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linux/sound/soc/codecs/ad1980.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

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/*
* ad1980.c -- ALSA Soc AD1980 codec support
*
* Copyright: Analog Device Inc.
* Author: Roy Huang <roy.huang@analog.com>
* Cliff Cai <cliff.cai@analog.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.
*/
#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 "ad1980.h"
static unsigned int ac97_read(struct snd_soc_codec *codec,
unsigned int reg);
static int ac97_write(struct snd_soc_codec *codec,
unsigned int reg, unsigned int val);
/*
* AD1980 register cache
*/
static const u16 ad1980_reg[] = {
0x0090, 0x8000, 0x8000, 0x8000, /* 0 - 6 */
0x0000, 0x0000, 0x8008, 0x8008, /* 8 - e */
0x8808, 0x8808, 0x0000, 0x8808, /* 10 - 16 */
0x8808, 0x0000, 0x8000, 0x0000, /* 18 - 1e */
0x0000, 0x0000, 0x0000, 0x0000, /* 20 - 26 */
0x03c7, 0x0000, 0xbb80, 0xbb80, /* 28 - 2e */
0xbb80, 0xbb80, 0x0000, 0x8080, /* 30 - 36 */
0x8080, 0x2000, 0x0000, 0x0000, /* 38 - 3e */
0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
0x8080, 0x0000, 0x0000, 0x0000, /* 60 - 66 */
0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
0x0000, 0x0000, 0x1001, 0x0000, /* 70 - 76 */
0x0000, 0x0000, 0x4144, 0x5370 /* 78 - 7e */
};
static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line",
"Stereo Mix", "Mono Mix", "Phone"};
static const struct soc_enum ad1980_cap_src =
SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 7, ad1980_rec_sel);
static const struct snd_kcontrol_new ad1980_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_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 31, 0),
SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 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("Phone Capture Volume", AC97_PHONE, 0, 31, 1),
SOC_SINGLE("Phone Capture Switch", AC97_PHONE, 15, 1, 1),
SOC_SINGLE("Mic Volume", AC97_MIC, 0, 31, 1),
SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
SOC_SINGLE("Stereo Mic Switch", AC97_AD_MISC, 6, 1, 0),
SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0),
SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1),
SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1),
SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1),
SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1),
SOC_ENUM("Capture Source", ad1980_cap_src),
SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
};
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_INT_PAGING:
case AC97_POWERDOWN:
case AC97_EXTENDED_STATUS:
case AC97_VENDOR_ID1:
case AC97_VENDOR_ID2:
return soc_ac97_ops.read(codec->ac97, reg);
default:
reg = reg >> 1;
if (reg >= ARRAY_SIZE(ad1980_reg))
return -EINVAL;
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(ad1980_reg))
cache[reg] = val;
return 0;
}
struct snd_soc_dai_driver ad1980_dai = {
.name = "ad1980-hifi",
.ac97_control = 1,
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 6,
.rates = SNDRV_PCM_RATE_48000,
.formats = SND_SOC_STD_AC97_FMTS, },
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_48000,
.formats = SND_SOC_STD_AC97_FMTS, },
};
EXPORT_SYMBOL_GPL(ad1980_dai);
static int ad1980_reset(struct snd_soc_codec *codec, int try_warm)
{
u16 retry_cnt = 0;
retry:
if (try_warm && soc_ac97_ops.warm_reset) {
soc_ac97_ops.warm_reset(codec->ac97);
if (ac97_read(codec, AC97_RESET) == 0x0090)
return 1;
}
soc_ac97_ops.reset(codec->ac97);
/* Set bit 16slot in register 74h, then every slot will has only 16
* bits. This command is sent out in 20bit mode, in which case the
* first nibble of data is eaten by the addr. (Tag is always 16 bit)*/
ac97_write(codec, AC97_AD_SERIAL_CFG, 0x9900);
if (ac97_read(codec, AC97_RESET) != 0x0090)
goto err;
return 0;
err:
while (retry_cnt++ < 10)
goto retry;
printk(KERN_ERR "AD1980 AC97 reset failed\n");
return -EIO;
}
static int ad1980_soc_probe(struct snd_soc_codec *codec)
{
int ret;
u16 vendor_id2;
u16 ext_status;
printk(KERN_INFO "AD1980 SoC Audio Codec\n");
ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
if (ret < 0) {
printk(KERN_ERR "ad1980: failed to register AC97 codec\n");
return ret;
}
ret = ad1980_reset(codec, 0);
if (ret < 0) {
printk(KERN_ERR "Failed to reset AD1980: AC97 link error\n");
goto reset_err;
}
/* Read out vendor ID to make sure it is ad1980 */
if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144)
goto reset_err;
vendor_id2 = ac97_read(codec, AC97_VENDOR_ID2);
if (vendor_id2 != 0x5370) {
if (vendor_id2 != 0x5374)
goto reset_err;
else
printk(KERN_WARNING "ad1980: "
"Found AD1981 - only 2/2 IN/OUT Channels "
"supported\n");
}
/* unmute captures and playbacks volume */
ac97_write(codec, AC97_MASTER, 0x0000);
ac97_write(codec, AC97_PCM, 0x0000);
ac97_write(codec, AC97_REC_GAIN, 0x0000);
ac97_write(codec, AC97_CENTER_LFE_MASTER, 0x0000);
ac97_write(codec, AC97_SURROUND_MASTER, 0x0000);
/*power on LFE/CENTER/Surround DACs*/
ext_status = ac97_read(codec, AC97_EXTENDED_STATUS);
ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
snd_soc_add_controls(codec, ad1980_snd_ac97_controls,
ARRAY_SIZE(ad1980_snd_ac97_controls));
return 0;
reset_err:
snd_soc_free_ac97_codec(codec);
return ret;
}
static int ad1980_soc_remove(struct snd_soc_codec *codec)
{
snd_soc_free_ac97_codec(codec);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_ad1980 = {
.probe = ad1980_soc_probe,
.remove = ad1980_soc_remove,
.reg_cache_size = ARRAY_SIZE(ad1980_reg),
.reg_word_size = sizeof(u16),
.reg_cache_step = 2,
.write = ac97_write,
.read = ac97_read,
};
static __devinit int ad1980_probe(struct platform_device *pdev)
{
return snd_soc_register_codec(&pdev->dev,
&soc_codec_dev_ad1980, &ad1980_dai, 1);
}
static int __devexit ad1980_remove(struct platform_device *pdev)
{
snd_soc_unregister_codec(&pdev->dev);
return 0;
}
static struct platform_driver ad1980_codec_driver = {
.driver = {
.name = "ad1980-codec",
.owner = THIS_MODULE,
},
.probe = ad1980_probe,
.remove = __devexit_p(ad1980_remove),
};
static int __init ad1980_init(void)
{
return platform_driver_register(&ad1980_codec_driver);
}
module_init(ad1980_init);
static void __exit ad1980_exit(void)
{
platform_driver_unregister(&ad1980_codec_driver);
}
module_exit(ad1980_exit);
MODULE_DESCRIPTION("ASoC ad1980 driver");
MODULE_AUTHOR("Roy Huang, Cliff Cai");
MODULE_LICENSE("GPL");
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