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Merge branch 'for-linus' of git://git.alsa-project.org/alsa-kernel 

* 'for-linus' of git://git.alsa-project.org/alsa-kernel: (179 commits)
  ALSA: Release v1.0.17
  ALSA: correct kcalloc usage
  ALSA: ALSA driver for SGI O2 audio board
  ALSA: asoc: kbuild - only show menus for the current ASoC CPU platform.
  ALSA: ALSA driver for SGI HAL2 audio device
  ALSA: hda - Fix FSC V5505 model
  ALSA: hda - Fix missing init for unsol events on micsense model
  ALSA: hda - Fix internal mic vref pin setup
  ALSA: hda: 92hd71bxx PC Beep
  ALSA: HDA - HP dc7600 with pci sub IDs 0x103c/0x3011 belongs to hp-3013 model
  ALSA: usb-audio: add some Yamaha USB MIDI quirks
  ALSA: usb-audio: fix Yamaha KX quirk
  ALSA: ASoC: Au12x0/Au1550 PSC Audio support
  ALSA: Add Yamaha KX49 (USB MIDI controller) to usbquirks.h
  ALSA: ASoC: pxa2xx-ac97: fix warning due to missing argument in fuction declaration
  ALSA: tosa: fix compilation with new DAPM API
  ALSA: wavefront - add const
  ALSA: remove CONFIG_KMOD from sound
  ALSA: Fix a const to non-const assignment in the Digigram VXpocket sound driver
  ALSA: Fix a const pointer usage warning in the Digigram VX soundcard driver
  ...
This commit is contained in:
Linus Torvalds 2008-07-14 13:26:07 -07:00
parent b7f80afa28 fe0a3fe324
commit b5cf43c47b
201 changed files with 16534 additions and 3954 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

17
Documentation/sound/alsa/ALSA-Configuration.txt
View File

@ -753,8 +753,11 @@ 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)
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
@ -845,7 +848,7 @@ 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 +856,10 @@ 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
@ -1091,7 +1098,7 @@ 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.
@ -2267,6 +2274,10 @@ 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
=======================================

4
Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
View File

@ -6127,8 +6127,8 @@ struct _snd_pcm_runtime {
<para>
<function>snd_printdd()</function> is compiled in only when
<constant>CONFIG_SND_DEBUG_DETECT</constant> is set. Please note
that <constant>DEBUG_DETECT</constant> is not set as default
<constant>CONFIG_SND_DEBUG_VERBOSE</constant> is set. Please note
that <constant>CONFIG_SND_DEBUG_VERBOSE</constant> is not set as default
even if you configure the alsa-driver with
<option>--with-debug=full</option> option. You need to give
explicitly <option>--with-debug=detect</option> option instead.

8
include/asm-mips/mach-au1x00/au1xxx_psc.h
View File

@ -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)

46
include/sound/ad1843.h Normal file
View File

@ -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 <vivien.chappelier@linux-mips.org>
* Copyright 2008 Thomas Bogendoerfer <tsbogend@franken.de>
*/
#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 */

3
include/sound/control.h
View File

@ -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;

8
include/sound/core.h
View File

@ -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

8
include/sound/cs4231-regs.h
View File

@ -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 */

3
include/sound/cs4231.h
View File

@ -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);

1
include/sound/emu10k1.h
View File

@ -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 */

2
include/sound/seq_kernel.h
View File

@ -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

42
include/sound/soc-dapm.h
View File

@ -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, \
@ -193,6 +200,7 @@ 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,
@ -205,25 +213,32 @@ 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,
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,
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);
/* dapm audio endpoint control */
int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
char *pin, int status);
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 {
@ -245,6 +260,18 @@ 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;
@ -277,6 +304,9 @@ 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 */

