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kernel_optimize_test/sound/soc/pxa/spitz.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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9.5 KiB
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/*
* spitz.c -- SoC audio for Sharp SL-Cxx00 models Spitz, Borzoi and Akita
*
* Copyright 2005 Wolfson Microelectronics PLC.
* Copyright 2005 Openedhand Ltd.
*
* Authors: Liam Girdwood <lrg@slimlogic.co.uk>
* Richard Purdie <richard@openedhand.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <linux/gpio.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/spitz.h>
#include "../codecs/wm8750.h"
#include "pxa2xx-i2s.h"
#define SPITZ_HP 0
#define SPITZ_MIC 1
#define SPITZ_LINE 2
#define SPITZ_HEADSET 3
#define SPITZ_HP_OFF 4
#define SPITZ_SPK_ON 0
#define SPITZ_SPK_OFF 1
/* audio clock in Hz - rounded from 12.235MHz */
#define SPITZ_AUDIO_CLOCK 12288000
static int spitz_jack_func;
static int spitz_spk_func;
static void spitz_ext_control(struct snd_soc_codec *codec)
{
if (spitz_spk_func == SPITZ_SPK_ON)
snd_soc_dapm_enable_pin(codec, "Ext Spk");
else
snd_soc_dapm_disable_pin(codec, "Ext Spk");
/* set up jack connection */
switch (spitz_jack_func) {
case SPITZ_HP:
/* enable and unmute hp jack, disable mic bias */
snd_soc_dapm_disable_pin(codec, "Headset Jack");
snd_soc_dapm_disable_pin(codec, "Mic Jack");
snd_soc_dapm_disable_pin(codec, "Line Jack");
snd_soc_dapm_enable_pin(codec, "Headphone Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 1);
gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
break;
case SPITZ_MIC:
/* enable mic jack and bias, mute hp */
snd_soc_dapm_disable_pin(codec, "Headphone Jack");
snd_soc_dapm_disable_pin(codec, "Headset Jack");
snd_soc_dapm_disable_pin(codec, "Line Jack");
snd_soc_dapm_enable_pin(codec, "Mic Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
case SPITZ_LINE:
/* enable line jack, disable mic bias and mute hp */
snd_soc_dapm_disable_pin(codec, "Headphone Jack");
snd_soc_dapm_disable_pin(codec, "Headset Jack");
snd_soc_dapm_disable_pin(codec, "Mic Jack");
snd_soc_dapm_enable_pin(codec, "Line Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
case SPITZ_HEADSET:
/* enable and unmute headset jack enable mic bias, mute L hp */
snd_soc_dapm_disable_pin(codec, "Headphone Jack");
snd_soc_dapm_enable_pin(codec, "Mic Jack");
snd_soc_dapm_disable_pin(codec, "Line Jack");
snd_soc_dapm_enable_pin(codec, "Headset Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
break;
case SPITZ_HP_OFF:
/* jack removed, everything off */
snd_soc_dapm_disable_pin(codec, "Headphone Jack");
snd_soc_dapm_disable_pin(codec, "Headset Jack");
snd_soc_dapm_disable_pin(codec, "Mic Jack");
snd_soc_dapm_disable_pin(codec, "Line Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
}
snd_soc_dapm_sync(codec);
}
static int spitz_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
/* check the jack status at stream startup */
spitz_ext_control(codec);
return 0;
}
static int spitz_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;
unsigned int clk = 0;
int ret = 0;
switch (params_rate(params)) {
case 8000:
case 16000:
case 48000:
case 96000:
clk = 12288000;
break;
case 11025:
case 22050:
case 44100:
clk = 11289600;
break;
}
/* set codec DAI configuration */
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;
/* set cpu DAI configuration */
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;
/* set the codec system clock for DAC and ADC */
ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
/* set the I2S system clock as input (unused) */
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
return 0;
}
static struct snd_soc_ops spitz_ops = {
.startup = spitz_startup,
.hw_params = spitz_hw_params,
};
static int spitz_get_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = spitz_jack_func;
return 0;
}
static int spitz_set_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
if (spitz_jack_func == ucontrol->value.integer.value[0])
