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Merge remote-tracking branches 'asoc/topic/stac9766', 'asoc/topic/sti', 'asoc/topic/sti-codec', 'asoc/topic/sunxi' and 'asoc/topic/tegra' into asoc-next

This commit is contained in:
Mark Brown 2016-12-12 15:53:21 +00:00
parent f617134f75 7aacbc7ff7 4db61af068 92591efabc 4a15b24a65 8a7a282b78
commit da644e2ede
23 changed files with 1815 additions and 462 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

65
Documentation/devicetree/bindings/sound/sun4i-codec.txt
View File

@ -1,8 +1,12 @@
* Allwinner A10 Codec
Required properties:
- compatible: must be either "allwinner,sun4i-a10-codec" or
"allwinner,sun7i-a20-codec"
- compatible: must be one of the following compatibles:
- "allwinner,sun4i-a10-codec"
- "allwinner,sun6i-a31-codec"
- "allwinner,sun7i-a20-codec"
- "allwinner,sun8i-a23-codec"
- "allwinner,sun8i-h3-codec"
- reg: must contain the registers location and length
- interrupts: must contain the codec interrupt
- dmas: DMA channels for tx and rx dma. See the DMA client binding,
@ -17,6 +21,43 @@ Required properties:
Optional properties:
- allwinner,pa-gpios: gpio to enable external amplifier
Required properties for the following compatibles:
- "allwinner,sun6i-a31-codec"
- "allwinner,sun8i-a23-codec"
- "allwinner,sun8i-h3-codec"
- resets: phandle to the reset control for this device
- allwinner,audio-routing: A list of the connections between audio components.
Each entry is a pair of strings, the first being the
connection's sink, the second being the connection's
source. Valid names include:
Audio pins on the SoC:
"HP"
"HPCOM"
"LINEIN"
"LINEOUT" (not on sun8i-a23)
"MIC1"
"MIC2"
"MIC3" (sun6i-a31 only)
Microphone biases from the SoC:
"HBIAS"
"MBIAS"
Board connectors:
"Headphone"
"Headset Mic"
"Line In"
"Line Out"
"Mic"
"Speaker"
Required properties for the following compatibles:
- "allwinner,sun8i-a23-codec"
- "allwinner,sun8i-h3-codec"
- allwinner,codec-analog-controls: A phandle to the codec analog controls
block in the PRCM.
Example:
codec: codec@01c22c00 {
#sound-dai-cells = <0>;
@ -28,3 +69,23 @@ codec: codec@01c22c00 {
dmas = <&dma 0 19>, <&dma 0 19>;
dma-names = "rx", "tx";
};
codec: codec@01c22c00 {
#sound-dai-cells = <0>;
compatible = "allwinner,sun6i-a31-codec";
reg = <0x01c22c00 0x98>;
interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>;
clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>;
clock-names = "apb", "codec";
resets = <&ccu RST_APB1_CODEC>;
dmas = <&dma 15>, <&dma 15>;
dma-names = "rx", "tx";
allwinner,audio-routing =
"Headphone", "HP",
"Speaker", "LINEOUT",
"LINEIN", "Line In",
"MIC1", "MBIAS",
"MIC1", "Mic",
"MIC2", "HBIAS",
"MIC2", "Headset Mic";
};

16
Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt Normal file
View File

@ -0,0 +1,16 @@
* Allwinner Codec Analog Controls
Required properties:
- compatible: must be one of the following compatibles:
- "allwinner,sun8i-a23-codec-analog"
- "allwinner,sun8i-h3-codec-analog"
Required properties if not a sub-node of the PRCM node:
- reg: must contain the registers location and length
Example:
prcm: prcm@01f01400 {
codec_analog: codec-analog {
compatible = "allwinner,sun8i-a23-codec-analog";
};
};

162
sound/soc/codecs/stac9766.c
View File

@ -18,6 +18,7 @@
#include <linux/slab.h>
#include <linux/module.h>
#include <linux/device.h>
#include <linux/regmap.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
@ -26,31 +27,56 @@
#include <sound/soc.h>
#include <sound/tlv.h>
#include "stac9766.h"
#define STAC9766_VENDOR_ID 0x83847666
#define STAC9766_VENDOR_ID_MASK 0xffffffff
/*
* STAC9766 register cache
*/
static const u16 stac9766_reg[] = {
0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
0x0000, 0x0000, 0x8008, 0x8008, /* e */
0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
#define AC97_STAC_DA_CONTROL 0x6A
#define AC97_STAC_ANALOG_SPECIAL 0x6E
#define AC97_STAC_STEREO_MIC 0x78
static const struct reg_default stac9766_reg_defaults[] = {
{ 0x02, 0x8000 },
{ 0x04, 0x8000 },
{ 0x06, 0x8000 },
{ 0x0a, 0x0000 },
{ 0x0c, 0x8008 },
{ 0x0e, 0x8008 },
{ 0x10, 0x8808 },
{ 0x12, 0x8808 },
{ 0x14, 0x8808 },
{ 0x16, 0x8808 },
{ 0x18, 0x8808 },
{ 0x1a, 0x0000 },
{ 0x1c, 0x8000 },
{ 0x20, 0x0000 },
{ 0x22, 0x0000 },
{ 0x28, 0x0a05 },
{ 0x2c, 0xbb80 },
{ 0x32, 0xbb80 },
{ 0x3a, 0x2000 },
{ 0x3e, 0x0100 },
{ 0x4c, 0x0300 },
{ 0x4e, 0xffff },
{ 0x50, 0x0000 },
{ 0x52, 0x0000 },
{ 0x54, 0x0000 },
{ 0x6a, 0x0000 },
{ 0x6e, 0x1000 },
{ 0x72, 0x0000 },
{ 0x78, 0x0000 },
};
static const struct regmap_config stac9766_regmap_config = {
.reg_bits = 16,
.reg_stride = 2,
.val_bits = 16,
.max_register = 0x78,
.cache_type = REGCACHE_RBTREE,
.volatile_reg = regmap_ac97_default_volatile,
.reg_defaults = stac9766_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(stac9766_reg_defaults),
};
static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
@ -139,71 +165,22 @@ static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
};
static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int val)
{
struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
u16 *cache = codec->reg_cache;
if (reg > AC97_STAC_PAGE0) {
stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
soc_ac97_ops->write(ac97, reg, val);
stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
return 0;
}
if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
return -EIO;
soc_ac97_ops->write(ac97, reg, val);
cache[reg / 2] = val;
return 0;
}
static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
unsigned int reg)
{
struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
u16 val = 0, *cache = codec->reg_cache;
if (reg > AC97_STAC_PAGE0) {
stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
val = soc_ac97_ops->read(ac97, reg - AC97_STAC_PAGE0);
stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
return val;
}
if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
return -EIO;
if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
reg == AC97_VENDOR_ID2) {
val = soc_ac97_ops->read(ac97, reg);
return val;
}
return cache[reg / 2];
}
static int ac97_analog_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned short reg, vra;
unsigned short reg;
vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
vra |= 0x1; /* enable variable rate audio */
vra &= ~0x4; /* disable SPDIF output */
stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
/* enable variable rate audio, disable SPDIF output */
snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x5, 0x1);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
reg = AC97_PCM_FRONT_DAC_RATE;
else
reg = AC97_PCM_LR_ADC_RATE;
return stac9766_ac97_write(codec, reg, runtime->rate);
return snd_soc_write(codec, reg, runtime->rate);
}
static int ac97_digital_prepare(struct snd_pcm_substream *substream,
@ -211,18 +188,16 @@ static int ac97_digital_prepare(struct snd_pcm_substream *substream,
{
struct snd_soc_codec *codec = dai->codec;
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned short reg, vra;
unsigned short reg;
stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);
snd_soc_write(codec, AC97_SPDIF, 0x2002);
vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
vra |= 0x5; /* Enable VRA and SPDIF out */
stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
/* Enable VRA and SPDIF out */
snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x5, 0x5);
reg = AC97_PCM_FRONT_DAC_RATE;
return stac9766_ac97_write(codec, reg, runtime->rate);
return snd_soc_write(codec, reg, runtime->rate);
}
static int stac9766_set_bias_level(struct snd_soc_codec *codec,
@ -232,11 +207,11 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_ON: /* full On */
case SND_SOC_BIAS_PREPARE: /* partial On */
case SND_SOC_BIAS_STANDBY: /* Off, with power */
stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
snd_soc_write(codec, AC97_POWERDOWN, 0x0000);
break;
case SND_SOC_BIAS_OFF: /* Off, without power */
/* disable everything including AC link */
stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
snd_soc_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
return 0;
@ -300,21 +275,34 @@ static struct snd_soc_dai_driver stac9766_dai[] = {
static int stac9766_codec_probe(struct snd_soc_codec *codec)
{
struct snd_ac97 *ac97;
struct regmap *regmap;
int ret;
ac97 = snd_soc_new_ac97_codec(codec, STAC9766_VENDOR_ID,
STAC9766_VENDOR_ID_MASK);
if (IS_ERR(ac97))
return PTR_ERR(ac97);
regmap = regmap_init_ac97(ac97, &stac9766_regmap_config);
if (IS_ERR(regmap)) {
ret = PTR_ERR(regmap);
goto err_free_ac97;
}
snd_soc_codec_init_regmap(codec, regmap);
snd_soc_codec_set_drvdata(codec, ac97);
return 0;
err_free_ac97:
snd_soc_free_ac97_codec(ac97);
return ret;
}
static int stac9766_codec_remove(struct snd_soc_codec *codec)
{
struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
snd_soc_codec_exit_regmap(codec);
snd_soc_free_ac97_codec(ac97);
return 0;
}
@ -324,17 +312,11 @@ static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
.controls = stac9766_snd_ac97_controls,
.num_controls = ARRAY_SIZE(stac9766_snd_ac97_controls),
},
.write = stac9766_ac97_write,
.read = stac9766_ac97_read,
.set_bias_level = stac9766_set_bias_level,
.suspend_bias_off = true,
.probe = stac9766_codec_probe,
.remove = stac9766_codec_remove,
.resume = stac9766_codec_resume,
.reg_cache_size = ARRAY_SIZE(stac9766_reg),
.reg_word_size = sizeof(u16),
.reg_cache_step = 2,
.reg_cache_default = stac9766_reg,
};
static int stac9766_probe(struct platform_device *pdev)