175
include/sound/soc.h
View File

@ -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 }
@ -91,6 +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, \
@ -102,6 +120,24 @@
.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
*/
@ -185,8 +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;
@ -221,6 +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
*/
@ -249,6 +305,12 @@ 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 {
@ -272,87 +334,45 @@ struct snd_soc_ops {
int (*trigger)(struct snd_pcm_substream *, int);
};
/* ASoC codec DAI ops */
struct snd_soc_codec_ops {
/* codec DAI clocking configuration */
int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai,
/* ASoC DAI ops */
struct snd_soc_dai_ops {
/* DAI clocking configuration */
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);
/* CPU 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,
/* DAI format configuration */
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);
};
/* 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;
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_codec_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;
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_dai *dai);
void (*remove)(struct platform_device *pdev,
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;
struct snd_soc_cpu_ops dai_ops;
struct snd_soc_dai_ops dai_ops;
/* DAI capabilities */
struct snd_soc_pcm_stream capture;
@ -360,7 +380,9 @@ 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 */
@ -374,7 +396,8 @@ 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;
@ -396,12 +419,12 @@ 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 */
struct snd_soc_codec_dai *dai;
struct snd_soc_dai *dai;
unsigned int num_dai;
};
@ -420,12 +443,12 @@ 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 *);
@ -439,8 +462,8 @@ 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;
@ -467,7 +490,8 @@ 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;
@ -482,6 +506,7 @@ 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;
};

2
include/sound/uda1341.h
View File

@ -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"
/*

4
include/sound/version.h
View File

@ -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 "1.0.17"
#define CONFIG_SND_DATE ""

34
sound/Kconfig
View File

@ -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 <file:Documentation/sound/oss/README.modules>; 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 <http://www.alsa-project.org/>
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

11
sound/aoa/Kconfig
View File

@ -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

4
sound/aoa/codecs/Kconfig
View File

@ -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

1
sound/aoa/fabrics/Kconfig
View File

@ -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---

1
sound/aoa/soundbus/Kconfig
View File

@ -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

21
sound/arm/Kconfig
View File

@ -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

2
sound/arm/sa11xx-uda1341.c
View File

@ -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

29
sound/core/Kconfig
View File

@ -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
@ -161,16 +150,17 @@ config SND_VERBOSE_PRINTK
config SND_DEBUG
bool "Debug"
depends on SND
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"
@ -184,4 +174,3 @@ config SND_PCM_XRUN_DEBUG
config SND_VMASTER
bool
depends on SND

7
sound/core/control.c
View File

@ -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;

67
sound/core/init.c
View File

@ -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<<idx2) {
if (module_slot_mismatch(module, idx2))
continue;
idx = idx2;
if (idx >= 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<<idx2) {
if (!slots[idx2] || !*slots[idx2]) {
idx = idx2;
break;
}
}
}
if (idx < 0)
err = -ENODEV;
else if (idx < snd_ecards_limit) {
if (snd_cards_lock & (1 << idx))
err = -EBUSY; /* invalid */
} else if (idx >= 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;

62
sound/core/memalloc.c
View File

@ -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

2
sound/core/seq/seq_clientmgr.c
View File

@ -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];

6
sound/core/seq/seq_device.c
View File

@ -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

8
sound/core/sound.c
View File

@ -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? */

6
sound/core/timer.c
View File

@ -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(&register_mutex);
timer = snd_timer_find(tid);
#ifdef CONFIG_KMOD
if (timer == NULL) {
#ifdef CONFIG_MODULES
if (!timer) {
mutex_unlock(&register_mutex);
snd_timer_request(tid);
mutex_lock(&register_mutex);

91
sound/drivers/Kconfig
View File

@ -1,15 +1,41 @@
# ALSA generic drivers
config SND_MPU401_UART
tristate
select SND_RAWMIDI
menu "Generic devices"
depends on SND!=n
config SND_OPL3_LIB
tristate
select SND_TIMER
select SND_HWDEP
config SND_OPL4_LIB
tristate
select SND_TIMER
select SND_HWDEP
config SND_VX_LIB
tristate
select SND_HWDEP
select SND_PCM
config SND_AC97_CODEC
tristate
select SND_PCM
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
depends on SND
select SND_PCM
help
If you don't have a sound card in your computer, you can include a
@ -35,33 +61,8 @@ config SND_PCSP
Say M if you don't.
Say Y only if you really know what you do.
config SND_MPU401_UART
tristate
select SND_RAWMIDI
config SND_OPL3_LIB
tristate
select SND_TIMER
select SND_HWDEP
config SND_OPL4_LIB
tristate
select SND_TIMER
select SND_HWDEP
config SND_VX_LIB
tristate
select SND_HWDEP
select SND_PCM
config SND_AC97_CODEC
tristate
select SND_PCM
select AC97_BUS
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
@ -90,7 +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
@ -102,7 +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
@ -115,7 +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
@ -131,7 +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
@ -142,7 +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
@ -153,7 +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
@ -163,4 +161,25 @@ config SND_ML403_AC97CR
To compile this driver as a module, choose M here: the module
will be called snd-ml403_ac97cr.
endmenu
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

