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linux/sound/soc/codecs/wm9705.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

430 lines
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/*
* wm9705.c -- ALSA Soc WM9705 codec support
*
* Copyright 2008 Ian Molton <spyro@f2s.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; Version 2 of the License only.
*
*/
#include <linux/init.h>
#include <linux/slab.h>
#include <linux/module.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include "wm9705.h"
/*
* WM9705 register cache
*/
static const u16 wm9705_reg[] = {
0x6150, 0x8000, 0x8000, 0x8000, /* 0x0 */
0x0000, 0x8000, 0x8008, 0x8008, /* 0x8 */
0x8808, 0x8808, 0x8808, 0x8808, /* 0x10 */
0x8808, 0x0000, 0x8000, 0x0000, /* 0x18 */
0x0000, 0x0000, 0x0000, 0x000f, /* 0x20 */
0x0605, 0x0000, 0xbb80, 0x0000, /* 0x28 */
0x0000, 0xbb80, 0x0000, 0x0000, /* 0x30 */
0x0000, 0x2000, 0x0000, 0x0000, /* 0x38 */
0x0000, 0x0000, 0x0000, 0x0000, /* 0x40 */
0x0000, 0x0000, 0x0000, 0x0000, /* 0x48 */
0x0000, 0x0000, 0x0000, 0x0000, /* 0x50 */
0x0000, 0x0000, 0x0000, 0x0000, /* 0x58 */
0x0000, 0x0000, 0x0000, 0x0000, /* 0x60 */
0x0000, 0x0000, 0x0000, 0x0000, /* 0x68 */
0x0000, 0x0808, 0x0000, 0x0006, /* 0x70 */
0x0000, 0x0000, 0x574d, 0x4c05, /* 0x78 */
};
static const struct snd_kcontrol_new wm9705_snd_ac97_controls[] = {
SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1),
SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1),
SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1),
SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1),
SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
SOC_SINGLE("PCBeep Playback Volume", AC97_PC_BEEP, 1, 15, 1),
SOC_SINGLE("Phone Playback Volume", AC97_PHONE, 0, 31, 1),
SOC_DOUBLE("Line Playback Volume", AC97_LINE, 8, 0, 31, 1),
SOC_DOUBLE("CD Playback Volume", AC97_CD, 8, 0, 31, 1),
SOC_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1),
SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 6, 1, 0),
SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0),
SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1),
};
static const char *wm9705_mic[] = {"Mic 1", "Mic 2"};
static const char *wm9705_rec_sel[] = {"Mic", "CD", "NC", "NC",
"Line", "Stereo Mix", "Mono Mix", "Phone"};
static const struct soc_enum wm9705_enum_mic =
SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, wm9705_mic);
static const struct soc_enum wm9705_enum_rec_l =
SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9705_rec_sel);
static const struct soc_enum wm9705_enum_rec_r =
SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9705_rec_sel);
/* Headphone Mixer */
static const struct snd_kcontrol_new wm9705_hp_mixer_controls[] = {
SOC_DAPM_SINGLE("PCBeep Playback Switch", AC97_PC_BEEP, 15, 1, 1),
SOC_DAPM_SINGLE("CD Playback Switch", AC97_CD, 15, 1, 1),
SOC_DAPM_SINGLE("Mic Playback Switch", AC97_MIC, 15, 1, 1),
SOC_DAPM_SINGLE("Phone Playback Switch", AC97_PHONE, 15, 1, 1),
SOC_DAPM_SINGLE("Line Playback Switch", AC97_LINE, 15, 1, 1),
};
/* Mic source */
static const struct snd_kcontrol_new wm9705_mic_src_controls =
SOC_DAPM_ENUM("Route", wm9705_enum_mic);
/* Capture source */
static const struct snd_kcontrol_new wm9705_capture_selectl_controls =
SOC_DAPM_ENUM("Route", wm9705_enum_rec_l);
static const struct snd_kcontrol_new wm9705_capture_selectr_controls =
SOC_DAPM_ENUM("Route", wm9705_enum_rec_r);
/* DAPM widgets */
static const struct snd_soc_dapm_widget wm9705_dapm_widgets[] = {
SND_SOC_DAPM_MUX("Mic Source", SND_SOC_NOPM, 0, 0,
&wm9705_mic_src_controls),
SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0,
&wm9705_capture_selectl_controls),
SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0,
&wm9705_capture_selectr_controls),
SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MIXER_NAMED_CTL("HP Mixer", SND_SOC_NOPM, 0, 0,
&wm9705_hp_mixer_controls[0],
ARRAY_SIZE(wm9705_hp_mixer_controls)),
SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_PGA("Headphone PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Speaker PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Line PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Line out PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mono PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Phone PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mic PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("PCBEEP PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("CD PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("ADC PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("HPOUTL"),
SND_SOC_DAPM_OUTPUT("HPOUTR"),
SND_SOC_DAPM_OUTPUT("LOUT"),
SND_SOC_DAPM_OUTPUT("ROUT"),
SND_SOC_DAPM_OUTPUT("MONOOUT"),
SND_SOC_DAPM_INPUT("PHONE"),
SND_SOC_DAPM_INPUT("LINEINL"),
SND_SOC_DAPM_INPUT("LINEINR"),
SND_SOC_DAPM_INPUT("CDINL"),
SND_SOC_DAPM_INPUT("CDINR"),
SND_SOC_DAPM_INPUT("PCBEEP"),
SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2"),
};
/* Audio map
* WM9705 has no switches to disable the route from the inputs to the HP mixer
* so in order to prevent active inputs from forcing the audio outputs to be
* constantly enabled, we use the mutes on those inputs to simulate such
* controls.
