linux/sound/soc/codecs/uda134x.c

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/*
* uda134x.c -- UDA134X ALSA SoC Codec driver
*
* Modifications by Christian Pellegrin <chripell@evolware.org>
*
* Copyright 2007 Dension Audio Systems Ltd.
* Author: Zoltan Devai
*
* Based on the WM87xx drivers by Liam Girdwood and Richard Purdie
*
* 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/module.h>
#include <linux/delay.h>
include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo <tj@kernel.org> Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org> Cc: Ingo Molnar <mingo@redhat.com> Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
2010-03-24 08:04:11 +00:00
#include <linux/slab.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/uda134x.h>
#include <sound/l3.h>
#include "uda134x.h"
#define UDA134X_RATES SNDRV_PCM_RATE_8000_48000
#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE)
struct uda134x_priv {
int sysclk;
int dai_fmt;
struct snd_pcm_substream *master_substream;
struct snd_pcm_substream *slave_substream;
struct regmap *regmap;
struct uda134x_platform_data *pd;
};
static const struct reg_default uda134x_reg_defaults[] = {
{ UDA134X_EA000, 0x04 },
{ UDA134X_EA001, 0x04 },
{ UDA134X_EA010, 0x04 },
{ UDA134X_EA011, 0x00 },
{ UDA134X_EA100, 0x00 },
{ UDA134X_EA101, 0x00 },
{ UDA134X_EA110, 0x00 },
{ UDA134X_EA111, 0x00 },
{ UDA134X_STATUS0, 0x00 },
{ UDA134X_STATUS1, 0x03 },
{ UDA134X_DATA000, 0x00 },
{ UDA134X_DATA001, 0x00 },
{ UDA134X_DATA010, 0x00 },
{ UDA134X_DATA011, 0x00 },
{ UDA134X_DATA1, 0x00 },
};
/*
* Write to the uda134x registers
*
*/
static int uda134x_regmap_write(void *context, unsigned int reg,
unsigned int value)
{
struct uda134x_platform_data *pd = context;
int ret;
u8 addr;
u8 data = value;
switch (reg) {
case UDA134X_STATUS0:
case UDA134X_STATUS1:
addr = UDA134X_STATUS_ADDR;
data |= (reg - UDA134X_STATUS0) << 7;
break;
case UDA134X_DATA000:
case UDA134X_DATA001:
case UDA134X_DATA010:
case UDA134X_DATA011:
addr = UDA134X_DATA0_ADDR;
data |= (reg - UDA134X_DATA000) << 6;
break;
case UDA134X_DATA1:
addr = UDA134X_DATA1_ADDR;
break;
default:
/* It's an extended address register */
addr = (reg | UDA134X_EXTADDR_PREFIX);
ret = l3_write(&pd->l3,
UDA134X_DATA0_ADDR, &addr, 1);
if (ret != 1)
return -EIO;
addr = UDA134X_DATA0_ADDR;
data = (value | UDA134X_EXTDATA_PREFIX);
break;
}
ret = l3_write(&pd->l3,
addr, &data, 1);
if (ret != 1)
return -EIO;
return 0;
}
static inline void uda134x_reset(struct snd_soc_component *component)
{
struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component);
unsigned int mask = 1<<6;
regmap_update_bits(uda134x->regmap, UDA134X_STATUS0, mask, mask);
msleep(1);
regmap_update_bits(uda134x->regmap, UDA134X_STATUS0, mask, 0);
}
static int uda134x_mute(struct snd_soc_dai *dai, int mute)
{
struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(dai->component);
unsigned int mask = 1<<2;
unsigned int val;
pr_debug("%s mute: %d\n", __func__, mute);
if (mute)
val = mask;
else
val = 0;
return regmap_update_bits(uda134x->regmap, UDA134X_DATA010, mask, val);
}
static int uda134x_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component);
struct snd_pcm_runtime *master_runtime;
if (uda134x->master_substream) {
master_runtime = uda134x->master_substream->runtime;
pr_debug("%s constraining to %d bits at %d\n", __func__,
master_runtime->sample_bits,
master_runtime->rate);
snd_pcm_hw_constraint_single(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE,
master_runtime->rate);
snd_pcm_hw_constraint_single(substream->runtime,
SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
master_runtime->sample_bits);
uda134x->slave_substream = substream;
} else