2
sound/drivers/vx/vx_hwdep.c
View File

@ -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;
}

6
sound/i2c/cs8427.c
View File

@ -23,6 +23,7 @@
#include <linux/slab.h>
#include <linux/delay.h>
#include <linux/init.h>
#include <asm/unaligned.h>
#include <sound/core.h>
#include <sound/control.h>
#include <sound/pcm.h>
@ -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;

2
sound/i2c/l3/uda1341.c
View File

@ -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 <linux/module.h>
#include <linux/init.h>
#include <linux/types.h>

61
sound/isa/Kconfig
View File

@ -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
@ -204,15 +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"
depends on SND
select SND_RAWMIDI
select SND_PCM
select SND_GUS_SYNTH
help
Say Y here to include support for Gravis UltraSound Classic
soundcards.
@ -222,11 +215,9 @@ config SND_GUSCLASSIC
config SND_GUSEXTREME
tristate "Gravis UltraSound Extreme"
depends on SND
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.
@ -236,10 +227,8 @@ config SND_GUSEXTREME
config SND_GUSMAX
tristate "Gravis UltraSound MAX"
depends on SND
select SND_RAWMIDI
select SND_CS4231_LIB
select SND_GUS_SYNTH
help
Say Y here to include support for Gravis UltraSound MAX
soundcards.
@ -249,10 +238,9 @@ 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
help
Say Y here to include support for AMD InterWave based
soundcards (Gravis UltraSound Plug & Play, STB SoundRage32,
@ -263,10 +251,9 @@ 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
help
Say Y here to include support for AMD InterWave based
soundcards with a TEA6330T bass and treble regulator
@ -277,7 +264,6 @@ 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 +276,6 @@ 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 +289,6 @@ 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
@ -318,10 +302,9 @@ config SND_OPTI92X_CS4231
config SND_OPTI93X
tristate "OPTi 82C93x"
depends on SND
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.
@ -331,7 +314,6 @@ 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 +327,6 @@ 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 +339,6 @@ 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 +351,6 @@ 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 +381,6 @@ 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 +391,6 @@ 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 +403,6 @@ 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 +424,5 @@ config SND_WAVEFRONT_FIRMWARE_IN_KERNEL
you need to install the firmware files from the
alsa-firmware package.
endmenu
endif # SND_ISA

118
sound/isa/cs423x/cs4231_lib.c
View File

@ -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);

1126
sound/isa/opti9xx/opti92x-ad1848.c
View File

File diff suppressed because it is too large Load Diff

2
sound/isa/sb/Makefile
View File

@ -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))

2
sound/isa/wavefront/wavefront_synth.c
View File

@ -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;

27
sound/mips/Kconfig
View File

@ -1,15 +1,34 @@
# 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_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
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) && 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

4
sound/mips/Makefile
View File

@ -3,6 +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

561
sound/mips/ad1843.c Normal file
View File

@ -0,0 +1,561 @@
/*
* AD1843 low level driver
*
* Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
* Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de>
*
* 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 <linux/init.h>
#include <linux/sched.h>
#include <linux/errno.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ad1843.h>
/*
* 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;
}

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/*
* Driver for A2 audio system used in SGI machines
* Copyright (c) 2008 Thomas Bogendoerfer <tsbogend@alpha.fanken.de>
*
* Based on OSS code from Ladislav Michl <ladis@linux-mips.org>, 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 <linux/kernel.h>
#include <linux/init.h>
#include <linux/interrupt.h>
#include <linux/dma-mapping.h>
#include <linux/platform_device.h>
#include <linux/io.h>
#include <asm/sgi/hpc3.h>
#include <asm/sgi/ip22.h>
#include <sound/core.h>
#include <sound/control.h>
#include <sound/pcm.h>
#include <sound/pcm-indirect.h>
#include <sound/initval.h>
#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(&regs->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), &regs->iar);
H2_INDIRECT_WAIT(regs);
ret = hal2_read(&regs->idr0) & 0xffff;
hal2_write(H2_READ_ADDR(addr) | 0x1, &regs->iar);
H2_INDIRECT_WAIT(regs);
ret |= (hal2_read(&regs->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, &regs->idr0);
hal2_write(0, &regs->idr1);
hal2_write(0, &regs->idr2);
hal2_write(0, &regs->idr3);
hal2_write(H2_WRITE_ADDR(addr), &regs->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, &regs->idr0);
hal2_write(val >> 16, &regs->idr1);
hal2_write(0, &regs->idr2);
hal2_write(0, &regs->idr3);
hal2_write(H2_WRITE_ADDR(addr), &regs->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), &regs->iar);
H2_INDIRECT_WAIT(regs);
hal2_write((hal2_read(&regs->idr0) & 0xffff) | bit, &regs->idr0);
hal2_write(0, &regs->idr1);
hal2_write(0, &regs->idr2);
hal2_write(0, &regs->idr3);
hal2_write(H2_WRITE_ADDR(addr), &regs->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), &regs->iar);
H2_INDIRECT_WAIT(regs);
hal2_write((hal2_read(&regs->idr0) & 0xffff) & ~bit, &regs->idr0);
hal2_write(0, &regs->idr1);
hal2_write(0, &regs->idr2);
hal2_write(0, &regs->idr3);
hal2_write(H2_WRITE_ADDR(addr), &regs->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);