17
sound/soc/codecs/stac9766.h
View File

@ -1,17 +0,0 @@
/*
* stac9766.h -- STAC9766 Soc Audio driver
*/
#ifndef _STAC9766_H
#define _STAC9766_H
#define AC97_STAC_PAGE0 0x1000
#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A)
#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E)
#define AC97_STAC_STEREO_MIC 0x78
/* STAC9766 DAI ID's */
#define STAC9766_DAI_AC97_ANALOG 0
#define STAC9766_DAI_AC97_DIGITAL 1
#endif

153
sound/soc/codecs/sti-sas.c
View File

@ -14,28 +14,8 @@
#include <sound/soc.h>
#include <sound/soc-dapm.h>
/* chipID supported */
#define CHIPID_STIH416 0
#define CHIPID_STIH407 1
/* DAC definitions */
/* stih416 DAC registers */
/* sysconf 2517: Audio-DAC-Control */
#define STIH416_AUDIO_DAC_CTRL 0x00000814
/* sysconf 2519: Audio-Gue-Control */
#define STIH416_AUDIO_GLUE_CTRL 0x0000081C
#define STIH416_DAC_NOT_STANDBY 0x3
#define STIH416_DAC_SOFTMUTE 0x4
#define STIH416_DAC_ANA_NOT_PWR 0x5
#define STIH416_DAC_NOT_PNDBG 0x6
#define STIH416_DAC_NOT_STANDBY_MASK BIT(STIH416_DAC_NOT_STANDBY)
#define STIH416_DAC_SOFTMUTE_MASK BIT(STIH416_DAC_SOFTMUTE)
#define STIH416_DAC_ANA_NOT_PWR_MASK BIT(STIH416_DAC_ANA_NOT_PWR)
#define STIH416_DAC_NOT_PNDBG_MASK BIT(STIH416_DAC_NOT_PNDBG)
/* stih407 DAC registers */
/* sysconf 5041: Audio-Gue-Control */
#define STIH407_AUDIO_GLUE_CTRL 0x000000A4
@ -63,14 +43,9 @@ enum {
STI_SAS_DAI_ANALOG_OUT,
};
static const struct reg_default stih416_sas_reg_defaults[] = {
{ STIH407_AUDIO_GLUE_CTRL, 0x00000040 },
{ STIH407_AUDIO_DAC_CTRL, 0x000000000 },
};
static const struct reg_default stih407_sas_reg_defaults[] = {
{ STIH416_AUDIO_DAC_CTRL, 0x000000000 },
{ STIH416_AUDIO_GLUE_CTRL, 0x00000040 },
{ STIH407_AUDIO_DAC_CTRL, 0x000000000 },
{ STIH407_AUDIO_GLUE_CTRL, 0x00000040 },
};
struct sti_dac_audio {
@ -89,7 +64,6 @@ struct sti_spdif_audio {
/* device data structure */
struct sti_sas_dev_data {
const int chipid; /* IC version */
const struct regmap_config *regmap;
const struct snd_soc_dai_ops *dac_ops; /* DAC function callbacks */
const struct snd_soc_dapm_widget *dapm_widgets; /* dapms declaration */
@ -150,13 +124,11 @@ static int sti_sas_init_sas_registers(struct snd_soc_codec *codec,
ret = snd_soc_update_bits(codec, STIH407_AUDIO_GLUE_CTRL,
SPDIF_BIPHASE_IDLE_MASK, 0);
if (ret < 0) {
dev_err(codec->dev, "Failed to update SPDIF registers");
dev_err(codec->dev, "Failed to update SPDIF registers\n");
return ret;
}
/* Init DAC configuration */
switch (data->dev_data->chipid) {
case CHIPID_STIH407:
/* init configuration */
ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
STIH407_DAC_STANDBY_MASK,
@ -170,31 +142,9 @@ static int sti_sas_init_sas_registers(struct snd_soc_codec *codec,
ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
STIH407_DAC_SOFTMUTE_MASK,
STIH407_DAC_SOFTMUTE_MASK);
break;
case CHIPID_STIH416:
ret = snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_NOT_STANDBY_MASK, 0);
if (!ret)
ret = snd_soc_update_bits(codec,
STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_ANA_NOT_PWR, 0);
if (!ret)
ret = snd_soc_update_bits(codec,
STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_NOT_PNDBG_MASK,
0);
if (!ret)
ret = snd_soc_update_bits(codec,
STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_SOFTMUTE_MASK,
STIH416_DAC_SOFTMUTE_MASK);
break;
default:
return -EINVAL;
}
if (ret < 0) {
dev_err(codec->dev, "Failed to update DAC registers");
dev_err(codec->dev, "Failed to update DAC registers\n");
return ret;
}
@ -217,37 +167,6 @@ static int sti_sas_dac_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return 0;
}
static int stih416_dac_probe(struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev);
struct sti_dac_audio *dac = &drvdata->dac;
/* Get reset control */
dac->rst = devm_reset_control_get(codec->dev, "dac_rst");
if (IS_ERR(dac->rst)) {
dev_err(dai->codec->dev,
"%s: ERROR: DAC reset control not defined !\n",
__func__);
dac->rst = NULL;
return -EFAULT;
}
/* Put the DAC into reset */
reset_control_assert(dac->rst);
return 0;
}
static const struct snd_soc_dapm_widget stih416_sas_dapm_widgets[] = {
SND_SOC_DAPM_PGA("DAC bandgap", STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_NOT_PNDBG_MASK, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV("DAC standby ana", STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_ANA_NOT_PWR, 0, NULL, 0),
SND_SOC_DAPM_DAC("DAC standby", "dac_p", STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_NOT_STANDBY, 0),
SND_SOC_DAPM_OUTPUT("DAC Output"),
};
static const struct snd_soc_dapm_widget stih407_sas_dapm_widgets[] = {
SND_SOC_DAPM_OUT_DRV("DAC standby ana", STIH407_AUDIO_DAC_CTRL,
STIH407_DAC_STANDBY_ANA, 1, NULL, 0),
@ -256,30 +175,11 @@ static const struct snd_soc_dapm_widget stih407_sas_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("DAC Output"),
};
static const struct snd_soc_dapm_route stih416_sas_route[] = {
{"DAC Output", NULL, "DAC bandgap"},
{"DAC Output", NULL, "DAC standby ana"},
{"DAC standby ana", NULL, "DAC standby"},
};
static const struct snd_soc_dapm_route stih407_sas_route[] = {
{"DAC Output", NULL, "DAC standby ana"},
{"DAC standby ana", NULL, "DAC standby"},
};
static int stih416_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream)
{
struct snd_soc_codec *codec = dai->codec;
if (mute) {
return snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_SOFTMUTE_MASK,
STIH416_DAC_SOFTMUTE_MASK);
} else {
return snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_SOFTMUTE_MASK, 0);
}
}
static int stih407_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream)
{
@ -392,13 +292,13 @@ static int sti_sas_prepare(struct snd_pcm_substream *substream,
switch (dai->id) {
case STI_SAS_DAI_SPDIF_OUT:
if ((drvdata->spdif.mclk / runtime->rate) != 128) {
dev_err(codec->dev, "unexpected mclk-fs ratio");
dev_err(codec->dev, "unexpected mclk-fs ratio\n");
return -EINVAL;
}
break;
case STI_SAS_DAI_ANALOG_OUT:
if ((drvdata->dac.mclk / runtime->rate) != 256) {
dev_err(codec->dev, "unexpected mclk-fs ratio");
dev_err(codec->dev, "unexpected mclk-fs ratio\n");
return -EINVAL;
}
break;
@ -407,13 +307,6 @@ static int sti_sas_prepare(struct snd_pcm_substream *substream,
return 0;
}
static const struct snd_soc_dai_ops stih416_dac_ops = {
.set_fmt = sti_sas_dac_set_fmt,
.mute_stream = stih416_sas_dac_mute,
.prepare = sti_sas_prepare,
.set_sysclk = sti_sas_set_sysclk,
};
static const struct snd_soc_dai_ops stih407_dac_ops = {
.set_fmt = sti_sas_dac_set_fmt,
.mute_stream = stih407_sas_dac_mute,
@ -434,31 +327,7 @@ static const struct regmap_config stih407_sas_regmap = {
.reg_write = sti_sas_write_reg,
};
static const struct regmap_config stih416_sas_regmap = {
.reg_bits = 32,
.val_bits = 32,
.max_register = STIH416_AUDIO_DAC_CTRL,
.reg_defaults = stih416_sas_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(stih416_sas_reg_defaults),
.volatile_reg = sti_sas_volatile_register,
.cache_type = REGCACHE_RBTREE,
.reg_read = sti_sas_read_reg,
.reg_write = sti_sas_write_reg,
};
static const struct sti_sas_dev_data stih416_data = {
.chipid = CHIPID_STIH416,
.regmap = &stih416_sas_regmap,
.dac_ops = &stih416_dac_ops,
.dapm_widgets = stih416_sas_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(stih416_sas_dapm_widgets),
.dapm_routes = stih416_sas_route,
.num_dapm_routes = ARRAY_SIZE(stih416_sas_route),
};
static const struct sti_sas_dev_data stih407_data = {
.chipid = CHIPID_STIH407,
.regmap = &stih407_sas_regmap,
.dac_ops = &stih407_dac_ops,
.dapm_widgets = stih407_sas_dapm_widgets,
@ -532,10 +401,6 @@ static struct snd_soc_codec_driver sti_sas_driver = {
};
static const struct of_device_id sti_sas_dev_match[] = {
{
.compatible = "st,stih416-sas-codec",
.data = &stih416_data,
},
{
.compatible = "st,stih407-sas-codec",
.data = &stih407_data,
@ -558,7 +423,7 @@ static int sti_sas_driver_probe(struct platform_device *pdev)
/* Populate data structure depending on compatibility */
of_id = of_match_node(sti_sas_dev_match, pnode);
if (!of_id->data) {
dev_err(&pdev->dev, "data associated to device is missing");
dev_err(&pdev->dev, "data associated to device is missing\n");
return -EINVAL;
}
@ -584,10 +449,6 @@ static int sti_sas_driver_probe(struct platform_device *pdev)
}
drvdata->spdif.regmap = drvdata->dac.regmap;
/* Set DAC dai probe */
if (drvdata->dev_data->chipid == CHIPID_STIH416)
sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].probe = stih416_dac_probe;
sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].ops = drvdata->dev_data->dac_ops;
/* Set dapms*/

