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linux-next/sound/soc/omap/n810.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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/*
* n810.c -- SoC audio for Nokia N810
*
* Copyright (C) 2008 Nokia Corporation
*
* Contact: Jarkko Nikula <jhnikula@gmail.com>
*
* 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.
*
* This program is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
* 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., 51 Franklin St, Fifth Floor, Boston, MA
* 02110-1301 USA
*
*/
#include <linux/clk.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <linux/gpio.h>
#include <plat/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
#include "../codecs/tlv320aic3x.h"
#define N810_HEADSET_AMP_GPIO 10
#define N810_SPEAKER_AMP_GPIO 101
enum {
N810_JACK_DISABLED,
N810_JACK_HP,
N810_JACK_HS,
N810_JACK_MIC,
};
static struct clk *sys_clkout2;
static struct clk *sys_clkout2_src;
static struct clk *func96m_clk;
static int n810_spk_func;
static int n810_jack_func;
static int n810_dmic_func;
static void n810_ext_control(struct snd_soc_codec *codec)
{
int hp = 0, line1l = 0;
switch (n810_jack_func) {
case N810_JACK_HS:
line1l = 1;
case N810_JACK_HP:
hp = 1;
break;
case N810_JACK_MIC:
line1l = 1;
break;
}
if (n810_spk_func)
snd_soc_dapm_enable_pin(codec, "Ext Spk");
else
snd_soc_dapm_disable_pin(codec, "Ext Spk");
if (hp)
snd_soc_dapm_enable_pin(codec, "Headphone Jack");
else
snd_soc_dapm_disable_pin(codec, "Headphone Jack");
if (line1l)
snd_soc_dapm_enable_pin(codec, "LINE1L");
else
snd_soc_dapm_disable_pin(codec, "LINE1L");
if (n810_dmic_func)
snd_soc_dapm_enable_pin(codec, "DMic");
else
snd_soc_dapm_disable_pin(codec, "DMic");
snd_soc_dapm_sync(codec);
}
static int n810_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
snd_pcm_hw_constraint_minmax(runtime,
SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
n810_ext_control(codec);
return clk_enable(sys_clkout2);
}
static void n810_shutdown(struct snd_pcm_substream *substream)
{
clk_disable(sys_clkout2);
}
static int n810_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int err;
/* Set codec DAI configuration */
err = snd_soc_dai_set_fmt(codec_dai,
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (err < 0)
return err;
/* Set cpu DAI configuration */
err = snd_soc_dai_set_fmt(cpu_dai,
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (err < 0)
return err;
/* Set the codec system clock for DAC and ADC */
err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000,
SND_SOC_CLOCK_IN);
return err;
}
static struct snd_soc_ops n810_ops = {
.startup = n810_startup,
.hw_params = n810_hw_params,
.shutdown = n810_shutdown,
};
static int n810_get_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = n810_spk_func;
return 0;
}
static int n810_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
if (n810_spk_func == ucontrol->value.integer.value[0])
return 0;
n810_spk_func = ucontrol->value.integer.value[0];
n810_ext_control(codec);
return 1;
}
static int n810_get_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = n810_jack_func;
return 0;
}
static int n810_set_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
if (n810_jack_func == ucontrol->value.integer.value[0])
return 0;
n810_jack_func = ucontrol->value.integer.value[0];
n810_ext_control(codec);
return 1;
}
static int n810_get_input(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = n810_dmic_func;
return 0;
}
static int n810_set_input(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
if (n810_dmic_func == ucontrol->value.integer.value[0])
return 0;
n810_dmic_func = ucontrol->value.integer.value[0];
n810_ext_control(codec);
return 1;
}
static int n810_spk_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
if (SND_SOC_DAPM_EVENT_ON(event))
gpio_set_value(N810_SPEAKER_AMP_GPIO, 1);
else
gpio_set_value(N810_SPEAKER_AMP_GPIO, 0);
return 0;
}
static int n810_jack_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
if (SND_SOC_DAPM_EVENT_ON(event))
gpio_set_value(N810_HEADSET_AMP_GPIO, 1);
else
gpio_set_value(N810_HEADSET_AMP_GPIO, 0);
return 0;
}
static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = {
SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event),
SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event),
SND_SOC_DAPM_MIC("DMic", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
{"Headphone Jack", NULL, "HPLOUT"},
{"Headphone Jack", NULL, "HPROUT"},
{"Ext Spk", NULL, "LLOUT"},
{"Ext Spk", NULL, "RLOUT"},
{"DMic Rate 64", NULL, "Mic Bias 2V"},
{"Mic Bias 2V", NULL, "DMic"},
};
static const char *spk_function[] = {"Off", "On"};
static const char *jack_function[] = {"Off", "Headphone", "Headset", "Mic"};
static const char *input_function[] = {"ADC", "Digital Mic"};
static const struct soc_enum n810_enum[] = {
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
};
static const struct snd_kcontrol_new aic33_n810_controls[] = {
SOC_ENUM_EXT("Speaker Function", n810_enum[0],
n810_get_spk, n810_set_spk),
SOC_ENUM_EXT("Jack Function", n810_enum[1],
n810_get_jack, n810_set_jack),
SOC_ENUM_EXT("Input Select", n810_enum[2],
n810_get_input, n810_set_input),
};
static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
int err;
/* Not connected */
snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
snd_soc_dapm_nc_pin(codec, "HPLCOM");
snd_soc_dapm_nc_pin(codec, "HPRCOM");
snd_soc_dapm_nc_pin(codec, "MIC3L");
snd_soc_dapm_nc_pin(codec, "MIC3R");
snd_soc_dapm_nc_pin(codec, "LINE1R");
snd_soc_dapm_nc_pin(codec, "LINE2L");
snd_soc_dapm_nc_pin(codec, "LINE2R");
/* Add N810 specific controls */
err = snd_soc_add_controls(codec, aic33_n810_controls,
ARRAY_SIZE(aic33_n810_controls));
if (err < 0)
return err;
/* Add N810 specific widgets */
snd_soc_dapm_new_controls(codec, aic33_dapm_widgets,
ARRAY_SIZE(aic33_dapm_widgets));
/* Set up N810 specific audio path audio_map */
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
snd_soc_dapm_sync(codec);
return 0;
}
/* Digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link n810_dai = {
.name = "TLV320AIC33",
.stream_name = "AIC33",
.cpu_dai_name = "omap-mcbsp-dai.1",
.platform_name = "omap-pcm-audio",
.codec_name = "tlv320aic3x-codec.2-0018",
.codec_dai_name = "tlv320aic3x-hifi",
.init = n810_aic33_init,
.ops = &n810_ops,
};
/* Audio machine driver */
static struct snd_soc_card snd_soc_n810 = {
.name = "N810",
.dai_link = &n810_dai,
.num_links = 1,
};
static struct platform_device *n810_snd_device;
static int __init n810_soc_init(void)
{
int err;
struct device *dev;
if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax()))
return -ENODEV;
n810_snd_device = platform_device_alloc("soc-audio", -1);
if (!n810_snd_device)
return -ENOMEM;
platform_set_drvdata(n810_snd_device, &snd_soc_n810);
err = platform_device_add(n810_snd_device);
if (err)
goto err1;
dev = &n810_snd_device->dev;
sys_clkout2_src = clk_get(dev, "sys_clkout2_src");
if (IS_ERR(sys_clkout2_src)) {
dev_err(dev, "Could not get sys_clkout2_src clock\n");
err = PTR_ERR(sys_clkout2_src);
goto err2;
}
sys_clkout2 = clk_get(dev, "sys_clkout2");
if (IS_ERR(sys_clkout2)) {
dev_err(dev, "Could not get sys_clkout2\n");
err = PTR_ERR(sys_clkout2);
goto err3;
}
/*
* Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use
* 96 MHz as its parent in order to get 12 MHz
*/
func96m_clk = clk_get(dev, "func_96m_ck");
if (IS_ERR(func96m_clk)) {
dev_err(dev, "Could not get func 96M clock\n");
err = PTR_ERR(func96m_clk);
goto err4;
}
clk_set_parent(sys_clkout2_src, func96m_clk);
clk_set_rate(sys_clkout2, 12000000);
BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) ||
(gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0));
gpio_direction_output(N810_HEADSET_AMP_GPIO, 0);
gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
return 0;
err4:
clk_put(sys_clkout2);
err3:
clk_put(sys_clkout2_src);
err2:
platform_device_del(n810_snd_device);
err1:
platform_device_put(n810_snd_device);
return err;
}
static void __exit n810_soc_exit(void)
{
gpio_free(N810_SPEAKER_AMP_GPIO);
gpio_free(N810_HEADSET_AMP_GPIO);
clk_put(sys_clkout2_src);
clk_put(sys_clkout2);
clk_put(func96m_clk);
platform_device_unregister(n810_snd_device);
}
module_init(n810_soc_init);
module_exit(n810_soc_exit);
MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
MODULE_DESCRIPTION("ALSA SoC Nokia N810");
MODULE_LICENSE("GPL");
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