uda134x->master_substream = substream;
return 0;
}
static void uda134x_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component);
if (uda134x->master_substream == substream)
uda134x->master_substream = uda134x->slave_substream;
uda134x->slave_substream = NULL;
}
static int uda134x_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component);
unsigned int hw_params = 0;
if (substream == uda134x->slave_substream) {
pr_debug("%s ignoring hw_params for slave substream\n",
__func__);
return 0;
}
pr_debug("%s sysclk: %d, rate:%d\n", __func__,
uda134x->sysclk, params_rate(params));
/* set SYSCLK / fs ratio */
switch (uda134x->sysclk / params_rate(params)) {
case 512:
break;
case 384:
hw_params |= (1<<4);
break;
case 256:
hw_params |= (1<<5);
break;
default:
printk(KERN_ERR "%s unsupported fs\n", __func__);
return -EINVAL;
}
pr_debug("%s dai_fmt: %d, params_format:%d\n", __func__,
uda134x->dai_fmt, params_format(params));
/* set DAI format and word length */
switch (uda134x->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
break;
case SND_SOC_DAIFMT_RIGHT_J:
switch (params_width(params)) {
case 16:
hw_params |= (1<<1);
break;
case 18:
hw_params |= (1<<2);
break;
case 20:
hw_params |= ((1<<2) | (1<<1));
break;
default:
printk(KERN_ERR "%s unsupported format (right)\n",
__func__);
return -EINVAL;
}
break;
case SND_SOC_DAIFMT_LEFT_J:
hw_params |= (1<<3);
break;
default:
printk(KERN_ERR "%s unsupported format\n", __func__);
return -EINVAL;
}
return regmap_update_bits(uda134x->regmap, UDA134X_STATUS0,
STATUS0_SYSCLK_MASK | STATUS0_DAIFMT_MASK, hw_params);
}
static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_component *component = codec_dai->component;
struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component);
pr_debug("%s clk_id: %d, freq: %u, dir: %d\n", __func__,
clk_id, freq, dir);
/* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
because the codec is slave. Of course limitations of the clock
master (the IIS controller) apply.
We'll error out on set_hw_params if it's not OK */
if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) {
uda134x->sysclk = freq;
return 0;
}
printk(KERN_ERR "%s unsupported sysclk\n", __func__);
return -EINVAL;
}
static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_component *component = codec_dai->component;
struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component);
pr_debug("%s fmt: %08X\n", __func__, fmt);
/* codec supports only full slave mode */
if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
printk(KERN_ERR "%s unsupported slave mode\n", __func__);
return -EINVAL;
}
/* no support for clock inversion */
if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
printk(KERN_ERR "%s unsupported clock inversion\n", __func__);
return -EINVAL;
}
/* We can't setup DAI format here as it depends on the word bit num */
/* so let's just store the value for later */
uda134x->dai_fmt = fmt;
return 0;
}
static int uda134x_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component);
struct uda134x_platform_data *pd = uda134x->pd;
pr_debug("%s bias level %d\n", __func__, level);
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
/* power on */
if (pd->power) {
pd->power(1);
regcache_sync(uda134x->regmap);
}
break;
case SND_SOC_BIAS_STANDBY:
break;
case SND_SOC_BIAS_OFF:
/* power off */
if (pd->power) {
pd->power(0);
regcache_mark_dirty(uda134x->regmap);
}
break;
}
return 0;
}
static const char *uda134x_dsp_setting[] = {"Flat", "Minimum1",
"Minimum2", "Maximum"};
static const char *uda134x_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
static const char *uda134x_mixmode[] = {"Differential", "Analog1",
"Analog2", "Both"};
static const struct soc_enum uda134x_mixer_enum[] = {