1
sound/soc/fsl/efika-audio-fabric.c
View File

@ -27,7 +27,6 @@
#include "mpc5200_dma.h"
#include "mpc5200_psc_ac97.h"
#include "../codecs/stac9766.h"
#define DRV_NAME "efika-audio-fabric"

43
sound/soc/sti/sti_uniperif.c
View File

@ -7,6 +7,7 @@
#include <linux/module.h>
#include <linux/pinctrl/consumer.h>
#include <linux/delay.h>
#include "uniperif.h"
@ -97,6 +98,28 @@ static const struct of_device_id snd_soc_sti_match[] = {
{},
};
int sti_uniperiph_reset(struct uniperif *uni)
{
int count = 10;
/* Reset uniperipheral uni */
SET_UNIPERIF_SOFT_RST_SOFT_RST(uni);
if (uni->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) {
while (GET_UNIPERIF_SOFT_RST_SOFT_RST(uni) && count) {
udelay(5);
count--;
}
}
if (!count) {
dev_err(uni->dev, "Failed to reset uniperif\n");
return -EIO;
}
return 0;
}
int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
unsigned int rx_mask, int slots,
int slot_width)
@ -293,7 +316,7 @@ static int sti_uniperiph_dai_suspend(struct snd_soc_dai *dai)
/* The uniperipheral should be in stopped state */
if (uni->state != UNIPERIF_STATE_STOPPED) {
dev_err(uni->dev, "%s: invalid uni state( %d)",
dev_err(uni->dev, "%s: invalid uni state( %d)\n",
__func__, (int)uni->state);
return -EBUSY;
}
@ -301,7 +324,7 @@ static int sti_uniperiph_dai_suspend(struct snd_soc_dai *dai)
/* Pinctrl: switch pinstate to sleep */
ret = pinctrl_pm_select_sleep_state(uni->dev);
if (ret)
dev_err(uni->dev, "%s: failed to select pinctrl state",
dev_err(uni->dev, "%s: failed to select pinctrl state\n",
__func__);
return ret;
@ -322,7 +345,7 @@ static int sti_uniperiph_dai_resume(struct snd_soc_dai *dai)
/* pinctrl: switch pinstate to default */
ret = pinctrl_pm_select_default_state(uni->dev);
if (ret)
dev_err(uni->dev, "%s: failed to select pinctrl state",
dev_err(uni->dev, "%s: failed to select pinctrl state\n",
__func__);
return ret;
@ -366,11 +389,12 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
const struct of_device_id *of_id;
const struct sti_uniperiph_dev_data *dev_data;
const char *mode;
int ret;
/* Populate data structure depending on compatibility */
of_id = of_match_node(snd_soc_sti_match, node);
if (!of_id->data) {
dev_err(dev, "data associated to device is missing");
dev_err(dev, "data associated to device is missing\n");
return -EINVAL;
}
dev_data = (struct sti_uniperiph_dev_data *)of_id->data;
@ -389,7 +413,7 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
uni->mem_region = platform_get_resource(priv->pdev, IORESOURCE_MEM, 0);
if (!uni->mem_region) {
dev_err(dev, "Failed to get memory resource");
dev_err(dev, "Failed to get memory resource\n");
return -ENODEV;
}
@ -403,7 +427,7 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
uni->irq = platform_get_irq(priv->pdev, 0);
if (uni->irq < 0) {
dev_err(dev, "Failed to get IRQ resource");
dev_err(dev, "Failed to get IRQ resource\n");
return -ENXIO;
}
@ -421,12 +445,15 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
dai_data->stream = dev_data->stream;
if (priv->dai_data.stream == SNDRV_PCM_STREAM_PLAYBACK) {
uni_player_init(priv->pdev, uni);
ret = uni_player_init(priv->pdev, uni);
stream = &dai->playback;
} else {
uni_reader_init(priv->pdev, uni);
ret = uni_reader_init(priv->pdev, uni);
stream = &dai->capture;
}
if (ret < 0)
return ret;
dai->ops = uni->dai_ops;
stream->stream_name = dai->name;

2
sound/soc/sti/uniperif.h
View File

@ -1397,6 +1397,8 @@ static inline int sti_uniperiph_get_unip_tdm_frame_size(struct uniperif *uni)
return (uni->tdm_slot.slots * uni->tdm_slot.slot_width / 8);
}
int sti_uniperiph_reset(struct uniperif *uni);
int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
unsigned int rx_mask, int slots,
int slot_width);

