Minki/linux
1
0
Fork 1
mirror of https://github.com/torvalds/linux.git synced 2024-11-17 17:41:44 +00:00
Code Issues Packages Projects Releases Wiki Activity
3ed37648e1
linux/sound/soc/codecs/stac9766.c
Liam Girdwood 022658beab ASoC: core: Add support for DAI and machine kcontrols. 
Currently ASoC can only add kcontrols using codec and platform component device
handles. It's also desirable to add kcontrols for DAIs (i.e. McBSP) and for
SoC card machine drivers too. This allows the kcontrol to have a direct handle to
the parent ASoC component DAI/SoC Card/Platform/Codec device and hence easily
get it's private data.

This change makes snd_soc_add_controls() static and wraps it in the folowing
calls (card and dai are new) :-

snd_soc_add_card_controls()
snd_soc_add_codec_controls()
snd_soc_add_dai_controls()
snd_soc_add_platform_controls()

This patch also does a lot of small mechanical changes in individual codec drivers
to replace snd_soc_add_controls() with snd_soc_add_codec_controls().

It also updates the McBSP DAI driver to use snd_soc_add_dai_controls().

Finally, it updates the existing machine drivers that register controls to either :-

1) Use snd_soc_add_card_controls() where no direct codec control is required.
2) Use snd_soc_add_codec_controls() where there is direct codec control.

In the case of 1) above we also update the machine drivers to get the correct
component data pointers from the kcontrol (rather than getting the machine pointer
via the codec pointer).

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-02-04 12:40:11 +00:00

415 lines
12 KiB
C
Raw Blame History

/*
* 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 "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->dapm.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 snd_soc_codec *codec)
{
stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int stac9766_codec_resume(struct snd_soc_codec *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);
return 0;
}
static const struct snd_soc_dai_ops stac9766_dai_ops_analog = {
.prepare = ac97_analog_prepare,
};
static const struct snd_soc_dai_ops stac9766_dai_ops_digital = {
.prepare = ac97_digital_prepare,
};
static struct snd_soc_dai_driver stac9766_dai[] = {
{
.name = "stac9766-hifi-analog",
.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-hifi-IEC958",
.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,
}
};
static int stac9766_codec_probe(struct snd_soc_codec *codec)
{
int ret = 0;
printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);
ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
if (ret < 0)
goto codec_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 codec_err;
}
stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
snd_soc_add_codec_controls(codec, stac9766_snd_ac97_controls,
ARRAY_SIZE(stac9766_snd_ac97_controls));
return 0;
codec_err:
snd_soc_free_ac97_codec(codec);
return ret;
}
static int stac9766_codec_remove(struct snd_soc_codec *codec)
{
snd_soc_free_ac97_codec(codec);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
.write = stac9766_ac97_write,
.read = stac9766_ac97_read,
.set_bias_level = stac9766_set_bias_level,
.probe = stac9766_codec_probe,
.remove = stac9766_codec_remove,
.suspend = stac9766_codec_suspend,
.resume = stac9766_codec_resume,
.reg_cache_size = ARRAY_SIZE(stac9766_reg),
.reg_word_size = sizeof(u16),
.reg_cache_step = 2,
.reg_cache_default = stac9766_reg,
};
static __devinit int stac9766_probe(struct platform_device *pdev)
{
return snd_soc_register_codec(&pdev->dev,
&soc_codec_dev_stac9766, stac9766_dai, ARRAY_SIZE(stac9766_dai));
}
static int __devexit stac9766_remove(struct platform_device *pdev)
{
snd_soc_unregister_codec(&pdev->dev);
return 0;
}
static struct platform_driver stac9766_codec_driver = {
.driver = {
.name = "stac9766-codec",
.owner = THIS_MODULE,
},
.probe = stac9766_probe,
.remove = __devexit_p(stac9766_remove),
};
module_platform_driver(stac9766_codec_driver);
MODULE_DESCRIPTION("ASoC stac9766 driver");
MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
MODULE_LICENSE("GPL");
Reference in New Issue View Git Blame Copy Permalink