linux/sound/soc/samsung/s3c24xx_uda134x.c

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/*
* Modifications by Christian Pellegrin <chripell@evolware.org>
*
* s3c24xx_uda134x.c -- S3C24XX_UDA134X ALSA SoC Audio board driver
*
* Copyright 2007 Dension Audio Systems Ltd.
* Author: Zoltan Devai
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#include <linux/clk.h>
#include <linux/gpio.h>
#include <linux/module.h>
#include <sound/soc.h>
#include <sound/s3c24xx_uda134x.h>
#include <plat/regs-iis.h>
#include "s3c24xx-i2s.h"
/* #define ENFORCE_RATES 1 */
/*
Unfortunately the S3C24XX in master mode has a limited capacity of
generating the clock for the codec. If you define this only rates
that are really available will be enforced. But be careful, most
user level application just want the usual sampling frequencies (8,
11.025, 22.050, 44.1 kHz) and anyway resampling is a costly
operation for embedded systems. So if you aren't very lucky or your
hardware engineer wasn't very forward-looking it's better to leave
this undefined. If you do so an approximate value for the requested
sampling rate in the range -/+ 5% will be chosen. If this in not
possible an error will be returned.
*/
static struct clk *xtal;
static struct clk *pclk;
/* this is need because we don't have a place where to keep the
* pointers to the clocks in each substream. We get the clocks only
* when we are actually using them so we don't block stuff like
* frequency change or oscillator power-off */
static int clk_users;
static DEFINE_MUTEX(clk_lock);
static unsigned int rates[33 * 2];
#ifdef ENFORCE_RATES
static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
.count = ARRAY_SIZE(rates),
.list = rates,
.mask = 0,
};
#endif
static struct platform_device *s3c24xx_uda134x_snd_device;
static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
{
int ret = 0;
#ifdef ENFORCE_RATES
struct snd_pcm_runtime *runtime = substream->runtime;
#endif
mutex_lock(&clk_lock);
pr_debug("%s %d\n", __func__, clk_users);
if (clk_users == 0) {
xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal");
if (IS_ERR(xtal)) {
printk(KERN_ERR "%s cannot get xtal\n", __func__);
ret = PTR_ERR(xtal);
} else {
pclk = clk_get(&s3c24xx_uda134x_snd_device->dev,
"pclk");
if (IS_ERR(pclk)) {
printk(KERN_ERR "%s cannot get pclk\n",
__func__);
clk_put(xtal);
ret = PTR_ERR(pclk);
}
}
if (!ret) {
int i, j;
for (i = 0; i < 2; i++) {
int fs = i ? 256 : 384;
rates[i*33] = clk_get_rate(xtal) / fs;
for (j = 1; j < 33; j++)
rates[i*33 + j] = clk_get_rate(pclk) /
(j * fs);
}
}
}
clk_users += 1;
mutex_unlock(&clk_lock);
if (!ret) {
#ifdef ENFORCE_RATES
ret = snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&hw_constraints_rates);
if (ret < 0)
printk(KERN_ERR "%s cannot set constraints\n",
__func__);
#endif
}
return ret;
}
static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream)
{
mutex_lock(&clk_lock);
pr_debug("%s %d\n", __func__, clk_users);
clk_users -= 1;
if (clk_users == 0) {
clk_put(xtal);
xtal = NULL;
clk_put(pclk);
pclk = NULL;
}
mutex_unlock(&clk_lock);
}
static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
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>
2010-03-18 04:15:21 +08:00
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
unsigned int clk = 0;
int ret = 0;
int clk_source, fs_mode;
unsigned long rate = params_rate(params);
long err, cerr;
unsigned int div;
int i, bi;
err = 999999;
bi = 0;
for (i = 0; i < 2*33; i++) {
cerr = rates[i] - rate;
if (cerr < 0)
cerr = -cerr;
if (cerr < err) {
err = cerr;
bi = i;
}
}
if (bi / 33 == 1)
fs_mode = S3C2410_IISMOD_256FS;
else
fs_mode = S3C2410_IISMOD_384FS;
if (bi % 33 == 0) {
clk_source = S3C24XX_CLKSRC_MPLL;
div = 1;
} else {
clk_source = S3C24XX_CLKSRC_PCLK;
div = bi % 33;
}
pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi);
clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate;
pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__,
fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS",
clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK",
div, clk, err);
if ((err * 100 / rate) > 5) {
printk(KERN_ERR "S3C24XX_UDA134X: effective frequency "
"too different from desired (%ld%%)\n",
err * 100 / rate);
return -EINVAL;
}