91
sound/soc/sti/uniperif_player.c
View File

@ -6,8 +6,6 @@
*/
#include <linux/clk.h>
#include <linux/delay.h>
#include <linux/io.h>
#include <linux/mfd/syscon.h>
#include <sound/asoundef.h>
@ -55,25 +53,6 @@ static const struct snd_pcm_hardware uni_player_pcm_hw = {
.buffer_bytes_max = 256 * PAGE_SIZE
};
static inline int reset_player(struct uniperif *player)
{
int count = 10;
if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) {
while (GET_UNIPERIF_SOFT_RST_SOFT_RST(player) && count) {
udelay(5);
count--;
}
}
if (!count) {
dev_err(player->dev, "Failed to reset uniperif");
return -EIO;
}
return 0;
}
/*
* uni_player_irq_handler
* In case of error audio stream is stopped; stop action is protected via PCM
@ -97,7 +76,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
/* Check for fifo error (underrun) */
if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(player))) {
dev_err(player->dev, "FIFO underflow error detected");
dev_err(player->dev, "FIFO underflow error detected\n");
/* Interrupt is just for information when underflow recovery */
if (player->underflow_enabled) {
@ -119,7 +98,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
/* Check for dma error (overrun) */
if (unlikely(status & UNIPERIF_ITS_DMA_ERROR_MASK(player))) {
dev_err(player->dev, "DMA error detected");
dev_err(player->dev, "DMA error detected\n");
/* Disable interrupt so doesn't continually fire */
SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player);
@ -135,11 +114,14 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
/* Check for underflow recovery done */
if (unlikely(status & UNIPERIF_ITM_UNDERFLOW_REC_DONE_MASK(player))) {
if (!player->underflow_enabled) {
dev_err(player->dev, "unexpected Underflow recovering");
dev_err(player->dev,
"unexpected Underflow recovering\n");
return -EPERM;
}
/* Read the underflow recovery duration */
tmp = GET_UNIPERIF_STATUS_1_UNDERFLOW_DURATION(player);
dev_dbg(player->dev, "Underflow recovered (%d LR clocks max)\n",
tmp);
/* Clear the underflow recovery duration */
SET_UNIPERIF_BIT_CONTROL_CLR_UNDERFLOW_DURATION(player);
@ -153,7 +135,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
/* Check if underflow recovery failed */
if (unlikely(status &
UNIPERIF_ITM_UNDERFLOW_REC_FAILED_MASK(player))) {
dev_err(player->dev, "Underflow recovery failed");
dev_err(player->dev, "Underflow recovery failed\n");
/* Stop the player */
snd_pcm_stream_lock(player->substream);
@ -336,7 +318,7 @@ static int uni_player_prepare_iec958(struct uniperif *player,
/* Oversampling must be multiple of 128 as iec958 frame is 32-bits */
if ((clk_div % 128) || (clk_div <= 0)) {
dev_err(player->dev, "%s: invalid clk_div %d",
dev_err(player->dev, "%s: invalid clk_div %d\n",
__func__, clk_div);
return -EINVAL;
}
@ -359,7 +341,7 @@ static int uni_player_prepare_iec958(struct uniperif *player,
SET_UNIPERIF_I2S_FMT_DATA_SIZE_24(player);
break;
default:
dev_err(player->dev, "format not supported");
dev_err(player->dev, "format not supported\n");
return -EINVAL;
}
@ -448,12 +430,12 @@ static int uni_player_prepare_pcm(struct uniperif *player,
* for 16 bits must be a multiple of 64
*/
if ((slot_width == 32) && (clk_div % 128)) {
dev_err(player->dev, "%s: invalid clk_div", __func__);
dev_err(player->dev, "%s: invalid clk_div\n", __func__);
return -EINVAL;
}
if ((slot_width == 16) && (clk_div % 64)) {
dev_err(player->dev, "%s: invalid clk_div", __func__);
dev_err(player->dev, "%s: invalid clk_div\n", __func__);
return -EINVAL;
}
@ -471,7 +453,7 @@ static int uni_player_prepare_pcm(struct uniperif *player,
SET_UNIPERIF_I2S_FMT_DATA_SIZE_16(player);
break;
default:
dev_err(player->dev, "subframe format not supported");
dev_err(player->dev, "subframe format not supported\n");
return -EINVAL;
}
@ -491,7 +473,7 @@ static int uni_player_prepare_pcm(struct uniperif *player,
break;
default:
dev_err(player->dev, "format not supported");
dev_err(player->dev, "format not supported\n");
return -EINVAL;
}
@ -504,7 +486,7 @@ static int uni_player_prepare_pcm(struct uniperif *player,
/* Number of channelsmust be even*/
if ((runtime->channels % 2) || (runtime->channels < 2) ||
(runtime->channels > 10)) {
dev_err(player->dev, "%s: invalid nb of channels", __func__);
dev_err(player->dev, "%s: invalid nb of channels\n", __func__);
return -EINVAL;
}
@ -762,7 +744,7 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
/* The player should be stopped */
if (player->state != UNIPERIF_STATE_STOPPED) {
dev_err(player->dev, "%s: invalid player state %d", __func__,
dev_err(player->dev, "%s: invalid player state %d\n", __func__,
player->state);
return -EINVAL;
}
@ -791,7 +773,8 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
/* Trigger limit must be an even number */
if ((!trigger_limit % 2) || (trigger_limit != 1 && transfer_size % 2) ||
(trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(player))) {
dev_err(player->dev, "invalid trigger limit %d", trigger_limit);
dev_err(player->dev, "invalid trigger limit %d\n",
trigger_limit);
return -EINVAL;
}
@ -812,7 +795,7 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
ret = uni_player_prepare_tdm(player, runtime);
break;
default:
dev_err(player->dev, "invalid player type");
dev_err(player->dev, "invalid player type\n");
return -EINVAL;
}
@ -852,16 +835,14 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
SET_UNIPERIF_I2S_FMT_PADDING_SONY_MODE(player);
break;
default:
dev_err(player->dev, "format not supported");
dev_err(player->dev, "format not supported\n");
return -EINVAL;
}
SET_UNIPERIF_I2S_FMT_NO_OF_SAMPLES_TO_READ(player, 0);
/* Reset uniperipheral player */
SET_UNIPERIF_SOFT_RST_SOFT_RST(player);
return reset_player(player);
return sti_uniperiph_reset(player);
}
static int uni_player_start(struct uniperif *player)
@ -870,13 +851,13 @@ static int uni_player_start(struct uniperif *player)
/* The player should be stopped */
if (player->state != UNIPERIF_STATE_STOPPED) {
dev_err(player->dev, "%s: invalid player state", __func__);
dev_err(player->dev, "%s: invalid player state\n", __func__);
return -EINVAL;
}
ret = clk_prepare_enable(player->clk);
if (ret) {
dev_err(player->dev, "%s: Failed to enable clock", __func__);
dev_err(player->dev, "%s: Failed to enable clock\n", __func__);
return ret;
}
@ -893,10 +874,7 @@ static int uni_player_start(struct uniperif *player)
SET_UNIPERIF_ITM_BSET_UNDERFLOW_REC_FAILED(player);
}
/* Reset uniperipheral player */
SET_UNIPERIF_SOFT_RST_SOFT_RST(player);
ret = reset_player(player);
ret = sti_uniperiph_reset(player);
if (ret < 0) {
clk_disable_unprepare(player->clk);
return ret;
@ -938,17 +916,14 @@ static int uni_player_stop(struct uniperif *player)
/* The player should not be in stopped state */
if (player->state == UNIPERIF_STATE_STOPPED) {
dev_err(player->dev, "%s: invalid player state", __func__);
dev_err(player->dev, "%s: invalid player state\n", __func__);
return -EINVAL;
}
/* Turn the player off */
SET_UNIPERIF_CTRL_OPERATION_OFF(player);
/* Soft reset the player */
SET_UNIPERIF_SOFT_RST_SOFT_RST(player);
ret = reset_player(player);
ret = sti_uniperiph_reset(player);
if (ret < 0)
return ret;
@ -973,7 +948,7 @@ int uni_player_resume(struct uniperif *player)
ret = regmap_field_write(player->clk_sel, 1);
if (ret) {
dev_err(player->dev,
"%s: Failed to select freq synth clock",
"%s: Failed to select freq synth clock\n",
__func__);
return ret;
}
@ -1070,7 +1045,7 @@ int uni_player_init(struct platform_device *pdev,
ret = uni_player_parse_dt_audio_glue(pdev, player);
if (ret < 0) {
dev_err(player->dev, "Failed to parse DeviceTree");
dev_err(player->dev, "Failed to parse DeviceTree\n");
return ret;
}
@ -1085,15 +1060,17 @@ int uni_player_init(struct platform_device *pdev,
/* Get uniperif resource */
player->clk = of_clk_get(pdev->dev.of_node, 0);
if (IS_ERR(player->clk))
if (IS_ERR(player->clk)) {
dev_err(player->dev, "Failed to get clock\n");
ret = PTR_ERR(player->clk);
}
/* Select the frequency synthesizer clock */
if (player->clk_sel) {
ret = regmap_field_write(player->clk_sel, 1);
if (ret) {
dev_err(player->dev,
"%s: Failed to select freq synth clock",
"%s: Failed to select freq synth clock\n",
__func__);
return ret;
}
@ -1105,7 +1082,7 @@ int uni_player_init(struct platform_device *pdev,
ret = regmap_field_write(player->valid_sel, player->id);
if (ret) {
dev_err(player->dev,
"%s: unable to connect to tdm bus", __func__);
"%s: unable to connect to tdm bus\n", __func__);
return ret;
}
}
@ -1113,8 +1090,10 @@ int uni_player_init(struct platform_device *pdev,
ret = devm_request_irq(&pdev->dev, player->irq,
uni_player_irq_handler, IRQF_SHARED,
dev_name(&pdev->dev), player);
if (ret < 0)
if (ret < 0) {
dev_err(player->dev, "unable to request IRQ %d\n", player->irq);
return ret;
}
mutex_init(&player->ctrl_lock);

41
sound/soc/sti/uniperif_reader.c
View File

@ -5,10 +5,6 @@
* License terms: GNU General Public License (GPL), version 2
*/
#include <linux/clk.h>
#include <linux/delay.h>
#include <linux/io.h>
#include <sound/soc.h>
#include "uniperif.h"
@ -52,7 +48,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id)
if (reader->state == UNIPERIF_STATE_STOPPED) {
/* Unexpected IRQ: do nothing */
dev_warn(reader->dev, "unexpected IRQ ");
dev_warn(reader->dev, "unexpected IRQ\n");
return IRQ_HANDLED;
}
@ -62,7 +58,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id)
/* Check for fifo overflow error */
if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) {
dev_err(reader->dev, "FIFO error detected");
dev_err(reader->dev, "FIFO error detected\n");
snd_pcm_stream_lock(reader->substream);
snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN);
@ -105,7 +101,7 @@ static int uni_reader_prepare_pcm(struct snd_pcm_runtime *runtime,
SET_UNIPERIF_I2S_FMT_DATA_SIZE_16(reader);
break;
default:
dev_err(reader->dev, "subframe format not supported");
dev_err(reader->dev, "subframe format not supported\n");
return -EINVAL;
}
@ -125,14 +121,14 @@ static int uni_reader_prepare_pcm(struct snd_pcm_runtime *runtime,
break;
default:
dev_err(reader->dev, "format not supported");
dev_err(reader->dev, "format not supported\n");
return -EINVAL;
}
/* Number of channels must be even */
if ((runtime->channels % 2) || (runtime->channels < 2) ||
(runtime->channels > 10)) {
dev_err(reader->dev, "%s: invalid nb of channels", __func__);
dev_err(reader->dev, "%s: invalid nb of channels\n", __func__);
return -EINVAL;
}
@ -186,11 +182,10 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
struct uniperif *reader = priv->dai_data.uni;
struct snd_pcm_runtime *runtime = substream->runtime;
int transfer_size, trigger_limit, ret;
int count = 10;
/* The reader should be stopped */
if (reader->state != UNIPERIF_STATE_STOPPED) {
dev_err(reader->dev, "%s: invalid reader state %d", __func__,
dev_err(reader->dev, "%s: invalid reader state %d\n", __func__,
reader->state);
return -EINVAL;
}
@ -219,7 +214,8 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
if ((!trigger_limit % 2) ||
(trigger_limit != 1 && transfer_size % 2) ||
(trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(reader))) {
dev_err(reader->dev, "invalid trigger limit %d", trigger_limit);
dev_err(reader->dev, "invalid trigger limit %d\n",
trigger_limit);
return -EINVAL;
}
@ -246,7 +242,7 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
SET_UNIPERIF_I2S_FMT_PADDING_SONY_MODE(reader);
break;
default:
dev_err(reader->dev, "format not supported");
dev_err(reader->dev, "format not supported\n");
return -EINVAL;
}
@ -287,25 +283,14 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
}
/* Reset uniperipheral reader */
SET_UNIPERIF_SOFT_RST_SOFT_RST(reader);
while (GET_UNIPERIF_SOFT_RST_SOFT_RST(reader)) {
udelay(5);
count--;
}
if (!count) {
dev_err(reader->dev, "Failed to reset uniperif");
return -EIO;
}
return 0;
return sti_uniperiph_reset(reader);
}
static int uni_reader_start(struct uniperif *reader)
{
/* The reader should be stopped */
if (reader->state != UNIPERIF_STATE_STOPPED) {
dev_err(reader->dev, "%s: invalid reader state", __func__);
dev_err(reader->dev, "%s: invalid reader state\n", __func__);
return -EINVAL;
}
@ -325,7 +310,7 @@ static int uni_reader_stop(struct uniperif *reader)
{
/* The reader should not be in stopped state */
if (reader->state == UNIPERIF_STATE_STOPPED) {
dev_err(reader->dev, "%s: invalid reader state", __func__);
dev_err(reader->dev, "%s: invalid reader state\n", __func__);
return -EINVAL;
}
@ -423,7 +408,7 @@ int uni_reader_init(struct platform_device *pdev,
uni_reader_irq_handler, IRQF_SHARED,
dev_name(&pdev->dev), reader);
if (ret < 0) {
dev_err(&pdev->dev, "Failed to request IRQ");
dev_err(&pdev->dev, "Failed to request IRQ\n");
return -EBUSY;
}

