4583..61b3955481c5 100644 --- a/sound/oss/msnd.h +++ b/sound/oss/msnd.h @@ -24,8 +24,6 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. * - * $Id: msnd.h,v 1.36 1999/03/21 17:05:42 andrewtv Exp $ - * ********************************************************************/ #ifndef __MSND_H #define __MSND_H diff --git a/sound/oss/msnd_classic.h b/sound/oss/msnd_classic.h index 7ffea5267f96..1a17dde2f650 100644 --- a/sound/oss/msnd_classic.h +++ b/sound/oss/msnd_classic.h @@ -24,8 +24,6 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. * - * $Id: msnd_classic.h,v 1.10 1999/03/21 17:36:09 andrewtv Exp $ - * ********************************************************************/ #ifndef __MSND_CLASSIC_H #define __MSND_CLASSIC_H diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c index f1f49ebf752e..bf27e008f465 100644 --- a/sound/oss/msnd_pinnacle.c +++ b/sound/oss/msnd_pinnacle.c @@ -29,13 +29,8 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. * - * $Id: msnd_pinnacle.c,v 1.8 2000/12/30 00:33:21 sycamore Exp $ - * * 12-3-2000 Modified IO port validation Steve Sycamore * - * - * $$$: msnd_pinnacle.c,v 1.75 1999/03/21 16:50:09 andrewtv $$$ $ - * ********************************************************************/ #include diff --git a/sound/oss/msnd_pinnacle.h b/sound/oss/msnd_pinnacle.h index cce911487004..c18d66cbbe3f 100644 --- a/sound/oss/msnd_pinnacle.h +++ b/sound/oss/msnd_pinnacle.h @@ -24,8 +24,6 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. * - * $Id: msnd_pinnacle.h,v 1.11 1999/03/21 17:36:09 andrewtv Exp $ - * ********************************************************************/ #ifndef __MSND_PINNACLE_H #define __MSND_PINNACLE_H From 89fe5117928b2c1272c9376362131ded561c91ad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 May 2008 16:10:37 +0200 Subject: [PATCH 040/179] sound: Convert to menuconfig Convert menu in sound Kconfig files to menuconfig and if. Signed-off-by: Takashi Iwai --- sound/Kconfig | 34 ++++++--------- sound/aoa/Kconfig | 11 +++-- sound/aoa/codecs/Kconfig | 4 -- sound/aoa/fabrics/Kconfig | 1 - sound/aoa/soundbus/Kconfig | 1 - sound/arm/Kconfig | 21 ++++++--- sound/core/Kconfig | 19 ++------- sound/drivers/Kconfig | 87 ++++++++++++++++++-------------------- sound/isa/Kconfig | 51 ++++++++-------------- sound/mips/Kconfig | 14 ++++-- sound/parisc/Kconfig | 13 ++++-- sound/pci/Kconfig | 81 ++++++----------------------------- sound/pcmcia/Kconfig | 15 ++++--- sound/ppc/Kconfig | 26 +++++------- sound/sh/Kconfig | 16 +++++-- sound/soc/Kconfig | 17 ++++---- sound/soc/at91/Kconfig | 2 +- sound/soc/codecs/Kconfig | 10 +---- sound/soc/davinci/Kconfig | 2 +- sound/soc/fsl/Kconfig | 2 +- sound/soc/pxa/Kconfig | 2 +- sound/soc/s3c24xx/Kconfig | 4 +- sound/soc/sh/Kconfig | 5 +-- sound/sparc/Kconfig | 17 +++++--- sound/spi/Kconfig | 13 ++++-- sound/usb/Kconfig | 16 ++++--- 26 files changed, 211 insertions(+), 273 deletions(-) diff --git a/sound/Kconfig b/sound/Kconfig index 4247406160e7..a37bee094eba 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -1,11 +1,9 @@ # sound/Config.in # -menu "Sound" - depends on HAS_IOMEM - -config SOUND +menuconfig SOUND tristate "Sound card support" + depends on HAS_IOMEM help If you have a sound card in your computer, i.e. if it can say more than an occasional beep, say Y. Be sure to have all the information @@ -28,22 +26,22 @@ config SOUND and read ; the module will be called soundcore. +if SOUND + source "sound/oss/dmasound/Kconfig" if !M68K -menu "Advanced Linux Sound Architecture" - depends on SOUND!=n - -config SND +menuconfig SND tristate "Advanced Linux Sound Architecture" - depends on SOUND help Say 'Y' or 'M' to enable ALSA (Advanced Linux Sound Architecture), the new base sound system. For more information, see +if SND + source "sound/core/Kconfig" source "sound/drivers/Kconfig" @@ -58,9 +56,7 @@ source "sound/aoa/Kconfig" source "sound/arm/Kconfig" -if SPI source "sound/spi/Kconfig" -endif source "sound/mips/Kconfig" @@ -80,22 +76,20 @@ source "sound/parisc/Kconfig" source "sound/soc/Kconfig" -endmenu +endif # SND -menu "Open Sound System" - depends on SOUND!=n - -config SOUND_PRIME +menuconfig SOUND_PRIME tristate "Open Sound System (DEPRECATED)" - depends on SOUND help Say 'Y' or 'M' to enable Open Sound System drivers. +if SOUND_PRIME + source "sound/oss/Kconfig" -endmenu +endif # SOUND_PRIME -endif +endif # !M68K config AC97_BUS tristate @@ -105,4 +99,4 @@ config AC97_BUS sound although they're sharing the AC97 bus. Concerned drivers should "select" this. -endmenu +endif # SOUND diff --git a/sound/aoa/Kconfig b/sound/aoa/Kconfig index 5d5813cec4c8..c081e18b9540 100644 --- a/sound/aoa/Kconfig +++ b/sound/aoa/Kconfig @@ -1,18 +1,17 @@ -menu "Apple Onboard Audio driver" - depends on SND!=n && PPC_PMAC - -config SND_AOA +menuconfig SND_AOA tristate "Apple Onboard Audio driver" - depends on SND + depends on PPC_PMAC select SND_PCM ---help--- This option enables the new driver for the various Apple Onboard Audio components. +if SND_AOA + source "sound/aoa/fabrics/Kconfig" source "sound/aoa/codecs/Kconfig" source "sound/aoa/soundbus/Kconfig" -endmenu +endif # SND_AOA diff --git a/sound/aoa/codecs/Kconfig b/sound/aoa/codecs/Kconfig index d5fbd6016e93..808eb11ebacd 100644 --- a/sound/aoa/codecs/Kconfig +++ b/sound/aoa/codecs/Kconfig @@ -1,6 +1,5 @@ config SND_AOA_ONYX tristate "support Onyx chip" - depends on SND_AOA select I2C select I2C_POWERMAC ---help--- @@ -10,7 +9,6 @@ config SND_AOA_ONYX #config SND_AOA_TOPAZ # tristate "support Topaz chips" -# depends on SND_AOA # ---help--- # This option enables support for the Topaz (CS84xx) # codec chips found in the latest Apple machines, @@ -19,7 +17,6 @@ config SND_AOA_ONYX config SND_AOA_TAS tristate "support TAS chips" - depends on SND_AOA select I2C select I2C_POWERMAC ---help--- @@ -29,7 +26,6 @@ config SND_AOA_TAS config SND_AOA_TOONIE tristate "support Toonie chip" - depends on SND_AOA ---help--- This option enables support for the toonie codec found in the Mac Mini. If you have a Mac Mini and diff --git a/sound/aoa/fabrics/Kconfig b/sound/aoa/fabrics/Kconfig index 50d7021ff677..3ca475a886b1 100644 --- a/sound/aoa/fabrics/Kconfig +++ b/sound/aoa/fabrics/Kconfig @@ -1,6 +1,5 @@ config SND_AOA_FABRIC_LAYOUT tristate "layout-id fabric" - depends on SND_AOA select SND_AOA_SOUNDBUS select SND_AOA_SOUNDBUS_I2S ---help--- diff --git a/sound/aoa/soundbus/Kconfig b/sound/aoa/soundbus/Kconfig index 7368b7ddfe0d..839d1137b9b2 100644 --- a/sound/aoa/soundbus/Kconfig +++ b/sound/aoa/soundbus/Kconfig @@ -1,6 +1,5 @@ config SND_AOA_SOUNDBUS tristate "Apple Soundbus support" - depends on SOUND select SND_PCM ---help--- This option enables the generic driver for the soundbus diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 2e4a5e0d16db..351e19ea3785 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -1,11 +1,19 @@ # ALSA ARM drivers -menu "ALSA ARM devices" - depends on SND!=n && ARM +menuconfig SND_ARM + bool "ARM sound devices" + depends on ARM + default y + help + Support for sound devices specific to ARM architectures. + Drivers that are implemented on ASoC can be found in + "ALSA for SoC audio support" section. + +if SND_ARM config SND_SA11XX_UDA1341 tristate "SA11xx UDA1341TS driver (iPaq H3600)" - depends on ARCH_SA1100 && SND && L3 + depends on ARCH_SA1100 && L3 select SND_PCM help Say Y here if you have a Compaq iPaq H3x00 handheld computer @@ -16,7 +24,7 @@ config SND_SA11XX_UDA1341 config SND_ARMAACI tristate "ARM PrimeCell PL041 AC Link support" - depends on SND && ARM_AMBA + depends on ARM_AMBA select SND_PCM select SND_AC97_CODEC @@ -26,11 +34,12 @@ config SND_PXA2XX_PCM config SND_PXA2XX_AC97 tristate "AC97 driver for the Intel PXA2xx chip" - depends on ARCH_PXA && SND + depends on ARCH_PXA select SND_PXA2XX_PCM select SND_AC97_CODEC help Say Y or M if you want to support any AC97 codec attached to the PXA2xx AC97 interface. -endmenu +endif # SND_ARM + diff --git a/sound/core/Kconfig b/sound/core/Kconfig index a8d71c6c8e75..db2111375784 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -1,24 +1,19 @@ # ALSA soundcard-configuration config SND_TIMER tristate - depends on SND config SND_PCM tristate select SND_TIMER - depends on SND config SND_HWDEP tristate - depends on SND config SND_RAWMIDI tristate - depends on SND config SND_SEQUENCER tristate "Sequencer support" - depends on SND select SND_TIMER help Say Y or M to enable MIDI sequencer and router support. This @@ -44,11 +39,9 @@ config SND_SEQ_DUMMY config SND_OSSEMUL bool - depends on SND config SND_MIXER_OSS tristate "OSS Mixer API" - depends on SND select SND_OSSEMUL help To enable OSS mixer API emulation (/dev/mixer*), say Y here @@ -61,7 +54,6 @@ config SND_MIXER_OSS config SND_PCM_OSS tristate "OSS PCM (digital audio) API" - depends on SND select SND_OSSEMUL select SND_PCM help @@ -84,7 +76,7 @@ config SND_PCM_OSS_PLUGINS config SND_SEQUENCER_OSS bool "OSS Sequencer API" - depends on SND && SND_SEQUENCER + depends on SND_SEQUENCER select SND_OSSEMUL help Say Y here to enable OSS sequencer emulation (both @@ -98,7 +90,7 @@ config SND_SEQUENCER_OSS config SND_RTCTIMER tristate "RTC Timer support" - depends on SND && RTC + depends on RTC select SND_TIMER help Say Y here to enable RTC timer support for ALSA. ALSA uses @@ -123,7 +115,6 @@ config SND_SEQ_RTCTIMER_DEFAULT config SND_DYNAMIC_MINORS bool "Dynamic device file minor numbers" - depends on SND help If you say Y here, the minor numbers of ALSA device files in /dev/snd/ are allocated dynamically. This allows you to have @@ -134,7 +125,6 @@ config SND_DYNAMIC_MINORS config SND_SUPPORT_OLD_API bool "Support old ALSA API" - depends on SND default y help Say Y here to support the obsolete ALSA PCM API (ver.0.9.0 rc3 @@ -142,7 +132,7 @@ config SND_SUPPORT_OLD_API config SND_VERBOSE_PROCFS bool "Verbose procfs contents" - depends on SND && PROC_FS + depends on PROC_FS default y help Say Y here to include code for verbose procfs contents (provides @@ -151,7 +141,6 @@ config SND_VERBOSE_PROCFS config SND_VERBOSE_PRINTK bool "Verbose printk" - depends on SND help Say Y here to enable verbose log messages. These messages will help to identify source file and position containing @@ -161,7 +150,6 @@ config SND_VERBOSE_PRINTK config SND_DEBUG bool "Debug" - depends on SND help Say Y here to enable ALSA debug code. @@ -184,4 +172,3 @@ config SND_PCM_XRUN_DEBUG config SND_VMASTER bool - depends on SND diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 159137bf4c11..79b4d3f25665 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -1,40 +1,3 @@ -# ALSA generic drivers - -menu "Generic devices" - depends on SND!=n - - -config SND_PCSP - tristate "PC-Speaker support (READ HELP!)" - depends on PCSPKR_PLATFORM && X86_PC && HIGH_RES_TIMERS - depends on INPUT - depends on EXPERIMENTAL - depends on SND - select SND_PCM - help - If you don't have a sound card in your computer, you can include a - driver for the PC speaker which allows it to act like a primitive - sound card. - This driver also replaces the pcspkr driver for beeps. - - You can compile this as a module which will be called snd-pcsp. - - WARNING: if you already have a soundcard, enabling this - driver may lead to a problem. Namely, it may get loaded - before the other sound driver of yours, making the - pc-speaker a default sound device. Which is likely not - what you want. To make this driver play nicely with other - sound driver, you can add this into your /etc/modprobe.conf: - options snd-pcsp index=2 - - You don't need this driver if you only want your pc-speaker to beep. - You don't need this driver if you have a tablet piezo beeper - in your PC instead of the real speaker. - - Say N if you have a sound card. - Say M if you don't. - Say Y only if you really know what you do. - config SND_MPU401_UART tristate select SND_RAWMIDI @@ -60,9 +23,46 @@ config SND_AC97_CODEC select AC97_BUS select SND_VMASTER +menuconfig SND_DRIVERS + bool "Generic sound devices" + default y + help + Support for generic sound devices. + +if SND_DRIVERS + +config SND_PCSP + tristate "PC-Speaker support (READ HELP!)" + depends on PCSPKR_PLATFORM && X86_PC && HIGH_RES_TIMERS + depends on INPUT + depends on EXPERIMENTAL + select SND_PCM + help + If you don't have a sound card in your computer, you can include a + driver for the PC speaker which allows it to act like a primitive + sound card. + This driver also replaces the pcspkr driver for beeps. + + You can compile this as a module which will be called snd-pcsp. + + WARNING: if you already have a soundcard, enabling this + driver may lead to a problem. Namely, it may get loaded + before the other sound driver of yours, making the + pc-speaker a default sound device. Which is likely not + what you want. To make this driver play nicely with other + sound driver, you can add this into your /etc/modprobe.conf: + options snd-pcsp index=2 + + You don't need this driver if you only want your pc-speaker to beep. + You don't need this driver if you have a tablet piezo beeper + in your PC instead of the real speaker. + + Say N if you have a sound card. + Say M if you don't. + Say Y only if you really know what you do. + config SND_DUMMY tristate "Dummy (/dev/null) soundcard" - depends on SND select SND_PCM help Say Y here to include the dummy driver. This driver does @@ -91,7 +91,6 @@ config SND_VIRMIDI config SND_MTPAV tristate "MOTU MidiTimePiece AV multiport MIDI" - depends on SND select SND_RAWMIDI help To use a MOTU MidiTimePiece AV multiport MIDI adapter @@ -103,7 +102,7 @@ config SND_MTPAV config SND_MTS64 tristate "ESI Miditerminal 4140 driver" - depends on SND && PARPORT + depends on PARPORT select SND_RAWMIDI help The ESI Miditerminal 4140 is a 4 In 4 Out MIDI Interface with @@ -116,7 +115,6 @@ config SND_MTS64 config SND_SERIAL_U16550 tristate "UART16550 serial MIDI driver" - depends on SND select SND_RAWMIDI help To include support for MIDI serial port interfaces, say Y here @@ -132,7 +130,6 @@ config SND_SERIAL_U16550 config SND_MPU401 tristate "Generic MPU-401 UART driver" - depends on SND select SND_MPU401_UART help Say Y here to include support for MIDI ports compatible with @@ -143,7 +140,7 @@ config SND_MPU401 config SND_PORTMAN2X4 tristate "Portman 2x4 driver" - depends on SND && PARPORT + depends on PARPORT select SND_RAWMIDI help Say Y here to include support for Midiman Portman 2x4 parallel @@ -154,7 +151,7 @@ config SND_PORTMAN2X4 config SND_ML403_AC97CR tristate "Xilinx ML403 AC97 Controller Reference" - depends on SND && XILINX_VIRTEX + depends on XILINX_VIRTEX select SND_AC97_CODEC help Say Y here to include support for the @@ -164,4 +161,4 @@ config SND_ML403_AC97CR To compile this driver as a module, choose M here: the module will be called snd-ml403_ac97cr. -endmenu +endif # SND_DRIVERS diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 2639a6ab8f2e..4575ba865910 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -21,12 +21,17 @@ config SND_SB16_DSP select SND_PCM select SND_SB_COMMON -menu "ISA devices" - depends on SND!=n && ISA && ISA_DMA_API +menuconfig SND_ISA + bool "ISA sound devices" + depends on ISA && ISA_DMA_API + default y + help + Support for sound devices connected via the ISA bus. + +if SND_ISA config SND_ADLIB tristate "AdLib FM card" - depends on SND select SND_OPL3_LIB help Say Y here to include support for AdLib FM cards. @@ -36,7 +41,7 @@ config SND_ADLIB config SND_AD1816A tristate "Analog Devices SoundPort AD1816A" - depends on SND && PNP && ISA + depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -50,7 +55,6 @@ config SND_AD1816A config SND_AD1848 tristate "Generic AD1848/CS4248 driver" - depends on SND select SND_AD1848_LIB help Say Y here to include support for AD1848 (Analog Devices) or @@ -64,7 +68,7 @@ config SND_AD1848 config SND_ALS100 tristate "Avance Logic ALS100/ALS120" - depends on SND && PNP && ISA + depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -78,7 +82,7 @@ config SND_ALS100 config SND_AZT2320 tristate "Aztech Systems AZT2320" - depends on SND && PNP && ISA + depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -92,7 +96,6 @@ config SND_AZT2320 config SND_CMI8330 tristate "C-Media CMI8330" - depends on SND select SND_AD1848_LIB select SND_SB16_DSP help @@ -104,7 +107,6 @@ config SND_CMI8330 config SND_CS4231 tristate "Generic Cirrus Logic CS4231 driver" - depends on SND select SND_MPU401_UART select SND_CS4231_LIB help @@ -116,7 +118,6 @@ config SND_CS4231 config SND_CS4232 tristate "Generic Cirrus Logic CS4232 driver" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_CS4231_LIB @@ -129,7 +130,6 @@ config SND_CS4232 config SND_CS4236 tristate "Generic Cirrus Logic CS4236+ driver" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_CS4231_LIB @@ -142,7 +142,7 @@ config SND_CS4236 config SND_DT019X tristate "Diamond Technologies DT-019X, Avance Logic ALS-007" - depends on SND && PNP && ISA + depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -156,7 +156,7 @@ config SND_DT019X config SND_ES968 tristate "Generic ESS ES968 driver" - depends on SND && PNP && ISA + depends on PNP select ISAPNP select SND_MPU401_UART select SND_SB8_DSP @@ -168,7 +168,6 @@ config SND_ES968 config SND_ES1688 tristate "Generic ESS ES688/ES1688 driver" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -181,7 +180,6 @@ config SND_ES1688 config SND_ES18XX tristate "Generic ESS ES18xx driver" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -193,7 +191,7 @@ config SND_ES18XX config SND_SC6000 tristate "Gallant SC-6000, Audio Excel DSP 16" - depends on SND && HAS_IOPORT + depends on HAS_IOPORT select SND_AD1848_LIB select SND_OPL3_LIB select SND_MPU401_UART @@ -209,7 +207,6 @@ config SND_GUS_SYNTH config SND_GUSCLASSIC tristate "Gravis UltraSound Classic" - depends on SND select SND_RAWMIDI select SND_PCM select SND_GUS_SYNTH @@ -222,7 +219,6 @@ config SND_GUSCLASSIC config SND_GUSEXTREME tristate "Gravis UltraSound Extreme" - depends on SND select SND_HWDEP select SND_MPU401_UART select SND_PCM @@ -236,7 +232,6 @@ config SND_GUSEXTREME config SND_GUSMAX tristate "Gravis UltraSound MAX" - depends on SND select SND_RAWMIDI select SND_CS4231_LIB select SND_GUS_SYNTH @@ -249,7 +244,7 @@ config SND_GUSMAX config SND_INTERWAVE tristate "AMD InterWave, Gravis UltraSound PnP" - depends on SND && PNP && ISA + depends on PNP select SND_RAWMIDI select SND_CS4231_LIB select SND_GUS_SYNTH @@ -263,7 +258,7 @@ config SND_INTERWAVE config SND_INTERWAVE_STB tristate "AMD InterWave + TEA6330T (UltraSound 32-Pro)" - depends on SND && PNP && ISA + depends on PNP select SND_RAWMIDI select SND_CS4231_LIB select SND_GUS_SYNTH @@ -277,7 +272,6 @@ config SND_INTERWAVE_STB config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_CS4231_LIB @@ -290,7 +284,6 @@ config SND_OPL3SA2 config SND_OPTI92X_AD1848 tristate "OPTi 82C92x - AD1848" - depends on SND select SND_OPL3_LIB select SND_OPL4_LIB select SND_MPU401_UART @@ -304,7 +297,6 @@ config SND_OPTI92X_AD1848 config SND_OPTI92X_CS4231 tristate "OPTi 82C92x - CS4231" - depends on SND select SND_OPL3_LIB select SND_OPL4_LIB select SND_MPU401_UART @@ -318,7 +310,6 @@ config SND_OPTI92X_CS4231 config SND_OPTI93X tristate "OPTi 82C93x" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -331,7 +322,6 @@ config SND_OPTI93X config SND_MIRO tristate "Miro miroSOUND PCM1pro/PCM12/PCM20radio driver" - depends on SND select SND_OPL4_LIB select SND_CS4231_LIB select SND_MPU401_UART @@ -345,7 +335,6 @@ config SND_MIRO config SND_SB8 tristate "Sound Blaster 1.0/2.0/Pro (8-bit)" - depends on SND select SND_OPL3_LIB select SND_RAWMIDI select SND_SB8_DSP @@ -358,7 +347,6 @@ config SND_SB8 config SND_SB16 tristate "Sound Blaster 16 (PnP)" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_SB16_DSP @@ -371,7 +359,6 @@ config SND_SB16 config SND_SBAWE tristate "Sound Blaster AWE (32,64) (PnP)" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_SB16_DSP @@ -402,7 +389,6 @@ config SND_SB16_CSP_FIRMWARE_IN_KERNEL config SND_SGALAXY tristate "Aztech Sound Galaxy" - depends on SND select SND_AD1848_LIB help Say Y here to include support for Aztech Sound Galaxy @@ -413,7 +399,6 @@ config SND_SGALAXY config SND_SSCAPE tristate "Ensoniq SoundScape PnP driver" - depends on SND select SND_HWDEP select SND_MPU401_UART select SND_CS4231_LIB @@ -426,7 +411,6 @@ config SND_SSCAPE config SND_WAVEFRONT tristate "Turtle Beach Maui,Tropez,Tropez+ (Wavefront)" - depends on SND select FW_LOADER select SND_OPL3_LIB select SND_MPU401_UART @@ -448,4 +432,5 @@ config SND_WAVEFRONT_FIRMWARE_IN_KERNEL you need to install the firmware files from the alsa-firmware package. -endmenu +endif # SND_ISA + diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 531f8ba96a71..bb26f6cf4c0a 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -1,15 +1,21 @@ # ALSA MIPS drivers -menu "ALSA MIPS devices" - depends on SND!=n && MIPS +menuconfig SND_MIPS + bool "MIPS sound devices" + depends on MIPS + default y + help + Support for sound devices of MIPS architectures. + +if SND_MIPS config SND_AU1X00 tristate "Au1x00 AC97 Port Driver" - depends on (SOC_AU1000 || SOC_AU1100 || SOC_AU1500) && SND + depends on SOC_AU1000 || SOC_AU1100 || SOC_AU1500 select SND_PCM select SND_AC97_CODEC help ALSA Sound driver for the Au1x00's AC97 port. -endmenu +endif # SND_MIPS diff --git a/sound/parisc/Kconfig b/sound/parisc/Kconfig index a5a7f9d75d05..9b61d95010f0 100644 --- a/sound/parisc/Kconfig +++ b/sound/parisc/Kconfig @@ -1,15 +1,20 @@ # ALSA PA-RISC drivers -menu "GSC devices" - depends on SND!=n && GSC +menuconfig SND_GSC + bool "GSC sound devices" + depends on GSC + default y + help + Support for GSC sound devices on PA-RISC architectures. + +if SND_GSC config SND_HARMONY tristate "Harmony/Vivace sound chip" - depends on SND select SND_PCM help Say 'Y' or 'M' to include support for the Harmony/Vivace sound chip found in most GSC-based PA-RISC workstations. It's frequently provided as part of the Lasi multi-function IC. -endmenu +endif # SND_GSC diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index d95fbb2b5b9f..b148c0b3ef3f 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -1,11 +1,16 @@ # ALSA PCI drivers -menu "PCI devices" - depends on SND!=n && PCI +menuconfig SND_PCI + bool "PCI sound devices" + depends on PCI + default y + help + Support for sound devices connected via the PCI bus. + +if SND_PCI config SND_AD1889 tristate "Analog Devices AD1889" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated AC97 sound @@ -17,7 +22,6 @@ config SND_AD1889 config SND_ALS300 tristate "Avance Logic ALS300/ALS300+" - depends on SND select SND_PCM select SND_AC97_CODEC select SND_OPL3_LIB @@ -29,7 +33,7 @@ config SND_ALS300 config SND_ALS4000 tristate "Avance Logic ALS4000" - depends on SND && ISA_DMA_API + depends on ISA_DMA_API select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -43,7 +47,6 @@ config SND_ALS4000 config SND_ALI5451 tristate "ALi M5451 PCI Audio Controller" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -57,7 +60,6 @@ config SND_ALI5451 config SND_ATIIXP tristate "ATI IXP AC97 Controller" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated AC97 sound @@ -69,7 +71,6 @@ config SND_ATIIXP config SND_ATIIXP_MODEM tristate "ATI IXP Modem" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated MC97 modem on @@ -80,7 +81,6 @@ config SND_ATIIXP_MODEM config SND_AU8810 tristate "Aureal Advantage" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -95,7 +95,6 @@ config SND_AU8810 config SND_AU8820 tristate "Aureal Vortex" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -109,7 +108,6 @@ config SND_AU8820 config SND_AU8830 tristate "Aureal Vortex 2" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -124,7 +122,6 @@ config SND_AU8830 config SND_AW2 tristate "Emagic Audiowerk 2" - depends on SND help Say Y here to include support for Emagic Audiowerk 2 soundcards. @@ -139,7 +136,7 @@ config SND_AW2 config SND_AZT3328 tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)" - depends on SND && EXPERIMENTAL + depends on EXPERIMENTAL select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -152,7 +149,6 @@ config SND_AZT3328 config SND_BT87X tristate "Bt87x Audio Capture" - depends on SND select SND_PCM help If you want to record audio from TV cards based on @@ -174,7 +170,6 @@ config SND_BT87X_OVERCLOCK config SND_CA0106 tristate "SB Audigy LS / Live 24bit" - depends on SND select SND_AC97_CODEC select SND_RAWMIDI select SND_VMASTER @@ -187,7 +182,6 @@ config SND_CA0106 config SND_CMIPCI tristate "C-Media 8338, 8738, 8768, 8770" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -201,13 +195,11 @@ config SND_CMIPCI config SND_OXYGEN_LIB tristate - depends on SND select SND_PCM select SND_MPU401_UART config SND_OXYGEN tristate "C-Media 8788 (Oxygen)" - depends on SND select SND_OXYGEN_LIB help Say Y here to include support for sound cards based on the @@ -225,7 +217,6 @@ config SND_OXYGEN config SND_CS4281 tristate "Cirrus Logic (Sound Fusion) CS4281" - depends on SND select SND_OPL3_LIB select SND_RAWMIDI select SND_AC97_CODEC @@ -237,7 +228,6 @@ config SND_CS4281 config SND_CS46XX tristate "Cirrus Logic (Sound Fusion) CS4280/CS461x/CS462x/CS463x" - depends on SND select SND_RAWMIDI select SND_AC97_CODEC help @@ -258,7 +248,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" - depends on SND && ISA_DMA_API + depends on ISA_DMA_API select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. @@ -268,7 +258,7 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" - depends on SND && X86 && !X86_64 + depends on X86 && !X86_64 select SND_PCM select SND_AC97_CODEC help @@ -286,7 +276,6 @@ config SND_CS5535AUDIO config SND_DARLA20 tristate "(Echoaudio) Darla20" - depends on SND select FW_LOADER select SND_PCM help @@ -297,7 +286,6 @@ config SND_DARLA20 config SND_GINA20 tristate "(Echoaudio) Gina20" - depends on SND select FW_LOADER select SND_PCM help @@ -308,7 +296,6 @@ config SND_GINA20 config SND_LAYLA20 tristate "(Echoaudio) Layla20" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -320,7 +307,6 @@ config SND_LAYLA20 config SND_DARLA24 tristate "(Echoaudio) Darla24" - depends on SND select FW_LOADER select SND_PCM help @@ -331,7 +317,6 @@ config SND_DARLA24 config SND_GINA24 tristate "(Echoaudio) Gina24" - depends on SND select FW_LOADER select SND_PCM help @@ -342,7 +327,6 @@ config SND_GINA24 config SND_LAYLA24 tristate "(Echoaudio) Layla24" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -354,7 +338,6 @@ config SND_LAYLA24 config SND_MONA tristate "(Echoaudio) Mona" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -366,7 +349,6 @@ config SND_MONA config SND_MIA tristate "(Echoaudio) Mia" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -378,7 +360,6 @@ config SND_MIA config SND_ECHO3G tristate "(Echoaudio) 3G cards" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -390,7 +371,6 @@ config SND_ECHO3G config SND_INDIGO tristate "(Echoaudio) Indigo" - depends on SND select FW_LOADER select SND_PCM help @@ -401,7 +381,6 @@ config SND_INDIGO config SND_INDIGOIO tristate "(Echoaudio) Indigo IO" - depends on SND select FW_LOADER select SND_PCM help @@ -412,7 +391,6 @@ config SND_INDIGOIO config SND_INDIGODJ tristate "(Echoaudio) Indigo DJ" - depends on SND select FW_LOADER select SND_PCM help @@ -423,7 +401,6 @@ config SND_INDIGODJ config SND_EMU10K1 tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" - depends on SND select FW_LOADER select SND_HWDEP select SND_RAWMIDI @@ -441,7 +418,6 @@ config SND_EMU10K1 config SND_EMU10K1X tristate "Emu10k1X (Dell OEM Version)" - depends on SND select SND_AC97_CODEC select SND_RAWMIDI help @@ -453,7 +429,6 @@ config SND_EMU10K1X config SND_ENS1370 tristate "(Creative) Ensoniq AudioPCI 1370" - depends on SND select SND_RAWMIDI select SND_PCM help @@ -464,7 +439,6 @@ config SND_ENS1370 config SND_ENS1371 tristate "(Creative) Ensoniq AudioPCI 1371/1373" - depends on SND select SND_RAWMIDI select SND_AC97_CODEC help @@ -476,7 +450,6 @@ config SND_ENS1371 config SND_ES1938 tristate "ESS ES1938/1946/1969 (Solo-1)" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC @@ -489,7 +462,6 @@ config SND_ES1938 config SND_ES1968 tristate "ESS ES1968/1978 (Maestro-1/2/2E)" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -501,7 +473,6 @@ config SND_ES1968 config SND_FM801 tristate "ForteMedia FM801" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC @@ -528,7 +499,6 @@ config SND_FM801_TEA575X config SND_HDA_INTEL tristate "Intel HD Audio" - depends on SND select SND_PCM select SND_VMASTER help @@ -637,7 +607,6 @@ config SND_HDA_POWER_SAVE_DEFAULT config SND_HDSP tristate "RME Hammerfall DSP Audio" - depends on SND select SND_HWDEP select SND_RAWMIDI select SND_PCM @@ -650,7 +619,6 @@ config SND_HDSP config SND_HDSPM tristate "RME Hammerfall DSP MADI" - depends on SND select SND_HWDEP select SND_RAWMIDI select SND_PCM @@ -663,7 +631,6 @@ config SND_HDSPM config SND_HIFIER tristate "TempoTec HiFier Fantasia" - depends on SND select SND_OXYGEN_LIB help Say Y here to include support for the MediaTek/TempoTec HiFier @@ -674,7 +641,6 @@ config SND_HIFIER config SND_ICE1712 tristate "ICEnsemble ICE1712 (Envy24)" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -691,7 +657,6 @@ config SND_ICE1712 config SND_ICE1724 tristate "ICE/VT1724/1720 (Envy24HT/PT)" - depends on SND select SND_RAWMIDI select SND_AC97_CODEC select SND_VMASTER @@ -709,7 +674,6 @@ config SND_ICE1724 config SND_INTEL8X0 tristate "Intel/SiS/nVidia/AMD/ALi AC97 Controller" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated AC97 sound @@ -722,7 +686,6 @@ config SND_INTEL8X0 config SND_INTEL8X0M tristate "Intel/SiS/nVidia/AMD MC97 Modem" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated MC97 modem on @@ -733,7 +696,6 @@ config SND_INTEL8X0M config SND_KORG1212 tristate "Korg 1212 IO" - depends on SND select FW_LOADER if !SND_KORG1212_FIRMWARE_IN_KERNEL select SND_PCM help @@ -753,7 +715,6 @@ config SND_KORG1212_FIRMWARE_IN_KERNEL config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" - depends on SND select FW_LOADER if !SND_MAESTRO3_FIRMWARE_IN_KERNEL select SND_AC97_CODEC help @@ -774,7 +735,6 @@ config SND_MAESTRO3_FIRMWARE_IN_KERNEL config SND_MIXART tristate "Digigram miXart" - depends on SND select SND_HWDEP select SND_PCM help @@ -786,7 +746,6 @@ config SND_MIXART config SND_NM256 tristate "NeoMagic NM256AV/ZX" - depends on SND select SND_AC97_CODEC help Say Y here to include support for NeoMagic NM256AV/ZX chips. @@ -796,7 +755,6 @@ config SND_NM256 config SND_PCXHR tristate "Digigram PCXHR" - depends on SND select SND_PCM select SND_HWDEP help @@ -807,7 +765,6 @@ config SND_PCXHR config SND_RIPTIDE tristate "Conexant Riptide" - depends on SND select FW_LOADER select SND_OPL3_LIB select SND_MPU401_UART @@ -820,7 +777,6 @@ config SND_RIPTIDE config SND_RME32 tristate "RME Digi32, 32/8, 32 PRO" - depends on SND select SND_PCM help Say Y to include support for RME Digi32, Digi32 PRO and @@ -832,7 +788,6 @@ config SND_RME32 config SND_RME96 tristate "RME Digi96, 96/8, 96/8 PRO" - depends on SND select SND_PCM help Say Y here to include support for RME Digi96, Digi96/8 and @@ -843,7 +798,6 @@ config SND_RME96 config SND_RME9652 tristate "RME Digi9652 (Hammerfall)" - depends on SND select SND_PCM help Say Y here to include support for RME Hammerfall (RME @@ -854,7 +808,7 @@ config SND_RME9652 config SND_SIS7019 tristate "SiS 7019 Audio Accelerator" - depends on SND && X86 && !X86_64 + depends on X86 && !X86_64 select SND_AC97_CODEC help Say Y here to include support for the SiS 7019 Audio Accelerator. @@ -864,7 +818,6 @@ config SND_SIS7019 config SND_SONICVIBES tristate "S3 SonicVibes" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC @@ -877,7 +830,6 @@ config SND_SONICVIBES config SND_TRIDENT tristate "Trident 4D-Wave DX/NX; SiS 7018" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -889,7 +841,6 @@ config SND_TRIDENT config SND_VIA82XX tristate "VIA 82C686A/B, 8233/8235 AC97 Controller" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -901,7 +852,6 @@ config SND_VIA82XX config SND_VIA82XX_MODEM tristate "VIA 82C686A/B, 8233 based Modems" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated MC97 modem on @@ -912,7 +862,6 @@ config SND_VIA82XX_MODEM config SND_VIRTUOSO tristate "Asus Virtuoso 100/200 (Xonar)" - depends on SND select SND_OXYGEN_LIB help Say Y here to include support for sound cards based on the @@ -923,7 +872,6 @@ config SND_VIRTUOSO config SND_VX222 tristate "Digigram VX222" - depends on SND select SND_VX_LIB help Say Y here to include support for Digigram VX222 soundcards. @@ -933,7 +881,6 @@ config SND_VX222 config SND_YMFPCI tristate "Yamaha YMF724/740/744/754" - depends on SND select FW_LOADER if !SND_YMFPCI_FIRMWARE_IN_KERNEL select SND_OPL3_LIB select SND_MPU401_UART @@ -975,4 +922,4 @@ config SND_AC97_POWER_SAVE_DEFAULT The default time-out value in seconds for AC97 automatic power-save mode. 0 means to disable the power-save mode. -endmenu +endif # SND_PCI diff --git a/sound/pcmcia/Kconfig b/sound/pcmcia/Kconfig index c9fa1a2bc58b..7fbb190adf6d 100644 --- a/sound/pcmcia/Kconfig +++ b/sound/pcmcia/Kconfig @@ -1,11 +1,16 @@ # ALSA PCMCIA drivers -menu "PCMCIA devices" - depends on SND!=n && PCMCIA +menuconfig SND_PCMCIA + bool "PCMCIA sound devices" + depends on PCMCIA + default y + help + Support for sound devices connected via the PCMCIA bus. + +if SND_PCMCIA && PCMCIA config SND_VXPOCKET tristate "Digigram VXpocket" - depends on SND && PCMCIA select SND_VX_LIB help Say Y here to include support for Digigram VXpocket and @@ -16,7 +21,6 @@ config SND_VXPOCKET config SND_PDAUDIOCF tristate "Sound Core PDAudioCF" - depends on SND && PCMCIA select SND_PCM help Say Y here to include support for Sound Core PDAudioCF @@ -25,4 +29,5 @@ config SND_PDAUDIOCF To compile this driver as a module, choose M here: the module will be called snd-pdaudiocf. -endmenu +endif # SND_PCMCIA + diff --git a/sound/ppc/Kconfig b/sound/ppc/Kconfig index cacb0b136883..777de2b17178 100644 --- a/sound/ppc/Kconfig +++ b/sound/ppc/Kconfig @@ -1,17 +1,17 @@ # ALSA PowerMac drivers -menu "ALSA PowerMac devices" - depends on SND!=n && PPC +menuconfig SND_PPC + bool "PowerPC sound devices" + depends on PPC64 || PPC32 + default y + help + Support for sound devices specific to PowerPC architectures. -comment "ALSA PowerMac requires I2C" - depends on SND && I2C=n - -comment "ALSA PowerMac requires INPUT" - depends on SND && INPUT=n +if SND_PPC config SND_POWERMAC tristate "PowerMac (AWACS, DACA, Burgundy, Tumbler, Keywest)" - depends on SND && I2C && INPUT && PPC_PMAC + depends on I2C && INPUT && PPC_PMAC select SND_PCM help Say Y here to include support for the integrated sound device. @@ -32,14 +32,9 @@ config SND_POWERMAC_AUTO_DRC Note that you can turn on/off DRC manually even without this option. -endmenu - -menu "ALSA PowerPC devices" - depends on SND!=n && ( PPC64 || PPC32 ) - config SND_PS3 tristate "PS3 Audio support" - depends on SND && PS3_PS3AV + depends on PS3_PS3AV select SND_PCM default m help @@ -52,4 +47,5 @@ config SND_PS3_DEFAULT_START_DELAY int "Startup delay time in ms" depends on SND_PS3 default "2000" -endmenu + +endif # SND_PPC diff --git a/sound/sh/Kconfig b/sound/sh/Kconfig index b7e08ef22a94..cfc143985802 100644 --- a/sound/sh/Kconfig +++ b/sound/sh/Kconfig @@ -1,14 +1,22 @@ # ALSA SH drivers -menu "SUPERH devices" - depends on SND!=n && SUPERH +menuconfig SND_SUPERH + bool "SUPERH sound devices" + depends on SUPERH + default y + help + Support for sound devices specific to SUPERH architectures. + Drivers that are implemented on ASoC can be found in + "ALSA for SoC audio support" section. + +if SND_SUPERH config SND_AICA tristate "Dreamcast Yamaha AICA sound" - depends on SH_DREAMCAST && SND + depends on SH_DREAMCAST select SND_PCM help ALSA Sound driver for the SEGA Dreamcast console. -endmenu +endif # SND_SUPERH diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 18f28ac4bfe8..fd7bc4f89072 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -2,15 +2,8 @@ # SoC audio configuration # -menu "System on Chip audio support" - depends on SND!=n - -config SND_SOC_AC97_BUS - bool - -config SND_SOC +menuconfig SND_SOC tristate "ALSA for SoC audio support" - depends on SND select SND_PCM ---help--- @@ -23,6 +16,11 @@ config SND_SOC This ASoC audio support can also be built as a module. If so, the module will be called snd-soc-core. +if SND_SOC + +config SND_SOC_AC97_BUS + bool + # All the supported Soc's source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" @@ -35,4 +33,5 @@ source "sound/soc/omap/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" -endmenu +endif # SND_SOC + diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig index 5cb93fd3a407..905186502e00 100644 --- a/sound/soc/at91/Kconfig +++ b/sound/soc/at91/Kconfig @@ -1,6 +1,6 @@ config SND_AT91_SOC tristate "SoC Audio for the Atmel AT91 System-on-Chip" - depends on ARCH_AT91 && SND_SOC + depends on ARCH_AT91 help Say Y or M if you want to add support for codecs attached to the AT91 SSC interface. You will also need diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3903ab7dfa4a..d4a5fe42f6e0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1,31 +1,25 @@ config SND_SOC_AC97_CODEC tristate - depends on SND_SOC + select SND_AC97_CODEC config SND_SOC_WM8731 tristate - depends on SND_SOC config SND_SOC_WM8750 tristate - depends on SND_SOC config SND_SOC_WM8753 tristate - depends on SND_SOC config SND_SOC_WM9712 tristate - depends on SND_SOC config SND_SOC_WM9713 tristate - depends on SND_SOC # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate - depends on SND_SOC # Cirrus Logic CS4270 Codec Hardware Mute Support # Select if you have external muting circuitry attached to your CS4270. @@ -43,4 +37,4 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_TLV320AIC3X tristate - depends on SND_SOC && I2C + depends on I2C diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 20680c551aab..8f7e33834902 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,6 +1,6 @@ config SND_DAVINCI_SOC tristate "SoC Audio for the TI DAVINCI chip" - depends on ARCH_DAVINCI && SND_SOC + depends on ARCH_DAVINCI help Say Y or M if you want to add support for codecs attached to the DAVINCI AC97 or I2S interface. You will also need diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 257101f44e9e..19802e27df4b 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -2,7 +2,7 @@ menu "ALSA SoC audio for Freescale SOCs" config SND_SOC_MPC8610 bool "ALSA SoC support for the MPC8610 SOC" - depends on SND_SOC && MPC8610_HPCD + depends on MPC8610_HPCD default y if MPC8610 help Say Y if you want to add support for codecs attached to the SSI diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 484f883459e0..18a40dc8ad9d 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,6 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" - depends on ARCH_PXA && SND_SOC + depends on ARCH_PXA help Say Y or M if you want to add support for codecs attached to the PXA2xx AC97, I2S or SSP interface. You will also need diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 1f6dbfc4caa8..b9f2353effeb 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,7 +1,6 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3C24XX chips" - depends on ARCH_S3C2410 && SND_SOC - select SND_PCM + depends on ARCH_S3C2410 help Say Y or M if you want to add support for codecs attached to the S3C24XX AC97, I2S or SSP interface. You will also need @@ -16,7 +15,6 @@ config SND_S3C2412_SOC_I2S config SND_S3C2443_SOC_AC97 tristate select AC97_BUS - select SND_AC97_CODEC select SND_SOC_AC97_BUS config SND_S3C24XX_SOC_NEO1973_WM8753 diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 4c1e013381c9..54bd604012af 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -3,7 +3,7 @@ menu "SoC Audio support for SuperH" config SND_SOC_PCM_SH7760 tristate "SoC Audio support for Renesas SH7760" - depends on CPU_SUBTYPE_SH7760 && SND_SOC && SH_DMABRG + depends on CPU_SUBTYPE_SH7760 && SH_DMABRG help Enable this option for SH7760 AC97/I2S audio support. @@ -13,10 +13,9 @@ config SND_SOC_PCM_SH7760 ## config SND_SOC_SH4_HAC + tristate select AC97_BUS select SND_SOC_AC97_BUS - select SND_AC97_CODEC - tristate config SND_SOC_SH4_SSI tristate diff --git a/sound/sparc/Kconfig b/sound/sparc/Kconfig index 079e22af074c..d75deba5617d 100644 --- a/sound/sparc/Kconfig +++ b/sound/sparc/Kconfig @@ -1,11 +1,17 @@ # ALSA Sparc drivers -menu "ALSA Sparc devices" - depends on SND!=n && SPARC +menuconfig SND_SPARC + bool "Sparc sound devices" + depends on SPARC + default y + help + Support for sound devices specific to Sun SPARC architectures. + +if SND_SPARC config SND_SUN_AMD7930 tristate "Sun AMD7930" - depends on SBUS && SND + depends on SBUS select SND_PCM help Say Y here to include support for AMD7930 sound device on Sun. @@ -15,7 +21,6 @@ config SND_SUN_AMD7930 config SND_SUN_CS4231 tristate "Sun CS4231" - depends on SND select SND_PCM help Say Y here to include support for CS4231 sound device on Sun. @@ -25,7 +30,7 @@ config SND_SUN_CS4231 config SND_SUN_DBRI tristate "Sun DBRI" - depends on SND && SBUS + depends on SBUS select SND_PCM help Say Y here to include support for DBRI sound device on Sun. @@ -33,4 +38,4 @@ config SND_SUN_DBRI To compile this driver as a module, choose M here: the module will be called snd-sun-dbri. -endmenu +endif # SND_SPARC diff --git a/sound/spi/Kconfig b/sound/spi/Kconfig index 0d08c29213c8..e6485be2e6f7 100644 --- a/sound/spi/Kconfig +++ b/sound/spi/Kconfig @@ -1,7 +1,13 @@ #SPI drivers -menu "SPI devices" - depends on SND != n +menuconfig SND_SPI + bool "SPI sound devices" + depends on SPI + default y + help + Support for sound devices connected via the SPI bus. + +if SND_SPI config SND_AT73C213 tristate "Atmel AT73C213 DAC driver" @@ -28,4 +34,5 @@ config SND_AT73C213_TARGET_BITRATE Set to 48000 Hz by default. -endmenu +endif # SND_SPI + diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 9351b8a765b9..ffcdc8f4ef66 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -1,11 +1,16 @@ # ALSA USB drivers -menu "USB devices" - depends on SND!=n && USB!=n +menuconfig SND_USB + bool "USB sound devices" + depends on USB + default y + help + Support for sound devices connected via the USB bus. + +if SND_USB && USB config SND_USB_AUDIO tristate "USB Audio/MIDI driver" - depends on SND && USB select SND_HWDEP select SND_RAWMIDI select SND_PCM @@ -18,7 +23,7 @@ config SND_USB_AUDIO config SND_USB_USX2Y tristate "Tascam US-122, US-224 and US-428 USB driver" - depends on SND && USB && (X86 || PPC || ALPHA) + depends on X86 || PPC || ALPHA select SND_HWDEP select SND_RAWMIDI select SND_PCM @@ -31,7 +36,6 @@ config SND_USB_USX2Y config SND_USB_CAIAQ tristate "Native Instruments USB audio devices" - depends on SND && USB select SND_HWDEP select SND_RAWMIDI select SND_PCM @@ -63,5 +67,5 @@ config SND_USB_CAIAQ_INPUT * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 -endmenu +endif # SND_USB From 6938d6b2a90d5e2ffaef037852652a1333502519 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 May 2008 16:11:26 +0200 Subject: [PATCH 041/179] [ALSA] Fix AC97 kconfig items The kconfig items related with AC97-powersave must be outside the CONFIG_SND_PCI range. And it'd be better together with CONFIG_SND_AC97. Signed-off-by: Takashi Iwai --- sound/drivers/Kconfig | 21 +++++++++++++++++++++ sound/pci/Kconfig | 21 --------------------- 2 files changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 79b4d3f25665..255fd18b9aec 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -161,4 +161,25 @@ config SND_ML403_AC97CR To compile this driver as a module, choose M here: the module will be called snd-ml403_ac97cr. +config SND_AC97_POWER_SAVE + bool "AC97 Power-Saving Mode" + depends on SND_AC97_CODEC && EXPERIMENTAL + default n + help + Say Y here to enable the aggressive power-saving support of + AC97 codecs. In this mode, the power-mode is dynamically + controlled at each open/close. + + The mode is activated by passing power_save=1 option to + snd-ac97-codec driver. You can toggle it dynamically over + sysfs, too. + +config SND_AC97_POWER_SAVE_DEFAULT + int "Default time-out for AC97 power-save mode" + depends on SND_AC97_POWER_SAVE + default 0 + help + The default time-out value in seconds for AC97 automatic + power-save mode. 0 means to disable the power-save mode. + endif # SND_DRIVERS diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index b148c0b3ef3f..8fe5dac39428 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -901,25 +901,4 @@ config SND_YMFPCI_FIRMWARE_IN_KERNEL for the YMFPCI driver. If you choose N here, you need to install the firmware files from the alsa-firmware package. -config SND_AC97_POWER_SAVE - bool "AC97 Power-Saving Mode" - depends on SND_AC97_CODEC && EXPERIMENTAL - default n - help - Say Y here to enable the aggressive power-saving support of - AC97 codecs. In this mode, the power-mode is dynamically - controlled at each open/close. - - The mode is activated by passing power_save=1 option to - snd-ac97-codec driver. You can toggle it dynamically over - sysfs, too. - -config SND_AC97_POWER_SAVE_DEFAULT - int "Default time-out for AC97 power-save mode" - depends on SND_AC97_POWER_SAVE - default 0 - help - The default time-out value in seconds for AC97 automatic - power-save mode. 0 means to disable the power-save mode. - endif # SND_PCI From 62cf872a8eec1f11aacbec0ac3fe3698bfa9b403 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 May 2008 12:15:15 +0200 Subject: [PATCH 042/179] [ALSA] Replace CONFIG_SND_DEBUG_DETECT with CONFIG_SND_DEBUG_VERBOSE Replace CONFIG_SND_DEBUG_DETECT with CONFIG_SND_DEBUG_VERBOSE to represent its meaning more better. This config isn't provided only for the detection but for more verbose debug prints in general. Signed-off-by: Takashi Iwai --- .../sound/alsa/ALSA-Configuration.txt | 2 +- .../alsa/DocBook/writing-an-alsa-driver.tmpl | 4 ++-- include/sound/core.h | 8 ++++---- sound/core/Kconfig | 10 ++++++---- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/pcxhr/pcxhr.c | 4 ++-- sound/pci/pcxhr/pcxhr_core.c | 18 +++++++++--------- 8 files changed, 26 insertions(+), 24 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 0bbee38acd26..e59569462cb9 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1091,7 +1091,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This occurs when the access to non-existing or non-working codec slot (likely a modem one) causes a stall of the communication via HD-audio bus. You can see which codec slots are probed by enabling - CONFIG_SND_DEBUG_DETECT, or simply from the file name of the codec + CONFIG_SND_DEBUG_VERBOSE, or simply from the file name of the codec proc files. Then limit the slots to probe by probe_mask option. For example, probe_mask=1 means to probe only the first slot, and probe_mask=4 means only the third slot. diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl index b03df4d4795c..e13c4e67029f 100644 --- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl @@ -6127,8 +6127,8 @@ struct _snd_pcm_runtime { snd_printdd() is compiled in only when - CONFIG_SND_DEBUG_DETECT is set. Please note - that DEBUG_DETECT is not set as default + CONFIG_SND_DEBUG_VERBOSE is set. Please note + that CONFIG_SND_DEBUG_VERBOSE is not set as default even if you configure the alsa-driver with option. You need to give explicitly option instead. diff --git a/include/sound/core.h b/include/sound/core.h index 695ee53488a3..558b96284bd2 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -412,13 +412,13 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) #endif /* CONFIG_SND_DEBUG */ -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE /** * snd_printdd - debug printk * @format: format string * * Works like snd_printk() for debugging purposes. - * Ignored when CONFIG_SND_DEBUG_DETECT is not set. + * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set. */ #define snd_printdd(format, args...) snd_printk(format, ##args) #else @@ -442,7 +442,7 @@ struct snd_pci_quirk { unsigned short subvendor; /* PCI subvendor ID */ unsigned short subdevice; /* PCI subdevice ID */ int value; /* value */ -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE const char *name; /* name of the device (optional) */ #endif }; @@ -450,7 +450,7 @@ struct snd_pci_quirk { #define _SND_PCI_QUIRK_ID(vend,dev) \ .subvendor = (vend), .subdevice = (dev) #define SND_PCI_QUIRK_ID(vend,dev) {_SND_PCI_QUIRK_ID(vend, dev)} -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE #define SND_PCI_QUIRK(vend,dev,xname,val) \ {_SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname)} #else diff --git a/sound/core/Kconfig b/sound/core/Kconfig index db2111375784..335d45ecde6a 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -153,12 +153,14 @@ config SND_DEBUG help Say Y here to enable ALSA debug code. -config SND_DEBUG_DETECT - bool "Debug detection" +config SND_DEBUG_VERBOSE + bool "More verbose debug" depends on SND_DEBUG help - Say Y here to enable extra-verbose log messages printed when - detecting devices. + Say Y here to enable extra-verbose debugging messages. + + Let me repeat: it enables EXTRA-VERBOSE DEBUGGING messages. + So, say Y only if you are ready to be annoyed. config SND_PCM_XRUN_DEBUG bool "Enable PCM ring buffer overrun/underrun debugging" diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a6be6e3e8716..d2e1093f8e97 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2335,7 +2335,7 @@ int snd_hda_check_board_config(struct hda_codec *codec, if (!tbl) return -1; if (tbl->value >= 0 && tbl->value < num_configs) { -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE char tmp[10]; const char *model = NULL; if (models) diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 2177d9af5334..6e18a422d993 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -88,7 +88,7 @@ static int hda_hwdep_ioctl_compat(struct snd_hwdep *hw, struct file *file, static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file) { -#ifndef CONFIG_SND_DEBUG_DETECT +#ifndef CONFIG_SND_DEBUG_VERBOSE if (!capable(CAP_SYS_RAWIO)) return -EACCES; #endif diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 7fdcdc8c6b64..2c7e25336795 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -516,7 +516,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg) int capture_mask = 0; int playback_mask = 0; -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE struct timeval my_tv1, my_tv2; do_gettimeofday(&my_tv1); #endif @@ -623,7 +623,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg) mutex_unlock(&mgr->setup_mutex); -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE do_gettimeofday(&my_tv2); snd_printdd("***TRIGGER TASKLET*** TIME = %ld (err = %x)\n", (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 78aa81feaa4a..abe5c59b72df 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -473,7 +473,7 @@ static struct pcxhr_cmd_info pcxhr_dsp_cmds[] = { [CMD_AUDIO_LEVEL_ADJUST] = { 0xc22000, 0, RMH_SSIZE_FIXED }, }; -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE static char* cmd_names[] = { [CMD_VERSION] = "CMD_VERSION", [CMD_SUPPORTED] = "CMD_SUPPORTED", @@ -549,7 +549,7 @@ static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) } } } -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (rmh->cmd_idx < CMD_LAST_INDEX) snd_printdd(" stat[%d]=%x\n", i, data); #endif @@ -597,7 +597,7 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) data |= 0x008000; /* MASK_MORE_THAN_1_WORD_COMMAND */ else data &= 0xff7fff; /* MASK_1_WORD_COMMAND */ -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (rmh->cmd_idx < CMD_LAST_INDEX) snd_printdd("MSG cmd[0]=%x (%s)\n", data, cmd_names[rmh->cmd_idx]); #endif @@ -624,7 +624,7 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) for (i=1; i < rmh->cmd_len; i++) { /* send other words */ data = rmh->cmd[i]; -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (rmh->cmd_idx < CMD_LAST_INDEX) snd_printdd(" cmd[%d]=%x\n", i, data); #endif @@ -847,7 +847,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m int state, i, err; int audio_mask; -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE struct timeval my_tv1, my_tv2; do_gettimeofday(&my_tv1); #endif @@ -894,7 +894,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m if (err) return err; } -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE do_gettimeofday(&my_tv2); snd_printdd("***SET PIPE STATE*** TIME = %ld (err = %x)\n", (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); @@ -951,7 +951,7 @@ static int pcxhr_handle_async_err(struct pcxhr_mgr *mgr, u32 err, enum pcxhr_async_err_src err_src, int pipe, int is_capture) { -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE static char* err_src_name[] = { [PCXHR_ERR_PIPE] = "Pipe", [PCXHR_ERR_STREAM] = "Stream", @@ -1169,7 +1169,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) mgr->dsp_time_last, dsp_time_new); mgr->dsp_time_err++; } -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (dsp_time_diff == 0) snd_printdd("ERROR DSP TIME NO DIFF time(%d)\n", dsp_time_new); else if (dsp_time_diff >= (2*PCXHR_GRANULARITY)) @@ -1208,7 +1208,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) mgr->src_it_dsp = reg; tasklet_hi_schedule(&mgr->msg_taskq); } -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (reg & PCXHR_FATAL_DSP_ERR) snd_printdd("FATAL DSP ERROR : %x\n", reg); #endif From 142054a389ebf7972b4eee822ad7c55ff852b649 Mon Sep 17 00:00:00 2001 From: Mike Rapoport Date: Mon, 26 May 2008 12:59:16 +0300 Subject: [PATCH 043/179] [ALSA] Add EM-X270 ASoC driver This patch adds ASoC support for EM-X270 machine. Signed-off-by: Mike Rapoport Signed-off-by: Takashi Iwai --- sound/soc/pxa/Kconfig | 9 ++++ sound/soc/pxa/Makefile | 3 +- sound/soc/pxa/em-x270.c | 102 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 113 insertions(+), 1 deletion(-) create mode 100644 sound/soc/pxa/em-x270.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 18a40dc8ad9d..12f6ac99b04c 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -62,3 +62,12 @@ config SND_PXA2XX_SOC_E800 help Say Y if you want to add support for SoC audio on the Toshiba e800 PDA + +config SND_PXA2XX_SOC_EM_X270 + tristate "SoC Audio support for CompuLab EM-x270" + depends on SND_PXA2XX_SOC && MACH_EM_X270 + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9712 + help + Say Y if you want to add support for SoC audio on + CompuLab EM-x270. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 04e5646f75ba..5bc8edf9dca9 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -13,10 +13,11 @@ snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o +snd-soc-em-x270-objs := em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o - +obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c new file mode 100644 index 000000000000..02dcac39cdf6 --- /dev/null +++ b/sound/soc/pxa/em-x270.c @@ -0,0 +1,102 @@ +/* + * em-x270.c -- SoC audio for EM-X270 + * + * Copyright 2007 CompuLab, Ltd. + * + * Author: Mike Rapoport + * + * Copied from tosa.c: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * Authors: Liam Girdwood + * Richard Purdie + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "../codecs/wm9712.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + +static struct snd_soc_dai_link em_x270_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], + }, +}; + +static struct snd_soc_machine em_x270 = { + .name = "EM-X270", + .dai_link = em_x270_dai, + .num_links = ARRAY_SIZE(em_x270_dai), +}; + +static struct snd_soc_device em_x270_snd_devdata = { + .machine = &em_x270, + .platform = &pxa2xx_soc_platform, + .codec_dev = &soc_codec_dev_wm9712, +}; + +static struct platform_device *em_x270_snd_device; + +static int __init em_x270_init(void) +{ + int ret; + + if (!machine_is_em_x270()) + return -ENODEV; + + em_x270_snd_device = platform_device_alloc("soc-audio", -1); + if (!em_x270_snd_device) + return -ENOMEM; + + platform_set_drvdata(em_x270_snd_device, &em_x270_snd_devdata); + em_x270_snd_devdata.dev = &em_x270_snd_device->dev; + ret = platform_device_add(em_x270_snd_device); + + if (ret) + platform_device_put(em_x270_snd_device); + + return ret; +} + +static void __exit em_x270_exit(void) +{ + platform_device_unregister(em_x270_snd_device); +} + +module_init(em_x270_init); +module_exit(em_x270_exit); + +/* Module information */ +MODULE_AUTHOR("Mike Rapoport"); +MODULE_DESCRIPTION("ALSA SoC EM-X270"); +MODULE_LICENSE("GPL"); From f269002e61446ed3410d8ca5f06ebca1e2760cb5 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 27 May 2008 11:44:55 +0200 Subject: [PATCH 044/179] [ALSA] hda - Add support of Teradici controller Add the new PCI ID 0x6549 0x1200 Teradici controller. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6ba7ac01d9f6..7f62196989a1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -197,6 +197,10 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define ATIHDMI_NUM_CAPTURE 0 #define ATIHDMI_NUM_PLAYBACK 1 +/* TERA has 4 playback and 3 capture */ +#define TERA_NUM_CAPTURE 3 +#define TERA_NUM_PLAYBACK 4 + /* this number is statically defined for simplicity */ #define MAX_AZX_DEV 16 @@ -384,6 +388,7 @@ enum { AZX_DRIVER_SIS, AZX_DRIVER_ULI, AZX_DRIVER_NVIDIA, + AZX_DRIVER_TERA, }; static char *driver_short_names[] __devinitdata = { @@ -395,6 +400,7 @@ static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", [AZX_DRIVER_NVIDIA] = "HDA NVidia", + [AZX_DRIVER_TERA] = "HDA Teradici", }; /* @@ -1106,6 +1112,7 @@ static unsigned int azx_max_codecs[] __devinitdata = { [AZX_DRIVER_SIS] = 3, /* FIXME: correct? */ [AZX_DRIVER_ULI] = 3, /* FIXME: correct? */ [AZX_DRIVER_NVIDIA] = 3, /* FIXME: correct? */ + [AZX_DRIVER_TERA] = 1, }; static int __devinit azx_codec_create(struct azx *chip, const char *model, @@ -2229,6 +2236,8 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA }, + /* Teradici */ + { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); From abbc9d1b25637b1948a4718fa8f7b257233136bc Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 27 May 2008 11:48:01 +0200 Subject: [PATCH 045/179] [ALSA] hda - Add ICH9 controller support (8086:2911) Added the missing PCI ID for ICH9 controller (8086:2911) Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7f62196989a1..6e2dc4f5ca7a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2184,6 +2184,7 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x8086, 0x27d8), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x269a), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x284b), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x2911), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x293e), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH }, From 6dda9f4a95905f2b38e79e3737da5e25397e6acb Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 27 May 2008 12:05:31 +0200 Subject: [PATCH 046/179] [ALSA] hda - Add ALC663 support Added the support of ALC663 codec, including specific models for ASUS M51VA, ASUS G71V, ASUS H13 and ASUS G50V. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- .../sound/alsa/ALSA-Configuration.txt | 6 +- sound/pci/hda/patch_realtek.c | 289 +++++++++++++++++- 2 files changed, 291 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index e59569462cb9..f48939e97aba 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -845,7 +845,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ALC269 basic Basic preset - ALC662 + ALC662/663 3stack-dig 3-stack (2-channel) with SPDIF 3stack-6ch 3-stack (6-channel) 3stack-6ch-dig 3-stack (6-channel) with SPDIF @@ -853,6 +853,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. lenovo-101e Lenovo laptop eeepc-p701 ASUS Eeepc P701 eeepc-ep20 ASUS Eeepc EP20 + m51va ASUS M51VA + g71v ASUS G71V + h13 ASUS H13 + g50v ASUS G50V auto auto-config reading BIOS (default) ALC882/885 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b0a2a262ece2..c659588e26d3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -163,6 +163,10 @@ enum { ALC662_LENOVO_101E, ALC662_ASUS_EEEPC_P701, ALC662_ASUS_EEEPC_EP20, + ALC663_ASUS_M51VA, + ALC663_ASUS_G71V, + ALC663_ASUS_H13, + ALC663_ASUS_G50V, ALC662_AUTO, ALC662_MODEL_LAST, }; @@ -13251,6 +13255,23 @@ static struct hda_input_mux alc662_eeepc_capture_source = { }, }; +static struct hda_input_mux alc663_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x1 }, + { "Line", 0x2 }, + }, +}; + +static struct hda_input_mux alc663_m51va_capture_source = { + .num_items = 2, + .items = { + { "Ext-Mic", 0x0 }, + { "D-Mic", 0x9 }, + }, +}; + #define alc662_mux_enum_info alc_mux_enum_info #define alc662_mux_enum_get alc_mux_enum_get #define alc662_mux_enum_put alc882_mux_enum_put @@ -13431,6 +13452,44 @@ static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc663_m51va_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("DMic Playback Switch", 0x23, 0x9, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc663_g71v_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc663_g50v_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc662_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -13501,6 +13560,11 @@ static struct hda_verb alc662_init_verbs[] = { {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + + /* always trun on EAPD */ + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, + { } }; @@ -13571,6 +13635,43 @@ static struct hda_verb alc662_auto_init_verbs[] = { { } }; +static struct hda_verb alc663_m51va_init_verbs[] = { + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ + + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)}, + + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + +static struct hda_verb alc663_g71v_init_verbs[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */ + /* {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, */ /* Headphone */ + + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ + + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT}, + {} +}; + +static struct hda_verb alc663_g50v_init_verbs[] = { + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ + + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + /* capture mixer elements */ static struct snd_kcontrol_new alc662_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), @@ -13692,6 +13793,125 @@ static void alc662_eeepc_ep20_inithook(struct hda_codec *codec) alc662_eeepc_ep20_automute(codec); } +static void alc663_m51va_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x21, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc663_m51va_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); + snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); + snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); + snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); +} + +static void alc663_m51va_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_m51va_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc663_m51va_mic_automute(codec); + break; + } +} + +static void alc663_m51va_inithook(struct hda_codec *codec) +{ + alc663_m51va_speaker_automute(codec); + alc663_m51va_mic_automute(codec); +} + +static void alc663_g71v_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x21, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc663_g71v_front_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc663_g71v_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_g71v_hp_automute(codec); + break; + case ALC880_FRONT_EVENT: + alc663_g71v_front_automute(codec); + break; + case ALC880_MIC_EVENT: + alc662_eeepc_mic_automute(codec); + break; + } +} + +static void alc663_g71v_inithook(struct hda_codec *codec) +{ + alc663_g71v_front_automute(codec); + alc663_g71v_hp_automute(codec); + alc662_eeepc_mic_automute(codec); +} + +static void alc663_g50v_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_m51va_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc662_eeepc_mic_automute(codec); + break; + } +} + +static void alc663_g50v_inithook(struct hda_codec *codec) +{ + alc663_m51va_speaker_automute(codec); + alc662_eeepc_mic_automute(codec); +} + #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc662_loopbacks alc880_loopbacks #endif @@ -13714,14 +13934,24 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC662_LENOVO_101E] = "lenovo-101e", [ALC662_ASUS_EEEPC_P701] = "eeepc-p701", [ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20", + [ALC663_ASUS_M51VA] = "m51va", + [ALC663_ASUS_G71V] = "g71v", + [ALC663_ASUS_H13] = "h13", + [ALC663_ASUS_G50V] = "g50v", [ALC662_AUTO] = "auto", }; static struct snd_pci_quirk alc662_cfg_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS G71V", ALC663_ASUS_G71V), + SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA), + SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS M51VA", ALC663_ASUS_G50V), SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), + SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13), {} }; @@ -13809,7 +14039,53 @@ static struct alc_config_preset alc662_presets[] = { .unsol_event = alc662_eeepc_ep20_unsol_event, .init_hook = alc662_eeepc_ep20_inithook, }, - + [ALC663_ASUS_M51VA] = { + .mixers = { alc663_m51va_mixer, alc662_capture_mixer}, + .init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc663_m51va_capture_source, + .unsol_event = alc663_m51va_unsol_event, + .init_hook = alc663_m51va_inithook, + }, + [ALC663_ASUS_G71V] = { + .mixers = { alc663_g71v_mixer, alc662_capture_mixer}, + .init_verbs = { alc662_init_verbs, alc663_g71v_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc662_eeepc_capture_source, + .unsol_event = alc663_g71v_unsol_event, + .init_hook = alc663_g71v_inithook, + }, + [ALC663_ASUS_H13] = { + .mixers = { alc663_m51va_mixer, alc662_capture_mixer}, + .init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc663_m51va_capture_source, + .unsol_event = alc663_m51va_unsol_event, + .init_hook = alc663_m51va_inithook, + }, + [ALC663_ASUS_G50V] = { + .mixers = { alc663_g50v_mixer, alc662_capture_mixer}, + .init_verbs = { alc662_init_verbs, alc663_g50v_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), + .channel_mode = alc662_3ST_6ch_modes, + .input_mux = &alc663_capture_source, + .unsol_event = alc663_g50v_unsol_event, + .init_hook = alc663_g50v_inithook, + }, }; @@ -14108,11 +14384,17 @@ static int patch_alc662(struct hda_codec *codec) if (board_config != ALC662_AUTO) setup_preset(spec, &alc662_presets[board_config]); - spec->stream_name_analog = "ALC662 Analog"; + if (codec->vendor_id == 0x10ec0663) { + spec->stream_name_analog = "ALC663 Analog"; + spec->stream_name_digital = "ALC663 Digital"; + } else { + spec->stream_name_analog = "ALC662 Analog"; + spec->stream_name_digital = "ALC662 Digital"; + } + spec->stream_analog_playback = &alc662_pcm_analog_playback; spec->stream_analog_capture = &alc662_pcm_analog_capture; - spec->stream_name_digital = "ALC662 Digital"; spec->stream_digital_playback = &alc662_pcm_digital_playback; spec->stream_digital_capture = &alc662_pcm_digital_capture; @@ -14151,6 +14433,7 @@ struct hda_codec_preset snd_hda_preset_realtek[] = { .patch = patch_alc883 }, { .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1", .patch = patch_alc662 }, + { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 }, From 531240ff520406c793a110e1c0f187d931f47d66 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 27 May 2008 12:10:25 +0200 Subject: [PATCH 047/179] [ALSA] hda - Fix vref pincap check in alc882 auto-detection Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c659588e26d3..8174050da15f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6357,7 +6357,9 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) continue; vref = PIN_IN; if (1 /*i <= AUTO_PIN_FRONT_MIC*/) { - if (snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP) & + unsigned int pincap; + pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + if ((pincap >> AC_PINCAP_VREF_SHIFT) & AC_PINCAP_VREF_80) vref = PIN_VREF80; } From 2f8932863d243a744ccd3dc005490ad9d2eae478 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 27 May 2008 12:14:47 +0200 Subject: [PATCH 048/179] [ALSA] hda - show correct codec chip in PCM stream names Show more exact codec chip name in the PCM stream name strings. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 47 +++++++++++++++++++++++++++++------ 1 file changed, 39 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8174050da15f..5770b9c3efa2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6491,14 +6491,20 @@ static int patch_alc882(struct hda_codec *codec) if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); - spec->stream_name_analog = "ALC882 Analog"; + if (codec->vendor_id == 0x10ec0885) { + spec->stream_name_analog = "ALC885 Analog"; + spec->stream_name_digital = "ALC885 Digital"; + } else { + spec->stream_name_analog = "ALC882 Analog"; + spec->stream_name_digital = "ALC882 Digital"; + } + spec->stream_analog_playback = &alc882_pcm_analog_playback; spec->stream_analog_capture = &alc882_pcm_analog_capture; /* FIXME: setup DAC5 */ /*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/ spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture; - spec->stream_name_digital = "ALC882 Digital"; spec->stream_digital_playback = &alc882_pcm_digital_playback; spec->stream_digital_capture = &alc882_pcm_digital_capture; @@ -8177,12 +8183,25 @@ static int patch_alc883(struct hda_codec *codec) if (board_config != ALC883_AUTO) setup_preset(spec, &alc883_presets[board_config]); - spec->stream_name_analog = "ALC883 Analog"; + switch (codec->vendor_id) { + case 0x10ec0888: + spec->stream_name_analog = "ALC888 Analog"; + spec->stream_name_digital = "ALC888 Digital"; + break; + case 0x10ec0889: + spec->stream_name_analog = "ALC889 Analog"; + spec->stream_name_digital = "ALC889 Digital"; + break; + default: + spec->stream_name_analog = "ALC883 Analog"; + spec->stream_name_digital = "ALC883 Digital"; + break; + } + spec->stream_analog_playback = &alc883_pcm_analog_playback; spec->stream_analog_capture = &alc883_pcm_analog_capture; spec->stream_analog_alt_capture = &alc883_pcm_analog_alt_capture; - spec->stream_name_digital = "ALC883 Digital"; spec->stream_digital_playback = &alc883_pcm_digital_playback; spec->stream_digital_capture = &alc883_pcm_digital_capture; @@ -10680,12 +10699,18 @@ static int patch_alc268(struct hda_codec *codec) if (board_config != ALC268_AUTO) setup_preset(spec, &alc268_presets[board_config]); - spec->stream_name_analog = "ALC268 Analog"; + if (codec->vendor_id == 0x10ec0267) { + spec->stream_name_analog = "ALC267 Analog"; + spec->stream_name_digital = "ALC267 Digital"; + } else { + spec->stream_name_analog = "ALC268 Analog"; + spec->stream_name_digital = "ALC268 Digital"; + } + spec->stream_analog_playback = &alc268_pcm_analog_playback; spec->stream_analog_capture = &alc268_pcm_analog_capture; spec->stream_analog_alt_capture = &alc268_pcm_analog_alt_capture; - spec->stream_name_digital = "ALC268 Digital"; spec->stream_digital_playback = &alc268_pcm_digital_playback; if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) @@ -13174,11 +13199,17 @@ static int patch_alc861vd(struct hda_codec *codec) if (board_config != ALC861VD_AUTO) setup_preset(spec, &alc861vd_presets[board_config]); - spec->stream_name_analog = "ALC861VD Analog"; + if (codec->vendor_id == 0x10ec0660) { + spec->stream_name_analog = "ALC660-VD Analog"; + spec->stream_name_digital = "ALC660-VD Digital"; + } else { + spec->stream_name_analog = "ALC861VD Analog"; + spec->stream_name_digital = "ALC861VD Digital"; + } + spec->stream_analog_playback = &alc861vd_pcm_analog_playback; spec->stream_analog_capture = &alc861vd_pcm_analog_capture; - spec->stream_name_digital = "ALC861VD Digital"; spec->stream_digital_playback = &alc861vd_pcm_digital_playback; spec->stream_digital_capture = &alc861vd_pcm_digital_capture; From f9423e7a94eb2dfef3503dde76d17eaf342ab962 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 27 May 2008 12:32:25 +0200 Subject: [PATCH 049/179] [ALSA] hda - Fix EAPD and COEF setups for realtek codecs Fixed EAPD and COEF setups for Realtek ALC662/663, 660-VD and 888 codecs. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 37 +++++++++++++++++++++++++++++++++-- 1 file changed, 35 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5770b9c3efa2..cb3e0283f0ce 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -780,6 +780,24 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) alc_sku_automute(codec); } +/* additional initialization for ALC888 variants */ +static void alc888_coef_init(struct hda_codec *codec) +{ + unsigned int tmp; + + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0); + tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); + if ((tmp & 0xf0) == 2) + /* alc888S-VC */ + snd_hda_codec_read(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x830); + else + /* alc888-VB */ + snd_hda_codec_read(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x3030); +} + /* 32-bit subsystem ID for BIOS loading in HD Audio codec. * 31 ~ 16 : Manufacture ID * 15 ~ 8 : SKU ID @@ -855,8 +873,10 @@ do_sku: case 0x10ec0267: case 0x10ec0268: case 0x10ec0269: + case 0x10ec0660: + case 0x10ec0662: + case 0x10ec0663: case 0x10ec0862: - case 0x10ec0662: case 0x10ec0889: snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_EAPD_BTLENABLE, 2); @@ -881,7 +901,6 @@ do_sku: case 0x10ec0882: case 0x10ec0883: case 0x10ec0885: - case 0x10ec0888: case 0x10ec0889: snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); @@ -893,6 +912,9 @@ do_sku: AC_VERB_SET_PROC_COEF, tmp | 0x2010); break; + case 0x10ec0888: + alc888_coef_init(codec); + break; case 0x10ec0267: case 0x10ec0268: snd_hda_codec_write(codec, 0x20, 0, @@ -8214,6 +8236,9 @@ static int patch_alc883(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC883_AUTO) spec->init_hook = alc883_auto_init; + else if (codec->vendor_id == 0x10ec0888) + spec->init_hook = alc888_coef_init; + #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc883_loopbacks; @@ -12662,6 +12687,12 @@ static struct hda_verb alc861vd_eapd_verbs[] = { { } }; +static struct hda_verb alc660vd_eapd_verbs[] = { + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, + { } +}; + static struct hda_verb alc861vd_lenovo_unsol_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -13202,6 +13233,8 @@ static int patch_alc861vd(struct hda_codec *codec) if (codec->vendor_id == 0x10ec0660) { spec->stream_name_analog = "ALC660-VD Analog"; spec->stream_name_digital = "ALC660-VD Digital"; + /* always turn on EAPD */ + spec->init_verbs[spec->num_init_verbs++] = alc660vd_eapd_verbs; } else { spec->stream_name_analog = "ALC861VD Analog"; spec->stream_name_digital = "ALC861VD Digital"; From a93bbaa77ea61c6bad684263a65f812b31bf9791 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 May 2008 17:59:24 +0200 Subject: [PATCH 050/179] [ALSA] Improve the slots option handling Fix and improve the slots option handling. The sound core tries to find the slot with the given module name first and assign if it's still available. If all pre-given slots are unavailable, then try to find another free slot. Also, when a module name begins with '!', it means the negative match: the slot will be given for any modules but that one. Signed-off-by: Takashi Iwai --- .../sound/alsa/ALSA-Configuration.txt | 4 ++ sound/core/init.c | 67 +++++++++++-------- 2 files changed, 44 insertions(+), 27 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index f48939e97aba..529073dea196 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -2271,6 +2271,10 @@ case above again, the first two slots are already reserved. If any other driver (e.g. snd-usb-audio) is loaded before snd-interwave or snd-ens1371, it will be assigned to the third or later slot. +When a module name is given with '!', the slot will be given for any +modules but that name. For example, "slots=!snd-pcsp" will reserve +the first slot for any modules but snd-pcsp. + ALSA PCM devices to OSS devices mapping ======================================= diff --git a/sound/core/init.c b/sound/core/init.c index ac0573416130..5c254d498ae0 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -46,17 +46,24 @@ static char *slots[SNDRV_CARDS]; module_param_array(slots, charp, NULL, 0444); MODULE_PARM_DESC(slots, "Module names assigned to the slots."); -/* return non-zero if the given index is already reserved for another +/* return non-zero if the given index is reserved for the given * module via slots option */ -static int module_slot_mismatch(struct module *module, int idx) +static int module_slot_match(struct module *module, int idx) { + int match = 1; #ifdef MODULE - char *s1, *s2; + const char *s1, *s2; + if (!module || !module->name || !slots[idx]) return 0; - s1 = slots[idx]; - s2 = module->name; + + s1 = module->name; + s2 = slots[idx]; + if (*s2 == '!') { + match = 0; /* negative match */ + s2++; + } /* compare module name strings * hyphens are handled as equivalent with underscore */ @@ -68,12 +75,12 @@ static int module_slot_mismatch(struct module *module, int idx) if (c2 == '-') c2 = '_'; if (c1 != c2) - return 1; + return !match; if (!c1) break; } -#endif - return 0; +#endif /* MODULE */ + return match; } #if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) @@ -129,7 +136,7 @@ struct snd_card *snd_card_new(int idx, const char *xid, struct module *module, int extra_size) { struct snd_card *card; - int err; + int err, idx2; if (extra_size < 0) extra_size = 0; @@ -144,35 +151,41 @@ struct snd_card *snd_card_new(int idx, const char *xid, err = 0; mutex_lock(&snd_card_mutex); if (idx < 0) { - int idx2; for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) /* idx == -1 == 0xffff means: take any free slot */ if (~snd_cards_lock & idx & 1<= snd_ecards_limit) - snd_ecards_limit = idx + 1; - break; + if (module_slot_match(module, idx2)) { + idx = idx2; + break; + } } - } else { - if (idx < snd_ecards_limit) { - if (snd_cards_lock & (1 << idx)) - err = -EBUSY; /* invalid */ - } else { - if (idx < SNDRV_CARDS) - snd_ecards_limit = idx + 1; /* increase the limit */ - else - err = -ENODEV; - } } - if (idx < 0 || err < 0) { + if (idx < 0) { + for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) + /* idx == -1 == 0xffff means: take any free slot */ + if (~snd_cards_lock & idx & 1<= SNDRV_CARDS) + err = -ENODEV; + if (err < 0) { mutex_unlock(&snd_card_mutex); snd_printk(KERN_ERR "cannot find the slot for index %d (range 0-%i), error: %d\n", idx, snd_ecards_limit - 1, err); goto __error; } snd_cards_lock |= 1 << idx; /* lock it */ + if (idx >= snd_ecards_limit) + snd_ecards_limit = idx + 1; /* increase the limit */ mutex_unlock(&snd_card_mutex); card->number = idx; card->module = module; From e13ac2e9b18bde51cf32c69c2209df25791ab3e5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 28 May 2008 17:58:05 +0100 Subject: [PATCH 051/179] [ALSA] ASoC: Add SOC_DOUBLE_S8_TLV control type The SOC_DOUBLE_S8_TLV control type was originally implemented in the UDA1380 driver by Philipp Zabel and was moved into the core by me. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- include/sound/soc.h | 15 +++++++++ sound/soc/soc-core.c | 72 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 87 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index bca9538d9e50..9fa2093e74eb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -73,6 +73,15 @@ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ .private_value = (reg_left) | ((shift) << 8) | \ ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } +#define SOC_DOUBLE_S8_TLV(xname, reg, min, max, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ + .put = snd_soc_put_volsw_s8, \ + .private_value = (reg) | (((signed char)max) << 16) | \ + (((signed char)min) << 24) } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .texts = xtexts } @@ -267,6 +276,12 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); /* SoC PCM stream information */ struct snd_soc_pcm_stream { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a3f091e0843a..f594ab888e17 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1589,6 +1589,78 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); +/** + * snd_soc_info_volsw_s8 - signed mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int max = (signed char)((kcontrol->private_value >> 16) & 0xff); + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = max-min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); + +/** + * snd_soc_get_volsw_s8 - signed mixer get callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to get the value of a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + int val = snd_soc_read(codec, reg); + + ucontrol->value.integer.value[0] = + ((signed char)(val & 0xff))-min; + ucontrol->value.integer.value[1] = + ((signed char)((val >> 8) & 0xff))-min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); + +/** + * snd_soc_put_volsw_sgn - signed mixer put callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to set the value of a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + unsigned short val; + + val = (ucontrol->value.integer.value[0]+min) & 0xff; + val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; + + return snd_soc_update_bits(codec, reg, 0xffff, val); +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); + static int __devinit snd_soc_init(void) { printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION); From b7482f52789266e2548be5d0f6420c9fc12428d8 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Wed, 28 May 2008 17:58:06 +0100 Subject: [PATCH 052/179] [ALSA] ASoC: Add UDA1380 driver The UDA1380 codec is used by the HTC Magician and a number of Samsung reference boards. This driver has had a long out of tree history, having originally been written by Giorgio Padrin and converted to ASoC by Richard Purdie. Since conversion to ASoC most of the work on the driver has been done by Philipp Zabel with some review and updates for new APIs by Liam Girdwood and Mark Brown. Signed-off-by: Richard Purdie Signed-off-by: Philipp Zabel Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/uda1380.c | 852 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/uda1380.h | 89 ++++ 4 files changed, 946 insertions(+) create mode 100644 sound/soc/codecs/uda1380.c create mode 100644 sound/soc/codecs/uda1380.