245
sound/mips/hal2.h Normal file
View File

@ -0,0 +1,245 @@
#ifndef __HAL2_H
#define __HAL2_H
/*
* Driver for HAL2 sound processors
* Copyright (c) 1999 Ulf Carlsson <ulfc@bun.falkenberg.se>
* Copyright (c) 2001, 2002, 2003 Ladislav Michl <ladis@linux-mips.org>
*
* 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 <linux/types.h>
/* 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 */

1006
sound/mips/sgio2audio.c Normal file
View File

File diff suppressed because it is too large Load Diff

49
sound/oss/Kconfig
View File

@ -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 <file:Documentation/sound/oss/MultiSound> 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"

7
sound/oss/dmasound/dmasound_core.c
View File

@ -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) ;

2
sound/oss/dmasound/dmasound_paula.c
View File

@ -710,7 +710,7 @@ static MACHINE machAmiga = {
/*** Config & Setup **********************************************************/
int __init dmasound_paula_init(void)
static int __init dmasound_paula_init(void)
{
int err;

2
sound/oss/dmasound/dmasound_q40.c
View File

@ -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;

2
sound/oss/msnd.c
View File

@ -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 <linux/module.h>

2
sound/oss/msnd.h
View File

@ -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

2
sound/oss/msnd_classic.h
View File

@ -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

5
sound/oss/msnd_pinnacle.c
View File

@ -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 <linux/kernel.h>

2
sound/oss/msnd_pinnacle.h
View File

@ -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

13
sound/parisc/Kconfig
View File

@ -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

104
sound/pci/Kconfig
View File

@ -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
@ -691,8 +657,7 @@ 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
@ -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
@ -954,25 +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.
endmenu
endif # SND_PCI

2
sound/pci/Makefile
View File

@ -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

12
sound/pci/ac97/Makefile
View File

@ -3,16 +3,8 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-ac97-codec-objs := ac97_codec.o ac97_pcm.o
ifneq ($(CONFIG_PROC_FS),)
snd-ac97-codec-objs += ac97_proc.o
endif
snd-ak4531-codec-objs := ak4531_codec.o
snd-ac97-codec-y := ac97_codec.o ac97_pcm.o
snd-ac97-codec-$(CONFIG_PROC_FS) += ac97_proc.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))

11
sound/pci/ac97/ac97_codec.c
View File

@ -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
/*
@ -2294,9 +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 */
@ -2362,7 +2365,7 @@ 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);

81
sound/pci/ac97/ac97_patch.c
View File

@ -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),
@ -3352,8 +3353,66 @@ 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 +3420,24 @@ 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;
}
@ -3633,7 +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;
}

34
sound/pci/ac97/ak4531_codec.c → sound/pci/ak4531_codec.c
View File

@ -28,9 +28,11 @@
#include <sound/ak4531_codec.h>
#include <sound/tlv.h>
/*
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
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);
@ -270,7 +272,7 @@ 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 +381,9 @@ 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 +479,8 @@ 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 +488,3 @@ 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)

2
sound/pci/au88x0/au88x0_game.c
View File

@ -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:

1243
sound/pci/azt3328.c
View File

File diff suppressed because it is too large Load Diff

207
sound/pci/azt3328.h
View File

@ -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) */
@ -54,7 +55,10 @@
#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 +76,26 @@
#define SOUNDFORMAT_FLAG_16BIT 0x0010
#define SOUNDFORMAT_FLAG_2CHANNELS 0x0020
/* define frequency helpers, for maximum value safety */
enum azf_freq_t {
#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 +121,171 @@
/** 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!!
* 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 /* 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_!
* 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)

5
sound/pci/ca0106/ca0106_main.c
View File

@ -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".