8
sound/soc/sunxi/Kconfig
View File

@ -9,6 +9,14 @@ config SND_SUN4I_CODEC
Select Y or M to add support for the Codec embedded in the Allwinner
A10 and affiliated SoCs.
config SND_SUN8I_CODEC_ANALOG
tristate "Allwinner sun8i Codec Analog Controls Support"
depends on MACH_SUN8I || COMPILE_TEST
select REGMAP
help
Say Y or M if you want to add support for the analog controls for
the codec embedded in newer Allwinner SoCs.
config SND_SUN4I_I2S
tristate "Allwinner A10 I2S Support"
select SND_SOC_GENERIC_DMAENGINE_PCM

1
sound/soc/sunxi/Makefile
View File

@ -1,3 +1,4 @@
obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o
obj-$(CONFIG_SND_SUN4I_I2S) += sun4i-i2s.o
obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o
obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o

835
sound/soc/sunxi/sun4i-codec.c
View File

File diff suppressed because it is too large Load Diff

101
sound/soc/sunxi/sun4i-i2s.c
View File

@ -93,6 +93,9 @@ struct sun4i_i2s {
struct clk *mod_clk;
struct regmap *regmap;
unsigned int mclk_freq;
struct snd_dmaengine_dai_dma_data capture_dma_data;
struct snd_dmaengine_dai_dma_data playback_dma_data;
};
@ -157,14 +160,24 @@ static int sun4i_i2s_get_mclk_div(struct sun4i_i2s *i2s,
}
static int sun4i_i2s_oversample_rates[] = { 128, 192, 256, 384, 512, 768 };
static bool sun4i_i2s_oversample_is_valid(unsigned int oversample)
{
int i;
for (i = 0; i < ARRAY_SIZE(sun4i_i2s_oversample_rates); i++)
if (sun4i_i2s_oversample_rates[i] == oversample)
return true;
return false;
}
static int sun4i_i2s_set_clk_rate(struct sun4i_i2s *i2s,
unsigned int rate,
unsigned int word_size)
{
unsigned int clk_rate;
unsigned int oversample_rate, clk_rate;
int bclk_div, mclk_div;
int ret, i;
int ret;
switch (rate) {
case 176400:
@ -196,21 +209,18 @@ static int sun4i_i2s_set_clk_rate(struct sun4i_i2s *i2s,
if (ret)
return ret;
/* Always favor the highest oversampling rate */
for (i = (ARRAY_SIZE(sun4i_i2s_oversample_rates) - 1); i >= 0; i--) {
unsigned int oversample_rate = sun4i_i2s_oversample_rates[i];
oversample_rate = i2s->mclk_freq / rate;
if (!sun4i_i2s_oversample_is_valid(oversample_rate))
return -EINVAL;
bclk_div = sun4i_i2s_get_bclk_div(i2s, oversample_rate,
word_size);
if (bclk_div < 0)
return -EINVAL;
mclk_div = sun4i_i2s_get_mclk_div(i2s, oversample_rate,
clk_rate,
rate);
if ((bclk_div >= 0) && (mclk_div >= 0))
break;
}
if ((bclk_div < 0) || (mclk_div < 0))
clk_rate, rate);
if (mclk_div < 0)
return -EINVAL;
regmap_write(i2s->regmap, SUN4I_I2S_CLK_DIV_REG,
@ -341,6 +351,27 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return 0;
}
static void sun4i_i2s_start_capture(struct sun4i_i2s *i2s)
{
/* Flush RX FIFO */
regmap_update_bits(i2s->regmap, SUN4I_I2S_FIFO_CTRL_REG,
SUN4I_I2S_FIFO_CTRL_FLUSH_RX,
SUN4I_I2S_FIFO_CTRL_FLUSH_RX);
/* Clear RX counter */
regmap_write(i2s->regmap, SUN4I_I2S_RX_CNT_REG, 0);
/* Enable RX Block */
regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
SUN4I_I2S_CTRL_RX_EN,
SUN4I_I2S_CTRL_RX_EN);
/* Enable RX DRQ */
regmap_update_bits(i2s->regmap, SUN4I_I2S_DMA_INT_CTRL_REG,
SUN4I_I2S_DMA_INT_CTRL_RX_DRQ_EN,
SUN4I_I2S_DMA_INT_CTRL_RX_DRQ_EN);
}
static void sun4i_i2s_start_playback(struct sun4i_i2s *i2s)
{
/* Flush TX FIFO */
@ -362,6 +393,18 @@ static void sun4i_i2s_start_playback(struct sun4i_i2s *i2s)
SUN4I_I2S_DMA_INT_CTRL_TX_DRQ_EN);
}
static void sun4i_i2s_stop_capture(struct sun4i_i2s *i2s)
{
/* Disable RX Block */
regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
SUN4I_I2S_CTRL_RX_EN,
0);
/* Disable RX DRQ */
regmap_update_bits(i2s->regmap, SUN4I_I2S_DMA_INT_CTRL_REG,
SUN4I_I2S_DMA_INT_CTRL_RX_DRQ_EN,
0);
}
static void sun4i_i2s_stop_playback(struct sun4i_i2s *i2s)
{
@ -388,7 +431,7 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
sun4i_i2s_start_playback(i2s);
else
return -EINVAL;
sun4i_i2s_start_capture(i2s);
break;
case SNDRV_PCM_TRIGGER_STOP:
@ -397,7 +440,7 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
sun4i_i2s_stop_playback(i2s);
else
return -EINVAL;
sun4i_i2s_stop_capture(i2s);
break;
default:
@ -447,9 +490,23 @@ static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream,
regmap_write(i2s->regmap, SUN4I_I2S_CTRL_REG, 0);
}
static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
if (clk_id != 0)
return -EINVAL;
i2s->mclk_freq = freq;
return 0;
}
static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = {
.hw_params = sun4i_i2s_hw_params,
.set_fmt = sun4i_i2s_set_fmt,
.set_sysclk = sun4i_i2s_set_sysclk,
.shutdown = sun4i_i2s_shutdown,
.startup = sun4i_i2s_startup,
.trigger = sun4i_i2s_trigger,
@ -459,7 +516,9 @@ static int sun4i_i2s_dai_probe(struct snd_soc_dai *dai)
{
struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, NULL);
snd_soc_dai_init_dma_data(dai,
&i2s->playback_dma_data,
&i2s->capture_dma_data);
snd_soc_dai_set_drvdata(dai, i2s);
@ -468,6 +527,13 @@ static int sun4i_i2s_dai_probe(struct snd_soc_dai *dai)
static struct snd_soc_dai_driver sun4i_i2s_dai = {
.probe = sun4i_i2s_dai_probe,
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_192000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.playback = {
.stream_name = "Playback",
.channels_min = 2,
@ -630,6 +696,9 @@ static int sun4i_i2s_probe(struct platform_device *pdev)
i2s->playback_dma_data.addr = res->start + SUN4I_I2S_FIFO_TX_REG;
i2s->playback_dma_data.maxburst = 4;
i2s->capture_dma_data.addr = res->start + SUN4I_I2S_FIFO_RX_REG;
i2s->capture_dma_data.maxburst = 4;
pm_runtime_enable(&pdev->dev);
if (!pm_runtime_enabled(&pdev->dev)) {
ret = sun4i_i2s_runtime_resume(&pdev->dev);