return 0;
spitz_jack_func = ucontrol->value.integer.value[0];
spitz_ext_control(codec);
return 1;
}
static int spitz_get_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = spitz_spk_func;
return 0;
}
static int spitz_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
if (spitz_spk_func == ucontrol->value.integer.value[0])
return 0;
spitz_spk_func = ucontrol->value.integer.value[0];
spitz_ext_control(codec);
return 1;
}
static int spitz_mic_bias(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
if (machine_is_borzoi() || machine_is_spitz())
gpio_set_value(SPITZ_GPIO_MIC_BIAS,
SND_SOC_DAPM_EVENT_ON(event));
if (machine_is_akita())
gpio_set_value(AKITA_GPIO_MIC_BIAS,
SND_SOC_DAPM_EVENT_ON(event));
return 0;
}
/* spitz machine dapm widgets */
static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
SND_SOC_DAPM_LINE("Line Jack", NULL),
/* headset is a mic and mono headphone */
SND_SOC_DAPM_HP("Headset Jack", NULL),
};
/* Spitz machine audio_map */
static const struct snd_soc_dapm_route audio_map[] = {
/* headphone connected to LOUT1, ROUT1 */
{"Headphone Jack", NULL, "LOUT1"},
{"Headphone Jack", NULL, "ROUT1"},
/* headset connected to ROUT1 and LINPUT1 with bias (def below) */
{"Headset Jack", NULL, "ROUT1"},
/* ext speaker connected to LOUT2, ROUT2 */
{"Ext Spk", NULL , "ROUT2"},
{"Ext Spk", NULL , "LOUT2"},
/* mic is connected to input 1 - with bias */
{"LINPUT1", NULL, "Mic Bias"},
{"Mic Bias", NULL, "Mic Jack"},
/* line is connected to input 1 - no bias */
{"LINPUT1", NULL, "Line Jack"},
};
static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
"Off"};
static const char *spk_function[] = {"On", "Off"};
static const struct soc_enum spitz_enum[] = {
SOC_ENUM_SINGLE_EXT(5, jack_function),
SOC_ENUM_SINGLE_EXT(2, spk_function),
};
static const struct snd_kcontrol_new wm8750_spitz_controls[] = {
SOC_ENUM_EXT("Jack Function", spitz_enum[0], spitz_get_jack,
spitz_set_jack),
SOC_ENUM_EXT("Speaker Function", spitz_enum[1], spitz_get_spk,
spitz_set_spk),
};
/*
* Logic for a wm8750 as connected on a Sharp SL-Cxx00 Device
*/
static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
int err;
/* NC codec pins */
snd_soc_dapm_nc_pin(codec, "RINPUT1");
snd_soc_dapm_nc_pin(codec, "LINPUT2");
snd_soc_dapm_nc_pin(codec, "RINPUT2");
snd_soc_dapm_nc_pin(codec, "LINPUT3");
snd_soc_dapm_nc_pin(codec, "RINPUT3");
snd_soc_dapm_nc_pin(codec, "OUT3");
snd_soc_dapm_nc_pin(codec, "MONO1");
/* Add spitz specific controls */
err = snd_soc_add_controls(codec, wm8750_spitz_controls,
ARRAY_SIZE(wm8750_spitz_controls));
if (err < 0)
return err;
/* Add spitz specific widgets */
snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
ARRAY_SIZE(wm8750_dapm_widgets));
/* Set up spitz specific audio paths */
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
snd_soc_dapm_sync(codec);
return 0;
}
/* spitz digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link spitz_dai = {
.name = "wm8750",
.stream_name = "WM8750",
.cpu_dai_name = "pxa-is2",
.codec_dai_name = "wm8750-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm8750-codec.0-001a",
.init = spitz_wm8750_init,
.ops = &spitz_ops,
};
/* spitz audio machine driver */
static struct snd_soc_card snd_soc_spitz = {
.name = "Spitz",
.dai_link = &spitz_dai,
.num_links = 1,
};
static struct platform_device *spitz_snd_device;
static int __init spitz_init(void)
{
int ret;
if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita()))
return -ENODEV;
spitz_snd_device = platform_device_alloc("soc-audio", -1);
if (!spitz_snd_device)
return -ENOMEM;
platform_set_drvdata(spitz_snd_device, &snd_soc_spitz);
ret = platform_device_add(spitz_snd_device);
if (ret)
platform_device_put(spitz_snd_device);
return ret;
}
static void __exit spitz_exit(void)
{
platform_device_unregister(spitz_snd_device);
}
module_init(spitz_init);
module_exit(spitz_exit);
MODULE_AUTHOR("Richard Purdie");
MODULE_DESCRIPTION("ALSA SoC Spitz");
MODULE_LICENSE("GPL");
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