SOC_ENUM_SINGLE(UDA134X_DATA010, 0, 0x04, uda134x_dsp_setting),
SOC_ENUM_SINGLE(UDA134X_DATA010, 3, 0x04, uda134x_deemph),
SOC_ENUM_SINGLE(UDA134X_EA010, 0, 0x04, uda134x_mixmode),
};
static const struct snd_kcontrol_new uda1341_snd_controls[] = {
SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
SOC_SINGLE("Capture Volume", UDA134X_EA010, 2, 0x07, 0),
SOC_SINGLE("Analog1 Volume", UDA134X_EA000, 0, 0x1F, 1),
SOC_SINGLE("Analog2 Volume", UDA134X_EA001, 0, 0x1F, 1),
SOC_SINGLE("Mic Sensitivity", UDA134X_EA010, 2, 7, 0),
SOC_SINGLE("Mic Volume", UDA134X_EA101, 0, 0x1F, 0),
SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0),
SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0),
SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]),
SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
SOC_ENUM("Input Mux", uda134x_mixer_enum[2]),
SOC_SINGLE("AGC Switch", UDA134X_EA100, 4, 1, 0),
SOC_SINGLE("AGC Target Volume", UDA134X_EA110, 0, 0x03, 1),
SOC_SINGLE("AGC Timing", UDA134X_EA110, 2, 0x07, 0),
SOC_SINGLE("DAC +6dB Switch", UDA134X_STATUS1, 6, 1, 0),
SOC_SINGLE("ADC +6dB Switch", UDA134X_STATUS1, 5, 1, 0),
SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0),
SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0),
SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0),
SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
};
static const struct snd_kcontrol_new uda1340_snd_controls[] = {
SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0),
SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0),
SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]),
SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
};
static const struct snd_kcontrol_new uda1345_snd_controls[] = {
SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
};
/* UDA1341 has the DAC/ADC power down in STATUS1 */
static const struct snd_soc_dapm_widget uda1341_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_STATUS1, 0, 0),
SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_STATUS1, 1, 0),
};
/* UDA1340/4/5 has the DAC/ADC pwoer down in DATA0 11 */
static const struct snd_soc_dapm_widget uda1340_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_DATA011, 0, 0),
SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_DATA011, 1, 0),
};
/* Common DAPM widgets */
static const struct snd_soc_dapm_widget uda134x_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("VINL1"),
SND_SOC_DAPM_INPUT("VINR1"),
SND_SOC_DAPM_INPUT("VINL2"),
SND_SOC_DAPM_INPUT("VINR2"),
SND_SOC_DAPM_OUTPUT("VOUTL"),
SND_SOC_DAPM_OUTPUT("VOUTR"),
};
static const struct snd_soc_dapm_route uda134x_dapm_routes[] = {
{ "ADC", NULL, "VINL1" },
{ "ADC", NULL, "VINR1" },
{ "ADC", NULL, "VINL2" },
{ "ADC", NULL, "VINR2" },
{ "VOUTL", NULL, "DAC" },
{ "VOUTR", NULL, "DAC" },
};
static const struct snd_soc_dai_ops uda134x_dai_ops = {
.startup = uda134x_startup,
.shutdown = uda134x_shutdown,
.hw_params = uda134x_hw_params,
.digital_mute = uda134x_mute,
.set_sysclk = uda134x_set_dai_sysclk,
.set_fmt = uda134x_set_dai_fmt,
};
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>
2010-03-17 20:15:21 +00:00
static struct snd_soc_dai_driver uda134x_dai = {
.name = "uda134x-hifi",
/* playback capabilities */
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = UDA134X_RATES,
.formats = UDA134X_FORMATS,
},
/* capture capabilities */
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = UDA134X_RATES,
.formats = UDA134X_FORMATS,
},
/* pcm operations */
.ops = &uda134x_dai_ops,
};
static int uda134x_soc_probe(struct snd_soc_component *component)
{
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component);
struct uda134x_platform_data *pd = uda134x->pd;