665
sound/soc/sunxi/sun8i-codec-analog.c Normal file
View File

@ -0,0 +1,665 @@
/*
* This driver supports the analog controls for the internal codec
* found in Allwinner's A31s, A23, A33 and H3 SoCs.
*
* Copyright 2016 Chen-Yu Tsai <wens@csie.org>
*
* 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.
*/
#include <linux/io.h>
#include <linux/kernel.h>
#include <linux/module.h>
#include <linux/of.h>
#include <linux/of_device.h>
#include <linux/platform_device.h>
#include <linux/regmap.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
/* Codec analog control register offsets and bit fields */
#define SUN8I_ADDA_HP_VOLC 0x00
#define SUN8I_ADDA_HP_VOLC_PA_CLK_GATE 7
#define SUN8I_ADDA_HP_VOLC_HP_VOL 0
#define SUN8I_ADDA_LOMIXSC 0x01
#define SUN8I_ADDA_LOMIXSC_MIC1 6
#define SUN8I_ADDA_LOMIXSC_MIC2 5
#define SUN8I_ADDA_LOMIXSC_PHONE 4
#define SUN8I_ADDA_LOMIXSC_PHONEN 3
#define SUN8I_ADDA_LOMIXSC_LINEINL 2
#define SUN8I_ADDA_LOMIXSC_DACL 1
#define SUN8I_ADDA_LOMIXSC_DACR 0
#define SUN8I_ADDA_ROMIXSC 0x02
#define SUN8I_ADDA_ROMIXSC_MIC1 6
#define SUN8I_ADDA_ROMIXSC_MIC2 5
#define SUN8I_ADDA_ROMIXSC_PHONE 4
#define SUN8I_ADDA_ROMIXSC_PHONEP 3
#define SUN8I_ADDA_ROMIXSC_LINEINR 2
#define SUN8I_ADDA_ROMIXSC_DACR 1
#define SUN8I_ADDA_ROMIXSC_DACL 0
#define SUN8I_ADDA_DAC_PA_SRC 0x03
#define SUN8I_ADDA_DAC_PA_SRC_DACAREN 7
#define SUN8I_ADDA_DAC_PA_SRC_DACALEN 6
#define SUN8I_ADDA_DAC_PA_SRC_RMIXEN 5
#define SUN8I_ADDA_DAC_PA_SRC_LMIXEN 4
#define SUN8I_ADDA_DAC_PA_SRC_RHPPAMUTE 3
#define SUN8I_ADDA_DAC_PA_SRC_LHPPAMUTE 2
#define SUN8I_ADDA_DAC_PA_SRC_RHPIS 1
#define SUN8I_ADDA_DAC_PA_SRC_LHPIS 0
#define SUN8I_ADDA_PHONEIN_GCTRL 0x04
#define SUN8I_ADDA_PHONEIN_GCTRL_PHONEPG 4
#define SUN8I_ADDA_PHONEIN_GCTRL_PHONENG 0
#define SUN8I_ADDA_LINEIN_GCTRL 0x05
#define SUN8I_ADDA_LINEIN_GCTRL_LINEING 4
#define SUN8I_ADDA_LINEIN_GCTRL_PHONEG 0
#define SUN8I_ADDA_MICIN_GCTRL 0x06
#define SUN8I_ADDA_MICIN_GCTRL_MIC1G 4
#define SUN8I_ADDA_MICIN_GCTRL_MIC2G 0
#define SUN8I_ADDA_PAEN_HP_CTRL 0x07
#define SUN8I_ADDA_PAEN_HP_CTRL_HPPAEN 7
#define SUN8I_ADDA_PAEN_HP_CTRL_LINEOUTEN 7 /* H3 specific */
#define SUN8I_ADDA_PAEN_HP_CTRL_HPCOM_FC 5
#define SUN8I_ADDA_PAEN_HP_CTRL_COMPTEN 4
#define SUN8I_ADDA_PAEN_HP_CTRL_PA_ANTI_POP_CTRL 2
#define SUN8I_ADDA_PAEN_HP_CTRL_LTRNMUTE 1
#define SUN8I_ADDA_PAEN_HP_CTRL_RTLNMUTE 0
#define SUN8I_ADDA_PHONEOUT_CTRL 0x08
#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUTG 5
#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUTEN 4
#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUT_MIC1 3
#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUT_MIC2 2
#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUT_RMIX 1
#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUT_LMIX 0
#define SUN8I_ADDA_PHONE_GAIN_CTRL 0x09
#define SUN8I_ADDA_PHONE_GAIN_CTRL_LINEOUT_VOL 3
#define SUN8I_ADDA_PHONE_GAIN_CTRL_PHONEPREG 0
#define SUN8I_ADDA_MIC2G_CTRL 0x0a
#define SUN8I_ADDA_MIC2G_CTRL_MIC2AMPEN 7
#define SUN8I_ADDA_MIC2G_CTRL_MIC2BOOST 4
#define SUN8I_ADDA_MIC2G_CTRL_LINEOUTLEN 3
#define SUN8I_ADDA_MIC2G_CTRL_LINEOUTREN 2
#define SUN8I_ADDA_MIC2G_CTRL_LINEOUTLSRC 1
#define SUN8I_ADDA_MIC2G_CTRL_LINEOUTRSRC 0
#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL 0x0b
#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_HMICBIASEN 7
#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MMICBIASEN 6
#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_HMICBIAS_MODE 5
#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1AMPEN 3
#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1BOOST 0
#define SUN8I_ADDA_LADCMIXSC 0x0c
#define SUN8I_ADDA_LADCMIXSC_MIC1 6
#define SUN8I_ADDA_LADCMIXSC_MIC2 5
#define SUN8I_ADDA_LADCMIXSC_PHONE 4
#define SUN8I_ADDA_LADCMIXSC_PHONEN 3
#define SUN8I_ADDA_LADCMIXSC_LINEINL 2
#define SUN8I_ADDA_LADCMIXSC_OMIXRL 1
#define SUN8I_ADDA_LADCMIXSC_OMIXRR 0
#define SUN8I_ADDA_RADCMIXSC 0x0d
#define SUN8I_ADDA_RADCMIXSC_MIC1 6
#define SUN8I_ADDA_RADCMIXSC_MIC2 5
#define SUN8I_ADDA_RADCMIXSC_PHONE 4
#define SUN8I_ADDA_RADCMIXSC_PHONEP 3
#define SUN8I_ADDA_RADCMIXSC_LINEINR 2
#define SUN8I_ADDA_RADCMIXSC_OMIXR 1
#define SUN8I_ADDA_RADCMIXSC_OMIXL 0
#define SUN8I_ADDA_RES 0x0e
#define SUN8I_ADDA_RES_MMICBIAS_SEL 4
#define SUN8I_ADDA_RES_PA_ANTI_POP_CTRL 0
#define SUN8I_ADDA_ADC_AP_EN 0x0f
#define SUN8I_ADDA_ADC_AP_EN_ADCREN 7
#define SUN8I_ADDA_ADC_AP_EN_ADCLEN 6
#define SUN8I_ADDA_ADC_AP_EN_ADCG 0
/* Analog control register access bits */
#define ADDA_PR 0x0 /* PRCM base + 0x1c0 */
#define ADDA_PR_RESET BIT(28)
#define ADDA_PR_WRITE BIT(24)
#define ADDA_PR_ADDR_SHIFT 16
#define ADDA_PR_ADDR_MASK GENMASK(4, 0)
#define ADDA_PR_DATA_IN_SHIFT 8
#define ADDA_PR_DATA_IN_MASK GENMASK(7, 0)
#define ADDA_PR_DATA_OUT_SHIFT 0
#define ADDA_PR_DATA_OUT_MASK GENMASK(7, 0)
/* regmap access bits */
static int adda_reg_read(void *context, unsigned int reg, unsigned int *val)
{
void __iomem *base = (void __iomem *)context;
u32 tmp;
/* De-assert reset */
writel(readl(base) | ADDA_PR_RESET, base);
/* Clear write bit */
writel(readl(base) & ~ADDA_PR_WRITE, base);
/* Set register address */
tmp = readl(base);
tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
writel(tmp, base);
/* Read back value */
*val = readl(base) & ADDA_PR_DATA_OUT_MASK;
return 0;
}
static int adda_reg_write(void *context, unsigned int reg, unsigned int val)
{
void __iomem *base = (void __iomem *)context;
u32 tmp;
/* De-assert reset */
writel(readl(base) | ADDA_PR_RESET, base);
/* Set register address */
tmp = readl(base);
tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
writel(tmp, base);
/* Set data to write */
tmp = readl(base);
tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT);
tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT;
writel(tmp, base);
/* Set write bit to signal a write */
writel(readl(base) | ADDA_PR_WRITE, base);
/* Clear write bit */
writel(readl(base) & ~ADDA_PR_WRITE, base);
return 0;
}
static const struct regmap_config adda_pr_regmap_cfg = {
.name = "adda-pr",
.reg_bits = 5,
.reg_stride = 1,
.val_bits = 8,
.reg_read = adda_reg_read,
.reg_write = adda_reg_write,
.fast_io = true,
.max_register = 24,
};
/* mixer controls */
static const struct snd_kcontrol_new sun8i_codec_mixer_controls[] = {
SOC_DAPM_DOUBLE_R("DAC Playback Switch",
SUN8I_ADDA_LOMIXSC,
SUN8I_ADDA_ROMIXSC,
SUN8I_ADDA_LOMIXSC_DACL, 1, 0),
SOC_DAPM_DOUBLE_R("DAC Reversed Playback Switch",
SUN8I_ADDA_LOMIXSC,
SUN8I_ADDA_ROMIXSC,
SUN8I_ADDA_LOMIXSC_DACR, 1, 0),
SOC_DAPM_DOUBLE_R("Line In Playback Switch",
SUN8I_ADDA_LOMIXSC,
SUN8I_ADDA_ROMIXSC,
SUN8I_ADDA_LOMIXSC_LINEINL, 1, 0),
SOC_DAPM_DOUBLE_R("Mic1 Playback Switch",
SUN8I_ADDA_LOMIXSC,
SUN8I_ADDA_ROMIXSC,
SUN8I_ADDA_LOMIXSC_MIC1, 1, 0),
SOC_DAPM_DOUBLE_R("Mic2 Playback Switch",
SUN8I_ADDA_LOMIXSC,
SUN8I_ADDA_ROMIXSC,
SUN8I_ADDA_LOMIXSC_MIC2, 1, 0),
};
/* ADC mixer controls */
static const struct snd_kcontrol_new sun8i_codec_adc_mixer_controls[] = {
SOC_DAPM_DOUBLE_R("Mixer Capture Switch",
SUN8I_ADDA_LADCMIXSC,
SUN8I_ADDA_RADCMIXSC,
SUN8I_ADDA_LADCMIXSC_OMIXRL, 1, 0),
SOC_DAPM_DOUBLE_R("Mixer Reversed Capture Switch",
SUN8I_ADDA_LADCMIXSC,
SUN8I_ADDA_RADCMIXSC,
SUN8I_ADDA_LADCMIXSC_OMIXRR, 1, 0),
SOC_DAPM_DOUBLE_R("Line In Capture Switch",
SUN8I_ADDA_LADCMIXSC,
SUN8I_ADDA_RADCMIXSC,
SUN8I_ADDA_LADCMIXSC_LINEINL, 1, 0),
SOC_DAPM_DOUBLE_R("Mic1 Capture Switch",
SUN8I_ADDA_LADCMIXSC,
SUN8I_ADDA_RADCMIXSC,
SUN8I_ADDA_LADCMIXSC_MIC1, 1, 0),
SOC_DAPM_DOUBLE_R("Mic2 Capture Switch",
SUN8I_ADDA_LADCMIXSC,
SUN8I_ADDA_RADCMIXSC,
SUN8I_ADDA_LADCMIXSC_MIC2, 1, 0),
};
/* volume / mute controls */
static const DECLARE_TLV_DB_SCALE(sun8i_codec_out_mixer_pregain_scale,
-450, 150, 0);
static const DECLARE_TLV_DB_RANGE(sun8i_codec_mic_gain_scale,
0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0),
);
static const struct snd_kcontrol_new sun8i_codec_common_controls[] = {
/* Mixer pre-gains */
SOC_SINGLE_TLV("Line In Playback Volume", SUN8I_ADDA_LINEIN_GCTRL,
SUN8I_ADDA_LINEIN_GCTRL_LINEING,
0x7, 0, sun8i_codec_out_mixer_pregain_scale),
SOC_SINGLE_TLV("Mic1 Playback Volume", SUN8I_ADDA_MICIN_GCTRL,
SUN8I_ADDA_MICIN_GCTRL_MIC1G,
0x7, 0, sun8i_codec_out_mixer_pregain_scale),
SOC_SINGLE_TLV("Mic2 Playback Volume",
SUN8I_ADDA_MICIN_GCTRL, SUN8I_ADDA_MICIN_GCTRL_MIC2G,
0x7, 0, sun8i_codec_out_mixer_pregain_scale),
/* Microphone Amp boost gains */
SOC_SINGLE_TLV("Mic1 Boost Volume", SUN8I_ADDA_MIC1G_MICBIAS_CTRL,
SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1BOOST, 0x7, 0,
sun8i_codec_mic_gain_scale),
SOC_SINGLE_TLV("Mic2 Boost Volume", SUN8I_ADDA_MIC2G_CTRL,
SUN8I_ADDA_MIC2G_CTRL_MIC2BOOST, 0x7, 0,
sun8i_codec_mic_gain_scale),
/* ADC */
SOC_SINGLE_TLV("ADC Gain Capture Volume", SUN8I_ADDA_ADC_AP_EN,
SUN8I_ADDA_ADC_AP_EN_ADCG, 0x7, 0,
sun8i_codec_out_mixer_pregain_scale),
};
static const struct snd_soc_dapm_widget sun8i_codec_common_widgets[] = {
/* ADC */
SND_SOC_DAPM_ADC("Left ADC", NULL, SUN8I_ADDA_ADC_AP_EN,
SUN8I_ADDA_ADC_AP_EN_ADCLEN, 0),
SND_SOC_DAPM_ADC("Right ADC", NULL, SUN8I_ADDA_ADC_AP_EN,
SUN8I_ADDA_ADC_AP_EN_ADCREN, 0),
/* DAC */
SND_SOC_DAPM_DAC("Left DAC", NULL, SUN8I_ADDA_DAC_PA_SRC,
SUN8I_ADDA_DAC_PA_SRC_DACALEN, 0),
SND_SOC_DAPM_DAC("Right DAC", NULL, SUN8I_ADDA_DAC_PA_SRC,
SUN8I_ADDA_DAC_PA_SRC_DACAREN, 0),
/*
* Due to this component and the codec belonging to separate DAPM
* contexts, we need to manually link the above widgets to their
* stream widgets at the card level.
*/
/* Line In */
SND_SOC_DAPM_INPUT("LINEIN"),
/* Microphone inputs */
SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2"),
/* Microphone Bias */
SND_SOC_DAPM_SUPPLY("MBIAS", SUN8I_ADDA_MIC1G_MICBIAS_CTRL,
SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MMICBIASEN,
0, NULL, 0),
/* Mic input path */
SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN8I_ADDA_MIC1G_MICBIAS_CTRL,
SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1AMPEN, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mic2 Amplifier", SUN8I_ADDA_MIC2G_CTRL,
SUN8I_ADDA_MIC2G_CTRL_MIC2AMPEN, 0, NULL, 0),
/* Mixers */
SND_SOC_DAPM_MIXER("Left Mixer", SUN8I_ADDA_DAC_PA_SRC,
SUN8I_ADDA_DAC_PA_SRC_LMIXEN, 0,
sun8i_codec_mixer_controls,
ARRAY_SIZE(sun8i_codec_mixer_controls)),
SND_SOC_DAPM_MIXER("Right Mixer", SUN8I_ADDA_DAC_PA_SRC,
SUN8I_ADDA_DAC_PA_SRC_RMIXEN, 0,
sun8i_codec_mixer_controls,
ARRAY_SIZE(sun8i_codec_mixer_controls)),
SND_SOC_DAPM_MIXER("Left ADC Mixer", SUN8I_ADDA_ADC_AP_EN,
SUN8I_ADDA_ADC_AP_EN_ADCLEN, 0,
sun8i_codec_adc_mixer_controls,
