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kernel-ark/sound/soc/codecs/stac9766.c
Tejun Heo 5a0e3ad6af include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h 
percpu.h is included by sched.h and module.h and thus ends up being
included when building most .c files.  percpu.h includes slab.h which
in turn includes gfp.h making everything defined by the two files
universally available and complicating inclusion dependencies.

percpu.h -> slab.h dependency is about to be removed.  Prepare for
this change by updating users of gfp and slab facilities include those
headers directly instead of assuming availability.  As this conversion
needs to touch large number of source files, the following script is
used as the basis of conversion.

  http://userweb.kernel.org/~tj/misc/slabh-sweep.py

The script does the followings.

* Scan files for gfp and slab usages and update includes such that
  only the necessary includes are there.  ie. if only gfp is used,
  gfp.h, if slab is used, slab.h.

* When the script inserts a new include, it looks at the include
  blocks and try to put the new include such that its order conforms
  to its surrounding.  It's put in the include block which contains
  core kernel includes, in the same order that the rest are ordered -
  alphabetical, Christmas tree, rev-Xmas-tree or at the end if there
  doesn't seem to be any matching order.

* If the script can't find a place to put a new include (mostly
  because the file doesn't have fitting include block), it prints out
  an error message indicating which .h file needs to be added to the
  file.

The conversion was done in the following steps.

1. The initial automatic conversion of all .c files updated slightly
   over 4000 files, deleting around 700 includes and adding ~480 gfp.h
   and ~3000 slab.h inclusions.  The script emitted errors for ~400
   files.

2. Each error was manually checked.  Some didn't need the inclusion,
   some needed manual addition while adding it to implementation .h or
   embedding .c file was more appropriate for others.  This step added
   inclusions to around 150 files.

3. The script was run again and the output was compared to the edits
   from #2 to make sure no file was left behind.

4. Several build tests were done and a couple of problems were fixed.
   e.g. lib/decompress_*.c used malloc/free() wrappers around slab
   APIs requiring slab.h to be added manually.

5. The script was run on all .h files but without automatically
   editing them as sprinkling gfp.h and slab.h inclusions around .h
   files could easily lead to inclusion dependency hell.  Most gfp.h
   inclusion directives were ignored as stuff from gfp.h was usually
   wildly available and often used in preprocessor macros.  Each
   slab.h inclusion directive was examined and added manually as
   necessary.

6. percpu.h was updated not to include slab.h.

7. Build test were done on the following configurations and failures
   were fixed.  CONFIG_GCOV_KERNEL was turned off for all tests (as my
   distributed build env didn't work with gcov compiles) and a few
   more options had to be turned off depending on archs to make things
   build (like ipr on powerpc/64 which failed due to missing writeq).

   * x86 and x86_64 UP and SMP allmodconfig and a custom test config.
   * powerpc and powerpc64 SMP allmodconfig
   * sparc and sparc64 SMP allmodconfig
   * ia64 SMP allmodconfig
   * s390 SMP allmodconfig
   * alpha SMP allmodconfig
   * um on x86_64 SMP allmodconfig

8. percpu.h modifications were reverted so that it could be applied as
   a separate patch and serve as bisection point.

Given the fact that I had only a couple of failures from tests on step
6, I'm fairly confident about the coverage of this conversion patch.
If there is a breakage, it's likely to be something in one of the arch
headers which should be easily discoverable easily on most builds of
the specific arch.