*/
static const struct snd_soc_dapm_route audio_map[] = {
/* HP mixer */
{"HP Mixer", "PCBeep Playback Switch", "PCBEEP PGA"},
{"HP Mixer", "CD Playback Switch", "CD PGA"},
{"HP Mixer", "Mic Playback Switch", "Mic PGA"},
{"HP Mixer", "Phone Playback Switch", "Phone PGA"},
{"HP Mixer", "Line Playback Switch", "Line PGA"},
{"HP Mixer", NULL, "Left DAC"},
{"HP Mixer", NULL, "Right DAC"},
/* mono mixer */
{"Mono Mixer", NULL, "HP Mixer"},
/* outputs */
{"Headphone PGA", NULL, "HP Mixer"},
{"HPOUTL", NULL, "Headphone PGA"},
{"HPOUTR", NULL, "Headphone PGA"},
{"Line out PGA", NULL, "HP Mixer"},
{"LOUT", NULL, "Line out PGA"},
{"ROUT", NULL, "Line out PGA"},
{"Mono PGA", NULL, "Mono Mixer"},
{"MONOOUT", NULL, "Mono PGA"},
/* inputs */
{"CD PGA", NULL, "CDINL"},
{"CD PGA", NULL, "CDINR"},
{"Line PGA", NULL, "LINEINL"},
{"Line PGA", NULL, "LINEINR"},
{"Phone PGA", NULL, "PHONE"},
{"Mic Source", "Mic 1", "MIC1"},
{"Mic Source", "Mic 2", "MIC2"},
{"Mic PGA", NULL, "Mic Source"},
{"PCBEEP PGA", NULL, "PCBEEP"},
/* Left capture selector */
{"Left Capture Source", "Mic", "Mic Source"},
{"Left Capture Source", "CD", "CDINL"},
{"Left Capture Source", "Line", "LINEINL"},
{"Left Capture Source", "Stereo Mix", "HP Mixer"},
{"Left Capture Source", "Mono Mix", "HP Mixer"},
{"Left Capture Source", "Phone", "PHONE"},
/* Right capture source */
{"Right Capture Source", "Mic", "Mic Source"},
{"Right Capture Source", "CD", "CDINR"},
{"Right Capture Source", "Line", "LINEINR"},
{"Right Capture Source", "Stereo Mix", "HP Mixer"},
{"Right Capture Source", "Mono Mix", "HP Mixer"},
{"Right Capture Source", "Phone", "PHONE"},
{"ADC PGA", NULL, "Left Capture Source"},
{"ADC PGA", NULL, "Right Capture Source"},
/* ADC's */
{"Left ADC", NULL, "ADC PGA"},
{"Right ADC", NULL, "ADC PGA"},
};
static int wm9705_add_widgets(struct snd_soc_codec *codec)
{
snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets,
ARRAY_SIZE(wm9705_dapm_widgets));
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
/* We use a register cache to enhance read performance. */
static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg)
{
u16 *cache = codec->reg_cache;
switch (reg) {
case AC97_RESET:
case AC97_VENDOR_ID1:
case AC97_VENDOR_ID2:
return soc_ac97_ops.read(codec->ac97, reg);
default:
reg = reg >> 1;
if (reg >= (ARRAY_SIZE(wm9705_reg)))
return -EIO;
return cache[reg];
}
}
static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int val)
{
u16 *cache = codec->reg_cache;
soc_ac97_ops.write(codec->ac97, reg, val);
reg = reg >> 1;
if (reg < (ARRAY_SIZE(wm9705_reg)))
cache[reg] = val;
return 0;
}
static int ac97_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
int reg;
u16 vra;
vra = ac97_read(codec, AC97_EXTENDED_STATUS);
ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
reg = AC97_PCM_FRONT_DAC_RATE;
else
reg = AC97_PCM_LR_ADC_RATE;
return ac97_write(codec, reg, runtime->rate);
}
#define WM9705_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000)
static struct snd_soc_dai_ops wm9705_dai_ops = {
.prepare = ac97_prepare,
};
static struct snd_soc_dai_driver wm9705_dai[] = {