ARRAY_SIZE(sun8i_codec_adc_mixer_controls)),
SND_SOC_DAPM_MIXER("Right ADC Mixer", SUN8I_ADDA_ADC_AP_EN,
SUN8I_ADDA_ADC_AP_EN_ADCREN, 0,
sun8i_codec_adc_mixer_controls,
ARRAY_SIZE(sun8i_codec_adc_mixer_controls)),
};
static const struct snd_soc_dapm_route sun8i_codec_common_routes[] = {
/* Microphone Routes */
{ "Mic1 Amplifier", NULL, "MIC1"},
{ "Mic2 Amplifier", NULL, "MIC2"},
/* Left Mixer Routes */
{ "Left Mixer", "DAC Playback Switch", "Left DAC" },
{ "Left Mixer", "DAC Reversed Playback Switch", "Right DAC" },
{ "Left Mixer", "Line In Playback Switch", "LINEIN" },
{ "Left Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" },
{ "Left Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" },
/* Right Mixer Routes */
{ "Right Mixer", "DAC Playback Switch", "Right DAC" },
{ "Right Mixer", "DAC Reversed Playback Switch", "Left DAC" },
{ "Right Mixer", "Line In Playback Switch", "LINEIN" },
{ "Right Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" },
{ "Right Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" },
/* Left ADC Mixer Routes */
{ "Left ADC Mixer", "Mixer Capture Switch", "Left Mixer" },
{ "Left ADC Mixer", "Mixer Reversed Capture Switch", "Right Mixer" },
{ "Left ADC Mixer", "Line In Capture Switch", "LINEIN" },
{ "Left ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" },
{ "Left ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" },
/* Right ADC Mixer Routes */
{ "Right ADC Mixer", "Mixer Capture Switch", "Right Mixer" },
{ "Right ADC Mixer", "Mixer Reversed Capture Switch", "Left Mixer" },
{ "Right ADC Mixer", "Line In Capture Switch", "LINEIN" },
{ "Right ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" },
{ "Right ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" },
/* ADC Routes */
{ "Left ADC", NULL, "Left ADC Mixer" },
{ "Right ADC", NULL, "Right ADC Mixer" },
};
/* headphone specific controls, widgets, and routes */
static const DECLARE_TLV_DB_SCALE(sun8i_codec_hp_vol_scale, -6300, 100, 1);
static const struct snd_kcontrol_new sun8i_codec_headphone_controls[] = {
SOC_SINGLE_TLV("Headphone Playback Volume",
SUN8I_ADDA_HP_VOLC,
SUN8I_ADDA_HP_VOLC_HP_VOL, 0x3f, 0,
sun8i_codec_hp_vol_scale),
SOC_DOUBLE("Headphone Playback Switch",
SUN8I_ADDA_DAC_PA_SRC,
SUN8I_ADDA_DAC_PA_SRC_LHPPAMUTE,
SUN8I_ADDA_DAC_PA_SRC_RHPPAMUTE, 1, 0),
};
static const char * const sun8i_codec_hp_src_enum_text[] = {
"DAC", "Mixer",
};
static SOC_ENUM_DOUBLE_DECL(sun8i_codec_hp_src_enum,
SUN8I_ADDA_DAC_PA_SRC,
SUN8I_ADDA_DAC_PA_SRC_LHPIS,
SUN8I_ADDA_DAC_PA_SRC_RHPIS,
sun8i_codec_hp_src_enum_text);
static const struct snd_kcontrol_new sun8i_codec_hp_src[] = {
SOC_DAPM_ENUM("Headphone Source Playback Route",
sun8i_codec_hp_src_enum),
};
static const struct snd_soc_dapm_widget sun8i_codec_headphone_widgets[] = {
SND_SOC_DAPM_MUX("Headphone Source Playback Route",
SND_SOC_NOPM, 0, 0, sun8i_codec_hp_src),
SND_SOC_DAPM_OUT_DRV("Headphone Amp", SUN8I_ADDA_PAEN_HP_CTRL,
SUN8I_ADDA_PAEN_HP_CTRL_HPPAEN, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("HPCOM Protection", SUN8I_ADDA_PAEN_HP_CTRL,
SUN8I_ADDA_PAEN_HP_CTRL_COMPTEN, 0, NULL, 0),
SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPCOM", SUN8I_ADDA_PAEN_HP_CTRL,
SUN8I_ADDA_PAEN_HP_CTRL_HPCOM_FC, 0x3, 0x3, 0),
SND_SOC_DAPM_OUTPUT("HP"),
};
static const struct snd_soc_dapm_route sun8i_codec_headphone_routes[] = {
{ "Headphone Source Playback Route", "DAC", "Left DAC" },
{ "Headphone Source Playback Route", "DAC", "Right DAC" },
{ "Headphone Source Playback Route", "Mixer", "Left Mixer" },
{ "Headphone Source Playback Route", "Mixer", "Right Mixer" },
{ "Headphone Amp", NULL, "Headphone Source Playback Route" },
{ "HPCOM", NULL, "HPCOM Protection" },
{ "HP", NULL, "Headphone Amp" },
};
static int sun8i_codec_add_headphone(struct snd_soc_component *cmpnt)
{
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cmpnt);
struct device *dev = cmpnt->dev;
int ret;
ret = snd_soc_add_component_controls(cmpnt,
sun8i_codec_headphone_controls,
ARRAY_SIZE(sun8i_codec_headphone_controls));
if (ret) {
dev_err(dev, "Failed to add Headphone controls: %d\n", ret);
return ret;
}
ret = snd_soc_dapm_new_controls(dapm, sun8i_codec_headphone_widgets,
ARRAY_SIZE(sun8i_codec_headphone_widgets));
if (ret) {
dev_err(dev, "Failed to add Headphone DAPM widgets: %d\n", ret);
return ret;
}
ret = snd_soc_dapm_add_routes(dapm, sun8i_codec_headphone_routes,
ARRAY_SIZE(sun8i_codec_headphone_routes));
if (ret) {
dev_err(dev, "Failed to add Headphone DAPM routes: %d\n", ret);
return ret;
}
return 0;
}
/* hmic specific widget */
static const struct snd_soc_dapm_widget sun8i_codec_hmic_widgets[] = {
SND_SOC_DAPM_SUPPLY("HBIAS", SUN8I_ADDA_MIC1G_MICBIAS_CTRL,
SUN8I_ADDA_MIC1G_MICBIAS_CTRL_HMICBIASEN,
0, NULL, 0),
};
static int sun8i_codec_add_hmic(struct snd_soc_component *cmpnt)
{
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cmpnt);
struct device *dev = cmpnt->dev;
int ret;
ret = snd_soc_dapm_new_controls(dapm, sun8i_codec_hmic_widgets,
ARRAY_SIZE(sun8i_codec_hmic_widgets));
if (ret)
dev_err(dev, "Failed to add Mic3 DAPM widgets: %d\n", ret);
return ret;
}
/* line out specific controls, widgets and routes */
static const DECLARE_TLV_DB_RANGE(sun8i_codec_lineout_vol_scale,
0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1),
2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0),
);
static const struct snd_kcontrol_new sun8i_codec_lineout_controls[] = {
SOC_SINGLE_TLV("Line Out Playback Volume",
SUN8I_ADDA_PHONE_GAIN_CTRL,
SUN8I_ADDA_PHONE_GAIN_CTRL_LINEOUT_VOL, 0x1f, 0,
sun8i_codec_lineout_vol_scale),
SOC_DOUBLE("Line Out Playback Switch",
SUN8I_ADDA_MIC2G_CTRL,
SUN8I_ADDA_MIC2G_CTRL_LINEOUTLEN,
SUN8I_ADDA_MIC2G_CTRL_LINEOUTREN, 1, 0),
};
static const char * const sun8i_codec_lineout_src_enum_text[] = {
"Stereo", "Mono Differential",
};
static SOC_ENUM_DOUBLE_DECL(sun8i_codec_lineout_src_enum,
SUN8I_ADDA_MIC2G_CTRL,
SUN8I_ADDA_MIC2G_CTRL_LINEOUTLSRC,
SUN8I_ADDA_MIC2G_CTRL_LINEOUTRSRC,
sun8i_codec_lineout_src_enum_text);
static const struct snd_kcontrol_new sun8i_codec_lineout_src[] = {
SOC_DAPM_ENUM("Line Out Source Playback Route",
sun8i_codec_lineout_src_enum),
};
static const struct snd_soc_dapm_widget sun8i_codec_lineout_widgets[] = {
SND_SOC_DAPM_MUX("Line Out Source Playback Route",
SND_SOC_NOPM, 0, 0, sun8i_codec_lineout_src),
/* It is unclear if this is a buffer or gate, model it as a supply */
SND_SOC_DAPM_SUPPLY("Line Out Enable", SUN8I_ADDA_PAEN_HP_CTRL,
SUN8I_ADDA_PAEN_HP_CTRL_LINEOUTEN, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("LINEOUT"),
};
static const struct snd_soc_dapm_route sun8i_codec_lineout_routes[] = {
{ "Line Out Source Playback Route", "Stereo", "Left Mixer" },
{ "Line Out Source Playback Route", "Stereo", "Right Mixer" },
{ "Line Out Source Playback Route", "Mono Differential", "Left Mixer" },
{ "Line Out Source Playback Route", "Mono Differential", "Right Mixer" },
{ "LINEOUT", NULL, "Line Out Source Playback Route" },
{ "LINEOUT", NULL, "Line Out Enable", },
};
static int sun8i_codec_add_lineout(struct snd_soc_component *cmpnt)
{
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cmpnt);
struct device *dev = cmpnt->dev;
int ret;
ret = snd_soc_add_component_controls(cmpnt,
sun8i_codec_lineout_controls,
ARRAY_SIZE(sun8i_codec_lineout_controls));
if (ret) {
dev_err(dev, "Failed to add Line Out controls: %d\n", ret);
return ret;
}
ret = snd_soc_dapm_new_controls(dapm, sun8i_codec_lineout_widgets,
ARRAY_SIZE(sun8i_codec_lineout_widgets));
if (ret) {
dev_err(dev, "Failed to add Line Out DAPM widgets: %d\n", ret);
return ret;
}
ret = snd_soc_dapm_add_routes(dapm, sun8i_codec_lineout_routes,
ARRAY_SIZE(sun8i_codec_lineout_routes));
if (ret) {
dev_err(dev, "Failed to add Line Out DAPM routes: %d\n", ret);
return ret;
}
return 0;
}
struct sun8i_codec_analog_quirks {
bool has_headphone;
bool has_hmic;
bool has_lineout;
};
static const struct sun8i_codec_analog_quirks sun8i_a23_quirks = {
.has_headphone = true,
.has_hmic = true,
};
static const struct sun8i_codec_analog_quirks sun8i_h3_quirks = {
.has_lineout = true,
};
static int sun8i_codec_analog_cmpnt_probe(struct snd_soc_component *cmpnt)
{
struct device *dev = cmpnt->dev;
const struct sun8i_codec_analog_quirks *quirks;
int ret;
/*
* This would never return NULL unless someone directly registers a
* platform device matching this driver's name, without specifying a
* device tree node.
*/
quirks = of_device_get_match_data(dev);
/* Add controls, widgets, and routes for individual features */
if (quirks->has_headphone) {
ret = sun8i_codec_add_headphone(cmpnt);
if (ret)
return ret;
}
if (quirks->has_hmic) {
ret = sun8i_codec_add_hmic(cmpnt);
if (ret)
return ret;
}
if (quirks->has_lineout) {
ret = sun8i_codec_add_lineout(cmpnt);
if (ret)
return ret;
}
return 0;
}
static const struct snd_soc_component_driver sun8i_codec_analog_cmpnt_drv = {
.controls = sun8i_codec_common_controls,
.num_controls = ARRAY_SIZE(sun8i_codec_common_controls),
.dapm_widgets = sun8i_codec_common_widgets,
.num_dapm_widgets = ARRAY_SIZE(sun8i_codec_common_widgets),
.dapm_routes = sun8i_codec_common_routes,
.num_dapm_routes = ARRAY_SIZE(sun8i_codec_common_routes),
.probe = sun8i_codec_analog_cmpnt_probe,
};
static const struct of_device_id sun8i_codec_analog_of_match[] = {
{
.compatible = "allwinner,sun8i-a23-codec-analog",
.data = &sun8i_a23_quirks,
},
{
.compatible = "allwinner,sun8i-h3-codec-analog",
.data = &sun8i_h3_quirks,
},
{}
};
MODULE_DEVICE_TABLE(of, sun8i_codec_analog_of_match);
static int sun8i_codec_analog_probe(struct platform_device *pdev)
{
struct resource *res;
struct regmap *regmap;
void __iomem *base;
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
base = devm_ioremap_resource(&pdev->dev, res);
if (IS_ERR(base)) {
dev_err(&pdev->dev, "Failed to map the registers\n");
return PTR_ERR(base);
}
regmap = devm_regmap_init(&pdev->dev, NULL, base, &adda_pr_regmap_cfg);
if (IS_ERR(regmap)) {
dev_err(&pdev->dev, "Failed to create regmap\n");
return PTR_ERR(regmap);
}
return devm_snd_soc_register_component(&pdev->dev,
&sun8i_codec_analog_cmpnt_drv,
NULL, 0);
}
static struct platform_driver sun8i_codec_analog_driver = {
.driver = {
.name = "sun8i-codec-analog",
.of_match_table = sun8i_codec_analog_of_match,
},
.probe = sun8i_codec_analog_probe,
};
module_platform_driver(sun8i_codec_analog_driver);
MODULE_DESCRIPTION("Allwinner internal codec analog controls driver");
MODULE_AUTHOR("Chen-Yu Tsai <wens@csie.org>");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:sun8i-codec-analog");