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d4a5fe42f6e0..7beefccfa821 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2,6 +2,9 @@ config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC +config SND_SOC_UDA1380 + tristate + config SND_SOC_WM8731 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4e1314c9d3ec..d5926a117078 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,4 +1,5 @@ snd-soc-ac97-objs := ac97.o +snd-soc-uda1380-objs := uda1380.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o @@ -8,6 +9,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o +obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c new file mode 100644 index 000000000000..cb50486201f1 --- /dev/null +++ b/sound/soc/codecs/uda1380.c @@ -0,0 +1,852 @@ +/* + * uda1380.c - Philips UDA1380 ALSA SoC audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Copyright (c) 2007 Philipp Zabel + * Improved support for DAPM and audio routing/mixing capabilities, + * added TLV support. + * + * Modified by Richard Purdie to fit into SoC + * codec model. + * + * Copyright (c) 2005 Giorgio Padrin + * Copyright 2005 Openedhand Ltd. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "uda1380.h" + +#define UDA1380_VERSION "0.6" +#define AUDIO_NAME "uda1380" + +/* + * uda1380 register cache + */ +static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = { + 0x0502, 0x0000, 0x0000, 0x3f3f, + 0x0202, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0xff00, 0x0000, 0x4800, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x8000, 0x0002, 0x0000, +}; + +/* + * read uda1380 register cache + */ +static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == UDA1380_RESET) + return 0; + if (reg >= UDA1380_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write uda1380 register cache + */ +static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= UDA1380_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the UDA1380 register space + */ +static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[3]; + + /* data is + * data[0] is register offset + * data[1] is MS byte + * data[2] is LS byte + */ + data[0] = reg; + data[1] = (value & 0xff00) >> 8; + data[2] = value & 0x00ff; + + uda1380_write_reg_cache(codec, reg, value); + + /* the interpolator & decimator regs must only be written when the + * codec DAI is active. + */ + if (!codec->active && (reg >= UDA1380_MVOL)) + return 0; + pr_debug("uda1380: hw write %x val %x\n", reg, value); + if (codec->hw_write(codec->control_data, data, 3) == 3) { + unsigned int val; + i2c_master_send(codec->control_data, data, 1); + i2c_master_recv(codec->control_data, data, 2); + val = (data[0]<<8) | data[1]; + if (val != value) { + pr_debug("uda1380: READ BACK VAL %x\n", + (data[0]<<8) | data[1]); + return -EIO; + } + return 0; + } else + return -EIO; +} + +#define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0) + +/* declarations of ALSA reg_elem_REAL controls */ +static const char *uda1380_deemp[] = { + "None", + "32kHz", + "44.1kHz", + "48kHz", + "96kHz", +}; +static const char *uda1380_input_sel[] = { + "Line", + "Mic + Line R", + "Line L", + "Mic", +}; +static const char *uda1380_output_sel[] = { + "DAC", + "Analog Mixer", +}; +static const char *uda1380_spf_mode[] = { + "Flat", + "Minimum1", + "Minimum2", + "Maximum" +}; +static const char *uda1380_capture_sel[] = { + "ADC", + "Digital Mixer" +}; +static const char *uda1380_sel_ns[] = { + "3rd-order", + "5th-order" +}; +static const char *uda1380_mix_control[] = { + "off", + "PCM only", + "before sound processing", + "after sound processing" +}; +static const char *uda1380_sdet_setting[] = { + "3200", + "4800", + "9600", + "19200" +}; +static const char *uda1380_os_setting[] = { + "single-speed", + "double-speed (no mixing)", + "quad-speed (no mixing)" +}; + +static const struct soc_enum uda1380_deemp_enum[] = { + SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp), + SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp), +}; +static const struct soc_enum uda1380_input_sel_enum = + SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel); /* SEL_MIC, SEL_LNA */ +static const struct soc_enum uda1380_output_sel_enum = + SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel); /* R02_EN_AVC */ +static const struct soc_enum uda1380_spf_enum = + SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode); /* M */ +static const struct soc_enum uda1380_capture_sel_enum = + SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel); /* SEL_SOURCE */ +static const struct soc_enum uda1380_sel_ns_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns); /* SEL_NS */ +static const struct soc_enum uda1380_mix_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control); /* MIX, MIX_POS */ +static const struct soc_enum uda1380_sdet_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting); /* SD_VALUE */ +static const struct soc_enum uda1380_os_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting); /* OS */ + +/* + * from -48 dB in 1.5 dB steps (mute instead of -49.5 dB) + */ +static DECLARE_TLV_DB_SCALE(amix_tlv, -4950, 150, 1); + +/* + * from -78 dB in 1 dB steps (3 dB steps, really. LSB are ignored), + * from -66 dB in 0.5 dB steps (2 dB steps, really) and + * from -52 dB in 0.25 dB steps + */ +static const unsigned int mvol_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 15, TLV_DB_SCALE_ITEM(-8200, 100, 1), + 16, 43, TLV_DB_SCALE_ITEM(-6600, 50, 0), + 44, 252, TLV_DB_SCALE_ITEM(-5200, 25, 0), +}; + +/* + * from -72 dB in 1.5 dB steps (6 dB steps really), + * from -66 dB in 0.75 dB steps (3 dB steps really), + * from -60 dB in 0.5 dB steps (2 dB steps really) and + * from -46 dB in 0.25 dB steps + */ +static const unsigned int vc_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 7, TLV_DB_SCALE_ITEM(-7800, 150, 1), + 8, 15, TLV_DB_SCALE_ITEM(-6600, 75, 0), + 16, 43, TLV_DB_SCALE_ITEM(-6000, 50, 0), + 44, 228, TLV_DB_SCALE_ITEM(-4600, 25, 0), +}; + +/* from 0 to 6 dB in 2 dB steps if SPF mode != flat */ +static DECLARE_TLV_DB_SCALE(tr_tlv, 0, 200, 0); + +/* from 0 to 24 dB in 2 dB steps, if SPF mode == maximum, otherwise cuts + * off at 18 dB max) */ +static DECLARE_TLV_DB_SCALE(bb_tlv, 0, 200, 0); + +/* from -63 to 24 dB in 0.5 dB steps (-128...48) */ +static DECLARE_TLV_DB_SCALE(dec_tlv, -6400, 50, 1); + +/* from 0 to 24 dB in 3 dB steps */ +static DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); + +/* from 0 to 30 dB in 2 dB steps */ +static DECLARE_TLV_DB_SCALE(vga_tlv, 0, 200, 0); + +static const struct snd_kcontrol_new uda1380_snd_controls[] = { + SOC_DOUBLE_TLV("Analog Mixer Volume", UDA1380_AMIX, 0, 8, 44, 1, amix_tlv), /* AVCR, AVCL */ + SOC_DOUBLE_TLV("Master Playback Volume", UDA1380_MVOL, 0, 8, 252, 1, mvol_tlv), /* MVCL, MVCR */ + SOC_SINGLE_TLV("ADC Playback Volume", UDA1380_MIXVOL, 8, 228, 1, vc_tlv), /* VC2 */ + SOC_SINGLE_TLV("PCM Playback Volume", UDA1380_MIXVOL, 0, 228, 1, vc_tlv), /* VC1 */ + SOC_ENUM("Sound Processing Filter", uda1380_spf_enum), /* M */ + SOC_DOUBLE_TLV("Tone Control - Treble", UDA1380_MODE, 4, 12, 3, 0, tr_tlv), /* TRL, TRR */ + SOC_DOUBLE_TLV("Tone Control - Bass", UDA1380_MODE, 0, 8, 15, 0, bb_tlv), /* BBL, BBR */ +/**/ SOC_SINGLE("Master Playback Switch", UDA1380_DEEMP, 14, 1, 1), /* MTM */ + SOC_SINGLE("ADC Playback Switch", UDA1380_DEEMP, 11, 1, 1), /* MT2 from decimation filter */ + SOC_ENUM("ADC Playback De-emphasis", uda1380_deemp_enum[0]), /* DE2 */ + SOC_SINGLE("PCM Playback Switch", UDA1380_DEEMP, 3, 1, 1), /* MT1, from digital data input */ + SOC_ENUM("PCM Playback De-emphasis", uda1380_deemp_enum[1]), /* DE1 */ + SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0), /* DA_POL_INV */ + SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum), /* SEL_NS */ + SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum), /* MIX_POS, MIX */ + SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0), /* SILENCE, force DAC output to silence */ + SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0), /* SDET_ON */ + SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum), /* SD_VALUE */ + SOC_ENUM("Oversampling Input", uda1380_os_enum), /* OS */ + SOC_DOUBLE_S8_TLV("ADC Capture Volume", UDA1380_DEC, -128, 48, dec_tlv), /* ML_DEC, MR_DEC */ +/**/ SOC_SINGLE("ADC Capture Switch", UDA1380_PGA, 15, 1, 1), /* MT_ADC */ + SOC_DOUBLE_TLV("Line Capture Volume", UDA1380_PGA, 0, 8, 8, 0, pga_tlv), /* PGA_GAINCTRLL, PGA_GAINCTRLR */ + SOC_SINGLE("ADC Polarity inverting Switch", UDA1380_ADC, 12, 1, 0), /* ADCPOL_INV */ + SOC_SINGLE_TLV("Mic Capture Volume", UDA1380_ADC, 8, 15, 0, vga_tlv), /* VGA_CTRL */ + SOC_SINGLE("DC Filter Bypass Switch", UDA1380_ADC, 1, 1, 0), /* SKIP_DCFIL (before decimator) */ + SOC_SINGLE("DC Filter Enable Switch", UDA1380_ADC, 0, 1, 0), /* EN_DCFIL (at output of decimator) */ + SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0), /* TODO: enum, see table 62 */ + SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1), /* AGC_LEVEL */ + /* -5.5, -8, -11.5, -14 dBFS */ + SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0), +}; + +/* add non dapm controls */ +static int uda1380_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Input mux */ +static const struct snd_kcontrol_new uda1380_input_mux_control = + SOC_DAPM_ENUM("Route", uda1380_input_sel_enum); + +/* Output mux */ +static const struct snd_kcontrol_new uda1380_output_mux_control = + SOC_DAPM_ENUM("Route", uda1380_output_sel_enum); + +/* Capture mux */ +static const struct snd_kcontrol_new uda1380_capture_mux_control = + SOC_DAPM_ENUM("Route", uda1380_capture_sel_enum); + + +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &uda1380_input_mux_control), + SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, 0, 0, + &uda1380_output_mux_control), + SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, + &uda1380_capture_mux_control), + SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0), + SND_SOC_DAPM_INPUT("VINM"), + SND_SOC_DAPM_INPUT("VINL"), + SND_SOC_DAPM_INPUT("VINR"), + SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("VOUTLHP"), + SND_SOC_DAPM_OUTPUT("VOUTRHP"), + SND_SOC_DAPM_OUTPUT("VOUTL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), + SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0), + SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + + /* output mux */ + {"HeadPhone Driver", NULL, "Output Mux"}, + {"VOUTR", NULL, "Output Mux"}, + {"VOUTL", NULL, "Output Mux"}, + + {"Analog Mixer", NULL, "VINR"}, + {"Analog Mixer", NULL, "VINL"}, + {"Analog Mixer", NULL, "DAC"}, + + {"Output Mux", "DAC", "DAC"}, + {"Output Mux", "Analog Mixer", "Analog Mixer"}, + + /* {"DAC", "Digital Mixer", "I2S" } */ + + /* headphone driver */ + {"VOUTLHP", NULL, "HeadPhone Driver"}, + {"VOUTRHP", NULL, "HeadPhone Driver"}, + + /* input mux */ + {"Left ADC", NULL, "Input Mux"}, + {"Input Mux", "Mic", "Mic LNA"}, + {"Input Mux", "Mic + Line R", "Mic LNA"}, + {"Input Mux", "Line L", "Left PGA"}, + {"Input Mux", "Line", "Left PGA"}, + + /* right input */ + {"Right ADC", "Mic + Line R", "Right PGA"}, + {"Right ADC", "Line", "Right PGA"}, + + /* inputs */ + {"Mic LNA", NULL, "VINM"}, + {"Left PGA", NULL, "VINL"}, + {"Right PGA", NULL, "VINR"}, +}; + +static int uda1380_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int uda1380_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int iface; + + /* set up DAI based upon fmt */ + iface = uda1380_read_reg_cache(codec, UDA1380_IFACE); + iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK); + + /* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= R01_SFORI_I2S | R01_SFORO_I2S; + break; + case SND_SOC_DAIFMT_LSB: + iface |= R01_SFORI_LSB16 | R01_SFORO_I2S; + break; + case SND_SOC_DAIFMT_MSB: + iface |= R01_SFORI_MSB | R01_SFORO_I2S; + } + + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) + iface |= R01_SIM; + + uda1380_write(codec, UDA1380_IFACE, iface); + + return 0; +} + +/* + * Flush reg cache + * We can only write the interpolator and decimator registers + * when the DAI is being clocked by the CPU DAI. It's up to the + * machine and cpu DAI driver to do this before we are called. + */ +static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + int reg, reg_start, reg_end, clk; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg_start = UDA1380_MVOL; + reg_end = UDA1380_MIXER; + } else { + reg_start = UDA1380_DEC; + reg_end = UDA1380_AGC; + } + + /* FIXME disable DAC_CLK */ + clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK); + + for (reg = reg_start; reg <= reg_end; reg++) { + pr_debug("uda1380: flush reg %x val %x:", reg, + uda1380_read_reg_cache(codec, reg)); + uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg)); + } + + /* FIXME enable DAC_CLK */ + uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK); + + return 0; +} + +static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + + /* set WSPLL power and divider if running from this clock */ + if (clk & R00_DAC_CLK) { + int rate = params_rate(params); + u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM); + clk &= ~0x3; /* clear SEL_LOOP_DIV */ + switch (rate) { + case 6250 ... 12500: + clk |= 0x0; + break; + case 12501 ... 25000: + clk |= 0x1; + break; + case 25001 ... 50000: + clk |= 0x2; + break; + case 50001 ... 100000: + clk |= 0x3; + break; + } + uda1380_write(codec, UDA1380_PM, R02_PON_PLL | pm); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clk |= R00_EN_DAC | R00_EN_INT; + else + clk |= R00_EN_ADC | R00_EN_DEC; + + uda1380_write(codec, UDA1380_CLK, clk); + return 0; +} + +static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + + /* shut down WSPLL power if running from this clock */ + if (clk & R00_DAC_CLK) { + u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM); + uda1380_write(codec, UDA1380_PM, ~R02_PON_PLL & pm); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clk &= ~(R00_EN_DAC | R00_EN_INT); + else + clk &= ~(R00_EN_ADC | R00_EN_DEC); + + uda1380_write(codec, UDA1380_CLK, clk); +} + +static int uda1380_mute(struct snd_soc_codec_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM; + + /* FIXME: mute(codec,0) is called when the magician clock is already + * set to WSPLL, but for some unknown reason writing to interpolator + * registers works only when clocked by SYSCLK */ + u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk); + if (mute) + uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM); + else + uda1380_write(codec, UDA1380_DEEMP, mute_reg); + uda1380_write(codec, UDA1380_CLK, clk); + return 0; +} + +static int uda1380_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int pm = uda1380_read_reg_cache(codec, UDA1380_PM); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm); + break; + case SND_SOC_BIAS_STANDBY: + uda1380_write(codec, UDA1380_PM, R02_PON_BIAS); + break; + case SND_SOC_BIAS_OFF: + uda1380_write(codec, UDA1380_PM, 0x0); + break; + } + codec->bias_level = level; + return 0; +} + +#define UDA1380_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +struct snd_soc_codec_dai uda1380_dai[] = { +{ + .name = "UDA1380", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + }, + .dai_ops = { + .digital_mute = uda1380_mute, + .set_fmt = uda1380_set_dai_fmt, + }, +}, +{ /* playback only - dual interface */ + .name = "UDA1380", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + }, + .dai_ops = { + .digital_mute = uda1380_mute, + .set_fmt = uda1380_set_dai_fmt, + }, +}, +{ /* capture only - dual interface*/ + .name = "UDA1380", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + }, + .dai_ops = { + .set_fmt = uda1380_set_dai_fmt, + }, +}, +}; +EXPORT_SYMBOL_GPL(uda1380_dai); + +static int uda1380_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int uda1380_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda1380_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the UDA1380 driver + * register mixer and dsp interfaces with the kernel + */ +static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "UDA1380"; + codec->owner = THIS_MODULE; + codec->read = uda1380_read_reg_cache; + codec->write = uda1380_write; + codec->set_bias_level = uda1380_set_bias_level; + codec->dai = uda1380_dai; + codec->num_dai = ARRAY_SIZE(uda1380_dai); + codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + codec->reg_cache_size = sizeof(uda1380_reg); + codec->reg_cache_step = 2; + uda1380_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + pr_err("uda1380: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* set clock input */ + switch (dac_clk) { + case UDA1380_DAC_CLK_SYSCLK: + uda1380_write(codec, UDA1380_CLK, 0); + break; + case UDA1380_DAC_CLK_WSPLL: + uda1380_write(codec, UDA1380_CLK, R00_DAC_CLK); + break; + } + + /* uda1380 init */ + uda1380_add_controls(codec); + uda1380_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + pr_err("uda1380: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *uda1380_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +#define I2C_DRIVERID_UDA1380 0xfefe /* liam - need a proper id */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver uda1380_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ + +static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = uda1380_socdev; + struct uda1380_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + pr_err("uda1380: failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = uda1380_init(socdev, setup->dac_clk); + if (ret < 0) { + pr_err("uda1380: failed to initialise UDA1380\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int uda1380_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int uda1380_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, uda1380_codec_probe); +} + +static struct i2c_driver uda1380_i2c_driver = { + .driver = { + .name = "UDA1380 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_UDA1380, + .attach_adapter = uda1380_i2c_attach, + .detach_client = uda1380_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "UDA1380", + .driver = &uda1380_i2c_driver, +}; +#endif + +static int uda1380_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct uda1380_setup_data *setup; + struct snd_soc_codec *codec; + int ret = 0; + + pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + uda1380_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&uda1380_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int uda1380_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&uda1380_i2c_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_uda1380 = { + .probe = uda1380_probe, + .remove = uda1380_remove, + .suspend = uda1380_suspend, + .resume = uda1380_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); + +MODULE_AUTHOR("Giorgio Padrin"); +MODULE_DESCRIPTION("Audio support for codec Philips UDA1380"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h new file mode 100644 index 000000000000..f9d885c8bf01 --- /dev/null +++ b/sound/soc/codecs/uda1380.h @@ -0,0 +1,89 @@ +/* + * Audio support for Philips UDA1380 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Copyright (c) 2005 Giorgio Padrin + */ + +#ifndef _UDA1380_H +#define _UDA1380_H + +#define UDA1380_CLK 0x00 +#define UDA1380_IFACE 0x01 +#define UDA1380_PM 0x02 +#define UDA1380_AMIX 0x03 +#define UDA1380_HP 0x04 +#define UDA1380_MVOL 0x10 +#define UDA1380_MIXVOL 0x11 +#define UDA1380_MODE 0x12 +#define UDA1380_DEEMP 0x13 +#define UDA1380_MIXER 0x14 +#define UDA1380_INTSTAT 0x18 +#define UDA1380_DEC 0x20 +#define UDA1380_PGA 0x21 +#define UDA1380_ADC 0x22 +#define UDA1380_AGC 0x23 +#define UDA1380_DECSTAT 0x28 +#define UDA1380_RESET 0x7f + +#define UDA1380_CACHEREGNUM 0x24 + +/* Register flags */ +#define R00_EN_ADC 0x0800 +#define R00_EN_DEC 0x0400 +#define R00_EN_DAC 0x0200 +#define R00_EN_INT 0x0100 +#define R00_DAC_CLK 0x0010 +#define R01_SFORI_I2S 0x0000 +#define R01_SFORI_LSB16 0x0100 +#define R01_SFORI_LSB18 0x0200 +#define R01_SFORI_LSB20 0x0300 +#define R01_SFORI_MSB 0x0500 +#define R01_SFORI_MASK 0x0700 +#define R01_SFORO_I2S 0x0000 +#define R01_SFORO_LSB16 0x0001 +#define R01_SFORO_LSB18 0x0002 +#define R01_SFORO_LSB20 0x0003 +#define R01_SFORO_LSB24 0x0004 +#define R01_SFORO_MSB 0x0005 +#define R01_SFORO_MASK 0x0007 +#define R01_SEL_SOURCE 0x0040 +#define R01_SIM 0x0010 +#define R02_PON_PLL 0x8000 +#define R02_PON_HP 0x2000 +#define R02_PON_DAC 0x0400 +#define R02_PON_BIAS 0x0100 +#define R02_EN_AVC 0x0080 +#define R02_PON_AVC 0x0040 +#define R02_PON_LNA 0x0010 +#define R02_PON_PGAL 0x0008 +#define R02_PON_ADCL 0x0004 +#define R02_PON_PGAR 0x0002 +#define R02_PON_ADCR 0x0001 +#define R13_MTM 0x4000 +#define R14_SILENCE 0x0080 +#define R14_SDET_ON 0x0040 +#define R21_MT_ADC 0x8000 +#define R22_SEL_LNA 0x0008 +#define R22_SEL_MIC 0x0004 +#define R22_SKIP_DCFIL 0x0002 +#define R23_AGC_EN 0x0001 + +struct uda1380_setup_data { + unsigned short i2c_address; + int dac_clk; +#define UDA1380_DAC_CLK_SYSCLK 0 +#define UDA1380_DAC_CLK_WSPLL 1 +}; + +#define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */ +#define UDA1380_DAI_PLAYBACK 1 /* playback DAI */ +#define UDA1380_DAI_CAPTURE 2 /* capture DAI */ + +extern struct snd_soc_codec_dai uda1380_dai[3]; +extern struct snd_soc_codec_device soc_codec_dev_uda1380; + +#endif /* _UDA1380_H */ From c18f68e4d809a517ed8df540bac2993a4f14d9a4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 29 May 2008 18:53:05 +0200 Subject: [PATCH 053/179] [ALSA] Clean up sound/pci/ac97/Makefile Signed-off-by: Takashi Iwai --- sound/pci/ac97/Makefile | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/pci/ac97/Makefile b/sound/pci/ac97/Makefile index 0be48b1a22d0..ae36950f2568 100644 --- a/sound/pci/ac97/Makefile +++ b/sound/pci/ac97/Makefile @@ -3,16 +3,11 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-ac97-codec-objs := ac97_codec.o ac97_pcm.o - -ifneq ($(CONFIG_PROC_FS),) -snd-ac97-codec-objs += ac97_proc.o -endif +snd-ac97-codec-y := ac97_codec.o ac97_pcm.o +snd-ac97-codec-$(CONFIG_PROC_FS) += ac97_proc.o snd-ak4531-codec-objs := ak4531_codec.o # Toplevel Module Dependency obj-$(CONFIG_SND_AC97_CODEC) += snd-ac97-codec.o obj-$(CONFIG_SND_ENS1370) += snd-ak4531-codec.o - -obj-m := $(sort $(obj-m)) From 23ce1547638443f0053dd674e728062c48ff0e39 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 May 2008 09:22:22 +0200 Subject: [PATCH 054/179] [ALSA] Make ak4531 local to ens1370 driver The ak4531 module is used only by ens1370 driver (and unlikely that any other will use it ever). Let's make it local to ens1370. Signed-off-by: Takashi Iwai --- sound/pci/Makefile | 2 +- sound/pci/ac97/Makefile | 3 --- sound/pci/{ac97 => }/ak4531_codec.c | 32 ++++++----------------------- 3 files changed, 7 insertions(+), 30 deletions(-) rename sound/pci/{ac97 => }/ak4531_codec.c (96%) diff --git a/sound/pci/Makefile b/sound/pci/Makefile index 85ef14bc8056..65b25d221cd2 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -13,7 +13,7 @@ snd-bt87x-objs := bt87x.o snd-cmipci-objs := cmipci.o snd-cs4281-objs := cs4281.o snd-cs5530-objs := cs5530.o -snd-ens1370-objs := ens1370.o +snd-ens1370-objs := ens1370.o ak4531_codec.o snd-ens1371-objs := ens1371.o snd-es1938-objs := es1938.o snd-es1968-objs := es1968.o diff --git a/sound/pci/ac97/Makefile b/sound/pci/ac97/Makefile index ae36950f2568..41fa322f0971 100644 --- a/sound/pci/ac97/Makefile +++ b/sound/pci/ac97/Makefile @@ -6,8 +6,5 @@ snd-ac97-codec-y := ac97_codec.o ac97_pcm.o snd-ac97-codec-$(CONFIG_PROC_FS) += ac97_proc.o -snd-ak4531-codec-objs := ak4531_codec.o - # Toplevel Module Dependency obj-$(CONFIG_SND_AC97_CODEC) += snd-ac97-codec.o -obj-$(CONFIG_SND_ENS1370) += snd-ak4531-codec.o diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ak4531_codec.c similarity index 96% rename from sound/pci/ac97/ak4531_codec.c rename to sound/pci/ak4531_codec.c index c0c1633999ea..6a99eed2d8b6 100644 --- a/sound/pci/ac97/ak4531_codec.c +++ b/sound/pci/ak4531_codec.c @@ -270,7 +270,7 @@ static const DECLARE_TLV_DB_SCALE(db_scale_master, -6200, 200, 0); static const DECLARE_TLV_DB_SCALE(db_scale_mono, -2800, 400, 0); static const DECLARE_TLV_DB_SCALE(db_scale_input, -5000, 200, 0); -static struct snd_kcontrol_new snd_ak4531_controls[] = { +static struct snd_kcontrol_new snd_ak4531_controls[] __devinitdata = { AK4531_DOUBLE_TLV("Master Playback Switch", 0, AK4531_LMASTER, AK4531_RMASTER, 7, 7, 1, 1, @@ -379,8 +379,9 @@ static u8 snd_ak4531_initial_map[0x19 + 1] = { 0x01 /* 19: Mic Amp Setup */ }; -int snd_ak4531_mixer(struct snd_card *card, struct snd_ak4531 *_ak4531, - struct snd_ak4531 **rak4531) +int __devinit snd_ak4531_mixer(struct snd_card *card, + struct snd_ak4531 *_ak4531, + struct snd_ak4531 **rak4531) { unsigned int idx; int err; @@ -476,7 +477,8 @@ static void snd_ak4531_proc_read(struct snd_info_entry *entry, ak4531->regs[AK4531_MIC_GAIN] & 1 ? "+30dB" : "+0dB"); } -static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531) +static void __devinit +snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531) { struct snd_info_entry *entry; @@ -484,25 +486,3 @@ static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak453 snd_info_set_text_ops(entry, ak4531, snd_ak4531_proc_read); } #endif - -EXPORT_SYMBOL(snd_ak4531_mixer); -#ifdef CONFIG_PM -EXPORT_SYMBOL(snd_ak4531_suspend); -EXPORT_SYMBOL(snd_ak4531_resume); -#endif - -/* - * INIT part - */ - -static int __init alsa_ak4531_init(void) -{ - return 0; -} - -static void __exit alsa_ak4531_exit(void) -{ -} - -module_init(alsa_ak4531_init) -module_exit(alsa_ak4531_exit) From 8bb8b453cb458d8f62411e78a4cfd6d860b503b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 May 2008 09:47:45 +0200 Subject: [PATCH 055/179] [ALSA] trident - clean up obsolete synth codes Clean up the unused synth codes in the memory handling of trident driver. Signed-off-by: Takashi Iwai --- sound/pci/trident/trident_memory.c | 178 ----------------------------- 1 file changed, 178 deletions(-) diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index df9b487fa17e..3fd7f1b29b0f 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -310,181 +310,3 @@ int snd_trident_free_pages(struct snd_trident *trident, mutex_unlock(&hdr->block_mutex); return 0; } - - -/*---------------------------------------------------------------- - * memory allocation using multiple pages (for synth) - *---------------------------------------------------------------- - * Unlike the DMA allocation above, non-contiguous pages are - * assigned to TLB. - *----------------------------------------------------------------*/ - -/* - */ -static int synth_alloc_pages(struct snd_trident *hw, struct snd_util_memblk *blk); -static int synth_free_pages(struct snd_trident *hw, struct snd_util_memblk *blk); - -/* - * allocate a synth sample area - */ -struct snd_util_memblk * -snd_trident_synth_alloc(struct snd_trident *hw, unsigned int size) -{ - struct snd_util_memblk *blk; - struct snd_util_memhdr *hdr = hw->tlb.memhdr; - - mutex_lock(&hdr->block_mutex); - blk = __snd_util_mem_alloc(hdr, size); - if (blk == NULL) { - mutex_unlock(&hdr->block_mutex); - return NULL; - } - if (synth_alloc_pages(hw, blk)) { - __snd_util_mem_free(hdr, blk); - mutex_unlock(&hdr->block_mutex); - return NULL; - } - mutex_unlock(&hdr->block_mutex); - return blk; -} - -EXPORT_SYMBOL(snd_trident_synth_alloc); - -/* - * free a synth sample area - */ -int -snd_trident_synth_free(struct snd_trident *hw, struct snd_util_memblk *blk) -{ - struct snd_util_memhdr *hdr = hw->tlb.memhdr; - - mutex_lock(&hdr->block_mutex); - synth_free_pages(hw, blk); - __snd_util_mem_free(hdr, blk); - mutex_unlock(&hdr->block_mutex); - return 0; -} - -EXPORT_SYMBOL(snd_trident_synth_free); - -/* - * reset TLB entry and free kernel page - */ -static void clear_tlb(struct snd_trident *trident, int page) -{ - void *ptr = page_to_ptr(trident, page); - dma_addr_t addr = page_to_addr(trident, page); - set_silent_tlb(trident, page); - if (ptr) { - struct snd_dma_buffer dmab; - dmab.dev.type = SNDRV_DMA_TYPE_DEV; - dmab.dev.dev = snd_dma_pci_data(trident->pci); - dmab.area = ptr; - dmab.addr = addr; - dmab.bytes = ALIGN_PAGE_SIZE; - snd_dma_free_pages(&dmab); - } -} - -/* check new allocation range */ -static void get_single_page_range(struct snd_util_memhdr *hdr, - struct snd_util_memblk *blk, - int *first_page_ret, int *last_page_ret) -{ - struct list_head *p; - struct snd_util_memblk *q; - int first_page, last_page; - first_page = firstpg(blk); - if ((p = blk->list.prev) != &hdr->block) { - q = list_entry(p, struct snd_util_memblk, list); - if (lastpg(q) == first_page) - first_page++; /* first page was already allocated */ - } - last_page = lastpg(blk); - if ((p = blk->list.next) != &hdr->block) { - q = list_entry(p, struct snd_util_memblk, list); - if (firstpg(q) == last_page) - last_page--; /* last page was already allocated */ - } - *first_page_ret = first_page; - *last_page_ret = last_page; -} - -/* - * allocate kernel pages and assign them to TLB - */ -static int synth_alloc_pages(struct snd_trident *hw, struct snd_util_memblk *blk) -{ - int page, first_page, last_page; - struct snd_dma_buffer dmab; - - firstpg(blk) = get_aligned_page(blk->offset); - lastpg(blk) = get_aligned_page(blk->offset + blk->size - 1); - get_single_page_range(hw->tlb.memhdr, blk, &first_page, &last_page); - - /* allocate a kernel page for each Trident page - - * fortunately Trident page size and kernel PAGE_SIZE is identical! - */ - for (page = first_page; page <= last_page; page++) { - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(hw->pci), - ALIGN_PAGE_SIZE, &dmab) < 0) - goto __fail; - if (! is_valid_page(dmab.addr)) { - snd_dma_free_pages(&dmab); - goto __fail; - } - set_tlb_bus(hw, page, (unsigned long)dmab.area, dmab.addr); - } - return 0; - -__fail: - /* release allocated pages */ - last_page = page - 1; - for (page = first_page; page <= last_page; page++) - clear_tlb(hw, page); - - return -ENOMEM; -} - -/* - * free pages - */ -static int synth_free_pages(struct snd_trident *trident, struct snd_util_memblk *blk) -{ - int page, first_page, last_page; - - get_single_page_range(trident->tlb.memhdr, blk, &first_page, &last_page); - for (page = first_page; page <= last_page; page++) - clear_tlb(trident, page); - - return 0; -} - -/* - * copy_from_user(blk + offset, data, size) - */ -int snd_trident_synth_copy_from_user(struct snd_trident *trident, - struct snd_util_memblk *blk, - int offset, const char __user *data, int size) -{ - int page, nextofs, end_offset, temp, temp1; - - offset += blk->offset; - end_offset = offset + size; - page = get_aligned_page(offset) + 1; - do { - nextofs = aligned_page_offset(page); - temp = nextofs - offset; - temp1 = end_offset - offset; - if (temp1 < temp) - temp = temp1; - if (copy_from_user(offset_ptr(trident, offset), data, temp)) - return -EFAULT; - offset = nextofs; - data += temp; - page++; - } while (offset < end_offset); - return 0; -} - -EXPORT_SYMBOL(snd_trident_synth_copy_from_user); From 72860b0f3c4a303498c9e16c8a4a38ddf23d56de Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 May 2008 11:25:03 +0200 Subject: [PATCH 056/179] sound: Clean up sound/oss/Kconfig Removed unnecessary dependencies, fix a wrong selection, and make CONFIG_SOUND_OSS menuconfig to simplify. Signed-off-by: Takashi Iwai --- sound/oss/Kconfig | 49 ++++++++++++++++++----------------------------- 1 file changed, 19 insertions(+), 30 deletions(-) diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 3be2dc1025b5..33940139844b 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -7,7 +7,7 @@ config SOUND_BCM_CS4297A tristate "Crystal Sound CS4297a (for Swarm)" - depends on SOUND_PRIME && SIBYTE_SWARM + depends on SIBYTE_SWARM help The BCM91250A has a Crystal CS4297a on synchronous serial port B (in addition to the DB-9 serial port). Say Y or M @@ -17,7 +17,7 @@ config SOUND_BCM_CS4297A config SOUND_VWSND tristate "SGI Visual Workstation Sound" - depends on SOUND_PRIME && X86_VISWS + depends on X86_VISWS help Say Y or M if you have an SGI Visual Workstation and you want to be able to use its on-board audio. Read @@ -26,19 +26,18 @@ config SOUND_VWSND config SOUND_HAL2 tristate "SGI HAL2 sound (EXPERIMENTAL)" - depends on SOUND_PRIME && SGI_IP22 && EXPERIMENTAL + depends on SGI_IP22 && EXPERIMENTAL help Say Y or M if you have an SGI Indy or Indigo2 system and want to be able to use its on-board A2 audio system. config SOUND_AU1550_AC97 tristate "Au1550/Au1200 AC97 Sound" - select SND_AC97_CODEC - depends on SOUND_PRIME && (SOC_AU1550 || SOC_AU1200) + depends on SOC_AU1550 || SOC_AU1200 config SOUND_TRIDENT tristate "Trident 4DWave DX/NX, SiS 7018 or ALi 5451 PCI Audio Core" - depends on SOUND_PRIME && PCI + depends on PCI ---help--- Say Y or M if you have a PCI sound card utilizing the Trident 4DWave-DX/NX chipset or your mother board chipset has SiS 7018 @@ -79,7 +78,7 @@ config SOUND_TRIDENT config SOUND_MSNDCLAS tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey" - depends on SOUND_PRIME && (m || !STANDALONE) && ISA + depends on (m || !STANDALONE) && ISA help Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or Monterey (not for the Pinnacle or Fiji). @@ -143,7 +142,7 @@ config MSNDCLAS_IO config SOUND_MSNDPIN tristate "Support for Turtle Beach MultiSound Pinnacle, Fiji" - depends on SOUND_PRIME && (m || !STANDALONE) && ISA + depends on (m || !STANDALONE) && ISA help Say M here if you have a Turtle Beach MultiSound Pinnacle or Fiji. See for important information @@ -229,7 +228,7 @@ config MSNDPIN_NONPNP configure the card's resources. comment "MSND Pinnacle DSP section will be configured to above parameters." - depends on SOUND_PRIME && SOUND_MSNDPIN=y && MSNDPIN_NONPNP + depends on SOUND_MSNDPIN=y && MSNDPIN_NONPNP config MSNDPIN_CFG hex "MSND Pinnacle config port 250,260,270" @@ -242,7 +241,7 @@ config MSNDPIN_CFG Mode". comment "Pinnacle-specific Device Configuration (0 disables)" - depends on SOUND_PRIME && SOUND_MSNDPIN=y && MSNDPIN_NONPNP + depends on SOUND_MSNDPIN=y && MSNDPIN_NONPNP config MSNDPIN_MPU_IO hex "MSND Pinnacle MPU I/O (e.g. 330)" @@ -294,7 +293,7 @@ config MSNDPIN_JOYSTICK_IO config MSND_FIFOSIZE int "MSND buffer size (kB)" - depends on SOUND_PRIME && (SOUND_MSNDPIN=y || SOUND_MSNDCLAS=y) + depends on SOUND_MSNDPIN=y || SOUND_MSNDCLAS=y default "128" help Configures the size of each audio buffer, in kilobytes, for @@ -302,9 +301,9 @@ config MSND_FIFOSIZE and Pinnacle). Larger values reduce the chance of data overruns at the expense of overall latency. If unsure, use the default. -config SOUND_OSS +menuconfig SOUND_OSS tristate "OSS sound modules" - depends on SOUND_PRIME && ISA_DMA_API && VIRT_TO_BUS + depends on ISA_DMA_API && VIRT_TO_BUS help OSS is the Open Sound System suite of sound card drivers. They make sound programming easier since they provide a common API. Say Y or @@ -312,16 +311,16 @@ config SOUND_OSS driver for your sound card above, then pick your driver from the list below. +if SOUND_OSS + config SOUND_TRACEINIT bool "Verbose initialisation" - depends on SOUND_OSS help Verbose soundcard initialization -- affects the format of autoprobe and initialization messages at boot time. config SOUND_DMAP bool "Persistent DMA buffers" - depends on SOUND_OSS ---help--- Linux can often have problems allocating DMA buffers for ISA sound cards on machines with more than 16MB of RAM. This is because ISA @@ -338,8 +337,6 @@ config SOUND_DMAP config SOUND_SSCAPE tristate "Ensoniq SoundScape support" - depends on SOUND_OSS - depends on VIRT_TO_BUS help Answer Y if you have a sound card based on the Ensoniq SoundScape chipset. Such cards are being manufactured at least by Ensoniq, Spea @@ -352,13 +349,11 @@ config SOUND_SSCAPE config SOUND_VMIDI tristate "Loopback MIDI device support" - depends on SOUND_OSS help Support for MIDI loopback on port 1 or 2. config SOUND_TRIX tristate "MediaTrix AudioTrix Pro support" - depends on SOUND_OSS help Answer Y if you have the AudioTriX Pro sound card manufactured by MediaTrix. @@ -382,7 +377,6 @@ config TRIX_BOOT_FILE config SOUND_MSS tristate "Microsoft Sound System support" - depends on SOUND_OSS ---help--- Again think carefully before answering Y to this question. It's safe to answer Y if you have the original Windows Sound System card @@ -414,7 +408,6 @@ config SOUND_MSS config SOUND_MPU401 tristate "MPU-401 support (NOT for SB16)" - depends on SOUND_OSS ---help--- Be careful with this question. The MPU401 interface is supported by all sound cards. However, some natively supported cards have their @@ -430,7 +423,6 @@ config SOUND_MPU401 config SOUND_PAS tristate "ProAudioSpectrum 16 support" - depends on SOUND_OSS ---help--- Answer Y only if you have a Pro Audio Spectrum 16, ProAudio Studio 16 or Logitech SoundMan 16 sound card. Answer N if you have some @@ -452,7 +444,6 @@ config PAS_JOYSTICK config SOUND_PSS tristate "PSS (AD1848, ADSP-2115, ESC614) support" - depends on SOUND_OSS help Answer Y or M if you have an Orchid SW32, Cardinal DSP16, Beethoven ADSP-16 or some other card based on the PSS chipset (AD1848 codec + @@ -495,7 +486,6 @@ config PSS_BOOT_FILE config SOUND_SB tristate "100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support" - depends on SOUND_OSS ---help--- Answer Y if you have an original Sound Blaster card made by Creative Labs or a 100% hardware compatible clone (like the Thunderboard or @@ -522,7 +512,6 @@ config SOUND_SB config SOUND_YM3812 tristate "Yamaha FM synthesizer (YM3812/OPL-3) support" - depends on SOUND_OSS ---help--- Answer Y if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4). Answering Y is usually a safe and recommended choice, however some @@ -538,7 +527,6 @@ config SOUND_YM3812 config SOUND_UART6850 tristate "6850 UART support" - depends on SOUND_OSS help This option enables support for MIDI interfaces based on the 6850 UART chip. This interface is rarely found on sound cards. It's safe @@ -549,7 +537,6 @@ config SOUND_UART6850 config SOUND_AEDSP16 tristate "Gallant Audio Cards (SC-6000 and SC-6600 based)" - depends on SOUND_OSS ---help--- Answer Y if you have a Gallant's Audio Excel DSP 16 card. This driver supports Audio Excel DSP 16 but not the III nor PnP versions @@ -630,14 +617,14 @@ endchoice config SOUND_VIDC tristate "VIDC 16-bit sound" - depends on ARM && (ARCH_ACORN || ARCH_CLPS7500) && SOUND_OSS + depends on ARM && (ARCH_ACORN || ARCH_CLPS7500) help 16-bit support for the VIDC onboard sound hardware found on Acorn machines. config SOUND_WAVEARTIST tristate "Netwinder WaveArtist" - depends on ARM && SOUND_OSS && ARCH_NETWINDER + depends on ARM && ARCH_NETWINDER help Say Y here to include support for the Rockwell WaveArtist sound system. This driver is mainly for the NetWinder. @@ -646,9 +633,11 @@ config SOUND_KAHLUA tristate "XpressAudio Sound Blaster emulation" depends on SOUND_SB +endif # SOUND_OSS + config SOUND_SH_DAC_AUDIO tristate "SuperH DAC audio support" - depends on SOUND_PRIME && CPU_SH3 + depends on CPU_SH3 config SOUND_SH_DAC_AUDIO_CHANNEL int "DAC channel" From 2621f0338ce4e3e57cc32a967f5a3d2999390fe3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 May 2008 11:27:24 +0200 Subject: [PATCH 057/179] [ALSA] trivial clean up of sound/isa/sb/Makefile Remove unneeded sort in sound/isa/sb/Makefile. Signed-off-by: Takashi Iwai --- sound/isa/sb/Makefile | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index c9d1c986d70e..1098a56b2f4b 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -34,5 +34,3 @@ ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o endif obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-emu8000-synth.o - -obj-m := $(sort $(obj-m)) From a5003fc04113c217370409beac812831cbf6e0ac Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 May 2008 09:49:41 +0200 Subject: [PATCH 058/179] [ALSA] emu10k1 - simplify page allocation for synth Simplify the page allocation of emu10k1 driver for emux synth support. Since these pages aren't be necessarily coherent, we can avoid expensive DMA-coherent routines. Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/memory.c | 66 ++++++++++++++++---------------------- 1 file changed, 28 insertions(+), 38 deletions(-) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 916c1dbcd53c..128eaca17a61 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -437,43 +437,46 @@ static void get_single_page_range(struct snd_util_memhdr *hdr, *last_page_ret = last_page; } +/* release allocated pages */ +static void __synth_free_pages(struct snd_emu10k1 *emu, int first_page, + int last_page) +{ + int page; + + for (page = first_page; page <= last_page; page++) { + free_page((unsigned long)emu->page_ptr_table[page]); + emu->page_addr_table[page] = 0; + emu->page_ptr_table[page] = NULL; + } +} + /* * allocate kernel pages */ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) { int page, first_page, last_page; - struct snd_dma_buffer dmab; emu10k1_memblk_init(blk); get_single_page_range(emu->memhdr, blk, &first_page, &last_page); /* allocate kernel pages */ for (page = first_page; page <= last_page; page++) { - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(emu->pci), - PAGE_SIZE, &dmab) < 0) - goto __fail; - if (! is_valid_page(emu, dmab.addr)) { - snd_dma_free_pages(&dmab); - goto __fail; + /* first try to allocate from <4GB zone */ + struct page *p = alloc_page(GFP_KERNEL | GFP_DMA32 | + __GFP_NOWARN); + if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) + /* try to allocate from <16MB zone */ + p = alloc_page(GFP_DMA | + __GFP_NORETRY | /* no OOM-killer */ + __GFP_NOWARN); + if (!p) { + __synth_free_pages(emu, first_page, page - 1); + return -ENOMEM; } - emu->page_addr_table[page] = dmab.addr; - emu->page_ptr_table[page] = dmab.area; + emu->page_addr_table[page] = page_to_phys(p); + emu->page_ptr_table[page] = page_address(p); } return 0; - -__fail: - /* release allocated pages */ - last_page = page - 1; - for (page = first_page; page <= last_page; page++) { - dmab.area = emu->page_ptr_table[page]; - dmab.addr = emu->page_addr_table[page]; - dmab.bytes = PAGE_SIZE; - snd_dma_free_pages(&dmab); - emu->page_addr_table[page] = 0; - emu->page_ptr_table[page] = NULL; - } - - return -ENOMEM; } /* @@ -481,23 +484,10 @@ __fail: */ static int synth_free_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) { - int page, first_page, last_page; - struct snd_dma_buffer dmab; + int first_page, last_page; get_single_page_range(emu->memhdr, blk, &first_page, &last_page); - dmab.dev.type = SNDRV_DMA_TYPE_DEV; - dmab.dev.dev = snd_dma_pci_data(emu->pci); - for (page = first_page; page <= last_page; page++) { - if (emu->page_ptr_table[page] == NULL) - continue; - dmab.area = emu->page_ptr_table[page]; - dmab.addr = emu->page_addr_table[page]; - dmab.bytes = PAGE_SIZE; - snd_dma_free_pages(&dmab); - emu->page_addr_table[page] = 0; - emu->page_ptr_table[page] = NULL; - } - + __synth_free_pages(emu, first_page, last_page); return 0; } From d2cd74b158d7214a556226e3312f9fb1de64d7ae Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Jun 2008 11:45:53 +0200 Subject: [PATCH 059/179] [ALSA] emu10k1 - Fix inverted Analog/Digital mixer switch on Audigy2 On Audigy2 Platinum, the Analog/Digital mixer switch is inverted. https://bugzilla.novell.com/show_bug.cgi?id=396204 The patch adds a simple workaround. There might be another device requiring a similar fix, too (or fix for audigy2 generically), but right now I fix only the known broken one. Signed-off-by: Takashi Iwai --- include/sound/emu10k1.h | 1 + sound/pci/emu10k1/emu10k1_main.c | 1 + sound/pci/emu10k1/emumixer.c | 13 ++++++++++--- 3 files changed, 12 insertions(+), 3 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 7b7b9b13b4dd..10ee28eac018 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1670,6 +1670,7 @@ struct snd_emu_chip_details { unsigned char spi_dac; /* SPI interface for DAC */ unsigned char i2c_adc; /* I2C interface for ADC */ unsigned char adc_1361t; /* Use Philips 1361T ADC */ + unsigned char invert_shared_spdif; /* analog/digital switch inverted */ const char *driver; const char *name; const char *id; /* for backward compatibility - can be NULL if not needed */ diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 548c9cc81af5..2f283ea6ad9a 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1528,6 +1528,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .adc_1361t = 1, /* 24 bit capture instead of 16bit. Fixes ALSA bug#324 */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .revision = 0x04, diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index fd221209abcb..f34bbfb705f5 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1578,6 +1578,10 @@ static int snd_emu10k1_shared_spdif_get(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = inl(emu->port + A_IOCFG) & A_IOCFG_GPOUT0 ? 1 : 0; else ucontrol->value.integer.value[0] = inl(emu->port + HCFG) & HCFG_GPOUT0 ? 1 : 0; + if (emu->card_capabilities->invert_shared_spdif) + ucontrol->value.integer.value[0] = + !ucontrol->value.integer.value[0]; + return 0; } @@ -1586,15 +1590,18 @@ static int snd_emu10k1_shared_spdif_put(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - unsigned int reg, val; + unsigned int reg, val, sw; int change = 0; + sw = ucontrol->value.integer.value[0]; + if (emu->card_capabilities->invert_shared_spdif) + sw = !sw; spin_lock_irqsave(&emu->reg_lock, flags); if ( emu->card_capabilities->i2c_adc) { /* Do nothing for Audigy 2 ZS Notebook */ } else if (emu->audigy) { reg = inl(emu->port + A_IOCFG); - val = ucontrol->value.integer.value[0] ? A_IOCFG_GPOUT0 : 0; + val = sw ? A_IOCFG_GPOUT0 : 0; change = (reg & A_IOCFG_GPOUT0) != val; if (change) { reg &= ~A_IOCFG_GPOUT0; @@ -1603,7 +1610,7 @@ static int snd_emu10k1_shared_spdif_put(struct snd_kcontrol *kcontrol, } } reg = inl(emu->port + HCFG); - val = ucontrol->value.integer.value[0] ? HCFG_GPOUT0 : 0; + val = sw ? HCFG_GPOUT0 : 0; change |= (reg & HCFG_GPOUT0) != val; if (change) { reg &= ~HCFG_GPOUT0; From 5a85bb02ba473aacd29f4039a74744f729443b31 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 3 Jun 2008 09:06:20 +0200 Subject: [PATCH 060/179] ALSA: Release v1.0.17rc1 Signed-off-by: Jaroslav Kysela --- include/sound/version.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/version.h b/include/sound/version.h index ed6fb2eb1eac..2474b8ee99f9 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ -/* include/version.h. Generated by alsa/ksync script. */ -#define CONFIG_SND_VERSION "1.0.16" +/* include/version.h */ +#define CONFIG_SND_VERSION "1.0.17rc1" #define CONFIG_SND_DATE "" From 17bba1b72d190742c99a140154f0abac9c1996c3 Mon Sep 17 00:00:00 2001 From: Jiang zhe Date: Wed, 4 Jun 2008 12:11:07 +0200 Subject: [PATCH 061/179] [ALSA] hda - support intel DG33 motherboards These two motherboards's pin configuration are not covered by driver. I wrote a new model to support them. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 107 ++++++++++++++++++++++++++++++++++ 1 file changed, 107 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cb3e0283f0ce..048cb84721fa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -209,6 +209,7 @@ enum { ALC883_MITAC, ALC883_CLEVO_M720, ALC883_FUJITSU_PI2515, + ALC883_3ST_6ch_INTEL, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6603,6 +6604,16 @@ static struct hda_input_mux alc883_capture_source = { }, }; +static struct hda_input_mux alc883_3stack_6ch_intel = { + .num_items = 4, + .items = { + { "Mic", 0x1 }, + { "Front Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + static struct hda_input_mux alc883_lenovo_101e_capture_source = { .num_items = 2, .items = { @@ -6683,6 +6694,48 @@ static struct hda_channel_mode alc883_3ST_6ch_modes[3] = { { 6, alc883_3ST_ch6_init }, }; +/* + * 2ch mode + */ +static struct hda_verb alc883_3ST_ch2_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* + * 4ch mode + */ +static struct hda_verb alc883_3ST_ch4_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ +static struct hda_verb alc883_3ST_ch6_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = { + { 2, alc883_3ST_ch2_intel_init }, + { 4, alc883_3ST_ch4_intel_init }, + { 6, alc883_3ST_ch6_intel_init }, +}; + /* * 6ch mode */ @@ -6915,6 +6968,45 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, + HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_fivestack_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -7763,6 +7855,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_MITAC] = "mitac", [ALC883_CLEVO_M720] = "clevo-m720", [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515", + [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", [ALC883_AUTO] = "auto", }; @@ -7820,6 +7913,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), + SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), + SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), {} }; @@ -7858,6 +7953,18 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, + [ALC883_3ST_6ch_INTEL] = { + .mixers = { alc883_3ST_6ch_intel_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes), + .channel_mode = alc883_3ST_6ch_intel_modes, + .need_dac_fix = 1, + .input_mux = &alc883_3stack_6ch_intel, + }, [ALC883_6ST_DIG] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs }, From 2c3bf9abb11dd8050cd2d153917d1746c8d5af05 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jun 2008 12:39:38 +0200 Subject: [PATCH 062/179] [ALSA] hda - Fix PLL gating control on Realtek codecs On some Realtek codecs, the analog PLL gating control bit must be set off while the default value is 1. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 46 +++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 048cb84721fa..7997e13b59a6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -285,6 +285,10 @@ struct alc_spec { #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif + + /* for PLL fix */ + hda_nid_t pll_nid; + unsigned int pll_coef_idx, pll_coef_bit; }; /* @@ -752,6 +756,38 @@ static struct hda_verb alc_gpio3_init_verbs[] = { { } }; +/* + * Fix hardware PLL issue + * On some codecs, the analog PLL gating control must be off while + * the default value is 1. + */ +static void alc_fix_pll(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int val; + + if (!spec->pll_nid) + return; + snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX, + spec->pll_coef_idx); + val = snd_hda_codec_read(codec, spec->pll_nid, 0, + AC_VERB_GET_PROC_COEF, 0); + snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX, + spec->pll_coef_idx); + snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_PROC_COEF, + val & ~(1 << spec->pll_coef_bit)); +} + +static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, + unsigned int coef_idx, unsigned int coef_bit) +{ + struct alc_spec *spec = codec->spec; + spec->pll_nid = nid; + spec->pll_coef_idx = coef_idx; + spec->pll_coef_bit = coef_bit; + alc_fix_pll(codec); +} + static void alc_sku_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -2400,6 +2436,8 @@ static int alc_init(struct hda_codec *codec) struct alc_spec *spec = codec->spec; unsigned int i; + alc_fix_pll(codec); + for (i = 0; i < spec->num_init_verbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); @@ -8286,6 +8324,8 @@ static int patch_alc883(struct hda_codec *codec) codec->spec = spec; + alc_fix_pll_init(codec, 0x20, 0x0a, 10); + board_config = snd_hda_check_board_config(codec, ALC883_MODEL_LAST, alc883_models, alc883_cfg_tbl); @@ -9886,6 +9926,8 @@ static int patch_alc262(struct hda_codec *codec) } #endif + alc_fix_pll_init(codec, 0x20, 0x0a, 10); + board_config = snd_hda_check_board_config(codec, ALC262_MODEL_LAST, alc262_models, alc262_cfg_tbl); @@ -11196,6 +11238,8 @@ static int patch_alc269(struct hda_codec *codec) codec->spec = spec; + alc_fix_pll_init(codec, 0x20, 0x04, 15); + board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST, alc269_models, alc269_cfg_tbl); @@ -14531,6 +14575,8 @@ static int patch_alc662(struct hda_codec *codec) codec->spec = spec; + alc_fix_pll_init(codec, 0x20, 0x04, 15); + board_config = snd_hda_check_board_config(codec, ALC662_MODEL_LAST, alc662_models, alc662_cfg_tbl); From 607d982bbea2a14b5b77cc7689f509d588e1e6da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jun 2008 12:41:21 +0200 Subject: [PATCH 063/179] [ALSA] hda - increase max_codecs of ICH to 4 It turned out that some ICH9-based boards use SD3 for the audio codec where the current driver code doesn't probe. Since we have a better codec slot check now, it must be safe to increase this to 4. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6e2dc4f5ca7a..dc68709e7569 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1104,7 +1104,7 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) */ static unsigned int azx_max_codecs[] __devinitdata = { - [AZX_DRIVER_ICH] = 3, + [AZX_DRIVER_ICH] = 4, /* Some ICH9 boards use SD3 */ [AZX_DRIVER_SCH] = 3, [AZX_DRIVER_ATI] = 4, [AZX_DRIVER_ATIHDMI] = 4, From 8b83afe0d21f07145ec34ac006656e4d3edc4bac Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Ville=20Syrj=C3=A4l=C3=A4?= Date: Tue, 3 Jun 2008 20:52:10 +0300 Subject: [PATCH 064/179] [ALSA] maestro3: Fix hw volume on HP OmniBook Make the hw volume buttons work correctly on some HP OmniBook laptops. The original quirk was apparently applied a bit too early and it was also lacking some critial register writes. This improved sequence was discovered by trial and error (like the original sequence). Tested and found working on OB500 and OB6000 laptops. Signed-off-by: Ville Syrjala Signed-off-by: Takashi Iwai --- sound/pci/maestro3.c | 42 +++++++++++++++++++++++++++--------------- 1 file changed, 27 insertions(+), 15 deletions(-) diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index a536c59fbea1..f4788dee05c3 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2427,6 +2427,29 @@ snd_m3_amp_enable(struct snd_m3 *chip, int enable) outw(0xffff, io + GPIO_MASK); } +static void +snd_m3_hv_init(struct snd_m3 *chip) +{ + unsigned long io = chip->iobase; + u16 val = GPI_VOL_DOWN | GPI_VOL_UP; + + if (!chip->is_omnibook) + return; + + /* + * Volume buttons on some HP OmniBook laptops + * require some GPIO magic to work correctly. + */ + outw(0xffff, io + GPIO_MASK); + outw(0x0000, io + GPIO_DATA); + + outw(~val, io + GPIO_MASK); + outw(inw(io + GPIO_DIRECTION) & ~val, io + GPIO_DIRECTION); + outw(val, io + GPIO_MASK); + + outw(0xffff, io + GPIO_MASK); +} + static int snd_m3_chip_init(struct snd_m3 *chip) { @@ -2442,21 +2465,6 @@ snd_m3_chip_init(struct snd_m3 *chip) DISABLE_LEGACY); pci_write_config_word(pcidev, PCI_LEGACY_AUDIO_CTRL, w); - if (chip->is_omnibook) { - /* - * Volume buttons on some HP OmniBook laptops don't work - * correctly. This makes them work for the most part. - * - * Volume up and down buttons on the laptop side work. - * Fn+cursor_up (volme up) works. - * Fn+cursor_down (volume down) doesn't work. - * Fn+F7 (mute) works acts as volume up. - */ - outw(~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_MASK); - outw(inw(io + GPIO_DIRECTION) & ~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DIRECTION); - outw((GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DATA); - outw(0xffff, io + GPIO_MASK); - } pci_read_config_dword(pcidev, PCI_ALLEGRO_CONFIG, &n); n &= ~(HV_CTRL_ENABLE | REDUCED_DEBOUNCE | HV_BUTTON_FROM_GD); n |= chip->hv_config; @@ -2642,6 +2650,8 @@ static int m3_resume(struct pci_dev *pci) snd_m3_enable_ints(chip); snd_m3_amp_enable(chip, 1); + snd_m3_hv_init(chip); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } @@ -2781,6 +2791,8 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, snd_m3_amp_enable(chip, 1); + snd_m3_hv_init(chip); + tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip); if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED, From 5d421516670e8009436e299bd25cff1a6d3a4707 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 5 Jun 2008 13:49:32 +0100 Subject: [PATCH 065/179] [ALSA] ASoC: Add WM8510 driver The WM8510 is a mono CODEC with speaker driver optimised for telephony applications, featuring: - 16/20/24/32 bit audio at data rates between 8kHz and 48kHz - On-chip PLL - Dual microphone inputs This driver was originally written by Liam Girdwood with updates from Brett Saunders, Geoffrey Wossum and myself. Signed-off-by: Liam Girdwood Signed-off-by: Brett Saunders Signed-off-by: Geoffrey Wossum Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8510.c | 836 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8510.h | 103 +++++ 4 files changed, 944 insertions(+) create mode 100644 sound/soc/codecs/wm8510.c create mode 100644 sound/soc/codecs/wm8510.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7beefccfa821..ef400b363440 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -5,6 +5,9 @@ config SND_SOC_AC97_CODEC config SND_SOC_UDA1380 tristate +config SND_SOC_WM8510 + tristate + config SND_SOC_WM8731 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d5926a117078..b92c66541663 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,5 +1,6 @@ snd-soc-ac97-objs := ac97.o snd-soc-uda1380-objs := uda1380.o +snd-soc-wm8510-objs := wm8510.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o @@ -10,6 +11,7 @@ snd-soc-tlv320aic3x-objs := tlv320aic3x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o +obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c new file mode 100644 index 000000000000..6f54cb0f50fd --- /dev/null +++ b/sound/soc/codecs/wm8510.c @@ -0,0 +1,836 @@ +/* + * wm8510.c -- WM8510 ALSA Soc Audio driver + * + * Copyright 2006 Wolfson Microelectronics PLC. + * + * Author: Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8510.h" + +#define AUDIO_NAME "wm8510" +#define WM8510_VERSION "0.6" + +/* + * Debug + */ + +#define WM8510_DEBUG 0 + +#ifdef WM8510_DEBUG +#define dbg(format, arg...) \ + printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) +#else +#define dbg(format, arg...) do {} while (0) +#endif +#define err(format, arg...) \ + printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) +#define info(format, arg...) \ + printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) +#define warn(format, arg...) \ + printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) + +struct snd_soc_codec_device soc_codec_dev_wm8510; + +/* + * wm8510 register cache + * We can't read the WM8510 register space when we are + * using 2 wire for device control, so we cache them instead. + */ +static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0050, 0x0000, 0x0140, 0x0000, + 0x0000, 0x0000, 0x0000, 0x00ff, + 0x0000, 0x0000, 0x0100, 0x00ff, + 0x0000, 0x0000, 0x012c, 0x002c, + 0x002c, 0x002c, 0x002c, 0x0000, + 0x0032, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0038, 0x000b, 0x0032, 0x0000, + 0x0008, 0x000c, 0x0093, 0x00e9, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0003, 0x0010, 0x0000, 0x0000, + 0x0000, 0x0002, 0x0001, 0x0000, + 0x0000, 0x0000, 0x0039, 0x0000, + 0x0000, +}; + +/* + * read wm8510 register cache + */ +static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == WM8510_RESET) + return 0; + if (reg >= WM8510_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write wm8510 register cache + */ +static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8510_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the WM8510 register space + */ +static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8510 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8510_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define wm8510_reset(c) wm8510_write(c, WM8510_RESET, 0) + +static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" }; +static const char *wm8510_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; +static const char *wm8510_alc[] = { "ALC", "Limiter" }; + +static const struct soc_enum wm8510_enum[] = { + SOC_ENUM_SINGLE(WM8510_COMP, 1, 4, wm8510_companding), /* adc */ + SOC_ENUM_SINGLE(WM8510_COMP, 3, 4, wm8510_companding), /* dac */ + SOC_ENUM_SINGLE(WM8510_DAC, 4, 4, wm8510_deemp), + SOC_ENUM_SINGLE(WM8510_ALC3, 8, 2, wm8510_alc), +}; + +static const struct snd_kcontrol_new wm8510_snd_controls[] = { + +SOC_SINGLE("Digital Loopback Switch", WM8510_COMP, 0, 1, 0), + +SOC_ENUM("DAC Companding", wm8510_enum[1]), +SOC_ENUM("ADC Companding", wm8510_enum[0]), + +SOC_ENUM("Playback De-emphasis", wm8510_enum[2]), +SOC_SINGLE("DAC Inversion Switch", WM8510_DAC, 0, 1, 0), + +SOC_SINGLE("Master Playback Volume", WM8510_DACVOL, 0, 127, 0), + +SOC_SINGLE("High Pass Filter Switch", WM8510_ADC, 8, 1, 0), +SOC_SINGLE("High Pass Cut Off", WM8510_ADC, 4, 7, 0), +SOC_SINGLE("ADC Inversion Switch", WM8510_COMP, 0, 1, 0), + +SOC_SINGLE("Capture Volume", WM8510_ADCVOL, 0, 127, 0), + +SOC_SINGLE("DAC Playback Limiter Switch", WM8510_DACLIM1, 8, 1, 0), +SOC_SINGLE("DAC Playback Limiter Decay", WM8510_DACLIM1, 4, 15, 0), +SOC_SINGLE("DAC Playback Limiter Attack", WM8510_DACLIM1, 0, 15, 0), + +SOC_SINGLE("DAC Playback Limiter Threshold", WM8510_DACLIM2, 4, 7, 0), +SOC_SINGLE("DAC Playback Limiter Boost", WM8510_DACLIM2, 0, 15, 0), + +SOC_SINGLE("ALC Enable Switch", WM8510_ALC1, 8, 1, 0), +SOC_SINGLE("ALC Capture Max Gain", WM8510_ALC1, 3, 7, 0), +SOC_SINGLE("ALC Capture Min Gain", WM8510_ALC1, 0, 7, 0), + +SOC_SINGLE("ALC Capture ZC Switch", WM8510_ALC2, 8, 1, 0), +SOC_SINGLE("ALC Capture Hold", WM8510_ALC2, 4, 7, 0), +SOC_SINGLE("ALC Capture Target", WM8510_ALC2, 0, 15, 0), + +SOC_ENUM("ALC Capture Mode", wm8510_enum[3]), +SOC_SINGLE("ALC Capture Decay", WM8510_ALC3, 4, 15, 0), +SOC_SINGLE("ALC Capture Attack", WM8510_ALC3, 0, 15, 0), + +SOC_SINGLE("ALC Capture Noise Gate Switch", WM8510_NGATE, 3, 1, 0), +SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8510_NGATE, 0, 7, 0), + +SOC_SINGLE("Capture PGA ZC Switch", WM8510_INPPGA, 7, 1, 0), +SOC_SINGLE("Capture PGA Volume", WM8510_INPPGA, 0, 63, 0), + +SOC_SINGLE("Speaker Playback ZC Switch", WM8510_SPKVOL, 7, 1, 0), +SOC_SINGLE("Speaker Playback Switch", WM8510_SPKVOL, 6, 1, 1), +SOC_SINGLE("Speaker Playback Volume", WM8510_SPKVOL, 0, 63, 0), +SOC_SINGLE("Speaker Boost", WM8510_OUTPUT, 2, 1, 0), + +SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0), +SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1), +}; + +/* add non dapm controls */ +static int wm8510_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8510_snd_controls[i], codec, + NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Speaker Output Mixer */ +static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 0), +}; + +/* Mono Output Mixer */ +static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8510_boost_controls[] = { +SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 0), +SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0), +SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0), +}; + +static const struct snd_kcontrol_new wm8510_micpga_controls[] = { +SOC_DAPM_SINGLE("MICP Switch", WM8510_INPUT, 0, 1, 0), +SOC_DAPM_SINGLE("MICN Switch", WM8510_INPUT, 1, 1, 0), +SOC_DAPM_SINGLE("AUX Switch", WM8510_INPUT, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8510_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Speaker Mixer", WM8510_POWER3, 2, 0, + &wm8510_speaker_mixer_controls[0], + ARRAY_SIZE(wm8510_speaker_mixer_controls)), +SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_POWER3, 3, 0, + &wm8510_mono_mixer_controls[0], + ARRAY_SIZE(wm8510_mono_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER2, 0, 0), +SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, + &wm8510_micpga_controls[0], + ARRAY_SIZE(wm8510_micpga_controls)), +SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, + &wm8510_boost_controls[0], + ARRAY_SIZE(wm8510_boost_controls)), + +SND_SOC_DAPM_MICBIAS("Mic Bias", WM8510_POWER1, 4, 0), + +SND_SOC_DAPM_INPUT("MICN"), +SND_SOC_DAPM_INPUT("MICP"), +SND_SOC_DAPM_INPUT("AUX"), +SND_SOC_DAPM_OUTPUT("MONOOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, + {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, + {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Microphone PGA */ + {"Mic PGA", "MICN Switch", "MICN"}, + {"Mic PGA", "MICP Switch", "MICP"}, + { "Mic PGA", "AUX Switch", "Aux Input" }, + + /* Boost Mixer */ + {"Boost Mixer", "Mic PGA Switch", "Mic PGA"}, + {"Boost Mixer", "Mic Volume", "MICP"}, + {"Boost Mixer", "Aux Volume", "Aux Input"}, + + {"ADC", NULL, "Boost Mixer"}, +}; + +static int wm8510_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8510_dapm_widgets, + ARRAY_SIZE(wm8510_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +struct pll_ { + unsigned int pre_div:4; /* prescale - 1 */ + unsigned int n:4; + unsigned int k; +}; + +static struct pll_ pll_div; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 24) * 10) + +static void pll_factors(unsigned int target, unsigned int source) +{ + unsigned long long Kpart; + unsigned int K, Ndiv, Nmod; + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div.pre_div = 1; + Ndiv = target / source; + } else + pll_div.pre_div = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8510 N value %d outwith recommended range!d\n", + Ndiv); + + pll_div.n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div.k = K; +} + +static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + if (freq_in == 0 || freq_out == 0) { + /* Clock CODEC directly from MCLK */ + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); + wm8510_write(codec, WM8510_CLOCK, reg & 0x0ff); + + /* Turn off PLL */ + reg = wm8510_read_reg_cache(codec, WM8510_POWER1); + wm8510_write(codec, WM8510_POWER1, reg & 0x1df); + return 0; + } + + pll_factors(freq_out*8, freq_in); + + wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); + wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18); + wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff); + wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff); + reg = wm8510_read_reg_cache(codec, WM8510_POWER1); + wm8510_write(codec, WM8510_POWER1, reg | 0x020); + + /* Run CODEC from PLL instead of MCLK */ + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); + wm8510_write(codec, WM8510_CLOCK, reg | 0x100); + + return 0; +} + +/* + * Configure WM8510 clock dividers. + */ +static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8510_OPCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf; + wm8510_write(codec, WM8510_GPIO, reg | div); + break; + case WM8510_MCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f; + wm8510_write(codec, WM8510_CLOCK, reg | div); + break; + case WM8510_ADCCLK: + reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7; + wm8510_write(codec, WM8510_ADC, reg | div); + break; + case WM8510_DACCLK: + reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7; + wm8510_write(codec, WM8510_DAC, reg | div); + break; + case WM8510_BCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3; + wm8510_write(codec, WM8510_CLOCK, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int wm8510_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + clk |= 0x0001; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0010; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0008; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x00018; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0180; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0100; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0080; + break; + default: + return -EINVAL; + } + + wm8510_write(codec, WM8510_IFACE, iface); + wm8510_write(codec, WM8510_CLOCK, clk); + return 0; +} + +static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f; + u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0020; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0040; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x0060; + break; + } + + /* filter coefficient */ + switch (params_rate(params)) { + case SNDRV_PCM_RATE_8000: + adn |= 0x5 << 1; + break; + case SNDRV_PCM_RATE_11025: + adn |= 0x4 << 1; + break; + case SNDRV_PCM_RATE_16000: + adn |= 0x3 << 1; + break; + case SNDRV_PCM_RATE_22050: + adn |= 0x2 << 1; + break; + case SNDRV_PCM_RATE_32000: + adn |= 0x1 << 1; + break; + case SNDRV_PCM_RATE_44100: + case SNDRV_PCM_RATE_48000: + break; + } + + wm8510_write(codec, WM8510_IFACE, iface); + wm8510_write(codec, WM8510_ADD, adn); + return 0; +} + +static int wm8510_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; + + if (mute) + wm8510_write(codec, WM8510_DAC, mute_reg | 0x40); + else + wm8510_write(codec, WM8510_DAC, mute_reg); + return 0; +} + +/* liam need to make this lower power with dapm */ +static int wm8510_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + + switch (level) { + case SND_SOC_BIAS_ON: + wm8510_write(codec, WM8510_POWER1, 0x1ff); + wm8510_write(codec, WM8510_POWER2, 0x1ff); + wm8510_write(codec, WM8510_POWER3, 0x1ff); + break; + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + wm8510_write(codec, WM8510_POWER1, 0x0); + wm8510_write(codec, WM8510_POWER2, 0x0); + wm8510_write(codec, WM8510_POWER3, 0x0); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8510_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +#define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_codec_dai wm8510_dai = { + .name = "WM8510 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8510_RATES, + .formats = WM8510_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8510_RATES, + .formats = WM8510_FORMATS,}, + .ops = { + .hw_params = wm8510_pcm_hw_params, + }, + .dai_ops = { + .digital_mute = wm8510_mute, + .set_fmt = wm8510_set_dai_fmt, + .set_clkdiv = wm8510_set_dai_clkdiv, + .set_pll = wm8510_set_dai_pll, + }, +}; +EXPORT_SYMBOL_GPL(wm8510_dai); + +static int wm8510_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8510_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8510_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the WM8510 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8510_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "WM8510"; + codec->owner = THIS_MODULE; + codec->read = wm8510_read_reg_cache; + codec->write = wm8510_write; + codec->set_bias_level = wm8510_set_bias_level; + codec->dai = &wm8510_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm8510_reg); + codec->reg_cache = kmemdup(wm8510_reg, sizeof(wm8510_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + wm8510_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8510_add_controls(codec); + wm8510_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to register card\n"); + goto card_err; + } + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *wm8510_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM8510 2 wire address is 0x1a + */ +#define I2C_DRIVERID_WM8510 0xfefe /* liam - need a proper id */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver wm8510_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ + +static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = wm8510_socdev; + struct wm8510_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + err("failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = wm8510_init(socdev); + if (ret < 0) { + err("failed to initialise WM8510\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int wm8510_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int wm8510_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, wm8510_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver wm8510_i2c_driver = { + .driver = { + .name = "WM8510 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_WM8510, + .attach_adapter = wm8510_i2c_attach, + .detach_client = wm8510_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "WM8510", + .driver = &wm8510_i2c_driver, +}; +#endif + +static int wm8510_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8510_setup_data *setup; + struct snd_soc_codec *codec; + int ret = 0; + + info("WM8510 Audio Codec %s", WM8510_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + wm8510_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&wm8510_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int wm8510_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8510_i2c_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8510 = { + .probe = wm8510_probe, + .remove = wm8510_remove, + .suspend = wm8510_suspend, + .resume = wm8510_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510); + +MODULE_DESCRIPTION("ASoC WM8510 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h new file mode 100644 index 000000000000..c862e7b7d530 --- /dev/null +++ b/sound/soc/codecs/wm8510.h @@ -0,0 +1,103 @@ +/* + * wm8510.h -- WM8510 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8510_H +#define _WM8510_H + +/* WM8510 register space */ + +#define WM8510_RESET 0x0 +#define WM8510_POWER1 0x1 +#define WM8510_POWER2 0x2 +#define WM8510_POWER3 0x3 +#define WM8510_IFACE 0x4 +#define WM8510_COMP 0x5 +#define WM8510_CLOCK 0x6 +#define WM8510_ADD 0x7 +#define WM8510_GPIO 0x8 +#define WM8510_DAC 0xa +#define WM8510_DACVOL 0xb +#define WM8510_ADC 0xe +#define WM8510_ADCVOL 0xf +#define WM8510_EQ1 0x12 +#define WM8510_EQ2 0x13 +#define WM8510_EQ3 0x14 +#define WM8510_EQ4 0x15 +#define WM8510_EQ5 0x16 +#define WM8510_DACLIM1 0x18 +#define WM8510_DACLIM2 0x19 +#define WM8510_NOTCH1 0x1b +#define WM8510_NOTCH2 0x1c +#define WM8510_NOTCH3 0x1d +#define WM8510_NOTCH4 0x1e +#define WM8510_ALC1 0x20 +#define WM8510_ALC2 0x21 +#define WM8510_ALC3 0x22 +#define WM8510_NGATE 0x23 +#define WM8510_PLLN 0x24 +#define WM8510_PLLK1 0x25 +#define WM8510_PLLK2 0x26 +#define WM8510_PLLK3 0x27 +#define WM8510_ATTEN 0x28 +#define WM8510_INPUT 0x2c +#define WM8510_INPPGA 0x2d +#define WM8510_ADCBOOST 0x2f +#define WM8510_OUTPUT 0x31 +#define WM8510_SPKMIX 0x32 +#define WM8510_SPKVOL 0x36 +#define WM8510_MONOMIX 0x38 + +#define WM8510_CACHEREGNUM 57 + +/* Clock divider Id's */ +#define WM8510_OPCLKDIV 0 +#define WM8510_MCLKDIV 1 +#define WM8510_ADCCLK 2 +#define WM8510_DACCLK 3 +#define WM8510_BCLKDIV 4 + +/* DAC clock dividers */ +#define WM8510_DACCLK_F2 (1 << 3) +#define WM8510_DACCLK_F4 (0 << 3) + +/* ADC clock dividers */ +#define WM8510_ADCCLK_F2 (1 << 3) +#define WM8510_ADCCLK_F4 (0 << 3) + +/* PLL Out dividers */ +#define WM8510_OPCLKDIV_1 (0 << 4) +#define WM8510_OPCLKDIV_2 (1 << 4) +#define WM8510_OPCLKDIV_3 (2 << 4) +#define WM8510_OPCLKDIV_4 (3 << 4) + +/* BCLK clock dividers */ +#define WM8510_BCLKDIV_1 (0 << 2) +#define WM8510_BCLKDIV_2 (1 << 2) +#define WM8510_BCLKDIV_4 (2 << 2) +#define WM8510_BCLKDIV_8 (3 << 2) +#define WM8510_BCLKDIV_16 (4 << 2) +#define WM8510_BCLKDIV_32 (5 << 2) + +/* MCLK clock dividers */ +#define WM8510_MCLKDIV_1 (0 << 5) +#define WM8510_MCLKDIV_1_5 (1 << 5) +#define WM8510_MCLKDIV_2 (2 << 5) +#define WM8510_MCLKDIV_3 (3 << 5) +#define WM8510_MCLKDIV_4 (4 << 5) +#define WM8510_MCLKDIV_6 (5 << 5) +#define WM8510_MCLKDIV_8 (6 << 5) +#define WM8510_MCLKDIV_12 (7 << 5) + +struct wm8510_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_codec_dai wm8510_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8510; + +#endif From f10485e79882bf09cf442a673511154987b43e3b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 5 Jun 2008 13:49:33 +0100 Subject: [PATCH 066/179] [ALSA] ASoC: Add WM8990 driver The WM8990 is a highly integrated ultra-low power hi-fi codec designed for handsets rich in multimedia features such as mobile TV, digital audio playback and gaming. The bulk of this driver was written by Liam Girdwood with some additional development and updates for new ASoC APIs by me. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8990.c | 1645 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8990.h | 832 +++++++++++++++++++ 4 files changed, 2482 insertions(+) create mode 100644 sound/soc/codecs/wm8990.c create mode 100644 sound/soc/codecs/wm8990.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ef400b363440..f439138db843 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,6 +17,9 @@ config SND_SOC_WM8750 config SND_SOC_WM8753 tristate +config SND_SOC_WM8990 + tristate + config SND_SOC_WM9712 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b92c66541663..d9826f35d8c0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,7 @@ snd-soc-wm8510-objs := wm8510.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o +snd-soc-wm8990-objs := wm8990.