{
.name = "wm9705-hifi",
.ac97_control = 1,
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
.channels_max = 2,
.rates = WM9705_AC97_RATES,
.formats = SND_SOC_STD_AC97_FMTS,
},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM9705_AC97_RATES,
.formats = SND_SOC_STD_AC97_FMTS,
},
.ops = &wm9705_dai_ops,
},
{
.name = "wm9705-aux",
.playback = {
.stream_name = "Aux Playback",
.channels_min = 1,
.channels_max = 1,
.rates = WM9705_AC97_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
}
};
static int wm9705_reset(struct snd_soc_codec *codec)
{
if (soc_ac97_ops.reset) {
soc_ac97_ops.reset(codec->ac97);
if (ac97_read(codec, 0) == wm9705_reg[0])
return 0; /* Success */
}
return -EIO;
}
#ifdef CONFIG_PM
static int wm9705_soc_suspend(struct snd_soc_codec *codec, pm_message_t msg)
{
soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff);
return 0;
}
static int wm9705_soc_resume(struct snd_soc_codec *codec)
{
int i, ret;
u16 *cache = codec->reg_cache;
ret = wm9705_reset(codec);
if (ret < 0) {
printk(KERN_ERR "could not reset AC97 codec\n");
return ret;
}
for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) {
soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
}
return 0;
}
#else
#define wm9705_soc_suspend NULL
#define wm9705_soc_resume NULL
#endif
static int wm9705_soc_probe(struct snd_soc_codec *codec)
{
int ret = 0;
printk(KERN_INFO "WM9705 SoC Audio Codec\n");
ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
if (ret < 0) {
printk(KERN_ERR "wm9705: failed to register AC97 codec\n");
return ret;
}
ret = wm9705_reset(codec);
if (ret)
goto reset_err;
snd_soc_add_controls(codec, wm9705_snd_ac97_controls,
ARRAY_SIZE(wm9705_snd_ac97_controls));
wm9705_add_widgets(codec);
return 0;
reset_err:
snd_soc_free_ac97_codec(codec);
return ret;
}
static int wm9705_soc_remove(struct snd_soc_codec *codec)
{
snd_soc_free_ac97_codec(codec);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_wm9705 = {
.probe = wm9705_soc_probe,
.remove = wm9705_soc_remove,
.suspend = wm9705_soc_suspend,
.resume = wm9705_soc_resume,
.read = ac97_read,
.write = ac97_write,
.reg_cache_size = sizeof(wm9705_reg),
.reg_word_size = sizeof(u16),
.reg_cache_step = 2,
.reg_cache_default = wm9705_reg,
};
static __devinit int wm9705_probe(struct platform_device *pdev)
{
return snd_soc_register_codec(&pdev->dev,
&soc_codec_dev_wm9705, wm9705_dai, ARRAY_SIZE(wm9705_dai));
}
static int __devexit wm9705_remove(struct platform_device *pdev)
{
snd_soc_unregister_codec(&pdev->dev);
return 0;
}
static struct platform_driver wm9705_codec_driver = {
.driver = {
.name = "wm9705-codec",
.owner = THIS_MODULE,
},
.probe = wm9705_probe,
.remove = __devexit_p(wm9705_remove),
};
static int __init wm9705_init(void)
{
return platform_driver_register(&wm9705_codec_driver);
}
module_init(wm9705_init);
static void __exit wm9705_exit(void)
{
platform_driver_unregister(&wm9705_codec_driver);
}
module_exit(wm9705_exit);
MODULE_DESCRIPTION("ASoC WM9705 driver");
MODULE_AUTHOR("Ian Molton");
MODULE_LICENSE("GPL v2");
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