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, fs_mode);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
S3C2410_IISMOD_32FS);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
S3C24XX_PRESCALE(div, div));
if (ret < 0)
return ret;
/* set the codec system clock for DAC and ADC */
ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk,
SND_SOC_CLOCK_OUT);
if (ret < 0)
return ret;
return 0;
}
static struct snd_soc_ops s3c24xx_uda134x_ops = {
.startup = s3c24xx_uda134x_startup,
.shutdown = s3c24xx_uda134x_shutdown,
.hw_params = s3c24xx_uda134x_hw_params,
};
static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
.name = "UDA134X",
.stream_name = "UDA134X",
.codec_name = "uda134x-codec",
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>
2010-03-18 04:15:21 +08:00
.codec_dai_name = "uda134x-hifi",
.cpu_dai_name = "s3c24xx-iis",
.ops = &s3c24xx_uda134x_ops,
.platform_name = "s3c24xx-iis",
};
static struct snd_soc_card snd_soc_s3c24xx_uda134x = {
.name = "S3C24XX_UDA134X",
.owner = THIS_MODULE,
.dai_link = &s3c24xx_uda134x_dai_link,
.num_links = 1,
};
static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins;
static void setdat(int v)
{
gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0);
}
static void setclk(int v)
{
gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0);
}
static void setmode(int v)
{
gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0);
}
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>
2010-03-18 04:15:21 +08:00
/* FIXME - This must be codec platform data but in which board file ?? */
static struct uda134x_platform_data s3c24xx_uda134x = {
.l3 = {
.setdat = setdat,
.setclk = setclk,
.setmode = setmode,
.data_hold = 1,
.data_setup = 1,
.clock_high = 1,
.mode_hold = 1,
.mode = 1,
.mode_setup = 1,
},
};
static int s3c24xx_uda134x_setup_pin(int pin, char *fun)
{
if (gpio_request(pin, "s3c24xx_uda134x") < 0) {
printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
"l3 %s pin already in use", fun);
return -EBUSY;
}
gpio_direction_output(pin, 0);
return 0;
}
static int s3c24xx_uda134x_probe(struct platform_device *pdev)
{
int ret;
printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n");
s3c24xx_uda134x_l3_pins = pdev->dev.platform_data;
if (s3c24xx_uda134x_l3_pins == NULL) {
printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
"unable to find platform data\n");
return -ENODEV;
}
s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power;
s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model;
if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data,
"data") < 0)
return -EBUSY;
if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk,
"clk") < 0) {
gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
return -EBUSY;
}
if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode,
"mode") < 0) {
gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
gpio_free(s3c24xx_uda134x_l3_pins->l3_clk);
return -EBUSY;
}
s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1);
if (!s3c24xx_uda134x_snd_device) {
printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
"Unable to register\n");
return -ENOMEM;
}
platform_set_drvdata(s3c24xx_uda134x_snd_device,
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>
2010-03-18 04:15:21 +08:00
&snd_soc_s3c24xx_uda134x);
platform_device_add_data(s3c24xx_uda134x_snd_device, &s3c24xx_uda134x, sizeof(s3c24xx_uda134x));
ret = platform_device_add(s3c24xx_uda134x_snd_device);
if (ret) {
printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n");
platform_device_put(s3c24xx_uda134x_snd_device);
}
return ret;
}
static int s3c24xx_uda134x_remove(struct platform_device *pdev)
{
platform_device_unregister(s3c24xx_uda134x_snd_device);
gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
gpio_free(s3c24xx_uda134x_l3_pins->l3_clk);
gpio_free(s3c24xx_uda134x_l3_pins->l3_mode);
return 0;
}
static struct platform_driver s3c24xx_uda134x_driver = {
.probe = s3c24xx_uda134x_probe,
.remove = s3c24xx_uda134x_remove,
.driver = {
.name = "s3c24xx_uda134x",
.owner = THIS_MODULE,
},
};
module_platform_driver(s3c24xx_uda134x_driver);
MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>");
MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver");
MODULE_LICENSE("GPL");