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o @@ -15,6 +16,7 @@ obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o +obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c new file mode 100644 index 000000000000..a7d25e2f2522 --- /dev/null +++ b/sound/soc/codecs/wm8990.c @@ -0,0 +1,1645 @@ +/* + * wm8990.c -- WM8990 ALSA Soc Audio driver + * + * Copyright 2008 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * lg@opensource.wolfsonmicro.com or linux@wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8990.h" + +#define AUDIO_NAME "wm8990" +#define WM8990_VERSION "0.2" + +/* + * Debug + */ + +#define WM8990_DEBUG 0 + +#ifdef WM8990_DEBUG +#define dbg(format, arg...) \ + printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) +#else +#define dbg(format, arg...) do {} while (0) +#endif +#define err(format, arg...) \ + printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) +#define info(format, arg...) \ + printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) +#define warn(format, arg...) \ + printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) + +/* codec private data */ +struct wm8990_priv { + unsigned int sysclk; + unsigned int pcmclk; +}; + +/* + * wm8990 register cache. Note that register 0 is not included in the + * cache. + */ +static const u16 wm8990_reg[] = { + 0x8990, /* R0 - Reset */ + 0x0000, /* R1 - Power Management (1) */ + 0x6000, /* R2 - Power Management (2) */ + 0x0000, /* R3 - Power Management (3) */ + 0x4050, /* R4 - Audio Interface (1) */ + 0x4000, /* R5 - Audio Interface (2) */ + 0x01C8, /* R6 - Clocking (1) */ + 0x0000, /* R7 - Clocking (2) */ + 0x0040, /* R8 - Audio Interface (3) */ + 0x0040, /* R9 - Audio Interface (4) */ + 0x0004, /* R10 - DAC CTRL */ + 0x00C0, /* R11 - Left DAC Digital Volume */ + 0x00C0, /* R12 - Right DAC Digital Volume */ + 0x0000, /* R13 - Digital Side Tone */ + 0x0100, /* R14 - ADC CTRL */ + 0x00C0, /* R15 - Left ADC Digital Volume */ + 0x00C0, /* R16 - Right ADC Digital Volume */ + 0x0000, /* R17 */ + 0x0000, /* R18 - GPIO CTRL 1 */ + 0x1000, /* R19 - GPIO1 & GPIO2 */ + 0x1010, /* R20 - GPIO3 & GPIO4 */ + 0x1010, /* R21 - GPIO5 & GPIO6 */ + 0x8000, /* R22 - GPIOCTRL 2 */ + 0x0800, /* R23 - GPIO_POL */ + 0x008B, /* R24 - Left Line Input 1&2 Volume */ + 0x008B, /* R25 - Left Line Input 3&4 Volume */ + 0x008B, /* R26 - Right Line Input 1&2 Volume */ + 0x008B, /* R27 - Right Line Input 3&4 Volume */ + 0x0000, /* R28 - Left Output Volume */ + 0x0000, /* R29 - Right Output Volume */ + 0x0066, /* R30 - Line Outputs Volume */ + 0x0022, /* R31 - Out3/4 Volume */ + 0x0079, /* R32 - Left OPGA Volume */ + 0x0079, /* R33 - Right OPGA Volume */ + 0x0003, /* R34 - Speaker Volume */ + 0x0003, /* R35 - ClassD1 */ + 0x0000, /* R36 */ + 0x0100, /* R37 - ClassD3 */ + 0x0000, /* R38 */ + 0x0000, /* R39 - Input Mixer1 */ + 0x0000, /* R40 - Input Mixer2 */ + 0x0000, /* R41 - Input Mixer3 */ + 0x0000, /* R42 - Input Mixer4 */ + 0x0000, /* R43 - Input Mixer5 */ + 0x0000, /* R44 - Input Mixer6 */ + 0x0000, /* R45 - Output Mixer1 */ + 0x0000, /* R46 - Output Mixer2 */ + 0x0000, /* R47 - Output Mixer3 */ + 0x0000, /* R48 - Output Mixer4 */ + 0x0000, /* R49 - Output Mixer5 */ + 0x0000, /* R50 - Output Mixer6 */ + 0x0180, /* R51 - Out3/4 Mixer */ + 0x0000, /* R52 - Line Mixer1 */ + 0x0000, /* R53 - Line Mixer2 */ + 0x0000, /* R54 - Speaker Mixer */ + 0x0000, /* R55 - Additional Control */ + 0x0000, /* R56 - AntiPOP1 */ + 0x0000, /* R57 - AntiPOP2 */ + 0x0000, /* R58 - MICBIAS */ + 0x0000, /* R59 */ + 0x0008, /* R60 - PLL1 */ + 0x0031, /* R61 - PLL2 */ + 0x0026, /* R62 - PLL3 */ +}; + +/* + * read wm8990 register cache + */ +static inline unsigned int wm8990_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1); + return cache[reg]; +} + +/* + * write wm8990 register cache + */ +static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1); + + /* Reset register is uncached */ + if (reg == 0) + return; + + cache[reg] = value; +} + +/* + * write to the wm8990 register space + */ +static int wm8990_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[3]; + + data[0] = reg & 0xFF; + data[1] = (value >> 8) & 0xFF; + data[2] = value & 0xFF; + + wm8990_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, data, 3) == 2) + return 0; + else + return -EIO; +} + +#define wm8990_reset(c) wm8990_write(c, WM8990_RESET, 0) + +static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600); + +static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000); + +static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, 0, -2100); + +static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600); + +static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0); + +static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0); + +static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763); + +static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0); + +static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int ret; + u16 val; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + + /* now hit the volume update bits (always bit 8) */ + val = wm8990_read_reg_cache(codec, reg); + return wm8990_write(codec, reg, val | 0x0100); +} + +#define SOC_WM899X_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert,\ + tlv_array) {\ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw, .put = wm899x_outpga_put_volsw_vu, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + + +static const char *wm8990_digital_sidetone[] = + {"None", "Left ADC", "Right ADC", "Reserved"}; + +static const struct soc_enum wm8990_left_digital_sidetone_enum = +SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE, + WM8990_ADC_TO_DACL_SHIFT, + WM8990_ADC_TO_DACL_MASK, + wm8990_digital_sidetone); + +static const struct soc_enum wm8990_right_digital_sidetone_enum = +SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE, + WM8990_ADC_TO_DACR_SHIFT, + WM8990_ADC_TO_DACR_MASK, + wm8990_digital_sidetone); + +static const char *wm8990_adcmode[] = + {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; + +static const struct soc_enum wm8990_right_adcmode_enum = +SOC_ENUM_SINGLE(WM8990_ADC_CTRL, + WM8990_ADC_HPF_CUT_SHIFT, + WM8990_ADC_HPF_CUT_MASK, + wm8990_adcmode); + +static const struct snd_kcontrol_new wm8990_snd_controls[] = { +/* INMIXL */ +SOC_SINGLE("LIN12 PGA Boost", WM8990_INPUT_MIXER3, WM8990_L12MNBST_BIT, 1, 0), +SOC_SINGLE("LIN34 PGA Boost", WM8990_INPUT_MIXER3, WM8990_L34MNBST_BIT, 1, 0), +/* INMIXR */ +SOC_SINGLE("RIN12 PGA Boost", WM8990_INPUT_MIXER3, WM8990_R12MNBST_BIT, 1, 0), +SOC_SINGLE("RIN34 PGA Boost", WM8990_INPUT_MIXER3, WM8990_R34MNBST_BIT, 1, 0), + +/* LOMIX */ +SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8990_OUTPUT_MIXER3, + WM8990_LLI3LOVOL_SHIFT, WM8990_LLI3LOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER3, + WM8990_LR12LOVOL_SHIFT, WM8990_LR12LOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER3, + WM8990_LL12LOVOL_SHIFT, WM8990_LL12LOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8990_OUTPUT_MIXER5, + WM8990_LRI3LOVOL_SHIFT, WM8990_LRI3LOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8990_OUTPUT_MIXER5, + WM8990_LRBLOVOL_SHIFT, WM8990_LRBLOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8990_OUTPUT_MIXER5, + WM8990_LRBLOVOL_SHIFT, WM8990_LRBLOVOL_MASK, 1, out_mix_tlv), + +/* ROMIX */ +SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8990_OUTPUT_MIXER4, + WM8990_RRI3ROVOL_SHIFT, WM8990_RRI3ROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER4, + WM8990_RL12ROVOL_SHIFT, WM8990_RL12ROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER4, + WM8990_RR12ROVOL_SHIFT, WM8990_RR12ROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8990_OUTPUT_MIXER6, + WM8990_RLI3ROVOL_SHIFT, WM8990_RLI3ROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8990_OUTPUT_MIXER6, + WM8990_RLBROVOL_SHIFT, WM8990_RLBROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8990_OUTPUT_MIXER6, + WM8990_RRBROVOL_SHIFT, WM8990_RRBROVOL_MASK, 1, out_mix_tlv), + +/* LOUT */ +SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8990_LEFT_OUTPUT_VOLUME, + WM8990_LOUTVOL_SHIFT, WM8990_LOUTVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("LOUT ZC", WM8990_LEFT_OUTPUT_VOLUME, WM8990_LOZC_BIT, 1, 0), + +/* ROUT */ +SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8990_RIGHT_OUTPUT_VOLUME, + WM8990_ROUTVOL_SHIFT, WM8990_ROUTVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("ROUT ZC", WM8990_RIGHT_OUTPUT_VOLUME, WM8990_ROZC_BIT, 1, 0), + +/* LOPGA */ +SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8990_LEFT_OPGA_VOLUME, + WM8990_LOPGAVOL_SHIFT, WM8990_LOPGAVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("LOPGA ZC Switch", WM8990_LEFT_OPGA_VOLUME, + WM8990_LOPGAZC_BIT, 1, 0), + +/* ROPGA */ +SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8990_RIGHT_OPGA_VOLUME, + WM8990_ROPGAVOL_SHIFT, WM8990_ROPGAVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("ROPGA ZC Switch", WM8990_RIGHT_OPGA_VOLUME, + WM8990_ROPGAZC_BIT, 1, 0), + +SOC_SINGLE("LON Mute Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_LONMUTE_BIT, 1, 0), +SOC_SINGLE("LOP Mute Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_LOPMUTE_BIT, 1, 0), +SOC_SINGLE("LOP Attenuation Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_LOATTN_BIT, 1, 0), +SOC_SINGLE("RON Mute Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_RONMUTE_BIT, 1, 0), +SOC_SINGLE("ROP Mute Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_ROPMUTE_BIT, 1, 0), +SOC_SINGLE("ROP Attenuation Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_ROATTN_BIT, 1, 0), + +SOC_SINGLE("OUT3 Mute Switch", WM8990_OUT3_4_VOLUME, + WM8990_OUT3MUTE_BIT, 1, 0), +SOC_SINGLE("OUT3 Attenuation Switch", WM8990_OUT3_4_VOLUME, + WM8990_OUT3ATTN_BIT, 1, 0), + +SOC_SINGLE("OUT4 Mute Switch", WM8990_OUT3_4_VOLUME, + WM8990_OUT4MUTE_BIT, 1, 0), +SOC_SINGLE("OUT4 Attenuation Switch", WM8990_OUT3_4_VOLUME, + WM8990_OUT4ATTN_BIT, 1, 0), + +SOC_SINGLE("Speaker Mode Switch", WM8990_CLASSD1, + WM8990_CDMODE_BIT, 1, 0), + +SOC_SINGLE("Speaker Output Attenuation Volume", WM8990_SPEAKER_VOLUME, + WM8990_SPKVOL_SHIFT, WM8990_SPKVOL_MASK, 0), +SOC_SINGLE("Speaker DC Boost Volume", WM8990_CLASSD3, + WM8990_DCGAIN_SHIFT, WM8990_DCGAIN_MASK, 0), +SOC_SINGLE("Speaker AC Boost Volume", WM8990_CLASSD3, + WM8990_ACGAIN_SHIFT, WM8990_ACGAIN_MASK, 0), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume", + WM8990_LEFT_DAC_DIGITAL_VOLUME, + WM8990_DACL_VOL_SHIFT, + WM8990_DACL_VOL_MASK, + 0, + out_dac_tlv), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume", + WM8990_RIGHT_DAC_DIGITAL_VOLUME, + WM8990_DACR_VOL_SHIFT, + WM8990_DACR_VOL_MASK, + 0, + out_dac_tlv), + +SOC_ENUM("Left Digital Sidetone", wm8990_left_digital_sidetone_enum), +SOC_ENUM("Right Digital Sidetone", wm8990_right_digital_sidetone_enum), + +SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8990_DIGITAL_SIDE_TONE, + WM8990_ADCL_DAC_SVOL_SHIFT, WM8990_ADCL_DAC_SVOL_MASK, 0, + out_sidetone_tlv), +SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8990_DIGITAL_SIDE_TONE, + WM8990_ADCR_DAC_SVOL_SHIFT, WM8990_ADCR_DAC_SVOL_MASK, 0, + out_sidetone_tlv), + +SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8990_ADC_CTRL, + WM8990_ADC_HPF_ENA_BIT, 1, 0), + +SOC_ENUM("ADC HPF Mode", wm8990_right_adcmode_enum), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume", + WM8990_LEFT_ADC_DIGITAL_VOLUME, + WM8990_ADCL_VOL_SHIFT, + WM8990_ADCL_VOL_MASK, + 0, + in_adc_tlv), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume", + WM8990_RIGHT_ADC_DIGITAL_VOLUME, + WM8990_ADCR_VOL_SHIFT, + WM8990_ADCR_VOL_MASK, + 0, + in_adc_tlv), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN12 Volume", + WM8990_LEFT_LINE_INPUT_1_2_VOLUME, + WM8990_LIN12VOL_SHIFT, + WM8990_LIN12VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("LIN12 ZC Switch", WM8990_LEFT_LINE_INPUT_1_2_VOLUME, + WM8990_LI12ZC_BIT, 1, 0), + +SOC_SINGLE("LIN12 Mute Switch", WM8990_LEFT_LINE_INPUT_1_2_VOLUME, + WM8990_LI12MUTE_BIT, 1, 0), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN34 Volume", + WM8990_LEFT_LINE_INPUT_3_4_VOLUME, + WM8990_LIN34VOL_SHIFT, + WM8990_LIN34VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("LIN34 ZC Switch", WM8990_LEFT_LINE_INPUT_3_4_VOLUME, + WM8990_LI34ZC_BIT, 1, 0), + +SOC_SINGLE("LIN34 Mute Switch", WM8990_LEFT_LINE_INPUT_3_4_VOLUME, + WM8990_LI34MUTE_BIT, 1, 0), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN12 Volume", + WM8990_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8990_RIN12VOL_SHIFT, + WM8990_RIN12VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("RIN12 ZC Switch", WM8990_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8990_RI12ZC_BIT, 1, 0), + +SOC_SINGLE("RIN12 Mute Switch", WM8990_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8990_RI12MUTE_BIT, 1, 0), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN34 Volume", + WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8990_RIN34VOL_SHIFT, + WM8990_RIN34VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("RIN34 ZC Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8990_RI34ZC_BIT, 1, 0), + +SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8990_RI34MUTE_BIT, 1, 0), + +}; + +/* add non dapm controls */ +static int wm8990_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8990_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8990_snd_controls[i], codec, + NULL)); + if (err < 0) + return err; + } + return 0; +} + +/* + * _DAPM_ Controls + */ + +static int inmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u16 reg, fakepower; + + reg = wm8990_read_reg_cache(w->codec, WM8990_POWER_MANAGEMENT_2); + fakepower = wm8990_read_reg_cache(w->codec, WM8990_INTDRIVBITS); + + if (fakepower & ((1 << WM8990_INMIXL_PWR_BIT) | + (1 << WM8990_AINLMUX_PWR_BIT))) { + reg |= WM8990_AINL_ENA; + } else { + reg &= ~WM8990_AINL_ENA; + } + + if (fakepower & ((1 << WM8990_INMIXR_PWR_BIT) | + (1 << WM8990_AINRMUX_PWR_BIT))) { + reg |= WM8990_AINR_ENA; + } else { + reg &= ~WM8990_AINL_ENA; + } + wm8990_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg); + + return 0; +} + +static int outmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u32 reg_shift = kcontrol->private_value & 0xfff; + int ret = 0; + u16 reg; + + switch (reg_shift) { + case WM8990_SPEAKER_MIXER | (WM8990_LDSPK_BIT << 8) : + reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER1); + if (reg & WM8990_LDLO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 1 LDLO Set\n"); + ret = -1; + } + break; + case WM8990_SPEAKER_MIXER | (WM8990_RDSPK_BIT << 8): + reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER2); + if (reg & WM8990_RDRO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 2 RDRO Set\n"); + ret = -1; + } + break; + case WM8990_OUTPUT_MIXER1 | (WM8990_LDLO_BIT << 8): + reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER); + if (reg & WM8990_LDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer LDSPK Set\n"); + ret = -1; + } + break; + case WM8990_OUTPUT_MIXER2 | (WM8990_RDRO_BIT << 8): + reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER); + if (reg & WM8990_RDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer RDSPK Set\n"); + ret = -1; + } + break; + } + + return ret; +} + +/* INMIX dB values */ +static const unsigned int in_mix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(-1200, 600), +}; + +/* Left In PGA Connections */ +static const struct snd_kcontrol_new wm8990_dapm_lin12_pga_controls[] = { +SOC_DAPM_SINGLE("LIN1 Switch", WM8990_INPUT_MIXER2, WM8990_LMN1_BIT, 1, 0), +SOC_DAPM_SINGLE("LIN2 Switch", WM8990_INPUT_MIXER2, WM8990_LMP2_BIT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8990_dapm_lin34_pga_controls[] = { +SOC_DAPM_SINGLE("LIN3 Switch", WM8990_INPUT_MIXER2, WM8990_LMN3_BIT, 1, 0), +SOC_DAPM_SINGLE("LIN4 Switch", WM8990_INPUT_MIXER2, WM8990_LMP4_BIT, 1, 0), +}; + +/* Right In PGA Connections */ +static const struct snd_kcontrol_new wm8990_dapm_rin12_pga_controls[] = { +SOC_DAPM_SINGLE("RIN1 Switch", WM8990_INPUT_MIXER2, WM8990_RMN1_BIT, 1, 0), +SOC_DAPM_SINGLE("RIN2 Switch", WM8990_INPUT_MIXER2, WM8990_RMP2_BIT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8990_dapm_rin34_pga_controls[] = { +SOC_DAPM_SINGLE("RIN3 Switch", WM8990_INPUT_MIXER2, WM8990_RMN3_BIT, 1, 0), +SOC_DAPM_SINGLE("RIN4 Switch", WM8990_INPUT_MIXER2, WM8990_RMP4_BIT, 1, 0), +}; + +/* INMIXL */ +static const struct snd_kcontrol_new wm8990_dapm_inmixl_controls[] = { +SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8990_INPUT_MIXER3, + WM8990_LDBVOL_SHIFT, WM8990_LDBVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8990_INPUT_MIXER5, WM8990_LI2BVOL_SHIFT, + 7, 0, in_mix_tlv), +SOC_DAPM_SINGLE("LINPGA12 Switch", WM8990_INPUT_MIXER3, WM8990_L12MNB_BIT, + 1, 0), +SOC_DAPM_SINGLE("LINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT, + 1, 0), +}; + +/* INMIXR */ +static const struct snd_kcontrol_new wm8990_dapm_inmixr_controls[] = { +SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8990_INPUT_MIXER4, + WM8990_RDBVOL_SHIFT, WM8990_RDBVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8990_INPUT_MIXER6, WM8990_RI2BVOL_SHIFT, + 7, 0, in_mix_tlv), +SOC_DAPM_SINGLE("RINPGA12 Switch", WM8990_INPUT_MIXER3, WM8990_L12MNB_BIT, + 1, 0), +SOC_DAPM_SINGLE("RINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT, + 1, 0), +}; + +/* AINLMUX */ +static const char *wm8990_ainlmux[] = + {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; + +static const struct soc_enum wm8990_ainlmux_enum = +SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINLMODE_SHIFT, + ARRAY_SIZE(wm8990_ainlmux), wm8990_ainlmux); + +static const struct snd_kcontrol_new wm8990_dapm_ainlmux_controls = +SOC_DAPM_ENUM("Route", wm8990_ainlmux_enum); + +/* DIFFINL */ + +/* AINRMUX */ +static const char *wm8990_ainrmux[] = + {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; + +static const struct soc_enum wm8990_ainrmux_enum = +SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINRMODE_SHIFT, + ARRAY_SIZE(wm8990_ainrmux), wm8990_ainrmux); + +static const struct snd_kcontrol_new wm8990_dapm_ainrmux_controls = +SOC_DAPM_ENUM("Route", wm8990_ainrmux_enum); + +/* RXVOICE */ +static const struct snd_kcontrol_new wm8990_dapm_rxvoice_controls[] = { +SOC_DAPM_SINGLE_TLV("LIN4/RXN", WM8990_INPUT_MIXER5, WM8990_LR4BVOL_SHIFT, + WM8990_LR4BVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("RIN4/RXP", WM8990_INPUT_MIXER6, WM8990_RL4BVOL_SHIFT, + WM8990_RL4BVOL_MASK, 0, in_mix_tlv), +}; + +/* LOMIX */ +static const struct snd_kcontrol_new wm8990_dapm_lomix_controls[] = { +SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LRBLO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LLBLO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LRI3LO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LLI3LO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LR12LO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LL12LO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8990_OUTPUT_MIXER1, + WM8990_LDLO_BIT, 1, 0), +}; + +/* ROMIX */ +static const struct snd_kcontrol_new wm8990_dapm_romix_controls[] = { +SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RLBRO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RRBRO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RLI3RO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RRI3RO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RL12RO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RR12RO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8990_OUTPUT_MIXER2, + WM8990_RDRO_BIT, 1, 0), +}; + +/* LONMIX */ +static const struct snd_kcontrol_new wm8990_dapm_lonmix_controls[] = { +SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8990_LINE_MIXER1, + WM8990_LLOPGALON_BIT, 1, 0), +SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8990_LINE_MIXER1, + WM8990_LROPGALON_BIT, 1, 0), +SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8990_LINE_MIXER1, + WM8990_LOPLON_BIT, 1, 0), +}; + +/* LOPMIX */ +static const struct snd_kcontrol_new wm8990_dapm_lopmix_controls[] = { +SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8990_LINE_MIXER1, + WM8990_LR12LOP_BIT, 1, 0), +SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8990_LINE_MIXER1, + WM8990_LL12LOP_BIT, 1, 0), +SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8990_LINE_MIXER1, + WM8990_LLOPGALOP_BIT, 1, 0), +}; + +/* RONMIX */ +static const struct snd_kcontrol_new wm8990_dapm_ronmix_controls[] = { +SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8990_LINE_MIXER2, + WM8990_RROPGARON_BIT, 1, 0), +SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8990_LINE_MIXER2, + WM8990_RLOPGARON_BIT, 1, 0), +SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8990_LINE_MIXER2, + WM8990_ROPRON_BIT, 1, 0), +}; + +/* ROPMIX */ +static const struct snd_kcontrol_new wm8990_dapm_ropmix_controls[] = { +SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8990_LINE_MIXER2, + WM8990_RL12ROP_BIT, 1, 0), +SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8990_LINE_MIXER2, + WM8990_RR12ROP_BIT, 1, 0), +SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8990_LINE_MIXER2, + WM8990_RROPGAROP_BIT, 1, 0), +}; + +/* OUT3MIX */ +static const struct snd_kcontrol_new wm8990_dapm_out3mix_controls[] = { +SOC_DAPM_SINGLE("OUT3MIX LIN4/RXP Bypass Switch", WM8990_OUT3_4_MIXER, + WM8990_LI4O3_BIT, 1, 0), +SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8990_OUT3_4_MIXER, + WM8990_LPGAO3_BIT, 1, 0), +}; + +/* OUT4MIX */ +static const struct snd_kcontrol_new wm8990_dapm_out4mix_controls[] = { +SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8990_OUT3_4_MIXER, + WM8990_RPGAO4_BIT, 1, 0), +SOC_DAPM_SINGLE("OUT4MIX RIN4/RXP Bypass Switch", WM8990_OUT3_4_MIXER, + WM8990_RI4O4_BIT, 1, 0), +}; + +/* SPKMIX */ +static const struct snd_kcontrol_new wm8990_dapm_spkmix_controls[] = { +SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8990_SPEAKER_MIXER, + WM8990_LI2SPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8990_SPEAKER_MIXER, + WM8990_LB2SPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8990_SPEAKER_MIXER, + WM8990_LOPGASPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8990_SPEAKER_MIXER, + WM8990_LDSPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8990_SPEAKER_MIXER, + WM8990_RDSPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8990_SPEAKER_MIXER, + WM8990_ROPGASPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8990_SPEAKER_MIXER, + WM8990_RL12ROP_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8990_SPEAKER_MIXER, + WM8990_RI2SPK_BIT, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8990_dapm_widgets[] = { +/* Input Side */ +/* Input Lines */ +SND_SOC_DAPM_INPUT("LIN1"), +SND_SOC_DAPM_INPUT("LIN2"), +SND_SOC_DAPM_INPUT("LIN3"), +SND_SOC_DAPM_INPUT("LIN4/RXN"), +SND_SOC_DAPM_INPUT("RIN3"), +SND_SOC_DAPM_INPUT("RIN4/RXP"), +SND_SOC_DAPM_INPUT("RIN1"), +SND_SOC_DAPM_INPUT("RIN2"), +SND_SOC_DAPM_INPUT("Internal ADC Source"), + +/* DACs */ +SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8990_POWER_MANAGEMENT_2, + WM8990_ADCL_ENA_BIT, 0), +SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8990_POWER_MANAGEMENT_2, + WM8990_ADCR_ENA_BIT, 0), + +/* Input PGAs */ +SND_SOC_DAPM_MIXER("LIN12 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_LIN12_ENA_BIT, + 0, &wm8990_dapm_lin12_pga_controls[0], + ARRAY_SIZE(wm8990_dapm_lin12_pga_controls)), +SND_SOC_DAPM_MIXER("LIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_LIN34_ENA_BIT, + 0, &wm8990_dapm_lin34_pga_controls[0], + ARRAY_SIZE(wm8990_dapm_lin34_pga_controls)), +SND_SOC_DAPM_MIXER("RIN12 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN12_ENA_BIT, + 0, &wm8990_dapm_rin12_pga_controls[0], + ARRAY_SIZE(wm8990_dapm_rin12_pga_controls)), +SND_SOC_DAPM_MIXER("RIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN34_ENA_BIT, + 0, &wm8990_dapm_rin34_pga_controls[0], + ARRAY_SIZE(wm8990_dapm_rin34_pga_controls)), + +/* INMIXL */ +SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0, + &wm8990_dapm_inmixl_controls[0], + ARRAY_SIZE(wm8990_dapm_inmixl_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* AINLMUX */ +SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0, + &wm8990_dapm_ainlmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* INMIXR */ +SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0, + &wm8990_dapm_inmixr_controls[0], + ARRAY_SIZE(wm8990_dapm_inmixr_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* AINRMUX */ +SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0, + &wm8990_dapm_ainrmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* Output Side */ +/* DACs */ +SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8990_POWER_MANAGEMENT_3, + WM8990_DACL_ENA_BIT, 0), +SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8990_POWER_MANAGEMENT_3, + WM8990_DACR_ENA_BIT, 0), + +/* LOMIX */ +SND_SOC_DAPM_MIXER_E("LOMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LOMIX_ENA_BIT, + 0, &wm8990_dapm_lomix_controls[0], + ARRAY_SIZE(wm8990_dapm_lomix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + +/* LONMIX */ +SND_SOC_DAPM_MIXER("LONMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LON_ENA_BIT, 0, + &wm8990_dapm_lonmix_controls[0], + ARRAY_SIZE(wm8990_dapm_lonmix_controls)), + +/* LOPMIX */ +SND_SOC_DAPM_MIXER("LOPMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LOP_ENA_BIT, 0, + &wm8990_dapm_lopmix_controls[0], + ARRAY_SIZE(wm8990_dapm_lopmix_controls)), + +/* OUT3MIX */ +SND_SOC_DAPM_MIXER("OUT3MIX", WM8990_POWER_MANAGEMENT_1, WM8990_OUT3_ENA_BIT, 0, + &wm8990_dapm_out3mix_controls[0], + ARRAY_SIZE(wm8990_dapm_out3mix_controls)), + +/* SPKMIX */ +SND_SOC_DAPM_MIXER_E("SPKMIX", WM8990_POWER_MANAGEMENT_1, WM8990_SPK_ENA_BIT, 0, + &wm8990_dapm_spkmix_controls[0], + ARRAY_SIZE(wm8990_dapm_spkmix_controls), outmixer_event, + SND_SOC_DAPM_PRE_REG), + +/* OUT4MIX */ +SND_SOC_DAPM_MIXER("OUT4MIX", WM8990_POWER_MANAGEMENT_1, WM8990_OUT4_ENA_BIT, 0, + &wm8990_dapm_out4mix_controls[0], + ARRAY_SIZE(wm8990_dapm_out4mix_controls)), + +/* ROPMIX */ +SND_SOC_DAPM_MIXER("ROPMIX", WM8990_POWER_MANAGEMENT_3, WM8990_ROP_ENA_BIT, 0, + &wm8990_dapm_ropmix_controls[0], + ARRAY_SIZE(wm8990_dapm_ropmix_controls)), + +/* RONMIX */ +SND_SOC_DAPM_MIXER("RONMIX", WM8990_POWER_MANAGEMENT_3, WM8990_RON_ENA_BIT, 0, + &wm8990_dapm_ronmix_controls[0], + ARRAY_SIZE(wm8990_dapm_ronmix_controls)), + +/* ROMIX */ +SND_SOC_DAPM_MIXER_E("ROMIX", WM8990_POWER_MANAGEMENT_3, WM8990_ROMIX_ENA_BIT, + 0, &wm8990_dapm_romix_controls[0], + ARRAY_SIZE(wm8990_dapm_romix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + +/* LOUT PGA */ +SND_SOC_DAPM_PGA("LOUT PGA", WM8990_POWER_MANAGEMENT_1, WM8990_LOUT_ENA_BIT, 0, + NULL, 0), + +/* ROUT PGA */ +SND_SOC_DAPM_PGA("ROUT PGA", WM8990_POWER_MANAGEMENT_1, WM8990_ROUT_ENA_BIT, 0, + NULL, 0), + +/* LOPGA */ +SND_SOC_DAPM_PGA("LOPGA", WM8990_POWER_MANAGEMENT_3, WM8990_LOPGA_ENA_BIT, 0, + NULL, 0), + +/* ROPGA */ +SND_SOC_DAPM_PGA("ROPGA", WM8990_POWER_MANAGEMENT_3, WM8990_ROPGA_ENA_BIT, 0, + NULL, 0), + +/* MICBIAS */ +SND_SOC_DAPM_MICBIAS("MICBIAS", WM8990_POWER_MANAGEMENT_1, + WM8990_MICBIAS_ENA_BIT, 0), + +SND_SOC_DAPM_OUTPUT("LON"), +SND_SOC_DAPM_OUTPUT("LOP"), +SND_SOC_DAPM_OUTPUT("OUT3"), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("SPKN"), +SND_SOC_DAPM_OUTPUT("SPKP"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_OUTPUT("OUT4"), +SND_SOC_DAPM_OUTPUT("ROP"), +SND_SOC_DAPM_OUTPUT("RON"), + +SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Make DACs turn on when playing even if not mixed into any outputs */ + {"Internal DAC Sink", NULL, "Left DAC"}, + {"Internal DAC Sink", NULL, "Right DAC"}, + + /* Make ADCs turn on when recording even if not mixed from any inputs */ + {"Left ADC", NULL, "Internal ADC Source"}, + {"Right ADC", NULL, "Internal ADC Source"}, + + /* Input Side */ + /* LIN12 PGA */ + {"LIN12 PGA", "LIN1 Switch", "LIN1"}, + {"LIN12 PGA", "LIN2 Switch", "LIN2"}, + /* LIN34 PGA */ + {"LIN34 PGA", "LIN3 Switch", "LIN3"}, + {"LIN34 PGA", "LIN4 Switch", "LIN4"}, + /* INMIXL */ + {"INMIXL", "Record Left Volume", "LOMIX"}, + {"INMIXL", "LIN2 Volume", "LIN2"}, + {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, + {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, + /* AILNMUX */ + {"AILNMUX", "INMIXL Mix", "INMIXL"}, + {"AILNMUX", "DIFFINL Mix", "LIN12PGA"}, + {"AILNMUX", "DIFFINL Mix", "LIN34PGA"}, + {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* ADC */ + {"Left ADC", NULL, "AILNMUX"}, + + /* RIN12 PGA */ + {"RIN12 PGA", "RIN1 Switch", "RIN1"}, + {"RIN12 PGA", "RIN2 Switch", "RIN2"}, + /* RIN34 PGA */ + {"RIN34 PGA", "RIN3 Switch", "RIN3"}, + {"RIN34 PGA", "RIN4 Switch", "RIN4"}, + /* INMIXL */ + {"INMIXR", "Record Right Volume", "ROMIX"}, + {"INMIXR", "RIN2 Volume", "RIN2"}, + {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, + {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, + /* AIRNMUX */ + {"AIRNMUX", "INMIXR Mix", "INMIXR"}, + {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"}, + {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"}, + {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"}, + {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* ADC */ + {"Right ADC", NULL, "AIRNMUX"}, + + /* LOMIX */ + {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"}, + {"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"}, + {"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"LOMIX", "LOMIX Right ADC Bypass Switch", "AINRMUX"}, + {"LOMIX", "LOMIX Left ADC Bypass Switch", "AINLMUX"}, + {"LOMIX", "LOMIX Left DAC Switch", "Left DAC"}, + + /* ROMIX */ + {"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"}, + {"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"}, + {"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"ROMIX", "ROMIX Right ADC Bypass Switch", "AINRMUX"}, + {"ROMIX", "ROMIX Left ADC Bypass Switch", "AINLMUX"}, + {"ROMIX", "ROMIX Right DAC Switch", "Right DAC"}, + + /* SPKMIX */ + {"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"}, + {"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"}, + {"SPKMIX", "SPKMIX LADC Bypass Switch", "AINLMUX"}, + {"SPKMIX", "SPKMIX RADC Bypass Switch", "AINRMUX"}, + {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"}, + {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"}, + {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"}, + {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"}, + + /* LONMIX */ + {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"}, + {"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"}, + {"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"}, + + /* LOPMIX */ + {"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"}, + + /* OUT3MIX */ + {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"}, + {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"}, + + /* OUT4MIX */ + {"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"}, + {"OUT4MIX", "OUT4MIX RIN4/RXP Bypass Switch", "RIN4/RXP"}, + + /* RONMIX */ + {"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"}, + {"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"}, + {"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"}, + + /* ROPMIX */ + {"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"}, + + /* Out Mixer PGAs */ + {"LOPGA", NULL, "LOMIX"}, + {"ROPGA", NULL, "ROMIX"}, + + {"LOUT PGA", NULL, "LOMIX"}, + {"ROUT PGA", NULL, "ROMIX"}, + + /* Output Pins */ + {"LON", NULL, "LONMIX"}, + {"LOP", NULL, "LOPMIX"}, + {"OUT", NULL, "OUT3MIX"}, + {"LOUT", NULL, "LOUT PGA"}, + {"SPKN", NULL, "SPKMIX"}, + {"ROUT", NULL, "ROUT PGA"}, + {"OUT4", NULL, "OUT4MIX"}, + {"ROP", NULL, "ROPMIX"}, + {"RON", NULL, "RONMIX"}, +}; + +static int wm8990_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8990_dapm_widgets, + ARRAY_SIZE(wm8990_dapm_widgets)); + + /* set up the WM8990 audio map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +/* PLL divisors */ +struct _pll_div { + u32 div2; + u32 n; + u32 k; +}; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 16) * 10) + +static void pll_factors(struct _pll_div *pll_div, unsigned int target, + unsigned int source) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod; + + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div->div2 = 1; + Ndiv = target / source; + } else + pll_div->div2 = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8990 N value outwith recommended range! N = %d\n", Ndiv); + + pll_div->n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div->k = K; +} + +static int wm8990_set_dai_pll(struct snd_soc_codec_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + u16 reg; + struct snd_soc_codec *codec = codec_dai->codec; + struct _pll_div pll_div; + + if (freq_in && freq_out) { + pll_factors(&pll_div, freq_out * 4, freq_in); + + /* Turn on PLL */ + reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2); + reg |= WM8990_PLL_ENA; + wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + + /* sysclk comes from PLL */ + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2); + wm8990_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC); + + /* set up N , fractional mode and pre-divisor if neccessary */ + wm8990_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM | + (pll_div.div2?WM8990_PRESCALE:0)); + wm8990_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8)); + wm8990_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF)); + } else { + /* Turn on PLL */ + reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2); + reg &= ~WM8990_PLL_ENA; + wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + } + return 0; +} + +/* + * Clock after PLL and dividers + */ +static int wm8990_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8990_priv *wm8990 = codec->private_data; + + wm8990->sysclk = freq; + return 0; +} + +/* + * Set's ADC and Voice DAC format. + */ +static int wm8990_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 audio1, audio3; + + audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1); + audio3 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_3); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + audio3 &= ~WM8990_AIF_MSTR1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + audio3 |= WM8990_AIF_MSTR1; + break; + default: + return -EINVAL; + } + + audio1 &= ~WM8990_AIF_FMT_MASK; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + audio1 |= WM8990_AIF_TMF_I2S; + audio1 &= ~WM8990_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_RIGHT_J: + audio1 |= WM8990_AIF_TMF_RIGHTJ; + audio1 &= ~WM8990_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_LEFT_J: + audio1 |= WM8990_AIF_TMF_LEFTJ; + audio1 &= ~WM8990_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_A: + audio1 |= WM8990_AIF_TMF_DSP; + audio1 &= ~WM8990_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_B: + audio1 |= WM8990_AIF_TMF_DSP | WM8990_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + + wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1); + wm8990_write(codec, WM8990_AUDIO_INTERFACE_3, audio3); + return 0; +} + +static int wm8990_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8990_MCLK_DIV: + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) & + ~WM8990_MCLK_DIV_MASK; + wm8990_write(codec, WM8990_CLOCKING_2, reg | div); + break; + case WM8990_DACCLK_DIV: + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) & + ~WM8990_DAC_CLKDIV_MASK; + wm8990_write(codec, WM8990_CLOCKING_2, reg | div); + break; + case WM8990_ADCCLK_DIV: + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) & + ~WM8990_ADC_CLKDIV_MASK; + wm8990_write(codec, WM8990_CLOCKING_2, reg | div); + break; + case WM8990_BCLK_DIV: + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_1) & + ~WM8990_BCLK_DIV_MASK; + wm8990_write(codec, WM8990_CLOCKING_1, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + */ +static int wm8990_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1); + + audio1 &= ~WM8990_AIF_WL_MASK; + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + audio1 |= WM8990_AIF_WL_20BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + audio1 |= WM8990_AIF_WL_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + audio1 |= WM8990_AIF_WL_32BITS; + break; + } + + wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1); + return 0; +} + +static int wm8990_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 val; + + val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL) & ~WM8990_DAC_MUTE; + + if (mute) + wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); + else + wm8990_write(codec, WM8990_DAC_CTRL, val); + + return 0; +} + +static int wm8990_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 val; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Enable all output discharge bits */ + wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | + WM8990_DIS_RLINE | WM8990_DIS_OUT3 | + WM8990_DIS_OUT4 | WM8990_DIS_LOUT | + WM8990_DIS_ROUT); + + /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_BUFDCOPEN | WM8990_POBCTRL | + WM8990_VMIDTOG); + + /* Delay to allow output caps to discharge */ + msleep(msecs_to_jiffies(300)); + + /* Disable VMIDTOG */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_BUFDCOPEN | WM8990_POBCTRL); + + /* disable all output discharge bits */ + wm8990_write(codec, WM8990_ANTIPOP1, 0); + + /* Enable outputs */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00); + + msleep(msecs_to_jiffies(50)); + + /* Enable VMID at 2x50k */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02); + + msleep(msecs_to_jiffies(100)); + + /* Enable VREF */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); + + msleep(msecs_to_jiffies(600)); + + /* Enable BUFIOEN */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_BUFDCOPEN | WM8990_POBCTRL | + WM8990_BUFIOEN); + + /* Disable outputs */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x3); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN); + } else { + /* ON -> standby */ + + } + break; + + case SND_SOC_BIAS_OFF: + /* Enable POBCTRL and SOFT_ST */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_POBCTRL | WM8990_BUFIOEN); + + /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_BUFDCOPEN | WM8990_POBCTRL | + WM8990_BUFIOEN); + + /* mute DAC */ + val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL); + wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); + + /* Enable any disabled outputs */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); + + /* Disable VMID */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01); + + msleep(msecs_to_jiffies(300)); + + /* Enable all output discharge bits */ + wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | + WM8990_DIS_RLINE | WM8990_DIS_OUT3 | + WM8990_DIS_OUT4 | WM8990_DIS_LOUT | + WM8990_DIS_ROUT); + + /* Disable VREF */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x0); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8990_write(codec, WM8990_ANTIPOP2, 0x0); + break; + } + + codec->bias_level = level; + return 0; +} + +#define WM8990_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define WM8990_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* + * The WM8990 supports 2 different and mutually exclusive DAI + * configurations. + * + * 1. ADC/DAC on Primary Interface + * 2. ADC on Primary Interface/DAC on secondary + */ +struct snd_soc_codec_dai wm8990_dai = { +/* ADC/DAC on primary */ + .name = "WM8990 ADC/DAC Primary", + .id = 1, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8990_RATES, + .formats = WM8990_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8990_RATES, + .formats = WM8990_FORMATS,}, + .ops = { + .hw_params = wm8990_hw_params,}, + .dai_ops = { + .digital_mute = wm8990_mute, + .set_fmt = wm8990_set_dai_fmt, + .set_clkdiv = wm8990_set_dai_clkdiv, + .set_pll = wm8990_set_dai_pll, + .set_sysclk = wm8990_set_dai_sysclk, + }, +}; +EXPORT_SYMBOL_GPL(wm8990_dai); + +static int wm8990_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + /* we only need to suspend if we are a valid card */ + if (!codec->card) + return 0; + + wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8990_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* we only need to resume if we are a valid card */ + if (!codec->card) + return 0; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) { + if (i + 1 == WM8990_RESET) + continue; + data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + + wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +/* + * initialise the WM8990 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8990_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + u16 reg; + int ret = 0; + + codec->name = "WM8990"; + codec->owner = THIS_MODULE; + codec->read = wm8990_read_reg_cache; + codec->write = wm8990_write; + codec->set_bias_level = wm8990_set_bias_level; + codec->dai = &wm8990_dai; + codec->num_dai = 2; + codec->reg_cache_size = ARRAY_SIZE(wm8990_reg); + codec->reg_cache = kmemdup(wm8990_reg, sizeof(wm8990_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + wm8990_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8990: failed to create pcms\n"); + goto pcm_err; + } + + /* charge output caps */ + codec->bias_level = SND_SOC_BIAS_OFF; + wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + reg = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_4); + wm8990_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1); + + reg = wm8990_read_reg_cache(codec, WM8990_GPIO1_GPIO2) & + ~WM8990_GPIO1_SEL_MASK; + wm8990_write(codec, WM8990_GPIO1_GPIO2, reg | 1); + + reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2); + wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA); + + wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); + wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); + + wm8990_add_controls(codec); + wm8990_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8990: failed to register card\n"); + goto card_err; + } + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ +static struct snd_soc_device *wm8990_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM891 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x34 + * high = 0x36 + */ +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver wm8990_i2c_driver; +static struct i2c_client client_template; + +static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = wm8990_socdev; + struct wm8990_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + err("failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = wm8990_init(socdev); + if (ret < 0) { + err("failed to initialise WM8990\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int wm8990_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int wm8990_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, wm8990_codec_probe); +} + +static struct i2c_driver wm8990_i2c_driver = { + .driver = { + .name = "WM8990 I2C Codec", + .owner = THIS_MODULE, + }, + .attach_adapter = wm8990_i2c_attach, + .detach_client = wm8990_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "WM8990", + .driver = &wm8990_i2c_driver, +}; +#endif + +static int wm8990_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8990_setup_data *setup; + struct snd_soc_codec *codec; + struct wm8990_priv *wm8990; + int ret = 0; + + info("WM8990 Audio Codec %s\n", WM8990_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + wm8990 = kzalloc(sizeof(struct wm8990_priv), GFP_KERNEL); + if (wm8990 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = wm8990; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + wm8990_socdev = socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&wm8990_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int wm8990_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8990_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8990 = { + .probe = wm8990_probe, + .remove = wm8990_remove, + .suspend = wm8990_suspend, + .resume = wm8990_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990); + +MODULE_DESCRIPTION("ASoC WM8990 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h new file mode 100644 index 000000000000..bf9f8823dfc9 --- /dev/null +++ b/sound/soc/codecs/wm8990.h @@ -0,0 +1,832 @@ +/* + * wm8990.h -- audio driver for WM8990 + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef __WM8990REGISTERDEFS_H__ +#define __WM8990REGISTERDEFS_H__ + +/* + * Register values. + */ +#define WM8990_RESET 0x00 +#define WM8990_POWER_MANAGEMENT_1 0x01 +#define WM8990_POWER_MANAGEMENT_2 0x02 +#define WM8990_POWER_MANAGEMENT_3 0x03 +#define WM8990_AUDIO_INTERFACE_1 0x04 +#define WM8990_AUDIO_INTERFACE_2 0x05 +#define WM8990_CLOCKING_1 0x06 +#define WM8990_CLOCKING_2 0x07 +#define WM8990_AUDIO_INTERFACE_3 0x08 +#define WM8990_AUDIO_INTERFACE_4 0x09 +#define WM8990_DAC_CTRL 0x0A +#define WM8990_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define WM8990_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define WM8990_DIGITAL_SIDE_TONE 0x0D +#define WM8990_ADC_CTRL 0x0E +#define WM8990_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define WM8990_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define WM8990_GPIO_CTRL_1 0x12 +#define WM8990_GPIO1_GPIO2 0x13 +#define WM8990_GPIO3_GPIO4 0x14 +#define WM8990_GPIO5_GPIO6 0x15 +#define WM8990_GPIOCTRL_2 0x16 +#define WM8990_GPIO_POL 0x17 +#define WM8990_LEFT_LINE_INPUT_1_2_VOLUME 0x18 +#define WM8990_LEFT_LINE_INPUT_3_4_VOLUME 0x19 +#define WM8990_RIGHT_LINE_INPUT_1_2_VOLUME 0x1A +#define WM8990_RIGHT_LINE_INPUT_3_4_VOLUME 0x1B +#define WM8990_LEFT_OUTPUT_VOLUME 0x1C +#define WM8990_RIGHT_OUTPUT_VOLUME 0x1D +#define WM8990_LINE_OUTPUTS_VOLUME 0x1E +#define WM8990_OUT3_4_VOLUME 0x1F +#define WM8990_LEFT_OPGA_VOLUME 0x20 +#define WM8990_RIGHT_OPGA_VOLUME 0x21 +#define WM8990_SPEAKER_VOLUME 0x22 +#define WM8990_CLASSD1 0x23 +#define WM8990_CLASSD3 0x25 +#define WM8990_INPUT_MIXER1 0x27 +#define WM8990_INPUT_MIXER2 0x28 +#define WM8990_INPUT_MIXER3 0x29 +#define WM8990_INPUT_MIXER4 0x2A +#define WM8990_INPUT_MIXER5 0x2B +#define WM8990_INPUT_MIXER6 0x2C +#define WM8990_OUTPUT_MIXER1 0x2D +#define WM8990_OUTPUT_MIXER2 0x2E +#define WM8990_OUTPUT_MIXER3 0x2F +#define WM8990_OUTPUT_MIXER4 0x30 +#define WM8990_OUTPUT_MIXER5 0x31 +#define WM8990_OUTPUT_MIXER6 0x32 +#define WM8990_OUT3_4_MIXER 0x33 +#define WM8990_LINE_MIXER1 0x34 +#define WM8990_LINE_MIXER2 0x35 +#define WM8990_SPEAKER_MIXER 0x36 +#define WM8990_ADDITIONAL_CONTROL 0x37 +#define WM8990_ANTIPOP1 0x38 +#define WM8990_ANTIPOP2 0x39 +#define WM8990_MICBIAS 0x3A +#define WM8990_PLL1 0x3C +#define WM8990_PLL2 0x3D +#define WM8990_PLL3 0x3E +#define WM8990_INTDRIVBITS 0x3F + +#define WM8990_REGISTER_COUNT 60 +#define WM8990_MAX_REGISTER 0x3F + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - Reset + */ +#define WM8990_SW_RESET_CHIP_ID_MASK 0xFFFF /* SW_RESET_CHIP_ID */ + +/* + * R1 (0x01) - Power Management (1) + */ +#define WM8990_SPK_ENA 0x1000 /* SPK_ENA */ +#define WM8990_SPK_ENA_BIT 12 +#define WM8990_OUT3_ENA 0x0800 /* OUT3_ENA */ +#define WM8990_OUT3_ENA_BIT 11 +#define WM8990_OUT4_ENA 0x0400 /* OUT4_ENA */ +#define WM8990_OUT4_ENA_BIT 10 +#define WM8990_LOUT_ENA 0x0200 /* LOUT_ENA */ +#define WM8990_LOUT_ENA_BIT 9 +#define WM8990_ROUT_ENA 0x0100 /* ROUT_ENA */ +#define WM8990_ROUT_ENA_BIT 8 +#define WM8990_MICBIAS_ENA 0x0010 /* MICBIAS_ENA */ +#define WM8990_MICBIAS_ENA_BIT 4 +#define WM8990_VMID_MODE_MASK 0x0006 /* VMID_MODE - [2:1] */ +#define WM8990_VREF_ENA 0x0001 /* VREF_ENA */ +#define WM8990_VREF_ENA_BIT 0 + +/* + * R2 (0x02) - Power Management (2) + */ +#define WM8990_PLL_ENA 0x8000 /* PLL_ENA */ +#define WM8990_PLL_ENA_BIT 15 +#define WM8990_TSHUT_ENA 0x4000 /* TSHUT_ENA */ +#define WM8990_TSHUT_ENA_BIT 14 +#define WM8990_TSHUT_OPDIS 0x2000 /* TSHUT_OPDIS */ +#define WM8990_TSHUT_OPDIS_BIT 13 +#define WM8990_OPCLK_ENA 0x0800 /* OPCLK_ENA */ +#define WM8990_OPCLK_ENA_BIT 11 +#define WM8990_AINL_ENA 0x0200 /* AINL_ENA */ +#define WM8990_AINL_ENA_BIT 9 +#define WM8990_AINR_ENA 0x0100 /* AINR_ENA */ +#define WM8990_AINR_ENA_BIT 8 +#define WM8990_LIN34_ENA 0x0080 /* LIN34_ENA */ +#define WM8990_LIN34_ENA_BIT 7 +#define WM8990_LIN12_ENA 0x0040 /* LIN12_ENA */ +#define WM8990_LIN12_ENA_BIT 6 +#define WM8990_RIN34_ENA 0x0020 /* RIN34_ENA */ +#define WM8990_RIN34_ENA_BIT 5 +#define WM8990_RIN12_ENA 0x0010 /* RIN12_ENA */ +#define WM8990_RIN12_ENA_BIT 4 +#define WM8990_ADCL_ENA 0x0002 /* ADCL_ENA */ +#define WM8990_ADCL_ENA_BIT 1 +#define WM8990_ADCR_ENA 0x0001 /* ADCR_ENA */ +#define WM8990_ADCR_ENA_BIT 0 + +/* + * R3 (0x03) - Power Management (3) + */ +#define WM8990_LON_ENA 0x2000 /* LON_ENA */ +#define WM8990_LON_ENA_BIT 13 +#define WM8990_LOP_ENA 0x1000 /* LOP_ENA */ +#define WM8990_LOP_ENA_BIT 12 +#define WM8990_RON_ENA 0x0800 /* RON_ENA */ +#define WM8990_RON_ENA_BIT 11 +#define WM8990_ROP_ENA 0x0400 /* ROP_ENA */ +#define WM8990_ROP_ENA_BIT 10 +#define WM8990_LOPGA_ENA 0x0080 /* LOPGA_ENA */ +#define WM8990_LOPGA_ENA_BIT 7 +#define WM8990_ROPGA_ENA 0x0040 /* ROPGA_ENA */ +#define WM8990_ROPGA_ENA_BIT 6 +#define WM8990_LOMIX_ENA 0x0020 /* LOMIX_ENA */ +#define WM8990_LOMIX_ENA_BIT 5 +#define WM8990_ROMIX_ENA 0x0010 /* ROMIX_ENA */ +#define WM8990_ROMIX_ENA_BIT 4 +#define WM8990_DACL_ENA 0x0002 /* DACL_ENA */ +#define WM8990_DACL_ENA_BIT 1 +#define WM8990_DACR_ENA 0x0001 /* DACR_ENA */ +#define WM8990_DACR_ENA_BIT 0 + +/* + * R4 (0x04) - Audio Interface (1) + */ +#define WM8990_AIFADCL_SRC 0x8000 /* AIFADCL_SRC */ +#define WM8990_AIFADCR_SRC 0x4000 /* AIFADCR_SRC */ +#define WM8990_AIFADC_TDM 0x2000 /* AIFADC_TDM */ +#define WM8990_AIFADC_TDM_CHAN 0x1000 /* AIFADC_TDM_CHAN */ +#define WM8990_AIF_BCLK_INV 0x0100 /* AIF_BCLK_INV */ +#define WM8990_AIF_LRCLK_INV 0x0080 /* AIF_LRCLK_INV */ +#define WM8990_AIF_WL_MASK 0x0060 /* AIF_WL - [6:5] */ +#define WM8990_AIF_WL_16BITS (0 << 5) +#define WM8990_AIF_WL_20BITS (1 << 5) +#define WM8990_AIF_WL_24BITS (2 << 5) +#define WM8990_AIF_WL_32BITS (3 << 5) +#define WM8990_AIF_FMT_MASK 0x0018 /* AIF_FMT - [4:3] */ +#define WM8990_AIF_TMF_RIGHTJ (0 << 3) +#define WM8990_AIF_TMF_LEFTJ (1 << 3) +#define WM8990_AIF_TMF_I2S (2 << 3) +#define WM8990_AIF_TMF_DSP (3 << 3) + +/* + * R5 (0x05) - Audio Interface (2) + */ +#define WM8990_DACL_SRC 0x8000 /* DACL_SRC */ +#define WM8990_DACR_SRC 0x4000 /* DACR_SRC */ +#define WM8990_AIFDAC_TDM 0x2000 /* AIFDAC_TDM */ +#define WM8990_AIFDAC_TDM_CHAN 0x1000 /* AIFDAC_TDM_CHAN */ +#define WM8990_DAC_BOOST_MASK 0x0C00 /* DAC_BOOST */ +#define WM8990_DAC_COMP 0x0010 /* DAC_COMP */ +#define WM8990_DAC_COMPMODE 0x0008 /* DAC_COMPMODE */ +#define WM8990_ADC_COMP 0x0004 /* ADC_COMP */ +#define WM8990_ADC_COMPMODE 0x0002 /* ADC_COMPMODE */ +#define WM8990_LOOPBACK 0x0001 /* LOOPBACK */ + +/* + * R6 (0x06) - Clocking (1) + */ +#define WM8990_TOCLK_RATE 0x8000 /* TOCLK_RATE */ +#define WM8990_TOCLK_ENA 0x4000 /* TOCLK_ENA */ +#define WM8990_OPCLKDIV_MASK 0x1E00 /* OPCLKDIV - [12:9] */ +#define WM8990_DCLKDIV_MASK 0x01C0 /* DCLKDIV - [8:6] */ +#define WM8990_BCLK_DIV_MASK 0x001E /* BCLK_DIV - [4:1] */ +#define WM8990_BCLK_DIV_1 (0x0 << 1) +#define WM8990_BCLK_DIV_1_5 (0x1 << 1) +#define WM8990_BCLK_DIV_2 (0x2 << 1) +#define WM8990_BCLK_DIV_3 (0x3 << 1) +#define WM8990_BCLK_DIV_4 (0x4 << 1) +#define WM8990_BCLK_DIV_5_5 (0x5 << 1) +#define WM8990_BCLK_DIV_6 (0x6 << 1) +#define WM8990_BCLK_DIV_8 (0x7 << 1) +#define WM8990_BCLK_DIV_11 (0x8 << 1) +#define WM8990_BCLK_DIV_12 (0x9 << 1) +#define WM8990_BCLK_DIV_16 (0xA << 1) +#define WM8990_BCLK_DIV_22 (0xB << 1) +#define WM8990_BCLK_DIV_24 (0xC << 1) +#define WM8990_BCLK_DIV_32 (0xD << 1) +#define WM8990_BCLK_DIV_44 (0xE << 1) +#define WM8990_BCLK_DIV_48 (0xF << 1) + +/* + * R7 (0x07) - Clocking (2) + */ +#define WM8990_MCLK_SRC 0x8000 /* MCLK_SRC */ +#define WM8990_SYSCLK_SRC 0x4000 /* SYSCLK_SRC */ +#define WM8990_CLK_FORCE 0x2000 /* CLK_FORCE */ +#define WM8990_MCLK_DIV_MASK 0x1800 /* MCLK_DIV - [12:11] */ +#define WM8990_MCLK_DIV_1 (0 << 11) +#define WM8990_MCLK_DIV_2 (2 << 11) +#define WM8990_MCLK_INV 0x0400 /* MCLK_INV */ +#define WM8990_ADC_CLKDIV_MASK 0x00E0 /* ADC_CLKDIV */ +#define WM8990_ADC_CLKDIV_1 (0 << 5) +#define WM8990_ADC_CLKDIV_1_5 (1 << 5) +#define WM8990_ADC_CLKDIV_2 (2 << 5) +#define WM8990_ADC_CLKDIV_3 (3 << 5) +#define WM8990_ADC_CLKDIV_4 (4 << 5) +#define WM8990_ADC_CLKDIV_5_5 (5 << 5) +#define WM8990_ADC_CLKDIV_6 (6 << 5) +#define WM8990_DAC_CLKDIV_MASK 0x001C /* DAC_CLKDIV - [4:2] */ +#define WM8990_DAC_CLKDIV_1 (0 << 2) +#define WM8990_DAC_CLKDIV_1_5 (1 << 2) +#define WM8990_DAC_CLKDIV_2 (2 << 2) +#define WM8990_DAC_CLKDIV_3 (3 << 2) +#define WM8990_DAC_CLKDIV_4 (4 << 2) +#define WM8990_DAC_CLKDIV_5_5 (5 << 2) +#define WM8990_DAC_CLKDIV_6 (6 << 2) + +/* + * R8 (0x08) - Audio Interface (3) + */ +#define WM8990_AIF_MSTR1 0x8000 /* AIF_MSTR1 */ +#define WM8990_AIF_MSTR2 0x4000 /* AIF_MSTR2 */ +#define WM8990_AIF_SEL 0x2000 /* AIF_SEL */ +#define WM8990_ADCLRC_DIR 0x0800 /* ADCLRC_DIR */ +#define WM8990_ADCLRC_RATE_MASK 0x07FF /* ADCLRC_RATE */ + +/* + * R9 (0x09) - Audio Interface (4) + */ +#define WM8990_ALRCGPIO1 0x8000 /* ALRCGPIO1 */ +#define WM8990_ALRCBGPIO6 0x4000 /* ALRCBGPIO6 */ +#define WM8990_AIF_TRIS 0x2000 /* AIF_TRIS */ +#define WM8990_DACLRC_DIR 0x0800 /* DACLRC_DIR */ +#define WM8990_DACLRC_RATE_MASK 0x07FF /* DACLRC_RATE */ + +/* + * R10 (0x0A) - DAC CTRL + */ +#define WM8990_AIF_LRCLKRATE 0x0400 /* AIF_LRCLKRATE */ +#define WM8990_DAC_MONO 0x0200 /* DAC_MONO */ +#define WM8990_DAC_SB_FILT 0x0100 /* DAC_SB_FILT */ +#define WM8990_DAC_MUTERATE 0x0080 /* DAC_MUTERATE */ +#define WM8990_DAC_MUTEMODE 0x0040 /* DAC_MUTEMODE */ +#define WM8990_DEEMP_MASK 0x0030 /* DEEMP - [5:4] */ +#define WM8990_DAC_MUTE 0x0004 /* DAC_MUTE */ +#define WM8990_DACL_DATINV 0x0002 /* DACL_DATINV */ +#define WM8990_DACR_DATINV 0x0001 /* DACR_DATINV */ + +/* + * R11 (0x0B) - Left DAC Digital Volume + */ +#define WM8990_DAC_VU 0x0100 /* DAC_VU */ +#define WM8990_DACL_VOL_MASK 0x00FF /* DACL_VOL - [7:0] */ +#define WM8990_DACL_VOL_SHIFT 0 +/* + * R12 (0x0C) - Right DAC Digital Volume + */ +#define WM8990_DAC_VU 0x0100 /* DAC_VU */ +#define WM8990_DACR_VOL_MASK 0x00FF /* DACR_VOL - [7:0] */ +#define WM8990_DACR_VOL_SHIFT 0 +/* + * R13 (0x0D) - Digital Side Tone + */ +#define WM8990_ADCL_DAC_SVOL_MASK 0x0F /* ADCL_DAC_SVOL */ +#define WM8990_ADCL_DAC_SVOL_SHIFT 9 +#define WM8990_ADCR_DAC_SVOL_MASK 0x0F /* ADCR_DAC_SVOL */ +#define WM8990_ADCR_DAC_SVOL_SHIFT 5 +#define WM8990_ADC_TO_DACL_MASK 0x03 /* ADC_TO_DACL - [3:2] */ +#define WM8990_ADC_TO_DACL_SHIFT 2 +#define WM8990_ADC_TO_DACR_MASK 0x03 /* ADC_TO_DACR - [1:0] */ +#define WM8990_ADC_TO_DACR_SHIFT 0 + +/* + * R14 (0x0E) - ADC CTRL + */ +#define WM8990_ADC_HPF_ENA 0x0100 /* ADC_HPF_ENA */ +#define WM8990_ADC_HPF_ENA_BIT 8 +#define WM8990_ADC_HPF_CUT_MASK 0x03 /* ADC_HPF_CUT - [6:5] */ +#define WM8990_ADC_HPF_CUT_SHIFT 5 +#define WM8990_ADCL_DATINV 0x0002 /* ADCL_DATINV */ +#define WM8990_ADCL_DATINV_BIT 1 +#define WM8990_ADCR_DATINV 0x0001 /* ADCR_DATINV */ +#define WM8990_ADCR_DATINV_BIT 0 + +/* + * R15 (0x0F) - Left ADC Digital Volume + */ +#define WM8990_ADC_VU 0x0100 /* ADC_VU */ +#define WM8990_ADCL_VOL_MASK 0x00FF /* ADCL_VOL - [7:0] */ +#define WM8990_ADCL_VOL_SHIFT 0 + +/* + * R16 (0x10) - Right ADC Digital Volume + */ +#define WM8990_ADC_VU 0x0100 /* ADC_VU */ +#define WM8990_ADCR_VOL_MASK 0x00FF /* ADCR_VOL - [7:0] */ +#define WM8990_ADCR_VOL_SHIFT 0 + +/* + * R18 (0x12) - GPIO CTRL 1 + */ +#define WM8990_IRQ 0x1000 /* IRQ */ +#define WM8990_TEMPOK 0x0800 /* TEMPOK */ +#define WM8990_MICSHRT 0x0400 /* MICSHRT */ +#define WM8990_MICDET 0x0200 /* MICDET */ +#define WM8990_PLL_LCK 0x0100 /* PLL_LCK */ +#define WM8990_GPI8_STATUS 0x0080 /* GPI8_STATUS */ +#define WM8990_GPI7_STATUS 0x0040 /* GPI7_STATUS */ +#define WM8990_GPIO6_STATUS 0x0020 /* GPIO6_STATUS */ +#define WM8990_GPIO5_STATUS 0x0010 /* GPIO5_STATUS */ +#define WM8990_GPIO4_STATUS 0x0008 /* GPIO4_STATUS */ +#define WM8990_GPIO3_STATUS 0x0004 /* GPIO3_STATUS */ +#define WM8990_GPIO2_STATUS 0x0002 /* GPIO2_STATUS */ +#define WM8990_GPIO1_STATUS 0x0001 /* GPIO1_STATUS */ + +/* + * R19 (0x13) - GPIO1 & GPIO2 + */ +#define WM8990_GPIO2_DEB_ENA 0x8000 /* GPIO2_DEB_ENA */ +#define WM8990_GPIO2_IRQ_ENA 0x4000 /* GPIO2_IRQ_ENA */ +#define WM8990_GPIO2_PU 0x2000 /* GPIO2_PU */ +#define WM8990_GPIO2_PD 0x1000 /* GPIO2_PD */ +#define WM8990_GPIO2_SEL_MASK 0x0F00 /* GPIO2_SEL - [11:8] */ +#define WM8990_GPIO1_DEB_ENA 0x0080 /* GPIO1_DEB_ENA */ +#define WM8990_GPIO1_IRQ_ENA 0x0040 /* GPIO1_IRQ_ENA */ +#define WM8990_GPIO1_PU 0x0020 /* GPIO1_PU */ +#define WM8990_GPIO1_PD 0x0010 /* GPIO1_PD */ +#define WM8990_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */ + +/* + * R20 (0x14) - GPIO3 & GPIO4 + */ +#define WM8990_GPIO4_DEB_ENA 0x8000 /* GPIO4_DEB_ENA */ +#define WM8990_GPIO4_IRQ_ENA 0x4000 /* GPIO4_IRQ_ENA */ +#define WM8990_GPIO4_PU 0x2000 /* GPIO4_PU */ +#define WM8990_GPIO4_PD 0x1000 /* GPIO4_PD */ +#define WM8990_GPIO4_SEL_MASK 0x0F00 /* GPIO4_SEL - [11:8] */ +#define WM8990_GPIO3_DEB_ENA 0x0080 /* GPIO3_DEB_ENA */ +#define WM8990_GPIO3_IRQ_ENA 0x0040 /* GPIO3_IRQ_ENA */ +#define WM8990_GPIO3_PU 0x0020 /* GPIO3_PU */ +#define WM8990_GPIO3_PD 0x0010 /* GPIO3_PD */ +#define WM8990_GPIO3_SEL_MASK 0x000F /* GPIO3_SEL - [3:0] */ + +/* + * R21 (0x15) - GPIO5 & GPIO6 + */ +#define WM8990_GPIO6_DEB_ENA 0x8000 /* GPIO6_DEB_ENA */ +#define WM8990_GPIO6_IRQ_ENA 0x4000 /* GPIO6_IRQ_ENA */ +#define WM8990_GPIO6_PU 0x2000 /* GPIO6_PU */ +#define WM8990_GPIO6_PD 0x1000 /* GPIO6_PD */ +#define WM8990_GPIO6_SEL_MASK 0x0F00 /* GPIO6_SEL - [11:8] */ +#define WM8990_GPIO5_DEB_ENA 0x0080 /* GPIO5_DEB_ENA */ +#define WM8990_GPIO5_IRQ_ENA 0x0040 /* GPIO5_IRQ_ENA */ +#define WM8990_GPIO5_PU 0x0020 /* GPIO5_PU */ +#define WM8990_GPIO5_PD 0x0010 /* GPIO5_PD */ +#define WM8990_GPIO5_SEL_MASK 0x000F /* GPIO5_SEL - [3:0] */ + +/* + * R22 (0x16) - GPIOCTRL 2 + */ +#define WM8990_RD_3W_ENA 0x8000 /* RD_3W_ENA */ +#define WM8990_MODE_3W4W 0x4000 /* MODE_3W4W */ +#define WM8990_TEMPOK_IRQ_ENA 0x0800 /* TEMPOK_IRQ_ENA */ +#define WM8990_MICSHRT_IRQ_ENA 0x0400 /* MICSHRT_IRQ_ENA */ +#define WM8990_MICDET_IRQ_ENA 0x0200 /* MICDET_IRQ_ENA */ +#define WM8990_PLL_LCK_IRQ_ENA 0x0100 /* PLL_LCK_IRQ_ENA */ +#define WM8990_GPI8_DEB_ENA 0x0080 /* GPI8_DEB_ENA */ +#define WM8990_GPI8_IRQ_ENA 0x0040 /* GPI8_IRQ_ENA */ +#define WM8990_GPI8_ENA 0x0010 /* GPI8_ENA */ +#define WM8990_GPI7_DEB_ENA 0x0008 /* GPI7_DEB_ENA */ +#define WM8990_GPI7_IRQ_ENA 0x0004 /* GPI7_IRQ_ENA */ +#define WM8990_GPI7_ENA 0x0001 /* GPI7_ENA */ + +/* + * R23 (0x17) - GPIO_POL + */ +#define WM8990_IRQ_INV 0x1000 /* IRQ_INV */ +#define WM8990_TEMPOK_POL 0x0800 /* TEMPOK_POL */ +#define WM8990_MICSHRT_POL 0x0400 /* MICSHRT_POL */ +#define WM8990_MICDET_POL 0x0200 /* MICDET_POL */ +#define WM8990_PLL_LCK_POL 0x0100 /* PLL_LCK_POL */ +#define WM8990_GPI8_POL 0x0080 /* GPI8_POL */ +#define WM8990_GPI7_POL 0x0040 /* GPI7_POL */ +#define WM8990_GPIO6_POL 0x0020 /* GPIO6_POL */ +#define WM8990_GPIO5_POL 0x0010 /* GPIO5_POL */ +#define WM8990_GPIO4_POL 0x0008 /* GPIO4_POL */ +#define WM8990_GPIO3_POL 0x0004 /* GPIO3_POL */ +#define WM8990_GPIO2_POL 0x0002 /* GPIO2_POL */ +#define WM8990_GPIO1_POL 0x0001 /* GPIO1_POL */ + +/* + * R24 (0x18) - Left Line Input 1&2 Volume + */ +#define WM8990_IPVU 0x0100 /* IPVU */ +#define WM8990_LI12MUTE 0x0080 /* LI12MUTE */ +#define WM8990_LI12MUTE_BIT 7 +#define WM8990_LI12ZC 0x0040 /* LI12ZC */ +#define WM8990_LI12ZC_BIT 6 +#define WM8990_LIN12VOL_MASK 0x001F /* LIN12VOL - [4:0] */ +#define WM8990_LIN12VOL_SHIFT 0 +/* + * R25 (0x19) - Left Line Input 3&4 Volume + */ +#define WM8990_IPVU 0x0100 /* IPVU */ +#define WM8990_LI34MUTE 0x0080 /* LI34MUTE */ +#define WM8990_LI34MUTE_BIT 7 +#define WM8990_LI34ZC 0x0040 /* LI34ZC */ +#define WM8990_LI34ZC_BIT 6 +#define WM8990_LIN34VOL_MASK 0x001F /* LIN34VOL - [4:0] */ +#define WM8990_LIN34VOL_SHIFT 0 + +/* + * R26 (0x1A) - Right Line Input 1&2 Volume + */ +#define WM8990_IPVU 0x0100 /* IPVU */ +#define WM8990_RI12MUTE 0x0080 /* RI12MUTE */ +#define WM8990_RI12MUTE_BIT 7 +#define WM8990_RI12ZC 0x0040 /* RI12ZC */ +#define WM8990_RI12ZC_BIT 6 +#define WM8990_RIN12VOL_MASK 0x001F /* RIN12VOL - [4:0] */ +#define WM8990_RIN12VOL_SHIFT 0 + +/* + * R27 (0x1B) - Right Line Input 3&4 Volume + */ +#define WM8990_IPVU 0x0100 /* IPVU */ +#define WM8990_RI34MUTE 0x0080 /* RI34MUTE */ +#define WM8990_RI34MUTE_BIT 7 +#define WM8990_RI34ZC 0x0040 /* RI34ZC */ +#define WM8990_RI34ZC_BIT 6 +#define WM8990_RIN34VOL_MASK 0x001F /* RIN34VOL - [4:0] */ +#define WM8990_RIN34VOL_SHIFT 0 + +/* + * R28 (0x1C) - Left Output Volume + */ +#define WM8990_OPVU 0x0100 /* OPVU */ +#define WM8990_LOZC 0x0080 /* LOZC */ +#define WM8990_LOZC_BIT 7 +#define WM8990_LOUTVOL_MASK 0x007F /* LOUTVOL - [6:0] */ +#define WM8990_LOUTVOL_SHIFT 0 +/* + * R29 (0x1D) - Right Output Volume + */ +#define WM8990_OPVU 0x0100 /* OPVU */ +#define WM8990_ROZC 0x0080 /* ROZC */ +#define WM8990_ROZC_BIT 7 +#define WM8990_ROUTVOL_MASK 0x007F /* ROUTVOL - [6:0] */ +#define WM8990_ROUTVOL_SHIFT 0 +/* + * R30 (0x1E) - Line Outputs Volume + */ +#define WM8990_LONMUTE 0x0040 /* LONMUTE */ +#define WM8990_LONMUTE_BIT 6 +#define WM8990_LOPMUTE 0x0020 /* LOPMUTE */ +#define WM8990_LOPMUTE_BIT 5 +#define WM8990_LOATTN 0x0010 /* LOATTN */ +#define WM8990_LOATTN_BIT 4 +#define WM8990_RONMUTE 0x0004 /* RONMUTE */ +#define WM8990_RONMUTE_BIT 2 +#define WM8990_ROPMUTE 0x0002 /* ROPMUTE */ +#define WM8990_ROPMUTE_BIT 1 +#define WM8990_ROATTN 0x0001 /* ROATTN */ +#define WM8990_ROATTN_BIT 0 + +/* + * R31 (0x1F) - Out3/4 Volume + */ +#define WM8990_OUT3MUTE 0x0020 /* OUT3MUTE */ +#define WM8990_OUT3MUTE_BIT 5 +#define WM8990_OUT3ATTN 0x0010 /* OUT3ATTN */ +#define WM8990_OUT3ATTN_BIT 4 +#define WM8990_OUT4MUTE 0x0002 /* OUT4MUTE */ +#define WM8990_OUT4MUTE_BIT 1 +#define WM8990_OUT4ATTN 0x0001 /* OUT4ATTN */ +#define WM8990_OUT4ATTN_BIT 0 + +/* + * R32 (0x20) - Left OPGA Volume + */ +#define WM8990_OPVU 0x0100 /* OPVU */ +#define WM8990_LOPGAZC 0x0080 /* LOPGAZC */ +#define WM8990_LOPGAZC_BIT 7 +#define WM8990_LOPGAVOL_MASK 0x007F /* LOPGAVOL - [6:0] */ +#define WM8990_LOPGAVOL_SHIFT 0 + +/* + * R33 (0x21) - Right OPGA Volume + */ +#define WM8990_OPVU 0x0100 /* OPVU */ +#define WM8990_ROPGAZC 0x0080 /* ROPGAZC */ +#define WM8990_ROPGAZC_BIT 7 +#define WM8990_ROPGAVOL_MASK 0x007F /* ROPGAVOL - [6:0] */ +#define WM8990_ROPGAVOL_SHIFT 0 +/* + * R34 (0x22) - Speaker Volume + */ +#define WM8990_SPKVOL_MASK 0x0003 /* SPKVOL - [1:0] */ +#define WM8990_SPKVOL_SHIFT 0 + +/* + * R35 (0x23) - ClassD1 + */ +#define WM8990_CDMODE 0x0100 /* CDMODE */ +#define WM8990_CDMODE_BIT 8 + +/* + * R37 (0x25) - ClassD3 + */ +#define WM8990_DCGAIN_MASK 0x0007 /* DCGAIN - [5:3] */ +#define WM8990_DCGAIN_SHIFT 3 +#define WM8990_ACGAIN_MASK 0x0007 /* ACGAIN - [2:0] */ +#define WM8990_ACGAIN_SHIFT 0 +/* + * R39 (0x27) - Input Mixer1 + */ +#define WM8990_AINLMODE_MASK 0x000C /* AINLMODE - [3:2] */ +#define WM8990_AINLMODE_SHIFT 2 +#define WM8990_AINRMODE_MASK 0x0003 /* AINRMODE - [1:0] */ +#define WM8990_AINRMODE_SHIFT 0 + +/* + * R40 (0x28) - Input Mixer2 + */ +#define WM8990_LMP4 0x0080 /* LMP4 */ +#define WM8990_LMP4_BIT 7 /* LMP4 */ +#define WM8990_LMN3 0x0040 /* LMN3 */ +#define WM8990_LMN3_BIT 6 /* LMN3 */ +#define WM8990_LMP2 0x0020 /* LMP2 */ +#define WM8990_LMP2_BIT 5 /* LMP2 */ +#define WM8990_LMN1 0x0010 /* LMN1 */ +#define WM8990_LMN1_BIT 4 /* LMN1 */ +#define WM8990_RMP4 0x0008 /* RMP4 */ +#define WM8990_RMP4_BIT 3 /* RMP4 */ +#define WM8990_RMN3 0x0004 /* RMN3 */ +#define WM8990_RMN3_BIT 2 /* RMN3 */ +#define WM8990_RMP2 0x0002 /* RMP2 */ +#define WM8990_RMP2_BIT 1 /* RMP2 */ +#define WM8990_RMN1 0x0001 /* RMN1 */ +#define WM8990_RMN1_BIT 0 /* RMN1 */ + +/* + * R41 (0x29) - Input Mixer3 + */ +#define WM8990_L34MNB 0x0100 /* L34MNB */ +#define WM8990_L34MNB_BIT 8 +#define WM8990_L34MNBST 0x0080 /* L34MNBST */ +#define WM8990_L34MNBST_BIT 7 +#define WM8990_L12MNB 0x0020 /* L12MNB */ +#define WM8990_L12MNB_BIT 5 +#define WM8990_L12MNBST 0x0010 /* L12MNBST */ +#define WM8990_L12MNBST_BIT 4 +#define WM8990_LDBVOL_MASK 0x0007 /* LDBVOL - [2:0] */ +#define WM8990_LDBVOL_SHIFT 0 + +/* + * R42 (0x2A) - Input Mixer4 + */ +#define WM8990_R34MNB 0x0100 /* R34MNB */ +#define WM8990_R34MNB_BIT 8 +#define WM8990_R34MNBST 0x0080 /* R34MNBST */ +#define WM8990_R34MNBST_BIT 7 +#define WM8990_R12MNB 0x0020 /* R12MNB */ +#define WM8990_R12MNB_BIT 5 +#define WM8990_R12MNBST 0x0010 /* R12MNBST */ +#define WM8990_R12MNBST_BIT 4 +#define WM8990_RDBVOL_MASK 0x0007 /* RDBVOL - [2:0] */ +#define WM8990_RDBVOL_SHIFT 0 + +/* + * R43 (0x2B) - Input Mixer5 + */ +#define WM8990_LI2BVOL_MASK 0x07 /* LI2BVOL - [8:6] */ +#define WM8990_LI2BVOL_SHIFT 6 +#define WM8990_LR4BVOL_MASK 0x07 /* LR4BVOL - [5:3] */ +#define WM8990_LR4BVOL_SHIFT 3 +#define WM8990_LL4BVOL_MASK 0x07 /* LL4BVOL - [2:0] */ +#define WM8990_LL4BVOL_SHIFT 0 + +/* + * R44 (0x2C) - Input Mixer6 + */ +#define WM8990_RI2BVOL_MASK 0x07 /* RI2BVOL - [8:6] */ +#define WM8990_RI2BVOL_SHIFT 6 +#define WM8990_RL4BVOL_MASK 0x07 /* RL4BVOL - [5:3] */ +#define WM8990_RL4BVOL_SHIFT 3 +#define WM8990_RR4BVOL_MASK 0x07 /* RR4BVOL - [2:0] */ +#define WM8990_RR4BVOL_SHIFT 0 + +/* + * R45 (0x2D) - Output Mixer1 + */ +#define WM8990_LRBLO 0x0080 /* LRBLO */ +#define WM8990_LRBLO_BIT 7 +#define WM8990_LLBLO 0x0040 /* LLBLO */ +#define WM8990_LLBLO_BIT 6 +#define WM8990_LRI3LO 0x0020 /* LRI3LO */ +#define WM8990_LRI3LO_BIT 5 +#define WM8990_LLI3LO 0x0010 /* LLI3LO */ +#define WM8990_LLI3LO_BIT 4 +#define WM8990_LR12LO 0x0008 /* LR12LO */ +#define WM8990_LR12LO_BIT 3 +#define WM8990_LL12LO 0x0004 /* LL12LO */ +#define WM8990_LL12LO_BIT 2 +#define WM8990_LDLO 0x0001 /* LDLO */ +#define WM8990_LDLO_BIT 0 + +/* + * R46 (0x2E) - Output Mixer2 + */ +#define WM8990_RLBRO 0x0080 /* RLBRO */ +#define WM8990_RLBRO_BIT 7 +#define WM8990_RRBRO 0x0040 /* RRBRO */ +#define WM8990_RRBRO_BIT 6 +#define WM8990_RLI3RO 0x0020 /* RLI3RO */ +#define WM8990_RLI3RO_BIT 5 +#define WM8990_RRI3RO 0x0010 /* RRI3RO */ +#define WM8990_RRI3RO_BIT 4 +#define WM8990_RL12RO 0x0008 /* RL12RO */ +#define WM8990_RL12RO_BIT 3 +#define WM8990_RR12RO 0x0004 /* RR12RO */ +#define WM8990_RR12RO_BIT 2 +#define WM8990_RDRO 0x0001 /* RDRO */ +#define WM8990_RDRO_BIT 0 + +/* + * R47 (0x2F) - Output Mixer3 + */ +#define WM8990_LLI3LOVOL_MASK 0x07 /* LLI3LOVOL - [8:6] */ +#define WM8990_LLI3LOVOL_SHIFT 6 +#define WM8990_LR12LOVOL_MASK 0x07 /* LR12LOVOL - [5:3] */ +#define WM8990_LR12LOVOL_SHIFT 3 +#define WM8990_LL12LOVOL_MASK 0x07 /* LL12LOVOL - [2:0] */ +#define WM8990_LL12LOVOL_SHIFT 0 + +/* + * R48 (0x30) - Output Mixer4 + */ +#define WM8990_RRI3ROVOL_MASK 0x07 /* RRI3ROVOL - [8:6] */ +#define WM8990_RRI3ROVOL_SHIFT 6 +#define WM8990_RL12ROVOL_MASK 0x07 /* RL12ROVOL - [5:3] */ +#define WM8990_RL12ROVOL_SHIFT 3 +#define WM8990_RR12ROVOL_MASK 0x07 /* RR12ROVOL - [2:0] */ +#define WM8990_RR12ROVOL_SHIFT 0 + +/* + * R49 (0x31) - Output Mixer5 + */ +#define WM8990_LRI3LOVOL_MASK 0x07 /* LRI3LOVOL - [8:6] */ +#define WM8990_LRI3LOVOL_SHIFT 6 +#define WM8990_LRBLOVOL_MASK 0x07 /* LRBLOVOL - [5:3] */ +#define WM8990_LRBLOVOL_SHIFT 3 +#define WM8990_LLBLOVOL_MASK 0x07 /* LLBLOVOL - [2:0] */ +#define WM8990_LLBLOVOL_SHIFT 0 + +/* + * R50 (0x32) - Output Mixer6 + */ +#define WM8990_RLI3ROVOL_MASK 0x07 /* RLI3ROVOL - [8:6] */ +#define WM8990_RLI3ROVOL_SHIFT 6 +#define WM8990_RLBROVOL_MASK 0x07 /* RLBROVOL - [5:3] */ +#define WM8990_RLBROVOL_SHIFT 3 +#define WM8990_RRBROVOL_MASK 0x07 /* RRBROVOL - [2:0] */ +#define WM8990_RRBROVOL_SHIFT 0 + +/* + * R51 (0x33) - Out3/4 Mixer + */ +#define WM8990_VSEL_MASK 0x0180 /* VSEL - [8:7] */ +#define WM8990_LI4O3 0x0020 /* LI4O3 */ +#define WM8990_LI4O3_BIT 5 +#define WM8990_LPGAO3 0x0010 /* LPGAO3 */ +#define WM8990_LPGAO3_BIT 4 +#define WM8990_RI4O4 0x0002 /* RI4O4 */ +#define WM8990_RI4O4_BIT 1 +#define WM8990_RPGAO4 0x0001 /* RPGAO4 */ +#define WM8990_RPGAO4_BIT 0 +/* + * R52 (0x34) - Line Mixer1 + */ +#define WM8990_LLOPGALON 0x0040 /* LLOPGALON */ +#define WM8990_LLOPGALON_BIT 6 +#define WM8990_LROPGALON 0x0020 /* LROPGALON */ +#define WM8990_LROPGALON_BIT 5 +#define WM8990_LOPLON 0x0010 /* LOPLON */ +#define WM8990_LOPLON_BIT 4 +#define WM8990_LR12LOP 0x0004 /* LR12LOP */ +#define WM8990_LR12LOP_BIT 2 +#define WM8990_LL12LOP 0x0002 /* LL12LOP */ +#define WM8990_LL12LOP_BIT 1 +#define WM8990_LLOPGALOP 0x0001 /* LLOPGALOP */ +#define WM8990_LLOPGALOP_BIT 0 +/* + * R53 (0x35) - Line Mixer2 + */ +#define WM8990_RROPGARON 0x0040 /* RROPGARON */ +#define WM8990_RROPGARON_BIT 6 +#define WM8990_RLOPGARON 0x0020 /* RLOPGARON */ +#define WM8990_RLOPGARON_BIT 5 +#define WM8990_ROPRON 0x0010 /* ROPRON */ +#define WM8990_ROPRON_BIT 4 +#define WM8990_RL12ROP 0x0004 /* RL12ROP */ +#define WM8990_RL12ROP_BIT 2 +#define WM8990_RR12ROP 0x0002 /* RR12ROP */ +#define WM8990_RR12ROP_BIT 1 +#define WM8990_RROPGAROP 0x0001 /* RROPGAROP */ +#define WM8990_RROPGAROP_BIT 0 + +/* + * R54 (0x36) - Speaker Mixer + */ +#define WM8990_LB2SPK 0x0080 /* LB2SPK */ +#define WM8990_LB2SPK_BIT 7 +#define WM8990_RB2SPK 0x0040 /* RB2SPK */ +#define WM8990_RB2SPK_BIT 6 +#define WM8990_LI2SPK 0x0020 /* LI2SPK */ +#define WM8990_LI2SPK_BIT 5 +#define WM8990_RI2SPK 0x0010 /* RI2SPK */ +#define WM8990_RI2SPK_BIT 4 +#define WM8990_LOPGASPK 0x0008 /* LOPGASPK */ +#define WM8990_LOPGASPK_BIT 3 +#define WM8990_ROPGASPK 0x0004 /* ROPGASPK */ +#define WM8990_ROPGASPK_BIT 2 +#define WM8990_LDSPK 0x0002 /* LDSPK */ +#define WM8990_LDSPK_BIT 1 +#define WM8990_RDSPK 0x0001 /* RDSPK */ +#define WM8990_RDSPK_BIT 0 + +/* + * R55 (0x37) - Additional Control + */ +#define WM8990_VROI 0x0001 /* VROI */ + +/* + * R56 (0x38) - AntiPOP1 + */ +#define WM8990_DIS_LLINE 0x0020 /* DIS_LLINE */ +#define WM8990_DIS_RLINE 0x0010 /* DIS_RLINE */ +#define WM8990_DIS_OUT3 0x0008 /* DIS_OUT3 */ +#define WM8990_DIS_OUT4 0x0004 /* DIS_OUT4 */ +#define WM8990_DIS_LOUT 0x0002 /* DIS_LOUT */ +#define WM8990_DIS_ROUT 0x0001 /* DIS_ROUT */ + +/* + * R57 (0x39) - AntiPOP2 + */ +#define WM8990_SOFTST 0x0040 /* SOFTST */ +#define WM8990_BUFIOEN 0x0008 /* BUFIOEN */ +#define WM8990_BUFDCOPEN 0x0004 /* BUFDCOPEN */ +#define WM8990_POBCTRL 0x0002 /* POBCTRL */ +#define WM8990_VMIDTOG 0x0001 /* VMIDTOG */ + +/* + * R58 (0x3A) - MICBIAS + */ +#define WM8990_MCDSCTH_MASK 0x00C0 /* MCDSCTH - [7:6] */ +#define WM8990_MCDTHR_MASK 0x0038 /* MCDTHR - [5:3] */ +#define WM8990_MCD 0x0004 /* MCD */ +#define WM8990_MBSEL 0x0001 /* MBSEL */ + +/* + * R60 (0x3C) - PLL1 + */ +#define WM8990_SDM 0x0080 /* SDM */ +#define WM8990_PRESCALE 0x0040 /* PRESCALE */ +#define WM8990_PLLN_MASK 0x000F /* PLLN - [3:0] */ + +/* + * R61 (0x3D) - PLL2 + */ +#define WM8990_PLLK1_MASK 0x00FF /* PLLK1 - [7:0] */ + +/* + * R62 (0x3E) - PLL3 + */ +#define WM8990_PLLK2_MASK 0x00FF /* PLLK2 - [7:0] */ + +/* + * R63 (0x3F) - Internal Driver Bits + */ +#define WM8990_INMIXL_PWR_BIT 0 +#define WM8990_AINLMUX_PWR_BIT 1 +#define WM8990_INMIXR_PWR_BIT 2 +#define WM8990_AINRMUX_PWR_BIT 3 + +struct wm8990_setup_data { + unsigned short i2c_address; +}; + +#define WM8990_MCLK_DIV 0 +#define WM8990_DACCLK_DIV 1 +#define WM8990_ADCCLK_DIV 2 +#define WM8990_BCLK_DIV 3 + +extern struct snd_soc_codec_dai wm8990_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8990; + +#endif /* __WM8990REGISTERDEFS_H__ */ +/*------------------------------ END OF FILE ---------------------------------*/ From 9aaca9683b014c4d718f32cbddb42e5b36ea3c0f Mon Sep 17 00:00:00 2001 From: Geoffrey Wossum Date: Thu, 5 Jun 2008 13:49:34 +0100 Subject: [PATCH 067/179] [ALSA] Revised AT32 ASoC Patch Attached is a revised version of my patch to add AT32 to ASoC. This cleans most of the style issues associated with the previous patch. Also fixes an issue with the playpaq_wm8510.c code depending on a non-released patch to th AT32 portmux support. Patch is against 2.6.24.3.atmel.3 kernel, the latest AVR32 kernel Atmel has released, with the linux-2.6-asoc patches from when v2.6.24 was tagged also applied. [Fixed up minor checkpatch issues and updated for current kernels -- broonie] Signed-off-by: Geoffrey Wossum Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 3 +- sound/soc/at32/Kconfig | 34 ++ sound/soc/at32/Makefile | 11 + sound/soc/at32/at32-pcm.c | 491 ++++++++++++++++++ sound/soc/at32/at32-pcm.h | 79 +++ sound/soc/at32/at32-ssc.c | 849 ++++++++++++++++++++++++++++++++ sound/soc/at32/at32-ssc.h | 59 +++ sound/soc/at32/playpaq_wm8510.c | 524 ++++++++++++++++++++ 9 files changed, 2050 insertions(+), 1 deletion(-) create mode 100644 sound/soc/at32/Kconfig create mode 100644 sound/soc/at32/Makefile create mode 100644 sound/soc/at32/at32-pcm.c create mode 100644 sound/soc/at32/at32-pcm.h create mode 100644 sound/soc/at32/at32-ssc.c create mode 100644 sound/soc/at32/at32-ssc.h create mode 100644 sound/soc/at32/playpaq_wm8510.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index fd7bc4f89072..b939e22db7b4 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -22,6 +22,7 @@ config SND_SOC_AC97_BUS bool # All the supported Soc's +source "sound/soc/at32/Kconfig" source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 782db2127108..3645f959c264 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/ +obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += omap/ diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig new file mode 100644 index 000000000000..b0765e86c085 --- /dev/null +++ b/sound/soc/at32/Kconfig @@ -0,0 +1,34 @@ +config SND_AT32_SOC + tristate "SoC Audio for the Atmel AT32 System-on-a-Chip" + depends on AVR32 && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the AT32 SSC interface. You will also need to + to select the audio interfaces to support below. + + +config SND_AT32_SOC_SSC + tristate + + + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_AT32_SOC && BOARD_PLAYPAQ + select SND_AT32_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + + + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile new file mode 100644 index 000000000000..c03e55ececeb --- /dev/null +++ b/sound/soc/at32/Makefile @@ -0,0 +1,11 @@ +# AT32 Platform Support +snd-soc-at32-objs := at32-pcm.o +snd-soc-at32-ssc-objs := at32-ssc.o + +obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o +obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c new file mode 100644 index 000000000000..cf76e89fae6e --- /dev/null +++ b/sound/soc/at32/at32-pcm.c @@ -0,0 +1,491 @@ +/* sound/soc/at32/at32-pcm.c + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-pcm.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "at32-pcm.h" + + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware at32_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE), + + .formats = SNDRV_PCM_FMTBIT_S16, + .period_bytes_min = 32, + .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */ + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct at32_runtime_data { + struct at32_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of DMA buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* Save PDC registers (for power management) */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *dmabuf = &substream->dma_buffer; + size_t size = at32_pcm_hardware.buffer_bytes_max; + + dmabuf->dev.type = SNDRV_DMA_TYPE_DEV; + dmabuf->dev.dev = pcm->card->dev; + dmabuf->private_data = NULL; + dmabuf->area = dma_alloc_coherent(pcm->card->dev, size, + &dmabuf->addr, GFP_KERNEL); + pr_debug("at32_pcm: preallocate_dma_buffer: " + "area=%p, addr=%p, size=%ld\n", + (void *)dmabuf->area, (void *)dmabuf->addr, size); + + if (!dmabuf->area) + return -ENOMEM; + + dmabuf->bytes = size; + return 0; +} + + + +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + static int count; + + count++; + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "underrun" : "overrun", params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + + snd_pcm_period_elapsed(substream); +} + + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int at32_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params + */ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = at32_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("hw_params: DMA for %s initialized " + "(dma_bytes=%ld, period_size=%ld)\n", + prtd->params->name, runtime->dma_bytes, prtd->period_size); + + return 0; +} + + + +static int at32_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + + + +static int at32_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + + return 0; +} + + +static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("at32_pcm_trigger: buffer_size = %ld, " + "dma_area = %p, dma_bytes = %ld\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("trigger: period_ptr=%lx, xpr=%x, " + "xcr=%d, xnpr=%x, xncr=%d\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%x, imr=%x\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + + + +static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + + + +static int at32_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + +out: + return ret; +} + + + +static int at32_pcm_close(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + + +static int at32_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + + + +static struct snd_pcm_ops at32_pcm_ops = { + .open = at32_pcm_open, + .close = at32_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = at32_pcm_hw_params, + .hw_free = at32_pcm_hw_free, + .prepare = at32_pcm_prepare, + .trigger = at32_pcm_trigger, + .pointer = at32_pcm_pointer, + .mmap = at32_pcm_mmap, +}; + + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 at32_pcm_dmamask = 0xffffffff; + +static int at32_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &at32_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = at32_pcm_preallocate_dma_buffer( + pcm, SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n"); + ret = at32_pcm_preallocate_dma_buffer( + pcm, SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + + +out: + return ret; +} + + + +static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (substream == NULL) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + + + +#ifdef CONFIG_PM +static int at32_pcm_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) + return 0; + prtd = runtime->private_data; + params = prtd->params; + + /* Disable the PDC and save the PDC registers */ + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + + + +static int at32_pcm_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) + return 0; + prtd = runtime->private_data; + params = prtd->params; + + /* Restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else /* CONFIG_PM */ +# define at32_pcm_suspend NULL +# define at32_pcm_resume NULL +#endif /* CONFIG_PM */ + + + +struct snd_soc_platform at32_soc_platform = { + .name = "at32-audio", + .pcm_ops = &at32_pcm_ops, + .pcm_new = at32_pcm_new, + .pcm_free = at32_pcm_free_dma_buffers, + .suspend = at32_pcm_suspend, + .resume = at32_pcm_resume, +}; +EXPORT_SYMBOL_GPL(at32_soc_platform); + + + +MODULE_AUTHOR("Geoffrey Wossum "); +MODULE_DESCRIPTION("Atmel AT32 PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h new file mode 100644 index 000000000000..2a52430417da --- /dev/null +++ b/sound/soc/at32/at32-pcm.h @@ -0,0 +1,79 @@ +/* sound/soc/at32/at32-pcm.h + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_PCM_H +#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__ + +#include + + +/* + * Registers and status bits that are required by the PCM driver + * TODO: Is ptcr really used? + */ +struct at32_pdc_regs { + u32 xpr; /* PDC RX/TX pointer */ + u32 xcr; /* PDC RX/TX counter */ + u32 xnpr; /* PDC next RX/TX pointer */ + u32 xncr; /* PDC next RX/TX counter */ + u32 ptcr; /* PDC transfer control */ +}; + + + +/* + * SSC mask info + */ +struct at32_ssc_mask { + u32 ssc_enable; /* SSC RX/TX enable */ + u32 ssc_disable; /* SSC RX/TX disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */ + u32 pdc_enable; /* PDC RX/TX enable */ + u32 pdc_disable; /* PDC RX/TX disable */ +}; + + + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct at32_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct at32_pdc_regs *pdc; /* PDC register info */ + struct at32_ssc_mask *mask; /* SSC mask info */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler) (u32, struct snd_pcm_substream *); +}; + + + +/* + * The AT32 ASoC platform driver + */ +extern struct snd_soc_platform at32_soc_platform; + + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* __SOUND_SOC_AT32_AT32_PCM_H */ diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c new file mode 100644 index 000000000000..0ca44107f183 --- /dev/null +++ b/sound/soc/at32/at32-ssc.c @@ -0,0 +1,849 @@ +/* sound/soc/at32/at32-ssc.c + * ASoC platform driver for AT32 using SSC as DAI + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-ssc.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +/* #define DEBUG */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "at32-pcm.h" +#include "at32-ssc.h" + + + +/*-------------------------------------------------------------------------*\ + * Constants +\*-------------------------------------------------------------------------*/ +#define NUM_SSC_DEVICES 3 + +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of . These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + + + +/*-------------------------------------------------------------------------*\ + * Module data +\*-------------------------------------------------------------------------*/ +/* + * SSC PDC registered required by the PCM DMA engine + */ +static struct at32_pdc_regs pdc_tx_reg = { + .xpr = SSC_PDC_TPR, + .xcr = SSC_PDC_TCR, + .xnpr = SSC_PDC_TNPR, + .xncr = SSC_PDC_TNCR, +}; + + + +static struct at32_pdc_regs pdc_rx_reg = { + .xpr = SSC_PDC_RPR, + .xcr = SSC_PDC_RCR, + .xnpr = SSC_PDC_RNPR, + .xncr = SSC_PDC_RNCR, +}; + + + +/* + * SSC and PDC status bits for transmit and receive + */ +static struct at32_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS), +}; + + + +static struct at32_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS), +}; + + + +/* + * DMA parameters for each SSC + */ +static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + { + { + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, +}; + + + +static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +}; + + + + +/*-------------------------------------------------------------------------*\ + * ISR +\*-------------------------------------------------------------------------*/ +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt + * handler in the PCM driver. + */ +static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id) +{ + struct at32_ssc_info *ssc_p = dev_id; + struct at32_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) & + ssc_readl(ssc_p->ssc->regs, IMR)); + + /* + * Loop through substreams attached to this SSC. If a DMA-related + * interrupt occured on that substream, call the DMA interrupt + * handler function, if one has been registered in the dma_param + * structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + + return IRQ_HANDLED; +} + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int at32_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + + + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void at32_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + int dir_mask; + + dma_params = ssc_p->dma_params[substream->stream]; + + if (dma_params != NULL) { + ssc_writel(dma_params->ssc->regs, CR, + dma_params->mask->ssc_disable); + pr_debug("%s disabled SSC_SR=0x%08x\n", + (substream->stream ? "receiver" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[substream->stream] = NULL; + } + + + dir_mask = 1 << substream->stream; + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock */ + pr_debug("at32-ssc: Stopping user %d clock\n", + ssc_p->ssc->user); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + /* clear the SSC dividers */ + ssc_p->cmr_div = 0; + ssc_p->tcmr_period = 0; + ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} + + + +/* + * Set the SSC system clock rate + */ +static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + /* TODO: What the heck do I do here? */ + return 0; +} + + + +/* + * Record DAI format for use by hw_params() + */ +static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, + unsigned int fmt) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + + + +/* + * Record SSC clock dividers for use in hw_params() + */ +static int at32_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, + int div_id, int div) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case AT32_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case AT32_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case AT32_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + + + +/* + * Configure the SSC + */ +static int at32_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int id = rtd->dai->cpu_dai->id; + struct at32_ssc_info *ssc_p = &ssc_info[id]; + struct at32_pcm_dma_params *dma_params; + int channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + + /* + * Currently, there is only one set of dma_params for each direction. + * If more are added, this code will have to be changed to select + * the proper set + */ + dma_params = &ssc_dma_params[id][substream->stream]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[substream->stream] = dma_params; + + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the PCM driver's hw_params() + * function. It should not be used for other purposes as it + * is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + + /* + * Determine sample size in bits and the PDC increment + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + + case SNDRV_PCM_FORMAT_S16: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + + case SNDRV_PCM_FORMAT_S24: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + + case SNDRV_PCM_FORMAT_S32: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + + default: + pr_warning("at32-ssc: Unsupported PCM format %d", + params_format(params)); + return -EINVAL; + } + pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n", + bits, dma_params->pdc_xfer_size, channels); + + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) + if (bits > 16) { + pr_warning("at32-ssc: " + "sample size %d is too large for I2S\n", + bits); + return -EINVAL; + } + + + /* + * Compute the SSC register settings + */ + switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRS clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, SSC_START_FALLING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(RFMR_FSLEN, bits - 1) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, SSC_START_FALLING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(TFMR_FSLEN, bits - 1) | + SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) | + SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clock. + * + * The SSC transmit clock is obtained from the BCLK signal + * on the TK line, and the SSC receive clock is generated from + * the transmit clock. + * + * For single channel data, one sample is transferred on the + * falling edge of the LRC clock. For two channel data, one + * sample is transferred on both edges of the LRC clock. + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n"); + start_event = ((channels == 1) ? + SSC_START_FALLING_RF : SSC_START_EDGE_RF); + + rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, start_event) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_CLOCK)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, start_event) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_NONE) | + SSC_BF(TCMR_CKS, SSC_CKS_PIN)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, 1) | + SSC_BF(RCMR_START, SSC_START_RISING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, 1) | + SSC_BF(TCMR_START, SSC_START_RISING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_RISING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(TFMR_DATNB, channels - 1) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + pr_warning("at32-ssc: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + + if (!ssc_p->initialized) { + /* enable peripheral clock */ + pr_debug("at32-ssc: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + pr_warning("at32-ssc: request irq failed (%d)\n", ret); + pr_debug("at32-ssc: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* Set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("at32-ssc: SSC initialized\n"); + return 0; +} + + + +static int at32_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + + dma_params = ssc_p->dma_params[substream->stream]; + + ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable); + + return 0; +} + + + +#ifdef CONFIG_PM +static int at32_ssc_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int at32_ssc_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define at32_ssc_suspend NULL +# define at32_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define AT32_SSC_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + + +#define AT32_SSC_FORMATS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \ + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) + + +struct snd_soc_cpu_dai at32_ssc_dai[NUM_SSC_DEVICES] = { + { + .name = "at32-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[0], + }, + { + .name = "at32-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[1], + }, + { + .name = "at32-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[2], + }, +}; +EXPORT_SYMBOL_GPL(at32_ssc_dai); + + +MODULE_AUTHOR("Geoffrey Wossum "); +MODULE_DESCRIPTION("AT32 SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h new file mode 100644 index 000000000000..3c6901a6c036 --- /dev/null +++ b/sound/soc/at32/at32-ssc.h @@ -0,0 +1,59 @@ +/* sound/soc/at32/at32-ssc.h + * ASoC SSC interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_SSC_H +#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__ + +#include +#include + +#include "at32-pcm.h" + + + +struct at32_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + + +struct at32_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct at32_pcm_dma_params *dma_params[2]; + struct at32_ssc_state ssc_state; +}; + + +/* SSC divider ids */ +#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ + + +extern struct snd_soc_cpu_dai at32_ssc_dai[]; + + + +#endif /* __SOUND_SOC_AT32_AT32_SSC_H */ diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c new file mode 100644 index 000000000000..d6b9fd5e6123 --- /dev/null +++ b/sound/soc/at32/playpaq_wm8510.c @@ -0,0 +1,524 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + * + * NOTE: If you don't have the AT32 enhanced portmux configured (which + * isn't currently in the mainline or Atmel patched kernel), you will + * need to set the MCLK pin (PA30) to peripheral A in your board initialization + * code. Something like: + * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); + * + */ + +/* #define DEBUG */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + /* + * set CPU and CODEC DAI configuration + */ + ret = codec_dai->dai_ops.set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CODEC DAI format (%d)\n", + ret); + return ret; + } + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU DAI format (%d)\n", + ret); + return ret; + } + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", + ret); + return ret; + } + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: " + "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning + ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = codec_dai->dai_ops.set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", + ret); + return ret; + } + + + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, + WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", + ret); + return ret; + } + + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .hw_params = playpaq_wm8510_hw_params, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + {"Ext Spk", NULL, "SPKOUTP"}, + {"Ext Spk", NULL, "SPKOUTN"}, + + {"Mic Bias", NULL, "Int Mic"}, + {"MICN", NULL, "Mic Bias"}, + {"MICP", NULL, "Mic Bias"}, + + /* Terminator */ + {NULL, NULL, NULL}, +}; + + + +static int playpaq_wm8510_init(struct snd_soc_codec *codec) +{ + int i; + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], intercon[i][2]); + } + + + /* always connected endpoints */ + snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); + snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_sync_endpoints(codec); + + + + /* Make CSB show PLL rate */ + codec->dai->dai_ops.set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); +#endif + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_free_pin(MCLK_PIN); +#endif + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + +MODULE_AUTHOR("Geoffrey Wossum "); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL"); From 40445ac5e93f1b97e1d7a8c5bed9d34bd89b7550 Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Thu, 5 Jun 2008 21:22:25 +0300 Subject: [PATCH 068/179] [ALSA] remove SND_GUS_SYNTH After the removal of the sequencer instrument layer SND_GUS_SYNTH was no longer used. Reported-by: Robert P. J. Day Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 4575ba865910..d649a0e966da 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -202,14 +202,10 @@ config SND_SC6000 To compile this driver as a module, choose M here: the module will be called snd-sc6000. -config SND_GUS_SYNTH - tristate - config SND_GUSCLASSIC tristate "Gravis UltraSound Classic" select SND_RAWMIDI select SND_PCM - select SND_GUS_SYNTH help Say Y here to include support for Gravis UltraSound Classic soundcards. @@ -222,7 +218,6 @@ config SND_GUSEXTREME select SND_HWDEP select SND_MPU401_UART select SND_PCM - select SND_GUS_SYNTH help Say Y here to include support for Gravis UltraSound Extreme soundcards. @@ -234,7 +229,6 @@ config SND_GUSMAX tristate "Gravis UltraSound MAX" select SND_RAWMIDI select SND_CS4231_LIB - select SND_GUS_SYNTH help Say Y here to include support for Gravis UltraSound MAX soundcards. @@ -247,7 +241,6 @@ config SND_INTERWAVE depends on PNP select SND_RAWMIDI select SND_CS4231_LIB - select SND_GUS_SYNTH help Say Y here to include support for AMD InterWave based soundcards (Gravis UltraSound Plug & Play, STB SoundRage32, @@ -261,7 +254,6 @@ config SND_INTERWAVE_STB depends on PNP select SND_RAWMIDI select SND_CS4231_LIB - select SND_GUS_SYNTH help Say Y here to include support for AMD InterWave based soundcards with a TEA6330T bass and treble regulator From 33b9cd8214d6e385fcd63c321252a359ee96d568 Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Thu, 5 Jun 2008 21:24:10 +0300 Subject: [PATCH 069/179] [ALSA] usbaudio.c: remove #ifndef CONFIG_USB_EHCI_SPLIT_ISO code Since USB_EHCI_SPLIT_ISO is now unconditionally enabled the #ifndef CONFIG_USB_EHCI_SPLIT_ISO became wrong. Reported-by: Robert P. J. Day Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 410be4aff1ba..b8cfb7c22768 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -819,10 +819,6 @@ static const char *usb_error_string(int err) return "device disabled"; case -EHOSTUNREACH: return "device suspended"; -#ifndef CONFIG_USB_EHCI_SPLIT_ISO - case -ENOSYS: - return "enable CONFIG_USB_EHCI_SPLIT_ISO to play through a hub"; -#endif case -EINVAL: case -EAGAIN: case -EFBIG: From e7bd3de08e5208b09fbad06dd80294ea0e86f7bf Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Thu, 5 Jun 2008 21:29:18 +0300 Subject: [PATCH 070/179] ALSA: make sparc/dbri.c:snd_dbri_proc() static This patch makes the needlessly global snd_dbri_proc() static. Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai --- sound/sparc/dbri.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 3d00e0797b11..ee2e1b4f3551 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2490,7 +2490,7 @@ static void dbri_debug_read(struct snd_info_entry *entry, } #endif -void __devinit snd_dbri_proc(struct snd_card *card) +static void __devinit snd_dbri_proc(struct snd_card *card) { struct snd_dbri *dbri = card->private_data; struct snd_info_entry *entry; From 0264b3b636f1ba5777426dd9b5bb5cbbeb5fcb34 Mon Sep 17 00:00:00 2001 From: Juergen Beisert Date: Fri, 6 Jun 2008 17:02:57 +0200 Subject: [PATCH 071/179] [ALSA] ac97: add support for wm9711 master left inv switch This patch adds support for Master Left Inv Switch on wm9711. At least required to drive the mono speaker on the PXA270 platfrom Signed-off-by: Juergen Beisert Acked-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_patch.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 92817f7d46d2..abe88adec37e 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -669,6 +669,7 @@ AC97_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), AC97_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), AC97_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0), +AC97_SINGLE("Master Left Inv Switch", AC97_MASTER, 6, 1, 0), AC97_SINGLE("Master ZC Switch", AC97_MASTER, 7, 1, 0), AC97_SINGLE("Headphone ZC Switch", AC97_HEADPHONE, 7, 1, 0), AC97_SINGLE("Mono ZC Switch", AC97_MASTER_MONO, 7, 1, 0), From 05a076de83b507238cd186e93daeaf15820d570e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 6 Jun 2008 17:13:53 +0100 Subject: [PATCH 072/179] [ALSA] ASoC: Fix default mono mixer configuration for WM8510 Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm8510.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 6f54cb0f50fd..152e6f21154f 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -71,7 +71,7 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { 0x0003, 0x0010, 0x0000, 0x0000, 0x0000, 0x0002, 0x0001, 0x0000, 0x0000, 0x0000, 0x0039, 0x0000, - 0x0000, + 0x0001, }; /* From 38f582270feb19ea8821cb64c5d2fdbd6bdd0bd9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 8 Jun 2008 09:17:27 +0200 Subject: [PATCH 073/179] [ALSA] ac97 - Fix power_save option value as time-out The power_save option was set as boot although it was meant to be a timeout value like the same option of snd-hda-intel originally. Now fixed to the same style. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_codec.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 45fd29017ddd..2d2f16e11082 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -49,8 +49,9 @@ MODULE_PARM_DESC(enable_loopback, "Enable AC97 ADC/DAC Loopback Control"); #ifdef CONFIG_SND_AC97_POWER_SAVE static int power_save = CONFIG_SND_AC97_POWER_SAVE_DEFAULT; -module_param(power_save, bool, 0644); -MODULE_PARM_DESC(power_save, "Enable AC97 power-saving control"); +module_param(power_save, int, 0644); +MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " + "(in second, 0 = disable)."); #endif /* @@ -2362,7 +2363,7 @@ int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup) * that open/close frequently) */ schedule_delayed_work(&ac97->power_work, - msecs_to_jiffies(2000)); + msecs_to_jiffies(power_save * 1000)); else { cancel_delayed_work(&ac97->power_work); update_power_regs(ac97); From 3ae5f36a368b143cfd8a6f7a988a012c1b46eda0 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 8 Jun 2008 07:57:53 +0200 Subject: [PATCH 074/179] [ALSA] opti93x: fix sound ouput for Opti930 This patch fixes silenced output from the Opti930. Signed-off-by: Krzysztof Helt Acked-by: Rene Herman Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/isa/opti9xx/opti92x-ad1848.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index fe1afc13a01d..fc15b88364b4 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -491,16 +491,9 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) break; #else /* OPTi93X */ - case OPTi9XX_HW_82C930: case OPTi9XX_HW_82C931: case OPTi9XX_HW_82C933: - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 | - (chip->hardware == OPTi9XX_HW_82C930 ? 0x00 : 0x04), - 0x34); - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x20, 0xbf); - /* + /* * The BTC 1817DW has QS1000 wavetable which is connected * to the serial digital input of the OPTI931. */ @@ -510,6 +503,13 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) * or digital input signal. */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01); + case OPTi9XX_HW_82C930: /* FALL THROUGH */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 | + (chip->hardware == OPTi9XX_HW_82C930 ? 0x00 : 0x04), + 0x34); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x20, 0xbf); break; #endif /* OPTi93X */ From 36ca6e133ae3a57318a3ad8a41b2f005e3934c29 Mon Sep 17 00:00:00 2001 From: Akio Idehara Date: Mon, 9 Jun 2008 22:57:40 +0900 Subject: [PATCH 075/179] [ALSA] hda - Add Toshiba dynabook SS RX1 support I have Toshiba dynabook SS RX1 and this patch adds that support. Signed-off-by: Akio Idehara Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7997e13b59a6..61f8c13b2bb9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9719,6 +9719,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", + ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), From 781711a93e8aae756e8ae07acef185a6dfe552b8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jun 2008 12:34:45 +0200 Subject: [PATCH 076/179] ALSA: emu10k1 - Fix page allocation with GFP_DMA Added the missing GFP_ATOMIC to page_alloc when called with GFP_DMA. GFP_KERNEL often results in stalls for ZONE_DMA, so GFP_ATOMIC is more prgmatic. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/emu10k1/memory.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 128eaca17a61..42943b4fcb7b 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -466,7 +466,7 @@ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk __GFP_NOWARN); if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) /* try to allocate from <16MB zone */ - p = alloc_page(GFP_DMA | + p = alloc_page(GFP_ATOMIC | GFP_DMA | __GFP_NORETRY | /* no OOM-killer */ __GFP_NOWARN); if (!p) { From 0a1b42db4bf9db233d1f29795086a9526e84c845 Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Tue, 10 Jun 2008 01:26:28 +0300 Subject: [PATCH 077/179] sound: sound/oss/dmasound/: cleanups This patch contains the following cleanups: - make the following needlessly global functions static: - dmasound_core.c: get_afmt_string() - dmasound_paula.c: dmasound_paula_init() - dmasound_q40.c: dmasound_q40_init() - remove the following unused global variable: - dmasound_core.c: software_input_volume Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/oss/dmasound/dmasound_core.c | 7 +------ sound/oss/dmasound/dmasound_paula.c | 2 +- sound/oss/dmasound/dmasound_q40.c | 2 +- 3 files changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index a003c0ea9303..95fc5c681755 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -211,10 +211,6 @@ static int state_unit = -1; static int irq_installed; #endif /* MODULE */ -/* software implemented recording volume! */ -uint software_input_volume = SW_INPUT_VOLUME_SCALE * SW_INPUT_VOLUME_DEFAULT; -EXPORT_SYMBOL(software_input_volume); - /* control over who can modify resources shared between play/record */ static mode_t shared_resource_owner; static int shared_resources_initialised; @@ -1188,7 +1184,7 @@ static struct { /* publish this function for use by low-level code, if required */ -char *get_afmt_string(int afmt) +static char *get_afmt_string(int afmt) { switch(afmt) { case AFMT_MU_LAW: @@ -1551,4 +1547,3 @@ EXPORT_SYMBOL(dmasound_catchRadius); EXPORT_SYMBOL(dmasound_ulaw2dma8); EXPORT_SYMBOL(dmasound_alaw2dma8); #endif -EXPORT_SYMBOL(get_afmt_string) ; diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c index 202e8103dc4d..06e9e88e4c05 100644 --- a/sound/oss/dmasound/dmasound_paula.c +++ b/sound/oss/dmasound/dmasound_paula.c @@ -710,7 +710,7 @@ static MACHINE machAmiga = { /*** Config & Setup **********************************************************/ -int __init dmasound_paula_init(void) +static int __init dmasound_paula_init(void) { int err; diff --git a/sound/oss/dmasound/dmasound_q40.c b/sound/oss/dmasound/dmasound_q40.c index b3379dd7ca5e..1855b14d90c3 100644 --- a/sound/oss/dmasound/dmasound_q40.c +++ b/sound/oss/dmasound/dmasound_q40.c @@ -611,7 +611,7 @@ static MACHINE machQ40 = { /*** Config & Setup **********************************************************/ -int __init dmasound_q40_init(void) +static int __init dmasound_q40_init(void) { if (MACH_IS_Q40) { dmasound.mach = machQ40; From 675f25d4d3ff6501cbce608bcc2333a56ec4c105 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jun 2008 17:53:20 +0200 Subject: [PATCH 078/179] ALSA: hda - Add bdl_pos_adj option Added a new option, bdl_pos_adj, to adjust the delay of IRQ-wakeup timing. Most HD-audio hardwares have a problem that a BDL IRQ is issued before actually the data and the DMA pointer are updated. We have already a mechanism to force to delay snd_pcm_period_elapsed() calls via workq, but this costs much CPU, and typically the delay is within one sample. Thus, it's more clever to adjust the BDL entries instead. The new option adds the size of the delay in frames. As default, it's set to 1 -- that is, one sample delay. Even the hardware is really correct, one sample delay is relatively harmless in comparison with reporting wrong positions. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 117 +++++++++++++++++++++++++++----------- 1 file changed, 85 insertions(+), 32 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index dc68709e7569..1ec3fd4c8940 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -58,6 +58,7 @@ static int position_fix[SNDRV_CARDS]; static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int single_cmd; static int enable_msi; +static int bdl_pos_adj = 1; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -77,6 +78,8 @@ MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs " "(for debugging only)."); module_param(enable_msi, int, 0444); MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)"); +module_param(bdl_pos_adj, int, 0644); +MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset"); #ifdef CONFIG_SND_HDA_POWER_SAVE /* power_save option is defined in hda_codec.c */ @@ -309,7 +312,8 @@ struct azx_dev { unsigned int opened :1; unsigned int running :1; - unsigned int irq_pending: 1; + unsigned int irq_pending :1; + unsigned int irq_ignore :1; }; /* CORB/RIRB */ @@ -943,6 +947,11 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); if (!azx_dev->substream || !azx_dev->running) continue; + /* ignore the first dummy IRQ (due to pos_adj) */ + if (azx_dev->irq_ignore) { + azx_dev->irq_ignore = 0; + continue; + } /* check whether this IRQ is really acceptable */ if (azx_position_ok(chip, azx_dev)) { azx_dev->irq_pending = 0; @@ -976,15 +985,54 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) } +/* + * set up a BDL entry + */ +static int setup_bdle(struct snd_pcm_substream *substream, + struct azx_dev *azx_dev, u32 **bdlp, + int ofs, int size, int with_ioc) +{ + struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream); + u32 *bdl = *bdlp; + + while (size > 0) { + dma_addr_t addr; + int chunk; + + if (azx_dev->frags >= AZX_MAX_BDL_ENTRIES) + return -EINVAL; + + addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs); + /* program the address field of the BDL entry */ + bdl[0] = cpu_to_le32((u32)addr); + bdl[1] = cpu_to_le32(upper_32bit(addr)); + /* program the size field of the BDL entry */ + chunk = PAGE_SIZE - (ofs % PAGE_SIZE); + if (size < chunk) + chunk = size; + bdl[2] = cpu_to_le32(chunk); + /* program the IOC to enable interrupt + * only when the whole fragment is processed + */ + size -= chunk; + bdl[3] = (size || !with_ioc) ? 0 : cpu_to_le32(0x01); + bdl += 4; + azx_dev->frags++; + ofs += chunk; + } + *bdlp = bdl; + return ofs; +} + /* * set up BDL entries */ static int azx_setup_periods(struct snd_pcm_substream *substream, struct azx_dev *azx_dev) { - struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream); u32 *bdl; int i, ofs, periods, period_bytes; + int pos_adj = 0; /* reset BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, 0); @@ -998,39 +1046,44 @@ static int azx_setup_periods(struct snd_pcm_substream *substream, bdl = (u32 *)azx_dev->bdl.area; ofs = 0; azx_dev->frags = 0; - for (i = 0; i < periods; i++) { - int size, rest; - if (i >= AZX_MAX_BDL_ENTRIES) { - snd_printk(KERN_ERR "Too many BDL entries: " - "buffer=%d, period=%d\n", - azx_dev->bufsize, period_bytes); - /* reset */ - azx_sd_writel(azx_dev, SD_BDLPL, 0); - azx_sd_writel(azx_dev, SD_BDLPU, 0); - return -EINVAL; + azx_dev->irq_ignore = 0; + if (bdl_pos_adj > 0) { + struct snd_pcm_runtime *runtime = substream->runtime; + pos_adj = (bdl_pos_adj * runtime->rate + 47999) / 48000; + if (!pos_adj) + pos_adj = 1; + pos_adj = frames_to_bytes(runtime, pos_adj); + if (pos_adj >= period_bytes) { + snd_printk(KERN_WARNING "Too big adjustment %d\n", + bdl_pos_adj); + pos_adj = 0; + } else { + ofs = setup_bdle(substream, azx_dev, + &bdl, ofs, pos_adj, 1); + if (ofs < 0) + goto error; + azx_dev->irq_ignore = 1; } - rest = period_bytes; - do { - dma_addr_t addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs); - /* program the address field of the BDL entry */ - bdl[0] = cpu_to_le32((u32)addr); - bdl[1] = cpu_to_le32(upper_32bit(addr)); - /* program the size field of the BDL entry */ - size = PAGE_SIZE - (ofs % PAGE_SIZE); - if (rest < size) - size = rest; - bdl[2] = cpu_to_le32(size); - /* program the IOC to enable interrupt - * only when the whole fragment is processed - */ - rest -= size; - bdl[3] = rest ? 0 : cpu_to_le32(0x01); - bdl += 4; - azx_dev->frags++; - ofs += size; - } while (rest > 0); + } + for (i = 0; i < periods; i++) { + if (i == periods - 1 && pos_adj) + ofs = setup_bdle(substream, azx_dev, &bdl, ofs, + period_bytes - pos_adj, 0); + else + ofs = setup_bdle(substream, azx_dev, &bdl, ofs, + period_bytes, 1); + if (ofs < 0) + goto error; } return 0; + + error: + snd_printk(KERN_ERR "Too many BDL entries: buffer=%d, period=%d\n", + azx_dev->bufsize, period_bytes); + /* reset */ + azx_sd_writel(azx_dev, SD_BDLPL, 0); + azx_sd_writel(azx_dev, SD_BDLPU, 0); + return -EINVAL; } /* From d2e1c97331d37a477ef7e9a272b4de4e53b893be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jun 2008 17:53:34 +0200 Subject: [PATCH 079/179] ALSA: hda - remove position_fix=3 position_fix=3 is the option to correct the DMA position with the FIFO size. But, it never worked correctly, and we have now more other workarounds for the DMA position fixes. Thus better to remove it. Also, change POS_FIX_NONE to POS_FIX_LPIB to represent its real role better. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/ALSA-Configuration.txt | 2 +- sound/pci/hda/hda_intel.c | 15 ++++++--------- 2 files changed, 7 insertions(+), 10 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 529073dea196..ab28f7b62f29 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -753,7 +753,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. [Multiple options for each card instance] model - force the model name - position_fix - Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size) + position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF) probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) [Single (global) options] diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1ec3fd4c8940..0ff8b9b9629e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -70,7 +70,7 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "Fix DMA pointer " - "(0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size)."); + "(0 = auto, 1 = none, 2 = POSBUF)."); module_param_array(probe_mask, int, NULL, 0444); MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1)."); module_param(single_cmd, bool, 0444); @@ -266,9 +266,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; /* position fix mode */ enum { POS_FIX_AUTO, - POS_FIX_NONE, + POS_FIX_LPIB, POS_FIX_POSBUF, - POS_FIX_FIFO, }; /* Defines for ATI HD Audio support in SB450 south bridge */ @@ -1506,8 +1505,6 @@ static unsigned int azx_get_position(struct azx *chip, } else { /* read LPIB */ pos = azx_sd_readl(azx_dev, SD_LPIB); - if (chip->position_fix == POS_FIX_FIFO) - pos += azx_dev->fifo_size; } if (pos >= azx_dev->bufsize) pos = 0; @@ -1542,7 +1539,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) printk(KERN_WARNING "hda-intel: Invalid position buffer, " "using LPIB read method instead.\n"); - chip->position_fix = POS_FIX_NONE; + chip->position_fix = POS_FIX_LPIB; pos = azx_get_position(chip, azx_dev); } else chip->position_fix = POS_FIX_POSBUF; @@ -1917,9 +1914,9 @@ static int azx_dev_free(struct snd_device *device) * white/black-listing for position_fix */ static struct snd_pci_quirk position_fix_list[] __devinitdata = { - SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_NONE), - SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_NONE), - SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_NONE), + SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), {} }; From 555e219ffaeecbd08d1603e1551188483e7e3b64 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jun 2008 17:53:34 +0200 Subject: [PATCH 080/179] ALSA: hda - bdl_pos_adj option to each instance The option bdl_pos_adj should be provided for each card instance instead of a global one because the value depends rather on each controller-chip. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 25 +++++++++++++++---------- 1 file changed, 15 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0ff8b9b9629e..ddae3c479a88 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -55,10 +55,10 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static char *model[SNDRV_CARDS]; static int position_fix[SNDRV_CARDS]; +static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1}; static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int single_cmd; static int enable_msi; -static int bdl_pos_adj = 1; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -71,6 +71,8 @@ MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "Fix DMA pointer " "(0 = auto, 1 = none, 2 = POSBUF)."); +module_param_array(bdl_pos_adj, int, NULL, 0644); +MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1)."); module_param(single_cmd, bool, 0444); @@ -78,8 +80,6 @@ MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs " "(for debugging only)."); module_param(enable_msi, int, 0444); MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)"); -module_param(bdl_pos_adj, int, 0644); -MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset"); #ifdef CONFIG_SND_HDA_POWER_SAVE /* power_save option is defined in hda_codec.c */ @@ -330,6 +330,7 @@ struct azx_rb { struct azx { struct snd_card *card; struct pci_dev *pci; + int dev_index; /* chip type specific */ int driver_type; @@ -1026,12 +1027,13 @@ static int setup_bdle(struct snd_pcm_substream *substream, /* * set up BDL entries */ -static int azx_setup_periods(struct snd_pcm_substream *substream, +static int azx_setup_periods(struct azx *chip, + struct snd_pcm_substream *substream, struct azx_dev *azx_dev) { u32 *bdl; int i, ofs, periods, period_bytes; - int pos_adj = 0; + int pos_adj; /* reset BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, 0); @@ -1046,15 +1048,16 @@ static int azx_setup_periods(struct snd_pcm_substream *substream, ofs = 0; azx_dev->frags = 0; azx_dev->irq_ignore = 0; - if (bdl_pos_adj > 0) { + pos_adj = bdl_pos_adj[chip->dev_index]; + if (pos_adj > 0) { struct snd_pcm_runtime *runtime = substream->runtime; - pos_adj = (bdl_pos_adj * runtime->rate + 47999) / 48000; + pos_adj = (pos_adj * runtime->rate + 47999) / 48000; if (!pos_adj) pos_adj = 1; pos_adj = frames_to_bytes(runtime, pos_adj); if (pos_adj >= period_bytes) { snd_printk(KERN_WARNING "Too big adjustment %d\n", - bdl_pos_adj); + bdl_pos_adj[chip->dev_index]); pos_adj = 0; } else { ofs = setup_bdle(substream, azx_dev, @@ -1063,7 +1066,8 @@ static int azx_setup_periods(struct snd_pcm_substream *substream, goto error; azx_dev->irq_ignore = 1; } - } + } else + pos_adj = 0; for (i = 0; i < periods; i++) { if (i == periods - 1 && pos_adj) ofs = setup_bdle(substream, azx_dev, &bdl, ofs, @@ -1388,7 +1392,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", azx_dev->bufsize, azx_dev->format_val); - if (azx_setup_periods(substream, azx_dev) < 0) + if (azx_setup_periods(chip, substream, azx_dev) < 0) return -EINVAL; azx_setup_controller(chip, azx_dev); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -2001,6 +2005,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->irq = -1; chip->driver_type = driver_type; chip->msi = enable_msi; + chip->dev_index = dev; INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work); chip->position_fix = check_position_fix(chip, position_fix[dev]); From 5c0d7bc103dd1ae85967fbcf70be8d1ae3aa3d79 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jun 2008 17:53:35 +0200 Subject: [PATCH 081/179] ALSA: hda - Fix bdl_pos_adj value for ATI SB chipsets ATI SB controllers seem to report the DMA ahead in the amount of FIFO. Thus bdl_pos_adj should be 32 for them as default. Also, the default value is set to -1, which means to make the driver to choose the appropriate value. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ddae3c479a88..9b2dc0669b94 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -55,7 +55,7 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static char *model[SNDRV_CARDS]; static int position_fix[SNDRV_CARDS]; -static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1}; +static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int single_cmd; static int enable_msi; @@ -2013,6 +2013,18 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->single_cmd = single_cmd; + if (bdl_pos_adj[dev] < 0) { + switch (chip->driver_type) { + case AZX_DRIVER_ATI: + case AZX_DRIVER_ATIHDMI: + bdl_pos_adj[dev] = 32; + break; + default: + bdl_pos_adj[dev] = 1; + break; + } + } + #if BITS_PER_LONG != 64 /* Fix up base address on ULI M5461 */ if (chip->driver_type == AZX_DRIVER_ULI) { From 8c53b96f229aa6e27f17855cbbf8e4243868a3a9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jun 2008 17:53:35 +0200 Subject: [PATCH 082/179] ALSA: hda - Add description of bdl_pos_adj option Added a brief description of the new bdl_pos_adj option to ALSA-Configuration.txt. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/ALSA-Configuration.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index ab28f7b62f29..72aff61e7315 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -755,6 +755,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. model - force the model name position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF) probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) + bdl_pos_adj - Specifies the DMA IRQ timing delay in samples. + Passing -1 will make the driver to choose the appropriate + value based on the controller chip. [Single (global) options] single_cmd - Use single immediate commands to communicate with From a6a950a8a8e1431e8527809339e089ac926a7d34 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jun 2008 17:53:35 +0200 Subject: [PATCH 083/179] ALSA: hda - Add a warning if pending IRQ is found The pending IRQ handling is a very hackish workaround and should be avoided as much as possible via a larger bdl_pos_adj option value. Put a warning message if this situation occurs so that the user may have a chance to notice that something is wrong. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9b2dc0669b94..ec4ae2d94000 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -374,6 +374,7 @@ struct azx { unsigned int single_cmd :1; unsigned int polling_mode :1; unsigned int msi :1; + unsigned int irq_pending_warned :1; /* for debugging */ unsigned int last_cmd; /* last issued command (to sync) */ @@ -1562,6 +1563,14 @@ static void azx_irq_pending_work(struct work_struct *work) struct azx *chip = container_of(work, struct azx, irq_pending_work); int i, pending; + if (!chip->irq_pending_warned) { + printk(KERN_WARNING + "hda-intel: IRQ timing workaround is activated " + "for card #%d. Suggest a bigger bdl_pos_adj.\n", + chip->card->number); + chip->irq_pending_warned = 1; + } + for (;;) { pending = 0; spin_lock_irq(&chip->reg_lock); From abf1f5aafc1939db1f252e33914a3689e0f5830f Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 9 Jun 2008 23:07:28 +0200 Subject: [PATCH 084/179] ALSA: opti93x: add support for Opti93x codec in cs4231-lib This patch adds support for WSS compatible Opti93x codec to the cs4231-lib. Signed-off-by: Krzysztof Helt Tested-by: Rene Herman Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/cs4231-regs.h | 8 +++ include/sound/cs4231.h | 3 + sound/isa/cs423x/cs4231_lib.c | 118 +++++++++++++++++++++++++++++++--- 3 files changed, 120 insertions(+), 9 deletions(-) diff --git a/include/sound/cs4231-regs.h b/include/sound/cs4231-regs.h index e8d1f3e31f9e..92647532c454 100644 --- a/include/sound/cs4231-regs.h +++ b/include/sound/cs4231-regs.h @@ -177,4 +177,12 @@ #define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */ #define CS4236_VERSION 0x9c /* chip version and ID */ +/* definitions for extended registers - OPTI93X */ +#define OPTi931_AUX_LEFT_INPUT 0x10 +#define OPTi931_AUX_RIGHT_INPUT 0x11 +#define OPTi93X_MIC_LEFT_INPUT 0x14 +#define OPTi93X_MIC_RIGHT_INPUT 0x15 +#define OPTi93X_OUT_LEFT 0x16 +#define OPTi93X_OUT_RIGHT 0x17 + #endif /* __SOUND_CS4231_REGS_H */ diff --git a/include/sound/cs4231.h b/include/sound/cs4231.h index 66055d702aa3..f0785f9f4ae4 100644 --- a/include/sound/cs4231.h +++ b/include/sound/cs4231.h @@ -58,6 +58,7 @@ /* compatible, but clones */ #define CS4231_HW_INTERWAVE 0x1000 /* InterWave chip */ #define CS4231_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ +#define CS4231_HW_OPTI93X 0x1102 /* Opti 930/931/933 */ /* defines for codec.hwshare */ #define CS4231_HWSHARE_IRQ (1<<0) @@ -120,6 +121,8 @@ unsigned char snd_cs4236_ext_in(struct snd_cs4231 *chip, unsigned char reg); void snd_cs4231_mce_up(struct snd_cs4231 *chip); void snd_cs4231_mce_down(struct snd_cs4231 *chip); +void snd_cs4231_overrange(struct snd_cs4231 *chip); + irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id); const char *snd_cs4231_chip_id(struct snd_cs4231 *chip); diff --git a/sound/isa/cs423x/cs4231_lib.c b/sound/isa/cs423x/cs4231_lib.c index 0aa8649e5c7f..521db705d179 100644 --- a/sound/isa/cs423x/cs4231_lib.c +++ b/sound/isa/cs423x/cs4231_lib.c @@ -119,6 +119,42 @@ static unsigned char snd_cs4231_original_image[32] = 0x00, /* 1f/31 - cbrl */ }; +static unsigned char snd_opti93x_original_image[32] = +{ + 0x00, /* 00/00 - l_mixout_outctrl */ + 0x00, /* 01/01 - r_mixout_outctrl */ + 0x88, /* 02/02 - l_cd_inctrl */ + 0x88, /* 03/03 - r_cd_inctrl */ + 0x88, /* 04/04 - l_a1/fm_inctrl */ + 0x88, /* 05/05 - r_a1/fm_inctrl */ + 0x80, /* 06/06 - l_dac_inctrl */ + 0x80, /* 07/07 - r_dac_inctrl */ + 0x00, /* 08/08 - ply_dataform_reg */ + 0x00, /* 09/09 - if_conf */ + 0x00, /* 0a/10 - pin_ctrl */ + 0x00, /* 0b/11 - err_init_reg */ + 0x0a, /* 0c/12 - id_reg */ + 0x00, /* 0d/13 - reserved */ + 0x00, /* 0e/14 - ply_upcount_reg */ + 0x00, /* 0f/15 - ply_lowcount_reg */ + 0x88, /* 10/16 - reserved/l_a1_inctrl */ + 0x88, /* 11/17 - reserved/r_a1_inctrl */ + 0x88, /* 12/18 - l_line_inctrl */ + 0x88, /* 13/19 - r_line_inctrl */ + 0x88, /* 14/20 - l_mic_inctrl */ + 0x88, /* 15/21 - r_mic_inctrl */ + 0x80, /* 16/22 - l_out_outctrl */ + 0x80, /* 17/23 - r_out_outctrl */ + 0x00, /* 18/24 - reserved */ + 0x00, /* 19/25 - reserved */ + 0x00, /* 1a/26 - reserved */ + 0x00, /* 1b/27 - reserved */ + 0x00, /* 1c/28 - cap_dataform_reg */ + 0x00, /* 1d/29 - reserved */ + 0x00, /* 1e/30 - cap_upcount_reg */ + 0x00 /* 1f/31 - cap_lowcount_reg */ +}; + /* * Basic I/O functions */ @@ -895,7 +931,7 @@ static int snd_cs4231_capture_prepare(struct snd_pcm_substream *substream) return 0; } -static void snd_cs4231_overrange(struct snd_cs4231 *chip) +void snd_cs4231_overrange(struct snd_cs4231 *chip) { unsigned long flags; unsigned char res; @@ -1054,8 +1090,11 @@ static int snd_cs4231_probe(struct snd_cs4231 *chip) chip->image[CS4231_IFACE_CTRL] = (chip->image[CS4231_IFACE_CTRL] & ~CS4231_SINGLE_DMA) | (chip->single_dma ? CS4231_SINGLE_DMA : 0); - chip->image[CS4231_ALT_FEATURE_1] = 0x80; - chip->image[CS4231_ALT_FEATURE_2] = chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; + if (chip->hardware != CS4231_HW_OPTI93X) { + chip->image[CS4231_ALT_FEATURE_1] = 0x80; + chip->image[CS4231_ALT_FEATURE_2] = + chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; + } ptr = (unsigned char *) &chip->image; snd_cs4231_mce_down(chip); spin_lock_irqsave(&chip->reg_lock, flags); @@ -1376,6 +1415,7 @@ const char *snd_cs4231_chip_id(struct snd_cs4231 *chip) case CS4231_HW_INTERWAVE: return "AMD InterWave"; case CS4231_HW_OPL3SA2: return chip->card->shortname; case CS4231_HW_AD1845: return "AD1845"; + case CS4231_HW_OPTI93X: return "OPTi 93x"; default: return "???"; } } @@ -1401,8 +1441,13 @@ static int snd_cs4231_new(struct snd_card *card, chip->rate_constraint = snd_cs4231_xrate; chip->set_playback_format = snd_cs4231_playback_format; chip->set_capture_format = snd_cs4231_capture_format; - memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); - + if (chip->hardware == CS4231_HW_OPTI93X) + memcpy(&chip->image, &snd_opti93x_original_image, + sizeof(snd_opti93x_original_image)); + else + memcpy(&chip->image, &snd_cs4231_original_image, + sizeof(snd_cs4231_original_image)); + *rchip = chip; return 0; } @@ -1790,6 +1835,48 @@ CS4231_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), CS4231_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1) }; +static struct snd_kcontrol_new snd_opti93x_controls[] = { +CS4231_DOUBLE("Master Playback Switch", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), +CS4231_DOUBLE("Master Playback Volume", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), +CS4231_DOUBLE("PCM Playback Switch", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), +CS4231_DOUBLE("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), +CS4231_DOUBLE("FM Playback Switch", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Line Playback Switch", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +CS4231_DOUBLE("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), +CS4231_DOUBLE("Mic Playback Switch", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Mic Playback Volume", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Mic Boost", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), +CS4231_DOUBLE("CD Playback Switch", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("CD Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Aux Playback Switch", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Capture Volume", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_cs4231_info_mux, + .get = snd_cs4231_get_mux, + .put = snd_cs4231_put_mux, +} +}; + int snd_cs4231_mixer(struct snd_cs4231 *chip) { struct snd_card *card; @@ -1802,10 +1889,22 @@ int snd_cs4231_mixer(struct snd_cs4231 *chip) strcpy(card->mixername, chip->pcm->name); - for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cs4231_controls[idx], chip))) < 0) - return err; - } + if (chip->hardware == CS4231_HW_OPTI93X) + for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_opti93x_controls[idx], + chip)); + if (err < 0) + return err; + } + else + for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_cs4231_controls[idx], + chip)); + if (err < 0) + return err; + } return 0; } @@ -1815,6 +1914,7 @@ EXPORT_SYMBOL(snd_cs4236_ext_out); EXPORT_SYMBOL(snd_cs4236_ext_in); EXPORT_SYMBOL(snd_cs4231_mce_up); EXPORT_SYMBOL(snd_cs4231_mce_down); +EXPORT_SYMBOL(snd_cs4231_overrange); EXPORT_SYMBOL(snd_cs4231_interrupt); EXPORT_SYMBOL(snd_cs4231_chip_id); EXPORT_SYMBOL(snd_cs4231_create); From 9f240a558f94bea839e14c46c8fccf93bdd35ef6 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 11 Jun 2008 12:26:32 +0200 Subject: [PATCH 085/179] ALSA: opti93x: use cs4231 lib This patch converts the Opti93x driver to use the cs4231 library instead of duplicating the code. Signed-off-by: Krzysztof Helt Tested-by: Rene Herman Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/isa/Kconfig | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 1110 +--------------------------- 2 files changed, 37 insertions(+), 1075 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index d649a0e966da..25347a25d63c 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -304,7 +304,7 @@ config SND_OPTI93X tristate "OPTi 82C93x" select SND_OPL3_LIB select SND_MPU401_UART - select SND_PCM + select SND_CS4231_LIB help Say Y here to include support for soundcards based on Opti 82C93x chips. diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index fc15b88364b4..41c047e665ec 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -33,15 +33,10 @@ #include #include #include -#ifdef CS4231 +#if defined(CS4231) || defined(OPTi93X) #include #else -#ifndef OPTi93X #include -#else -#include -#include -#endif /* OPTi93X */ #endif /* CS4231 */ #include #include @@ -109,7 +104,6 @@ module_param(dma2, int, 0444); MODULE_PARM_DESC(dma2, "2nd dma # for opti9xx driver."); #endif /* CS4231 || OPTi93X */ -#define OPTi9XX_HW_DETECT 0 #define OPTi9XX_HW_82C928 1 #define OPTi9XX_HW_82C929 2 #define OPTi9XX_HW_82C924 3 @@ -123,105 +117,12 @@ MODULE_PARM_DESC(dma2, "2nd dma # for opti9xx driver."); #ifdef OPTi93X -#define OPTi93X_INDEX 0x00 -#define OPTi93X_DATA 0x01 #define OPTi93X_STATUS 0x02 -#define OPTi93X_DDATA 0x03 #define OPTi93X_PORT(chip, r) ((chip)->port + OPTi93X_##r) -#define OPTi93X_MIXOUT_LEFT 0x00 -#define OPTi93X_MIXOUT_RIGHT 0x01 -#define OPTi93X_CD_LEFT_INPUT 0x02 -#define OPTi93X_CD_RIGHT_INPUT 0x03 -#define OPTi930_AUX_LEFT_INPUT 0x04 -#define OPTi930_AUX_RIGHT_INPUT 0x05 -#define OPTi931_FM_LEFT_INPUT 0x04 -#define OPTi931_FM_RIGHT_INPUT 0x05 -#define OPTi93X_DAC_LEFT 0x06 -#define OPTi93X_DAC_RIGHT 0x07 -#define OPTi93X_PLAY_FORMAT 0x08 -#define OPTi93X_IFACE_CONF 0x09 -#define OPTi93X_PIN_CTRL 0x0a -#define OPTi93X_ERR_INIT 0x0b -#define OPTi93X_ID 0x0c -#define OPTi93X_PLAY_UPR_CNT 0x0e -#define OPTi93X_PLAY_LWR_CNT 0x0f -#define OPTi931_AUX_LEFT_INPUT 0x10 -#define OPTi931_AUX_RIGHT_INPUT 0x11 -#define OPTi93X_LINE_LEFT_INPUT 0x12 -#define OPTi93X_LINE_RIGHT_INPUT 0x13 -#define OPTi93X_MIC_LEFT_INPUT 0x14 -#define OPTi93X_MIC_RIGHT_INPUT 0x15 -#define OPTi93X_OUT_LEFT 0x16 -#define OPTi93X_OUT_RIGHT 0x17 -#define OPTi93X_CAPT_FORMAT 0x1c -#define OPTi93X_CAPT_UPR_CNT 0x1e -#define OPTi93X_CAPT_LWR_CNT 0x1f - -#define OPTi93X_TRD 0x20 -#define OPTi93X_MCE 0x40 -#define OPTi93X_INIT 0x80 - -#define OPTi93X_MIXOUT_MIC_GAIN 0x20 -#define OPTi93X_MIXOUT_LINE 0x00 -#define OPTi93X_MIXOUT_CD 0x40 -#define OPTi93X_MIXOUT_MIC 0x80 -#define OPTi93X_MIXOUT_MIXER 0xc0 - -#define OPTi93X_STEREO 0x10 -#define OPTi93X_LINEAR_8 0x00 -#define OPTi93X_ULAW_8 0x20 -#define OPTi93X_LINEAR_16_LIT 0x40 -#define OPTi93X_ALAW_8 0x60 -#define OPTi93X_ADPCM_16 0xa0 -#define OPTi93X_LINEAR_16_BIG 0xc0 - -#define OPTi93X_CAPTURE_PIO 0x80 -#define OPTi93X_PLAYBACK_PIO 0x40 -#define OPTi93X_AUTOCALIB 0x08 -#define OPTi93X_SINGLE_DMA 0x04 -#define OPTi93X_CAPTURE_ENABLE 0x02 -#define OPTi93X_PLAYBACK_ENABLE 0x01 - -#define OPTi93X_IRQ_ENABLE 0x02 - -#define OPTi93X_DMA_REQUEST 0x10 -#define OPTi93X_CALIB_IN_PROGRESS 0x20 - #define OPTi93X_IRQ_PLAYBACK 0x04 #define OPTi93X_IRQ_CAPTURE 0x08 - -struct snd_opti93x { - unsigned long port; - struct resource *res_port; - int irq; - int dma1; - int dma2; - - struct snd_opti9xx *chip; - unsigned short hardware; - unsigned char image[32]; - - unsigned char mce_bit; - unsigned short mode; - int mute; - - spinlock_t lock; - - struct snd_card *card; - struct snd_pcm *pcm; - struct snd_pcm_substream *playback_substream; - struct snd_pcm_substream *capture_substream; - unsigned int p_dma_size; - unsigned int c_dma_size; -}; - -#define OPTi93X_MODE_NONE 0x00 -#define OPTi93X_MODE_PLAY 0x01 -#define OPTi93X_MODE_CAPTURE 0x02 -#define OPTi93X_MODE_OPEN (OPTi93X_MODE_PLAY | OPTi93X_MODE_CAPTURE) - #endif /* OPTi93X */ struct snd_opti9xx { @@ -234,6 +135,7 @@ struct snd_opti9xx { unsigned long mc_base_size; #ifdef OPTi93X unsigned long mc_indir_index; + struct snd_cs4231 *codec; #endif /* OPTi93X */ unsigned long pwd_reg; @@ -654,979 +556,23 @@ __skip_mpu: #ifdef OPTi93X -static unsigned char snd_opti93x_default_image[32] = -{ - 0x00, /* 00/00 - l_mixout_outctrl */ - 0x00, /* 01/01 - r_mixout_outctrl */ - 0x88, /* 02/02 - l_cd_inctrl */ - 0x88, /* 03/03 - r_cd_inctrl */ - 0x88, /* 04/04 - l_a1/fm_inctrl */ - 0x88, /* 05/05 - r_a1/fm_inctrl */ - 0x80, /* 06/06 - l_dac_inctrl */ - 0x80, /* 07/07 - r_dac_inctrl */ - 0x00, /* 08/08 - ply_dataform_reg */ - 0x00, /* 09/09 - if_conf */ - 0x00, /* 0a/10 - pin_ctrl */ - 0x00, /* 0b/11 - err_init_reg */ - 0x0a, /* 0c/12 - id_reg */ - 0x00, /* 0d/13 - reserved */ - 0x00, /* 0e/14 - ply_upcount_reg */ - 0x00, /* 0f/15 - ply_lowcount_reg */ - 0x88, /* 10/16 - reserved/l_a1_inctrl */ - 0x88, /* 11/17 - reserved/r_a1_inctrl */ - 0x88, /* 12/18 - l_line_inctrl */ - 0x88, /* 13/19 - r_line_inctrl */ - 0x88, /* 14/20 - l_mic_inctrl */ - 0x88, /* 15/21 - r_mic_inctrl */ - 0x80, /* 16/22 - l_out_outctrl */ - 0x80, /* 17/23 - r_out_outctrl */ - 0x00, /* 18/24 - reserved */ - 0x00, /* 19/25 - reserved */ - 0x00, /* 1a/26 - reserved */ - 0x00, /* 1b/27 - reserved */ - 0x00, /* 1c/28 - cap_dataform_reg */ - 0x00, /* 1d/29 - reserved */ - 0x00, /* 1e/30 - cap_upcount_reg */ - 0x00 /* 1f/31 - cap_lowcount_reg */ -}; - - -static int snd_opti93x_busy_wait(struct snd_opti93x *chip) -{ - int timeout; - - for (timeout = 250; timeout-- > 0; udelay(10)) - if (!(inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_INIT)) - return 0; - - snd_printk("chip still busy.\n"); - return -EBUSY; -} - -static unsigned char snd_opti93x_in(struct snd_opti93x *chip, unsigned char reg) -{ - snd_opti93x_busy_wait(chip); - outb(chip->mce_bit | (reg & 0x1f), OPTi93X_PORT(chip, INDEX)); - return inb(OPTi93X_PORT(chip, DATA)); -} - -static void snd_opti93x_out(struct snd_opti93x *chip, unsigned char reg, - unsigned char value) -{ - snd_opti93x_busy_wait(chip); - outb(chip->mce_bit | (reg & 0x1f), OPTi93X_PORT(chip, INDEX)); - outb(value, OPTi93X_PORT(chip, DATA)); -} - -static void snd_opti93x_out_image(struct snd_opti93x *chip, unsigned char reg, - unsigned char value) -{ - snd_opti93x_out(chip, reg, chip->image[reg] = value); -} - -static void snd_opti93x_out_mask(struct snd_opti93x *chip, unsigned char reg, - unsigned char mask, unsigned char value) -{ - snd_opti93x_out_image(chip, reg, - (chip->image[reg] & ~mask) | (value & mask)); -} - - -static void snd_opti93x_mce_up(struct snd_opti93x *chip) -{ - snd_opti93x_busy_wait(chip); - - chip->mce_bit = OPTi93X_MCE; - if (!(inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_MCE)) - outb(chip->mce_bit, OPTi93X_PORT(chip, INDEX)); -} - -static void snd_opti93x_mce_down(struct snd_opti93x *chip) -{ - snd_opti93x_busy_wait(chip); - - chip->mce_bit = 0; - if (inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_MCE) - outb(chip->mce_bit, OPTi93X_PORT(chip, INDEX)); -} - -#define snd_opti93x_mute_reg(chip, reg, mute) \ - snd_opti93x_out(chip, reg, mute ? 0x80 : chip->image[reg]); - -static void snd_opti93x_mute(struct snd_opti93x *chip, int mute) -{ - mute = mute ? 1 : 0; - if (chip->mute == mute) - return; - - chip->mute = mute; - - snd_opti93x_mute_reg(chip, OPTi93X_CD_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_CD_RIGHT_INPUT, mute); - switch (chip->hardware) { - case OPTi9XX_HW_82C930: - snd_opti93x_mute_reg(chip, OPTi930_AUX_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi930_AUX_RIGHT_INPUT, mute); - break; - case OPTi9XX_HW_82C931: - case OPTi9XX_HW_82C933: - snd_opti93x_mute_reg(chip, OPTi931_FM_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_FM_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_AUX_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_AUX_RIGHT_INPUT, mute); - } - snd_opti93x_mute_reg(chip, OPTi93X_DAC_LEFT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_DAC_RIGHT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_LINE_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_LINE_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_MIC_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_MIC_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_OUT_LEFT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_OUT_RIGHT, mute); -} - - -static unsigned int snd_opti93x_get_count(unsigned char format, - unsigned int size) -{ - switch (format & 0xe0) { - case OPTi93X_LINEAR_16_LIT: - case OPTi93X_LINEAR_16_BIG: - size >>= 1; - break; - case OPTi93X_ADPCM_16: - return size >> 2; - } - return (format & OPTi93X_STEREO) ? (size >> 1) : size; -} - -static unsigned int rates[] = { 5512, 6615, 8000, 9600, 11025, 16000, - 18900, 22050, 27428, 32000, 33075, 37800, - 44100, 48000 }; -#define RATES ARRAY_SIZE(rates) - -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = RATES, - .list = rates, - .mask = 0, -}; - -static unsigned char bits[] = { 0x01, 0x0f, 0x00, 0x0e, 0x03, 0x02, - 0x05, 0x07, 0x04, 0x06, 0x0d, 0x09, - 0x0b, 0x0c}; - -static unsigned char snd_opti93x_get_freq(unsigned int rate) -{ - unsigned int i; - - for (i = 0; i < RATES; i++) { - if (rate == rates[i]) - return bits[i]; - } - snd_BUG(); - return bits[RATES-1]; -} - -static unsigned char snd_opti93x_get_format(struct snd_opti93x *chip, - unsigned int format, int channels) -{ - unsigned char retval = OPTi93X_LINEAR_8; - - switch (format) { - case SNDRV_PCM_FORMAT_MU_LAW: - retval = OPTi93X_ULAW_8; - break; - case SNDRV_PCM_FORMAT_A_LAW: - retval = OPTi93X_ALAW_8; - break; - case SNDRV_PCM_FORMAT_S16_LE: - retval = OPTi93X_LINEAR_16_LIT; - break; - case SNDRV_PCM_FORMAT_S16_BE: - retval = OPTi93X_LINEAR_16_BIG; - break; - case SNDRV_PCM_FORMAT_IMA_ADPCM: - retval = OPTi93X_ADPCM_16; - } - return (channels > 1) ? (retval | OPTi93X_STEREO) : retval; -} - - -static void snd_opti93x_playback_format(struct snd_opti93x *chip, unsigned char fmt) -{ - unsigned char mask; - - snd_opti93x_mute(chip, 1); - - snd_opti93x_mce_up(chip); - mask = (chip->mode & OPTi93X_MODE_CAPTURE) ? 0xf0 : 0xff; - snd_opti93x_out_mask(chip, OPTi93X_PLAY_FORMAT, mask, fmt); - snd_opti93x_mce_down(chip); - - snd_opti93x_mute(chip, 0); -} - -static void snd_opti93x_capture_format(struct snd_opti93x *chip, unsigned char fmt) -{ - snd_opti93x_mute(chip, 1); - - snd_opti93x_mce_up(chip); - if (!(chip->mode & OPTi93X_MODE_PLAY)) - snd_opti93x_out_mask(chip, OPTi93X_PLAY_FORMAT, 0x0f, fmt); - else - fmt = chip->image[OPTi93X_PLAY_FORMAT] & 0xf0; - snd_opti93x_out_image(chip, OPTi93X_CAPT_FORMAT, fmt); - snd_opti93x_mce_down(chip); - - snd_opti93x_mute(chip, 0); -} - - -static int snd_opti93x_open(struct snd_opti93x *chip, unsigned int mode) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - if (chip->mode & mode) { - spin_unlock_irqrestore(&chip->lock, flags); - return -EAGAIN; - } - - if (!(chip->mode & OPTi93X_MODE_OPEN)) { - outb(0x00, OPTi93X_PORT(chip, STATUS)); - snd_opti93x_out_mask(chip, OPTi93X_PIN_CTRL, - OPTi93X_IRQ_ENABLE, OPTi93X_IRQ_ENABLE); - chip->mode = mode; - } - else - chip->mode |= mode; - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static void snd_opti93x_close(struct snd_opti93x *chip, unsigned int mode) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - chip->mode &= ~mode; - if (chip->mode & OPTi93X_MODE_OPEN) { - spin_unlock_irqrestore(&chip->lock, flags); - return; - } - - snd_opti93x_mute(chip, 1); - - outb(0, OPTi93X_PORT(chip, STATUS)); - snd_opti93x_out_mask(chip, OPTi93X_PIN_CTRL, OPTi93X_IRQ_ENABLE, - ~OPTi93X_IRQ_ENABLE); - - snd_opti93x_mce_up(chip); - snd_opti93x_out_image(chip, OPTi93X_IFACE_CONF, 0x00); - snd_opti93x_mce_down(chip); - chip->mode = 0; - - snd_opti93x_mute(chip, 0); - spin_unlock_irqrestore(&chip->lock, flags); -} - -static int snd_opti93x_trigger(struct snd_pcm_substream *substream, - unsigned char what, int cmd) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_STOP: - { - unsigned int what = 0; - struct snd_pcm_substream *s; - snd_pcm_group_for_each_entry(s, substream) { - if (s == chip->playback_substream) { - what |= OPTi93X_PLAYBACK_ENABLE; - snd_pcm_trigger_done(s, substream); - } else if (s == chip->capture_substream) { - what |= OPTi93X_CAPTURE_ENABLE; - snd_pcm_trigger_done(s, substream); - } - } - spin_lock(&chip->lock); - if (cmd == SNDRV_PCM_TRIGGER_START) { - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, what, what); - if (what & OPTi93X_CAPTURE_ENABLE) - udelay(50); - } else - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, what, 0x00); - spin_unlock(&chip->lock); - break; - } - default: - return -EINVAL; - } - return 0; -} - -static int snd_opti93x_playback_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_opti93x_trigger(substream, - OPTi93X_PLAYBACK_ENABLE, cmd); -} - -static int snd_opti93x_capture_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_opti93x_trigger(substream, - OPTi93X_CAPTURE_ENABLE, cmd); -} - -static int snd_opti93x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); -} - - -static int snd_opti93x_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_lib_free_pages(substream); - return 0; -} - - -static int snd_opti93x_playback_prepare(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned char format; - unsigned int count = snd_pcm_lib_period_bytes(substream); - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - - spin_lock_irqsave(&chip->lock, flags); - - chip->p_dma_size = size; - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, - OPTi93X_PLAYBACK_ENABLE | OPTi93X_PLAYBACK_PIO, - ~(OPTi93X_PLAYBACK_ENABLE | OPTi93X_PLAYBACK_PIO)); - - snd_dma_program(chip->dma1, runtime->dma_addr, size, - DMA_MODE_WRITE | DMA_AUTOINIT); - - format = snd_opti93x_get_freq(runtime->rate); - format |= snd_opti93x_get_format(chip, runtime->format, - runtime->channels); - snd_opti93x_playback_format(chip, format); - format = chip->image[OPTi93X_PLAY_FORMAT]; - - count = snd_opti93x_get_count(format, count) - 1; - snd_opti93x_out_image(chip, OPTi93X_PLAY_LWR_CNT, count); - snd_opti93x_out_image(chip, OPTi93X_PLAY_UPR_CNT, count >> 8); - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static int snd_opti93x_capture_prepare(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned char format; - unsigned int count = snd_pcm_lib_period_bytes(substream); - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - - spin_lock_irqsave(&chip->lock, flags); - - chip->c_dma_size = size; - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, - OPTi93X_CAPTURE_ENABLE | OPTi93X_CAPTURE_PIO, 0); - - snd_dma_program(chip->dma2, runtime->dma_addr, size, - DMA_MODE_READ | DMA_AUTOINIT); - - format = snd_opti93x_get_freq(runtime->rate); - format |= snd_opti93x_get_format(chip, runtime->format, - runtime->channels); - snd_opti93x_capture_format(chip, format); - format = chip->image[OPTi93X_CAPT_FORMAT]; - - count = snd_opti93x_get_count(format, count) - 1; - snd_opti93x_out_image(chip, OPTi93X_CAPT_LWR_CNT, count); - snd_opti93x_out_image(chip, OPTi93X_CAPT_UPR_CNT, count >> 8); - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static snd_pcm_uframes_t snd_opti93x_playback_pointer(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[OPTi93X_IFACE_CONF] & OPTi93X_PLAYBACK_ENABLE)) - return 0; - - ptr = snd_dma_pointer(chip->dma1, chip->p_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - -static snd_pcm_uframes_t snd_opti93x_capture_pointer(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[OPTi93X_IFACE_CONF] & OPTi93X_CAPTURE_ENABLE)) - return 0; - - ptr = snd_dma_pointer(chip->dma2, chip->c_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - - -static void snd_opti93x_overrange(struct snd_opti93x *chip) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - if (snd_opti93x_in(chip, OPTi93X_ERR_INIT) & (0x08 | 0x02)) - chip->capture_substream->runtime->overrange++; - - spin_unlock_irqrestore(&chip->lock, flags); -} - static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) { - struct snd_opti93x *codec = dev_id; + struct snd_cs4231 *codec = dev_id; + struct snd_opti9xx *chip = codec->card->private_data; unsigned char status; - status = snd_opti9xx_read(codec->chip, OPTi9XX_MC_REG(11)); + status = snd_opti9xx_read(chip, OPTi9XX_MC_REG(11)); if ((status & OPTi93X_IRQ_PLAYBACK) && codec->playback_substream) snd_pcm_period_elapsed(codec->playback_substream); if ((status & OPTi93X_IRQ_CAPTURE) && codec->capture_substream) { - snd_opti93x_overrange(codec); + snd_cs4231_overrange(codec); snd_pcm_period_elapsed(codec->capture_substream); } outb(0x00, OPTi93X_PORT(codec, STATUS)); return IRQ_HANDLED; } - -static struct snd_pcm_hardware snd_opti93x_playback = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), - .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, - .rate_min = 5512, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware snd_opti93x_capture = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), - .rates = SNDRV_PCM_RATE_8000_48000, - .rate_min = 5512, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -static int snd_opti93x_playback_open(struct snd_pcm_substream *substream) -{ - int error; - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - if ((error = snd_opti93x_open(chip, OPTi93X_MODE_PLAY)) < 0) - return error; - snd_pcm_set_sync(substream); - chip->playback_substream = substream; - runtime->hw = snd_opti93x_playback; - snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - return error; -} - -static int snd_opti93x_capture_open(struct snd_pcm_substream *substream) -{ - int error; - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - if ((error = snd_opti93x_open(chip, OPTi93X_MODE_CAPTURE)) < 0) - return error; - runtime->hw = snd_opti93x_capture; - snd_pcm_set_sync(substream); - chip->capture_substream = substream; - snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - return error; -} - -static int snd_opti93x_playback_close(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - chip->playback_substream = NULL; - snd_opti93x_close(chip, OPTi93X_MODE_PLAY); - return 0; -} - -static int snd_opti93x_capture_close(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - chip->capture_substream = NULL; - snd_opti93x_close(chip, OPTi93X_MODE_CAPTURE); - return 0; -} - - -static void snd_opti93x_init(struct snd_opti93x *chip) -{ - unsigned long flags; - int i; - - spin_lock_irqsave(&chip->lock, flags); - snd_opti93x_mce_up(chip); - - for (i = 0; i < 32; i++) - snd_opti93x_out_image(chip, i, snd_opti93x_default_image[i]); - - snd_opti93x_mce_down(chip); - spin_unlock_irqrestore(&chip->lock, flags); -} - -static int snd_opti93x_probe(struct snd_opti93x *chip) -{ - unsigned long flags; - unsigned char val; - - spin_lock_irqsave(&chip->lock, flags); - val = snd_opti93x_in(chip, OPTi93X_ID) & 0x0f; - spin_unlock_irqrestore(&chip->lock, flags); - - return (val == 0x0a) ? 0 : -ENODEV; -} - -static int snd_opti93x_free(struct snd_opti93x *chip) -{ - release_and_free_resource(chip->res_port); - if (chip->dma1 >= 0) { - disable_dma(chip->dma1); - free_dma(chip->dma1); - } - if (chip->dma2 >= 0) { - disable_dma(chip->dma2); - free_dma(chip->dma2); - } - if (chip->irq >= 0) { - free_irq(chip->irq, chip); - } - kfree(chip); - return 0; -} - -static int snd_opti93x_dev_free(struct snd_device *device) -{ - struct snd_opti93x *chip = device->device_data; - return snd_opti93x_free(chip); -} - -static const char *snd_opti93x_chip_id(struct snd_opti93x *codec) -{ - switch (codec->hardware) { - case OPTi9XX_HW_82C930: return "82C930"; - case OPTi9XX_HW_82C931: return "82C931"; - case OPTi9XX_HW_82C933: return "82C933"; - default: return "???"; - } -} - -static int snd_opti93x_create(struct snd_card *card, struct snd_opti9xx *chip, - int dma1, int dma2, - struct snd_opti93x **rcodec) -{ - static struct snd_device_ops ops = { - .dev_free = snd_opti93x_dev_free, - }; - int error; - struct snd_opti93x *codec; - - *rcodec = NULL; - codec = kzalloc(sizeof(*codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - codec->irq = -1; - codec->dma1 = -1; - codec->dma2 = -1; - - if ((codec->res_port = request_region(chip->wss_base + 4, 4, "OPTI93x CODEC")) == NULL) { - snd_printk(KERN_ERR "opti9xx: can't grab port 0x%lx\n", chip->wss_base + 4); - snd_opti93x_free(codec); - return -EBUSY; - } - if (request_dma(dma1, "OPTI93x - 1")) { - snd_printk(KERN_ERR "opti9xx: can't grab DMA1 %d\n", dma1); - snd_opti93x_free(codec); - return -EBUSY; - } - codec->dma1 = chip->dma1; - if (request_dma(dma2, "OPTI93x - 2")) { - snd_printk(KERN_ERR "opti9xx: can't grab DMA2 %d\n", dma2); - snd_opti93x_free(codec); - return -EBUSY; - } - codec->dma2 = chip->dma2; - - if (request_irq(chip->irq, snd_opti93x_interrupt, IRQF_DISABLED, DEV_NAME" - WSS", codec)) { - snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); - snd_opti93x_free(codec); - return -EBUSY; - } - - codec->card = card; - codec->port = chip->wss_base + 4; - codec->irq = chip->irq; - - spin_lock_init(&codec->lock); - codec->hardware = chip->hardware; - codec->chip = chip; - - if ((error = snd_opti93x_probe(codec))) { - snd_opti93x_free(codec); - return error; - } - - snd_opti93x_init(codec); - - /* Register device */ - if ((error = snd_device_new(card, SNDRV_DEV_LOWLEVEL, codec, &ops)) < 0) { - snd_opti93x_free(codec); - return error; - } - - *rcodec = codec; - return 0; -} - -static struct snd_pcm_ops snd_opti93x_playback_ops = { - .open = snd_opti93x_playback_open, - .close = snd_opti93x_playback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_opti93x_hw_params, - .hw_free = snd_opti93x_hw_free, - .prepare = snd_opti93x_playback_prepare, - .trigger = snd_opti93x_playback_trigger, - .pointer = snd_opti93x_playback_pointer, -}; - -static struct snd_pcm_ops snd_opti93x_capture_ops = { - .open = snd_opti93x_capture_open, - .close = snd_opti93x_capture_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_opti93x_hw_params, - .hw_free = snd_opti93x_hw_free, - .prepare = snd_opti93x_capture_prepare, - .trigger = snd_opti93x_capture_trigger, - .pointer = snd_opti93x_capture_pointer, -}; - -static int snd_opti93x_pcm(struct snd_opti93x *codec, int device, struct snd_pcm **rpcm) -{ - int error; - struct snd_pcm *pcm; - - if ((error = snd_pcm_new(codec->card, "OPTi 82C93X", device, 1, 1, &pcm)) < 0) - return error; - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_opti93x_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_opti93x_capture_ops); - - pcm->private_data = codec; - pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; - - strcpy(pcm->name, snd_opti93x_chip_id(codec)); - - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_isa_data(), - 64*1024, codec->dma1 > 3 || codec->dma2 > 3 ? 128*1024 : 64*1024); - - codec->pcm = pcm; - if (rpcm) - *rpcm = pcm; - return 0; -} - -/* - * MIXER part - */ - -static int snd_opti93x_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - static char *texts[4] = { - "Line1", "Aux", "Mic", "Mix" - }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 2; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int snd_opti93x_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.enumerated.item[0] = (chip->image[OPTi93X_MIXOUT_LEFT] & OPTi93X_MIXOUT_MIXER) >> 6; - ucontrol->value.enumerated.item[1] = (chip->image[OPTi93X_MIXOUT_RIGHT] & OPTi93X_MIXOUT_MIXER) >> 6; - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static int snd_opti93x_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - unsigned short left, right; - int change; - - if (ucontrol->value.enumerated.item[0] > 3 || - ucontrol->value.enumerated.item[1] > 3) - return -EINVAL; - left = ucontrol->value.enumerated.item[0] << 6; - right = ucontrol->value.enumerated.item[1] << 6; - spin_lock_irqsave(&chip->lock, flags); - left = (chip->image[OPTi93X_MIXOUT_LEFT] & ~OPTi93X_MIXOUT_MIXER) | left; - right = (chip->image[OPTi93X_MIXOUT_RIGHT] & ~OPTi93X_MIXOUT_MIXER) | right; - change = left != chip->image[OPTi93X_MIXOUT_LEFT] || - right != chip->image[OPTi93X_MIXOUT_RIGHT]; - snd_opti93x_out_image(chip, OPTi93X_MIXOUT_LEFT, left); - snd_opti93x_out_image(chip, OPTi93X_MIXOUT_RIGHT, right); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -#if 0 - -#define OPTi93X_SINGLE(xname, xindex, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_opti93x_info_single, \ - .get = snd_opti93x_get_single, .put = snd_opti93x_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } - -static int snd_opti93x_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 16) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_opti93x_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.integer.value[0] = (chip->image[reg] >> shift) & mask; - spin_unlock_irqrestore(&chip->lock, flags); - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - return 0; -} - -static int snd_opti93x_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - int change; - unsigned short val; - - val = (ucontrol->value.integer.value[0] & mask); - if (invert) - val = mask - val; - val <<= shift; - spin_lock_irqsave(&chip->lock, flags); - val = (chip->image[reg] & ~(mask << shift)) | val; - change = val != chip->image[reg]; - snd_opti93x_out(chip, reg, val); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -#endif /* single */ - -#define OPTi93X_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_opti93x_info_double, \ - .get = snd_opti93x_get_double, .put = snd_opti93x_put_double, \ - .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } - -#define OPTi93X_DOUBLE_INVERT_INVERT(xctl) \ - do { xctl.private_value ^= 22; } while (0) -#define OPTi93X_DOUBLE_CHANGE_REGS(xctl, left_reg, right_reg) \ - do { xctl.private_value &= ~0x0000ffff; \ - xctl.private_value |= left_reg | (right_reg << 8); } while (0) - -static int snd_opti93x_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 24) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_opti93x_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.integer.value[0] = (chip->image[left_reg] >> shift_left) & mask; - ucontrol->value.integer.value[1] = (chip->image[right_reg] >> shift_right) & mask; - spin_unlock_irqrestore(&chip->lock, flags); - if (invert) { - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1]; - } - return 0; -} - -static int snd_opti93x_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - int change; - unsigned short val1, val2; - - val1 = ucontrol->value.integer.value[0] & mask; - val2 = ucontrol->value.integer.value[1] & mask; - if (invert) { - val1 = mask - val1; - val2 = mask - val2; - } - val1 <<= shift_left; - val2 <<= shift_right; - spin_lock_irqsave(&chip->lock, flags); - val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; - val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; - change = val1 != chip->image[left_reg] || val2 != chip->image[right_reg]; - snd_opti93x_out_image(chip, left_reg, val1); - snd_opti93x_out_image(chip, right_reg, val2); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -static struct snd_kcontrol_new snd_opti93x_controls[] __devinitdata = { -OPTi93X_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Master Playback Volume", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), -OPTi93X_DOUBLE("PCM Playback Switch", 0, OPTi93X_DAC_LEFT, OPTi93X_DAC_RIGHT, 7, 7, 1, 1), -OPTi93X_DOUBLE("PCM Playback Volume", 0, OPTi93X_DAC_LEFT, OPTi93X_DAC_RIGHT, 0, 0, 31, 1), -OPTi93X_DOUBLE("FM Playback Switch", 0, OPTi931_FM_LEFT_INPUT, OPTi931_FM_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("FM Playback Volume", 0, OPTi931_FM_LEFT_INPUT, OPTi931_FM_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Line Playback Switch", 0, OPTi93X_LINE_LEFT_INPUT, OPTi93X_LINE_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Line Playback Volume", 0, OPTi93X_LINE_LEFT_INPUT, OPTi93X_LINE_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Mic Playback Switch", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Mic Playback Volume", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Mic Boost", 0, OPTi93X_MIXOUT_LEFT, OPTi93X_MIXOUT_RIGHT, 5, 5, 1, 1), -OPTi93X_DOUBLE("CD Playback Switch", 0, OPTi93X_CD_LEFT_INPUT, OPTi93X_CD_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("CD Playback Volume", 0, OPTi93X_CD_LEFT_INPUT, OPTi93X_CD_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Aux Playback Switch", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Aux Playback Volume", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Capture Volume", 0, OPTi93X_MIXOUT_LEFT, OPTi93X_MIXOUT_RIGHT, 0, 0, 15, 0), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = snd_opti93x_info_mux, - .get = snd_opti93x_get_mux, - .put = snd_opti93x_put_mux, -} -}; - -static int __devinit snd_opti93x_mixer(struct snd_opti93x *chip) -{ - struct snd_card *card; - struct snd_kcontrol_new knew; - int err; - unsigned int idx; - - snd_assert(chip != NULL && chip->card != NULL, return -EINVAL); - - card = chip->card; - - strcpy(card->mixername, snd_opti93x_chip_id(chip)); - - for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { - knew = snd_opti93x_controls[idx]; - if (chip->hardware == OPTi9XX_HW_82C930) { - if (strstr(knew.name, "FM")) /* skip FM controls */ - continue; - else if (strcmp(knew.name, "Mic Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - else if (strstr(knew.name, "Aux")) - OPTi93X_DOUBLE_CHANGE_REGS(knew, OPTi930_AUX_LEFT_INPUT, OPTi930_AUX_RIGHT_INPUT); - else if (strcmp(knew.name, "PCM Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - else if (strcmp(knew.name, "Master Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - } - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_opti93x_controls[idx], chip))) < 0) - return err; - } - return 0; -} - #endif /* OPTi93X */ static int __devinit snd_card_opti9xx_detect(struct snd_card *card, @@ -1739,8 +685,16 @@ static void snd_card_opti9xx_free(struct snd_card *card) { struct snd_opti9xx *chip = card->private_data; - if (chip) + if (chip) { +#ifdef OPTi93X + struct snd_cs4231 *codec = chip->codec; + if (codec->irq > 0) { + disable_irq(codec->irq); + free_irq(codec->irq, codec); + } +#endif release_and_free_resource(chip->res_mc_base); + } } static int __devinit snd_opti9xx_probe(struct snd_card *card) @@ -1748,11 +702,11 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; int error; struct snd_opti9xx *chip = card->private_data; -#if defined(OPTi93X) - struct snd_opti93x *codec; -#elif defined(CS4231) +#if defined(CS4231) || defined(OPTi93X) struct snd_cs4231 *codec; +#ifdef CS4231 struct snd_timer *timer; +#endif #else struct snd_ad1848 *codec; #endif @@ -1784,26 +738,34 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) if ((error = snd_opti9xx_configure(chip))) return error; -#if defined(OPTi93X) - if ((error = snd_opti93x_create(card, chip, chip->dma1, chip->dma2, &codec))) - return error; - if ((error = snd_opti93x_pcm(codec, 0, &pcm)) < 0) - return error; - if ((error = snd_opti93x_mixer(codec)) < 0) - return error; -#elif defined(CS4231) +#if defined(CS4231) || defined(OPTi93X) if ((error = snd_cs4231_create(card, chip->wss_base + 4, -1, chip->irq, chip->dma1, chip->dma2, - CS4231_HW_DETECT, - 0, +#ifdef CS4231 + CS4231_HW_DETECT, 0, +#else /* OPTi93x */ + CS4231_HW_OPTI93X, CS4231_HWSHARE_IRQ, +#endif &codec)) < 0) return error; +#ifdef OPTi93X + chip->codec = codec; +#endif if ((error = snd_cs4231_pcm(codec, 0, &pcm)) < 0) return error; if ((error = snd_cs4231_mixer(codec)) < 0) return error; +#ifdef CS4231 if ((error = snd_cs4231_timer(codec, 0, &timer)) < 0) return error; +#else /* OPTI93X */ + error = request_irq(chip->irq, snd_opti93x_interrupt, + IRQF_DISABLED, DEV_NAME" - WSS", codec); + if (error < 0) { + snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); + return error; + } +#endif #else if ((error = snd_ad1848_create(card, chip->wss_base + 4, chip->irq, chip->dma1, From d751b233bb8568f1de1ccbe3824ca69090326251 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:06 +0100 Subject: [PATCH 086/179] ALSA: ASoC: Fix register cache sizes for Wolfson codecs The register cache size is used by the codec_reg sysfs file which works in terms of the register cache access functions rather than in terms of raw access to the cache so the size specified needs to be in terms of the number of elements. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 5acf43ab104e..77880537a3cd 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -534,7 +534,7 @@ static int wm8731_init(struct snd_soc_device *socdev) codec->set_bias_level = wm8731_set_bias_level; codec->dai = &wm8731_dai; codec->num_dai = 1; - codec->reg_cache_size = sizeof(wm8731_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8731_reg); codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 1f11ad24551a..1ae670a98c5c 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -798,7 +798,7 @@ static int wm8750_init(struct snd_soc_device *socdev) codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; - codec->reg_cache_size = sizeof(wm8750_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8750_reg); codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index c32e6326be6c..285c5eaefe02 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1557,7 +1557,7 @@ static int wm8753_init(struct snd_soc_device *socdev) codec->set_bias_level = wm8753_set_bias_level; codec->dai = wm8753_dai; codec->num_dai = 2; - codec->reg_cache_size = sizeof(wm8753_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8753_reg); codec->reg_cache = kmemdup(wm8753_reg, sizeof(wm8753_reg), GFP_KERNEL); if (codec->reg_cache == NULL) From 8ddd440768e04e0c158c2bfa6178928e1f1eef3e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:07 +0100 Subject: [PATCH 087/179] ALSA: ASoC: Fix register cache size for UDA1380 The register cache size is used by the codec_reg sysfs file which works in terms of the register cache access functions rather than in terms of raw access to the cache so the size specified needs to be in terms of the number of elements. Signed-off-by: Mark Brown Cc: Philipp Zabel Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/uda1380.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index cb50486201f1..6d5335b14d51 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -655,8 +655,8 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; - codec->reg_cache_size = sizeof(uda1380_reg); - codec->reg_cache_step = 2; + codec->reg_cache_size = ARRAY_SIZE(uda1380_reg); + codec->reg_cache_step = 1; uda1380_reset(codec); /* register pcms */ From ae2ff19180d7dcac438e399f874d4de230b7f326 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:08 +0100 Subject: [PATCH 088/179] ALSA: ASoC: Tweak tlv320aicx reg_cache_size MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit ASoC codec drivers frequently set the register cache size using sizeof() rather than ARRAY_SIZE(). For tlv320aicx either is correct since the registers are 8 bit but update to use ARRAY_SIZE() for clarity. Signed-off-by: Mark Brown Acked-by: Jarkko Nikula  Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index dc8a38d9e53a..528c26acdbe2 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1021,7 +1021,7 @@ static int aic3x_init(struct snd_soc_device *socdev) codec->set_bias_level = aic3x_set_bias_level; codec->dai = &aic3x_dai; codec->num_dai = 1; - codec->reg_cache_size = sizeof(aic3x_reg); + codec->reg_cache_size = ARRAY_SIZE(aic3x_reg); codec->reg_cache = kmemdup(aic3x_reg, sizeof(aic3x_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; From dbac7cb8a7fcae34ffda04e4bc30673c2ea841ae Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:09 +0100 Subject: [PATCH 089/179] ALSA: ASoC: Add missing includes Ensure that DAIs are prototyped in the codec drivers that define them. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/ac97.c | 1 + sound/soc/codecs/wm9712.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 7bf2081b46fb..300d007f2339 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -21,6 +21,7 @@ #include #include #include +#include "ac97.h" #define AC97_VERSION "0.6" diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index d9789f1c8903..28ac66f6b6b2 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -22,6 +22,7 @@ #include #include #include +#include "wm9712.h" #define WM9712_VERSION "0.4" From bdb92876f0a9d2b431199e385732ede89ff0b97d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:10 +0100 Subject: [PATCH 090/179] ALSA: ASoC: Pass the DAI being configured into CPU DAI probe and remove This allows per-DAI initialisation to be done by the CPU DAI drivers. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 6 ++++-- sound/soc/davinci/davinci-i2s.c | 6 ++++-- sound/soc/pxa/pxa2xx-ac97.c | 3 ++- sound/soc/s3c24xx/s3c2412-i2s.c | 3 ++- sound/soc/s3c24xx/s3c2443-ac97.c | 6 ++++-- sound/soc/s3c24xx/s3c24xx-i2s.c | 3 ++- sound/soc/soc-core.c | 6 +++--- 7 files changed, 21 insertions(+), 12 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9fa2093e74eb..56d2224c2c07 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -358,8 +358,10 @@ struct snd_soc_cpu_dai { unsigned char type; /* DAI callbacks */ - int (*probe)(struct platform_device *pdev); - void (*remove)(struct platform_device *pdev); + int (*probe)(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai); + void (*remove)(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai); int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); int (*resume)(struct platform_device *pdev, diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index c421774b33ee..c3b545ccff72 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -295,7 +295,8 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } -static int davinci_i2s_probe(struct platform_device *pdev) +static int davinci_i2s_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; @@ -356,7 +357,8 @@ err_release_region: return ret; } -static void davinci_i2s_remove(struct platform_device *pdev) +static void davinci_i2s_remove(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 97ec2d90547c..cb947956ed1a 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -310,7 +310,8 @@ static int pxa2xx_ac97_resume(struct platform_device *pdev, #define pxa2xx_ac97_resume NULL #endif -static int pxa2xx_ac97_probe(struct platform_device *pdev) +static int pxa2xx_ac97_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { int ret; diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index c4a46dd589b3..c463a82dec3a 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -601,7 +601,8 @@ struct clk *s3c2412_get_iisclk(void) EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); -static int s3c2412_i2s_probe(struct platform_device *pdev) +static int s3c2412_i2s_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { DBG("Entered %s\n", __func__); diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 0eed140dcd9b..533565b61b2f 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -209,7 +209,8 @@ static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { .dma_size = 4, }; -static int s3c2443_ac97_probe(struct platform_device *pdev) +static int s3c2443_ac97_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { int ret; u32 ac_glbctrl; @@ -260,7 +261,8 @@ static int s3c2443_ac97_probe(struct platform_device *pdev) return ret; } -static void s3c2443_ac97_remove(struct platform_device *pdev) +static void s3c2443_ac97_remove(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { free_irq(IRQ_S3C244x_AC97, NULL); clk_disable(s3c24xx_ac97.ac97_clk); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 4c52f7946d9e..42e96b5ff825 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -377,7 +377,8 @@ u32 s3c24xx_i2s_get_clockrate(void) } EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); -static int s3c24xx_i2s_probe(struct platform_device *pdev) +static int s3c24xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { DBG("Entered %s\n", __func__); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f594ab888e17..c96a6184d66e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -768,7 +768,7 @@ static int soc_probe(struct platform_device *pdev) for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->probe) { - ret = cpu_dai->probe(pdev); + ret = cpu_dai->probe(pdev, cpu_dai); if (ret < 0) goto cpu_dai_err; } @@ -798,7 +798,7 @@ cpu_dai_err: for (i--; i >= 0; i--) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) - cpu_dai->remove(pdev); + cpu_dai->remove(pdev, cpu_dai); } if (machine->remove) @@ -827,7 +827,7 @@ static int soc_remove(struct platform_device *pdev) for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) - cpu_dai->remove(pdev); + cpu_dai->remove(pdev, cpu_dai); } if (machine->remove) From d133b0ce6ceaf744184c7cb9f95ba92f333c246a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:11 +0100 Subject: [PATCH 091/179] ALSA: ASoC: Fix WM9713 voice PCM slave mode configuration Reported-by: Rodolfo Giometti Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm9713.