1
sound/pci/emu10k1/emu10k1_main.c
View File

@ -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,

13
sound/pci/emu10k1/emumixer.c
View File

@ -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;

69
sound/pci/emu10k1/memory.c
View File

@ -437,43 +437,49 @@ 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))) {
if (p)
__free_page(p);
/* try to allocate from <16MB zone */
p = alloc_page(GFP_ATOMIC | GFP_DMA |
__GFP_NORETRY | /* no OOM-killer */
__GFP_NOWARN);
}
emu->page_addr_table[page] = dmab.addr;
emu->page_ptr_table[page] = dmab.area;
if (!p) {
__synth_free_pages(emu, first_page, page - 1);
return -ENOMEM;
}
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 +487,10 @@ __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;
}

2
sound/pci/hda/hda_codec.c
View File

@ -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)

2
sound/pci/hda/hda_codec.h
View File

@ -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

2
sound/pci/hda/hda_hwdep.c
View File

@ -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

308
sound/pci/hda/hda_intel.c
View File

@ -55,6 +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 probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
static int single_cmd;
static int enable_msi;
@ -69,7 +70,9 @@ 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(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);
@ -197,6 +200,10 @@ 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
@ -259,9 +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 */
@ -285,6 +291,7 @@ 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 +308,11 @@ 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;
unsigned int irq_ignore :1;
};
/* CORB/RIRB */
@ -323,6 +330,7 @@ struct azx_rb {
struct azx {
struct snd_card *card;
struct pci_dev *pci;
int dev_index;
/* chip type specific */
int driver_type;
@ -366,9 +374,13 @@ 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) */
/* for pending irqs */
struct work_struct irq_pending_work;
};
/* driver types */
@ -381,6 +393,7 @@ enum {
AZX_DRIVER_SIS,
AZX_DRIVER_ULI,
AZX_DRIVER_NVIDIA,
AZX_DRIVER_TERA,
};
static char *driver_short_names[] __devinitdata = {
@ -392,6 +405,7 @@ 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",
};
/*
@ -426,11 +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);
/*
@ -461,7 +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);
@ -476,7 +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);
@ -847,7 +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;
}
@ -908,6 +917,8 @@ 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 +941,23 @@ 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;
/* 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;
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);
}
}
}
@ -959,59 +982,107 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
/*
* set up BDL entries
* set up a BDL entry
*/
static int azx_setup_periods(struct snd_pcm_substream *substream,
struct azx_dev *azx_dev)
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_32_bits(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 azx *chip,
struct snd_pcm_substream *substream,
struct azx_dev *azx_dev)
{
u32 *bdl;
int i, ofs, periods, period_bytes;
int pos_adj;
/* reset BDL address */
azx_sd_writel(azx_dev, SD_BDLPL, 0);
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 */
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;
pos_adj = bdl_pos_adj[chip->dev_index];
if (pos_adj > 0) {
struct snd_pcm_runtime *runtime = substream->runtime;
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[chip->dev_index]);
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);
} else
pos_adj = 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;
}
/*
@ -1062,7 +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 ||
@ -1085,7 +1156,7 @@ 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,
@ -1093,6 +1164,7 @@ 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,
@ -1316,7 +1388,7 @@ 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)
@ -1421,35 +1493,113 @@ 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)
pos += azx_dev->fifo_size;
}
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_LPIB;
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;
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);
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 +1826,7 @@ 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 +1883,7 @@ 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);
@ -1770,9 +1922,9 @@ 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),
{}
};
@ -1857,12 +2009,25 @@ 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]);
check_probe_mask(chip, dev);
chip->single_cmd = single_cmd;
if (bdl_pos_adj[dev] < 0) {
switch (chip->driver_type) {
case AZX_DRIVER_ICH:
bdl_pos_adj[dev] = 1;
break;
default:
bdl_pos_adj[dev] = 32;
break;
}
}
#if BITS_PER_LONG != 64
/* Fix up base address on ULI M5461 */
if (chip->driver_type == AZX_DRIVER_ULI) {
@ -2089,6 +2254,7 @@ 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 },
@ -2141,6 +2307,8 @@ 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);

5
sound/pci/hda/hda_proc.c
View File

@ -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)