2
sound/soc/tegra/tegra_alc5632.c
View File

@ -65,7 +65,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static struct snd_soc_ops tegra_alc5632_asoc_ops = {
static const struct snd_soc_ops tegra_alc5632_asoc_ops = {
.hw_params = tegra_alc5632_asoc_hw_params,
};

2
sound/soc/tegra/tegra_max98090.c
View File

@ -93,7 +93,7 @@ static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static struct snd_soc_ops tegra_max98090_ops = {
static const struct snd_soc_ops tegra_max98090_ops = {
.hw_params = tegra_max98090_asoc_hw_params,
};

2
sound/soc/tegra/tegra_rt5640.c
View File

@ -76,7 +76,7 @@ static int tegra_rt5640_asoc_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static struct snd_soc_ops tegra_rt5640_ops = {
static const struct snd_soc_ops tegra_rt5640_ops = {
.hw_params = tegra_rt5640_asoc_hw_params,
};

2
sound/soc/tegra/tegra_rt5677.c
View File

@ -93,7 +93,7 @@ static int tegra_rt5677_event_hp(struct snd_soc_dapm_widget *w,
return 0;
}
static struct snd_soc_ops tegra_rt5677_ops = {
static const struct snd_soc_ops tegra_rt5677_ops = {
.hw_params = tegra_rt5677_asoc_hw_params,
};

2
sound/soc/tegra/tegra_sgtl5000.c
View File

@ -82,7 +82,7 @@ static int tegra_sgtl5000_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static struct snd_soc_ops tegra_sgtl5000_ops = {
static const struct snd_soc_ops tegra_sgtl5000_ops = {
.hw_params = tegra_sgtl5000_hw_params,
};

2
sound/soc/tegra/tegra_wm8753.c
View File

@ -89,7 +89,7 @@ static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static struct snd_soc_ops tegra_wm8753_ops = {
static const struct snd_soc_ops tegra_wm8753_ops = {
.hw_params = tegra_wm8753_hw_params,
};

2
sound/soc/tegra/tegra_wm8903.c
View File

@ -96,7 +96,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static struct snd_soc_ops tegra_wm8903_ops = {
static const struct snd_soc_ops tegra_wm8903_ops = {
.hw_params = tegra_wm8903_hw_params,
};

2
sound/soc/tegra/trimslice.c
View File

@ -74,7 +74,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static struct snd_soc_ops trimslice_asoc_ops = {
static const struct snd_soc_ops trimslice_asoc_ops = {
.hw_params = trimslice_asoc_hw_params,
};