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 4f516a5a5616..aba330118784 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -876,7 +876,7 @@ static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, gpio |= 0x0018; break; case SND_SOC_DAIFMT_CBS_CFS: - reg |= 0x0200; + reg |= 0x2000; gpio |= 0x001a; break; case SND_SOC_DAIFMT_CBS_CFM: From 2cc8c609798957b90adf90b5bfb9859d1643fade Mon Sep 17 00:00:00 2001 From: Mike Montour Date: Wed, 11 Jun 2008 13:47:12 +0100 Subject: [PATCH 092/179] ALSA: ASoC: Add TLV information to remaining WM8753 controls Signed-off-by: Mike Montour Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm8753.c | 63 +++++++++++++++++++++++++++------------ 1 file changed, 44 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 285c5eaefe02..00b481183d45 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -260,28 +260,50 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, return 1; } -static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600); +static const DECLARE_TLV_DB_SCALE(rec_mix_tlv, -1500, 300, 0); +static const DECLARE_TLV_DB_SCALE(mic_preamp_tlv, 1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +static const unsigned int out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + /* 0000000 - 0101111 = "Analogue mute" */ + 0, 48, TLV_DB_SCALE_ITEM(-25500, 0, 0), + 48, 127, TLV_DB_SCALE_ITEM(-7300, 100, 0), +}; +static const DECLARE_TLV_DB_SCALE(mix_tlv, -1500, 300, 0); +static const DECLARE_TLV_DB_SCALE(voice_mix_tlv, -1200, 300, 0); +static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0); static const struct snd_kcontrol_new wm8753_snd_controls[] = { -SOC_DOUBLE_R("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0), +SOC_DOUBLE_R_TLV("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0, dac_tlv), -SOC_DOUBLE_R("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0), +SOC_DOUBLE_R_TLV("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0, + adc_tlv), -SOC_DOUBLE_R("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, 0, 127, 0), -SOC_DOUBLE_R("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, 127, 0), +SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, + 0, 127, 0, out_tlv), +SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, + 127, 0, out_tlv), -SOC_SINGLE("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0), +SOC_SINGLE_TLV("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0, out_tlv), -SOC_DOUBLE_R("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, 1), -SOC_DOUBLE_R("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, 7, 1), -SOC_DOUBLE_R("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, 1), +SOC_DOUBLE_R_TLV("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, + 1, mix_tlv), +SOC_DOUBLE_R_TLV("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, + 7, 1, mix_tlv), +SOC_DOUBLE_R_TLV("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, + 1, voice_mix_tlv), -SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, 1, 0), -SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0), +SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, + 1, 0), +SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, + 1, 0), -SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1), -SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1), -SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1), +SOC_SINGLE_TLV("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1, mix_tlv), +SOC_SINGLE_TLV("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1, + mix_tlv), +SOC_SINGLE_TLV("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1, + voice_mix_tlv), SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0), SOC_ENUM("Bass Boost", wm8753_enum[0]), @@ -291,10 +313,13 @@ SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 15, 1), SOC_SINGLE("Treble Volume", WM8753_TREBLE, 0, 15, 1), SOC_ENUM("Treble Cut-off", wm8753_enum[2]), -SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1, rec_mix_tlv), -SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1, rec_mix_tlv), +SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1, + rec_mix_tlv), +SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1, + rec_mix_tlv), -SOC_DOUBLE_R("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0), +SOC_DOUBLE_R_TLV("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0, + pga_tlv), SOC_DOUBLE_R("Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0), SOC_DOUBLE_R("Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 1), @@ -326,8 +351,8 @@ SOC_ENUM("De-emphasis", wm8753_enum[8]), SOC_ENUM("Playback Mono Mix", wm8753_enum[9]), SOC_ENUM("Playback Phase", wm8753_enum[10]), -SOC_SINGLE("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0), -SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0), +SOC_SINGLE_TLV("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0, mic_preamp_tlv), +SOC_SINGLE_TLV("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0, mic_preamp_tlv), SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai), From 10144c09a0d6a62e1d56e25f142743c7a00e5dba Mon Sep 17 00:00:00 2001 From: Mike Montour Date: Wed, 11 Jun 2008 13:47:13 +0100 Subject: [PATCH 093/179] ALSA: ASoC: Add SOC_SINGLE_EXT_TLV control type Signed-off-by: Mike Montour Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 56d2224c2c07..1f5c62181002 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -100,6 +100,15 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) } +#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmask, xinvert,\ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ From a2e31a595ea74120a5e9de91fce56dd835edff94 Mon Sep 17 00:00:00 2001 From: Mike Montour Date: Wed, 11 Jun 2008 13:47:14 +0100 Subject: [PATCH 094/179] ALSA: ASoC: Add TLV information to the LM4857 controls on the GTA01 Signed-off-by: Mike Montour Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index c1a0161bc72e..34851238dea0 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include @@ -474,13 +475,16 @@ static const struct soc_enum neo_scenario_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), }; +static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); +static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); + static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { - SOC_SINGLE_EXT("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, mono_tlv), SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], lm4857_get_mode, lm4857_set_mode), SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], From 6e132fa668d48a5a6fae6c24f6104e44b1ec1049 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 11 Jun 2008 13:47:15 +0100 Subject: [PATCH 095/179] ALSA: ASoC: Cover also Nokia N810 WiMAX Edition in N810 machine driver Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/omap/n810.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c32487b6c7b0..c168a64e3162 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -265,7 +265,7 @@ static int __init n810_soc_init(void) int err; struct device *dev; - if (!machine_is_nokia_n810()) + if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) return -ENODEV; n810_snd_device = platform_device_alloc("soc-audio", -1); From 4549915cdf468019d2fd86485dd8d2ab426b8d85 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Jun 2008 16:27:24 +0200 Subject: [PATCH 096/179] ALSA: hda - Fix stac9205_cfg_tbl Sort stac9205_cfg_table in the order of id numbers, and removed the duplicated (obsoleted) entries for 0x01fc and 0x01fd. This doesn't change the driver behavior since the old entries are all secondary. The duplication occured due to commit dfe495d0, and the old entries were introduced by commit ae0a8ed8. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a4f44a00bae8..1ef3dd6e8a64 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1754,12 +1754,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { "unknown Dell", STAC_9205_DELL_M42), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f8, "Dell Precision", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021c, - "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f9, "Dell Precision", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021b, - "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fa, "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fc, @@ -1770,18 +1766,14 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ff, "Dell Precision M4300", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0206, - "Dell Precision", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1, - "Dell Inspiron", STAC_9205_DELL_M44), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f2, - "Dell Inspiron", STAC_9205_DELL_M44), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fc, - "Dell Inspiron", STAC_9205_DELL_M44), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fd, - "Dell Inspiron", STAC_9205_DELL_M44), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0204, "unknown Dell", STAC_9205_DELL_M42), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0206, + "Dell Precision", STAC_9205_DELL_M43), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021b, + "Dell Precision", STAC_9205_DELL_M43), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021c, + "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021f, "Dell Inspiron", STAC_9205_DELL_M44), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, From 149b154452b18dedd4559cdd65b965ae9011e425 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Jun 2008 17:43:36 +0200 Subject: [PATCH 097/179] ALSA: hda - Remove unused mutex Removed unused mutex from patch_*.c. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 5 ----- sound/pci/hda/patch_conexant.c | 4 ---- 2 files changed, 9 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index a99e86d74278..55ce897ef0b3 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -23,7 +23,6 @@ #include #include #include -#include #include #include "hda_codec.h" @@ -64,7 +63,6 @@ struct ad198x_spec { /* PCM information */ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ - struct mutex amp_mutex; /* PCM volume/mute control mutex */ unsigned int spdif_route; /* dynamic controls, init_verbs and input_mux */ @@ -3177,7 +3175,6 @@ static int patch_ad1884(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -3847,7 +3844,6 @@ static int patch_ad1884a(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -4152,7 +4148,6 @@ static int patch_ad1882(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 6; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 36fd85260035..e3df8ac37076 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -82,7 +82,6 @@ struct conexant_spec { /* PCM information */ struct hda_pcm pcm_rec[2]; /* used in build_pcms() */ - struct mutex amp_mutex; /* PCM volume/mute control mutex */ unsigned int spdif_route; /* dynamic controls, init_verbs and input_mux */ @@ -928,7 +927,6 @@ static int patch_cxt5045(struct hda_codec *codec) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -1477,7 +1475,6 @@ static int patch_cxt5047(struct hda_codec *codec) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -1736,7 +1733,6 @@ static int patch_cxt5051(struct hda_codec *codec) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; codec->patch_ops = conexant_patch_ops; From d3bd67cdbb5d12f30dfd1b627fd95bd6dcc7ff1e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Jun 2008 18:17:26 +0200 Subject: [PATCH 098/179] ALSA: make snd_ctl_elem_read() and snd_ctl_elem_write() static snd_ctl_elem_read() and snd_ctl_elem_write() are no longer used by any other drivers. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/control.h | 3 --- sound/core/control.c | 7 ++++--- 2 files changed, 4 insertions(+), 6 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 3dc1291f52db..4721b4bba053 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -129,9 +129,6 @@ int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn); #define snd_ctl_unregister_ioctl_compat(fcn) #endif -int snd_ctl_elem_read(struct snd_card *card, struct snd_ctl_elem_value *control); -int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, struct snd_ctl_elem_value *control); - static inline unsigned int snd_ctl_get_ioffnum(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id) { return id->numid - kctl->id.numid; diff --git a/sound/core/control.c b/sound/core/control.c index 01a1a5af47bb..281b2e2ef0ea 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -684,7 +684,8 @@ static int snd_ctl_elem_info_user(struct snd_ctl_file *ctl, return result; } -int snd_ctl_elem_read(struct snd_card *card, struct snd_ctl_elem_value *control) +static int snd_ctl_elem_read(struct snd_card *card, + struct snd_ctl_elem_value *control) { struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; @@ -734,8 +735,8 @@ static int snd_ctl_elem_read_user(struct snd_card *card, return result; } -int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, - struct snd_ctl_elem_value *control) +static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, + struct snd_ctl_elem_value *control) { struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; From aafc4412be31306e5c38bd4e4581066507503adc Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Fri, 13 Jun 2008 18:04:33 +0200 Subject: [PATCH 099/179] ALSA: hda: Add support for 92HD73xxx codecs Added support for new family of IDT codecs. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 47 +++++++++++++++++++++++++--------- 1 file changed, 35 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1ef3dd6e8a64..c4f3489376c5 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -636,21 +636,28 @@ static struct hda_verb stac92hd71bxx_core_init[] = { { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, }; +#define HD_DISABLE_PORTF 3 static struct hda_verb stac92hd71bxx_analog_core_init[] = { + /* start of config #1 */ + + /* connect port 0f to audio mixer */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ + /* unmute right and left channels for node 0x0f */ + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* start of config #2 */ + /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* connect ports 0d and 0f to audio mixer */ + /* connect port 0d to audio mixer */ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute dac0 input in audio mixer */ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ + /* unmute right and left channels for nodes 0x0a, 0xd */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; @@ -1317,13 +1324,13 @@ static unsigned int ref92hd71bxx_pin_configs[10] = { 0x90a000f0, 0x01452050, }; -static unsigned int dell_m4_1_pin_configs[13] = { +static unsigned int dell_m4_1_pin_configs[10] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, 0x40f000f0, 0x4f0000f0, }; -static unsigned int dell_m4_2_pin_configs[13] = { +static unsigned int dell_m4_2_pin_configs[10] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x044413b0, @@ -3095,13 +3102,16 @@ static int stac92xx_init(struct hda_codec *codec) 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); int def_conf = snd_hda_codec_read(codec, spec->pwr_nids[i], 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = get_defcfg_connect(def_conf); /* outputs are only ports capable of power management * any attempts on powering down a input port cause the * referenced VREF to act quirky. */ if (pinctl & AC_PINCTL_IN_EN) continue; - if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) + /* skip any ports that don't have jacks since presence + * detection is useless */ + if (def_conf && def_conf != AC_JACK_PORT_FIXED) continue; enable_pin_detect(codec, spec->pwr_nids[i], event | i); codec->patch_ops.unsol_event(codec, (event | i) << 26); @@ -3606,6 +3616,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); + spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, @@ -3634,6 +3645,19 @@ again: spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; break; + case 0x111d7608: /* 5 Port with Analog Mixer */ + /* no output amps */ + spec->num_pwrs = 0; + spec->mixer = stac92hd71bxx_analog_mixer; + + /* disable VSW */ + spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; + stac92xx_set_config_reg(codec, 0xf, 0x40f000f0); + break; + case 0x111d7603: /* 6 Port with Analog Mixer */ + /* no output amps */ + spec->num_pwrs = 0; + /* fallthru */ default: spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; @@ -3652,15 +3676,13 @@ again: spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids; spec->dmux_nids = stac92hd71bxx_dmux_nids; + spec->pwr_nids = stac92hd71bxx_pwr_nids; spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); spec->num_dmics = STAC92HD71BXX_NUM_DMICS; spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); - spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); - spec->pwr_nids = stac92hd71bxx_pwr_nids; - spec->multiout.num_dacs = 1; spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids; @@ -4298,10 +4320,11 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, + { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d7608, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx }, From abb68c26ba15f8e84e580a40c0b1bc349cb534b0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 13 Jun 2008 16:24:04 +0100 Subject: [PATCH 100/179] ALSA: ASoC: Check for exact register match in wm97xx_reset() To provide added robustness in case an AC97 controller reads back all zeros in error cases check for an exact match when testing to see if resets have brought the codec back. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm9712.c | 4 ++-- sound/soc/codecs/wm9713.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 28ac66f6b6b2..47390113bd0e 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -589,12 +589,12 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == wm9712_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index aba330118784..0db963747369 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1076,12 +1076,12 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == wm9713_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != wm9713_reg[0]) return -EIO; return 0; } From 6ed2597883b1b03ca94f62f0cfe908314cba6d6b Mon Sep 17 00:00:00 2001 From: Andy Green Date: Fri, 13 Jun 2008 16:24:05 +0100 Subject: [PATCH 101/179] ALSA: ASoC: Don't block system resume On OpenMoko soc-audio resume is taking 700ms of the whole resume time of 1.3s, dominated by writes to the codec over I2C. This patch shunts the resume guts into a workqueue which then is done asynchronously. The "card" is locked using the ALSA power state APIs as suggested by Mark Brown. [Added fix for race with resume to suspend and fixed a couple of nits from checkpatch -- broonie.] Signed-off-by: Andy Green Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 46 +++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 44 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1f5c62181002..340223a8f24c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -510,6 +510,7 @@ struct snd_soc_device { struct snd_soc_codec *codec; struct snd_soc_codec_device *codec_dev; struct delayed_work delayed_work; + struct work_struct deferred_resume_work; void *codec_data; }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c96a6184d66e..b931039632c5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -639,6 +639,16 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_codec *codec = socdev->codec; int i; + /* Due to the resume being scheduled into a workqueue we could + * suspend before that's finished - wait for it to complete. + */ + snd_power_lock(codec->card); + snd_power_wait(codec->card, SNDRV_CTL_POWER_D0); + snd_power_unlock(codec->card); + + /* we're going to block userspace touching us until resume completes */ + snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot); + /* mute any active DAC's */ for (i = 0; i < machine->num_links; i++) { struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; @@ -691,16 +701,27 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) return 0; } -/* powers up audio subsystem after a suspend */ -static int soc_resume(struct platform_device *pdev) +/* deferred resume work, so resume can complete before we finished + * setting our codec back up, which can be very slow on I2C + */ +static void soc_resume_deferred(struct work_struct *work) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_device *socdev = container_of(work, + struct snd_soc_device, + deferred_resume_work); struct snd_soc_machine *machine = socdev->machine; struct snd_soc_platform *platform = socdev->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; struct snd_soc_codec *codec = socdev->codec; + struct platform_device *pdev = to_platform_device(socdev->dev); int i; + /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, + * so userspace apps are blocked from touching us + */ + + dev_info(socdev->dev, "starting resume work\n"); + if (machine->resume_pre) machine->resume_pre(pdev); @@ -742,6 +763,22 @@ static int soc_resume(struct platform_device *pdev) if (machine->resume_post) machine->resume_post(pdev); + dev_info(socdev->dev, "resume work completed\n"); + + /* userspace can access us now we are back as we were before */ + snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0); +} + +/* powers up audio subsystem after a suspend */ +static int soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + dev_info(socdev->dev, "scheduling resume work\n"); + + if (!schedule_work(&socdev->deferred_resume_work)) + dev_err(socdev->dev, "work item may be lost\n"); + return 0; } @@ -788,6 +825,9 @@ static int soc_probe(struct platform_device *pdev) /* DAPM stream work */ INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work); + /* deferred resume work */ + INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred); + return 0; platform_err: From 0c6341ace5efb14a0da08482e299b3c158e0d5fb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Jun 2008 20:50:27 +0200 Subject: [PATCH 102/179] ALSA: hda - bdl_pos_adj=32 as default Use bdl_pos_adj=32 as default except for Intel hardwares confirmed to work with bdl_pos_adj=1. Looks like ATI and NVidia require this higher value. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ec4ae2d94000..539d20102c32 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2024,12 +2024,11 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, if (bdl_pos_adj[dev] < 0) { switch (chip->driver_type) { - case AZX_DRIVER_ATI: - case AZX_DRIVER_ATIHDMI: - bdl_pos_adj[dev] = 32; + case AZX_DRIVER_ICH: + bdl_pos_adj[dev] = 1; break; default: - bdl_pos_adj[dev] = 1; + bdl_pos_adj[dev] = 32; break; } } From 766979e09d302315f314bfd96fdd83f8f9896d9c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Jun 2008 20:53:56 +0200 Subject: [PATCH 103/179] ALSA: hda - use upper_32_bits() Use the standard upper_32_bits() instead of own macro. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 539d20102c32..16715a68ba5e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -440,11 +440,6 @@ static char *driver_short_names[] __devinitdata = { /* for pcm support */ #define get_azx_dev(substream) (substream->runtime->private_data) -/* Get the upper 32bit of the given dma_addr_t - * Compiler should optimize and eliminate the code if dma_addr_t is 32bit - */ -#define upper_32bit(addr) (sizeof(addr) > 4 ? (u32)((addr) >> 32) : (u32)0) - static int azx_acquire_irq(struct azx *chip, int do_disconnect); /* @@ -475,7 +470,7 @@ static void azx_init_cmd_io(struct azx *chip) chip->corb.addr = chip->rb.addr; chip->corb.buf = (u32 *)chip->rb.area; azx_writel(chip, CORBLBASE, (u32)chip->corb.addr); - azx_writel(chip, CORBUBASE, upper_32bit(chip->corb.addr)); + azx_writel(chip, CORBUBASE, upper_32_bits(chip->corb.addr)); /* set the corb size to 256 entries (ULI requires explicitly) */ azx_writeb(chip, CORBSIZE, 0x02); @@ -490,7 +485,7 @@ static void azx_init_cmd_io(struct azx *chip) chip->rirb.addr = chip->rb.addr + 2048; chip->rirb.buf = (u32 *)(chip->rb.area + 2048); azx_writel(chip, RIRBLBASE, (u32)chip->rirb.addr); - azx_writel(chip, RIRBUBASE, upper_32bit(chip->rirb.addr)); + azx_writel(chip, RIRBUBASE, upper_32_bits(chip->rirb.addr)); /* set the rirb size to 256 entries (ULI requires explicitly) */ azx_writeb(chip, RIRBSIZE, 0x02); @@ -861,7 +856,7 @@ static void azx_init_chip(struct azx *chip) /* program the position buffer */ azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr); - azx_writel(chip, DPUBASE, upper_32bit(chip->posbuf.addr)); + azx_writel(chip, DPUBASE, upper_32_bits(chip->posbuf.addr)); chip->initialized = 1; } @@ -1006,7 +1001,7 @@ static int setup_bdle(struct snd_pcm_substream *substream, addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs); /* program the address field of the BDL entry */ bdl[0] = cpu_to_le32((u32)addr); - bdl[1] = cpu_to_le32(upper_32bit(addr)); + bdl[1] = cpu_to_le32(upper_32_bits(addr)); /* program the size field of the BDL entry */ chunk = PAGE_SIZE - (ofs % PAGE_SIZE); if (size < chunk) @@ -1138,7 +1133,7 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) /* lower BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl.addr); /* upper BDL address */ - azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl.addr)); + azx_sd_writel(azx_dev, SD_BDLPU, upper_32_bits(azx_dev->bdl.addr)); /* enable the position buffer */ if (chip->position_fix == POS_FIX_POSBUF || From b7d2399242b639f0b0be5f901506a7180d961b64 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 16 Jun 2008 09:42:25 +0200 Subject: [PATCH 104/179] ALSA: Release 1.0.17rc2 Signed-off-by: Jaroslav Kysela --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/version.h b/include/sound/version.h index 2474b8ee99f9..1802cc593333 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.17rc1" +#define CONFIG_SND_VERSION "1.0.17rc2" #define CONFIG_SND_DATE "" From 6719292af5513da52aa876d0e7ac4367a0851845 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 16 Jun 2008 10:39:34 +0200 Subject: [PATCH 105/179] ALSA: Remove duplicate MODULE_AUTHOR/DESCRIPTION/LICENCE from snd-ens1370.ko But comment only extra code in ak4531_codec.c for history. Signed-off-by: Jaroslav Kysela --- sound/pci/ak4531_codec.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/ak4531_codec.c b/sound/pci/ak4531_codec.c index 6a99eed2d8b6..33d37b1c42fc 100644 --- a/sound/pci/ak4531_codec.c +++ b/sound/pci/ak4531_codec.c @@ -28,9 +28,11 @@ #include #include +/* MODULE_AUTHOR("Jaroslav Kysela "); MODULE_DESCRIPTION("Universal routines for AK4531 codec"); MODULE_LICENSE("GPL"); +*/ #ifdef CONFIG_PROC_FS static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531); From 9f515b6898d0f2c1b9eb34ae9986794045f34e2b Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 17 Jun 2008 16:20:13 +0200 Subject: [PATCH 106/179] ALSA: emu10k1 - fix possible memory leak in memory allocation routines The leak was introduced in "[ALSA] emu10k1 - simplify page allocation for synth" commit. Signed-off-by: Jaroslav Kysela --- sound/pci/emu10k1/memory.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 42943b4fcb7b..759e29f89478 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -464,11 +464,17 @@ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk /* first try to allocate from <4GB zone */ struct page *p = alloc_page(GFP_KERNEL | GFP_DMA32 | __GFP_NOWARN); - if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) + if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) { /* try to allocate from <16MB zone */ - p = alloc_page(GFP_ATOMIC | GFP_DMA | + struct page *p1 = + alloc_page(GFP_ATOMIC | GFP_DMA | __GFP_NORETRY | /* no OOM-killer */ __GFP_NOWARN); + /* free page outside dma_mask range */ + if (p) + free_page((unsigned long)page_address(p)); + p = p1; + } if (!p) { __synth_free_pages(emu, first_page, page - 1); return -ENOMEM; From 284373059f9605442cac6453780f6aaecf9abac1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Jun 2008 16:30:27 +0200 Subject: [PATCH 107/179] ALSA: emu10k1 - simplify the last fix Clean up the previous commit for fixing memory leaks. Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/memory.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 759e29f89478..7d379f5131fb 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -465,15 +465,12 @@ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk struct page *p = alloc_page(GFP_KERNEL | GFP_DMA32 | __GFP_NOWARN); if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) { + if (p) + __free_page(p); /* try to allocate from <16MB zone */ - struct page *p1 = - alloc_page(GFP_ATOMIC | GFP_DMA | + p = alloc_page(GFP_ATOMIC | GFP_DMA | __GFP_NORETRY | /* no OOM-killer */ __GFP_NOWARN); - /* free page outside dma_mask range */ - if (p) - free_page((unsigned long)page_address(p)); - p = p1; } if (!p) { __synth_free_pages(emu, first_page, page - 1); From 83544994a6a592073e9f6b937c6da12cf5ed2d1e Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Fri, 13 Jun 2008 14:02:31 -0500 Subject: [PATCH 108/179] ALSA: Fix register programming in Freescale MPC8610 HPCD sound driver Fix the Freescale MPC8610 HPCD sound driver so that it programs the DMACR and PMUXCR registers in the global utilities correctly. Signed-off-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/fsl/mpc8610_hpcd.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index a00aac7a71f1..8820c3f8d171 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -58,9 +58,9 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) sound_device->dev.platform_data; /* Program the signal routing between the SSI and the DMA */ - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[0], CCSR_GUTS_DMACR_DEV_SSI); - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[1], CCSR_GUTS_DMACR_DEV_SSI); guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, @@ -170,9 +170,9 @@ int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) /* Restore the signal routing */ - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[0], 0); - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[1], 0); switch (machine_data->ssi_id) { @@ -182,7 +182,7 @@ int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) break; case 1: clrsetbits_be32(&machine_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA); break; } From 43785eaeb1cfb8aed3cf8027f298b242f88fdc45 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Jun 2008 15:47:26 +0200 Subject: [PATCH 109/179] ALSA: hda - Fix wrong volumes in AD1988 auto-probe mode Don't create mixer volume elements for Headphone and Speaker if they use the same DAC as normal line-outs on AD1988. Otherwise the amp value gets screwed up, e.g. https://bugzilla.novell.com/show_bug.cgi?id=398255 Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 32 +++++++++++++++++++++----------- 1 file changed, 21 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 55ce897ef0b3..05ca027fcc01 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2621,7 +2621,7 @@ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, { struct ad198x_spec *spec = codec->spec; hda_nid_t nid; - int idx, err; + int i, idx, err; char name[32]; if (! pin) @@ -2629,16 +2629,26 @@ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, idx = ad1988_pin_idx(pin); nid = ad1988_idx_to_dac(codec, idx); - /* specify the DAC as the extra output */ - if (! spec->multiout.hp_nid) - spec->multiout.hp_nid = nid; - else - spec->multiout.extra_out_nid[0] = nid; - /* control HP volume/switch on the output mixer amp */ - sprintf(name, "%s Playback Volume", pfx); - if ((err = add_control(spec, AD_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) - return err; + /* check whether the corresponding DAC was already taken */ + for (i = 0; i < spec->autocfg.line_outs; i++) { + hda_nid_t pin = spec->autocfg.line_out_pins[i]; + hda_nid_t dac = ad1988_idx_to_dac(codec, ad1988_pin_idx(pin)); + if (dac == nid) + break; + } + if (i >= spec->autocfg.line_outs) { + /* specify the DAC as the extra output */ + if (!spec->multiout.hp_nid) + spec->multiout.hp_nid = nid; + else + spec->multiout.extra_out_nid[0] = nid; + /* control HP volume/switch on the output mixer amp */ + sprintf(name, "%s Playback Volume", pfx); + err = add_control(spec, AD_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } nid = ad1988_mixer_nids[idx]; sprintf(name, "%s Playback Switch", pfx); if ((err = add_control(spec, AD_CTL_BIND_MUTE, name, From f52845ad771a2b62bc06bc940f16c8f6296654ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Jun 2008 16:35:22 +0200 Subject: [PATCH 110/179] ALSA: ca0106 - Add entry for another MSI K8N Diamond MB Added an entry for another MSI K8N Diamond mobo with SSID 1102:1009. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ca0106/ca0106_main.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index ecbe79b67e43..2f8b28add276 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -249,6 +249,11 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "MSI K8N Diamond MB [SB0438]", .gpio_type = 2, .i2c_adc = 1 } , + /* Another MSI K8N Diamond MB, which has apprently a different SSID */ + { .serial = 0x10091102, + .name = "MSI K8N Diamond MB", + .gpio_type = 2, + .i2c_adc = 1 } , /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". From 1301a964cef2759d46b4ce366eeeee1f869731f9 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Tue, 17 Jun 2008 19:19:34 +0100 Subject: [PATCH 111/179] ALSA: ASoC: fix PM=n build Fix sound/soc build failure when CONFIG_PM=n: linux-next-20080617/sound/soc/soc-core.c:829: error: 'soc_resume_deferred' undeclared (first use in this function) soc3.out:make[3]: *** [sound/soc/soc-core.o] Error 1 Signed-off-by: Randy Dunlap Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b931039632c5..bdbbc6a980fa 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -825,8 +825,10 @@ static int soc_probe(struct platform_device *pdev) /* DAPM stream work */ INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work); +#ifdef CONFIG_PM /* deferred resume work */ INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred); +#endif return 0; From 28bb171169247754bae47bad0609208333e7a34b Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Wed, 18 Jun 2008 13:45:13 -0700 Subject: [PATCH 112/179] alsa: add annotations to bitwise type snd_pcm_hw_param_t Fully half of all alsa sparse warnings are from snd_pcm_hw_param_t degrading to integer type, this goes a long way towards eliminating them. Signed-off-by: Harvey Harrison Signed-off-by: Jaroslav Kysela --- include/sound/asound.h | 12 ++++++++---- include/sound/pcm.h | 32 ++++++++++++++++---------------- 2 files changed, 24 insertions(+), 20 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 3eaf155b850d..0309da2f11d3 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -302,6 +302,8 @@ typedef int __bitwise snd_pcm_hw_param_t; #define SNDRV_PCM_HW_PARAM_SUBFORMAT ((__force snd_pcm_hw_param_t) 2) /* Subformat */ #define SNDRV_PCM_HW_PARAM_FIRST_MASK SNDRV_PCM_HW_PARAM_ACCESS #define SNDRV_PCM_HW_PARAM_LAST_MASK SNDRV_PCM_HW_PARAM_SUBFORMAT +#define SNDRV_PCM_HW_PARAM_MASK_INDEX(var) \ + ((__force int)(var) - (__force int)SNDRV_PCM_HW_PARAM_FIRST_MASK) #define SNDRV_PCM_HW_PARAM_SAMPLE_BITS ((__force snd_pcm_hw_param_t) 8) /* Bits per sample */ #define SNDRV_PCM_HW_PARAM_FRAME_BITS ((__force snd_pcm_hw_param_t) 9) /* Bits per frame */ @@ -317,6 +319,8 @@ typedef int __bitwise snd_pcm_hw_param_t; #define SNDRV_PCM_HW_PARAM_TICK_TIME ((__force snd_pcm_hw_param_t) 19) /* Approx tick duration in us */ #define SNDRV_PCM_HW_PARAM_FIRST_INTERVAL SNDRV_PCM_HW_PARAM_SAMPLE_BITS #define SNDRV_PCM_HW_PARAM_LAST_INTERVAL SNDRV_PCM_HW_PARAM_TICK_TIME +#define SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(var) \ + ((__force int)(var) - (__force int)SNDRV_PCM_HW_PARAM_FIRST_INTERVAL) #define SNDRV_PCM_HW_PARAMS_NORESAMPLE (1<<0) /* avoid rate resampling */ @@ -336,11 +340,11 @@ struct snd_mask { struct snd_pcm_hw_params { unsigned int flags; - struct snd_mask masks[SNDRV_PCM_HW_PARAM_LAST_MASK - - SNDRV_PCM_HW_PARAM_FIRST_MASK + 1]; + struct snd_mask masks[ + SNDRV_PCM_HW_PARAM_MASK_INDEX(SNDRV_PCM_HW_PARAM_LAST_MASK) + 1]; struct snd_mask mres[5]; /* reserved masks */ - struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL - - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1]; + struct snd_interval intervals[ + SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(SNDRV_PCM_HW_PARAM_LAST_INTERVAL) + 1]; struct snd_interval ires[9]; /* reserved intervals */ unsigned int rmask; /* W: requested masks */ unsigned int cmask; /* R: changed masks */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 51d58ccda2d8..5315b53f9b07 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -209,10 +209,10 @@ struct snd_pcm_hw_rule { }; struct snd_pcm_hw_constraints { - struct snd_mask masks[SNDRV_PCM_HW_PARAM_LAST_MASK - - SNDRV_PCM_HW_PARAM_FIRST_MASK + 1]; - struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL - - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1]; + struct snd_mask masks[ + SNDRV_PCM_HW_PARAM_MASK_INDEX(SNDRV_PCM_HW_PARAM_LAST_MASK) + 1]; + struct snd_interval intervals[ + SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(SNDRV_PCM_HW_PARAM_LAST_INTERVAL) + 1]; unsigned int rules_num; unsigned int rules_all; struct snd_pcm_hw_rule *rules; @@ -221,13 +221,13 @@ struct snd_pcm_hw_constraints { static inline struct snd_mask *constrs_mask(struct snd_pcm_hw_constraints *constrs, snd_pcm_hw_param_t var) { - return &constrs->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK]; + return &constrs->masks[SNDRV_PCM_HW_PARAM_MASK_INDEX(var)]; } static inline struct snd_interval *constrs_interval(struct snd_pcm_hw_constraints *constrs, snd_pcm_hw_param_t var) { - return &constrs->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]; + return &constrs->intervals[SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(var)]; } struct snd_ratnum { @@ -761,40 +761,40 @@ static inline void snd_pcm_trigger_done(struct snd_pcm_substream *substream, substream->runtime->trigger_master = master; } -static inline int hw_is_mask(int var) +static inline int hw_is_mask(snd_pcm_hw_param_t var) { - return var >= SNDRV_PCM_HW_PARAM_FIRST_MASK && - var <= SNDRV_PCM_HW_PARAM_LAST_MASK; + return (__force int)var >= (__force int)SNDRV_PCM_HW_PARAM_FIRST_MASK && + (__force int)var <= (__force int)SNDRV_PCM_HW_PARAM_LAST_MASK; } -static inline int hw_is_interval(int var) +static inline int hw_is_interval(snd_pcm_hw_param_t var) { - return var >= SNDRV_PCM_HW_PARAM_FIRST_INTERVAL && - var <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; + return (__force int)var >= (__force int)SNDRV_PCM_HW_PARAM_FIRST_INTERVAL && + (__force int)var <= (__force int)SNDRV_PCM_HW_PARAM_LAST_INTERVAL; } static inline struct snd_mask *hw_param_mask(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { - return ¶ms->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK]; + return ¶ms->masks[SNDRV_PCM_HW_PARAM_MASK_INDEX(var)]; } static inline struct snd_interval *hw_param_interval(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { - return ¶ms->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]; + return ¶ms->intervals[SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(var)]; } static inline const struct snd_mask *hw_param_mask_c(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { - return ¶ms->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK]; + return ¶ms->masks[SNDRV_PCM_HW_PARAM_MASK_INDEX(var)]; } static inline const struct snd_interval *hw_param_interval_c(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { - return ¶ms->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]; + return ¶ms->intervals[SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(var)]; } #define params_access(p) snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_ACCESS)) From a1855d802fb62718192eb7e180161b08adff4e73 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Jun 2008 15:41:37 +0200 Subject: [PATCH 113/179] ALSA: hda - Fix digital converter proc output AC_VERB_GET_DIGI_CONVERT_2 isn't actually implemented but reserved. The whole SIC bits are returned from DIGI_CONVERT_1. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_proc.c | 5 ++--- 2 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index dcd390b2bbaa..efc682888b31 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -78,7 +78,7 @@ enum { #define AC_VERB_GET_BEEP_CONTROL 0x0f0a #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c #define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d -#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e +#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */ #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 5633f77f8f3b..1e5aff5c48d1 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -366,8 +366,6 @@ static void print_digital_conv(struct snd_info_buffer *buffer, { unsigned int digi1 = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); - unsigned int digi2 = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT_2, 0); snd_iprintf(buffer, " Digital:"); if (digi1 & AC_DIG1_ENABLE) snd_iprintf(buffer, " Enabled"); @@ -386,7 +384,8 @@ static void print_digital_conv(struct snd_info_buffer *buffer, if (digi1 & AC_DIG1_LEVEL) snd_iprintf(buffer, " GenLevel"); snd_iprintf(buffer, "\n"); - snd_iprintf(buffer, " Digital category: 0x%x\n", digi2 & AC_DIG2_CC); + snd_iprintf(buffer, " Digital category: 0x%x\n", + (digi1 >> 8) & AC_DIG2_CC); } static const char *get_pwr_state(u32 state) From eabe3228a80728a21f871ccc86c72c170ca1a2dc Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 26 Jun 2008 08:51:31 +0200 Subject: [PATCH 114/179] [ALSA] Revert "alsa: add annotations to bitwise type snd_pcm_hw_param_t" This reverts commit 36b34d2437104f323e09d7c6af6451d3c0b9c0cd. From: Al Viro WIW, *all* this stuff is not bitwise at all. For crying out loud, half of these types are routinely used as array indices and loop variables... If anything, we want a different set of allowed operations - subtraction between elements of type (yielding integer), addition/subtraction of integer types not bigger than ours (yielding our type), comparisons, assignments (=, +=, -=, passing to function as argument, return from function, initializers) and second/third arguments in ?:. With 0 *not* being allowed as a constant of such type. It's not bitwise; we may use the same infrastructure in sparse, but it should be a separate class of types (__attribute__((affine))). dma_addr_t is another candidate for the same treatment, but there we'll need helpers for conversions to hw-acceptable form (dma_to_le32(), etc.) and gradual conversion of drivers. ALSA ones and pm mess are absolutely straightforward cases, though. Signed-off-by: Jaroslav Kysela --- include/sound/asound.h | 12 ++++-------- include/sound/pcm.h | 32 ++++++++++++++++---------------- 2 files changed, 20 insertions(+), 24 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 0309da2f11d3..3eaf155b850d 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -302,8 +302,6 @@ typedef int __bitwise snd_pcm_hw_param_t; #define SNDRV_PCM_HW_PARAM_SUBFORMAT ((__force snd_pcm_hw_param_t) 2) /* Subformat */ #define SNDRV_PCM_HW_PARAM_FIRST_MASK SNDRV_PCM_HW_PARAM_ACCESS #define SNDRV_PCM_HW_PARAM_LAST_MASK SNDRV_PCM_HW_PARAM_SUBFORMAT -#define SNDRV_PCM_HW_PARAM_MASK_INDEX(var) \ - ((__force int)(var) - (__force int)SNDRV_PCM_HW_PARAM_FIRST_MASK) #define SNDRV_PCM_HW_PARAM_SAMPLE_BITS ((__force snd_pcm_hw_param_t) 8) /* Bits per sample */ #define SNDRV_PCM_HW_PARAM_FRAME_BITS ((__force snd_pcm_hw_param_t) 9) /* Bits per frame */ @@ -319,8 +317,6 @@ typedef int __bitwise snd_pcm_hw_param_t; #define SNDRV_PCM_HW_PARAM_TICK_TIME ((__force snd_pcm_hw_param_t) 19) /* Approx tick duration in us */ #define SNDRV_PCM_HW_PARAM_FIRST_INTERVAL SNDRV_PCM_HW_PARAM_SAMPLE_BITS #define SNDRV_PCM_HW_PARAM_LAST_INTERVAL SNDRV_PCM_HW_PARAM_TICK_TIME -#define SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(var) \ - ((__force int)(var) - (__force int)SNDRV_PCM_HW_PARAM_FIRST_INTERVAL) #define SNDRV_PCM_HW_PARAMS_NORESAMPLE (1<<0) /* avoid rate resampling */ @@ -340,11 +336,11 @@ struct snd_mask { struct snd_pcm_hw_params { unsigned int flags; - struct snd_mask masks[ - SNDRV_PCM_HW_PARAM_MASK_INDEX(SNDRV_PCM_HW_PARAM_LAST_MASK) + 1]; + struct snd_mask masks[SNDRV_PCM_HW_PARAM_LAST_MASK - + SNDRV_PCM_HW_PARAM_FIRST_MASK + 1]; struct snd_mask mres[5]; /* reserved masks */ - struct snd_interval intervals[ - SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(SNDRV_PCM_HW_PARAM_LAST_INTERVAL) + 1]; + struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL - + SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1]; struct snd_interval ires[9]; /* reserved intervals */ unsigned int rmask; /* W: requested masks */ unsigned int cmask; /* R: changed masks */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 5315b53f9b07..51d58ccda2d8 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -209,10 +209,10 @@ struct snd_pcm_hw_rule { }; struct snd_pcm_hw_constraints { - struct snd_mask masks[ - SNDRV_PCM_HW_PARAM_MASK_INDEX(SNDRV_PCM_HW_PARAM_LAST_MASK) + 1]; - struct snd_interval intervals[ - SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(SNDRV_PCM_HW_PARAM_LAST_INTERVAL) + 1]; + struct snd_mask masks[SNDRV_PCM_HW_PARAM_LAST_MASK - + SNDRV_PCM_HW_PARAM_FIRST_MASK + 1]; + struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL - + SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1]; unsigned int rules_num; unsigned int rules_all; struct snd_pcm_hw_rule *rules; @@ -221,13 +221,13 @@ struct snd_pcm_hw_constraints { static inline struct snd_mask *constrs_mask(struct snd_pcm_hw_constraints *constrs, snd_pcm_hw_param_t var) { - return &constrs->masks[SNDRV_PCM_HW_PARAM_MASK_INDEX(var)]; + return &constrs->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK]; } static inline struct snd_interval *constrs_interval(struct snd_pcm_hw_constraints *constrs, snd_pcm_hw_param_t var) { - return &constrs->intervals[SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(var)]; + return &constrs->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]; } struct snd_ratnum { @@ -761,40 +761,40 @@ static inline void snd_pcm_trigger_done(struct snd_pcm_substream *substream, substream->runtime->trigger_master = master; } -static inline int hw_is_mask(snd_pcm_hw_param_t var) +static inline int hw_is_mask(int var) { - return (__force int)var >= (__force int)SNDRV_PCM_HW_PARAM_FIRST_MASK && - (__force int)var <= (__force int)SNDRV_PCM_HW_PARAM_LAST_MASK; + return var >= SNDRV_PCM_HW_PARAM_FIRST_MASK && + var <= SNDRV_PCM_HW_PARAM_LAST_MASK; } -static inline int hw_is_interval(snd_pcm_hw_param_t var) +static inline int hw_is_interval(int var) { - return (__force int)var >= (__force int)SNDRV_PCM_HW_PARAM_FIRST_INTERVAL && - (__force int)var <= (__force int)SNDRV_PCM_HW_PARAM_LAST_INTERVAL; + return var >= SNDRV_PCM_HW_PARAM_FIRST_INTERVAL && + var <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; } static inline struct snd_mask *hw_param_mask(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { - return ¶ms->masks[SNDRV_PCM_HW_PARAM_MASK_INDEX(var)]; + return ¶ms->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK]; } static inline struct snd_interval *hw_param_interval(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { - return ¶ms->intervals[SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(var)]; + return ¶ms->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]; } static inline const struct snd_mask *hw_param_mask_c(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { - return ¶ms->masks[SNDRV_PCM_HW_PARAM_MASK_INDEX(var)]; + return ¶ms->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK]; } static inline const struct snd_interval *hw_param_interval_c(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { - return ¶ms->intervals[SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(var)]; + return ¶ms->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]; } #define params_access(p) snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_ACCESS)) From 3e0e469fa216ec70c93b1593821b759d19ee2e6b Mon Sep 17 00:00:00 2001 From: Travis Place Date: Fri, 20 Jun 2008 16:51:45 +0200 Subject: [PATCH 115/179] ALSA: hda - Added model selection for iMac 24" Added the SSID of a known iMac 24" to automatically use ALC885_IMAC24 quirk. Signed-off-by: Travis Place Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 61f8c13b2bb9..d96a8762b287 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6192,6 +6192,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), + SND_PCI_QUIRK(0x106b, 0x00a0, "Apple iMac 24''", ALC885_IMAC24), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), From 1d11604eebfc91204e3182b48d3d920f51b09d40 Mon Sep 17 00:00:00 2001 From: Travis Place Date: Mon, 23 Jun 2008 11:42:30 +0200 Subject: [PATCH 116/179] ALSA: hda - Added SSID for 'Fujitsu Siemens Amilo M1451G' laptop Add the SSID for the "Fujitsu Siemens Amilo M1451G" laptop to patch_realtek.c , so that it uses ALC880_FUJITSU by default. Signed-off-by: Travis Place Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d96a8762b287..4edde6094cd1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3074,6 +3074,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), + SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL), SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW), From 981bcead3f2279a1ec6fb5f2c57aff79ed61a700 Mon Sep 17 00:00:00 2001 From: Pierre Ossman Date: Mon, 23 Jun 2008 11:45:47 +0200 Subject: [PATCH 117/179] ALSA: trident - pause s/pdif output Stop the S/PDIF DMA engine and output when the device is told to pause. It will keep on looping the current buffer contents if this isn't done. Signed-off-by: Pierre Ossman Tested-by: Rene Herman Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/trident/trident_main.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index bbcee2c09ae4..a69b4206c69e 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -1590,7 +1590,10 @@ static int snd_trident_trigger(struct snd_pcm_substream *substream, if (spdif_flag) { if (trident->device != TRIDENT_DEVICE_ID_SI7018) { outl(trident->spdif_pcm_bits, TRID_REG(trident, NX_SPCSTATUS)); - outb(trident->spdif_pcm_ctrl, TRID_REG(trident, NX_SPCTRL_SPCSO + 3)); + val = trident->spdif_pcm_ctrl; + if (!go) + val &= ~(0x28); + outb(val, TRID_REG(trident, NX_SPCTRL_SPCSO + 3)); } else { outl(trident->spdif_pcm_bits, TRID_REG(trident, SI_SPDIF_CS)); val = inl(TRID_REG(trident, SI_SERIAL_INTF_CTRL)) | SPDIF_EN; From 627d3e7abca30d6e86787c98dd7cbac0233bc5a9 Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Mon, 23 Jun 2008 11:50:47 +0200 Subject: [PATCH 118/179] ALSA: PCI168 snd-azt3328: some more fixups - fix problem with codec register 0x6a being write-only by adding a software shadow register (caused annoying noise after module loading due to _toggling_ between gameport and audio bits instead of configuring them properly) - rename several "Wave" mixer controls to "PCM", since this is what Wine and several other apps are looking for (IOW, _requiring_) and this is what AC97 specs use as naming, too, thus I'd guess it's what these controls are - cleanup, small optimizations Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/azt3328.c | 109 ++++++++++++++++++++++++++------------------ sound/pci/azt3328.h | 16 +++++-- 2 files changed, 76 insertions(+), 49 deletions(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index b832333c3023..22f18f3cfbc9 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -289,6 +289,12 @@ struct snd_azf3328 { struct pci_dev *pci; int irq; + /* register 0x6a is write-only, thus need to remember setting. + * If we need to add more registers here, then we might try to fold this + * into some transparent combined shadow register handling with + * CONFIG_PM register storage below, but that's slightly difficult. */ + u16 shadow_reg_codec_6AH; + #ifdef CONFIG_PM /* register value containers for power management * Note: not always full I/O range preserved (just like Win driver!) */ @@ -324,21 +330,6 @@ snd_azf3328_io_reg_setb(unsigned reg, u8 mask, int do_set) return 0; } -static int -snd_azf3328_io_reg_setw(unsigned reg, u16 mask, int do_set) -{ - u16 prev = inw(reg), new; - - new = (do_set) ? (prev|mask) : (prev & ~mask); - /* we need to always write the new value no matter whether it differs - * or not, since some register bits don't indicate their setting */ - outw(new, reg); - if (new != prev) - return 1; - - return 0; -} - static inline void snd_azf3328_codec_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) { @@ -662,7 +653,7 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, "pre 3D", "post 3D" }; struct azf3328_mixer_reg reg; - const char *p = NULL; + const char * const *p = NULL; snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -673,20 +664,20 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, if (reg.reg == IDX_MIXER_ADVCTL2) { switch(reg.lchan_shift) { case 8: /* modem out sel */ - p = texts1[uinfo->value.enumerated.item]; + p = texts1; break; case 9: /* mono sel source */ - p = texts2[uinfo->value.enumerated.item]; + p = texts2; break; case 15: /* PCM Out Path */ - p = texts4[uinfo->value.enumerated.item]; + p = texts4; break; } } else if (reg.reg == IDX_MIXER_REC_SELECT) - p = texts3[uinfo->value.enumerated.item]; + p = texts3; - strcpy(uinfo->value.enumerated.name, p); + strcpy(uinfo->value.enumerated.name, p[uinfo->value.enumerated.item]); return 0; } @@ -745,9 +736,11 @@ snd_azf3328_put_mixer_enum(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new snd_azf3328_mixer_controls[] __devinitdata = { AZF3328_MIXER_SWITCH("Master Playback Switch", IDX_MIXER_PLAY_MASTER, 15, 1), AZF3328_MIXER_VOL_STEREO("Master Playback Volume", IDX_MIXER_PLAY_MASTER, 0x1f, 1), - AZF3328_MIXER_SWITCH("Wave Playback Switch", IDX_MIXER_WAVEOUT, 15, 1), - AZF3328_MIXER_VOL_STEREO("Wave Playback Volume", IDX_MIXER_WAVEOUT, 0x1f, 1), - AZF3328_MIXER_SWITCH("Wave 3D Bypass Playback Switch", IDX_MIXER_ADVCTL2, 7, 1), + AZF3328_MIXER_SWITCH("PCM Playback Switch", IDX_MIXER_WAVEOUT, 15, 1), + AZF3328_MIXER_VOL_STEREO("PCM Playback Volume", + IDX_MIXER_WAVEOUT, 0x1f, 1), + AZF3328_MIXER_SWITCH("PCM 3D Bypass Playback Switch", + IDX_MIXER_ADVCTL2, 7, 1), AZF3328_MIXER_SWITCH("FM Playback Switch", IDX_MIXER_FMSYNTH, 15, 1), AZF3328_MIXER_VOL_STEREO("FM Playback Volume", IDX_MIXER_FMSYNTH, 0x1f, 1), AZF3328_MIXER_SWITCH("CD Playback Switch", IDX_MIXER_CDAUDIO, 15, 1), @@ -874,7 +867,7 @@ snd_azf3328_hw_free(struct snd_pcm_substream *substream) static void snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, unsigned reg, - unsigned int bitrate, + enum azf_freq_t bitrate, unsigned int format_width, unsigned int channels ) @@ -958,15 +951,29 @@ snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328 *chip, snd_azf3328_codec_setfmt(chip, reg, AZF_FREQ_4000, 8, 1); } +static void +snd_azf3328_codec_reg_6AH_update(struct snd_azf3328 *chip, + unsigned bitmask, + int enable +) +{ + if (enable) + chip->shadow_reg_codec_6AH &= ~bitmask; + else + chip->shadow_reg_codec_6AH |= bitmask; + snd_azf3328_dbgplay("6AH_update mask 0x%04x enable %d: val 0x%04x\n", + bitmask, enable, chip->shadow_reg_codec_6AH); + snd_azf3328_codec_outw(chip, IDX_IO_6AH, chip->shadow_reg_codec_6AH); +} + static inline void snd_azf3328_codec_enable(struct snd_azf3328 *chip, int enable) { + snd_azf3328_dbgplay("codec_enable %d\n", enable); /* no idea what exactly is being done here, but I strongly assume it's * PM related */ - snd_azf3328_io_reg_setw( - chip->codec_io+IDX_IO_6AH, - IO_6A_PAUSE_PLAYBACK_BIT8, - !enable + snd_azf3328_codec_reg_6AH_update( + chip, IO_6A_PAUSE_PLAYBACK_BIT8, enable ); } @@ -1404,10 +1411,8 @@ snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, int enable) static inline void snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, int enable) { - snd_azf3328_io_reg_setw( - chip->codec_io+IDX_IO_6AH, - IO_6A_SOMETHING2_GAMEPORT, - !enable + snd_azf3328_codec_reg_6AH_update( + chip, IO_6A_SOMETHING2_GAMEPORT, enable ); } @@ -1525,8 +1530,6 @@ snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) { struct gameport *gp; - int io_port = chip->game_io; - chip->gameport = gp = gameport_allocate_port(); if (!gp) { printk(KERN_ERR "azt3328: cannot alloc memory for gameport\n"); @@ -1536,7 +1539,7 @@ snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) gameport_set_name(gp, "AZF3328 Gameport"); gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci)); gameport_set_dev_parent(gp, &chip->pci->dev); - gp->io = io_port; + gp->io = chip->game_io; gameport_set_port_data(gp, chip); gp->open = snd_azf3328_gameport_open; @@ -1577,6 +1580,15 @@ snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip) /******************************************************************/ +static inline void +snd_azf3328_irq_log_unknown_type(u8 which) +{ + snd_azf3328_dbgplay( + "azt3328: unknown IRQ type (%x) occurred, please report!\n", + which + ); +} + static irqreturn_t snd_azf3328_interrupt(int irq, void *dev_id) { @@ -1594,11 +1606,14 @@ snd_azf3328_interrupt(int irq, void *dev_id) )) return IRQ_NONE; /* must be interrupt for another device */ - snd_azf3328_dbgplay("Interrupt %ld!\nIDX_IO_PLAY_FLAGS %04x, IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", - irq_count++ /* debug-only */, - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), - status); + snd_azf3328_dbgplay( + "irq_count %ld! IDX_IO_PLAY_FLAGS %04x, " + "IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", + irq_count++ /* debug-only */, + snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), + snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), + status + ); if (status & IRQ_TIMER) { /* snd_azf3328_dbgplay("timer %ld\n", @@ -1631,9 +1646,9 @@ snd_azf3328_interrupt(int irq, void *dev_id) ) ); } else - snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); + printk(KERN_WARNING "azt3328: irq handler problem!\n"); if (which & IRQ_PLAY_SOMETHING) - snd_azf3328_dbgplay("azt3328: unknown play IRQ type occurred, please report!\n"); + snd_azf3328_irq_log_unknown_type(which); } if (status & IRQ_RECORDING) { spin_lock(&chip->reg_lock); @@ -1653,9 +1668,9 @@ snd_azf3328_interrupt(int irq, void *dev_id) ) ); } else - snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); + printk(KERN_WARNING "azt3328: irq handler problem!\n"); if (which & IRQ_REC_SOMETHING) - snd_azf3328_dbgplay("azt3328: unknown rec IRQ type occurred, please report!\n"); + snd_azf3328_irq_log_unknown_type(which); } if (status & IRQ_GAMEPORT) snd_azf3328_gameport_interrupt(chip); @@ -2311,6 +2326,10 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) chip->saved_regs_codec[reg] = inw(chip->codec_io + reg * 2); + + /* manually store the one currently relevant write-only reg, too */ + chip->saved_regs_codec[IDX_IO_6AH / 2] = chip->shadow_reg_codec_6AH; + for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) chip->saved_regs_game[reg] = inw(chip->game_io + reg * 2); for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg) diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h index 3448fd626f80..7e3e8942d073 100644 --- a/sound/pci/azt3328.h +++ b/sound/pci/azt3328.h @@ -1,7 +1,8 @@ #ifndef __SOUND_AZT3328_H #define __SOUND_AZT3328_H -/* "PU" == "power-up value", as tested on PCI168 PCI rev. 10 */ +/* "PU" == "power-up value", as tested on PCI168 PCI rev. 10 + * "WRITE_ONLY" == register does not indicate actual bit values */ /*** main I/O area port indices ***/ /* (only 0x70 of 0x80 bytes saved/restored by Windows driver) */ @@ -76,7 +77,7 @@ #define SOUNDFORMAT_FLAG_2CHANNELS 0x0020 /* define frequency helpers, for maximum value safety */ -enum { +enum azf_freq_t { #define AZF_FREQ(rate) AZF_FREQ_##rate = rate AZF_FREQ(4000), AZF_FREQ(4800), @@ -150,11 +151,18 @@ enum { #define IO_68_RANDOM_TOGGLE1 0x0100 /* toggles randomly */ #define IO_68_RANDOM_TOGGLE2 0x0200 /* toggles randomly */ /* umm, nope, behaviour of these bits changes depending on what we wrote - * to 0x6b!! */ + * to 0x6b!! + * And they change upon playback/stop, too: + * Writing a value to 0x68 will display this exact value during playback, + * too but when stopped it can fall back to a rather different + * seemingly random value). Hmm, possibly this is a register which + * has a remote shadow which needs proper device supply which only exists + * in case playback is active? Or is this driver-induced? + */ /* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); * actually inhibits PCM playback!!! maybe power management??: */ -#define IDX_IO_6AH 0x6A +#define IDX_IO_6AH 0x6A /* WRITE_ONLY! */ /* bit 5: enabling this will activate permanent counting of bytes 2/3 * at gameport I/O (0xb402/3) (equal values each) and cause * gameport legacy I/O at 0x0200 to be _DISABLED_! From 796d2ca84859d1fdb11ff06cd9707ffab5642fca Mon Sep 17 00:00:00 2001 From: Richard Purdie Date: Mon, 23 Jun 2008 14:51:28 +0100 Subject: [PATCH 119/179] ALSA: ASoC: Add AK4535 driver The AK4535 codec is included in some HP iPAQ systems. This driver was originally written by Richard Purdie and with some bug fixes from Milan Plzik. While out of tree it has also had some mechanical updates for new APIs and current best practices from Liam Girdwood, Graeme Gregory and Mark Brown. Signed-off-by: Richard Purdie Signed-off-by: Milan Plzik Signed-off-by: Liam Girdwood Signed-off-by: Graeme Gregory Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4535.c | 696 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4535.h | 46 +++ 4 files changed, 747 insertions(+) create mode 100644 sound/soc/codecs/ak4535.c create mode 100644 sound/soc/codecs/ak4535.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f439138db843..1db04a28a53d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2,6 +2,9 @@ config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC +config SND_SOC_AK4535 + tristate + config SND_SOC_UDA1380 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d9826f35d8c0..d7b97abcf729 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,4 +1,5 @@ snd-soc-ac97-objs := ac97.o +snd-soc-ak4535-objs := ak4535.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8731-objs := wm8731.o @@ -11,6 +12,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o +obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c new file mode 100644 index 000000000000..469266e881d8 --- /dev/null +++ b/sound/soc/codecs/ak4535.c @@ -0,0 +1,696 @@ +/* + * ak4535.c -- AK4535 ALSA Soc Audio driver + * + * Copyright 2005 Openedhand Ltd. + * + * Author: Richard Purdie + * + * Based on wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "ak4535.h" + +#define AUDIO_NAME "ak4535" +#define AK4535_VERSION "0.3" + +struct snd_soc_codec_device soc_codec_dev_ak4535; + +/* codec private data */ +struct ak4535_priv { + unsigned int sysclk; +}; + +/* + * ak4535 register cache + */ +static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { + 0x0000, 0x0080, 0x0000, 0x0003, + 0x0002, 0x0000, 0x0011, 0x0001, + 0x0000, 0x0040, 0x0036, 0x0010, + 0x0000, 0x0000, 0x0057, 0x0000, +}; + +/* + * read ak4535 register cache + */ +static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4535_CACHEREGNUM) + return -1; + return cache[reg]; +} + +static inline unsigned int ak4535_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 data; + data = reg; + + if (codec->hw_write(codec->control_data, &data, 1) != 1) + return -EIO; + + if (codec->hw_read(codec->control_data, &data, 1) != 1) + return -EIO; + + return data; +}; + +/* + * write ak4535 register cache + */ +static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4535_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the AK4535 register space + */ +static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK4535 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + ak4535_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static int ak4535_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r = 0; + + for (i = 0; i < AK4535_CACHEREGNUM; i++) + r |= ak4535_write(codec, i, cache[i]); + + return r; +}; + +static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; +static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"}; +static const char *ak4535_hp_out[] = {"Stereo", "Mono"}; +static const char *ak4535_deemp[] = {"44.1kHz", "Off", "48kHz", "32kHz"}; +static const char *ak4535_mic_select[] = {"Internal", "External"}; + +static const struct soc_enum ak4535_enum[] = { + SOC_ENUM_SINGLE(AK4535_SIG1, 7, 2, ak4535_mono_gain), + SOC_ENUM_SINGLE(AK4535_SIG1, 6, 2, ak4535_mono_out), + SOC_ENUM_SINGLE(AK4535_MODE2, 2, 2, ak4535_hp_out), + SOC_ENUM_SINGLE(AK4535_DAC, 0, 4, ak4535_deemp), + SOC_ENUM_SINGLE(AK4535_MIC, 1, 2, ak4535_mic_select), +}; + +static const struct snd_kcontrol_new ak4535_snd_controls[] = { + SOC_SINGLE("ALC2 Switch", AK4535_SIG1, 1, 1, 0), + SOC_ENUM("Mono 1 Output", ak4535_enum[1]), + SOC_ENUM("Mono 1 Gain", ak4535_enum[0]), + SOC_ENUM("Headphone Output", ak4535_enum[2]), + SOC_ENUM("Playback Deemphasis", ak4535_enum[3]), + SOC_SINGLE("Bass Volume", AK4535_DAC, 2, 3, 0), + SOC_SINGLE("Mic Boost (+20dB) Switch", AK4535_MIC, 0, 1, 0), + SOC_ENUM("Mic Select", ak4535_enum[4]), + SOC_SINGLE("ALC Operation Time", AK4535_TIMER, 0, 3, 0), + SOC_SINGLE("ALC Recovery Time", AK4535_TIMER, 2, 3, 0), + SOC_SINGLE("ALC ZC Time", AK4535_TIMER, 4, 3, 0), + SOC_SINGLE("ALC 1 Switch", AK4535_ALC1, 5, 1, 0), + SOC_SINGLE("ALC 2 Switch", AK4535_ALC1, 6, 1, 0), + SOC_SINGLE("ALC Volume", AK4535_ALC2, 0, 127, 0), + SOC_SINGLE("Capture Volume", AK4535_PGA, 0, 127, 0), + SOC_SINGLE("Left Playback Volume", AK4535_LATT, 0, 127, 1), + SOC_SINGLE("Right Playback Volume", AK4535_RATT, 0, 127, 1), + SOC_SINGLE("AUX Bypass Volume", AK4535_VOL, 0, 15, 0), + SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0), +}; + +/* add non dapm controls */ +static int ak4535_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&ak4535_snd_controls[i], codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Mono 1 Mixer */ +static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0), + SOC_DAPM_SINGLE("Mono Playback Switch", AK4535_SIG1, 5, 1, 0), +}; + +/* Stereo Mixer */ +static const struct snd_kcontrol_new ak4535_stereo_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG2, 4, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", AK4535_SIG2, 7, 1, 0), + SOC_DAPM_SINGLE("Aux Bypass Switch", AK4535_SIG2, 5, 1, 0), +}; + +/* Input Mixer */ +static const struct snd_kcontrol_new ak4535_input_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Capture Switch", AK4535_MIC, 2, 1, 0), + SOC_DAPM_SINGLE("Aux Capture Switch", AK4535_MIC, 5, 1, 0), +}; + +/* Input mux */ +static const struct snd_kcontrol_new ak4535_input_mux_control = + SOC_DAPM_ENUM("Input Select", ak4535_enum[4]); + +/* HP L switch */ +static const struct snd_kcontrol_new ak4535_hpl_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 1, 1, 1); + +/* HP R switch */ +static const struct snd_kcontrol_new ak4535_hpr_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 0, 1, 1); + +/* mono 2 switch */ +static const struct snd_kcontrol_new ak4535_mono2_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG1, 0, 1, 0); + +/* Line out switch */ +static const struct snd_kcontrol_new ak4535_line_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 6, 1, 0); + +/* ak4535 dapm widgets */ +static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_stereo_mixer_controls[0], + ARRAY_SIZE(ak4535_stereo_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_mono1_mixer_controls[0], + ARRAY_SIZE(ak4535_mono1_mixer_controls)), + SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_input_mixer_controls[0], + ARRAY_SIZE(ak4535_input_mixer_controls)), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ak4535_input_mux_control), + SND_SOC_DAPM_DAC("DAC", "Playback", AK4535_PM2, 0, 0), + SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0, + &ak4535_mono2_control), + /* speaker powersave bit */ + SND_SOC_DAPM_PGA("Speaker Enable", AK4535_MODE2, 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ak4535_line_control), + SND_SOC_DAPM_SWITCH("Left HP Enable", SND_SOC_NOPM, 0, 0, + &ak4535_hpl_control), + SND_SOC_DAPM_SWITCH("Right HP Enable", SND_SOC_NOPM, 0, 0, + &ak4535_hpr_control), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPP"), + SND_SOC_DAPM_OUTPUT("SPN"), + SND_SOC_DAPM_OUTPUT("MOUT1"), + SND_SOC_DAPM_OUTPUT("MOUT2"), + SND_SOC_DAPM_OUTPUT("MICOUT"), + SND_SOC_DAPM_ADC("ADC", "Capture", AK4535_PM1, 0, 0), + SND_SOC_DAPM_PGA("Spk Amp", AK4535_PM2, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP R Amp", AK4535_PM2, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP L Amp", AK4535_PM2, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic", AK4535_PM1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line Out", AK4535_PM1, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", AK4535_PM1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX In", AK4535_PM1, 2, 0, NULL, 0), + + SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4535_MIC, 3, 0), + SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4535_MIC, 4, 0), + SND_SOC_DAPM_INPUT("MICIN"), + SND_SOC_DAPM_INPUT("MICEXT"), + SND_SOC_DAPM_INPUT("AUX"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("AIN"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /*stereo mixer */ + {"Stereo Mixer", "Playback Switch", "DAC"}, + {"Stereo Mixer", "Mic Sidetone Switch", "Mic"}, + {"Stereo Mixer", "Aux Bypass Switch", "AUX In"}, + + /* mono1 mixer */ + {"Mono1 Mixer", "Mic Sidetone Switch", "Mic"}, + {"Mono1 Mixer", "Mono Playback Switch", "DAC"}, + + /* Mic */ + {"Mic", NULL, "AIN"}, + {"Input Mux", "Internal", "Mic Int Bias"}, + {"Input Mux", "External", "Mic Ext Bias"}, + {"Mic Int Bias", NULL, "MICIN"}, + {"Mic Ext Bias", NULL, "MICEXT"}, + {"MICOUT", NULL, "Input Mux"}, + + /* line out */ + {"LOUT", NULL, "Line Out Enable"}, + {"ROUT", NULL, "Line Out Enable"}, + {"Line Out Enable", "Switch", "Line Out"}, + {"Line Out", NULL, "Stereo Mixer"}, + + /* mono1 out */ + {"MOUT1", NULL, "Mono Out"}, + {"Mono Out", NULL, "Mono1 Mixer"}, + + /* left HP */ + {"HPL", NULL, "Left HP Enable"}, + {"Left HP Enable", "Switch", "HP L Amp"}, + {"HP L Amp", NULL, "Stereo Mixer"}, + + /* right HP */ + {"HPR", NULL, "Right HP Enable"}, + {"Right HP Enable", "Switch", "HP R Amp"}, + {"HP R Amp", NULL, "Stereo Mixer"}, + + /* speaker */ + {"SPP", NULL, "Speaker Enable"}, + {"SPN", NULL, "Speaker Enable"}, + {"Speaker Enable", "Switch", "Spk Amp"}, + {"Spk Amp", NULL, "MIN"}, + + /* mono 2 */ + {"MOUT2", NULL, "Mono 2 Enable"}, + {"Mono 2 Enable", "Switch", "Stereo Mixer"}, + + /* Aux In */ + {"Aux In", NULL, "AUX"}, + + /* ADC */ + {"ADC", NULL, "Input Mixer"}, + {"Input Mixer", "Mic Capture Switch", "Mic"}, + {"Input Mixer", "Aux Capture Switch", "Aux In"}, +}; + +static int ak4535_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ak4535_dapm_widgets, + ARRAY_SIZE(ak4535_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int ak4535_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4535_priv *ak4535 = codec->private_data; + + ak4535->sysclk = freq; + return 0; +} + +static int ak4535_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct ak4535_priv *ak4535 = codec->private_data; + u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5); + int rate = params_rate(params), fs = 256; + + if (rate) + fs = ak4535->sysclk / rate; + + /* set fs */ + switch (fs) { + case 1024: + mode2 |= (0x2 << 5); + break; + case 512: + mode2 |= (0x1 << 5); + break; + case 256: + break; + } + + /* set rate */ + ak4535_write(codec, AK4535_MODE2, mode2); + return 0; +} + +static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 mode1 = 0; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + mode1 = 0x0002; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode1 = 0x0001; + break; + default: + return -EINVAL; + } + + /* use 32 fs for BCLK to save power */ + mode1 |= 0x4; + + ak4535_write(codec, AK4535_MODE1, mode1); + return 0; +} + +static int ak4535_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf; + if (!mute) + ak4535_write(codec, AK4535_DAC, mute_reg); + else + ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + return 0; +} + +static int ak4535_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 i; + + switch (level) { + case SND_SOC_BIAS_ON: + ak4535_mute(codec->dai, 0); + break; + case SND_SOC_BIAS_PREPARE: + ak4535_mute(codec->dai, 1); + break; + case SND_SOC_BIAS_STANDBY: + i = ak4535_read_reg_cache(codec, AK4535_PM1); + ak4535_write(codec, AK4535_PM1, i | 0x80); + i = ak4535_read_reg_cache(codec, AK4535_PM2); + ak4535_write(codec, AK4535_PM2, i & (~0x80)); + break; + case SND_SOC_BIAS_OFF: + i = ak4535_read_reg_cache(codec, AK4535_PM1); + ak4535_write(codec, AK4535_PM1, i & (~0x80)); + break; + } + codec->bias_level = level; + return 0; +} + +#define AK4535_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +struct snd_soc_codec_dai ak4535_dai = { + .name = "AK4535", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AK4535_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AK4535_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .hw_params = ak4535_hw_params, + }, + .dai_ops = { + .set_fmt = ak4535_set_dai_fmt, + .digital_mute = ak4535_mute, + .set_sysclk = ak4535_set_dai_sysclk, + }, +}; +EXPORT_SYMBOL_GPL(ak4535_dai); + +static int ak4535_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ak4535_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + ak4535_sync(codec); + ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + ak4535_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the AK4535 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak4535_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "AK4535"; + codec->owner = THIS_MODULE; + codec->read = ak4535_read_reg_cache; + codec->write = ak4535_write; + codec->set_bias_level = ak4535_set_bias_level; + codec->dai = &ak4535_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(ak4535_reg); + codec->reg_cache = kmemdup(ak4535_reg, sizeof(ak4535_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak4535: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ak4535_add_controls(codec); + ak4535_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak4535: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + + return ret; +} + +static struct snd_soc_device *ak4535_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +#define I2C_DRIVERID_AK4535 0xfefe /* liam - need a proper id */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver ak4535_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ +static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = ak4535_socdev; + struct ak4535_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + printk(KERN_ERR "failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = ak4535_init(socdev); + if (ret < 0) { + printk(KERN_ERR "failed to initialise AK4535\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int ak4535_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int ak4535_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, ak4535_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver ak4535_i2c_driver = { + .driver = { + .name = "AK4535 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_AK4535, + .attach_adapter = ak4535_i2c_attach, + .detach_client = ak4535_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "AK4535", + .driver = &ak4535_i2c_driver, +}; +#endif + +static int ak4535_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct ak4535_setup_data *setup; + struct snd_soc_codec *codec; + struct ak4535_priv *ak4535; + int ret = 0; + + printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + ak4535 = kzalloc(sizeof(struct ak4535_priv), GFP_KERNEL); + if (ak4535 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = ak4535; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ak4535_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->hw_read = (hw_read_t)i2c_master_recv; + ret = i2c_add_driver(&ak4535_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int ak4535_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&ak4535_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4535 = { + .probe = ak4535_probe, + .remove = ak4535_remove, + .suspend = ak4535_suspend, + .resume = ak4535_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535); + +MODULE_DESCRIPTION("Soc AK4535 driver"); +MODULE_AUTHOR("Richard Purdie"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h new file mode 100644 index 000000000000..fc686ddf753a --- /dev/null +++ b/sound/soc/codecs/ak4535.h @@ -0,0 +1,46 @@ +/* + * ak4535.h -- AK4535 Soc Audio driver + * + * Copyright 2005 Openedhand Ltd. + * + * Author: Richard Purdie + * + * Based on wm8753.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AK4535_H +#define _AK4535_H + +/* AK4535 register space */ + +#define AK4535_PM1 0x0 +#define AK4535_PM2 0x1 +#define AK4535_SIG1 0x2 +#define AK4535_SIG2 0x3 +#define AK4535_MODE1 0x4 +#define AK4535_MODE2 0x5 +#define AK4535_DAC 0x6 +#define AK4535_MIC 0x7 +#define AK4535_TIMER 0x8 +#define AK4535_ALC1 0x9 +#define AK4535_ALC2 0xa +#define AK4535_PGA 0xb +#define AK4535_LATT 0xc +#define AK4535_RATT 0xd +#define AK4535_VOL 0xe +#define AK4535_STATUS 0xf + +#define AK4535_CACHEREGNUM 0x10 + +struct ak4535_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_codec_dai ak4535_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4535; + +#endif From a5c95e90c1baa9c1114875264bbd283526eb8377 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Jun 2008 14:51:29 +0100 Subject: [PATCH 120/179] ALSA: ASoC: Replace custom debug macros with pr_ equivalents Several ASoC codec drivers use custom macros equivalent to the standard pr_ macros, most of which are not actually used. Replace these custom macros with the standard ones. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm8510.c | 25 +++---------------------- sound/soc/codecs/wm8731.c | 25 +++---------------------- sound/soc/codecs/wm8750.c | 25 +++---------------------- sound/soc/codecs/wm8753.c | 25 +++---------------------- sound/soc/codecs/wm8990.c | 25 +++---------------------- 5 files changed, 15 insertions(+), 110 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 152e6f21154f..b549f6753aba 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -30,25 +30,6 @@ #define AUDIO_NAME "wm8510" #define WM8510_VERSION "0.6" -/* - * Debug - */ - -#define WM8510_DEBUG 0 - -#ifdef WM8510_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif -#define err(format, arg...) \ - printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -#define info(format, arg...) \ - printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -#define warn(format, arg...) \ - printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) - struct snd_soc_codec_device soc_codec_dev_wm8510; /* @@ -721,13 +702,13 @@ static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) ret = i2c_attach_client(i2c); if (ret < 0) { - err("failed to attach codec at addr %x\n", addr); + pr_err("failed to attach codec at addr %x\n", addr); goto err; } ret = wm8510_init(socdev); if (ret < 0) { - err("failed to initialise WM8510\n"); + pr_err("failed to initialise WM8510\n"); goto err; } return ret; @@ -777,7 +758,7 @@ static int wm8510_probe(struct platform_device *pdev) struct snd_soc_codec *codec; int ret = 0; - info("WM8510 Audio Codec %s", WM8510_VERSION); + pr_info("WM8510 Audio Codec %s", WM8510_VERSION); setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 77880537a3cd..3ff42ad65ede 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -31,25 +31,6 @@ #define AUDIO_NAME "wm8731" #define WM8731_VERSION "0.13" -/* - * Debug - */ - -#define WM8731_DEBUG 0 - -#ifdef WM8731_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif -#define err(format, arg...) \ - printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -#define info(format, arg...) \ - printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -#define warn(format, arg...) \ - printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) - struct snd_soc_codec_device soc_codec_dev_wm8731; /* codec private data */ @@ -624,13 +605,13 @@ static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind) ret = i2c_attach_client(i2c); if (ret < 0) { - err("failed to attach codec at addr %x\n", addr); + pr_err("failed to attach codec at addr %x\n", addr); goto err; } ret = wm8731_init(socdev); if (ret < 0) { - err("failed to initialise WM8731\n"); + pr_err("failed to initialise WM8731\n"); goto err; } return ret; @@ -681,7 +662,7 @@ static int wm8731_probe(struct platform_device *pdev) struct wm8731_priv *wm8731; int ret = 0; - info("WM8731 Audio Codec %s", WM8731_VERSION); + pr_info("WM8731 Audio Codec %s", WM8731_VERSION); setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 1ae670a98c5c..eb460c9aa63e 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -31,25 +31,6 @@ #define AUDIO_NAME "WM8750" #define WM8750_VERSION "0.12" -/* - * Debug - */ - -#define WM8750_DEBUG 0 - -#ifdef WM8750_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif -#define err(format, arg...) \ - printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -#define info(format, arg...) \ - printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -#define warn(format, arg...) \ - printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) - /* codec private data */ struct wm8750_priv { unsigned int sysclk; @@ -896,13 +877,13 @@ static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind) ret = i2c_attach_client(i2c); if (ret < 0) { - err("failed to attach codec at addr %x\n", addr); + pr_err("failed to attach codec at addr %x\n", addr); goto err; } ret = wm8750_init(socdev); if (ret < 0) { - err("failed to initialise WM8750\n"); + pr_err("failed to initialise WM8750\n"); goto err; } return ret; @@ -953,7 +934,7 @@ static int wm8750_probe(struct platform_device *pdev) struct wm8750_priv *wm8750; int ret = 0; - info("WM8750 Audio Codec %s", WM8750_VERSION); + pr_info("WM8750 Audio Codec %s", WM8750_VERSION); codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) return -ENOMEM; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 00b481183d45..be01a738f184 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -55,25 +55,6 @@ #define AUDIO_NAME "wm8753" #define WM8753_VERSION "0.16" -/* - * Debug - */ - -#define WM8753_DEBUG 0 - -#ifdef WM8753_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif -#define err(format, arg...) \ - printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -#define info(format, arg...) \ - printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -#define warn(format, arg...) \ - printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) - static int caps_charge = 2000; module_param(caps_charge, int, 0); MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)"); @@ -1689,13 +1670,13 @@ static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind) ret = i2c_attach_client(i2c); if (ret < 0) { - err("failed to attach codec at addr %x\n", addr); + pr_err("failed to attach codec at addr %x\n", addr); goto err; } ret = wm8753_init(socdev); if (ret < 0) { - err("failed to initialise WM8753\n"); + pr_err("failed to initialise WM8753\n"); goto err; } @@ -1747,7 +1728,7 @@ static int wm8753_probe(struct platform_device *pdev) struct wm8753_priv *wm8753; int ret = 0; - info("WM8753 Audio Codec %s", WM8753_VERSION); + pr_info("WM8753 Audio Codec %s", WM8753_VERSION); setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index a7d25e2f2522..a1371b73ba7d 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -33,25 +33,6 @@ #define AUDIO_NAME "wm8990" #define WM8990_VERSION "0.2" -/* - * Debug - */ - -#define WM8990_DEBUG 0 - -#ifdef WM8990_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif -#define err(format, arg...) \ - printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -#define info(format, arg...) \ - printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -#define warn(format, arg...) \ - printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) - /* codec private data */ struct wm8990_priv { unsigned int sysclk; @@ -1524,13 +1505,13 @@ static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind) ret = i2c_attach_client(i2c); if (ret < 0) { - err("failed to attach codec at addr %x\n", addr); + pr_err("failed to attach codec at addr %x\n", addr); goto err; } ret = wm8990_init(socdev); if (ret < 0) { - err("failed to initialise WM8990\n"); + pr_err("failed to initialise WM8990\n"); goto err; } return ret; @@ -1579,7 +1560,7 @@ static int wm8990_probe(struct platform_device *pdev) struct wm8990_priv *wm8990; int ret = 0; - info("WM8990 Audio Codec %s\n", WM8990_VERSION); + pr_info("WM8990 Audio Codec %s\n", WM8990_VERSION); setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); From e2be2ccf9416bb4e0eb5f851967e79261f41d7e5 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 25 Jun 2008 14:42:07 +0300 Subject: [PATCH 121/179] ALSA: ASoC: Add support for generic DAPM register modifier widget This generic register modifier widget is for updating multiple codec register bits at once when the widget changes its power state. Signed-off-by: Jarkko Nikula Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc-dapm.h | 14 ++++++++++++++ sound/soc/soc-dapm.c | 19 +++++++++++++++++++ 2 files changed, 33 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index f8223fae5804..b2849538cbfe 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -130,6 +130,13 @@ { .