38
sound/pci/hda/patch_analog.c
View File

@ -23,7 +23,6 @@
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/pci.h>
#include <linux/mutex.h>
#include <sound/core.h>
#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 */
@ -1618,6 +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),
@ -2623,7 +2622,7 @@ 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)
@ -2631,16 +2630,26 @@ 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,
@ -3177,7 +3186,6 @@ 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 +3855,6 @@ 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 +4159,6 @@ 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;

33
sound/pci/hda/patch_conexant.c
View File

@ -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 */
@ -687,7 +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},
@ -907,10 +906,12 @@ 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),
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),
@ -928,7 +929,6 @@ 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;
@ -963,6 +963,7 @@ 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;
@ -1007,15 +1008,19 @@ 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;
}
@ -1477,7 +1482,6 @@ 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 +1740,6 @@ 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;

548
sound/pci/hda/patch_realtek.c
View File

@ -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,
};
@ -205,6 +209,7 @@ enum {
ALC883_MITAC,
ALC883_CLEVO_M720,
ALC883_FUJITSU_PI2515,
ALC883_3ST_6ch_INTEL,
ALC883_AUTO,
ALC883_MODEL_LAST,
};
@ -280,6 +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;
};
/*
@ -747,6 +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;
@ -776,6 +817,24 @@ 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
@ -851,8 +910,10 @@ 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);
@ -877,7 +938,6 @@ 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);
@ -889,6 +949,9 @@ 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,
@ -2373,6 +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]);
@ -3009,6 +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),
@ -5101,7 +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),
@ -6127,6 +6193,7 @@ 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),
@ -6353,7 +6420,9 @@ 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;
}
@ -6450,8 +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:
@ -6485,14 +6555,20 @@ 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;
@ -6569,6 +6645,16 @@ 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 = {
@ -6649,6 +6735,48 @@ 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
*/
@ -6881,15 +7009,54 @@ 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),
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),
@ -7729,6 +7896,7 @@ 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",
};
@ -7786,6 +7954,8 @@ 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),
{}
};
@ -7824,6 +7994,18 @@ 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 },
@ -8145,6 +8327,8 @@ 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);
@ -8171,12 +8355,25 @@ 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;
@ -8189,6 +8386,9 @@ 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;
@ -9522,6 +9722,8 @@ 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),
@ -9729,6 +9931,8 @@ 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);
@ -10674,12 +10878,18 @@ 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))
@ -11033,6 +11243,8 @@ 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);
@ -12631,6 +12843,12 @@ 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)},
@ -12786,6 +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),
{}
};
@ -13168,11 +13387,19 @@ 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";
/* 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";
}
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;
@ -13251,6 +13478,23 @@ 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 +13675,44 @@ 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 +13783,11 @@ 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 +13858,43 @@ 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 +14016,125 @@ 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 +14157,24 @@ 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 +14262,53 @@ 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,
},
};
@ -14082,6 +14581,8 @@ 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);
@ -14108,11 +14609,17 @@ 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 +14658,7 @@ 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 },

71
sound/pci/hda/patch_sigmatel.c
View File

@ -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)},
{}
};
@ -818,6 +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 */
@ -1317,13 +1327,13 @@ 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,
@ -1754,12 +1764,8 @@ 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 +1776,14 @@ 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,
@ -3103,13 +3105,16 @@ 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);
@ -3614,6 +3619,7 @@ 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,
@ -3642,6 +3648,19 @@ 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;
@ -3653,22 +3672,19 @@ 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;
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;
@ -4306,10 +4322,11 @@ 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 },

10
sound/pci/ice1712/envy24ht.h
View File

@ -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

2
sound/pci/ice1712/ice1712.h
View File

@ -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 */

215
sound/pci/ice1712/ice1724.c
View File

@ -32,7 +32,7 @@
#include <linux/mutex.h>
#include <sound/core.h>
#include <sound/info.h>
#include <sound/mpu401.h>
#include <sound/rawmidi.h>
#include <sound/initval.h>
#include <sound/asoundef.h>
@ -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));
}
}

42
sound/pci/maestro3.c
View File

@ -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,

4
sound/pci/nm256/nm256.c
View File

@ -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;

35
sound/pci/oxygen/hifier.c
View File

@ -28,7 +28,7 @@
MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
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;
@ -62,16 +62,28 @@ 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_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);
ak4396_write(chip, AK4396_LCH_ATT, 0);
ak4396_write(chip, AK4396_RCH_ATT, 0);
hifier_registers_init(chip);
snd_component_add(chip->card, "AK4396");
snd_component_add(chip->card, "CS5340");
@ -100,12 +112,6 @@ 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;
@ -140,6 +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,
@ -180,6 +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)