const struct snd_soc_dapm_widget *widgets;
unsigned num_widgets;
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>
2010-03-17 20:15:21 +00:00
int ret;
printk(KERN_INFO "UDA134X SoC Audio Codec\n");
switch (pd->model) {
case UDA134X_UDA1340:
case UDA134X_UDA1341:
case UDA134X_UDA1344:
case UDA134X_UDA1345:
break;
default:
printk(KERN_ERR "UDA134X SoC codec: "
"unsupported model %d\n",
pd->model);
return -EINVAL;
}
if (pd->power)
pd->power(1);
uda134x_reset(component);
if (pd->model == UDA134X_UDA1341) {
widgets = uda1341_dapm_widgets;
num_widgets = ARRAY_SIZE(uda1341_dapm_widgets);
} else {
widgets = uda1340_dapm_widgets;
num_widgets = ARRAY_SIZE(uda1340_dapm_widgets);
}
ret = snd_soc_dapm_new_controls(dapm, widgets, num_widgets);
if (ret) {
printk(KERN_ERR "%s failed to register dapm controls: %d",
__func__, ret);
return ret;
}
switch (pd->model) {
case UDA134X_UDA1340:
case UDA134X_UDA1344:
ret = snd_soc_add_component_controls(component, uda1340_snd_controls,
ARRAY_SIZE(uda1340_snd_controls));
break;
case UDA134X_UDA1341:
ret = snd_soc_add_component_controls(component, uda1341_snd_controls,
ARRAY_SIZE(uda1341_snd_controls));
break;
case UDA134X_UDA1345:
ret = snd_soc_add_component_controls(component, uda1345_snd_controls,
ARRAY_SIZE(uda1345_snd_controls));
break;
default:
printk(KERN_ERR "%s unknown codec type: %d",
__func__, pd->model);
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>
2010-03-17 20:15:21 +00:00
return -EINVAL;
}
if (ret < 0) {
printk(KERN_ERR "UDA134X: failed to register controls\n");
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>
2010-03-17 20:15:21 +00:00
return ret;
}
return 0;
}
static const struct snd_soc_component_driver soc_component_dev_uda134x = {
.probe = uda134x_soc_probe,
.set_bias_level = uda134x_set_bias_level,
.dapm_widgets = uda134x_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(uda134x_dapm_widgets),
.dapm_routes = uda134x_dapm_routes,
.num_dapm_routes = ARRAY_SIZE(uda134x_dapm_routes),
.suspend_bias_off = 1,
.idle_bias_on = 1,
.use_pmdown_time = 1,
.endianness = 1,
.non_legacy_dai_naming = 1,
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>
2010-03-17 20:15:21 +00:00
};
static const struct regmap_config uda134x_regmap_config = {
.reg_bits = 8,
.val_bits = 8,
.max_register = UDA134X_DATA1,
.reg_defaults = uda134x_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(uda134x_reg_defaults),
.cache_type = REGCACHE_RBTREE,
.reg_write = uda134x_regmap_write,
};
static int uda134x_codec_probe(struct platform_device *pdev)
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>
2010-03-17 20:15:21 +00:00
{
struct uda134x_platform_data *pd = pdev->dev.platform_data;
struct uda134x_priv *uda134x;
int ret;
if (!pd) {
dev_err(&pdev->dev, "Missing L3 bitbang function\n");
return -ENODEV;
}
uda134x = devm_kzalloc(&pdev->dev, sizeof(*uda134x), GFP_KERNEL);
if (!uda134x)
return -ENOMEM;
uda134x->pd = pd;
platform_set_drvdata(pdev, uda134x);
if (pd->l3.use_gpios) {
ret = l3_set_gpio_ops(&pdev->dev, &uda134x->pd->l3);
if (ret < 0)
return ret;
}
uda134x->regmap = devm_regmap_init(&pdev->dev, NULL, pd,
&uda134x_regmap_config);
if (IS_ERR(uda134x->regmap))
return PTR_ERR(uda134x->regmap);
return devm_snd_soc_register_component(&pdev->dev,
&soc_component_dev_uda134x, &uda134x_dai, 1);
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>
2010-03-17 20:15:21 +00:00
}
static struct platform_driver uda134x_codec_driver = {
.driver = {
.name = "uda134x-codec",
},
.probe = uda134x_codec_probe,
};
module_platform_driver(uda134x_codec_driver);
MODULE_DESCRIPTION("UDA134X ALSA soc codec driver");
MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>");
MODULE_LICENSE("GPL");