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert} +/* generic register modifier widget */ +#define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ +{ .id = wid, .name = wname, .kcontrols = NULL, .num_kcontrols = 0, \ + .reg = -((wreg) + 1), .shift = wshift, .mask = wmask, \ + .on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \ + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD} + /* dapm kcontrol types */ #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -227,6 +234,10 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); +/* event handler for register modifier widget - used by the soc-dapm */ +int dapm_reg_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); + /* dapm audio endpoint control */ int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, char *pin, int status); @@ -298,6 +309,9 @@ struct snd_soc_dapm_widget { unsigned char shift; /* bits to shift */ unsigned int saved_value; /* widget saved value */ unsigned int value; /* widget current value */ + unsigned int mask; /* non-shifted mask */ + unsigned int on_val; /* on state value */ + unsigned int off_val; /* off state value */ unsigned char power:1; /* block power status */ unsigned char invert:1; /* invert the power bit */ unsigned char active:1; /* active stream on DAC, ADC's */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 728f3ac2f304..25363829e605 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -442,6 +442,25 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return con; } +/* + * Handler for generic register modifier widget. + */ +int dapm_reg_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + unsigned int val; + + if (SND_SOC_DAPM_EVENT_ON(event)) + val = w->on_val; + else + val = w->off_val; + + snd_soc_update_bits(w->codec, -(w->reg + 1), + w->mask << w->shift, val << w->shift); + + return 0; +} + /* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- From 0bd72a3d49fc9fcac0202e4a2f394c7ee82af914 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 25 Jun 2008 14:42:08 +0300 Subject: [PATCH 122/179] ALSA: ASoC: TLV320AIC3X: Use register modifier widget for mic bias Signed-off-by: Jarkko Nikula Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 528c26acdbe2..0c7452f932e7 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -456,10 +456,12 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { &aic3x_right_line2_mux_controls), /* Mic Bias */ - SND_SOC_DAPM_MICBIAS("Mic Bias 2V", MICBIAS_CTRL, 6, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias 2.5V", MICBIAS_CTRL, 7, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 6, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V", + MICBIAS_CTRL, 6, 3, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2.5V", + MICBIAS_CTRL, 6, 3, 2, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias AVDD", + MICBIAS_CTRL, 6, 3, 3, 0), /* Left PGA to Left Output bypass */ SND_SOC_DAPM_MIXER("Left PGA Bypass Mixer", SND_SOC_NOPM, 0, 0, From 81971a14947b344ee7f6e3781f7ef36d8996f57a Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 25 Jun 2008 14:58:45 +0300 Subject: [PATCH 123/179] ALSA: ASoC: TLV320AIC3X: Modify only interface related bits in aic3x_set_dai_fmt Those two serial data interface control register bits have also other functions and they can be set before aic3x_set_dai_fmt is called. Signed-off-by: Jarkko Nikula Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 0c7452f932e7..29dc0ec3dd23 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -809,8 +809,10 @@ static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = codec->private_data; - u8 iface_areg = 0; - u8 iface_breg = 0; + u8 iface_areg, iface_breg; + + iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; + iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { From ee15ffdb140b1a4fda5f6a3a8b9994031de5edb4 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 25 Jun 2008 14:58:46 +0300 Subject: [PATCH 124/179] ALSA: ASoC: TLV320AIC3X: Add support for digital microphone input AIC33 and AIC34 codecs in TLV320AIC3x family support digital microphone input. When enabled, the codec ADC takes bitstream input to low-pass filter from GPIO2 instead of its own delta-sigma modulator while providing oversampling clock through GPIO1. Signed-off-by: Jarkko Nikula Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 29dc0ec3dd23..4f0bf26f170b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -455,6 +455,27 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line2_mux_controls), + /* + * Not a real mic bias widget but similar function. This is for dynamic + * control of GPIO1 digital mic modulator clock output function when + * using digital mic. + */ + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "GPIO1 dmic modclk", + AIC3X_GPIO1_REG, 4, 0xf, + AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK, + AIC3X_GPIO1_FUNC_DISABLED), + + /* + * Also similar function like mic bias. Selects digital mic with + * configurable oversampling rate instead of ADC converter. + */ + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 128", + AIC3X_ASD_INTF_CTRLA, 0, 3, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 64", + AIC3X_ASD_INTF_CTRLA, 0, 3, 2, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32", + AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), + /* Mic Bias */ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V", MICBIAS_CTRL, 6, 3, 1, 0), @@ -570,6 +591,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, {"Left ADC", NULL, "Left PGA Mixer"}, + {"Left ADC", NULL, "GPIO1 dmic modclk"}, /* Right Input */ {"Right Line1R Mux", "single-ended", "LINE1R"}, @@ -583,6 +605,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Right ADC", NULL, "Right PGA Mixer"}, + {"Right ADC", NULL, "GPIO1 dmic modclk"}, /* Left PGA Bypass */ {"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"}, @@ -643,6 +666,14 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line Out", NULL, "Right Line2 Bypass Mixer"}, {"Mono Out", NULL, "Right Line2 Bypass Mixer"}, {"Right HP Out", NULL, "Right Line2 Bypass Mixer"}, + + /* + * Logical path between digital mic enable and GPIO1 modulator clock + * output function + */ + {"GPIO1 dmic modclk", NULL, "DMic Rate 128"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 64"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 32"}, }; static int aic3x_add_widgets(struct snd_soc_codec *codec) From 90b9e47671d561fdf2802aac4ac749b6c86405e0 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 25 Jun 2008 14:58:47 +0300 Subject: [PATCH 125/179] ALSA: ASoC: Add digital mic configuration to N810 machine driver Signed-off-by: Jarkko Nikula Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/omap/n810.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c168a64e3162..767b39f339a8 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -46,11 +46,13 @@ static struct clk *func96m_clk; static int n810_spk_func; static int n810_jack_func; +static int n810_dmic_func; static void n810_ext_control(struct snd_soc_codec *codec) { snd_soc_dapm_set_endpoint(codec, "Ext Spk", n810_spk_func); snd_soc_dapm_set_endpoint(codec, "Headphone Jack", n810_jack_func); + snd_soc_dapm_set_endpoint(codec, "DMic", n810_dmic_func); snd_soc_dapm_sync_endpoints(codec); } @@ -150,6 +152,28 @@ static int n810_set_jack(struct snd_kcontrol *kcontrol, return 1; } +static int n810_get_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = n810_dmic_func; + + return 0; +} + +static int n810_set_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (n810_dmic_func == ucontrol->value.integer.value[0]) + return 0; + + n810_dmic_func = ucontrol->value.integer.value[0]; + n810_ext_control(codec); + + return 1; +} + static int n810_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { @@ -175,6 +199,7 @@ static int n810_jack_event(struct snd_soc_dapm_widget *w, static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event), SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event), + SND_SOC_DAPM_MIC("DMic", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -183,13 +208,18 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "LLOUT"}, {"Ext Spk", NULL, "RLOUT"}, + + {"DMic Rate 64", NULL, "Mic Bias 2V"}, + {"Mic Bias 2V", NULL, "DMic"}, }; static const char *spk_function[] = {"Off", "On"}; static const char *jack_function[] = {"Off", "Headphone"}; +static const char *input_function[] = {"ADC", "Digital Mic"}; static const struct soc_enum n810_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function), }; static const struct snd_kcontrol_new aic33_n810_controls[] = { @@ -197,6 +227,8 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = { n810_get_spk, n810_set_spk), SOC_ENUM_EXT("Jack Function", n810_enum[1], n810_get_jack, n810_set_jack), + SOC_ENUM_EXT("Input Select", n810_enum[2], + n810_get_input, n810_set_input), }; static int n810_aic33_init(struct snd_soc_codec *codec) @@ -248,6 +280,8 @@ static struct snd_soc_machine snd_soc_machine_n810 = { /* Audio private data */ static struct aic3x_setup_data n810_aic33_setup = { .i2c_address = 0x18, + .gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED, + .gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT, }; /* Audio subsystem */ From c7e0757a461db87157bbf92c4f5afda87cbe05f6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Jun 2008 14:42:51 +0200 Subject: [PATCH 126/179] ALSA: hda - Add MacBook 3.1 support MacBook 3.1 is compatible with model=mbp3. Added the codec SSID 10b6:3600. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4edde6094cd1..c7d4bbb01798 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6519,8 +6519,9 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b1000: /* iMac 24 */ board_config = ALC885_IMAC24; break; - case 0x106b00a1: /* Macbook */ + case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ case 0x106b2c00: /* Macbook Pro rev3 */ + case 0x106b3600: /* Macbook 3.1 */ board_config = ALC885_MBP3; break; default: From 4a3b6983232cd296ea260e06461d031f10065d63 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jun 2008 17:17:00 +0200 Subject: [PATCH 127/179] ALSA: ymfpci - fix initial volume for 44.1kHz output YDSXGR_BUF441OUTVOL register isn't initialized properly. This may lead to a silent output at full volume of unchanged "Wave Playback Volume". http://bugzilla.kernel.org/show_bug.cgi?id=10963 Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ymfpci/ymfpci_main.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 29b3056c5109..7129df5f315b 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -2205,6 +2205,7 @@ static int __devinit snd_ymfpci_memalloc(struct snd_ymfpci *chip) for (reg = 0x80; reg < 0xc0; reg += 4) snd_ymfpci_writel(chip, reg, 0); snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0x3fff3fff); + snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0x3fff3fff); snd_ymfpci_writel(chip, YDSXGR_ZVOUTVOL, 0x3fff3fff); snd_ymfpci_writel(chip, YDSXGR_SPDIFOUTVOL, 0x3fff3fff); snd_ymfpci_writel(chip, YDSXGR_NATIVEADCINVOL, 0x3fff3fff); @@ -2324,6 +2325,7 @@ int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state) chip->saved_regs[i] = snd_ymfpci_readl(chip, saved_regs_index[i]); chip->saved_ydsxgr_mode = snd_ymfpci_readl(chip, YDSXGR_MODE); snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0); + snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0); snd_ymfpci_disable_dsp(chip); pci_disable_device(pci); pci_save_state(pci); From 031005f78c8d0aebc17ddf7a34af9ffd48034d7d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Jun 2008 14:49:58 +0200 Subject: [PATCH 128/179] ALSA: hda - disable amp override on non-HP machines Some machines with Cx5045 seem to have no problem with the volume over 0dB on NID 0x17. Disable amp overrides for other devices but HP. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e3df8ac37076..64df8100593d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1005,15 +1005,19 @@ static int patch_cxt5045(struct hda_codec *codec) #endif } - /* - * Fix max PCM level to 0 dB - * (originall it has 0x2b steps with 0dB offset 0x14) - */ - snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, - (0x14 << AC_AMPCAP_OFFSET_SHIFT) | - (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); + switch (codec->subsystem_id >> 16) { + case 0x103c: + /* HP laptop has a really bad sound over 0dB on NID 0x17. + * Fix max PCM level to 0 dB + * (originall it has 0x2b steps with 0dB offset 0x14) + */ + snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, + (0x14 << AC_AMPCAP_OFFSET_SHIFT) | + (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + break; + } return 0; } From 69776bf3a7263a4e47751f475d540b0f73bf8337 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 27 Jun 2008 10:11:58 +0200 Subject: [PATCH 129/179] ALSA: Release v1.0.17rc3 Signed-off-by: Jaroslav Kysela --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/version.h b/include/sound/version.h index 1802cc593333..29a5b227312d 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.17rc2" +#define CONFIG_SND_VERSION "1.0.17rc3" #define CONFIG_SND_DATE "" From 942fd1ebf6072fdaa07ae6d77a4f58b39c1bfdf9 Mon Sep 17 00:00:00 2001 From: Walter Sheets Date: Fri, 27 Jun 2008 11:53:31 +0200 Subject: [PATCH 130/179] ALSA: via82xx - Add VIA audio device #1841 to ac97_quirk list Signed-off-by: Walter Sheets Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/via82xx.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index b585cc3e4c47..6781be9e3078 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1756,6 +1756,12 @@ static struct ac97_quirk ac97_quirks[] = { .name = "ECS L7VMM2", .type = AC97_TUNE_HP_ONLY }, + { + .subvendor = 0x1019, + .subdevice = 0x1841, + .name = "ECS K7VTA3", + .type = AC97_TUNE_HP_ONLY + }, { .subvendor = 0x1849, .subdevice = 0x3059, From 6cabf6b0694951288520512e0b68c78523acc9fd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Jun 2008 11:55:20 +0200 Subject: [PATCH 131/179] ALSA: Fix AC97 power down Some laptops don't like PR3 powerdown. Do PR3 powerdown only for the real power-saving. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_codec.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 2d2f16e11082..07364c00768a 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2295,9 +2295,11 @@ static void snd_ac97_powerdown(struct snd_ac97 *ac97) power |= AC97_PD_PR0 | AC97_PD_PR1; /* ADC & DAC powerdown */ snd_ac97_write(ac97, AC97_POWERDOWN, power); udelay(100); - power |= AC97_PD_PR2 | AC97_PD_PR3; /* Analog Mixer powerdown */ + power |= AC97_PD_PR2; /* Analog Mixer powerdown (Vref on) */ snd_ac97_write(ac97, AC97_POWERDOWN, power); if (ac97_is_power_save_mode(ac97)) { + power |= AC97_PD_PR3; /* Analog Mixer powerdown */ + snd_ac97_write(ac97, AC97_POWERDOWN, power); udelay(100); /* AC-link powerdown, internal Clk disable */ /* FIXME: this may cause click noises on some boards */ From 4d20f70a787e03530786152ea27a6ed157205cdc Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 27 Jun 2008 14:07:57 +0300 Subject: [PATCH 132/179] ALSA: ASoC: TLV320AIC3X: Add mixer control for ADC highpass filter Signed-off-by: Jarkko Nikula Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 6 ++++++ sound/soc/codecs/tlv320aic3x.h | 2 ++ 2 files changed, 8 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 4f0bf26f170b..d13830623db1 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -223,6 +223,8 @@ static const char *aic3x_right_hpcom_mux[] = { "differential of HPROUT", "constant VCM", "single-ended", "differential of HPLCOM", "external feedback" }; static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; +static const char *aic3x_adc_hpf[] = + { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; #define LDAC_ENUM 0 #define RDAC_ENUM 1 @@ -232,6 +234,7 @@ static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; #define LINE1R_ENUM 5 #define LINE2L_ENUM 6 #define LINE2R_ENUM 7 +#define ADC_HPF_ENUM 8 static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), @@ -242,6 +245,7 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; static const struct snd_kcontrol_new aic3x_snd_controls[] = { @@ -292,6 +296,8 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Input */ SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0), SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), + + SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; /* add non dapm controls */ diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index c1dd1ac0ceac..e6009461063f 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -37,6 +37,8 @@ #define AIC3X_ASD_INTF_CTRLB 9 /* Audio overflow status and PLL R value programming register */ #define AIC3X_OVRF_STATUS_AND_PLLR_REG 11 +/* Audio codec digital filter control register */ +#define AIC3X_CODEC_DFILT_CTRL 12 /* ADC PGA Gain control registers */ #define LADC_VOL 15 From 0701e0640f4a1de2b509cfee508216275f81d812 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Jun 2008 16:30:57 +0200 Subject: [PATCH 133/179] ALSA: ALSA: hda - Fix ALC883 medion model ALC883 medion model doesn't unmute the proper amps so no output can be heard. Replaced the mute switches to behave just like other models. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c7d4bbb01798..081d1a557065 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7050,13 +7050,13 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { static struct snd_kcontrol_new alc883_fivestack_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), From 470eaf6be78424fc499a5039e5d5fe58bace2bc3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Jun 2008 16:40:10 +0200 Subject: [PATCH 134/179] ALSA: hda - Add missing Thinkpad Z60m support Added the missing SSID of Thinkpad Z60m for model=thinkpad with AD1981HD. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 05ca027fcc01..e8003d99f0bf 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1616,6 +1616,7 @@ static const char *ad1981_models[AD1981_MODELS] = { static struct snd_pci_quirk ad1981_cfg_tbl[] = { SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD), + SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD), /* All HP models */ SND_PCI_QUIRK(0x103c, 0, "HP nx", AD1981_HP), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA), From 6a9dccd61253f361760802d8d19c28fa83ea83f5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 1 Jul 2008 14:52:05 +0200 Subject: [PATCH 135/179] ALSA: ALSA: hda - Fix speaker output on Toshiba P105 Toshiba Satellite P105 with cx5045 has no HP pin but only a speaker pin and does the speaker-muting on hardware. Thus the matching model is laptop-micsense. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 64df8100593d..95e3367d8879 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -906,6 +906,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530), SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P105", CXT5045_LAPTOP_MICSENSE), SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP_HPMICSENSE), From 25552e87aa4102f048f1a8a8ddc87c96593c304e Mon Sep 17 00:00:00 2001 From: Mike Rapoport Date: Wed, 2 Jul 2008 13:19:23 +0300 Subject: [PATCH 136/179] ALSA: ac97 - fix patch_ucb1400 for proper resume Replace 'snd_ac97_write' with snd_ac97_write_cache' in pacth_ucb1400 to allow proper codec wakeup. Signed-off-by: Mike Rapoport Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_patch.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index abe88adec37e..0746e9ccc20b 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3710,7 +3710,7 @@ static int patch_ucb1400(struct snd_ac97 * ac97) { ac97->build_ops = &patch_ucb1400_ops; /* enable headphone driver and smart low power mode by default */ - snd_ac97_write(ac97, 0x6a, 0x0050); - snd_ac97_write(ac97, 0x6c, 0x0030); + snd_ac97_write_cache(ac97, 0x6a, 0x0050); + snd_ac97_write_cache(ac97, 0x6c, 0x0030); return 0; } From 5415552d21150dbfbff012de2b59e685ea232b1e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Jul 2008 11:51:19 +0100 Subject: [PATCH 137/179] ALSA: ASoC: Advertise 16000Hz rate for WM9713 PCM interface Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm9713.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 0db963747369..a4806189044d 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1001,9 +1001,18 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); } -#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000) +#define WM9713_RATES (SNDRV_PCM_RATE_8000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define WM9713_PCM_RATES (SNDRV_PCM_RATE_8000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) #define WM9713_PCM_FORMATS \ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ @@ -1051,13 +1060,13 @@ struct snd_soc_codec_dai wm9713_dai[] = { .stream_name = "Voice Playback", .channels_min = 1, .channels_max = 1, - .rates = WM9713_RATES, + .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, .capture = { .stream_name = "Voice Capture", .channels_min = 1, .channels_max = 2, - .rates = WM9713_RATES, + .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, .ops = { .hw_params = wm9713_pcm_hw_params, From 15e4c72f520d1db9adc38ba157547a7c1fca45b2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Jul 2008 11:51:20 +0100 Subject: [PATCH 138/179] ALSA: ASoC: Make pop/click debug wait times dynamically configurable DAPM supports adding a compile time configurable delay to the widget power sequences, aiding diagnosis of problems with pops and clicks being generated during them. This patch converts this to be configurable at run time via a sysfs file. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-dapm.c | 67 +++++++++++++++++++++++++++++++++++--------- 1 file changed, 53 insertions(+), 14 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 25363829e605..7c2dd4ec8df1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -54,16 +54,6 @@ #define dbg(format, arg...) #endif -#define POP_DEBUG 0 -#if POP_DEBUG -#define POP_TIME 500 /* 500 msecs - change if pop debug is too fast */ -#define pop_wait(time) schedule_timeout_uninterruptible(msecs_to_jiffies(time)) -#define pop_dbg(format, arg...) printk(format, ## arg); pop_wait(POP_TIME) -#else -#define pop_dbg(format, arg...) -#define pop_wait(time) -#endif - /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic, @@ -80,6 +70,28 @@ static int dapm_status = 1; module_param(dapm_status, int, 0); MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries"); +static unsigned int pop_time; + +static void pop_wait(void) +{ + if (pop_time) + schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time)); +} + +static void pop_dbg(const char *fmt, ...) +{ + va_list args; + + va_start(args, fmt); + + if (pop_time) { + vprintk(fmt, args); + pop_wait(); + } + + va_end(args); +} + /* create a new dapm widget */ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( const struct snd_soc_dapm_widget *_widget) @@ -217,9 +229,9 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) change = old != new; if (change) { pop_dbg("pop test %s : %s in %d ms\n", widget->name, - widget->power ? "on" : "off", POP_TIME); + widget->power ? "on" : "off", pop_time); snd_soc_write(codec, widget->reg, new); - pop_wait(POP_TIME); + pop_wait(); } dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change); return change; @@ -803,20 +815,47 @@ static ssize_t dapm_widget_show(struct device *dev, static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); +/* pop/click delay times */ +static ssize_t dapm_pop_time_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + return sprintf(buf, "%d\n", pop_time); +} + +static ssize_t dapm_pop_time_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) + +{ + if (strict_strtoul(buf, 10, &pop_time) < 0) + printk(KERN_ERR "Unable to parse pop_time setting\n"); + + return count; +} + +static DEVICE_ATTR(dapm_pop_time, 0744, dapm_pop_time_show, + dapm_pop_time_store); + int snd_soc_dapm_sys_add(struct device *dev) { int ret = 0; - if (dapm_status) + if (dapm_status) { ret = device_create_file(dev, &dev_attr_dapm_widget); + if (ret == 0) + ret = device_create_file(dev, &dev_attr_dapm_pop_time); + } + return ret; } static void snd_soc_dapm_sys_remove(struct device *dev) { - if (dapm_status) + if (dapm_status) { + device_remove_file(dev, &dev_attr_dapm_pop_time); device_remove_file(dev, &dev_attr_dapm_widget); + } } /* free all dapm widgets and resources */ From 3f775987030cf7ff922765c18ddd0d311b4b46ef Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Thu, 3 Jul 2008 09:33:10 +0200 Subject: [PATCH 139/179] ALSA: ASoC: AC97 codec PM Simple suspend/resume for AC97 ASoC codec. Signed-off-by: Manuel Lauss Acked-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/ac97.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 300d007f2339..e4516f3ce64c 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -144,9 +144,34 @@ static int ac97_soc_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +static int ac97_soc_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_ac97_suspend(socdev->codec->ac97); + + return 0; +} + +static int ac97_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_ac97_resume(socdev->codec->ac97); + + return 0; +} +#else +#define ac97_soc_suspend NULL +#define ac97_soc_resume NULL +#endif + struct snd_soc_codec_device soc_codec_dev_ac97 = { .probe = ac97_soc_probe, .remove = ac97_soc_remove, + .suspend = ac97_soc_suspend, + .resume = ac97_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ac97); From 73ead4855d98e330f2840523e1648575b9f2aae9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 4 Jul 2008 16:01:14 +0100 Subject: [PATCH 140/179] ALSA: ASoC: Fix warning from strict_strtoul() Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-dapm.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7c2dd4ec8df1..91cbbefefb0a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -827,7 +827,11 @@ static ssize_t dapm_pop_time_store(struct device *dev, const char *buf, size_t count) { - if (strict_strtoul(buf, 10, &pop_time) < 0) + unsigned long val; + + if (strict_strtoul(buf, 10, &val) >= 0) + pop_time = val; + else printk(KERN_ERR "Unable to parse pop_time setting\n"); return count; From be321a890c25272965129ffe4b3b59a519fcf583 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Jul 2008 16:04:04 +0200 Subject: [PATCH 141/179] ALSA: hda - Add support for Lenovo 3000 N200 Added the model entry (model=lenovo) for Lenovo N3000 N200 laptop with ALC861-VD. Reference below: https://bugzilla.novell.com/show_bug.cgi?id=406425 Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 081d1a557065..92a709be3c4e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13004,6 +13004,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), + SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 N200", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG), {} }; From a5302181e5321664047f75715242aac4e0bbd17c Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 13:35:17 +0100 Subject: [PATCH 142/179] ALSA: asoc: core - refactored DAPM pin control API. Refactored snd_soc_dapm_set_endpoint() to snd_soc_dapm_enable_pin() and snd_soc_dapm_disable_pin(). Renamed snd_soc_dapm_sync_endpoints() to snd_soc_dapm_sync(). Renamed snd_soc_dapm_get_endpoint_status() to snd_soc_dapm_get_pin_status(). Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc-dapm.h | 15 ++-- sound/soc/at91/eti_b1_wm8731.c | 10 +-- sound/soc/codecs/tlv320aic3x.c | 4 +- sound/soc/davinci/davinci-evm.c | 16 ++-- sound/soc/omap/n810.c | 27 +++++-- sound/soc/pxa/corgi.c | 42 ++++++----- sound/soc/pxa/poodle.c | 24 +++--- sound/soc/pxa/spitz.c | 62 +++++++-------- sound/soc/pxa/tosa.c | 30 ++++---- sound/soc/s3c24xx/neo1973_wm8753.c | 116 ++++++++++++++--------------- sound/soc/sh/sh7760-ac97.c | 2 +- sound/soc/soc-dapm.c | 87 ++++++++++++++-------- 12 files changed, 233 insertions(+), 202 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index b2849538cbfe..3030fdc6981d 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -234,16 +234,11 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); -/* event handler for register modifier widget - used by the soc-dapm */ -int dapm_reg_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event); - -/* dapm audio endpoint control */ -int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, - char *pin, int status); -int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec, - char *pin); -int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec); +/* dapm audio pin control and status */ +int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_sync(struct snd_soc_codec *codec); /* dapm widget types */ enum snd_soc_dapm_type { diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 4a383a4a0ff1..ad971e7061fe 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -216,14 +216,14 @@ static int eti_b1_wm8731_init(struct snd_soc_codec *codec) snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); /* always connected */ - snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d13830623db1..954d39b7c040 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -29,7 +29,7 @@ * --------------------------------------- * * Hence the machine layer should disable unsupported inputs/outputs by - * snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0), etc. + * snd_soc_dapm_disable_pin(codec, "MONO_LOUT"), etc. */ #include @@ -206,7 +206,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, } if (found) - snd_soc_dapm_sync_endpoints(widget->codec); + snd_soc_dapm_sync(widget->codec); } ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 4c70a0ed3397..091eae3a9631 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -103,17 +103,17 @@ static int evm_aic3x_init(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); /* not connected */ - snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); - snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); - snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); + snd_soc_dapm_disable_pin(codec, "HPLCOM"); + snd_soc_dapm_disable_pin(codec, "HPRCOM"); /* always connected */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line In", 1); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 767b39f339a8..74f4599b4d7e 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -50,11 +50,22 @@ static int n810_dmic_func; static void n810_ext_control(struct snd_soc_codec *codec) { - snd_soc_dapm_set_endpoint(codec, "Ext Spk", n810_spk_func); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", n810_jack_func); - snd_soc_dapm_set_endpoint(codec, "DMic", n810_dmic_func); + if (n810_spk_func) + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); - snd_soc_dapm_sync_endpoints(codec); + if (n810_jack_func) + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + if (n810_dmic_func) + snd_soc_dapm_enable_pin(codec, "DMic"); + else + snd_soc_dapm_disable_pin(codec, "DMic); + + snd_soc_dapm_sync(codec); } static int n810_startup(struct snd_pcm_substream *substream) @@ -236,9 +247,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec) int i, err; /* Not connected */ - snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); - snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); - snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); + snd_soc_dapm_disable_pin(codec, "HPLCOM"); + snd_soc_dapm_disable_pin(codec, "HPRCOM"); /* Add N810 specific controls */ for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) { @@ -255,7 +266,7 @@ static int n810_aic33_init(struct snd_soc_codec *codec) /* Set up N810 specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index edeea63e80e8..db18ef68b691 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -50,47 +50,51 @@ static int corgi_spk_func; static void corgi_ext_control(struct snd_soc_codec *codec) { - int spk = 0, mic = 0, line = 0, hp = 0, hs = 0; - /* set up jack connection */ switch (corgi_jack_func) { case CORGI_HP: - hp = 1; /* set = unmute headphone */ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_MIC: - mic = 1; /* reset = mute headphone */ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_LINE: - line = 1; reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_HEADSET: - hs = 1; - mic = 1; reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); break; } if (corgi_spk_func == CORGI_SPK_ON) - spk = 1; - - /* set the enpoints to their new connetion states */ - snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", mic); - snd_soc_dapm_set_endpoint(codec, "Line Jack", line); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* signal a DAPM event */ - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int corgi_startup(struct snd_pcm_substream *substream) @@ -285,8 +289,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); /* Add corgi specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { @@ -303,7 +307,7 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) /* Set up corgi specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 810f1fe158ab..36cbf69f5f82 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -48,8 +48,6 @@ static int poodle_spk_func; static void poodle_ext_control(struct snd_soc_codec *codec) { - int spk = 0; - /* set up jack connection */ if (poodle_jack_func == POODLE_HP) { /* set = unmute headphone */ @@ -57,23 +55,23 @@ static void poodle_ext_control(struct snd_soc_codec *codec) POODLE_LOCOMO_GPIO_MUTE_L, 1); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 1); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); } else { locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_L, 0); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 0); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); } - if (poodle_spk_func == POODLE_SPK_ON) - spk = 1; - /* set the enpoints to their new connetion states */ - snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk); + if (poodle_spk_func == POODLE_SPK_ON) + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* signal a DAPM event */ - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int poodle_startup(struct snd_pcm_substream *substream) @@ -248,9 +246,9 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "MICIN", 1); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_enable_pin(codec, "MICIN"); /* Add poodle specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) { @@ -267,7 +265,7 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) /* Set up poodle specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 092b5c776b40..ec18163fdddd 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -51,60 +51,60 @@ static int spitz_spk_func; static void spitz_ext_control(struct snd_soc_codec *codec) { if (spitz_spk_func == SPITZ_SPK_ON) - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); else - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 0); + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* set up jack connection */ switch (spitz_jack_func) { case SPITZ_HP: /* enable and unmute hp jack, disable mic bias */ - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_HP_OFF: /* jack removed, everything off */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; } - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int spitz_startup(struct snd_pcm_substream *substream) @@ -291,13 +291,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) int i, err; /* NC codec pins */ - snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0); - snd_soc_dapm_set_endpoint(codec, "LINPUT2", 0); - snd_soc_dapm_set_endpoint(codec, "RINPUT2", 0); - snd_soc_dapm_set_endpoint(codec, "LINPUT3", 0); - snd_soc_dapm_set_endpoint(codec, "RINPUT3", 0); - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "MONO", 0); + snd_soc_dapm_disable_pin(codec, "RINPUT1"); + snd_soc_dapm_disable_pin(codec, "LINPUT2"); + snd_soc_dapm_disable_pin(codec, "RINPUT2"); + snd_soc_dapm_disable_pin(codec, "LINPUT3"); + snd_soc_dapm_disable_pin(codec, "RINPUT3"); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "MONO"); /* Add spitz specific controls */ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { @@ -314,7 +314,7 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) /* Set up spitz specific audio paths */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 465ff0f458ef..dba7689c5080 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -52,29 +52,31 @@ static int tosa_spk_func; static void tosa_ext_control(struct snd_soc_codec *codec) { - int spk = 0, mic_int = 0, hp = 0, hs = 0; - /* set up jack connection */ switch (tosa_jack_func) { case TOSA_HP: - hp = 1; + snd_soc_dapm_disable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case TOSA_MIC_INT: - mic_int = 1; + snd_soc_dapm_enable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case TOSA_HEADSET: - hs = 1; + snd_soc_dapm_disable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); break; } if (tosa_spk_func == TOSA_SPK_ON) - spk = 1; + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); - snd_soc_dapm_set_endpoint(codec, "Speaker", spk); - snd_soc_dapm_set_endpoint(codec, "Mic (Internal)", mic_int); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int tosa_startup(struct snd_pcm_substream *substream) @@ -191,8 +193,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "MONOOUT"); /* add tosa specific controls */ for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) { @@ -209,7 +211,7 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) /* set up tosa specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 34851238dea0..f053e85ff608 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -250,77 +250,77 @@ static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) switch (neo1973_scenario) { case NEO_AUDIO_OFF: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 1); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_HANDSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 1); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_HEADSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; default: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); } - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } @@ -511,12 +511,12 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) DBG("Entered %s\n", __func__); /* set up NC codec pins */ - snd_soc_dapm_set_endpoint(codec, "LOUT2", 0); - snd_soc_dapm_set_endpoint(codec, "ROUT2", 0); - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "OUT4", 0); - snd_soc_dapm_set_endpoint(codec, "LINE1", 0); - snd_soc_dapm_set_endpoint(codec, "LINE2", 0); + snd_soc_dapm_disable_pin(codec, "LOUT2"); + snd_soc_dapm_disable_pin(codec, "ROUT2"); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "OUT4"); + snd_soc_dapm_disable_pin(codec, "LINE1"); + snd_soc_dapm_disable_pin(codec, "LINE2"); /* set endpoints to default mode */ @@ -539,7 +539,7 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) err = snd_soc_dapm_add_routes(codec, dapm_routes, ARRAY_SIZE(dapm_routes)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 2f91de84c5c7..846d1b3a6305 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -25,7 +25,7 @@ extern struct snd_soc_platform sh7760_soc_platform; static int machine_init(struct snd_soc_codec *codec) { - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 91cbbefefb0a..94296b5dc58f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -880,8 +880,25 @@ static void dapm_free_widgets(struct snd_soc_codec *codec) } } +static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, + char *pin, int status) +{ + struct snd_soc_dapm_widget *w; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (!strcmp(w->name, pin)) { + dbg("dapm: %s: pin %s\n", codec->name, pin); + w->connected = status; + return 0; + } + } + + dbg("dapm: %s: configuring unknown pin %s\n", codec->name, pin); + return -EINVAL; +} + /** - * snd_soc_dapm_sync_endpoints - scan and power dapm paths + * snd_soc_dapm_sync - scan and power dapm paths * @codec: audio codec * * Walks all dapm audio paths and powers widgets according to their @@ -889,11 +906,11 @@ static void dapm_free_widgets(struct snd_soc_codec *codec) * * Returns 0 for success. */ -int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec) +int snd_soc_dapm_sync(struct snd_soc_codec *codec) { return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); } -EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints); +EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, const char *sink, const char *control, const char *source) @@ -1441,53 +1458,57 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, } /** - * snd_soc_dapm_set_endpoint - set audio endpoint status - * @codec: audio codec - * @endpoint: audio signal endpoint (or start point) - * @status: point status + * snd_soc_dapm_enable_pin - enable pin. + * @snd_soc_codec: SoC codec + * @pin: pin name * - * Set audio endpoint status - connected or disconnected. - * - * Returns 0 for success else error. + * Enables input/output pin and it's parents or children widgets iff there is + * a valid audio route and active audio stream. + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. */ -int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, - char *endpoint, int status) +int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin) { - struct snd_soc_dapm_widget *w; - - list_for_each_entry(w, &codec->dapm_widgets, list) { - if (!strcmp(w->name, endpoint)) { - w->connected = status; - return 0; - } - } - - return -ENODEV; + return snd_soc_dapm_set_pin(codec, pin, 1); } -EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint); +EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); /** - * snd_soc_dapm_get_endpoint_status - get audio endpoint status - * @codec: audio codec - * @endpoint: audio signal endpoint (or start point) + * snd_soc_dapm_disable_pin - disable pin. + * @codec: SoC codec + * @pin: pin name * - * Get audio endpoint status - connected or disconnected. - * - * Returns status + * Disables input/output pin and it's parents or children widgets. + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. */ -int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec, - char *endpoint) +int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin) +{ + return snd_soc_dapm_set_pin(codec, pin, 0); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); + +/** + * snd_soc_dapm_get_pin_status - get audio pin status + * @codec: audio codec + * @pin: audio signal pin endpoint (or start point) + * + * Get audio pin status - connected or disconnected. + * + * Returns 1 for connected otherwise 0. + */ +int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin) { struct snd_soc_dapm_widget *w; list_for_each_entry(w, &codec->dapm_widgets, list) { - if (!strcmp(w->name, endpoint)) + if (!strcmp(w->name, pin)) return w->connected; } return 0; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_get_endpoint_status); +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status); /** * snd_soc_dapm_free - free dapm resources From 3c4b266fe642bcaebe2b95edb56c3f8802924ff9 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:07:17 +0100 Subject: [PATCH 143/179] ALSA: asoc: core - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch series merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai in preparation for further ASoC v2 patches. This merger removes duplication in both DAI structures and simplifies the API for other users. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 71 ++++++++++++++------------------------------ sound/soc/soc-core.c | 50 +++++++++++++++---------------- 2 files changed, 48 insertions(+), 73 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 340223a8f24c..778e57e74dc8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -221,8 +221,7 @@ struct snd_soc_pcm_stream; struct snd_soc_ops; struct snd_soc_dai_mode; struct snd_soc_pcm_runtime; -struct snd_soc_codec_dai; -struct snd_soc_cpu_dai; +struct snd_soc_dai; struct snd_soc_codec; struct snd_soc_machine_config; struct soc_enum; @@ -317,50 +316,24 @@ struct snd_soc_ops { /* ASoC DAI ops */ struct snd_soc_dai_ops { /* DAI clocking configuration */ - int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai, + int (*set_sysclk)(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir); - int (*set_pll)(struct snd_soc_codec_dai *codec_dai, + int (*set_pll)(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out); - int (*set_clkdiv)(struct snd_soc_codec_dai *codec_dai, - int div_id, int div); + int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); /* DAI format configuration */ - int (*set_fmt)(struct snd_soc_codec_dai *codec_dai, - unsigned int fmt); - int (*set_tdm_slot)(struct snd_soc_codec_dai *codec_dai, + int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); + int (*set_tdm_slot)(struct snd_soc_dai *dai, unsigned int mask, int slots); - int (*set_tristate)(struct snd_soc_codec_dai *, int tristate); + int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* digital mute */ - int (*digital_mute)(struct snd_soc_codec_dai *, int mute); + int (*digital_mute)(struct snd_soc_dai *dai, int mute); }; -/* SoC Codec DAI */ -struct snd_soc_codec_dai { - char *name; - int id; - unsigned char type; - - /* DAI capabilities */ - struct snd_soc_pcm_stream playback; - struct snd_soc_pcm_stream capture; - - /* DAI runtime info */ - struct snd_soc_codec *codec; - unsigned int active; - unsigned char pop_wait:1; - - /* ops */ - struct snd_soc_ops ops; - struct snd_soc_dai_ops dai_ops; - - /* DAI private data */ - void *private_data; -}; - -/* SoC CPU DAI */ -struct snd_soc_cpu_dai { - +/* SoC DAI (Digital Audio Interface) */ +struct snd_soc_dai { /* DAI description */ char *name; unsigned int id; @@ -368,13 +341,13 @@ struct snd_soc_cpu_dai { /* DAI callbacks */ int (*probe)(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai); + struct snd_soc_dai *dai); void (*remove)(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai); + struct snd_soc_dai *dai); int (*suspend)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); int (*resume)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); /* ops */ struct snd_soc_ops ops; @@ -386,7 +359,9 @@ struct snd_soc_cpu_dai { /* DAI runtime info */ struct snd_pcm_runtime *runtime; - unsigned char active:1; + struct snd_soc_codec *codec; + unsigned int active; + unsigned char pop_wait:1; void *dma_data; /* DAI private data */ @@ -428,7 +403,7 @@ struct snd_soc_codec { struct delayed_work delayed_work; /* codec DAI's */ - struct snd_soc_codec_dai *dai; + struct snd_soc_dai *dai; unsigned int num_dai; }; @@ -447,12 +422,12 @@ struct snd_soc_platform { int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); int (*suspend)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); int (*resume)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); /* pcm creation and destruction */ - int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, + int (*pcm_new)(struct snd_card *, struct snd_soc_dai *, struct snd_pcm *); void (*pcm_free)(struct snd_pcm *); @@ -466,8 +441,8 @@ struct snd_soc_dai_link { char *stream_name; /* Stream name */ /* DAI */ - struct snd_soc_codec_dai *codec_dai; - struct snd_soc_cpu_dai *cpu_dai; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; /* machine stream operations */ struct snd_soc_ops *ops; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bdbbc6a980fa..4d626b47b2ff 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -134,8 +134,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); @@ -272,7 +272,7 @@ static void close_delayed_work(struct work_struct *work) struct snd_soc_device *socdev = container_of(work, struct snd_soc_device, delayed_work.work); struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_codec_dai *codec_dai; + struct snd_soc_dai *codec_dai; int i; mutex_lock(&pcm_mutex); @@ -323,8 +323,8 @@ static int soc_codec_close(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; mutex_lock(&pcm_mutex); @@ -384,8 +384,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; int ret = 0; @@ -489,8 +489,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); @@ -559,8 +559,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; mutex_lock(&pcm_mutex); @@ -594,8 +594,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret; if (codec_dai->ops.trigger) { @@ -651,7 +651,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* mute any active DAC's */ for (i = 0; i < machine->num_links; i++) { - struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; + struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 1); } @@ -664,7 +664,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) machine->suspend_pre(pdev, state); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); if (platform->suspend) @@ -690,7 +690,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) codec_dev->suspend(pdev, state); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); } @@ -726,7 +726,7 @@ static void soc_resume_deferred(struct work_struct *work) machine->resume_pre(pdev); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); } @@ -747,13 +747,13 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for (i = 0; i < machine->num_links; i++) { - struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; + struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 0); } for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); if (platform->resume) @@ -803,7 +803,7 @@ static int soc_probe(struct platform_device *pdev) } for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->probe) { ret = cpu_dai->probe(pdev, cpu_dai); if (ret < 0) @@ -838,7 +838,7 @@ platform_err: cpu_dai_err: for (i--; i >= 0; i--) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) cpu_dai->remove(pdev, cpu_dai); } @@ -867,7 +867,7 @@ static int soc_remove(struct platform_device *pdev) codec_dev->remove(pdev); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) cpu_dai->remove(pdev, cpu_dai); } @@ -895,8 +895,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, struct snd_soc_dai_link *dai_link, int num) { struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_codec_dai *codec_dai = dai_link->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = dai_link->cpu_dai; + struct snd_soc_dai *codec_dai = dai_link->codec_dai; + struct snd_soc_dai *cpu_dai = dai_link->cpu_dai; struct snd_soc_pcm_runtime *rtd; struct snd_pcm *pcm; char new_name[64]; @@ -1211,7 +1211,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->codec; #ifdef CONFIG_SND_SOC_AC97_BUS - struct snd_soc_codec_dai *codec_dai; + struct snd_soc_dai *codec_dai; int i; #endif From 453ba20b2800bc07a3eaf220010dc0127cc552fd Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:07:29 +0100 Subject: [PATCH 144/179] ALSA: asoc: at32 - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch series merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for AT32 platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/at32/at32-pcm.c | 6 +++--- sound/soc/at32/at32-ssc.c | 12 ++++++------ sound/soc/at32/at32-ssc.h | 2 +- sound/soc/at32/playpaq_wm8510.c | 6 +++--- 4 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c index cf76e89fae6e..435f1daf177c 100644 --- a/sound/soc/at32/at32-pcm.c +++ b/sound/soc/at32/at32-pcm.c @@ -366,7 +366,7 @@ static struct snd_pcm_ops at32_pcm_ops = { static u64 at32_pcm_dmamask = 0xffffffff; static int at32_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -422,7 +422,7 @@ static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) #ifdef CONFIG_PM static int at32_pcm_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at32_runtime_data *prtd; @@ -447,7 +447,7 @@ static int at32_pcm_suspend(struct platform_device *pdev, static int at32_pcm_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at32_runtime_data *prtd; diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c index 0ca44107f183..4ef6492c902e 100644 --- a/sound/soc/at32/at32-ssc.c +++ b/sound/soc/at32/at32-ssc.c @@ -336,7 +336,7 @@ static void at32_ssc_shutdown(struct snd_pcm_substream *substream) /* * Set the SSC system clock rate */ -static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { /* TODO: What the heck do I do here? */ @@ -348,7 +348,7 @@ static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Record DAI format for use by hw_params() */ -static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -362,7 +362,7 @@ static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, /* * Record SSC clock dividers for use in hw_params() */ -static int at32_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -670,7 +670,7 @@ static int at32_ssc_prepare(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int at32_ssc_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p; @@ -699,7 +699,7 @@ static int at32_ssc_suspend(struct platform_device *pdev, static int at32_ssc_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p; u32 cr; @@ -746,7 +746,7 @@ static int at32_ssc_resume(struct platform_device *pdev, SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) -struct snd_soc_cpu_dai at32_ssc_dai[NUM_SSC_DEVICES] = { +struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = { { .name = "at32-ssc0", .id = 0, diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h index 3c6901a6c036..3c052dbbe460 100644 --- a/sound/soc/at32/at32-ssc.h +++ b/sound/soc/at32/at32-ssc.h @@ -52,7 +52,7 @@ struct at32_ssc_info { #define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ -extern struct snd_soc_cpu_dai at32_ssc_dai[]; +extern struct snd_soc_dai at32_ssc_dai[]; diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index d6b9fd5e6123..18ac7d7391c1 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -82,7 +82,7 @@ static struct clk *_pll0; #if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( struct snd_pcm_hw_params *params, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p = cpu_dai->private_data; struct ssc_device *ssc = ssc_p->ssc; @@ -132,8 +132,8 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct at32_ssc_info *ssc_p = cpu_dai->private_data; struct ssc_device *ssc = ssc_p->ssc; unsigned int pll_out = 0, bclk = 0, mclk_div = 0; From d37ae539a1d76da8fe5a939ce8b6d818501c8716 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:07:37 +0100 Subject: [PATCH 145/179] ALSA: asoc: at91 - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the AT91 platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/at91/at91-pcm.c | 6 +++--- sound/soc/at91/at91-ssc.c | 12 ++++++------ sound/soc/at91/at91-ssc.h | 2 +- sound/soc/at91/eti_b1_wm8731.c | 8 ++++---- 4 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c index ccac6bd2889c..d47492b2b6e5 100644 --- a/sound/soc/at91/at91-pcm.c +++ b/sound/soc/at91/at91-pcm.c @@ -318,7 +318,7 @@ static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, static u64 at91_pcm_dmamask = 0xffffffff; static int at91_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -367,7 +367,7 @@ static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm) #ifdef CONFIG_PM static int at91_pcm_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at91_runtime_data *prtd; @@ -392,7 +392,7 @@ static int at91_pcm_suspend(struct platform_device *pdev, } static int at91_pcm_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at91_runtime_data *prtd; diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index bc35d00a38f8..c3625b665c5a 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -281,7 +281,7 @@ static void at91_ssc_shutdown(struct snd_pcm_substream *substream) /* * Record the SSC system clock rate. */ -static int at91_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { /* @@ -303,7 +303,7 @@ static int at91_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Record the DAI format for use in hw_params(). */ -static int at91_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -315,7 +315,7 @@ static int at91_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, /* * Record SSC clock dividers for use in hw_params(). */ -static int at91_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -634,7 +634,7 @@ static int at91_ssc_prepare(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int at91_ssc_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at91_ssc_info *ssc_p; @@ -662,7 +662,7 @@ static int at91_ssc_suspend(struct platform_device *pdev, } static int at91_ssc_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at91_ssc_info *ssc_p; @@ -700,7 +700,7 @@ static int at91_ssc_resume(struct platform_device *pdev, #define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_cpu_dai at91_ssc_dai[NUM_SSC_DEVICES] = { +struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = { { .name = "at91-ssc0", .id = 0, .type = SND_SOC_DAI_PCM, diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h index b188f973df9f..6b7bf382d06f 100644 --- a/sound/soc/at91/at91-ssc.h +++ b/sound/soc/at91/at91-ssc.h @@ -21,7 +21,7 @@ #define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ #define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ -extern struct snd_soc_cpu_dai at91_ssc_dai[]; +extern struct snd_soc_dai at91_ssc_dai[]; #endif /* _AT91_SSC_H */ diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index ad971e7061fe..9d4213963c26 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -53,8 +53,8 @@ static struct clk *pllb_clk; static int eti_b1_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* cpu clock is the AT91 master clock sent to the SSC */ @@ -87,8 +87,8 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; #ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE From 9cb132d743cf39b3bbe4288e9035217e7237a0bb Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:07:42 +0100 Subject: [PATCH 146/179] ALSA: asoc: davinci - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the DaVinci platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/davinci/davinci-evm.c | 4 ++-- sound/soc/davinci/davinci-i2s.c | 14 +++++++------- sound/soc/davinci/davinci-i2s.h | 2 +- sound/soc/davinci/davinci-pcm.c | 2 +- 4 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 091eae3a9631..4249e6a85743 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -33,8 +33,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret = 0; /* set codec DAI configuration */ diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index c3b545ccff72..5ebf1ff71c4c 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -147,7 +147,7 @@ static void davinci_mcbsp_stop(struct snd_pcm_substream *substream) static int davinci_i2s_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; cpu_dai->dma_data = dev->dma_params[substream->stream]; @@ -155,7 +155,7 @@ static int davinci_i2s_startup(struct snd_pcm_substream *substream) return 0; } -static int davinci_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct davinci_mcbsp_dev *dev = cpu_dai->private_data; @@ -296,11 +296,11 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) } static int davinci_i2s_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea; struct evm_snd_platform_data *pdata; @@ -358,11 +358,11 @@ err_release_region: } static void davinci_i2s_remove(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev = cpu_dai->private_data; struct resource *mem; @@ -378,7 +378,7 @@ static void davinci_i2s_remove(struct platform_device *pdev, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 -struct snd_soc_cpu_dai davinci_i2s_dai = { +struct snd_soc_dai davinci_i2s_dai = { .name = "davinci-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h index 9592d17db320..c5b091807eec 100644 --- a/sound/soc/davinci/davinci-i2s.h +++ b/sound/soc/davinci/davinci-i2s.h @@ -12,6 +12,6 @@ #ifndef _DAVINCI_I2S_H #define _DAVINCI_I2S_H -extern struct snd_soc_cpu_dai davinci_i2s_dai; +extern struct snd_soc_dai davinci_i2s_dai; #endif diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 6a76927c9971..6a5e56a782bb 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -350,7 +350,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) static u64 davinci_pcm_dmamask = 0xffffffff; static int davinci_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret; From e550e17ffeb8cf8db27724eaf2ad05f77388afb9 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:07:52 +0100 Subject: [PATCH 147/179] ALSA: asoc: codecs - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the codec drivers. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ac97.h | 2 +- sound/soc/codecs/ak4535.c | 8 ++++---- sound/soc/codecs/ak4535.h | 2 +- sound/soc/codecs/cs4270.c | 8 ++++---- sound/soc/codecs/cs4270.h | 2 +- sound/soc/codecs/tlv320aic3x.c | 8 ++++---- sound/soc/codecs/tlv320aic3x.h | 2 +- sound/soc/codecs/uda1380.c | 6 +++--- sound/soc/codecs/uda1380.h | 2 +- sound/soc/codecs/wm8510.c | 10 +++++----- sound/soc/codecs/wm8510.h | 2 +- sound/soc/codecs/wm8731.c | 8 ++++---- sound/soc/codecs/wm8731.h | 2 +- sound/soc/codecs/wm8750.c | 8 ++++---- sound/soc/codecs/wm8750.h | 2 +- sound/soc/codecs/wm8753.c | 28 ++++++++++++++-------------- sound/soc/codecs/wm8753.h | 2 +- sound/soc/codecs/wm8990.c | 12 ++++++------ sound/soc/codecs/wm8990.h | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9712.h | 2 +- sound/soc/codecs/wm9713.c | 10 +++++----- sound/soc/codecs/wm9713.h | 2 +- 24 files changed, 67 insertions(+), 67 deletions(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index e4516f3ce64c..61fd96ca7bc7 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -41,7 +41,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai ac97_dai = { +struct snd_soc_dai ac97_dai = { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97, .playback = { diff --git a/sound/soc/codecs/ac97.h b/sound/soc/codecs/ac97.h index 2bf6d69fd069..281aa42e2bbb 100644 --- a/sound/soc/codecs/ac97.h +++ b/sound/soc/codecs/ac97.h @@ -14,6 +14,6 @@ #define __LINUX_SND_SOC_AC97_H extern struct snd_soc_codec_device soc_codec_dev_ac97; -extern struct snd_soc_codec_dai ac97_dai; +extern struct snd_soc_dai ac97_dai; #endif diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 469266e881d8..b26003c4f3e8 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -329,7 +329,7 @@ static int ak4535_add_widgets(struct snd_soc_codec *codec) return 0; } -static int ak4535_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -369,7 +369,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -394,7 +394,7 @@ static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int ak4535_mute(struct snd_soc_codec_dai *dai, int mute) +static int ak4535_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf; @@ -436,7 +436,7 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai ak4535_dai = { +struct snd_soc_dai ak4535_dai = { .name = "AK4535", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h index fc686ddf753a..e9fe30e2c056 100644 --- a/sound/soc/codecs/ak4535.h +++ b/sound/soc/codecs/ak4535.h @@ -40,7 +40,7 @@ struct ak4535_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai ak4535_dai; +extern struct snd_soc_dai ak4535_dai; extern struct snd_soc_codec_device soc_codec_dev_ak4535; #endif diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e73fcfd9f5cd..9deb8c74fdfd 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -201,7 +201,7 @@ static struct { * driver what the input settings can be. This would need to be implemented * for stand-alone mode to work. */ -static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -251,7 +251,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, * data for playback only, but ASoC currently does not support different * formats for playback vs. record. */ -static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { struct snd_soc_codec *codec = codec_dai->codec; @@ -471,7 +471,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, * board does not have the MUTEA or MUTEB pins connected to such circuitry, * then this function will do nothing. */ -static int cs4270_mute(struct snd_soc_codec_dai *dai, int mute) +static int cs4270_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; int reg6; @@ -667,7 +667,7 @@ error: #endif /* USE_I2C*/ -struct snd_soc_codec_dai cs4270_dai = { +struct snd_soc_dai cs4270_dai = { .name = "CS4270", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/cs4270.h b/sound/soc/codecs/cs4270.h index 0ced49b7804d..adc6cd9667d4 100644 --- a/sound/soc/codecs/cs4270.h +++ b/sound/soc/codecs/cs4270.h @@ -16,7 +16,7 @@ * The ASoC codec DAI structure for the CS4270. Assign this structure to * the .codec_dai field of your machine driver's snd_soc_dai_link structure. */ -extern struct snd_soc_codec_dai cs4270_dai; +extern struct snd_soc_dai cs4270_dai; /* * The ASoC codec device structure for the CS4270. Assign this structure diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 954d39b7c040..b1dce5f459db 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -814,7 +814,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, return 0; } -static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute) +static int aic3x_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u8 ldac_reg = aic3x_read_reg_cache(codec, LDAC_VOL) & ~MUTE_ON; @@ -831,7 +831,7 @@ static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -841,7 +841,7 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, return 0; } -static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -990,7 +990,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected); #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_codec_dai aic3x_dai = { +struct snd_soc_dai aic3x_dai = { .name = "aic3x", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index e6009461063f..d76c079b86e7 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -228,7 +228,7 @@ struct aic3x_setup_data { unsigned int gpio_func[2]; }; -extern struct snd_soc_codec_dai aic3x_dai; +extern struct snd_soc_dai aic3x_dai; extern struct snd_soc_codec_device soc_codec_dev_aic3x; #endif /* _AIC3X_H */ diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 6d5335b14d51..a52d6d9e007a 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -372,7 +372,7 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec) return 0; } -static int uda1380_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -499,7 +499,7 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream) uda1380_write(codec, UDA1380_CLK, clk); } -static int uda1380_mute(struct snd_soc_codec_dai *codec_dai, int mute) +static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM; @@ -542,7 +542,7 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai uda1380_dai[] = { +struct snd_soc_dai uda1380_dai[] = { { .name = "UDA1380", .playback = { diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h index f9d885c8bf01..50c603e2c9f2 100644 --- a/sound/soc/codecs/uda1380.h +++ b/sound/soc/codecs/uda1380.h @@ -83,7 +83,7 @@ struct uda1380_setup_data { #define UDA1380_DAI_PLAYBACK 1 /* playback DAI */ #define UDA1380_DAI_CAPTURE 2 /* capture DAI */ -extern struct snd_soc_codec_dai uda1380_dai[3]; +extern struct snd_soc_dai uda1380_dai[3]; extern struct snd_soc_codec_device soc_codec_dev_uda1380; #endif /* _UDA1380_H */ diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index b549f6753aba..67325fd95447 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -332,7 +332,7 @@ static void pll_factors(unsigned int target, unsigned int source) pll_div.k = K; } -static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; @@ -368,7 +368,7 @@ static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai, /* * Configure WM8510 clock dividers. */ -static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -402,7 +402,7 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8510_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -510,7 +510,7 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8510_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8510_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; @@ -554,7 +554,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, #define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_codec_dai wm8510_dai = { +struct snd_soc_dai wm8510_dai = { .name = "WM8510 HiFi", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h index c862e7b7d530..f5d2e42eb3f4 100644 --- a/sound/soc/codecs/wm8510.h +++ b/sound/soc/codecs/wm8510.h @@ -97,7 +97,7 @@ struct wm8510_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai wm8510_dai; +extern struct snd_soc_dai wm8510_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8510; #endif diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 3ff42ad65ede..369d39c3f745 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -318,7 +318,7 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream) } } -static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8731_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7; @@ -330,7 +330,7 @@ static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -349,7 +349,7 @@ static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, } -static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -443,7 +443,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -struct snd_soc_codec_dai wm8731_dai = { +struct snd_soc_dai wm8731_dai = { .name = "WM8731", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h index 5bcab6a7afb4..99f2e3c60e33 100644 --- a/sound/soc/codecs/wm8731.h +++ b/sound/soc/codecs/wm8731.h @@ -38,7 +38,7 @@ struct wm8731_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai wm8731_dai; +extern struct snd_soc_dai wm8731_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8731; #endif diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index eb460c9aa63e..e23cb09f0d14 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -536,7 +536,7 @@ static inline int get_coeff(int mclk, int rate) return -EINVAL; } -static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8750_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -554,7 +554,7 @@ static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, return -EINVAL; } -static int wm8750_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -647,7 +647,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8750_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7; @@ -692,7 +692,7 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec, #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -struct snd_soc_codec_dai wm8750_dai = { +struct snd_soc_dai wm8750_dai = { .name = "WM8750", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/wm8750.h b/sound/soc/codecs/wm8750.h index a97a54a6348e..8ef30e628b21 100644 --- a/sound/soc/codecs/wm8750.h +++ b/sound/soc/codecs/wm8750.h @@ -61,7 +61,7 @@ struct wm8750_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai wm8750_dai; +extern struct snd_soc_dai wm8750_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8750; #endif diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index be01a738f184..8604809f0c36 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -740,7 +740,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8753_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { u16 reg, enable; @@ -863,7 +863,7 @@ static int get_coeff(int mclk, int rate) /* * Clock after PLL and dividers */ -static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -890,7 +890,7 @@ static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, /* * Set's ADC and Voice DAC format. */ -static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -960,7 +960,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, /* * Set's PCM dai fmt and BCLK. */ -static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1026,7 +1026,7 @@ static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1054,7 +1054,7 @@ static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, /* * Set's HiFi DAC format. */ -static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1087,7 +1087,7 @@ static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, /* * Set's I2S DAI format. */ -static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1195,7 +1195,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1210,7 +1210,7 @@ static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_pcm_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0) @@ -1218,7 +1218,7 @@ static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1233,7 +1233,7 @@ static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1250,7 +1250,7 @@ static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8753_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7; @@ -1316,7 +1316,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, * 3. Voice disabled - HIFI over HIFI * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture */ -static const struct snd_soc_codec_dai wm8753_all_dai[] = { +static const struct snd_soc_dai wm8753_all_dai[] = { /* DAI HiFi mode 1 */ { .name = "WM8753 HiFi", .id = 1, @@ -1456,7 +1456,7 @@ static const struct snd_soc_codec_dai wm8753_all_dai[] = { }, }; -struct snd_soc_codec_dai wm8753_dai[2]; +struct snd_soc_dai wm8753_dai[2]; EXPORT_SYMBOL_GPL(wm8753_dai); static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode) diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index 95e2a1f53169..44f5f1ff0cc7 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -120,7 +120,7 @@ struct wm8753_setup_data { #define WM8753_DAI_HIFI 0 #define WM8753_DAI_VOICE 1 -extern struct snd_soc_codec_dai wm8753_dai[2]; +extern struct snd_soc_dai wm8753_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_wm8753; #endif diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index a1371b73ba7d..3ecce5168e94 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1029,7 +1029,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8990_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { u16 reg; @@ -1065,7 +1065,7 @@ static int wm8990_set_dai_pll(struct snd_soc_codec_dai *codec_dai, /* * Clock after PLL and dividers */ -static int wm8990_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1078,7 +1078,7 @@ static int wm8990_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, /* * Set's ADC and Voice DAC format. */ -static int wm8990_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1131,7 +1131,7 @@ static int wm8990_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8990_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1196,7 +1196,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8990_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8990_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 val; @@ -1329,7 +1329,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, * 1. ADC/DAC on Primary Interface * 2. ADC on Primary Interface/DAC on secondary */ -struct snd_soc_codec_dai wm8990_dai = { +struct snd_soc_dai wm8990_dai = { /* ADC/DAC on primary */ .name = "WM8990 ADC/DAC Primary", .id = 1, diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h index bf9f8823dfc9..6bea57485283 100644 --- a/sound/soc/codecs/wm8990.h +++ b/sound/soc/codecs/wm8990.h @@ -825,7 +825,7 @@ struct wm8990_setup_data { #define WM8990_ADCCLK_DIV 2 #define WM8990_BCLK_DIV 3 -extern struct snd_soc_codec_dai wm8990_dai; +extern struct snd_soc_dai wm8990_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8990; #endif /* __WM8990REGISTERDEFS_H__ */ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 47390113bd0e..9fc8edd82225 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -532,7 +532,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai wm9712_dai[] = { +struct snd_soc_dai wm9712_dai[] = { { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97_BUS, diff --git a/sound/soc/codecs/wm9712.h b/sound/soc/codecs/wm9712.h index 719105d61e65..d29e8a18ca6d 100644 --- a/sound/soc/codecs/wm9712.h +++ b/sound/soc/codecs/wm9712.h @@ -8,7 +8,7 @@ #define WM9712_DAI_AC97_HIFI 0 #define WM9712_DAI_AC97_AUX 1 -extern struct snd_soc_codec_dai wm9712_dai[2]; +extern struct snd_soc_dai wm9712_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_wm9712; #endif diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index a4806189044d..38d1fe0971fc 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -789,7 +789,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, return 0; } -static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; @@ -800,7 +800,7 @@ static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai, * Tristate the PCM DAI lines, tristate can be disabled by calling * wm9713_set_dai_fmt() */ -static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai, int tristate) { struct snd_soc_codec *codec = codec_dai->codec; @@ -816,7 +816,7 @@ static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai, * Configure WM9713 clock dividers. * Voice DAC needs 256 FS */ -static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -858,7 +858,7 @@ static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1018,7 +1018,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) -struct snd_soc_codec_dai wm9713_dai[] = { +struct snd_soc_dai wm9713_dai[] = { { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97_BUS, diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h index d357b6c8134b..63b8d81756e3 100644 --- a/sound/soc/codecs/wm9713.h +++ b/sound/soc/codecs/wm9713.h @@ -46,7 +46,7 @@ #define WM9713_DAI_PCM_VOICE 2 extern struct snd_soc_codec_device soc_codec_dev_wm9713; -extern struct snd_soc_codec_dai wm9713_dai[3]; +extern struct snd_soc_dai wm9713_dai[3]; int wm9713_reset(struct snd_soc_codec *codec, int try_warm); From 8cf7b2b393b26b9ba1ccbeeb271a88e51d58f3e3 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:08:00 +0100 Subject: [PATCH 148/179] ALSA: asoc: fsl - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the Freescale PPC platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_dma.h | 2 +- sound/soc/fsl/fsl_ssi.c | 24 ++++++++++++------------ sound/soc/fsl/fsl_ssi.h | 4 ++-- sound/soc/fsl/mpc8610_hpcd.c | 4 ++-- 5 files changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 78de7168d2ba..da2bc5902864 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -282,7 +282,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) * once for each .dai_link in the machine driver's snd_soc_machine * structure. */ -static int fsl_dma_new(struct snd_card *card, struct snd_soc_codec_dai *dai, +static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { static u64 fsl_dma_dmamask = DMA_BIT_MASK(32); diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h index 430a6ce8b0d0..385d4a42603c 100644 --- a/sound/soc/fsl/fsl_dma.h +++ b/sound/soc/fsl/fsl_dma.h @@ -126,7 +126,7 @@ struct fsl_dma_link_descriptor { u8 res[4]; /* Reserved */ } __attribute__ ((aligned(32), packed)); -/* DMA information needed to create a snd_soc_cpu_dai object +/* DMA information needed to create a snd_soc_dai object * * ssi_stx_phys: bus address of SSI STX register to use * ssi_srx_phys: bus address of SSI SRX register to use diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f588545698f3..71bff33f5528 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -82,7 +82,7 @@ struct fsl_ssi_private { struct device *dev; unsigned int playback; unsigned int capture; - struct snd_soc_cpu_dai cpu_dai; + struct snd_soc_dai cpu_dai; struct device_attribute dev_attr; struct { @@ -479,7 +479,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) * @freq: the frequency of the given clock ID, currently ignored * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master) */ -static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int fsl_ssi_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -497,7 +497,7 @@ static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, * * @format: one of SND_SOC_DAIFMT_xxx */ -static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) +static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) { return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; } @@ -505,7 +505,7 @@ static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) /** * fsl_ssi_dai_template: template CPU DAI for the SSI */ -static struct snd_soc_cpu_dai fsl_ssi_dai_template = { +static struct snd_soc_dai fsl_ssi_dai_template = { .playback = { /* The SSI does not support monaural audio. */ .channels_min = 2, @@ -569,15 +569,15 @@ static ssize_t fsl_sysfs_ssi_show(struct device *dev, } /** - * fsl_ssi_create_dai: create a snd_soc_cpu_dai structure + * fsl_ssi_create_dai: create a snd_soc_dai structure * - * This function is called by the machine driver to create a snd_soc_cpu_dai + * This function is called by the machine driver to create a snd_soc_dai * structure. The function creates an ssi_private object, which contains - * the snd_soc_cpu_dai. It also creates the sysfs statistics device. + * the snd_soc_dai. It also creates the sysfs statistics device. */ -struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) +struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) { - struct snd_soc_cpu_dai *fsl_ssi_dai; + struct snd_soc_dai *fsl_ssi_dai; struct fsl_ssi_private *ssi_private; int ret = 0; struct device_attribute *dev_attr; @@ -588,7 +588,7 @@ struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) return NULL; } memcpy(&ssi_private->cpu_dai, &fsl_ssi_dai_template, - sizeof(struct snd_soc_cpu_dai)); + sizeof(struct snd_soc_dai)); fsl_ssi_dai = &ssi_private->cpu_dai; dev_attr = &ssi_private->dev_attr; @@ -623,11 +623,11 @@ struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) EXPORT_SYMBOL_GPL(fsl_ssi_create_dai); /** - * fsl_ssi_destroy_dai: destroy the snd_soc_cpu_dai object + * fsl_ssi_destroy_dai: destroy the snd_soc_dai object * * This function undoes the operations of fsl_ssi_create_dai() */ -void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai) +void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai) { struct fsl_ssi_private *ssi_private = container_of(fsl_ssi_dai, struct fsl_ssi_private, cpu_dai); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index c5ce88e15651..83b44d700e33 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -217,8 +217,8 @@ struct fsl_ssi_info { struct device *dev; }; -struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); -void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai); +struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); +void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai); #endif diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 8820c3f8d171..59d7e49bd661 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -96,8 +96,8 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct mpc8610_hpcd_data *machine_data = rtd->socdev->dev->platform_data; int ret = 0; From 8687eb8bded8c3c5842a85bd0c30e43fc5a3e0e0 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:08:07 +0100 Subject: [PATCH 149/179] ALSA: asoc: omap - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the Omap platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/omap/n810.c | 4 ++-- sound/soc/omap/omap-mcbsp.c | 16 ++++++++-------- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.c | 2 +- 4 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 74f4599b4d7e..d1233c01398f 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -86,8 +86,8 @@ static int n810_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int err; /* Set codec DAI configuration */ diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 40d87e6d0de8..00b0c9d73cd4 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -103,7 +103,7 @@ static const unsigned long omap2420_mcbsp_port[][2] = {}; static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); int err = 0; @@ -116,7 +116,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); if (!cpu_dai->active) { @@ -128,7 +128,7 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); int err = 0; @@ -157,7 +157,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; @@ -223,7 +223,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, * This must be called before _set_clkdiv and _set_sysclk since McBSP register * cache is initialized here */ -static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); @@ -292,7 +292,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, return 0; } -static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); @@ -347,7 +347,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, return 0; } -static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -376,7 +376,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, return err; } -struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS] = { +struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = { { .name = "omap-mcbsp-dai", .id = 0, diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 9965fd4b0427..ed8afb550671 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -44,6 +44,6 @@ enum omap_mcbsp_div { */ #define NUM_LINKS 1 -extern struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS]; +extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS]; #endif diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 62370202c649..e092f3d836d0 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -316,7 +316,7 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -int omap_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, +int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; From 917f93ac80a1c007d4a3ce269a3712f93a75728f Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:08:11 +0100 Subject: [PATCH 150/179] ALSA: asoc: pxa - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the PXA platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/pxa/corgi.c | 4 ++-- sound/soc/pxa/poodle.c | 4 ++-- sound/soc/pxa/pxa2xx-ac97.c | 14 +++++++------- sound/soc/pxa/pxa2xx-ac97.h | 2 +- sound/soc/pxa/pxa2xx-i2s.c | 14 +++++++------- sound/soc/pxa/pxa2xx-i2s.h | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 2 +- sound/soc/pxa/spitz.c | 4 ++-- 8 files changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index db18ef68b691..782afbf7ada5 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -123,8 +123,8 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 36cbf69f5f82..ce25b6bf340d 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -102,8 +102,8 @@ static int poodle_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index cb947956ed1a..b458667538cd 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -283,7 +283,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = { #ifdef CONFIG_PM static int pxa2xx_ac97_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { GCR |= GCR_ACLINK_OFF; clk_disable(ac97_clk); @@ -291,7 +291,7 @@ static int pxa2xx_ac97_suspend(struct platform_device *pdev, } static int pxa2xx_ac97_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); @@ -311,7 +311,7 @@ static int pxa2xx_ac97_resume(struct platform_device *pdev, #endif static int pxa2xx_ac97_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { int ret; @@ -373,7 +373,7 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out; @@ -387,7 +387,7 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out; @@ -401,7 +401,7 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; @@ -419,7 +419,7 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, * There is only 1 physical AC97 interface for pxa2xx, but it * has extra fifo's that can be used for aux DACs and ADCs. */ -struct snd_soc_cpu_dai pxa_ac97_dai[] = { +struct snd_soc_dai pxa_ac97_dai[] = { { .name = "pxa2xx-ac97", .id = 0, diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h index b8ccfee095c4..e390de8edcd4 100644 --- a/sound/soc/pxa/pxa2xx-ac97.h +++ b/sound/soc/pxa/pxa2xx-ac97.h @@ -14,7 +14,7 @@ #define PXA2XX_DAI_AC97_AUX 1 #define PXA2XX_DAI_AC97_MIC 2 -extern struct snd_soc_cpu_dai pxa_ac97_dai[3]; +extern struct snd_soc_dai pxa_ac97_dai[3]; /* platform data */ extern struct snd_ac97_bus_ops pxa2xx_ac97_ops; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 35090c2870ff..9c06553b9267 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -77,7 +77,7 @@ static struct pxa2xx_gpio gpio_bus[] = { static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (!cpu_dai->active) { SACR0 |= SACR0_RST; @@ -98,7 +98,7 @@ static int pxa_i2s_wait(void) return 0; } -static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { /* interface format */ @@ -124,7 +124,7 @@ static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, return 0; } -static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { if (clk_id != PXA2XX_I2S_SYSCLK) @@ -140,7 +140,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx); pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx); @@ -237,7 +237,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int pxa2xx_i2s_suspend(struct platform_device *dev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { if (!dai->active) return 0; @@ -255,7 +255,7 @@ static int pxa2xx_i2s_suspend(struct platform_device *dev, } static int pxa2xx_i2s_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { if (!dai->active) return 0; @@ -280,7 +280,7 @@ static int pxa2xx_i2s_resume(struct platform_device *pdev, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) -struct snd_soc_cpu_dai pxa_i2s_dai = { +struct snd_soc_dai pxa_i2s_dai = { .name = "pxa2xx-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/pxa/pxa2xx-i2s.h b/sound/soc/pxa/pxa2xx-i2s.h index 4435bd9f884f..e2def441153e 100644 --- a/sound/soc/pxa/pxa2xx-i2s.h +++ b/sound/soc/pxa/pxa2xx-i2s.h @@ -15,6 +15,6 @@ /* I2S clock */ #define PXA2XX_I2S_SYSCLK 0 -extern struct snd_soc_cpu_dai pxa_i2s_dai; +extern struct snd_soc_dai pxa_i2s_dai; #endif diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 01ad7bf716b7..2df03ee5819e 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -330,7 +330,7 @@ static void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK; -int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, +int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index ec18163fdddd..fd1abc7b08d8 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -121,8 +121,8 @@ static int spitz_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int clk = 0; int ret = 0; From 1992a6fbd929196aebe95e0e7b04c4da66c3bfec Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:08:24 +0100 Subject: [PATCH 151/179] ALSA: asoc: s3c24xx - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the S3C24xx platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 12 ++++++------ sound/soc/s3c24xx/s3c2412-i2s.c | 14 +++++++------- sound/soc/s3c24xx/s3c2412-i2s.h | 2 +- sound/soc/s3c24xx/s3c2443-ac97.c | 10 +++++----- sound/soc/s3c24xx/s3c24xx-ac97.h | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 14 +++++++------- sound/soc/s3c24xx/s3c24xx-i2s.h | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- 8 files changed, 29 insertions(+), 29 deletions(-) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index f053e85ff608..51a4ce3dbd19 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -66,8 +66,8 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int pll_out = 0, bclk = 0; int ret = 0; unsigned long iis_clkrate; @@ -156,7 +156,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; DBG("Entered %s\n", __func__); @@ -176,7 +176,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; unsigned int pcmdiv = 0; int ret = 0; unsigned long iis_clkrate; @@ -222,7 +222,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; DBG("Entered %s\n", __func__); @@ -546,7 +546,7 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) /* * BT Codec DAI */ -static struct snd_soc_cpu_dai bt_dai = { +static struct snd_soc_dai bt_dai = { .name = "Bluetooth", .id = 0, .type = SND_SOC_DAI_PCM, diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index c463a82dec3a..ee4676ed1283 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -295,7 +295,7 @@ static inline int s3c2412_snd_is_clkmaster(void) /* * Set S3C2412 I2S DAI format */ -static int s3c2412_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { u32 iismod; @@ -500,7 +500,7 @@ EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate); /* * Set S3C2412 Clock source */ -static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); @@ -528,7 +528,7 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Set S3C2412 Clock dividers */ -static int s3c2412_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; @@ -602,7 +602,7 @@ EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); static int s3c2412_i2s_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { DBG("Entered %s\n", __func__); @@ -648,7 +648,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev, #ifdef CONFIG_PM static int s3c2412_i2s_suspend(struct platform_device *dev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; u32 iismod; @@ -676,7 +676,7 @@ static int s3c2412_i2s_suspend(struct platform_device *dev, } static int s3c2412_i2s_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; @@ -708,7 +708,7 @@ static int s3c2412_i2s_resume(struct platform_device *pdev, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -struct snd_soc_cpu_dai s3c2412_i2s_dai = { +struct snd_soc_dai s3c2412_i2s_dai = { .name = "s3c2412-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h index 27f48e1ffa86..aac08a25e541 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.h +++ b/sound/soc/s3c24xx/s3c2412-i2s.h @@ -24,7 +24,7 @@ extern struct clk *s3c2412_get_iisclk(void); -extern struct snd_soc_cpu_dai s3c2412_i2s_dai; +extern struct snd_soc_dai s3c2412_i2s_dai; struct s3c2412_rate_calc { unsigned int clk_div; /* for prescaler */ diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 533565b61b2f..783349b7fede 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -210,7 +210,7 @@ static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { }; static int s3c2443_ac97_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { int ret; u32 ac_glbctrl; @@ -262,7 +262,7 @@ static int s3c2443_ac97_probe(struct platform_device *pdev, } static void s3c2443_ac97_remove(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { free_irq(IRQ_S3C244x_AC97, NULL); clk_disable(s3c24xx_ac97.ac97_clk); @@ -274,7 +274,7 @@ static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_out; @@ -316,7 +316,7 @@ static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; @@ -352,7 +352,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -struct snd_soc_cpu_dai s3c2443_ac97_dai[] = { +struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "s3c2443-ac97", .id = 0, diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h index bf03e8ed16c3..a96dcadf28b4 100644 --- a/sound/soc/s3c24xx/s3c24xx-ac97.h +++ b/sound/soc/s3c24xx/s3c24xx-ac97.h @@ -26,6 +26,6 @@ #define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97 #endif -extern struct snd_soc_cpu_dai s3c2443_ac97_dai[]; +extern struct snd_soc_dai s3c2443_ac97_dai[]; #endif /*S3C24XXAC97_H_*/ diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 42e96b5ff825..397524282b57 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -205,7 +205,7 @@ static inline int s3c24xx_snd_is_clkmaster(void) /* * Set S3C24xx I2S DAI format */ -static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { u32 iismod; @@ -313,7 +313,7 @@ exit_err: /* * Set S3C24xx Clock source */ -static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -339,7 +339,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Set S3C24xx Clock dividers */ -static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { u32 reg; @@ -378,7 +378,7 @@ u32 s3c24xx_i2s_get_clockrate(void) EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); static int s3c24xx_i2s_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { DBG("Entered %s\n", __func__); @@ -411,7 +411,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev, #ifdef CONFIG_PM static int s3c24xx_i2s_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { DBG("Entered %s\n", __func__); @@ -426,7 +426,7 @@ static int s3c24xx_i2s_suspend(struct platform_device *pdev, } static int s3c24xx_i2s_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { DBG("Entered %s\n", __func__); clk_enable(s3c24xx_i2s.iis_clk); @@ -449,7 +449,7 @@ static int s3c24xx_i2s_resume(struct platform_device *pdev, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -struct snd_soc_cpu_dai s3c24xx_i2s_dai = { +struct snd_soc_dai s3c24xx_i2s_dai = { .name = "s3c24xx-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.h b/sound/soc/s3c24xx/s3c24xx-i2s.h index 537b4ecce8a3..726d91cf4e1c 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.h +++ b/sound/soc/s3c24xx/s3c24xx-i2s.h @@ -32,6 +32,6 @@ u32 s3c24xx_i2s_get_clockrate(void); -extern struct snd_soc_cpu_dai s3c24xx_i2s_dai; +extern struct snd_soc_dai s3c24xx_i2s_dai; #endif /*S3C24XXI2S_H_*/ diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index ef599745159c..cef79b34dc6f 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -429,7 +429,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 s3c24xx_pcm_dmamask = DMA_32BIT_MASK; static int s3c24xx_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; From 53640650ffe07f6201ecae0b0dca078b97327247 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:08:29 +0100 Subject: [PATCH 152/179] ALSA: asoc: sh - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the SuperH platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/sh/dma-sh7760.c | 2 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/sh7760-ac97.c | 2 +- sound/soc/sh/ssi.c | 8 ++++---- 4 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 7a3ce80d6727..9faa12622d09 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -326,7 +326,7 @@ static void camelot_pcm_free(struct snd_pcm *pcm) } static int camelot_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, + struct snd_soc_dai *dai, struct snd_pcm *pcm) { /* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index b7b676b3d671..df7bc345c320 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -266,7 +266,7 @@ static int hac_hw_params(struct snd_pcm_substream *substream, #define AC97_FMTS \ SNDRV_PCM_FMTBIT_S16_LE -struct snd_soc_cpu_dai sh4_hac_dai[] = { +struct snd_soc_dai sh4_hac_dai[] = { { .name = "HAC0", .id = 0, diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 846d1b3a6305..92bfaf4774a7 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -20,7 +20,7 @@ #define IPSEL 0xFE400034 /* platform specific structs can be declared here */ -extern struct snd_soc_cpu_dai sh4_hac_dai[2]; +extern struct snd_soc_dai sh4_hac_dai[2]; extern struct snd_soc_platform sh7760_soc_platform; static int machine_init(struct snd_soc_codec *codec) diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 3388bc3d62d1..55c3464163ab 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -208,7 +208,7 @@ static int ssi_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, +static int ssi_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id]; @@ -222,7 +222,7 @@ static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, * This divider is used to generate the SSI_SCK (I2S bitclock) from the * clock at the HAC_BIT_CLK ("oversampling clock") pin. */ -static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) +static int ssi_set_clkdiv(struct snd_soc_dai *dai, int did, int div) { struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; unsigned long ssicr; @@ -245,7 +245,7 @@ static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) return 0; } -static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) +static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; unsigned long ssicr = SSIREG(SSICR); @@ -332,7 +332,7 @@ static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) -struct snd_soc_cpu_dai sh4_ssi_dai[] = { +struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI0", .id = 0, From c1286b8671135006eeb66b0267fa2895561b78e4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Jul 2008 19:26:03 +0100 Subject: [PATCH 153/179] ALSA: ASoC: Switch DAPM to use of standard DEBUG macro DAPM contains debug output controlled by a DAPM_DEBUG macro. Change this to be controlled by the standard DEBUG, dropping the custom dbg() macro as we go. Also fix the error printed when configuring an unknown pin to be an unconditionally displayed error rather than debug output. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-dapm.c | 22 ++++++++++------------ 1 file changed, 10 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 94296b5dc58f..d18ebc627861 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -45,13 +45,10 @@ #include /* debug */ -#define DAPM_DEBUG 0 -#if DAPM_DEBUG +#ifdef DEBUG #define dump_dapm(codec, action) dbg_dump_dapm(codec, action) -#define dbg(format, arg...) printk(format, ## arg) #else #define dump_dapm(codec, action) -#define dbg(format, arg...) #endif /* dapm power sequences - make this per codec in the future */ @@ -233,7 +230,8 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) snd_soc_write(codec, widget->reg, new); pop_wait(); } - dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change); + pr_debug("reg %x old %x new %x change %d\n", widget->reg, + old, new, change); return change; } @@ -591,8 +589,8 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) /* call any power change event handlers */ if (power_change) { if (w->event) { - dbg("power %s event for %s flags %x\n", - w->power ? "on" : "off", w->name, w->event_flags); + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", w->name, w->event_flags); if (power) { /* power up event */ if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { @@ -634,7 +632,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) return ret; } -#if DAPM_DEBUG +#ifdef DEBUG static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) { struct snd_soc_dapm_widget *w; @@ -887,13 +885,13 @@ static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, list_for_each_entry(w, &codec->dapm_widgets, list) { if (!strcmp(w->name, pin)) { - dbg("dapm: %s: pin %s\n", codec->name, pin); + pr_debug("dapm: %s: pin %s\n", codec->name, pin); w->connected = status; return 0; } } - dbg("dapm: %s: configuring unknown pin %s\n", codec->name, pin); + pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin); return -EINVAL; } @@ -1397,8 +1395,8 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, { if (!w->sname) continue; - dbg("widget %s\n %s stream %s event %d\n", w->name, w->sname, - stream, event); + pr_debug("widget %s\n %s stream %s event %d\n", + w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { switch(event) { case SND_SOC_DAPM_STREAM_START: From 3fccd8b1bf716f1314c76689a3554ea37ad282b6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Jul 2008 19:26:04 +0100 Subject: [PATCH 154/179] ALSA: ASoC: Dump DAPM state for non-stream changes Explicit DAPM syncs are likely to cause DAPM state updates, as are mixer and mux configuration changes, so display the DAPM status after them too. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-dapm.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d18ebc627861..2c87061c2a6b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -717,8 +717,10 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, path->connect = 0; /* old connection must be powered down */ } - if (found) + if (found) { dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); + dump_dapm(widget->codec, "mux power update"); + } return 0; } @@ -754,8 +756,10 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, break; } - if (found) + if (found) { dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); + dump_dapm(widget->codec, "mixer power update"); + } return 0; } @@ -906,7 +910,9 @@ static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, */ int snd_soc_dapm_sync(struct snd_soc_codec *codec) { - return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); + int ret = dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); + dump_dapm(codec, "sync"); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); From 7e2574050e6af203f0c94915c98c53ce1fc48044 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Tue, 8 Jul 2008 11:50:09 +0200 Subject: [PATCH 155/179] ALSA: hda - removed redundant gpio_mask An gpio_mask value was defined twice needlessly. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c4f3489376c5..a6d138831e26 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3669,7 +3669,6 @@ again: /* GPIO0 High = EAPD */ spec->gpio_mask = 0x01; spec->gpio_dir = 0x01; - spec->gpio_mask = 0x01; spec->gpio_data = 0x01; spec->mux_nids = stac92hd71bxx_mux_nids; From 8c6529dbf881303920a415c2d14a500218661949 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 8 Jul 2008 13:19:13 +0100 Subject: [PATCH 156/179] ALSA: asoc: core - add Digital Audio Interface (DAI) control functions. This patch adds several functions for DAI control and config and replaces the current method of calling function pointers within the DAI struct. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 21 +++++++ sound/soc/soc-core.c | 140 ++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 153 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 778e57e74dc8..1890d87c5204 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -256,6 +256,27 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); +/* Digital Audio Interface clocking API.*/ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir); + +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div); + +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out); + +/* Digital Audio interface formatting */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); + +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots); + +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); + +/* Digital Audio Interface mute */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); + /* *Controls */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4d626b47b2ff..83f1190293a8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -434,8 +434,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) else { codec_dai->pop_wait = 0; cancel_delayed_work(&socdev->delayed_work); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0); } } else { /* no delayed work - do we need to power up codec */ @@ -454,8 +453,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_START); snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0); } else { /* codec already powered - power on widgets */ @@ -467,8 +465,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(codec, codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + + snd_soc_dai_digital_mute(codec_dai, 0); } } @@ -566,8 +564,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock(&pcm_mutex); /* apply codec digital mute */ - if (!codec->active && codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 1); + if (!codec->active) + snd_soc_dai_digital_mute(codec_dai, 1); /* free any machine hw params */ if (machine->ops && machine->ops->hw_free) @@ -1703,6 +1701,132 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); +/** + * snd_soc_dai_set_sysclk - configure DAI system or master clock. + * @dai: DAI + * @clk_id: DAI specific clock ID + * @freq: new clock frequency in Hz + * @dir: new clock direction - input/output. + * + * Configures the DAI master (MCLK) or system (SYSCLK) clocking. + */ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); + +/** + * snd_soc_dai_set_clkdiv - configure DAI clock dividers. + * @dai: DAI + * @clk_id: DAI specific clock divider ID + * @div: new clock divisor. + * + * Configures the clock dividers. This is used to derive the best DAI bit and + * frame clocks from the system or master clock. It's best to set the DAI bit + * and frame clocks as low as possible to save system power. + */ +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div) +{ + if (dai->dai_ops.set_clkdiv) + return dai->dai_ops.set_clkdiv(dai, div_id, div); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); + +/** + * snd_soc_dai_set_pll - configure DAI PLL. + * @dai: DAI + * @pll_id: DAI specific PLL ID + * @freq_in: PLL input clock frequency in Hz + * @freq_out: requested PLL output clock frequency in Hz + * + * Configures and enables PLL to generate output clock based on input clock. + */ +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + if (dai->dai_ops.set_pll) + return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); + +/** + * snd_soc_dai_set_fmt - configure DAI hardware audio format. + * @dai: DAI + * @clk_id: DAI specific clock ID + * @fmt: SND_SOC_DAIFMT_ format value. + * + * Configures the DAI hardware format and clocking. + */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + if (dai->dai_ops.set_fmt) + return dai->dai_ops.set_fmt(dai, fmt); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); + +/** + * snd_soc_dai_set_tdm_slot - configure DAI TDM. + * @dai: DAI + * @mask: DAI specific mask representing used slots. + * @slots: Number of slots in use. + * + * Configures a DAI for TDM operation. Both mask and slots are codec and DAI + * specific. + */ +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_tdm_slot(dai, mask, slots); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); + +/** + * snd_soc_dai_set_tristate - configure DAI system or master clock. + * @dai: DAI + * @tristate: tristate enable + * + * Tristates the DAI so that others can use it. + */ +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_tristate(dai, tristate); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); + +/** + * snd_soc_dai_digital_mute - configure DAI system or master clock. + * @dai: DAI + * @mute: mute enable + * + * Mutes the DAI DAC. + */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) +{ + if (dai->dai_ops.digital_mute) + return dai->dai_ops.digital_mute(dai, mute); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); + static int __devinit snd_soc_init(void) { printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION); From 64105cfd65df74fdf82c1d053b2c9953304a94ea Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 8 Jul 2008 13:19:18 +0100 Subject: [PATCH 157/179] ALSA: asoc: machines - add Digital Audio Interface (DAI) control functions. This patch adds several functions for DAI control and config and replaces the current method of calling function pointers within the DAI struct within the machine drivers. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/at32/playpaq_wm8510.c | 16 ++++----- sound/soc/at91/eti_b1_wm8731.c | 18 +++++----- sound/soc/davinci/davinci-evm.c | 6 ++-- sound/soc/fsl/mpc8610_hpcd.c | 58 +++++++++++++----------------- sound/soc/omap/n810.c | 6 ++-- sound/soc/pxa/corgi.c | 8 ++--- sound/soc/pxa/poodle.c | 8 ++--- sound/soc/pxa/spitz.c | 8 ++--- sound/soc/s3c24xx/neo1973_wm8753.c | 26 +++++++------- 9 files changed, 71 insertions(+), 83 deletions(-) diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index 18ac7d7391c1..fd62f2569758 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -210,14 +210,14 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, /* * set CPU and CODEC DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, fmt); + ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { pr_warning("playpaq_wm8510: " "Failed to set CODEC DAI format (%d)\n", ret); return ret; } - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, fmt); + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); if (ret < 0) { pr_warning("playpaq_wm8510: " "Failed to set CPU DAI format (%d)\n", @@ -233,14 +233,13 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", cd.cmr_div, cd.period); - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, - AT32_SSC_CMR_DIV, cd.cmr_div); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", ret); return ret; } - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, cd.period); if (ret < 0) { pr_warning("playpaq_wm8510: " @@ -260,7 +259,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, #if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); if (ret < 0) { pr_warning ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", @@ -270,7 +269,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, #endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - ret = codec_dai->dai_ops.set_pll(codec_dai, 0, + ret = snd_soc_dai_set_pll(codec_dai, 0, clk_get_rate(CODEC_CLK), pll_out); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", @@ -279,8 +278,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, } - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, - WM8510_MCLKDIV, mclk_div); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", ret); diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 9d4213963c26..d532de954241 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -58,13 +58,13 @@ static int eti_b1_startup(struct snd_pcm_substream *substream) int ret; /* cpu clock is the AT91 master clock sent to the SSC */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, AT91_SYSCLK_MCK, + ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK, 60000000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* codec system clock is supplied by PCK1, set to 12MHz */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, 12000000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -96,13 +96,13 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, int cmr_div, period; /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; @@ -141,17 +141,17 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, } /* set the MCK divider for BCLK */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); if (ret < 0) return ret; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* set the BCLK divider for DACLRC */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_TCMR_PERIOD, period); } else { /* set the BCLK divider for ADCLRC */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_RCMR_PERIOD, period); } if (ret < 0) @@ -163,13 +163,13 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, */ /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 4249e6a85743..5e2c306399ed 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -38,19 +38,19 @@ static int evm_hw_params(struct snd_pcm_substream *substream, int ret = 0; /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF); if (ret < 0) return ret; /* set the codec system clock */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, + ret = snd_soc_dai_set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, SND_SOC_CLOCK_OUT); if (ret < 0) return ret; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 59d7e49bd661..4bdc9d8fc90e 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -103,55 +103,45 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) int ret = 0; /* Tell the CPU driver what the serial protocol is. */ - if (cpu_dai->dai_ops.set_fmt) { - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, - machine_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver audio format\n"); - return ret; - } + ret = snd_soc_dai_set_fmt(cpu_dai, machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver audio format\n"); + return ret; } /* Tell the codec driver what the serial protocol is. */ - if (codec_dai->dai_ops.set_fmt) { - ret = codec_dai->dai_ops.set_fmt(codec_dai, - machine_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver audio format\n"); - return ret; - } + ret = snd_soc_dai_set_fmt(codec_dai, machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver audio format\n"); + return ret; } /* * Tell the CPU driver what the clock frequency is, and whether it's a * slave or master. */ - if (cpu_dai->dai_ops.set_sysclk) { - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, 0, - machine_data->clk_frequency, - machine_data->cpu_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver clock parameters\n"); - return ret; - } + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, + machine_data->clk_frequency, + machine_data->cpu_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver clock parameters\n"); + return ret; } /* * Tell the codec driver what the MCLK frequency is, and whether it's * a slave or master. */ - if (codec_dai->dai_ops.set_sysclk) { - ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, - machine_data->clk_frequency, - machine_data->codec_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver clock params\n"); - return ret; - } + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + machine_data->clk_frequency, + machine_data->codec_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver clock params\n"); + return ret; } return 0; diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index d1233c01398f..e53c055412c7 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -91,7 +91,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, int err; /* Set codec DAI configuration */ - err = codec_dai->dai_ops.set_fmt(codec_dai, + err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); @@ -99,7 +99,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, return err; /* Set cpu DAI configuration */ - err = cpu_dai->dai_ops.set_fmt(cpu_dai, + err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); @@ -107,7 +107,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, return err; /* Set the codec system clock for DAC and ADC */ - err = codec_dai->dai_ops.set_sysclk(codec_dai, 0, 12000000, + err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN); return err; diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 782afbf7ada5..c0294464a23a 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -143,25 +143,25 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index ce25b6bf340d..65a4e9a8c39e 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -122,25 +122,25 @@ static int poodle_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index fd1abc7b08d8..64385797da5d 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -141,25 +141,25 @@ static int spitz_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8750_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 51a4ce3dbd19..4d7a9aa15f1a 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -108,44 +108,44 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set MCLK division for sample rate */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, S3C2410_IISMOD_32FS); if (ret < 0) return ret; /* set codec BCLK division for sample rate */ - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); if (ret < 0) return ret; /* set prescaler division for sample rate */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, S3C24XX_PRESCALE(4, 4)); if (ret < 0) return ret; /* codec PLL input is PCLK/4 */ - ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, iis_clkrate / 4, pll_out); if (ret < 0) return ret; @@ -161,7 +161,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) DBG("Entered %s\n", __func__); /* disable the PLL */ - return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); } /* @@ -194,24 +194,24 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, /* todo: gg check mode (DSP_B) against CSR datasheet */ /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set codec PCM division for sample rate */ - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); if (ret < 0) return ret; /* configue and enable PLL for 12.288MHz output */ - ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, iis_clkrate / 4, 12288000); if (ret < 0) return ret; @@ -227,7 +227,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) DBG("Entered %s\n", __func__); /* disable the PLL */ - return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); } static struct snd_soc_ops neo1973_voice_ops = { From ac8615baf279f9bcb90c8ae017321d63a94e1762 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 8 Jul 2008 13:20:39 +0100 Subject: [PATCH 158/179] ALSA: asoc: n810 - fix build error. This patch adds a missing " that was recently introduced (removed) in the DAI struct merge patch series. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/omap/n810.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index e53c055412c7..02cec96859b8 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -63,7 +63,7 @@ static void n810_ext_control(struct snd_soc_codec *codec) if (n810_dmic_func) snd_soc_dapm_enable_pin(codec, "DMic"); else - snd_soc_dapm_disable_pin(codec, "DMic); + snd_soc_dapm_disable_pin(codec, "DMic"); snd_soc_dapm_sync(codec); } From 6f48413dd0f28d81a20fba2e264dd1b9f205893e Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 8 Jul 2008 13:21:52 +0100 Subject: [PATCH 159/179] ALSA: asoc: at32 - DAI struct merge and enable_pin() change. This adds support for the recent DAI struct merge and new DAPM pin API. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/at32/playpaq_wm8510.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index fd62f2569758..fee5f8e58957 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -341,15 +341,15 @@ static int playpaq_wm8510_init(struct snd_soc_codec *codec) } - /* always connected endpoints */ - snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); - snd_soc_dapm_sync_endpoints(codec); + /* always connected pins */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_sync(codec); /* Make CSB show PLL rate */ - codec->dai->dai_ops.set_clkdiv(codec->dai, WM8510_OPCLKDIV, + snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, WM8510_OPCLKDIV_1 | 4); return 0; From bd5d06eb8f0b9e0eb117b9c45c254d577ec2e273 Mon Sep 17 00:00:00 2001 From: David Howells Date: Tue, 8 Jul 2008 17:30:30 +0100 Subject: [PATCH 160/179] ALSA: Fix a const pointer usage warning in the Digigram VX soundcard driver Fix a const pointer usage warning in the Digigram VX soundcard driver. A const pointer is being passed to copy_from_user() to load the firmware into. This is okay in this case because the function has allocated the firmware struct itself, but the const qualifier is part of the firmware struct - so the patch casts the const away. Signed-off-by: David Howells Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/drivers/vx/vx_hwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/vx/vx_hwdep.c b/sound/drivers/vx/vx_hwdep.c index 1dfe6948e6ff..efd22e92bced 100644 --- a/sound/drivers/vx/vx_hwdep.c +++ b/sound/drivers/vx/vx_hwdep.c @@ -183,7 +183,7 @@ static int vx_hwdep_dsp_load(struct snd_hwdep *hw, kfree(fw); return -ENOMEM; } - if (copy_from_user(fw->data, dsp->image, dsp->length)) { + if (copy_from_user((void *)fw->data, dsp->image, dsp->length)) { free_fw(fw); return -EFAULT; } From 7fc077fba5f8896c6fed3b35c5a10e7fdae82bbe Mon Sep 17 00:00:00 2001 From: David Howells Date: Tue, 8 Jul 2008 17:52:10 +0100 Subject: [PATCH 161/179] ALSA: Fix a const to non-const assignment in the Digigram VXpocket sound driver Fix a const to non-const pointer assignment warning in the Digigram VXpocket sound driver. This may be due to patch 0aa4937648b91e9e6d3879b2cbeaa5f0c9863ac0. Signed-off-by: David Howells Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pcmcia/vx/vxp_ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c index 157b0b539f39..99bf2a65a6f5 100644 --- a/sound/pcmcia/vx/vxp_ops.c +++ b/sound/pcmcia/vx/vxp_ops.c @@ -151,7 +151,7 @@ static int vxp_load_xilinx_binary(struct vx_core *_chip, const struct firmware * unsigned int i; int c; int regCSUER, regRUER; - unsigned char *image; + const unsigned char *image; unsigned char data; /* Switch to programmation mode */ From ee2da99782d717b264d5c309a2084ca85e9a64ff Mon Sep 17 00:00:00 2001 From: Johannes Berg Date: Wed, 9 Jul 2008 10:28:41 +0200 Subject: [PATCH 162/179] ALSA: remove CONFIG_KMOD from sound A bunch of things in alsa depend on CONFIG_KMOD, use CONFIG_MODULES instead where the dependency is needed at all. Signed-off-by: Johannes Berg Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/seq_kernel.h | 2 +- sound/core/seq/seq_clientmgr.c | 2 +- sound/core/seq/seq_device.c | 6 ++---- sound/core/sound.c | 8 ++++---- sound/core/timer.c | 6 +++--- sound/ppc/daca.c | 2 -- sound/ppc/tumbler.c | 2 -- 7 files changed, 11 insertions(+), 17 deletions(-) diff --git a/include/sound/seq_kernel.h b/include/sound/seq_kernel.h index f023c1b97f8c..3d9afb6a8c9c 100644 --- a/include/sound/seq_kernel.h +++ b/include/sound/seq_kernel.h @@ -105,7 +105,7 @@ int snd_seq_event_port_attach(int client, struct snd_seq_port_callback *pcbp, int cap, int type, int midi_channels, int midi_voices, char *portname); int snd_seq_event_port_detach(int client, int port); -#ifdef CONFIG_KMOD +#ifdef CONFIG_MODULES void snd_seq_autoload_lock(void); void snd_seq_autoload_unlock(void); #else diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 47cfa5186e34..7a1545d2d953 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -148,7 +148,7 @@ struct snd_seq_client *snd_seq_client_use_ptr(int clientid) return NULL; } spin_unlock_irqrestore(&clients_lock, flags); -#ifdef CONFIG_KMOD +#ifdef CONFIG_MODULES if (!in_interrupt()) { static char client_requested[SNDRV_SEQ_GLOBAL_CLIENTS]; static char card_requested[SNDRV_CARDS]; diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 2f00ad28a2b7..05410e536a4f 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -124,7 +124,7 @@ static void snd_seq_device_info(struct snd_info_entry *entry, * load all registered drivers (called from seq_clientmgr.c) */ -#ifdef CONFIG_KMOD +#ifdef CONFIG_MODULES /* avoid auto-loading during module_init() */ static int snd_seq_in_init; void snd_seq_autoload_lock(void) @@ -140,7 +140,7 @@ void snd_seq_autoload_unlock(void) void snd_seq_device_load_drivers(void) { -#ifdef CONFIG_KMOD +#ifdef CONFIG_MODULES struct ops_list *ops; /* Calling request_module during module_init() @@ -566,7 +566,5 @@ EXPORT_SYMBOL(snd_seq_device_load_drivers); EXPORT_SYMBOL(snd_seq_device_new); EXPORT_SYMBOL(snd_seq_device_register_driver); EXPORT_SYMBOL(snd_seq_device_unregister_driver); -#ifdef CONFIG_KMOD EXPORT_SYMBOL(snd_seq_autoload_lock); EXPORT_SYMBOL(snd_seq_autoload_unlock); -#endif diff --git a/sound/core/sound.c b/sound/core/sound.c index 6c8ab48c689a..09a94953745a 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -60,14 +60,14 @@ EXPORT_SYMBOL(snd_ecards_limit); static struct snd_minor *snd_minors[SNDRV_OS_MINORS]; static DEFINE_MUTEX(sound_mutex); -#ifdef CONFIG_KMOD +#ifdef CONFIG_MODULES /** * snd_request_card - try to load the card module * @card: the card number * * Tries to load the module "snd-card-X" for the given card number - * via KMOD. Returns immediately if already loaded. + * via request_module. Returns immediately if already loaded. */ void snd_request_card(int card) { @@ -92,7 +92,7 @@ static void snd_request_other(int minor) request_module(str); } -#endif /* request_module support */ +#endif /* modular kernel */ /** * snd_lookup_minor_data - get user data of a registered device @@ -132,7 +132,7 @@ static int snd_open(struct inode *inode, struct file *file) return -ENODEV; mptr = snd_minors[minor]; if (mptr == NULL) { -#ifdef CONFIG_KMOD +#ifdef CONFIG_MODULES int dev = SNDRV_MINOR_DEVICE(minor); if (dev == SNDRV_MINOR_CONTROL) { /* /dev/aloadC? */ diff --git a/sound/core/timer.c b/sound/core/timer.c index 9d8184a2c2d0..0af337efc64e 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -146,7 +146,7 @@ static struct snd_timer *snd_timer_find(struct snd_timer_id *tid) return NULL; } -#ifdef CONFIG_KMOD +#ifdef CONFIG_MODULES static void snd_timer_request(struct snd_timer_id *tid) { @@ -259,8 +259,8 @@ int snd_timer_open(struct snd_timer_instance **ti, /* open a master instance */ mutex_lock(®ister_mutex); timer = snd_timer_find(tid); -#ifdef CONFIG_KMOD - if (timer == NULL) { +#ifdef CONFIG_MODULES + if (!timer) { mutex_unlock(®ister_mutex); snd_timer_request(tid); mutex_lock(®ister_mutex); diff --git a/sound/ppc/daca.c b/sound/ppc/daca.c index ca9452901a50..8a5b29031933 100644 --- a/sound/ppc/daca.c +++ b/sound/ppc/daca.c @@ -249,9 +249,7 @@ int __init snd_pmac_daca_init(struct snd_pmac *chip) int i, err; struct pmac_daca *mix; -#ifdef CONFIG_KMOD request_module("i2c-powermac"); -#endif /* CONFIG_KMOD */ mix = kzalloc(sizeof(*mix), GFP_KERNEL); if (! mix) diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 3f8d7164cef9..009df8dd37a8 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1350,9 +1350,7 @@ int __init snd_pmac_tumbler_init(struct snd_pmac *chip) struct device_node *tas_node, *np; char *chipname; -#ifdef CONFIG_KMOD request_module("i2c-powermac"); -#endif /* CONFIG_KMOD */ mix = kzalloc(sizeof(*mix), GFP_KERNEL); if (! mix) From af13452820d075cb7153fe155ca796e64038d7e5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Jul 2008 19:13:30 +0200 Subject: [PATCH 163/179] ALSA: wavefront - add const Fix const to non-const pointer cast warning in wavefront_synth.c. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/isa/wavefront/wavefront_synth.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 95eeca163354..0bb9b9256601 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1939,7 +1939,7 @@ static int __devinit wavefront_download_firmware (snd_wavefront_t *dev, char *path) { - unsigned char *buf; + const unsigned char *buf; int len, err; int section_cnt_downloaded = 0; const struct firmware *firmware; From 76d39d0a1947fee4c29976e507fb93188eaa2b38 Mon Sep 17 00:00:00 2001 From: Dmitry Baryshkov Date: Tue, 8 Jul 2008 19:45:20 +0400 Subject: [PATCH 164/179] ALSA: tosa: fix compilation with new DAPM API Signed-off-by: Dmitry Baryshkov Acked-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/pxa/tosa.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index dba7689c5080..b6edb61a3a30 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -153,7 +153,7 @@ SND_SOC_DAPM_SPK("Speaker", NULL), }; /* tosa audio map */ -static const snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route audio_map[] = { /* headphone connected to HPOUTL, HPOUTR */ {"Headphone Jack", NULL, "HPOUTL"}, @@ -205,7 +205,7 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) } /* add tosa specific widgets */ - snd_soc_dapm_new_controls(codec, &tosa_dapm_widgets, + snd_soc_dapm_new_controls(codec, tosa_dapm_widgets, ARRAY_SIZE(tosa_dapm_widgets)); /* set up tosa specific audio path audio_map */ From a84534dc179d3e9f501e5a12598cc5271cd28f5b Mon Sep 17 00:00:00 2001 From: Dmitry Baryshkov Date: Tue, 8 Jul 2008 19:45:42 +0400 Subject: [PATCH 165/179] ALSA: ASoC: pxa2xx-ac97: fix warning due to missing argument in fuction declaration Signed-off-by: Dmitry Baryshkov Acked-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/pxa/pxa2xx-ac97.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index b458667538cd..059af815ea0c 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -356,7 +356,8 @@ static int pxa2xx_ac97_probe(struct platform_device *pdev, return ret; } -static void pxa2xx_ac97_remove(struct platform_device *pdev) +static void pxa2xx_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) { GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); From bf41534506a0572c06c8f34d12aa489be4c8780e Mon Sep 17 00:00:00 2001 From: Richard Chan Date: Wed, 9 Jul 2008 19:39:07 +0200 Subject: [PATCH 166/179] ALSA: Add Yamaha KX49 (USB MIDI controller) to usbquirks.h This patch is for the Yamaha USB MIDI controller KX49. http://www.yamahasynth.com/products/kx/index.html It has a 3-port MIDI interface and an HID interface (it has a tiny keyboard subset). Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/usb/usbquirks.h | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index b7ab3ee7647e..d675050210dc 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -210,6 +210,7 @@ YAMAHA_DEVICE(0x1042, NULL), YAMAHA_DEVICE(0x1043, NULL), YAMAHA_DEVICE(0x1044, NULL), YAMAHA_DEVICE(0x1045, NULL), +YAMAHA_DEVICE(0x104e, "KX49"), YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), From 4a161d235b68eb7234f40106560c488a1bdb3851 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Wed, 9 Jul 2008 16:27:56 +0200 Subject: [PATCH 167/179] ALSA: ASoC: Au12x0/Au1550 PSC Audio support Audio for Au12x0/Au1550 PSCs in AC97 and I2S mode, for ASoC v1 framework. - DBDMA, AC97 and I2S drivers - sample AC97 machine code (Db1200) Signed-off-by: Manuel Lauss Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/asm-mips/mach-au1x00/au1xxx_psc.h | 8 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/au1x/Kconfig | 32 ++ sound/soc/au1x/Makefile | 13 + sound/soc/au1x/dbdma2.c | 421 ++++++++++++++++++++++ sound/soc/au1x/psc-ac97.c | 387 ++++++++++++++++++++ sound/soc/au1x/psc-i2s.c | 414 +++++++++++++++++++++ sound/soc/au1x/psc.h | 53 +++ sound/soc/au1x/sample-ac97.c | 144 ++++++++ 10 files changed, 1474 insertions(+), 1 deletion(-) create mode 100644 sound/soc/au1x/Kconfig create mode 100644 sound/soc/au1x/Makefile create mode 100644 sound/soc/au1x/dbdma2.c create mode 100644 sound/soc/au1x/psc-ac97.c create mode 100644 sound/soc/au1x/psc-i2s.c create mode 100644 sound/soc/au1x/psc.h create mode 100644 sound/soc/au1x/sample-ac97.c diff --git a/include/asm-mips/mach-au1x00/au1xxx_psc.h b/include/asm-mips/mach-au1x00/au1xxx_psc.h index dae4eca2417e..892b7f168eb4 100644 --- a/include/asm-mips/mach-au1x00/au1xxx_psc.h +++ b/include/asm-mips/mach-au1x00/au1xxx_psc.h @@ -204,6 +204,14 @@ typedef struct psc_i2s { u32 psc_i2sudf; } psc_i2s_t; +#define PSC_I2SCFG_OFFSET 0x08 +#define PSC_I2SMASK_OFFSET 0x0C +#define PSC_I2SPCR_OFFSET 0x10 +#define PSC_I2SSTAT_OFFSET 0x14 +#define PSC_I2SEVENT_OFFSET 0x18 +#define PSC_I2SRXTX_OFFSET 0x1C +#define PSC_I2SUDF_OFFSET 0x20 + /* I2S Config Register. */ #define PSC_I2SCFG_RT_MASK (3 << 30) #define PSC_I2SCFG_RT_FIFO1 (0 << 30) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index b939e22db7b4..f743530add8f 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -24,6 +24,7 @@ config SND_SOC_AC97_BUS # All the supported Soc's source "sound/soc/at32/Kconfig" source "sound/soc/at91/Kconfig" +source "sound/soc/au1x/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 3645f959c264..933a66d30804 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -2,4 +2,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ -obj-$(CONFIG_SND_SOC) += omap/ +obj-$(CONFIG_SND_SOC) += omap/ au1x/ diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig new file mode 100644 index 000000000000..410a893aa66b --- /dev/null +++ b/sound/soc/au1x/Kconfig @@ -0,0 +1,32 @@ +## +## Au1200/Au1550 PSC + DBDMA +## +config SND_SOC_AU1XPSC + tristate "SoC Audio for Au1200/Au1250/Au1550" + depends on SOC_AU1200 || SOC_AU1550 + help + This option enables support for the Programmable Serial + Controllers in AC97 and I2S mode, and the Descriptor-Based DMA + Controller (DBDMA) as found on the Au1200/Au1250/Au1550 SoC. + +config SND_SOC_AU1XPSC_I2S + tristate + +config SND_SOC_AU1XPSC_AC97 + tristate + select AC97_BUS + select SND_AC97_CODEC + select SND_SOC_AC97_BUS + + +## +## Boards +## +config SND_SOC_SAMPLE_PSC_AC97 + tristate "Sample Au12x0/Au1550 PSC AC97 sound machine" + depends on SND_SOC_AU1XPSC + select SND_SOC_AU1XPSC_AC97 + select SND_SOC_AC97_CODEC + help + This is a sample AC97 sound machine for use in Au12x0/Au1550 + based systems which have audio on PSC1 (e.g. Db1200 demoboard). diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile new file mode 100644 index 000000000000..6c6950b8003a --- /dev/null +++ b/sound/soc/au1x/Makefile @@ -0,0 +1,13 @@ +# Au1200/Au1550 PSC audio +snd-soc-au1xpsc-dbdma-objs := dbdma2.o +snd-soc-au1xpsc-i2s-objs := psc-i2s.o +snd-soc-au1xpsc-ac97-objs := psc-ac97.o + +obj-$(CONFIG_SND_SOC_AU1XPSC) += snd-soc-au1xpsc-dbdma.o +obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o +obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o + +# Boards +snd-soc-sample-ac97-objs := sample-ac97.o + +obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c new file mode 100644 index 000000000000..1466d9328800 --- /dev/null +++ b/sound/soc/au1x/dbdma2.c @@ -0,0 +1,421 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (c) 2007-2008 MSC Vertriebsges.m.b.H., + * Manuel Lauss + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * DMA glue for Au1x-PSC audio. + * + * NOTE: all of these drivers can only work with a SINGLE instance + * of a PSC. Multiple independent audio devices are impossible + * with ASoC v1. + */ + + +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include +#include +#include + +#include "psc.h" + +/*#define PCM_DEBUG*/ + +#define MSG(x...) printk(KERN_INFO "au1xpsc_pcm: " x) +#ifdef PCM_DEBUG +#define DBG MSG +#else +#define DBG(x...) do {} while (0) +#endif + +struct au1xpsc_audio_dmadata { + /* DDMA control data */ + unsigned int ddma_id; /* DDMA direction ID for this PSC */ + u32 ddma_chan; /* DDMA context */ + + /* PCM context (for irq handlers) */ + struct snd_pcm_substream *substream; + unsigned long curr_period; /* current segment DDMA is working on */ + unsigned long q_period; /* queue period(s) */ + unsigned long dma_area; /* address of queued DMA area */ + unsigned long dma_area_s; /* start address of DMA area */ + unsigned long pos; /* current byte position being played */ + unsigned long periods; /* number of SG segments in total */ + unsigned long period_bytes; /* size in bytes of one SG segment */ + + /* runtime data */ + int msbits; +}; + +/* instance data. There can be only one, MacLeod!!!! */ +static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2]; + +/* + * These settings are somewhat okay, at least on my machine audio plays + * almost skip-free. Especially the 64kB buffer seems to help a LOT. + */ +#define AU1XPSC_PERIOD_MIN_BYTES 1024 +#define AU1XPSC_BUFFER_MIN_BYTES 65536 + +#define AU1XPSC_PCM_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | \ + SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE | \ + 0) + +/* PCM hardware DMA capabilities - platform specific */ +static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = AU1XPSC_PCM_FMTS, + .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES, + .period_bytes_max = 4096 * 1024 - 1, + .periods_min = 2, + .periods_max = 4096, /* 2 to as-much-as-you-like */ + .buffer_bytes_max = 4096 * 1024 - 1, + .fifo_size = 16, /* fifo entries of AC97/I2S PSC */ +}; + +static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd) +{ + au1xxx_dbdma_put_source_flags(cd->ddma_chan, + (void *)phys_to_virt(cd->dma_area), + cd->period_bytes, DDMA_FLAGS_IE); + + /* update next-to-queue period */ + ++cd->q_period; + cd->dma_area += cd->period_bytes; + if (cd->q_period >= cd->periods) { + cd->q_period = 0; + cd->dma_area = cd->dma_area_s; + } +} + +static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd) +{ + au1xxx_dbdma_put_dest_flags(cd->ddma_chan, + (void *)phys_to_virt(cd->dma_area), + cd->period_bytes, DDMA_FLAGS_IE); + + /* update next-to-queue period */ + ++cd->q_period; + cd->dma_area += cd->period_bytes; + if (cd->q_period >= cd->periods) { + cd->q_period = 0; + cd->dma_area = cd->dma_area_s; + } +} + +static void au1x_pcm_dmatx_cb(int irq, void *dev_id) +{ + struct au1xpsc_audio_dmadata *cd = dev_id; + + cd->pos += cd->period_bytes; + if (++cd->curr_period >= cd->periods) { + cd->pos = 0; + cd->curr_period = 0; + } + snd_pcm_period_elapsed(cd->substream); + au1x_pcm_queue_tx(cd); +} + +static void au1x_pcm_dmarx_cb(int irq, void *dev_id) +{ + struct au1xpsc_audio_dmadata *cd = dev_id; + + cd->pos += cd->period_bytes; + if (++cd->curr_period >= cd->periods) { + cd->pos = 0; + cd->curr_period = 0; + } + snd_pcm_period_elapsed(cd->substream); + au1x_pcm_queue_rx(cd); +} + +static void au1x_pcm_dbdma_free(struct au1xpsc_audio_dmadata *pcd) +{ + if (pcd->ddma_chan) { + au1xxx_dbdma_stop(pcd->ddma_chan); + au1xxx_dbdma_reset(pcd->ddma_chan); + au1xxx_dbdma_chan_free(pcd->ddma_chan); + pcd->ddma_chan = 0; + pcd->msbits = 0; + } +} + +/* in case of missing DMA ring or changed TX-source / RX-dest bit widths, + * allocate (or reallocate) a 2-descriptor DMA ring with bit depth according + * to ALSA-supplied sample depth. This is due to limitations in the dbdma api + * (cannot adjust source/dest widths of already allocated descriptor ring). + */ +static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd, + int stype, int msbits) +{ + /* DMA only in 8/16/32 bit widths */ + if (msbits == 24) + msbits = 32; + + /* check current config: correct bits and descriptors allocated? */ + if ((pcd->ddma_chan) && (msbits == pcd->msbits)) + goto out; /* all ok! */ + + au1x_pcm_dbdma_free(pcd); + + if (stype == PCM_RX) + pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id, + DSCR_CMD0_ALWAYS, + au1x_pcm_dmarx_cb, (void *)pcd); + else + pcd->ddma_chan = au1xxx_dbdma_chan_alloc(DSCR_CMD0_ALWAYS, + pcd->ddma_id, + au1x_pcm_dmatx_cb, (void *)pcd); + + if (!pcd->ddma_chan) + return -ENOMEM;; + + au1xxx_dbdma_set_devwidth(pcd->ddma_chan, msbits); + au1xxx_dbdma_ring_alloc(pcd->ddma_chan, 2); + + pcd->msbits = msbits; + + au1xxx_dbdma_stop(pcd->ddma_chan); + au1xxx_dbdma_reset(pcd->ddma_chan); + +out: + return 0; +} + +static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct au1xpsc_audio_dmadata *pcd; + int stype, ret; + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) + goto out; + + stype = SUBSTREAM_TYPE(substream); + pcd = au1xpsc_audio_pcmdma[stype]; + + DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " + "runtime->min_align %d\n", + (unsigned long)runtime->dma_area, + (unsigned long)runtime->dma_addr, runtime->dma_bytes, + runtime->min_align); + + DBG("bits %d frags %d frag_bytes %d is_rx %d\n", params->msbits, + params_periods(params), params_period_bytes(params), stype); + + ret = au1x_pcm_dbdma_realloc(pcd, stype, params->msbits); + if (ret) { + MSG("DDMA channel (re)alloc failed!\n"); + goto out; + } + + pcd->substream = substream; + pcd->period_bytes = params_period_bytes(params); + pcd->periods = params_periods(params); + pcd->dma_area_s = pcd->dma_area = (unsigned long)runtime->dma_addr; + pcd->q_period = 0; + pcd->curr_period = 0; + pcd->pos = 0; + + ret = 0; +out: + return ret; +} + +static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_lib_free_pages(substream); + return 0; +} + +static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct au1xpsc_audio_dmadata *pcd = + au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]; + + au1xxx_dbdma_reset(pcd->ddma_chan); + + if (SUBSTREAM_TYPE(substream) == PCM_RX) { + au1x_pcm_queue_rx(pcd); + au1x_pcm_queue_rx(pcd); + } else { + au1x_pcm_queue_tx(pcd); + au1x_pcm_queue_tx(pcd); + } + + return 0; +} + +static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + u32 c = au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->ddma_chan; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + au1xxx_dbdma_start(c); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + au1xxx_dbdma_stop(c); + break; + default: + return -EINVAL; + } + return 0; +} + +static snd_pcm_uframes_t +au1xpsc_pcm_pointer(struct snd_pcm_substream *substream) +{ + return bytes_to_frames(substream->runtime, + au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->pos); +} + +static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) +{ + snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware); + return 0; +} + +static int au1xpsc_pcm_close(struct snd_pcm_substream *substream) +{ + au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]); + return 0; +} + +struct snd_pcm_ops au1xpsc_pcm_ops = { + .open = au1xpsc_pcm_open, + .close = au1xpsc_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = au1xpsc_pcm_hw_params, + .hw_free = au1xpsc_pcm_hw_free, + .prepare = au1xpsc_pcm_prepare, + .trigger = au1xpsc_pcm_trigger, + .pointer = au1xpsc_pcm_pointer, +}; + +static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int au1xpsc_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1); + + return 0; +} + +static int au1xpsc_pcm_probe(struct platform_device *pdev) +{ + struct resource *r; + int ret; + + if (au1xpsc_audio_pcmdma[PCM_TX] || au1xpsc_audio_pcmdma[PCM_RX]) + return -EBUSY; + + /* TX DMA */ + au1xpsc_audio_pcmdma[PCM_TX] + = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); + if (!au1xpsc_audio_pcmdma[PCM_TX]) + return -ENOMEM; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r) { + ret = -ENODEV; + goto out1; + } + (au1xpsc_audio_pcmdma[PCM_TX])->ddma_id = r->start; + + /* RX DMA */ + au1xpsc_audio_pcmdma[PCM_RX] + = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); + if (!au1xpsc_audio_pcmdma[PCM_RX]) + return -ENOMEM; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) { + ret = -ENODEV; + goto out2; + } + (au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start; + + return 0; + +out2: + kfree(au1xpsc_audio_pcmdma[PCM_RX]); + au1xpsc_audio_pcmdma[PCM_RX] = NULL; +out1: + kfree(au1xpsc_audio_pcmdma[PCM_TX]); + au1xpsc_audio_pcmdma[PCM_TX] = NULL; + return ret; +} + +static int au1xpsc_pcm_remove(struct platform_device *pdev) +{ + int i; + + for (i = 0; i < 2; i++) { + if (au1xpsc_audio_pcmdma[i]) { + au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]); + kfree(au1xpsc_audio_pcmdma[i]); + au1xpsc_audio_pcmdma[i] = NULL; + } + } + + return 0; +} + +/* au1xpsc audio platform */ +struct snd_soc_platform au1xpsc_soc_platform = { + .name = "au1xpsc-pcm-dbdma", + .probe = au1xpsc_pcm_probe, + .remove = au1xpsc_pcm_remove, + .pcm_ops = &au1xpsc_pcm_ops, + .pcm_new = au1xpsc_pcm_new, + .pcm_free = au1xpsc_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); + +static int __init au1xpsc_audio_dbdma_init(void) +{ + au1xpsc_audio_pcmdma[PCM_TX] = NULL; + au1xpsc_audio_pcmdma[PCM_RX] = NULL; + return 0; +} + +static void __exit au1xpsc_audio_dbdma_exit(void) +{ +} + +module_init(au1xpsc_audio_dbdma_init); +module_exit(au1xpsc_audio_dbdma_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver"); +MODULE_AUTHOR("Manuel Lauss "); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c new file mode 100644 index 000000000000..57facbad6825 --- /dev/null +++ b/sound/soc/au1x/psc-ac97.c @@ -0,0 +1,387 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (c) 2007-2008 MSC Vertriebsges.m.b.H., + * Manuel Lauss + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Au1xxx-PSC AC97 glue. + * + * NOTE: all of these drivers can only work with a SINGLE instance + * of a PSC. Multiple independent audio devices are impossible + * with ASoC v1. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "psc.h" + +#define AC97_DIR \ + (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_48000 + +#define AC97_FMTS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE) + +#define AC97PCR_START(stype) \ + ((stype) == PCM_TX ? PSC_AC97PCR_TS : PSC_AC97PCR_RS) +#define AC97PCR_STOP(stype) \ + ((stype) == PCM_TX ? PSC_AC97PCR_TP : PSC_AC97PCR_RP) +#define AC97PCR_CLRFIFO(stype) \ + ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) + +/* instance data. There can be only one, MacLeod!!!! */ +static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; + +/* AC97 controller reads codec register */ +static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + unsigned short data, tmo; + + au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata)); + au_sync(); + + tmo = 1000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + udelay(2); + + if (!tmo) + data = 0xffff; + else + data = au_readl(AC97_CDC(pscdata)) & 0xffff; + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + return data; +} + +/* AC97 controller writes to codec register */ +static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + unsigned int tmo; + + au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata)); + au_sync(); + tmo = 1000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + au_sync(); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); +} + +/* AC97 controller asserts a warm reset */ +static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + + au_writel(PSC_AC97RST_SNC, AC97_RST(pscdata)); + au_sync(); + msleep(10); + au_writel(0, AC97_RST(pscdata)); + au_sync(); +} + +static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + int i; + + /* disable PSC during cold reset */ + au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(pscdata)); + au_sync(); + + /* issue cold reset */ + au_writel(PSC_AC97RST_RST, AC97_RST(pscdata)); + au_sync(); + msleep(500); + au_writel(0, AC97_RST(pscdata)); + au_sync(); + + /* enable PSC */ + au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata)); + au_sync(); + + /* wait for PSC to indicate it's ready */ + i = 100000; + while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) + au_sync(); + + if (i == 0) { + printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); + return; + } + + /* enable the ac97 function */ + au_writel(pscdata->cfg | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* wait for AC97 core to become ready */ + i = 100000; + while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) + au_sync(); + if (i == 0) + printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); +} + +/* AC97 controller operations */ +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = au1xpsc_ac97_read, + .write = au1xpsc_ac97_write, + .reset = au1xpsc_ac97_cold_reset, + .warm_reset = au1xpsc_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + unsigned long r, stat; + int chans, stype = SUBSTREAM_TYPE(substream); + + chans = params_channels(params); + + r = au_readl(AC97_CFG(pscdata)); + stat = au_readl(AC97_STAT(pscdata)); + + /* already active? */ + if (stat & (PSC_AC97STAT_TB | PSC_AC97STAT_RB)) { + /* reject parameters not currently set up */ + if ((PSC_AC97CFG_GET_LEN(r) != params->msbits) || + (pscdata->rate != params_rate(params))) + return -EINVAL; + } else { + /* disable AC97 device controller first */ + au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */ + r &= ~PSC_AC97CFG_LEN_MASK; + r |= PSC_AC97CFG_SET_LEN(params->msbits); + + /* channels: enable slots for front L/R channel */ + if (stype == PCM_TX) { + r &= ~PSC_AC97CFG_TXSLOT_MASK; + r |= PSC_AC97CFG_TXSLOT_ENA(3); + r |= PSC_AC97CFG_TXSLOT_ENA(4); + } else { + r &= ~PSC_AC97CFG_RXSLOT_MASK; + r |= PSC_AC97CFG_RXSLOT_ENA(3); + r |= PSC_AC97CFG_RXSLOT_ENA(4); + } + + /* finally enable the AC97 controller again */ + au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + pscdata->cfg = r; + pscdata->rate = params_rate(params); + } + + return 0; +} + +static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + int ret, stype = SUBSTREAM_TYPE(substream); + + ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + au_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); + au_sync(); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); + au_sync(); + break; + default: + ret = -EINVAL; + } + return ret; +} + +static int au1xpsc_ac97_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + int ret; + struct resource *r; + unsigned long sel; + + if (au1xpsc_ac97_workdata) + return -EBUSY; + + au1xpsc_ac97_workdata = + kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); + if (!au1xpsc_ac97_workdata) + return -ENOMEM; + + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r) { + ret = -ENODEV; + goto out0; + } + + ret = -EBUSY; + au1xpsc_ac97_workdata->ioarea = + request_mem_region(r->start, r->end - r->start + 1, + "au1xpsc_ac97"); + if (!au1xpsc_ac97_workdata->ioarea) + goto out0; + + au1xpsc_ac97_workdata->mmio = ioremap(r->start, 0xffff); + if (!au1xpsc_ac97_workdata->mmio) + goto out1; + + /* configuration: max dma trigger threshold, enable ac97 */ + au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | + PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; + + /* preserve PSC clock source set up by platform (dev.platform_data + * is already occupied by soc layer) + */ + sel = au_readl(PSC_SEL(au1xpsc_ac97_workdata)) & PSC_SEL_CLK_MASK; + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(0, PSC_SEL(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(au1xpsc_ac97_workdata)); + au_sync(); + /* next up: cold reset. Dont check for PSC-ready now since + * there may not be any codec clock yet. + */ + + return 0; + +out1: + release_resource(au1xpsc_ac97_workdata->ioarea); + kfree(au1xpsc_ac97_workdata->ioarea); +out0: + kfree(au1xpsc_ac97_workdata); + au1xpsc_ac97_workdata = NULL; + return ret; +} + +static void au1xpsc_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* disable PSC completely */ + au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_sync(); + + iounmap(au1xpsc_ac97_workdata->mmio); + release_resource(au1xpsc_ac97_workdata->ioarea); + kfree(au1xpsc_ac97_workdata->ioarea); + kfree(au1xpsc_ac97_workdata); + au1xpsc_ac97_workdata = NULL; +} + +static int au1xpsc_ac97_suspend(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* save interesting registers and disable PSC */ + au1xpsc_ac97_workdata->pm[0] = + au_readl(PSC_SEL(au1xpsc_ac97_workdata)); + + au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_sync(); + + return 0; +} + +static int au1xpsc_ac97_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* restore PSC clock config */ + au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE, + PSC_SEL(au1xpsc_ac97_workdata)); + au_sync(); + + /* after this point the ac97 core will cold-reset the codec. + * During cold-reset the PSC is reinitialized and the last + * configuration set up in hw_params() is restored. + */ + return 0; +} + +struct snd_soc_dai au1xpsc_ac97_dai = { + .name = "au1xpsc_ac97", + .type = SND_SOC_DAI_AC97, + .probe = au1xpsc_ac97_probe, + .remove = au1xpsc_ac97_remove, + .suspend = au1xpsc_ac97_suspend, + .resume = au1xpsc_ac97_resume, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .trigger = au1xpsc_ac97_trigger, + .hw_params = au1xpsc_ac97_hw_params, + }, +}; +EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); + +static int __init au1xpsc_ac97_init(void) +{ + au1xpsc_ac97_workdata = NULL; + return 0; +} + +static void __exit au1xpsc_ac97_exit(void) +{ +} + +module_init(au1xpsc_ac97_init); +module_exit(au1xpsc_ac97_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); +MODULE_AUTHOR("Manuel Lauss "); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c new file mode 100644 index 000000000000..ba4b5c199f21 --- /dev/null +++ b/sound/soc/au1x/psc-i2s.c @@ -0,0 +1,414 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (c) 2007-2008 MSC Vertriebsges.m.b.H., + * Manuel Lauss + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Au1xxx-PSC I2S glue. + * + * NOTE: all of these drivers can only work with a SINGLE instance + * of a PSC. Multiple independent audio devices are impossible + * with ASoC v1. + * NOTE: so far only PSC slave mode (bit- and frameclock) is supported. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "psc.h" + +/* supported I2S DAI hardware formats */ +#define AU1XPSC_I2S_DAIFMT \ + (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | \ + SND_SOC_DAIFMT_NB_NF) + +/* supported I2S direction */ +#define AU1XPSC_I2S_DIR \ + (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) + +#define AU1XPSC_I2S_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define AU1XPSC_I2S_FMTS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +#define I2SSTAT_BUSY(stype) \ + ((stype) == PCM_TX ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB) +#define I2SPCR_START(stype) \ + ((stype) == PCM_TX ? PSC_I2SPCR_TS : PSC_I2SPCR_RS) +#define I2SPCR_STOP(stype) \ + ((stype) == PCM_TX ? PSC_I2SPCR_TP : PSC_I2SPCR_RP) +#define I2SPCR_CLRFIFO(stype) \ + ((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC) + + +/* instance data. There can be only one, MacLeod!!!! */ +static struct au1xpsc_audio_data *au1xpsc_i2s_workdata; + +static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + unsigned long ct; + int ret; + + ret = -EINVAL; + + ct = pscdata->cfg; + + ct &= ~(PSC_I2SCFG_XM | PSC_I2SCFG_MLJ); /* left-justified */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ct |= PSC_I2SCFG_XM; /* enable I2S mode */ + break; + case SND_SOC_DAIFMT_MSB: + break; + case SND_SOC_DAIFMT_LSB: + ct |= PSC_I2SCFG_MLJ; /* LSB (right-) justified */ + break; + default: + goto out; + } + + ct &= ~(PSC_I2SCFG_BI | PSC_I2SCFG_WI); /* IB-IF */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + ct |= PSC_I2SCFG_BI | PSC_I2SCFG_WI; + break; + case SND_SOC_DAIFMT_NB_IF: + ct |= PSC_I2SCFG_BI; + break; + case SND_SOC_DAIFMT_IB_NF: + ct |= PSC_I2SCFG_WI; + break; + case SND_SOC_DAIFMT_IB_IF: + break; + default: + goto out; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: /* CODEC master */ + ct |= PSC_I2SCFG_MS; /* PSC I2S slave mode */ + break; + case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */ + ct &= ~PSC_I2SCFG_MS; /* PSC I2S Master mode */ + break; + default: + goto out; + } + + pscdata->cfg = ct; + ret = 0; +out: + return ret; +} + +static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + + int cfgbits; + unsigned long stat; + + /* check if the PSC is already streaming data */ + stat = au_readl(I2S_STAT(pscdata)); + if (stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB)) { + /* reject parameters not currently set up in hardware */ + cfgbits = au_readl(I2S_CFG(pscdata)); + if ((PSC_I2SCFG_GET_LEN(cfgbits) != params->msbits) || + (params_rate(params) != pscdata->rate)) + return -EINVAL; + } else { + /* set sample bitdepth */ + pscdata->cfg &= ~(0x1f << 4); + pscdata->cfg |= PSC_I2SCFG_SET_LEN(params->msbits); + /* remember current rate for other stream */ + pscdata->rate = params_rate(params); + } + return 0; +} + +/* Configure PSC late: on my devel systems the codec is I2S master and + * supplies the i2sbitclock __AND__ i2sMclk (!) to the PSC unit. ASoC + * uses aggressive PM and switches the codec off when it is not in use + * which also means the PSC unit doesn't get any clocks and is therefore + * dead. That's why this chunk here gets called from the trigger callback + * because I can be reasonably certain the codec is driving the clocks. + */ +static int au1xpsc_i2s_configure(struct au1xpsc_audio_data *pscdata) +{ + unsigned long tmo; + + /* bring PSC out of sleep, and configure I2S unit */ + au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata)); + au_sync(); + + tmo = 1000000; + while (!(au_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_SR) && tmo) + tmo--; + + if (!tmo) + goto psc_err; + + au_writel(0, I2S_CFG(pscdata)); + au_sync(); + au_writel(pscdata->cfg | PSC_I2SCFG_DE_ENABLE, I2S_CFG(pscdata)); + au_sync(); + + /* wait for I2S controller to become ready */ + tmo = 1000000; + while (!(au_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_DR) && tmo) + tmo--; + + if (tmo) + return 0; + +psc_err: + au_writel(0, I2S_CFG(pscdata)); + au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata)); + au_sync(); + return -ETIMEDOUT; +} + +static int au1xpsc_i2s_start(struct au1xpsc_audio_data *pscdata, int stype) +{ + unsigned long tmo, stat; + int ret; + + ret = 0; + + /* if both TX and RX are idle, configure the PSC */ + stat = au_readl(I2S_STAT(pscdata)); + if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) { + ret = au1xpsc_i2s_configure(pscdata); + if (ret) + goto out; + } + + au_writel(I2SPCR_CLRFIFO(stype), I2S_PCR(pscdata)); + au_sync(); + au_writel(I2SPCR_START(stype), I2S_PCR(pscdata)); + au_sync(); + + /* wait for start confirmation */ + tmo = 1000000; + while (!(au_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo) + tmo--; + + if (!tmo) { + au_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata)); + au_sync(); + ret = -ETIMEDOUT; + } +out: + return ret; +} + +static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) +{ + unsigned long tmo, stat; + + au_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata)); + au_sync(); + + /* wait for stop confirmation */ + tmo = 1000000; + while ((au_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo) + tmo--; + + /* if both TX and RX are idle, disable PSC */ + stat = au_readl(I2S_STAT(pscdata)); + if (!(stat & (PSC_I2SSTAT_RB | PSC_I2SSTAT_RB))) { + au_writel(0, I2S_CFG(pscdata)); + au_sync(); + au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata)); + au_sync(); + } + return 0; +} + +static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + int ret, stype = SUBSTREAM_TYPE(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + ret = au1xpsc_i2s_start(pscdata, stype); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + ret = au1xpsc_i2s_stop(pscdata, stype); + break; + default: + ret = -EINVAL; + } + return ret; +} + +static int au1xpsc_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct resource *r; + unsigned long sel; + int ret; + + if (au1xpsc_i2s_workdata) + return -EBUSY; + + au1xpsc_i2s_workdata = + kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); + if (!au1xpsc_i2s_workdata) + return -ENOMEM; + + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r) { + ret = -ENODEV; + goto out0; + } + + ret = -EBUSY; + au1xpsc_i2s_workdata->ioarea = + request_mem_region(r->start, r->end - r->start + 1, + "au1xpsc_i2s"); + if (!au1xpsc_i2s_workdata->ioarea) + goto out0; + + au1xpsc_i2s_workdata->mmio = ioremap(r->start, 0xffff); + if (!au1xpsc_i2s_workdata->mmio) + goto out1; + + /* preserve PSC clock source set up by platform (dev.platform_data + * is already occupied by soc layer) + */ + sel = au_readl(PSC_SEL(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK; + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(au1xpsc_i2s_workdata)); + au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_sync(); + + /* preconfigure: set max rx/tx fifo depths */ + au1xpsc_i2s_workdata->cfg |= + PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8; + + /* don't wait for I2S core to become ready now; clocks may not + * be running yet; depending on clock input for PSC a wait might + * time out. + */ + + return 0; + +out1: + release_resource(au1xpsc_i2s_workdata->ioarea); + kfree(au1xpsc_i2s_workdata->ioarea); +out0: + kfree(au1xpsc_i2s_workdata); + au1xpsc_i2s_workdata = NULL; + return ret; +} + +static void au1xpsc_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_sync(); + + iounmap(au1xpsc_i2s_workdata->mmio); + release_resource(au1xpsc_i2s_workdata->ioarea); + kfree(au1xpsc_i2s_workdata->ioarea); + kfree(au1xpsc_i2s_workdata); + au1xpsc_i2s_workdata = NULL; +} + +static int au1xpsc_i2s_suspend(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + /* save interesting register and disable PSC */ + au1xpsc_i2s_workdata->pm[0] = + au_readl(PSC_SEL(au1xpsc_i2s_workdata)); + + au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_sync(); + + return 0; +} + +static int au1xpsc_i2s_resume(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + /* select I2S mode and PSC clock */ + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(0, PSC_SEL(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(au1xpsc_i2s_workdata->pm[0], + PSC_SEL(au1xpsc_i2s_workdata)); + au_sync(); + + return 0; +} + +struct snd_soc_dai au1xpsc_i2s_dai = { + .name = "au1xpsc_i2s", + .type = SND_SOC_DAI_I2S, + .probe = au1xpsc_i2s_probe, + .remove = au1xpsc_i2s_remove, + .suspend = au1xpsc_i2s_suspend, + .resume = au1xpsc_i2s_resume, + .playback = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .capture = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .ops = { + .trigger = au1xpsc_i2s_trigger, + .hw_params = au1xpsc_i2s_hw_params, + }, + .dai_ops = { + .set_fmt = au1xpsc_i2s_set_fmt, + }, +}; +EXPORT_SYMBOL(au1xpsc_i2s_dai); + +static int __init au1xpsc_i2s_init(void) +{ + au1xpsc_i2s_workdata = NULL; + return 0; +} + +static void __exit au1xpsc_i2s_exit(void) +{ +} + +module_init(au1xpsc_i2s_init); +module_exit(au1xpsc_i2s_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver"); +MODULE_AUTHOR("Manuel Lauss "); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h new file mode 100644 index 000000000000..8fdb1a04a07b --- /dev/null +++ b/sound/soc/au1x/psc.h @@ -0,0 +1,53 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (c) 2007-2008 MSC Vertriebsges.m.b.H., + * Manuel Lauss + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * NOTE: all of these drivers can only work with a SINGLE instance + * of a PSC. Multiple independent audio devices are impossible + * with ASoC v1. + */ + +#ifndef _AU1X_PCM_H +#define _AU1X_PCM_H + +extern struct snd_soc_dai au1xpsc_ac97_dai; +extern struct snd_soc_dai au1xpsc_i2s_dai; +extern struct snd_soc_platform au1xpsc_soc_platform; +extern struct snd_ac97_bus_ops soc_ac97_ops; + +struct au1xpsc_audio_data { + void __iomem *mmio; + + unsigned long cfg; + unsigned long rate; + + unsigned long pm[2]; + struct resource *ioarea; +}; + +#define PCM_TX 0 +#define PCM_RX 1 + +#define SUBSTREAM_TYPE(substream) \ + ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX) + +/* easy access macros */ +#define PSC_CTRL(x) ((unsigned long)((x)->mmio) + PSC_CTRL_OFFSET) +#define PSC_SEL(x) ((unsigned long)((x)->mmio) + PSC_SEL_OFFSET) +#define I2S_STAT(x) ((unsigned long)((x)->mmio) + PSC_I2SSTAT_OFFSET) +#define I2S_CFG(x) ((unsigned long)((x)->mmio) + PSC_I2SCFG_OFFSET) +#define I2S_PCR(x) ((unsigned long)((x)->mmio) + PSC_I2SPCR_OFFSET) +#define AC97_CFG(x) ((unsigned long)((x)->mmio) + PSC_AC97CFG_OFFSET) +#define AC97_CDC(x) ((unsigned long)((x)->mmio) + PSC_AC97CDC_OFFSET) +#define AC97_EVNT(x) ((unsigned long)((x)->mmio) + PSC_AC97EVNT_OFFSET) +#define AC97_PCR(x) ((unsigned long)((x)->mmio) + PSC_AC97PCR_OFFSET) +#define AC97_RST(x) ((unsigned long)((x)->mmio) + PSC_AC97RST_OFFSET) +#define AC97_STAT(x) ((unsigned long)((x)->mmio) + PSC_AC97STAT_OFFSET) + +#endif diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c new file mode 100644 index 000000000000..f75ae7f62c3d --- /dev/null +++ b/sound/soc/au1x/sample-ac97.c @@ -0,0 +1,144 @@ +/* + * Sample Au12x0/Au1550 PSC AC97 sound machine. + * + * Copyright (c) 2007-2008 Manuel Lauss + * + * This program is free software; you can redistribute it and/or modify + * it under the terms outlined in the file COPYING at the root of this + * source archive. + * + * This is a very generic AC97 sound machine driver for boards which + * have (AC97) audio at PSC1 (e.g. DB1200 demoboards). + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/ac97.h" +#include "psc.h" + +static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_sync(codec); + return 0; +} + +static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &au1xpsc_ac97_dai, /* see psc-ac97.c */ + .codec_dai = &ac97_dai, /* see codecs/ac97.c */ + .init = au1xpsc_sample_ac97_init, + .ops = NULL, +}; + +static struct snd_soc_machine au1xpsc_sample_ac97_machine = { + .name = "Au1xxx PSC AC97 Audio", + .dai_link = &au1xpsc_sample_ac97_dai, + .num_links = 1, +}; + +static struct snd_soc_device au1xpsc_sample_ac97_devdata = { + .machine = &au1xpsc_sample_ac97_machine, + .