76
sound/pci/oxygen/oxygen.c
View File

@ -43,7 +43,7 @@
MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
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;
@ -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,
@ -99,20 +100,35 @@ 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)
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_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);
@ -120,9 +136,16 @@ 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);
}
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");
}
@ -133,12 +156,23 @@ 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)
{
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);
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_registers_init(chip);
snd_component_add(chip->card, "WM8785");
}
@ -158,6 +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)
{
@ -183,18 +223,6 @@ 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;
@ -256,6 +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,
@ -283,6 +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,
@ -331,6 +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)

14
sound/pci/oxygen/oxygen.h
View File

@ -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
@ -78,6 +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 {
@ -89,6 +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,
@ -117,6 +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 */

22
sound/pci/oxygen/oxygen_io.c
View File

@ -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");
}

106
sound/pci/oxygen/oxygen_lib.c
View File

@ -32,7 +32,7 @@
MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
MODULE_DESCRIPTION("C-Media CMI8788 helper library");
MODULE_LICENSE("GPL");
MODULE_LICENSE("GPL v2");
static irqreturn_t oxygen_interrupt(int dummy, void *dev_id)
@ -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));
@ -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);
@ -455,7 +459,7 @@ 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;
@ -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 */

53
sound/pci/oxygen/oxygen_pcm.c
View File

@ -24,6 +24,16 @@
#include <sound/pcm_params.h>
#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] = {
@ -155,6 +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;
@ -517,6 +533,7 @@ 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:
@ -663,12 +680,14 @@ 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);
@ -688,7 +707,8 @@ 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) {
@ -718,7 +738,8 @@ 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;
}