Signed-off-by: Tejun Heo <tj@kernel.org>
Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org>
Cc: Ingo Molnar <mingo@redhat.com>
Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
2010-03-30 22:02:32 +09:00

446 lines
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/*
* stac9766.c -- ALSA SoC STAC9766 codec support
*
* Copyright 2009 Jon Smirl, Digispeaker
* Author: Jon Smirl <jonsmirl@gmail.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
* Features:-
*
* o Support for AC97 Codec, S/PDIF
*/
#include <linux/init.h>
#include <linux/slab.h>
#include <linux/module.h>
#include <linux/device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/initval.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#include <sound/soc-of-simple.h>
#include "stac9766.h"
#define STAC9766_VERSION "0.10"
/*
* STAC9766 register cache
*/
static const u16 stac9766_reg[] = {
0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
0x0000, 0x0000, 0x8008, 0x8008, /* e */
0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
};
static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
"Line", "Stereo Mix", "Mono Mix", "Phone"};
static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
static const char *stac9766_record_all_mux[] = {"All analog",
"Analog plus DAC"};
static const char *stac9766_boost1[] = {"0dB", "10dB"};
static const char *stac9766_boost2[] = {"0dB", "20dB"};
static const char *stac9766_stereo_mic[] = {"Off", "On"};
static const struct soc_enum stac9766_record_enum =
SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
static const struct soc_enum stac9766_mono_enum =
SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
static const struct soc_enum stac9766_mic_enum =
SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
static const struct soc_enum stac9766_SPDIF_enum =
SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
static const struct soc_enum stac9766_popbypass_enum =
SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
static const struct soc_enum stac9766_record_all_enum =
SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
stac9766_record_all_mux);
static const struct soc_enum stac9766_boost1_enum =
SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
static const struct soc_enum stac9766_boost2_enum =
SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
static const struct soc_enum stac9766_stereo_mic_enum =
SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic);
static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);
static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
master_tlv),
SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
master_tlv),
SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),
SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),
SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),
SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),
SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),
SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
SOC_ENUM("Record All Mux", stac9766_record_all_enum),
SOC_ENUM("Record Mux", stac9766_record_enum),
SOC_ENUM("Mono Mux", stac9766_mono_enum),
SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
};
static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int val)
{
u16 *cache = codec->reg_cache;
if (reg > AC97_STAC_PAGE0) {
stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
soc_ac97_ops.write(codec->ac97, reg, val);
stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
return 0;
}
if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
return -EIO;
soc_ac97_ops.write(codec->ac97, reg, val);
cache[reg / 2] = val;
return 0;
}
static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
unsigned int reg)
{
u16 val = 0, *cache = codec->reg_cache;
if (reg > AC97_STAC_PAGE0) {
stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0);
stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
return val;
}
if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
return -EIO;
if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
reg == AC97_VENDOR_ID2) {
val = soc_ac97_ops.read(codec->ac97, reg);
return val;
}
return cache[reg / 2];
}
static int ac97_analog_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned short reg, vra;
vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
vra |= 0x1; /* enable variable rate audio */
vra &= ~0x4; /* disable SPDIF output */
stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
reg = AC97_PCM_FRONT_DAC_RATE;
else
reg = AC97_PCM_LR_ADC_RATE;
return stac9766_ac97_write(codec, reg, runtime->rate);
}
static int ac97_digital_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned short reg, vra;
stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);
vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
vra |= 0x5; /* Enable VRA and SPDIF out */
stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
reg = AC97_PCM_FRONT_DAC_RATE;
return stac9766_ac97_write(codec, reg, runtime->rate);
}
static int stac9766_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
switch (level) {
case SND_SOC_BIAS_ON: /* full On */
case SND_SOC_BIAS_PREPARE: /* partial On */
case SND_SOC_BIAS_STANDBY: /* Off, with power */
stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
break;
case SND_SOC_BIAS_OFF: /* Off, without power */
/* disable everything including AC link */
stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
codec->bias_level = level;
return 0;
}
static int stac9766_reset(struct snd_soc_codec *codec, int try_warm)
{
if (try_warm && soc_ac97_ops.warm_reset) {
soc_ac97_ops.warm_reset(codec->ac97);
if (stac9766_ac97_read(codec, 0) == stac9766_reg[0])
return 1;
}
soc_ac97_ops.reset(codec->ac97);
if (soc_ac97_ops.warm_reset)
soc_ac97_ops.warm_reset(codec->ac97);
if (stac9766_ac97_read(codec, 0) != stac9766_reg[0])
return -EIO;
return 0;
}
static int stac9766_codec_suspend(struct platform_device *pdev,
pm_message_t state)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int stac9766_codec_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
u16 id, reset;
reset = 0;
/* give the codec an AC97 warm reset to start the link */
reset:
if (reset > 5) {
printk(KERN_ERR "stac9766 failed to resume");
return -EIO;
}
codec->ac97->bus->ops->warm_reset(codec->ac97);
id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2);
if (id != 0x4c13) {
stac9766_reset(codec, 0);
reset++;
goto reset;
}
stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
stac9766_set_bias_level(codec, SND_SOC_BIAS_ON);
return 0;
}
static struct snd_soc_dai_ops stac9766_dai_ops_analog = {
.prepare = ac97_analog_prepare,
};
static struct snd_soc_dai_ops stac9766_dai_ops_digital = {
.prepare = ac97_digital_prepare,
};
struct snd_soc_dai stac9766_dai[] = {
{
.name = "stac9766 analog",
.id = 0,
.ac97_control = 1,
/* stream cababilities */
.playback = {
.stream_name = "stac9766 analog",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SND_SOC_STD_AC97_FMTS,
},
.capture = {
.stream_name = "stac9766 analog",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SND_SOC_STD_AC97_FMTS,
},
/* alsa ops */
.ops = &stac9766_dai_ops_analog,
},
{
.name = "stac9766 IEC958",
.id = 1,
.ac97_control = 1,
/* stream cababilities */
.playback = {
.stream_name = "stac9766 IEC958",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_32000 | \
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE,
},
/* alsa ops */
.ops = &stac9766_dai_ops_digital,
}
};
EXPORT_SYMBOL_GPL(stac9766_dai);
static int stac9766_codec_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec;
int ret = 0;
printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);
socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (socdev->card->codec == NULL)
return -ENOMEM;
codec = socdev->card->codec;
mutex_init(&codec->mutex);
codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg),
GFP_KERNEL);
if (codec->reg_cache == NULL) {
ret = -ENOMEM;
goto cache_err;
}
codec->reg_cache_size = sizeof(stac9766_reg);
codec->reg_cache_step = 2;
codec->name = "STAC9766";
codec->owner = THIS_MODULE;
codec->dai = stac9766_dai;
codec->num_dai = ARRAY_SIZE(stac9766_dai);
codec->write = stac9766_ac97_write;
codec->read = stac9766_ac97_read;
codec->set_bias_level = stac9766_set_bias_level;
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
if (ret < 0)
goto codec_err;
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0)
goto pcm_err;
/* do a cold reset for the controller and then try
* a warm reset followed by an optional cold reset for codec */
stac9766_reset(codec, 0);
ret = stac9766_reset(codec, 1);
if (ret < 0) {
printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n");
goto reset_err;
}
stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
snd_soc_add_controls(codec, stac9766_snd_ac97_controls,
ARRAY_SIZE(stac9766_snd_ac97_controls));
return 0;
reset_err:
snd_soc_free_pcms(socdev);
pcm_err:
snd_soc_free_ac97_codec(codec);
codec_err:
kfree(codec->private_data);
cache_err:
kfree(socdev->card->codec);
socdev->card->codec = NULL;
return ret;
}
static int stac9766_codec_remove(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
if (codec == NULL)
return 0;
snd_soc_free_pcms(socdev);
snd_soc_free_ac97_codec(codec);
kfree(codec->reg_cache);
kfree(codec);
return 0;
}
struct snd_soc_codec_device soc_codec_dev_stac9766 = {
.probe = stac9766_codec_probe,
.remove = stac9766_codec_remove,
.suspend = stac9766_codec_suspend,
.resume = stac9766_codec_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766);
MODULE_DESCRIPTION("ASoC stac9766 driver");
MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
MODULE_LICENSE("GPL");
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