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linux-next/sound/soc/jz4740/qi_lb60.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

163 lines
3.9 KiB
C

/*
* Copyright (C) 2009, Lars-Peter Clausen <lars@metafoo.de>
*
* 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.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <linux/gpio.h>
#include "../codecs/jz4740.h"
#include "jz4740-pcm.h"
#include "jz4740-i2s.h"
#define QI_LB60_SND_GPIO JZ_GPIO_PORTB(29)
#define QI_LB60_AMP_GPIO JZ_GPIO_PORTD(4)
static int qi_lb60_spk_event(struct snd_soc_dapm_widget *widget,
struct snd_kcontrol *ctrl, int event)
{
int on = 0;
if (event & SND_SOC_DAPM_POST_PMU)
on = 1;
else if (event & SND_SOC_DAPM_PRE_PMD)
on = 0;
gpio_set_value(QI_LB60_SND_GPIO, on);
gpio_set_value(QI_LB60_AMP_GPIO, on);
return 0;
}
static const struct snd_soc_dapm_widget qi_lb60_widgets[] = {
SND_SOC_DAPM_SPK("Speaker", qi_lb60_spk_event),
SND_SOC_DAPM_MIC("Mic", NULL),
};
static const struct snd_soc_dapm_route qi_lb60_routes[] = {
{"Mic", NULL, "MIC"},
{"Speaker", NULL, "LOUT"},
{"Speaker", NULL, "ROUT"},
};
#define QI_LB60_DAIFMT (SND_SOC_DAIFMT_I2S | \
SND_SOC_DAIFMT_NB_NF | \
SND_SOC_DAIFMT_CBM_CFM)
static int qi_lb60_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int ret;
snd_soc_dapm_nc_pin(codec, "LIN");
snd_soc_dapm_nc_pin(codec, "RIN");
ret = snd_soc_dai_set_fmt(cpu_dai, QI_LB60_DAIFMT);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cpu dai format: %d\n", ret);
return ret;
}
snd_soc_dapm_new_controls(codec, qi_lb60_widgets, ARRAY_SIZE(qi_lb60_widgets));
snd_soc_dapm_add_routes(codec, qi_lb60_routes, ARRAY_SIZE(qi_lb60_routes));
snd_soc_dapm_sync(codec);
return 0;
}
static struct snd_soc_dai_link qi_lb60_dai = {
.name = "jz4740",
.stream_name = "jz4740",
.cpu_dai_name = "jz4740-i2s",
.platform_name = "jz4740-pmc-audio",
.codec_dai_name = "jz4740-hifi",
.codec_name = "jz4740-codec",
.init = qi_lb60_codec_init,
};
static struct snd_soc_card qi_lb60 = {
.name = "QI LB60",
.dai_link = &qi_lb60_dai,
.num_links = 1,
};
static struct platform_device *qi_lb60_snd_device;
static int __init qi_lb60_init(void)
{
int ret;
qi_lb60_snd_device = platform_device_alloc("soc-audio", -1);
if (!qi_lb60_snd_device)
return -ENOMEM;
ret = gpio_request(QI_LB60_SND_GPIO, "SND");
if (ret) {
pr_err("qi_lb60 snd: Failed to request SND GPIO(%d): %d\n",
QI_LB60_SND_GPIO, ret);
goto err_device_put;
}
ret = gpio_request(QI_LB60_AMP_GPIO, "AMP");
if (ret) {
pr_err("qi_lb60 snd: Failed to request AMP GPIO(%d): %d\n",
QI_LB60_AMP_GPIO, ret);
goto err_gpio_free_snd;
}
gpio_direction_output(QI_LB60_SND_GPIO, 0);
gpio_direction_output(QI_LB60_AMP_GPIO, 0);
platform_set_drvdata(qi_lb60_snd_device, &qi_lb60);
ret = platform_device_add(qi_lb60_snd_device);
if (ret) {
pr_err("qi_lb60 snd: Failed to add snd soc device: %d\n", ret);
goto err_unset_pdata;
}
return 0;
err_unset_pdata:
platform_set_drvdata(qi_lb60_snd_device, NULL);
/*err_gpio_free_amp:*/
gpio_free(QI_LB60_AMP_GPIO);
err_gpio_free_snd:
gpio_free(QI_LB60_SND_GPIO);
err_device_put:
platform_device_put(qi_lb60_snd_device);
return ret;
}
module_init(qi_lb60_init);
static void __exit qi_lb60_exit(void)
{
gpio_free(QI_LB60_AMP_GPIO);
gpio_free(QI_LB60_SND_GPIO);
platform_device_unregister(qi_lb60_snd_device);
}
module_exit(qi_lb60_exit);
MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support");
MODULE_LICENSE("GPL v2");