358
sound/pci/oxygen/virtuoso.c
View File

@ -79,7 +79,7 @@
MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
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;
@ -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,
@ -159,6 +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;
@ -170,32 +181,59 @@ 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)
{
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 pcm1796_init(struct oxygen *chip)
{
struct xonar_data *data = chip->model_data;
unsigned int i;
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);
pcm1796_write(chip, i, 21, 0);
}
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;
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, 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, 21, 0);
pcm1796_write(chip, i, 16, 0x0f); /* set ATL/ATR after ATLD */
pcm1796_write(chip, i, 17, 0x0f);
}
pcm1796_init(chip);
oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT);
oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT);
@ -217,151 +255,6 @@ static void xonar_d2x_init(struct oxygen *chip)
xonar_d2_init(chip);
}
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;
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);
/* configure */
cs4398_write(chip, 2, CS4398_FM_SINGLE |
CS4398_DEM_NONE | CS4398_DIF_LJUST);
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);
cs4362a_write(chip, 0x03, CS4362A_MUTEC_6 | CS4362A_AMUTE |
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, 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, 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, 0x0d, 0x7f | CS4362A_MUTE);
cs4362a_write(chip, 0x0e, 0x7f | CS4362A_MUTE);
/* clear power down */
cs4398_write(chip, 8, CS4398_CPEN);
cs4362a_write(chip, 0x01, CS4362A_CPEN);
oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL,
GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE);
oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA,
GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE);
xonar_common_init(chip);
snd_component_add(chip->card, "CS4398");
snd_component_add(chip->card, "CS4362A");
snd_component_add(chip->card, "CS5361");
}
static void xonar_cleanup(struct oxygen *chip)
{
struct xonar_data *data = chip->model_data;
oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit);
}
static void xonar_dx_cleanup(struct oxygen *chip)
{
xonar_cleanup(chip);
cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN);
oxygen_clear_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC);
}
static void set_pcm1796_params(struct oxygen *chip,
struct snd_pcm_hw_params *params)
{
unsigned int i;
u8 value;
value = params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64;
for (i = 0; i < 4; ++i)
pcm1796_write(chip, i, 20, value);
}
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)
{
unsigned int value;
if (params_rate(params) <= 54000)
value = GPIO_CS53x1_M_SINGLE;
else if (params_rate(params) <= 108000)
value = GPIO_CS53x1_M_DOUBLE;
else
value = GPIO_CS53x1_M_QUAD;
oxygen_write16_masked(chip, OXYGEN_GPIO_DATA,
value, GPIO_CS53x1_M_MASK);
}
static void set_cs43xx_params(struct oxygen *chip,
struct snd_pcm_hw_params *params)
{
u8 fm_cs4398, fm_cs4362a;
fm_cs4398 = CS4398_DEM_NONE | CS4398_DIF_LJUST;
fm_cs4362a = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L;
if (params_rate(params) <= 50000) {
fm_cs4398 |= CS4398_FM_SINGLE;
fm_cs4362a |= CS4362A_FM_SINGLE;
} else if (params_rate(params) <= 100000) {
fm_cs4398 |= CS4398_FM_DOUBLE;
fm_cs4362a |= CS4362A_FM_DOUBLE;
} else {
fm_cs4398 |= CS4398_FM_QUAD;
fm_cs4362a |= 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);
}
static void update_cs4362a_volumes(struct oxygen *chip)
{
u8 mute;
@ -393,6 +286,141 @@ static void update_cs43xx_mute(struct oxygen *chip)
update_cs4362a_volumes(chip);
}
static void cs43xx_init(struct oxygen *chip)
{
struct xonar_data *data = chip->model_data;
/* set CPEN (control port mode) and power down */
cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN);
cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN);
/* configure */
cs4398_write(chip, 2, data->cs4398_fm);
cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L);
cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP |
CS4398_ZERO_CROSS | CS4398_SOFT_RAMP);
cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST);
cs4362a_write(chip, 0x03, CS4362A_MUTEC_6 | CS4362A_AMUTE |
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, data->cs4362a_fm);
cs4362a_write(chip, 0x09, data->cs4362a_fm);
cs4362a_write(chip, 0x0c, data->cs4362a_fm);
update_cs43xx_volume(chip);
update_cs43xx_mute(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);
oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA,
GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE);
xonar_common_init(chip);
snd_component_add(chip->card, "CS4398");
snd_component_add(chip->card, "CS4362A");
snd_component_add(chip->card, "CS5361");
}
static void xonar_cleanup(struct oxygen *chip)
{
struct xonar_data *data = chip->model_data;
oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit);
}
static void xonar_dx_cleanup(struct oxygen *chip)
{
xonar_cleanup(chip);
cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN);
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)
{
struct xonar_data *data = chip->model_data;
unsigned int i;
data->pcm1796_oversampling =
params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64;
for (i = 0; i < 4; ++i)
pcm1796_write(chip, i, 20, data->pcm1796_oversampling);
}
static void set_cs53x1_params(struct oxygen *chip,
struct snd_pcm_hw_params *params)
{
unsigned int value;
if (params_rate(params) <= 54000)
value = GPIO_CS53x1_M_SINGLE;
else if (params_rate(params) <= 108000)
value = GPIO_CS53x1_M_DOUBLE;
else
value = GPIO_CS53x1_M_QUAD;
oxygen_write16_masked(chip, OXYGEN_GPIO_DATA,
value, GPIO_CS53x1_M_MASK);
}
static void set_cs43xx_params(struct oxygen *chip,
struct snd_pcm_hw_params *params)
{
struct xonar_data *data = chip->model_data;
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) {
data->cs4398_fm |= CS4398_FM_SINGLE;
data->cs4362a_fm |= CS4362A_FM_SINGLE;
} else if (params_rate(params) <= 100000) {
data->cs4398_fm |= CS4398_FM_DOUBLE;
data->cs4362a_fm |= CS4362A_FM_DOUBLE;
} else {
data->cs4398_fm |= CS4398_FM_QUAD;
data->cs4362a_fm |= CS4362A_FM_QUAD;
}
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 xonar_gpio_changed(struct oxygen *chip)
{
struct xonar_data *data = chip->model_data;
@ -535,6 +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,
@ -563,6 +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,
@ -592,6 +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,
@ -636,6 +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)

4
sound/pci/pcxhr/pcxhr.c
View File

@ -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);

18
sound/pci/pcxhr/pcxhr_core.c
View File

@ -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

5
sound/pci/trident/trident_main.c
View File

@ -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;

178
sound/pci/trident/trident_memory.c
View File

@ -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);

6
sound/pci/via82xx.c
View File

@ -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,

2
sound/pci/ymfpci/ymfpci_main.c
View File

@ -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);

15
sound/pcmcia/Kconfig
View File

@ -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

2
sound/pcmcia/vx/vxp_ops.c
View File

@ -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 */

26
sound/ppc/Kconfig
View File

@ -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

2
sound/ppc/daca.c
View File

@ -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)

2
sound/ppc/tumbler.c
View File

@ -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)

16
sound/sh/Kconfig
View File

@ -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

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