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linux/sound/soc/codecs/stac9766.c
Liam Girdwood f0fba2ad1b ASoC: multi-component - ASoC Multi-Component Support 
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.

 struct snd_soc_codec    --->  struct snd_soc_codec (device data)
                          +->  struct snd_soc_codec_driver (driver data)

 struct snd_soc_platform --->  struct snd_soc_platform (device data)
                          +->  struct snd_soc_platform_driver (driver data)

 struct snd_soc_dai      --->  struct snd_soc_dai (device data)
                          +->  struct snd_soc_dai_driver (driver data)

 struct snd_soc_device   --->  deleted

This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.

The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.

This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.

Other notable multi-component changes:-

 * Stream operations now de-reference less structures.
 * close_delayed work() now runs on a DAI basis rather than looping all DAIs
   in a card.
 * PM suspend()/resume() operations can now handle N CODECs and Platforms
   per sound card.
 * Added soc_bind_dai_link() to bind the component devices to the sound card.
 * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
   DAI link components.
 * sysfs entries can now be registered per component per card.
 * snd_soc_new_pcms() functionailty rolled into dai_link_probe().
 * snd_soc_register_codec() now does all the codec list and mutex init.

This patch changes the probe() and remove() of the CODEC drivers as follows:-

 o Make CODEC driver a platform driver
 o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
 o Removed all static codec pointers (drivers now support > 1 codec dev)
 o snd_soc_register_pcms() now done by core.
 o snd_soc_register_dai() folded into snd_soc_register_codec().

CS4270 portions:
Acked-by: Timur Tabi <timur@freescale.com>

Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>

TI CODEC and OMAP fixes
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>

Samsung platform and misc fixes :-
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>

MPC8610 and PPC fixes.
Signed-off-by: Timur Tabi <timur@freescale.com>

i.MX fixes and some core fixes.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>

J4740 platform fixes:-
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>

CC: Tony Lindgren <tony@atomide.com>
CC: Nicolas Ferre <nicolas.ferre@atmel.com>
CC: Kevin Hilman <khilman@deeprootsystems.com>
CC: Sascha Hauer <s.hauer@pengutronix.de>
CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
CC: Daniel Gloeckner <dg@emlix.com>
CC: Manuel Lauss <mano@roarinelk.homelinux.net>
CC: Mike Frysinger <vapier.adi@gmail.com>
CC: Arnaud Patard <apatard@mandriva.com>
CC: Wan ZongShun <mcuos.com@gmail.com>

Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-08-12 14:00:00 +01:00

425 lines
12 KiB
C
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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 "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 snd_soc_codec *codec,
pm_message_t state)
{
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 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,
};
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_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 = sizeof(stac9766_reg),
.reg_word_size = sizeof(u16),
.reg_cache_step = 2,
};
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),
};
static int __init stac9766_init(void)
{
return platform_driver_register(&stac9766_codec_driver);
}
module_init(stac9766_init);
static void __exit stac9766_exit(void)
{
platform_driver_unregister(&stac9766_codec_driver);
}
module_exit(stac9766_exit);
MODULE_DESCRIPTION("ASoC stac9766 driver");
MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
MODULE_LICENSE("GPL");
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