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct resource au1xpsc_psc1_res[] = { + [0] = { + .start = CPHYSADDR(PSC1_BASE_ADDR), + .end = CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff, + .flags = IORESOURCE_MEM, + }, + [1] = { +#ifdef CONFIG_SOC_AU1200 + .start = AU1200_PSC1_INT, + .end = AU1200_PSC1_INT, +#elif defined(CONFIG_SOC_AU1550) + .start = AU1550_PSC1_INT, + .end = AU1550_PSC1_INT, +#endif + .flags = IORESOURCE_IRQ, + }, + [2] = { + .start = DSCR_CMD0_PSC1_TX, + .end = DSCR_CMD0_PSC1_TX, + .flags = IORESOURCE_DMA, + }, + [3] = { + .start = DSCR_CMD0_PSC1_RX, + .end = DSCR_CMD0_PSC1_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device *au1xpsc_sample_ac97_dev; + +static int __init au1xpsc_sample_ac97_load(void) +{ + int ret; + +#ifdef CONFIG_SOC_AU1200 + unsigned long io; + + /* modify sys_pinfunc for AC97 on PSC1 */ + io = au_readl(SYS_PINFUNC); + io |= SYS_PINFUNC_P1C; + io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B); + au_writel(io, SYS_PINFUNC); + au_sync(); +#endif + + ret = -ENOMEM; + + /* setup PSC clock source for AC97 part: external clock provided + * by codec. The psc-ac97.c driver depends on this setting! + */ + au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET); + au_sync(); + + au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1); + if (!au1xpsc_sample_ac97_dev) + goto out; + + au1xpsc_sample_ac97_dev->resource = + kmemdup(au1xpsc_psc1_res, sizeof(struct resource) * + ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL); + au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res); + au1xpsc_sample_ac97_dev->id = 1; + + platform_set_drvdata(au1xpsc_sample_ac97_dev, + &au1xpsc_sample_ac97_devdata); + au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev; + ret = platform_device_add(au1xpsc_sample_ac97_dev); + + if (ret) { + platform_device_put(au1xpsc_sample_ac97_dev); + au1xpsc_sample_ac97_dev = NULL; + } + +out: + return ret; +} + +static void __exit au1xpsc_sample_ac97_exit(void) +{ + platform_device_unregister(au1xpsc_sample_ac97_dev); +} + +module_init(au1xpsc_sample_ac97_load); +module_exit(au1xpsc_sample_ac97_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine"); +MODULE_AUTHOR("Manuel Lauss "); From c85ceac97487580e39daaf57858513246e425647 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 10 Jul 2008 11:01:06 +0200 Subject: [PATCH 168/179] ALSA: usb-audio: fix Yamaha KX quirk We have to restrict the quirk to interface 0 because the second interface is not MIDI but HID. Additionally, this product ID is used by all four KX models, so it is better to read the product name from the device. Signed-off-by: Clemens Ladisch --- sound/usb/usbquirks.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index d675050210dc..3eb12feeae5c 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -210,7 +210,7 @@ YAMAHA_DEVICE(0x1042, NULL), YAMAHA_DEVICE(0x1043, NULL), YAMAHA_DEVICE(0x1044, NULL), YAMAHA_DEVICE(0x1045, NULL), -YAMAHA_DEVICE(0x104e, "KX49"), +YAMAHA_INTERFACE(0x104e, 0, NULL), YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), From 83a1a3974f33d45d6631363738db737624a32e82 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 10 Jul 2008 11:05:42 +0200 Subject: [PATCH 169/179] ALSA: usb-audio: add some Yamaha USB MIDI quirks Add quirk entries for four Yamaha USB MIDI devices. Signed-off-by: Clemens Ladisch --- sound/usb/usbquirks.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 3eb12feeae5c..9ea726c049c6 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -211,6 +211,10 @@ YAMAHA_DEVICE(0x1043, NULL), YAMAHA_DEVICE(0x1044, NULL), YAMAHA_DEVICE(0x1045, NULL), YAMAHA_INTERFACE(0x104e, 0, NULL), +YAMAHA_DEVICE(0x104f, NULL), +YAMAHA_DEVICE(0x1050, NULL), +YAMAHA_DEVICE(0x1051, NULL), +YAMAHA_DEVICE(0x1052, NULL), YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), From 34ec8a0ae138c2787a550b930d29a9cce4900cee Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 10 Jul 2008 14:49:19 +0200 Subject: [PATCH 170/179] ALSA: HDA - HP dc7600 with pci sub IDs 0x103c/0x3011 belongs to hp-3013 model As reported and tested by an RedHat customer, HP dc7600 with pci sub IDs 0x103c/0x3011 works with the hp-3013 model and not with the hp only model. Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 92a709be3c4e..2807bc840d26 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5167,7 +5167,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013), - SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP), + SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3013, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3014, "HP", ALC260_HP), From f7c5dda23a9f4b7f8977612154daef44cc0f423b Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 10 Jul 2008 17:49:11 +0200 Subject: [PATCH 171/179] ALSA: hda: 92hd71bxx PC Beep Added volume controls for the analog PC Beep on 92hd71bxx codecs. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a6d138831e26..08cb77f51880 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -825,6 +825,9 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), { } /* end */ From 4090dffb1438e03a434e3747b14321440561d956 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 12 Jul 2008 12:02:45 +0200 Subject: [PATCH 172/179] ALSA: hda - Fix internal mic vref pin setup Set the vref80 to the internal mic pin 0x12 for Cx5045. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 95e3367d8879..71c4072c9266 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -686,7 +686,7 @@ static struct snd_kcontrol_new cxt5045_mixers_hp530[] = { static struct hda_verb cxt5045_init_verbs[] = { /* Line in, Mic */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, /* HP, Amp */ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, From 86376df6ad0f1a7a1118cd53b0cfd679524f5436 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 12 Jul 2008 12:04:05 +0200 Subject: [PATCH 173/179] ALSA: hda - Fix missing init for unsol events on micsense model Fixed the missing initialization for unsolicited events on Cx5045 micsense model. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 71c4072c9266..8295994665e7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -962,6 +962,7 @@ static int patch_cxt5045(struct hda_codec *codec) codec->patch_ops.init = cxt5045_init; break; case CXT5045_LAPTOP_MICSENSE: + codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; spec->input_mux = &cxt5045_capture_source; spec->num_init_verbs = 2; spec->init_verbs[1] = cxt5045_mic_sense_init_verbs; From 9e4641541e9681a568483133813332cfafa34d86 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 12 Jul 2008 12:05:25 +0200 Subject: [PATCH 174/179] ALSA: hda - Fix FSC V5505 model model=laptop-hpmicsense matches better to FSC V5505 laptop. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 8295994665e7..7c1eb23f0cec 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -910,7 +910,8 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", + CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE), From 787dba37a6ff5c80c67f37c081712a6e4af92e25 Mon Sep 17 00:00:00 2001 From: Thomas Bogendoerfer Date: Sat, 12 Jul 2008 12:12:20 +0200 Subject: [PATCH 175/179] ALSA: ALSA driver for SGI HAL2 audio device This patch adds a new ALSA driver for the audio device found inside many older SGI workstation (Indy, Indigo2). The hardware uses a SGI custom chip, which feeds two codec chips, an IEC chip and a synth chip. Currently only one of the codecs is supported. This driver already has the same functionality as the HAL2 OSS driver and will replace it. Signed-off-by: Thomas Bogendoerfer Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/mips/Kconfig | 7 + sound/mips/Makefile | 2 + sound/mips/hal2.c | 947 ++++++++++++++++++++++++++++++++++++++++++++ sound/mips/hal2.h | 245 ++++++++++++ 4 files changed, 1201 insertions(+) create mode 100644 sound/mips/hal2.c create mode 100644 sound/mips/hal2.h diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index bb26f6cf4c0a..2a61cade4ac3 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -9,6 +9,13 @@ menuconfig SND_MIPS if SND_MIPS +config SND_SGI_HAL2 + tristate "SGI HAL2 Audio" + depends on SGI_HAS_HAL2 + help + Sound support for the SGI Indy and Indigo2 Workstation. + + config SND_AU1X00 tristate "Au1x00 AC97 Port Driver" depends on SOC_AU1000 || SOC_AU1100 || SOC_AU1500 diff --git a/sound/mips/Makefile b/sound/mips/Makefile index 47afed971fba..63f4a9c0a8d9 100644 --- a/sound/mips/Makefile +++ b/sound/mips/Makefile @@ -3,6 +3,8 @@ # snd-au1x00-objs := au1x00.o +snd-sgi-hal2-objs := hal2.o # Toplevel Module Dependency obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o +obj-$(CONFIG_SND_SGI_HAL2) += snd-sgi-hal2.o diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c new file mode 100644 index 000000000000..db495be01861 --- /dev/null +++ b/sound/mips/hal2.c @@ -0,0 +1,947 @@ +/* + * Driver for A2 audio system used in SGI machines + * Copyright (c) 2008 Thomas Bogendoerfer + * + * Based on OSS code from Ladislav Michl , which + * was based on code from Ulf Carlsson + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ +#include +#include +#include +#include +#include +#include + +#include +#include + +#include +#include +#include +#include +#include + +#include "hal2.h" + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SGI HAL2 soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SGI HAL2 soundcard."); +MODULE_DESCRIPTION("ALSA driver for SGI HAL2 audio"); +MODULE_AUTHOR("Thomas Bogendoerfer"); +MODULE_LICENSE("GPL"); + + +#define H2_BLOCK_SIZE 1024 +#define H2_BUF_SIZE 16384 + +struct hal2_pbus { + struct hpc3_pbus_dmacregs *pbus; + int pbusnr; + unsigned int ctrl; /* Current state of pbus->pbdma_ctrl */ +}; + +struct hal2_desc { + struct hpc_dma_desc desc; + u32 pad; /* padding */ +}; + +struct hal2_codec { + struct snd_pcm_indirect pcm_indirect; + struct snd_pcm_substream *substream; + + unsigned char *buffer; + dma_addr_t buffer_dma; + struct hal2_desc *desc; + dma_addr_t desc_dma; + int desc_count; + struct hal2_pbus pbus; + int voices; /* mono/stereo */ + unsigned int sample_rate; + unsigned int master; /* Master frequency */ + unsigned short mod; /* MOD value */ + unsigned short inc; /* INC value */ +}; + +#define H2_MIX_OUTPUT_ATT 0 +#define H2_MIX_INPUT_GAIN 1 + +struct snd_hal2 { + struct snd_card *card; + + struct hal2_ctl_regs *ctl_regs; /* HAL2 ctl registers */ + struct hal2_aes_regs *aes_regs; /* HAL2 aes registers */ + struct hal2_vol_regs *vol_regs; /* HAL2 vol registers */ + struct hal2_syn_regs *syn_regs; /* HAL2 syn registers */ + + struct hal2_codec dac; + struct hal2_codec adc; +}; + +#define H2_INDIRECT_WAIT(regs) while (hal2_read(®s->isr) & H2_ISR_TSTATUS); + +#define H2_READ_ADDR(addr) (addr | (1<<7)) +#define H2_WRITE_ADDR(addr) (addr) + +static inline u32 hal2_read(u32 *reg) +{ + return __raw_readl(reg); +} + +static inline void hal2_write(u32 val, u32 *reg) +{ + __raw_writel(val, reg); +} + + +static u32 hal2_i_read32(struct snd_hal2 *hal2, u16 addr) +{ + u32 ret; + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(H2_READ_ADDR(addr), ®s->iar); + H2_INDIRECT_WAIT(regs); + ret = hal2_read(®s->idr0) & 0xffff; + hal2_write(H2_READ_ADDR(addr) | 0x1, ®s->iar); + H2_INDIRECT_WAIT(regs); + ret |= (hal2_read(®s->idr0) & 0xffff) << 16; + return ret; +} + +static void hal2_i_write16(struct snd_hal2 *hal2, u16 addr, u16 val) +{ + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(val, ®s->idr0); + hal2_write(0, ®s->idr1); + hal2_write(0, ®s->idr2); + hal2_write(0, ®s->idr3); + hal2_write(H2_WRITE_ADDR(addr), ®s->iar); + H2_INDIRECT_WAIT(regs); +} + +static void hal2_i_write32(struct snd_hal2 *hal2, u16 addr, u32 val) +{ + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(val & 0xffff, ®s->idr0); + hal2_write(val >> 16, ®s->idr1); + hal2_write(0, ®s->idr2); + hal2_write(0, ®s->idr3); + hal2_write(H2_WRITE_ADDR(addr), ®s->iar); + H2_INDIRECT_WAIT(regs); +} + +static void hal2_i_setbit16(struct snd_hal2 *hal2, u16 addr, u16 bit) +{ + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(H2_READ_ADDR(addr), ®s->iar); + H2_INDIRECT_WAIT(regs); + hal2_write((hal2_read(®s->idr0) & 0xffff) | bit, ®s->idr0); + hal2_write(0, ®s->idr1); + hal2_write(0, ®s->idr2); + hal2_write(0, ®s->idr3); + hal2_write(H2_WRITE_ADDR(addr), ®s->iar); + H2_INDIRECT_WAIT(regs); +} + +static void hal2_i_clearbit16(struct snd_hal2 *hal2, u16 addr, u16 bit) +{ + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(H2_READ_ADDR(addr), ®s->iar); + H2_INDIRECT_WAIT(regs); + hal2_write((hal2_read(®s->idr0) & 0xffff) & ~bit, ®s->idr0); + hal2_write(0, ®s->idr1); + hal2_write(0, ®s->idr2); + hal2_write(0, ®s->idr3); + hal2_write(H2_WRITE_ADDR(addr), ®s->iar); + H2_INDIRECT_WAIT(regs); +} + +static int hal2_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + switch ((int)kcontrol->private_value) { + case H2_MIX_OUTPUT_ATT: + uinfo->value.integer.max = 31; + break; + case H2_MIX_INPUT_GAIN: + uinfo->value.integer.max = 15; + break; + } + return 0; +} + +static int hal2_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_hal2 *hal2 = snd_kcontrol_chip(kcontrol); + u32 tmp; + int l, r; + + switch ((int)kcontrol->private_value) { + case H2_MIX_OUTPUT_ATT: + tmp = hal2_i_read32(hal2, H2I_DAC_C2); + if (tmp & H2I_C2_MUTE) { + l = 0; + r = 0; + } else { + l = 31 - ((tmp >> H2I_C2_L_ATT_SHIFT) & 31); + r = 31 - ((tmp >> H2I_C2_R_ATT_SHIFT) & 31); + } + break; + case H2_MIX_INPUT_GAIN: + tmp = hal2_i_read32(hal2, H2I_ADC_C2); + l = (tmp >> H2I_C2_L_GAIN_SHIFT) & 15; + r = (tmp >> H2I_C2_R_GAIN_SHIFT) & 15; + break; + } + ucontrol->value.integer.value[0] = l; + ucontrol->value.integer.value[1] = r; + + return 0; +} + +static int hal2_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_hal2 *hal2 = snd_kcontrol_chip(kcontrol); + u32 old, new; + int l, r; + + l = ucontrol->value.integer.value[0]; + r = ucontrol->value.integer.value[1]; + + switch ((int)kcontrol->private_value) { + case H2_MIX_OUTPUT_ATT: + old = hal2_i_read32(hal2, H2I_DAC_C2); + new = old & ~(H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE); + if (l | r) { + l = 31 - l; + r = 31 - r; + new |= (l << H2I_C2_L_ATT_SHIFT); + new |= (r << H2I_C2_R_ATT_SHIFT); + } else + new |= H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE; + hal2_i_write32(hal2, H2I_DAC_C2, new); + break; + case H2_MIX_INPUT_GAIN: + old = hal2_i_read32(hal2, H2I_ADC_C2); + new = old & ~(H2I_C2_L_GAIN_M | H2I_C2_R_GAIN_M); + new |= (l << H2I_C2_L_GAIN_SHIFT); + new |= (r << H2I_C2_R_GAIN_SHIFT); + hal2_i_write32(hal2, H2I_ADC_C2, new); + break; + } + return old != new; +} + +static struct snd_kcontrol_new hal2_ctrl_headphone __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = H2_MIX_OUTPUT_ATT, + .info = hal2_gain_info, + .get = hal2_gain_get, + .put = hal2_gain_put, +}; + +static struct snd_kcontrol_new hal2_ctrl_mic __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Capture Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = H2_MIX_INPUT_GAIN, + .info = hal2_gain_info, + .get = hal2_gain_get, + .put = hal2_gain_put, +}; + +static int __devinit hal2_mixer_create(struct snd_hal2 *hal2) +{ + int err; + + /* mute DAC */ + hal2_i_write32(hal2, H2I_DAC_C2, + H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE); + /* mute ADC */ + hal2_i_write32(hal2, H2I_ADC_C2, 0); + + err = snd_ctl_add(hal2->card, + snd_ctl_new1(&hal2_ctrl_headphone, hal2)); + if (err < 0) + return err; + + err = snd_ctl_add(hal2->card, + snd_ctl_new1(&hal2_ctrl_mic, hal2)); + if (err < 0) + return err; + + return 0; +} + +static irqreturn_t hal2_interrupt(int irq, void *dev_id) +{ + struct snd_hal2 *hal2 = dev_id; + irqreturn_t ret = IRQ_NONE; + + /* decide what caused this interrupt */ + if (hal2->dac.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) { + snd_pcm_period_elapsed(hal2->dac.substream); + ret = IRQ_HANDLED; + } + if (hal2->adc.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) { + snd_pcm_period_elapsed(hal2->adc.substream); + ret = IRQ_HANDLED; + } + return ret; +} + +static int hal2_compute_rate(struct hal2_codec *codec, unsigned int rate) +{ + unsigned short mod; + + if (44100 % rate < 48000 % rate) { + mod = 4 * 44100 / rate; + codec->master = 44100; + } else { + mod = 4 * 48000 / rate; + codec->master = 48000; + } + + codec->inc = 4; + codec->mod = mod; + rate = 4 * codec->master / mod; + + return rate; +} + +static void hal2_set_dac_rate(struct snd_hal2 *hal2) +{ + unsigned int master = hal2->dac.master; + int inc = hal2->dac.inc; + int mod = hal2->dac.mod; + + hal2_i_write16(hal2, H2I_BRES1_C1, (master == 44100) ? 1 : 0); + hal2_i_write32(hal2, H2I_BRES1_C2, + ((0xffff & (inc - mod - 1)) << 16) | inc); +} + +static void hal2_set_adc_rate(struct snd_hal2 *hal2) +{ + unsigned int master = hal2->adc.master; + int inc = hal2->adc.inc; + int mod = hal2->adc.mod; + + hal2_i_write16(hal2, H2I_BRES2_C1, (master == 44100) ? 1 : 0); + hal2_i_write32(hal2, H2I_BRES2_C2, + ((0xffff & (inc - mod - 1)) << 16) | inc); +} + +static void hal2_setup_dac(struct snd_hal2 *hal2) +{ + unsigned int fifobeg, fifoend, highwater, sample_size; + struct hal2_pbus *pbus = &hal2->dac.pbus; + + /* Now we set up some PBUS information. The PBUS needs information about + * what portion of the fifo it will use. If it's receiving or + * transmitting, and finally whether the stream is little endian or big + * endian. The information is written later, on the start call. + */ + sample_size = 2 * hal2->dac.voices; + /* Fifo should be set to hold exactly four samples. Highwater mark + * should be set to two samples. */ + highwater = (sample_size * 2) >> 1; /* halfwords */ + fifobeg = 0; /* playback is first */ + fifoend = (sample_size * 4) >> 3; /* doublewords */ + pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_LD | + (highwater << 8) | (fifobeg << 16) | (fifoend << 24); + /* We disable everything before we do anything at all */ + pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; + hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX); + /* Setup the HAL2 for playback */ + hal2_set_dac_rate(hal2); + /* Set endianess */ + hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECTX); + /* Set DMA bus */ + hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr)); + /* We are using 1st Bresenham clock generator for playback */ + hal2_i_write16(hal2, H2I_DAC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT) + | (1 << H2I_C1_CLKID_SHIFT) + | (hal2->dac.voices << H2I_C1_DATAT_SHIFT)); +} + +static void hal2_setup_adc(struct snd_hal2 *hal2) +{ + unsigned int fifobeg, fifoend, highwater, sample_size; + struct hal2_pbus *pbus = &hal2->adc.pbus; + + sample_size = 2 * hal2->adc.voices; + highwater = (sample_size * 2) >> 1; /* halfwords */ + fifobeg = (4 * 4) >> 3; /* record is second */ + fifoend = (4 * 4 + sample_size * 4) >> 3; /* doublewords */ + pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_RCV | HPC3_PDMACTRL_LD | + (highwater << 8) | (fifobeg << 16) | (fifoend << 24); + pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; + hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR); + /* Setup the HAL2 for record */ + hal2_set_adc_rate(hal2); + /* Set endianess */ + hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECR); + /* Set DMA bus */ + hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr)); + /* We are using 2nd Bresenham clock generator for record */ + hal2_i_write16(hal2, H2I_ADC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT) + | (2 << H2I_C1_CLKID_SHIFT) + | (hal2->adc.voices << H2I_C1_DATAT_SHIFT)); +} + +static void hal2_start_dac(struct snd_hal2 *hal2) +{ + struct hal2_pbus *pbus = &hal2->dac.pbus; + + pbus->pbus->pbdma_dptr = hal2->dac.desc_dma; + pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT; + /* enable DAC */ + hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX); +} + +static void hal2_start_adc(struct snd_hal2 *hal2) +{ + struct hal2_pbus *pbus = &hal2->adc.pbus; + + pbus->pbus->pbdma_dptr = hal2->adc.desc_dma; + pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT; + /* enable ADC */ + hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR); +} + +static inline void hal2_stop_dac(struct snd_hal2 *hal2) +{ + hal2->dac.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; + /* The HAL2 itself may remain enabled safely */ +} + +static inline void hal2_stop_adc(struct snd_hal2 *hal2) +{ + hal2->adc.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; +} + +static int hal2_alloc_dmabuf(struct hal2_codec *codec) +{ + struct hal2_desc *desc; + dma_addr_t desc_dma, buffer_dma; + int count = H2_BUF_SIZE / H2_BLOCK_SIZE; + int i; + + codec->buffer = dma_alloc_noncoherent(NULL, H2_BUF_SIZE, + &buffer_dma, GFP_KERNEL); + if (!codec->buffer) + return -ENOMEM; + desc = dma_alloc_noncoherent(NULL, count * sizeof(struct hal2_desc), + &desc_dma, GFP_KERNEL); + if (!desc) { + dma_free_noncoherent(NULL, H2_BUF_SIZE, + codec->buffer, buffer_dma); + return -ENOMEM; + } + codec->buffer_dma = buffer_dma; + codec->desc_dma = desc_dma; + codec->desc = desc; + for (i = 0; i < count; i++) { + desc->desc.pbuf = buffer_dma + i * H2_BLOCK_SIZE; + desc->desc.cntinfo = HPCDMA_XIE | H2_BLOCK_SIZE; + desc->desc.pnext = (i == count - 1) ? + desc_dma : desc_dma + (i + 1) * sizeof(struct hal2_desc); + desc++; + } + dma_cache_sync(NULL, codec->desc, count * sizeof(struct hal2_desc), + DMA_TO_DEVICE); + codec->desc_count = count; + return 0; +} + +static void hal2_free_dmabuf(struct hal2_codec *codec) +{ + dma_free_noncoherent(NULL, codec->desc_count * sizeof(struct hal2_desc), + codec->desc, codec->desc_dma); + dma_free_noncoherent(NULL, H2_BUF_SIZE, codec->buffer, + codec->buffer_dma); +} + +static struct snd_pcm_hardware hal2_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER), + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 65536, + .period_bytes_min = 1024, + .period_bytes_max = 65536, + .periods_min = 2, + .periods_max = 1024, +}; + +static int hal2_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int err; + + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (err < 0) + return err; + + return 0; +} + +static int hal2_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int hal2_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + int err; + + runtime->hw = hal2_pcm_hw; + + err = hal2_alloc_dmabuf(&hal2->dac); + if (err) + return err; + return 0; +} + +static int hal2_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + + hal2_free_dmabuf(&hal2->dac); + return 0; +} + +static int hal2_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct hal2_codec *dac = &hal2->dac; + + dac->voices = runtime->channels; + dac->sample_rate = hal2_compute_rate(dac, runtime->rate); + memset(&dac->pcm_indirect, 0, sizeof(dac->pcm_indirect)); + dac->pcm_indirect.hw_buffer_size = H2_BUF_SIZE; + dac->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream); + dac->substream = substream; + hal2_setup_dac(hal2); + return 0; +} + +static int hal2_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + hal2->dac.pcm_indirect.hw_io = hal2->dac.buffer_dma; + hal2->dac.pcm_indirect.hw_data = 0; + substream->ops->ack(substream); + hal2_start_dac(hal2); + break; + case SNDRV_PCM_TRIGGER_STOP: + hal2_stop_dac(hal2); + break; + default: + return -EINVAL; + } + return 0; +} + +static snd_pcm_uframes_t +hal2_playback_pointer(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *dac = &hal2->dac; + + return snd_pcm_indirect_playback_pointer(substream, &dac->pcm_indirect, + dac->pbus.pbus->pbdma_bptr); +} + +static void hal2_playback_transfer(struct snd_pcm_substream *substream, + struct snd_pcm_indirect *rec, size_t bytes) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + unsigned char *buf = hal2->dac.buffer + rec->hw_data; + + memcpy(buf, substream->runtime->dma_area + rec->sw_data, bytes); + dma_cache_sync(NULL, buf, bytes, DMA_TO_DEVICE); + +} + +static int hal2_playback_ack(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *dac = &hal2->dac; + + dac->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2; + snd_pcm_indirect_playback_transfer(substream, + &dac->pcm_indirect, + hal2_playback_transfer); + return 0; +} + +static int hal2_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *adc = &hal2->adc; + int err; + + runtime->hw = hal2_pcm_hw; + + err = hal2_alloc_dmabuf(adc); + if (err) + return err; + return 0; +} + +static int hal2_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + + hal2_free_dmabuf(&hal2->adc); + return 0; +} + +static int hal2_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct hal2_codec *adc = &hal2->adc; + + adc->voices = runtime->channels; + adc->sample_rate = hal2_compute_rate(adc, runtime->rate); + memset(&adc->pcm_indirect, 0, sizeof(adc->pcm_indirect)); + adc->pcm_indirect.hw_buffer_size = H2_BUF_SIZE; + adc->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2; + adc->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream); + adc->substream = substream; + hal2_setup_adc(hal2); + return 0; +} + +static int hal2_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + hal2->adc.pcm_indirect.hw_io = hal2->adc.buffer_dma; + hal2->adc.pcm_indirect.hw_data = 0; + printk(KERN_DEBUG "buffer_dma %x\n", hal2->adc.buffer_dma); + hal2_start_adc(hal2); + break; + case SNDRV_PCM_TRIGGER_STOP: + hal2_stop_adc(hal2); + break; + default: + return -EINVAL; + } + return 0; +} + +static snd_pcm_uframes_t +hal2_capture_pointer(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *adc = &hal2->adc; + + return snd_pcm_indirect_capture_pointer(substream, &adc->pcm_indirect, + adc->pbus.pbus->pbdma_bptr); +} + +static void hal2_capture_transfer(struct snd_pcm_substream *substream, + struct snd_pcm_indirect *rec, size_t bytes) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + unsigned char *buf = hal2->adc.buffer + rec->hw_data; + + dma_cache_sync(NULL, buf, bytes, DMA_FROM_DEVICE); + memcpy(substream->runtime->dma_area + rec->sw_data, buf, bytes); +} + +static int hal2_capture_ack(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *adc = &hal2->adc; + + snd_pcm_indirect_capture_transfer(substream, + &adc->pcm_indirect, + hal2_capture_transfer); + return 0; +} + +static struct snd_pcm_ops hal2_playback_ops = { + .open = hal2_playback_open, + .close = hal2_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hal2_pcm_hw_params, + .hw_free = hal2_pcm_hw_free, + .prepare = hal2_playback_prepare, + .trigger = hal2_playback_trigger, + .pointer = hal2_playback_pointer, + .ack = hal2_playback_ack, +}; + +static struct snd_pcm_ops hal2_capture_ops = { + .open = hal2_capture_open, + .close = hal2_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hal2_pcm_hw_params, + .hw_free = hal2_pcm_hw_free, + .prepare = hal2_capture_prepare, + .trigger = hal2_capture_trigger, + .pointer = hal2_capture_pointer, + .ack = hal2_capture_ack, +}; + +static int __devinit hal2_pcm_create(struct snd_hal2 *hal2) +{ + struct snd_pcm *pcm; + int err; + + /* create first pcm device with one outputs and one input */ + err = snd_pcm_new(hal2->card, "SGI HAL2 Audio", 0, 1, 1, &pcm); + if (err < 0) + return err; + + pcm->private_data = hal2; + strcpy(pcm->name, "SGI HAL2"); + + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &hal2_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &hal2_capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + 0, 1024 * 1024); + + return 0; +} + +static int hal2_dev_free(struct snd_device *device) +{ + struct snd_hal2 *hal2 = device->device_data; + + free_irq(SGI_HPCDMA_IRQ, hal2); + kfree(hal2); + return 0; +} + +static struct snd_device_ops hal2_ops = { + .dev_free = hal2_dev_free, +}; + +static void hal2_init_codec(struct hal2_codec *codec, struct hpc3_regs *hpc3, + int index) +{ + codec->pbus.pbusnr = index; + codec->pbus.pbus = &hpc3->pbdma[index]; +} + +static int hal2_detect(struct snd_hal2 *hal2) +{ + unsigned short board, major, minor; + unsigned short rev; + + /* reset HAL2 */ + hal2_write(0, &hal2->ctl_regs->isr); + + /* release reset */ + hal2_write(H2_ISR_GLOBAL_RESET_N | H2_ISR_CODEC_RESET_N, + &hal2->ctl_regs->isr); + + + hal2_i_write16(hal2, H2I_RELAY_C, H2I_RELAY_C_STATE); + rev = hal2_read(&hal2->ctl_regs->rev); + if (rev & H2_REV_AUDIO_PRESENT) + return -ENODEV; + + board = (rev & H2_REV_BOARD_M) >> 12; + major = (rev & H2_REV_MAJOR_CHIP_M) >> 4; + minor = (rev & H2_REV_MINOR_CHIP_M); + + printk(KERN_INFO "SGI HAL2 revision %i.%i.%i\n", + board, major, minor); + + return 0; +} + +static int hal2_create(struct snd_card *card, struct snd_hal2 **rchip) +{ + struct snd_hal2 *hal2; + struct hpc3_regs *hpc3 = hpc3c0; + int err; + + hal2 = kzalloc(sizeof(struct snd_hal2), GFP_KERNEL); + if (!hal2) + return -ENOMEM; + + hal2->card = card; + + if (request_irq(SGI_HPCDMA_IRQ, hal2_interrupt, IRQF_SHARED, + "SGI HAL2", hal2)) { + printk(KERN_ERR "HAL2: Can't get irq %d\n", SGI_HPCDMA_IRQ); + kfree(hal2); + return -EAGAIN; + } + + hal2->ctl_regs = (struct hal2_ctl_regs *)hpc3->pbus_extregs[0]; + hal2->aes_regs = (struct hal2_aes_regs *)hpc3->pbus_extregs[1]; + hal2->vol_regs = (struct hal2_vol_regs *)hpc3->pbus_extregs[2]; + hal2->syn_regs = (struct hal2_syn_regs *)hpc3->pbus_extregs[3]; + + if (hal2_detect(hal2) < 0) { + kfree(hal2); + return -ENODEV; + } + + hal2_init_codec(&hal2->dac, hpc3, 0); + hal2_init_codec(&hal2->adc, hpc3, 1); + + /* + * All DMA channel interfaces in HAL2 are designed to operate with + * PBUS programmed for 2 cycles in D3, 2 cycles in D4 and 2 cycles + * in D5. HAL2 is a 16-bit device which can accept both big and little + * endian format. It assumes that even address bytes are on high + * portion of PBUS (15:8) and assumes that HPC3 is programmed to + * accept a live (unsynchronized) version of P_DREQ_N from HAL2. + */ +#define HAL2_PBUS_DMACFG ((0 << HPC3_DMACFG_D3R_SHIFT) | \ + (2 << HPC3_DMACFG_D4R_SHIFT) | \ + (2 << HPC3_DMACFG_D5R_SHIFT) | \ + (0 << HPC3_DMACFG_D3W_SHIFT) | \ + (2 << HPC3_DMACFG_D4W_SHIFT) | \ + (2 << HPC3_DMACFG_D5W_SHIFT) | \ + HPC3_DMACFG_DS16 | \ + HPC3_DMACFG_EVENHI | \ + HPC3_DMACFG_RTIME | \ + (8 << HPC3_DMACFG_BURST_SHIFT) | \ + HPC3_DMACFG_DRQLIVE) + /* + * Ignore what's mentioned in the specification and write value which + * works in The Real World (TM) + */ + hpc3->pbus_dmacfg[hal2->dac.pbus.pbusnr][0] = 0x8208844; + hpc3->pbus_dmacfg[hal2->adc.pbus.pbusnr][0] = 0x8208844; + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, hal2, &hal2_ops); + if (err < 0) { + free_irq(SGI_HPCDMA_IRQ, hal2); + kfree(hal2); + return err; + } + *rchip = hal2; + return 0; +} + +static int __devinit hal2_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct snd_hal2 *chip; + int err; + + card = snd_card_new(index, id, THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + err = hal2_create(card, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + snd_card_set_dev(card, &pdev->dev); + + err = hal2_pcm_create(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + err = hal2_mixer_create(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "SGI HAL2 Audio"); + strcpy(card->shortname, "SGI HAL2 Audio"); + sprintf(card->longname, "%s irq %i", + card->shortname, + SGI_HPCDMA_IRQ); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + platform_set_drvdata(pdev, card); + return 0; +} + +static int __exit hal2_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + + snd_card_free(card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver hal2_driver = { + .probe = hal2_probe, + .remove = __devexit_p(hal2_remove), + .driver = { + .name = "sgihal2", + .owner = THIS_MODULE, + } +}; + +static int __init alsa_card_hal2_init(void) +{ + return platform_driver_register(&hal2_driver); +} + +static void __exit alsa_card_hal2_exit(void) +{ + platform_driver_unregister(&hal2_driver); +} + +module_init(alsa_card_hal2_init); +module_exit(alsa_card_hal2_exit); diff --git a/sound/mips/hal2.h b/sound/mips/hal2.h new file mode 100644 index 000000000000..f19828bc64e0 --- /dev/null +++ b/sound/mips/hal2.h @@ -0,0 +1,245 @@ +#ifndef __HAL2_H +#define __HAL2_H + +/* + * Driver for HAL2 sound processors + * Copyright (c) 1999 Ulf Carlsson + * Copyright (c) 2001, 2002, 2003 Ladislav Michl + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include + +/* Indirect status register */ + +#define H2_ISR_TSTATUS 0x01 /* RO: transaction status 1=busy */ +#define H2_ISR_USTATUS 0x02 /* RO: utime status bit 1=armed */ +#define H2_ISR_QUAD_MODE 0x04 /* codec mode 0=indigo 1=quad */ +#define H2_ISR_GLOBAL_RESET_N 0x08 /* chip global reset 0=reset */ +#define H2_ISR_CODEC_RESET_N 0x10 /* codec/synth reset 0=reset */ + +/* Revision register */ + +#define H2_REV_AUDIO_PRESENT 0x8000 /* RO: audio present 0=present */ +#define H2_REV_BOARD_M 0x7000 /* RO: bits 14:12, board revision */ +#define H2_REV_MAJOR_CHIP_M 0x00F0 /* RO: bits 7:4, major chip revision */ +#define H2_REV_MINOR_CHIP_M 0x000F /* RO: bits 3:0, minor chip revision */ + +/* Indirect address register */ + +/* + * Address of indirect internal register to be accessed. A write to this + * register initiates read or write access to the indirect registers in the + * HAL2. Note that there af four indirect data registers for write access to + * registers larger than 16 byte. + */ + +#define H2_IAR_TYPE_M 0xF000 /* bits 15:12, type of functional */ + /* block the register resides in */ + /* 1=DMA Port */ + /* 9=Global DMA Control */ + /* 2=Bresenham */ + /* 3=Unix Timer */ +#define H2_IAR_NUM_M 0x0F00 /* bits 11:8 instance of the */ + /* blockin which the indirect */ + /* register resides */ + /* If IAR_TYPE_M=DMA Port: */ + /* 1=Synth In */ + /* 2=AES In */ + /* 3=AES Out */ + /* 4=DAC Out */ + /* 5=ADC Out */ + /* 6=Synth Control */ + /* If IAR_TYPE_M=Global DMA Control: */ + /* 1=Control */ + /* If IAR_TYPE_M=Bresenham: */ + /* 1=Bresenham Clock Gen 1 */ + /* 2=Bresenham Clock Gen 2 */ + /* 3=Bresenham Clock Gen 3 */ + /* If IAR_TYPE_M=Unix Timer: */ + /* 1=Unix Timer */ +#define H2_IAR_ACCESS_SELECT 0x0080 /* 1=read 0=write */ +#define H2_IAR_PARAM 0x000C /* Parameter Select */ +#define H2_IAR_RB_INDEX_M 0x0003 /* Read Back Index */ + /* 00:word0 */ + /* 01:word1 */ + /* 10:word2 */ + /* 11:word3 */ +/* + * HAL2 internal addressing + * + * The HAL2 has "indirect registers" (idr) which are accessed by writing to the + * Indirect Data registers. Write the address to the Indirect Address register + * to transfer the data. + * + * We define the H2IR_* to the read address and H2IW_* to the write address and + * H2I_* to be fields in whatever register is referred to. + * + * When we write to indirect registers which are larger than one word (16 bit) + * we have to fill more than one indirect register before writing. When we read + * back however we have to read several times, each time with different Read + * Back Indexes (there are defs for doing this easily). + */ + +/* + * Relay Control + */ +#define H2I_RELAY_C 0x9100 +#define H2I_RELAY_C_STATE 0x01 /* state of RELAY pin signal */ + +/* DMA port enable */ + +#define H2I_DMA_PORT_EN 0x9104 +#define H2I_DMA_PORT_EN_SY_IN 0x01 /* Synth_in DMA port */ +#define H2I_DMA_PORT_EN_AESRX 0x02 /* AES receiver DMA port */ +#define H2I_DMA_PORT_EN_AESTX 0x04 /* AES transmitter DMA port */ +#define H2I_DMA_PORT_EN_CODECTX 0x08 /* CODEC transmit DMA port */ +#define H2I_DMA_PORT_EN_CODECR 0x10 /* CODEC receive DMA port */ + +#define H2I_DMA_END 0x9108 /* global dma endian select */ +#define H2I_DMA_END_SY_IN 0x01 /* Synth_in DMA port */ +#define H2I_DMA_END_AESRX 0x02 /* AES receiver DMA port */ +#define H2I_DMA_END_AESTX 0x04 /* AES transmitter DMA port */ +#define H2I_DMA_END_CODECTX 0x08 /* CODEC transmit DMA port */ +#define H2I_DMA_END_CODECR 0x10 /* CODEC receive DMA port */ + /* 0=b_end 1=l_end */ + +#define H2I_DMA_DRV 0x910C /* global PBUS DMA enable */ + +#define H2I_SYNTH_C 0x1104 /* Synth DMA control */ + +#define H2I_AESRX_C 0x1204 /* AES RX dma control */ + +#define H2I_C_TS_EN 0x20 /* Timestamp enable */ +#define H2I_C_TS_FRMT 0x40 /* Timestamp format */ +#define H2I_C_NAUDIO 0x80 /* Sign extend */ + +/* AESRX CTL, 16 bit */ + +#define H2I_AESTX_C 0x1304 /* AES TX DMA control */ +#define H2I_AESTX_C_CLKID_SHIFT 3 /* Bresenham Clock Gen 1-3 */ +#define H2I_AESTX_C_CLKID_M 0x18 +#define H2I_AESTX_C_DATAT_SHIFT 8 /* 1=mono 2=stereo (3=quad) */ +#define H2I_AESTX_C_DATAT_M 0x300 + +/* CODEC registers */ + +#define H2I_DAC_C1 0x1404 /* DAC DMA control, 16 bit */ +#define H2I_DAC_C2 0x1408 /* DAC DMA control, 32 bit */ +#define H2I_ADC_C1 0x1504 /* ADC DMA control, 16 bit */ +#define H2I_ADC_C2 0x1508 /* ADC DMA control, 32 bit */ + +/* Bits in CTL1 register */ + +#define H2I_C1_DMA_SHIFT 0 /* DMA channel */ +#define H2I_C1_DMA_M 0x7 +#define H2I_C1_CLKID_SHIFT 3 /* Bresenham Clock Gen 1-3 */ +#define H2I_C1_CLKID_M 0x18 +#define H2I_C1_DATAT_SHIFT 8 /* 1=mono 2=stereo (3=quad) */ +#define H2I_C1_DATAT_M 0x300 + +/* Bits in CTL2 register */ + +#define H2I_C2_R_GAIN_SHIFT 0 /* right a/d input gain */ +#define H2I_C2_R_GAIN_M 0xf +#define H2I_C2_L_GAIN_SHIFT 4 /* left a/d input gain */ +#define H2I_C2_L_GAIN_M 0xf0 +#define H2I_C2_R_SEL 0x100 /* right input select */ +#define H2I_C2_L_SEL 0x200 /* left input select */ +#define H2I_C2_MUTE 0x400 /* mute */ +#define H2I_C2_DO1 0x00010000 /* digital output port bit 0 */ +#define H2I_C2_DO2 0x00020000 /* digital output port bit 1 */ +#define H2I_C2_R_ATT_SHIFT 18 /* right d/a output - */ +#define H2I_C2_R_ATT_M 0x007c0000 /* attenuation */ +#define H2I_C2_L_ATT_SHIFT 23 /* left d/a output - */ +#define H2I_C2_L_ATT_M 0x0f800000 /* attenuation */ + +#define H2I_SYNTH_MAP_C 0x1104 /* synth dma handshake ctrl */ + +/* Clock generator CTL 1, 16 bit */ + +#define H2I_BRES1_C1 0x2104 +#define H2I_BRES2_C1 0x2204 +#define H2I_BRES3_C1 0x2304 + +#define H2I_BRES_C1_SHIFT 0 /* 0=48.0 1=44.1 2=aes_rx */ +#define H2I_BRES_C1_M 0x03 + +/* Clock generator CTL 2, 32 bit */ + +#define H2I_BRES1_C2 0x2108 +#define H2I_BRES2_C2 0x2208 +#define H2I_BRES3_C2 0x2308 + +#define H2I_BRES_C2_INC_SHIFT 0 /* increment value */ +#define H2I_BRES_C2_INC_M 0xffff +#define H2I_BRES_C2_MOD_SHIFT 16 /* modcontrol value */ +#define H2I_BRES_C2_MOD_M 0xffff0000 /* modctrl=0xffff&(modinc-1) */ + +/* Unix timer, 64 bit */ + +#define H2I_UTIME 0x3104 +#define H2I_UTIME_0_LD 0xffff /* microseconds, LSB's */ +#define H2I_UTIME_1_LD0 0x0f /* microseconds, MSB's */ +#define H2I_UTIME_1_LD1 0xf0 /* tenths of microseconds */ +#define H2I_UTIME_2_LD 0xffff /* seconds, LSB's */ +#define H2I_UTIME_3_LD 0xffff /* seconds, MSB's */ + +struct hal2_ctl_regs { + u32 _unused0[4]; + u32 isr; /* 0x10 Status Register */ + u32 _unused1[3]; + u32 rev; /* 0x20 Revision Register */ + u32 _unused2[3]; + u32 iar; /* 0x30 Indirect Address Register */ + u32 _unused3[3]; + u32 idr0; /* 0x40 Indirect Data Register 0 */ + u32 _unused4[3]; + u32 idr1; /* 0x50 Indirect Data Register 1 */ + u32 _unused5[3]; + u32 idr2; /* 0x60 Indirect Data Register 2 */ + u32 _unused6[3]; + u32 idr3; /* 0x70 Indirect Data Register 3 */ +}; + +struct hal2_aes_regs { + u32 rx_stat[2]; /* Status registers */ + u32 rx_cr[2]; /* Control registers */ + u32 rx_ud[4]; /* User data window */ + u32 rx_st[24]; /* Channel status data */ + + u32 tx_stat[1]; /* Status register */ + u32 tx_cr[3]; /* Control registers */ + u32 tx_ud[4]; /* User data window */ + u32 tx_st[24]; /* Channel status data */ +}; + +struct hal2_vol_regs { + u32 right; /* Right volume */ + u32 left; /* Left volume */ +}; + +struct hal2_syn_regs { + u32 _unused0[2]; + u32 page; /* DOC Page register */ + u32 regsel; /* DOC Register selection */ + u32 dlow; /* DOC Data low */ + u32 dhigh; /* DOC Data high */ + u32 irq; /* IRQ Status */ + u32 dram; /* DRAM Access */ +}; + +#endif /* __HAL2_H */ From 1e066322c26562621811effb1eb14097bc67a9ee Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 11 Jul 2008 14:05:17 +0100 Subject: [PATCH 176/179] ALSA: asoc: kbuild - only show menus for the current ASoC CPU platform. We don't want to see ASoC platform menus for other non selected architectures in our config. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/fsl/Kconfig | 4 ---- sound/soc/omap/Kconfig | 4 ---- 2 files changed, 8 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 19802e27df4b..3368ace60977 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,5 +1,3 @@ -menu "ALSA SoC audio for Freescale SOCs" - config SND_SOC_MPC8610 bool "ALSA SoC support for the MPC8610 SOC" depends on MPC8610_HPCD @@ -16,5 +14,3 @@ config SND_SOC_MPC8610_HPCD default y if MPC8610_HPCD help Say Y if you want to enable audio on the Freescale MPC8610 HPCD. - -endmenu diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 0230d83e8e5e..aea27e70043c 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,5 +1,3 @@ -menu "SoC Audio for the Texas Instruments OMAP" - config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" depends on ARCH_OMAP && SND_SOC @@ -15,5 +13,3 @@ config SND_OMAP_SOC_N810 select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. - -endmenu From 862c2c0a61c515f2e9f63f689215bcf99a607eaf Mon Sep 17 00:00:00 2001 From: Thomas Bogendoerfer Date: Sat, 12 Jul 2008 22:43:50 +0200 Subject: [PATCH 177/179] ALSA: ALSA driver for SGI O2 audio board This patch adds a new ALSA driver for the audio device found inside most of the SGI O2 workstation. The hardware uses a SGI custom chip, which feeds a AD codec chip. Signed-off-by: Thomas Bogendoerfer Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/ad1843.h | 46 ++ sound/mips/Kconfig | 6 + sound/mips/Makefile | 2 + sound/mips/ad1843.c | 561 ++++++++++++++++++++++ sound/mips/sgio2audio.c | 1006 +++++++++++++++++++++++++++++++++++++++ 5 files changed, 1621 insertions(+) create mode 100644 include/sound/ad1843.h create mode 100644 sound/mips/ad1843.c create mode 100644 sound/mips/sgio2audio.c diff --git a/include/sound/ad1843.h b/include/sound/ad1843.h new file mode 100644 index 000000000000..b236a9d1d6e4 --- /dev/null +++ b/include/sound/ad1843.h @@ -0,0 +1,46 @@ +/* + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + * + * Copyright 2003 Vivien Chappelier + * Copyright 2008 Thomas Bogendoerfer + */ + +#ifndef __SOUND_AD1843_H +#define __SOUND_AD1843_H + +struct snd_ad1843 { + void *chip; + int (*read)(void *chip, int reg); + int (*write)(void *chip, int reg, int val); +}; + +#define AD1843_GAIN_RECLEV 0 +#define AD1843_GAIN_LINE 1 +#define AD1843_GAIN_LINE_2 2 +#define AD1843_GAIN_MIC 3 +#define AD1843_GAIN_PCM_0 4 +#define AD1843_GAIN_PCM_1 5 +#define AD1843_GAIN_SIZE (AD1843_GAIN_PCM_1+1) + +int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id); +int ad1843_get_gain(struct snd_ad1843 *ad1843, int id); +int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval); +int ad1843_get_recsrc(struct snd_ad1843 *ad1843); +int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc); +void ad1843_setup_dac(struct snd_ad1843 *ad1843, + unsigned int id, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels); +void ad1843_shutdown_dac(struct snd_ad1843 *ad1843, + unsigned int id); +void ad1843_setup_adc(struct snd_ad1843 *ad1843, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels); +void ad1843_shutdown_adc(struct snd_ad1843 *ad1843); +int ad1843_init(struct snd_ad1843 *ad1843); + +#endif /* __SOUND_AD1843_H */ diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 2a61cade4ac3..a9823fad85c2 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -9,6 +9,12 @@ menuconfig SND_MIPS if SND_MIPS +config SND_SGI_O2 + tristate "SGI O2 Audio" + depends on SGI_IP32 + help + Sound support for the SGI O2 Workstation. + config SND_SGI_HAL2 tristate "SGI HAL2 Audio" depends on SGI_HAS_HAL2 diff --git a/sound/mips/Makefile b/sound/mips/Makefile index 63f4a9c0a8d9..861ec0a574b4 100644 --- a/sound/mips/Makefile +++ b/sound/mips/Makefile @@ -3,8 +3,10 @@ # snd-au1x00-objs := au1x00.o +snd-sgi-o2-objs := sgio2audio.o ad1843.o snd-sgi-hal2-objs := hal2.o # Toplevel Module Dependency obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o +obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o obj-$(CONFIG_SND_SGI_HAL2) += snd-sgi-hal2.o diff --git a/sound/mips/ad1843.c b/sound/mips/ad1843.c new file mode 100644 index 000000000000..c624510ec374 --- /dev/null +++ b/sound/mips/ad1843.c @@ -0,0 +1,561 @@ +/* + * AD1843 low level driver + * + * Copyright 2003 Vivien Chappelier + * Copyright 2008 Thomas Bogendoerfer + * + * inspired from vwsnd.c (SGI VW audio driver) + * Copyright 1999 Silicon Graphics, Inc. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include +#include +#include +#include +#include +#include + +/* + * AD1843 bitfield definitions. All are named as in the AD1843 data + * sheet, with ad1843_ prepended and individual bit numbers removed. + * + * E.g., bits LSS0 through LSS2 become ad1843_LSS. + * + * Only the bitfields we need are defined. + */ + +struct ad1843_bitfield { + char reg; + char lo_bit; + char nbits; +}; + +static const struct ad1843_bitfield + ad1843_PDNO = { 0, 14, 1 }, /* Converter Power-Down Flag */ + ad1843_INIT = { 0, 15, 1 }, /* Clock Initialization Flag */ + ad1843_RIG = { 2, 0, 4 }, /* Right ADC Input Gain */ + ad1843_RMGE = { 2, 4, 1 }, /* Right ADC Mic Gain Enable */ + ad1843_RSS = { 2, 5, 3 }, /* Right ADC Source Select */ + ad1843_LIG = { 2, 8, 4 }, /* Left ADC Input Gain */ + ad1843_LMGE = { 2, 12, 1 }, /* Left ADC Mic Gain Enable */ + ad1843_LSS = { 2, 13, 3 }, /* Left ADC Source Select */ + ad1843_RD2M = { 3, 0, 5 }, /* Right DAC 2 Mix Gain/Atten */ + ad1843_RD2MM = { 3, 7, 1 }, /* Right DAC 2 Mix Mute */ + ad1843_LD2M = { 3, 8, 5 }, /* Left DAC 2 Mix Gain/Atten */ + ad1843_LD2MM = { 3, 15, 1 }, /* Left DAC 2 Mix Mute */ + ad1843_RX1M = { 4, 0, 5 }, /* Right Aux 1 Mix Gain/Atten */ + ad1843_RX1MM = { 4, 7, 1 }, /* Right Aux 1 Mix Mute */ + ad1843_LX1M = { 4, 8, 5 }, /* Left Aux 1 Mix Gain/Atten */ + ad1843_LX1MM = { 4, 15, 1 }, /* Left Aux 1 Mix Mute */ + ad1843_RX2M = { 5, 0, 5 }, /* Right Aux 2 Mix Gain/Atten */ + ad1843_RX2MM = { 5, 7, 1 }, /* Right Aux 2 Mix Mute */ + ad1843_LX2M = { 5, 8, 5 }, /* Left Aux 2 Mix Gain/Atten */ + ad1843_LX2MM = { 5, 15, 1 }, /* Left Aux 2 Mix Mute */ + ad1843_RMCM = { 7, 0, 5 }, /* Right Mic Mix Gain/Atten */ + ad1843_RMCMM = { 7, 7, 1 }, /* Right Mic Mix Mute */ + ad1843_LMCM = { 7, 8, 5 }, /* Left Mic Mix Gain/Atten */ + ad1843_LMCMM = { 7, 15, 1 }, /* Left Mic Mix Mute */ + ad1843_HPOS = { 8, 4, 1 }, /* Headphone Output Voltage Swing */ + ad1843_HPOM = { 8, 5, 1 }, /* Headphone Output Mute */ + ad1843_MPOM = { 8, 6, 1 }, /* Mono Output Mute */ + ad1843_RDA1G = { 9, 0, 6 }, /* Right DAC1 Analog/Digital Gain */ + ad1843_RDA1GM = { 9, 7, 1 }, /* Right DAC1 Analog Mute */ + ad1843_LDA1G = { 9, 8, 6 }, /* Left DAC1 Analog/Digital Gain */ + ad1843_LDA1GM = { 9, 15, 1 }, /* Left DAC1 Analog Mute */ + ad1843_RDA2G = { 10, 0, 6 }, /* Right DAC2 Analog/Digital Gain */ + ad1843_RDA2GM = { 10, 7, 1 }, /* Right DAC2 Analog Mute */ + ad1843_LDA2G = { 10, 8, 6 }, /* Left DAC2 Analog/Digital Gain */ + ad1843_LDA2GM = { 10, 15, 1 }, /* Left DAC2 Analog Mute */ + ad1843_RDA1AM = { 11, 7, 1 }, /* Right DAC1 Digital Mute */ + ad1843_LDA1AM = { 11, 15, 1 }, /* Left DAC1 Digital Mute */ + ad1843_RDA2AM = { 12, 7, 1 }, /* Right DAC2 Digital Mute */ + ad1843_LDA2AM = { 12, 15, 1 }, /* Left DAC2 Digital Mute */ + ad1843_ADLC = { 15, 0, 2 }, /* ADC Left Sample Rate Source */ + ad1843_ADRC = { 15, 2, 2 }, /* ADC Right Sample Rate Source */ + ad1843_DA1C = { 15, 8, 2 }, /* DAC1 Sample Rate Source */ + ad1843_DA2C = { 15, 10, 2 }, /* DAC2 Sample Rate Source */ + ad1843_C1C = { 17, 0, 16 }, /* Clock 1 Sample Rate Select */ + ad1843_C2C = { 20, 0, 16 }, /* Clock 2 Sample Rate Select */ + ad1843_C3C = { 23, 0, 16 }, /* Clock 3 Sample Rate Select */ + ad1843_DAADL = { 25, 4, 2 }, /* Digital ADC Left Source Select */ + ad1843_DAADR = { 25, 6, 2 }, /* Digital ADC Right Source Select */ + ad1843_DAMIX = { 25, 14, 1 }, /* DAC Digital Mix Enable */ + ad1843_DRSFLT = { 25, 15, 1 }, /* Digital Reampler Filter Mode */ + ad1843_ADLF = { 26, 0, 2 }, /* ADC Left Channel Data Format */ + ad1843_ADRF = { 26, 2, 2 }, /* ADC Right Channel Data Format */ + ad1843_ADTLK = { 26, 4, 1 }, /* ADC Transmit Lock Mode Select */ + ad1843_SCF = { 26, 7, 1 }, /* SCLK Frequency Select */ + ad1843_DA1F = { 26, 8, 2 }, /* DAC1 Data Format Select */ + ad1843_DA2F = { 26, 10, 2 }, /* DAC2 Data Format Select */ + ad1843_DA1SM = { 26, 14, 1 }, /* DAC1 Stereo/Mono Mode Select */ + ad1843_DA2SM = { 26, 15, 1 }, /* DAC2 Stereo/Mono Mode Select */ + ad1843_ADLEN = { 27, 0, 1 }, /* ADC Left Channel Enable */ + ad1843_ADREN = { 27, 1, 1 }, /* ADC Right Channel Enable */ + ad1843_AAMEN = { 27, 4, 1 }, /* Analog to Analog Mix Enable */ + ad1843_ANAEN = { 27, 7, 1 }, /* Analog Channel Enable */ + ad1843_DA1EN = { 27, 8, 1 }, /* DAC1 Enable */ + ad1843_DA2EN = { 27, 9, 1 }, /* DAC2 Enable */ + ad1843_DDMEN = { 27, 12, 1 }, /* DAC2 to DAC1 Mix Enable */ + ad1843_C1EN = { 28, 11, 1 }, /* Clock Generator 1 Enable */ + ad1843_C2EN = { 28, 12, 1 }, /* Clock Generator 2 Enable */ + ad1843_C3EN = { 28, 13, 1 }, /* Clock Generator 3 Enable */ + ad1843_PDNI = { 28, 15, 1 }; /* Converter Power Down */ + +/* + * The various registers of the AD1843 use three different formats for + * specifying gain. The ad1843_gain structure parameterizes the + * formats. + */ + +struct ad1843_gain { + int negative; /* nonzero if gain is negative. */ + const struct ad1843_bitfield *lfield; + const struct ad1843_bitfield *rfield; + const struct ad1843_bitfield *lmute; + const struct ad1843_bitfield *rmute; +}; + +static const struct ad1843_gain ad1843_gain_RECLEV = { + .negative = 0, + .lfield = &ad1843_LIG, + .rfield = &ad1843_RIG +}; +static const struct ad1843_gain ad1843_gain_LINE = { + .negative = 1, + .lfield = &ad1843_LX1M, + .rfield = &ad1843_RX1M, + .lmute = &ad1843_LX1MM, + .rmute = &ad1843_RX1MM +}; +static const struct ad1843_gain ad1843_gain_LINE_2 = { + .negative = 1, + .lfield = &ad1843_LDA2G, + .rfield = &ad1843_RDA2G, + .lmute = &ad1843_LDA2GM, + .rmute = &ad1843_RDA2GM +}; +static const struct ad1843_gain ad1843_gain_MIC = { + .negative = 1, + .lfield = &ad1843_LMCM, + .rfield = &ad1843_RMCM, + .lmute = &ad1843_LMCMM, + .rmute = &ad1843_RMCMM +}; +static const struct ad1843_gain ad1843_gain_PCM_0 = { + .negative = 1, + .lfield = &ad1843_LDA1G, + .rfield = &ad1843_RDA1G, + .lmute = &ad1843_LDA1GM, + .rmute = &ad1843_RDA1GM +}; +static const struct ad1843_gain ad1843_gain_PCM_1 = { + .negative = 1, + .lfield = &ad1843_LD2M, + .rfield = &ad1843_RD2M, + .lmute = &ad1843_LD2MM, + .rmute = &ad1843_RD2MM +}; + +static const struct ad1843_gain *ad1843_gain[AD1843_GAIN_SIZE] = +{ + &ad1843_gain_RECLEV, + &ad1843_gain_LINE, + &ad1843_gain_LINE_2, + &ad1843_gain_MIC, + &ad1843_gain_PCM_0, + &ad1843_gain_PCM_1, +}; + +/* read the current value of an AD1843 bitfield. */ + +static int ad1843_read_bits(struct snd_ad1843 *ad1843, + const struct ad1843_bitfield *field) +{ + int w; + + w = ad1843->read(ad1843->chip, field->reg); + return w >> field->lo_bit & ((1 << field->nbits) - 1); +} + +/* + * write a new value to an AD1843 bitfield and return the old value. + */ + +static int ad1843_write_bits(struct snd_ad1843 *ad1843, + const struct ad1843_bitfield *field, + int newval) +{ + int w, mask, oldval, newbits; + + w = ad1843->read(ad1843->chip, field->reg); + mask = ((1 << field->nbits) - 1) << field->lo_bit; + oldval = (w & mask) >> field->lo_bit; + newbits = (newval << field->lo_bit) & mask; + w = (w & ~mask) | newbits; + ad1843->write(ad1843->chip, field->reg, w); + + return oldval; +} + +/* + * ad1843_read_multi reads multiple bitfields from the same AD1843 + * register. It uses a single read cycle to do it. (Reading the + * ad1843 requires 256 bit times at 12.288 MHz, or nearly 20 + * microseconds.) + * + * Called like this. + * + * ad1843_read_multi(ad1843, nfields, + * &ad1843_FIELD1, &val1, + * &ad1843_FIELD2, &val2, ...); + */ + +static void ad1843_read_multi(struct snd_ad1843 *ad1843, int argcount, ...) +{ + va_list ap; + const struct ad1843_bitfield *fp; + int w = 0, mask, *value, reg = -1; + + va_start(ap, argcount); + while (--argcount >= 0) { + fp = va_arg(ap, const struct ad1843_bitfield *); + value = va_arg(ap, int *); + if (reg == -1) { + reg = fp->reg; + w = ad1843->read(ad1843->chip, reg); + } + + mask = (1 << fp->nbits) - 1; + *value = w >> fp->lo_bit & mask; + } + va_end(ap); +} + +/* + * ad1843_write_multi stores multiple bitfields into the same AD1843 + * register. It uses one read and one write cycle to do it. + * + * Called like this. + * + * ad1843_write_multi(ad1843, nfields, + * &ad1843_FIELD1, val1, + * &ad1843_FIELF2, val2, ...); + */ + +static void ad1843_write_multi(struct snd_ad1843 *ad1843, int argcount, ...) +{ + va_list ap; + int reg; + const struct ad1843_bitfield *fp; + int value; + int w, m, mask, bits; + + mask = 0; + bits = 0; + reg = -1; + + va_start(ap, argcount); + while (--argcount >= 0) { + fp = va_arg(ap, const struct ad1843_bitfield *); + value = va_arg(ap, int); + if (reg == -1) + reg = fp->reg; + else + BUG_ON(reg != fp->reg); + m = ((1 << fp->nbits) - 1) << fp->lo_bit; + mask |= m; + bits |= (value << fp->lo_bit) & m; + } + va_end(ap); + + if (~mask & 0xFFFF) + w = ad1843->read(ad1843->chip, reg); + else + w = 0; + w = (w & ~mask) | bits; + ad1843->write(ad1843->chip, reg, w); +} + +int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id) +{ + const struct ad1843_gain *gp = ad1843_gain[id]; + int ret; + + ret = (1 << gp->lfield->nbits); + if (!gp->lmute) + ret -= 1; + return ret; +} + +/* + * ad1843_get_gain reads the specified register and extracts the gain value + * using the supplied gain type. + */ + +int ad1843_get_gain(struct snd_ad1843 *ad1843, int id) +{ + int lg, rg, lm, rm; + const struct ad1843_gain *gp = ad1843_gain[id]; + unsigned short mask = (1 << gp->lfield->nbits) - 1; + + ad1843_read_multi(ad1843, 2, gp->lfield, &lg, gp->rfield, &rg); + if (gp->negative) { + lg = mask - lg; + rg = mask - rg; + } + if (gp->lmute) { + ad1843_read_multi(ad1843, 2, gp->lmute, &lm, gp->rmute, &rm); + if (lm) + lg = 0; + if (rm) + rg = 0; + } + return lg << 0 | rg << 8; +} + +/* + * Set an audio channel's gain. + * + * Returns the new gain, which may be lower than the old gain. + */ + +int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval) +{ + const struct ad1843_gain *gp = ad1843_gain[id]; + unsigned short mask = (1 << gp->lfield->nbits) - 1; + + int lg = (newval >> 0) & mask; + int rg = (newval >> 8) & mask; + int lm = (lg == 0) ? 1 : 0; + int rm = (rg == 0) ? 1 : 0; + + if (gp->negative) { + lg = mask - lg; + rg = mask - rg; + } + if (gp->lmute) + ad1843_write_multi(ad1843, 2, gp->lmute, lm, gp->rmute, rm); + ad1843_write_multi(ad1843, 2, gp->lfield, lg, gp->rfield, rg); + return ad1843_get_gain(ad1843, id); +} + +/* Returns the current recording source */ + +int ad1843_get_recsrc(struct snd_ad1843 *ad1843) +{ + int val = ad1843_read_bits(ad1843, &ad1843_LSS); + + if (val < 0 || val > 2) { + val = 2; + ad1843_write_multi(ad1843, 2, + &ad1843_LSS, val, &ad1843_RSS, val); + } + return val; +} + +/* + * Set recording source. + * + * Returns newsrc on success, -errno on failure. + */ + +int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc) +{ + if (newsrc < 0 || newsrc > 2) + return -EINVAL; + + ad1843_write_multi(ad1843, 2, &ad1843_LSS, newsrc, &ad1843_RSS, newsrc); + return newsrc; +} + +/* Setup ad1843 for D/A conversion. */ + +void ad1843_setup_dac(struct snd_ad1843 *ad1843, + unsigned int id, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels) +{ + int ad_fmt = 0, ad_mode = 0; + + switch (fmt) { + case SNDRV_PCM_FORMAT_S8: + ad_fmt = 0; + break; + case SNDRV_PCM_FORMAT_U8: + ad_fmt = 0; + break; + case SNDRV_PCM_FORMAT_S16_LE: + ad_fmt = 1; + break; + case SNDRV_PCM_FORMAT_MU_LAW: + ad_fmt = 2; + break; + case SNDRV_PCM_FORMAT_A_LAW: + ad_fmt = 3; + break; + default: + break; + } + + switch (channels) { + case 2: + ad_mode = 0; + break; + case 1: + ad_mode = 1; + break; + default: + break; + } + + if (id) { + ad1843_write_bits(ad1843, &ad1843_C2C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_DA2SM, ad_mode, + &ad1843_DA2F, ad_fmt); + } else { + ad1843_write_bits(ad1843, &ad1843_C1C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_DA1SM, ad_mode, + &ad1843_DA1F, ad_fmt); + } +} + +void ad1843_shutdown_dac(struct snd_ad1843 *ad1843, unsigned int id) +{ + if (id) + ad1843_write_bits(ad1843, &ad1843_DA2F, 1); + else + ad1843_write_bits(ad1843, &ad1843_DA1F, 1); +} + +void ad1843_setup_adc(struct snd_ad1843 *ad1843, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels) +{ + int da_fmt = 0; + + switch (fmt) { + case SNDRV_PCM_FORMAT_S8: da_fmt = 0; break; + case SNDRV_PCM_FORMAT_U8: da_fmt = 0; break; + case SNDRV_PCM_FORMAT_S16_LE: da_fmt = 1; break; + case SNDRV_PCM_FORMAT_MU_LAW: da_fmt = 2; break; + case SNDRV_PCM_FORMAT_A_LAW: da_fmt = 3; break; + default: break; + } + + ad1843_write_bits(ad1843, &ad1843_C3C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_ADLF, da_fmt, &ad1843_ADRF, da_fmt); +} + +void ad1843_shutdown_adc(struct snd_ad1843 *ad1843) +{ + /* nothing to do */ +} + +/* + * Fully initialize the ad1843. As described in the AD1843 data + * sheet, section "START-UP SEQUENCE". The numbered comments are + * subsection headings from the data sheet. See the data sheet, pages + * 52-54, for more info. + * + * return 0 on success, -errno on failure. */ + +int ad1843_init(struct snd_ad1843 *ad1843) +{ + unsigned long later; + + if (ad1843_read_bits(ad1843, &ad1843_INIT) != 0) { + printk(KERN_ERR "ad1843: AD1843 won't initialize\n"); + return -EIO; + } + + ad1843_write_bits(ad1843, &ad1843_SCF, 1); + + /* 4. Put the conversion resources into standby. */ + ad1843_write_bits(ad1843, &ad1843_PDNI, 0); + later = jiffies + msecs_to_jiffies(500); + + while (ad1843_read_bits(ad1843, &ad1843_PDNO)) { + if (time_after(jiffies, later)) { + printk(KERN_ERR + "ad1843: AD1843 won't power up\n"); + return -EIO; + } + schedule_timeout_interruptible(5); + } + + /* 5. Power up the clock generators and enable clock output pins. */ + ad1843_write_multi(ad1843, 3, + &ad1843_C1EN, 1, + &ad1843_C2EN, 1, + &ad1843_C3EN, 1); + + /* 6. Configure conversion resources while they are in standby. */ + + /* DAC1/2 use clock 1/2 as source, ADC uses clock 3. Always. */ + ad1843_write_multi(ad1843, 4, + &ad1843_DA1C, 1, + &ad1843_DA2C, 2, + &ad1843_ADLC, 3, + &ad1843_ADRC, 3); + + /* 7. Enable conversion resources. */ + ad1843_write_bits(ad1843, &ad1843_ADTLK, 1); + ad1843_write_multi(ad1843, 7, + &ad1843_ANAEN, 1, + &ad1843_AAMEN, 1, + &ad1843_DA1EN, 1, + &ad1843_DA2EN, 1, + &ad1843_DDMEN, 1, + &ad1843_ADLEN, 1, + &ad1843_ADREN, 1); + + /* 8. Configure conversion resources while they are enabled. */ + + /* set gain to 0 for all channels */ + ad1843_set_gain(ad1843, AD1843_GAIN_RECLEV, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_LINE, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_LINE_2, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_MIC, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_PCM_0, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_PCM_1, 0); + + /* Unmute all channels. */ + /* DAC1 */ + ad1843_write_multi(ad1843, 2, &ad1843_LDA1GM, 0, &ad1843_RDA1GM, 0); + /* DAC2 */ + ad1843_write_multi(ad1843, 2, &ad1843_LDA2GM, 0, &ad1843_RDA2GM, 0); + + /* Set default recording source to Line In and set + * mic gain to +20 dB. + */ + ad1843_set_recsrc(ad1843, 2); + ad1843_write_multi(ad1843, 2, &ad1843_LMGE, 1, &ad1843_RMGE, 1); + + /* Set Speaker Out level to +/- 4V and unmute it. */ + ad1843_write_multi(ad1843, 3, + &ad1843_HPOS, 1, + &ad1843_HPOM, 0, + &ad1843_MPOM, 0); + + return 0; +} diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c new file mode 100644 index 000000000000..4c63504348dc --- /dev/null +++ b/sound/mips/sgio2audio.c @@ -0,0 +1,1006 @@ +/* + * Sound driver for Silicon Graphics O2 Workstations A/V board audio. + * + * Copyright 2003 Vivien Chappelier + * Copyright 2008 Thomas Bogendoerfer + * Mxier part taken from mace_audio.c: + * Copyright 2007 Thorben Jändling + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include +#include +#include +#define SNDRV_GET_ID +#include +#include + + +MODULE_AUTHOR("Vivien Chappelier "); +MODULE_DESCRIPTION("SGI O2 Audio"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard."); + + +#define AUDIO_CONTROL_RESET BIT(0) /* 1: reset audio interface */ +#define AUDIO_CONTROL_CODEC_PRESENT BIT(1) /* 1: codec detected */ + +#define CODEC_CONTROL_WORD_SHIFT 0 +#define CODEC_CONTROL_READ BIT(16) +#define CODEC_CONTROL_ADDRESS_SHIFT 17 + +#define CHANNEL_CONTROL_RESET BIT(10) /* 1: reset channel */ +#define CHANNEL_DMA_ENABLE BIT(9) /* 1: enable DMA transfer */ +#define CHANNEL_INT_THRESHOLD_DISABLED (0 << 5) /* interrupt disabled */ +#define CHANNEL_INT_THRESHOLD_25 (1 << 5) /* int on buffer >25% full */ +#define CHANNEL_INT_THRESHOLD_50 (2 << 5) /* int on buffer >50% full */ +#define CHANNEL_INT_THRESHOLD_75 (3 << 5) /* int on buffer >75% full */ +#define CHANNEL_INT_THRESHOLD_EMPTY (4 << 5) /* int on buffer empty */ +#define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */ +#define CHANNEL_INT_THRESHOLD_FULL (6 << 5) /* int on buffer empty */ +#define CHANNEL_INT_THRESHOLD_NOT_FULL (7 << 5) /* int on buffer !empty */ + +#define CHANNEL_RING_SHIFT 12 +#define CHANNEL_RING_SIZE (1 << CHANNEL_RING_SHIFT) +#define CHANNEL_RING_MASK (CHANNEL_RING_SIZE - 1) + +#define CHANNEL_LEFT_SHIFT 40 +#define CHANNEL_RIGHT_SHIFT 8 + +struct snd_sgio2audio_chan { + int idx; + struct snd_pcm_substream *substream; + int pos; + snd_pcm_uframes_t size; + spinlock_t lock; +}; + +/* definition of the chip-specific record */ +struct snd_sgio2audio { + struct snd_card *card; + + /* codec */ + struct snd_ad1843 ad1843; + spinlock_t ad1843_lock; + + /* channels */ + struct snd_sgio2audio_chan channel[3]; + + /* resources */ + void *ring_base; + dma_addr_t ring_base_dma; +}; + +/* AD1843 access */ + +/* + * read_ad1843_reg returns the current contents of a 16 bit AD1843 register. + * + * Returns unsigned register value on success, -errno on failure. + */ +static int read_ad1843_reg(void *priv, int reg) +{ + struct snd_sgio2audio *chip = priv; + int val; + unsigned long flags; + + spin_lock_irqsave(&chip->ad1843_lock, flags); + + writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) | + CODEC_CONTROL_READ, &mace->perif.audio.codec_control); + wmb(); + val = readq(&mace->perif.audio.codec_control); /* flush bus */ + udelay(200); + + val = readq(&mace->perif.audio.codec_read); + + spin_unlock_irqrestore(&chip->ad1843_lock, flags); + return val; +} + +/* + * write_ad1843_reg writes the specified value to a 16 bit AD1843 register. + */ +static int write_ad1843_reg(void *priv, int reg, int word) +{ + struct snd_sgio2audio *chip = priv; + int val; + unsigned long flags; + + spin_lock_irqsave(&chip->ad1843_lock, flags); + + writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) | + (word << CODEC_CONTROL_WORD_SHIFT), + &mace->perif.audio.codec_control); + wmb(); + val = readq(&mace->perif.audio.codec_control); /* flush bus */ + udelay(200); + + spin_unlock_irqrestore(&chip->ad1843_lock, flags); + return 0; +} + +static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843, + (int)kcontrol->private_value); + return 0; +} + +static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int vol; + + vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value); + + ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF; + ucontrol->value.integer.value[1] = vol & 0xFF; + + return 0; +} + +static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int newvol, oldvol; + + oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value); + newvol = (ucontrol->value.integer.value[0] << 8) | + ucontrol->value.integer.value[1]; + + newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value, + newvol); + + return newvol != oldvol; +} + +static int sgio2audio_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *texts[3] = { + "Cam Mic", "Mic", "Line" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 3; + if (uinfo->value.enumerated.item >= 3) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int sgio2audio_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843); + return 0; +} + +static int sgio2audio_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int newsrc, oldsrc; + + oldsrc = ad1843_get_recsrc(&chip->ad1843); + newsrc = ad1843_set_recsrc(&chip->ad1843, + ucontrol->value.enumerated.item[0]); + + return newsrc != oldsrc; +} + +/* dac1/pcm0 mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_PCM_0, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* dac2/pcm1 mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_PCM_1, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* record level mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_reclevel __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_RECLEV, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* record level source control */ +static struct snd_kcontrol_new sgio2audio_ctrl_recsource __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = sgio2audio_source_info, + .get = sgio2audio_source_get, + .put = sgio2audio_source_put, +}; + +/* line mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_line __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_LINE, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* cd mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_cd __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Playback Volume", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_LINE_2, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* mic mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_mic __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_MIC, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + + +static int __devinit snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip) +{ + int err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_recsource, chip)); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_line, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_cd, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_mic, chip)); + if (err < 0) + return err; + + return 0; +} + +/* low-level audio interface DMA */ + +/* get data out of bounce buffer, count must be a multiple of 32 */ +/* returns 1 if a period has elapsed */ +static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip, + unsigned int ch, unsigned int count) +{ + int ret; + unsigned long src_base, src_pos, dst_mask; + unsigned char *dst_base; + int dst_pos; + u64 *src; + s16 *dst; + u64 x; + unsigned long flags; + struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT); + src_pos = readq(&mace->perif.audio.chan[ch].read_ptr); + dst_base = runtime->dma_area; + dst_pos = chip->channel[ch].pos; + dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1; + + /* check if a period has elapsed */ + chip->channel[ch].size += (count >> 3); /* in frames */ + ret = chip->channel[ch].size >= runtime->period_size; + chip->channel[ch].size %= runtime->period_size; + + while (count) { + src = (u64 *)(src_base + src_pos); + dst = (s16 *)(dst_base + dst_pos); + + x = *src; + dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff; + dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff; + + src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK; + dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask; + count -= sizeof(u64); + } + + writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */ + chip->channel[ch].pos = dst_pos; + + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return ret; +} + +/* put some DMA data in bounce buffer, count must be a multiple of 32 */ +/* returns 1 if a period has elapsed */ +static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip, + unsigned int ch, unsigned int count) +{ + int ret; + s64 l, r; + unsigned long dst_base, dst_pos, src_mask; + unsigned char *src_base; + int src_pos; + u64 *dst; + s16 *src; + unsigned long flags; + struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT); + dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr); + src_base = runtime->dma_area; + src_pos = chip->channel[ch].pos; + src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1; + + /* check if a period has elapsed */ + chip->channel[ch].size += (count >> 3); /* in frames */ + ret = chip->channel[ch].size >= runtime->period_size; + chip->channel[ch].size %= runtime->period_size; + + while (count) { + src = (s16 *)(src_base + src_pos); + dst = (u64 *)(dst_base + dst_pos); + + l = src[0]; /* sign extend */ + r = src[1]; /* sign extend */ + + *dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) | + ((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT); + + dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK; + src_pos = (src_pos + 2 * sizeof(s16)) & src_mask; + count -= sizeof(u64); + } + + writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */ + chip->channel[ch].pos = src_pos; + + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return ret; +} + +static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + int ch = chan->idx; + + /* reset DMA channel */ + writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control); + udelay(10); + writeq(0, &mace->perif.audio.chan[ch].control); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* push a full buffer */ + snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32); + } + /* set DMA to wake on 50% empty and enable interrupt */ + writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50, + &mace->perif.audio.chan[ch].control); + return 0; +} + +static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + + writeq(0, &mace->perif.audio.chan[chan->idx].control); + return 0; +} + +static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + struct snd_sgio2audio *chip; + int count, ch; + + substream = chan->substream; + chip = snd_pcm_substream_chip(substream); + ch = chan->idx; + + /* empty the ring */ + count = CHANNEL_RING_SIZE - + readq(&mace->perif.audio.chan[ch].depth) - 32; + if (snd_sgio2audio_dma_pull_frag(chip, ch, count)) + snd_pcm_period_elapsed(substream); + + return IRQ_HANDLED; +} + +static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + struct snd_sgio2audio *chip; + int count, ch; + + substream = chan->substream; + chip = snd_pcm_substream_chip(substream); + ch = chan->idx; + /* fill the ring */ + count = CHANNEL_RING_SIZE - + readq(&mace->perif.audio.chan[ch].depth) - 32; + if (snd_sgio2audio_dma_push_frag(chip, ch, count)) + snd_pcm_period_elapsed(substream); + + return IRQ_HANDLED; +} + +static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + + substream = chan->substream; + snd_sgio2audio_dma_stop(substream); + snd_sgio2audio_dma_start(substream); + return IRQ_HANDLED; +} + +/* PCM part */ +/* PCM hardware definition */ +static struct snd_pcm_hardware snd_sgio2audio_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER), + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 65536, + .period_bytes_min = 32768, + .period_bytes_max = 65536, + .periods_min = 1, + .periods_max = 1024, +}; + +/* PCM playback open callback */ +static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[1]; + return 0; +} + +static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[2]; + return 0; +} + +/* PCM capture open callback */ +static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[0]; + return 0; +} + +/* PCM close callback */ +static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->private_data = NULL; + return 0; +} + + +/* hw_params callback */ +static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int size = params_buffer_bytes(hw_params); + + /* alloc virtual 'dma' area */ + if (runtime->dma_area) + vfree(runtime->dma_area); + runtime->dma_area = vmalloc(size); + if (runtime->dma_area == NULL) + return -ENOMEM; + runtime->dma_bytes = size; + return 0; +} + +/* hw_free callback */ +static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream) +{ + if (substream->runtime->dma_area) + vfree(substream->runtime->dma_area); + substream->runtime->dma_area = NULL; + return 0; +} + +/* prepare callback */ +static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + int ch = chan->idx; + unsigned long flags; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + /* Setup the pseudo-dma transfer pointers. */ + chip->channel[ch].pos = 0; + chip->channel[ch].size = 0; + chip->channel[ch].substream = substream; + + /* set AD1843 format */ + /* hardware format is always S16_LE */ + switch (substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + ad1843_setup_dac(&chip->ad1843, + ch - 1, + runtime->rate, + SNDRV_PCM_FORMAT_S16_LE, + runtime->channels); + break; + case SNDRV_PCM_STREAM_CAPTURE: + ad1843_setup_adc(&chip->ad1843, + runtime->rate, + SNDRV_PCM_FORMAT_S16_LE, + runtime->channels); + break; + } + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return 0; +} + +/* trigger callback */ +static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* start the PCM engine */ + snd_sgio2audio_dma_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + /* stop the PCM engine */ + snd_sgio2audio_dma_stop(substream); + break; + default: + return -EINVAL; + } + return 0; +} + +/* pointer callback */ +static snd_pcm_uframes_t +snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + + /* get the current hardware pointer */ + return bytes_to_frames(substream->runtime, + chip->channel[chan->idx].pos); +} + +/* get the physical page pointer on the given offset */ +static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream, + unsigned long offset) +{ + return vmalloc_to_page(substream->runtime->dma_area + offset); +} + +/* operators */ +static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { + .open = snd_sgio2audio_playback1_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { + .open = snd_sgio2audio_playback2_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +static struct snd_pcm_ops snd_sgio2audio_capture_ops = { + .open = snd_sgio2audio_capture_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +/* + * definitions of capture are omitted here... + */ + +/* create a pcm device */ +static int __devinit snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip) +{ + struct snd_pcm *pcm; + int err; + + /* create first pcm device with one outputs and one input */ + err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm); + if (err < 0) + return err; + + pcm->private_data = chip; + strcpy(pcm->name, "SGI O2 DAC1"); + + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_sgio2audio_playback1_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_sgio2audio_capture_ops); + + /* create second pcm device with one outputs and no input */ + err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm); + if (err < 0) + return err; + + pcm->private_data = chip; + strcpy(pcm->name, "SGI O2 DAC2"); + + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_sgio2audio_playback2_ops); + + return 0; +} + +static struct { + int idx; + int irq; + irqreturn_t (*isr)(int, void *); + const char *desc; +} snd_sgio2_isr_table[] = { + { + .idx = 0, + .irq = MACEISA_AUDIO1_DMAT_IRQ, + .isr = snd_sgio2audio_dma_in_isr, + .desc = "Capture DMA Channel 0" + }, { + .idx = 0, + .irq = MACEISA_AUDIO1_OF_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Capture Overflow" + }, { + .idx = 1, + .irq = MACEISA_AUDIO2_DMAT_IRQ, + .isr = snd_sgio2audio_dma_out_isr, + .desc = "Playback DMA Channel 1" + }, { + .idx = 1, + .irq = MACEISA_AUDIO2_MERR_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Memory Error Channel 1" + }, { + .idx = 2, + .irq = MACEISA_AUDIO3_DMAT_IRQ, + .isr = snd_sgio2audio_dma_out_isr, + .desc = "Playback DMA Channel 2" + }, { + .idx = 2, + .irq = MACEISA_AUDIO3_MERR_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Memory Error Channel 2" + } +}; + +/* ALSA driver */ + +static int snd_sgio2audio_free(struct snd_sgio2audio *chip) +{ + int i; + + /* reset interface */ + writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control); + udelay(1); + writeq(0, &mace->perif.audio.control); + + /* release IRQ's */ + for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) + free_irq(snd_sgio2_isr_table[i].irq, + &chip->channel[snd_sgio2_isr_table[i].idx]); + + dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE, + chip->ring_base, chip->ring_base_dma); + + /* release card data */ + kfree(chip); + return 0; +} + +static int snd_sgio2audio_dev_free(struct snd_device *device) +{ + struct snd_sgio2audio *chip = device->device_data; + + return snd_sgio2audio_free(chip); +} + +static struct snd_device_ops ops = { + .dev_free = snd_sgio2audio_dev_free, +}; + +static int __devinit snd_sgio2audio_create(struct snd_card *card, + struct snd_sgio2audio **rchip) +{ + struct snd_sgio2audio *chip; + int i, err; + + *rchip = NULL; + + /* check if a codec is attached to the interface */ + /* (Audio or Audio/Video board present) */ + if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT)) + return -ENOENT; + + chip = kzalloc(sizeof(struct snd_sgio2audio), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + + chip->card = card; + + chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE, + &chip->ring_base_dma, GFP_USER); + if (chip->ring_base == NULL) { + printk(KERN_ERR + "sgio2audio: could not allocate ring buffers\n"); + kfree(chip); + return -ENOMEM; + } + + spin_lock_init(&chip->ad1843_lock); + + /* initialize channels */ + for (i = 0; i < 3; i++) { + spin_lock_init(&chip->channel[i].lock); + chip->channel[i].idx = i; + } + + /* allocate IRQs */ + for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) { + if (request_irq(snd_sgio2_isr_table[i].irq, + snd_sgio2_isr_table[i].isr, + 0, + snd_sgio2_isr_table[i].desc, + &chip->channel[snd_sgio2_isr_table[i].idx])) { + snd_sgio2audio_free(chip); + printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n", + snd_sgio2_isr_table[i].irq); + return -EBUSY; + } + } + + /* reset the interface */ + writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control); + udelay(1); + writeq(0, &mace->perif.audio.control); + msleep_interruptible(1); /* give time to recover */ + + /* set ring base */ + writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase); + + /* attach the AD1843 codec */ + chip->ad1843.read = read_ad1843_reg; + chip->ad1843.write = write_ad1843_reg; + chip->ad1843.chip = chip; + + /* initialize the AD1843 codec */ + err = ad1843_init(&chip->ad1843); + if (err < 0) { + snd_sgio2audio_free(chip); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_sgio2audio_free(chip); + return err; + } + *rchip = chip; + return 0; +} + +static int __devinit snd_sgio2audio_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct snd_sgio2audio *chip; + int err; + + card = snd_card_new(index, id, THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + err = snd_sgio2audio_create(card, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + snd_card_set_dev(card, &pdev->dev); + + err = snd_sgio2audio_new_pcm(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_sgio2audio_new_mixer(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "SGI O2 Audio"); + strcpy(card->shortname, "SGI O2 Audio"); + sprintf(card->longname, "%s irq %i-%i", + card->shortname, + MACEISA_AUDIO1_DMAT_IRQ, + MACEISA_AUDIO3_MERR_IRQ); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + platform_set_drvdata(pdev, card); + return 0; +} + +static int __exit snd_sgio2audio_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + + snd_card_free(card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver sgio2audio_driver = { + .probe = snd_sgio2audio_probe, + .remove = __devexit_p(snd_sgio2audio_remove), + .driver = { + .name = "sgio2audio", + .owner = THIS_MODULE, + } +}; + +static int __init alsa_card_sgio2audio_init(void) +{ + return platform_driver_register(&sgio2audio_driver); +} + +static void __exit alsa_card_sgio2audio_exit(void) +{ + platform_driver_unregister(&sgio2audio_driver); +} + +module_init(alsa_card_sgio2audio_init) +module_exit(alsa_card_sgio2audio_exit) From 80ca9a706b458d09b8cc8d5258bb61957f66ca5e Mon Sep 17 00:00:00 2001 From: Milton Miller Date: Sun, 13 Jul 2008 13:58:12 +0200 Subject: [PATCH 178/179] ALSA: correct kcalloc usage kcalloc is supposed to be called with the count as its first argument and the element size as the second. Both arguments are size_t so does not affect correctness. This callsite is during module_init and therefore not performance critical. Another patch will optimize the case when the count is variable but the size is fixed. Signed-off-by: Milton Miller Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/nm256/nm256.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 7efb838d18a6..06d13e717114 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1302,8 +1302,8 @@ snd_nm256_mixer(struct nm256 *chip) .read = snd_nm256_ac97_read, }; - chip->ac97_regs = kcalloc(sizeof(short), - ARRAY_SIZE(nm256_ac97_init_val), GFP_KERNEL); + chip->ac97_regs = kcalloc(ARRAY_SIZE(nm256_ac97_init_val), + sizeof(short), GFP_KERNEL); if (! chip->ac97_regs) return -ENOMEM; From fe0a3fe324811385b64790d42079bf534798a0cd Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 14 Jul 2008 09:54:43 +0200 Subject: [PATCH 179/179] ALSA: Release v1.0.17 Signed-off-by: Jaroslav Kysela --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/version.h b/include/sound/version.h index 29a5b227312d..6b78aff273a8 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.17rc3" +#define CONFIG_SND_VERSION "1.0.17" #define CONFIG_SND_DATE ""