android_kernel_xiaomi_sm8350/sound/soc/omap/zoom2.c
Liam Girdwood ce6120cca2 ASoC: Decouple DAPM from CODECs
Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.

This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.

This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.

Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.

Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Manuel Lauss <manuel.lauss@googlemail.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Cc: Kevin Hilman <khilman@deeprootsystems.com>
Cc: Ryan Mallon <ryan@bluewatersys.com>
Cc: Timur Tabi <timur@freescale.com>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org>
Cc: Wan ZongShun <mcuos.com@gmail.com>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Jassi Brar <jassi.brar@samsung.com>
Cc: Daniel Gloeckner <dg@emlix.com>
Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-06 11:28:29 -04:00

293 lines
7.6 KiB
C

/*
* zoom2.c -- SoC audio for Zoom2
*
* Author: Misael Lopez Cruz <x0052729@ti.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/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 <mach/gpio.h>
#include <mach/board-zoom.h>
#include <plat/mcbsp.h>
/* Register descriptions for twl4030 codec part */
#include <linux/mfd/twl4030-codec.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
#define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15)
static int zoom2_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 ret;
/* Set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai,
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0) {
printk(KERN_ERR "can't set codec DAI configuration\n");
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_CBM_CFM);
if (ret < 0) {
printk(KERN_ERR "can't set cpu DAI configuration\n");
return ret;
}
/* Set the codec system clock for DAC and ADC */
ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
SND_SOC_CLOCK_IN);
if (ret < 0) {
printk(KERN_ERR "can't set codec system clock\n");
return ret;
}
return 0;
}
static struct snd_soc_ops zoom2_ops = {
.hw_params = zoom2_hw_params,
};
static int zoom2_hw_voice_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 ret;
/* Set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai,
SND_SOC_DAIFMT_DSP_A |
SND_SOC_DAIFMT_IB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (ret) {
printk(KERN_ERR "can't set codec DAI configuration\n");
return ret;
}
/* Set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai,
SND_SOC_DAIFMT_DSP_A |
SND_SOC_DAIFMT_IB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0) {
printk(KERN_ERR "can't set cpu DAI configuration\n");
return ret;
}
/* Set the codec system clock for DAC and ADC */
ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
SND_SOC_CLOCK_IN);
if (ret < 0) {
printk(KERN_ERR "can't set codec system clock\n");
return ret;
}
return 0;
}
static struct snd_soc_ops zoom2_voice_ops = {
.hw_params = zoom2_hw_voice_params,
};
/* Zoom2 machine DAPM */
static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = {
SND_SOC_DAPM_MIC("Ext Mic", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_HP("Headset Stereophone", NULL),
SND_SOC_DAPM_LINE("Aux In", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
/* External Mics: MAINMIC, SUBMIC with bias*/
{"MAINMIC", NULL, "Mic Bias 1"},
{"SUBMIC", NULL, "Mic Bias 2"},
{"Mic Bias 1", NULL, "Ext Mic"},
{"Mic Bias 2", NULL, "Ext Mic"},
/* External Speakers: HFL, HFR */
{"Ext Spk", NULL, "HFL"},
{"Ext Spk", NULL, "HFR"},
/* Headset Stereophone: HSOL, HSOR */
{"Headset Stereophone", NULL, "HSOL"},
{"Headset Stereophone", NULL, "HSOR"},
/* Headset Mic: HSMIC with bias */
{"HSMIC", NULL, "Headset Mic Bias"},
{"Headset Mic Bias", NULL, "Headset Mic"},
/* Aux In: AUXL, AUXR */
{"Aux In", NULL, "AUXL"},
{"Aux In", NULL, "AUXR"},
};
static int zoom2_twl4030_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
/* Add Zoom2 specific widgets */
ret = snd_soc_dapm_new_controls(dapm, zoom2_twl4030_dapm_widgets,
ARRAY_SIZE(zoom2_twl4030_dapm_widgets));
if (ret)
return ret;
/* Set up Zoom2 specific audio path audio_map */
snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
/* Zoom2 connected pins */
snd_soc_dapm_enable_pin(dapm, "Ext Mic");
snd_soc_dapm_enable_pin(dapm, "Ext Spk");
snd_soc_dapm_enable_pin(dapm, "Headset Mic");
snd_soc_dapm_enable_pin(dapm, "Headset Stereophone");
snd_soc_dapm_enable_pin(dapm, "Aux In");
/* TWL4030 not connected pins */
snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
snd_soc_dapm_nc_pin(dapm, "EARPIECE");
snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
snd_soc_dapm_nc_pin(dapm, "CARKITL");
snd_soc_dapm_nc_pin(dapm, "CARKITR");
ret = snd_soc_dapm_sync(dapm);
return ret;
}
static int zoom2_twl4030_voice_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
unsigned short reg;
/* Enable voice interface */
reg = codec->driver->read(codec, TWL4030_REG_VOICE_IF);
reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
codec->driver->write(codec, TWL4030_REG_VOICE_IF, reg);
return 0;
}
/* Digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link zoom2_dai[] = {
{
.name = "TWL4030 I2S",
.stream_name = "TWL4030 Audio",
.cpu_dai_name = "omap-mcbsp-dai.1",
.codec_dai_name = "twl4030-hifi",
.platform_name = "omap-pcm-audio",
.codec_name = "twl4030-codec",
.init = zoom2_twl4030_init,
.ops = &zoom2_ops,
},
{
.name = "TWL4030 PCM",
.stream_name = "TWL4030 Voice",
.cpu_dai_name = "omap-mcbsp-dai.2",
.codec_dai_name = "twl4030-voice",
.platform_name = "omap-pcm-audio",
.codec_name = "twl4030-codec",
.init = zoom2_twl4030_voice_init,
.ops = &zoom2_voice_ops,
},
};
/* Audio machine driver */
static struct snd_soc_card snd_soc_zoom2 = {
.name = "Zoom2",
.dai_link = zoom2_dai,
.num_links = ARRAY_SIZE(zoom2_dai),
};
static struct platform_device *zoom2_snd_device;
static int __init zoom2_soc_init(void)
{
int ret;
if (!machine_is_omap_zoom2())
return -ENODEV;
printk(KERN_INFO "Zoom2 SoC init\n");
zoom2_snd_device = platform_device_alloc("soc-audio", -1);
if (!zoom2_snd_device) {
printk(KERN_ERR "Platform device allocation failed\n");
return -ENOMEM;
}
platform_set_drvdata(zoom2_snd_device, &snd_soc_zoom2);
ret = platform_device_add(zoom2_snd_device);
if (ret)
goto err1;
BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0);
gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0);
BUG_ON(gpio_request(ZOOM2_HEADSET_EXTMUTE_GPIO, "ext_mute") < 0);
gpio_direction_output(ZOOM2_HEADSET_EXTMUTE_GPIO, 0);
return 0;
err1:
printk(KERN_ERR "Unable to add platform device\n");
platform_device_put(zoom2_snd_device);
return ret;
}
module_init(zoom2_soc_init);
static void __exit zoom2_soc_exit(void)
{
gpio_free(ZOOM2_HEADSET_MUX_GPIO);
gpio_free(ZOOM2_HEADSET_EXTMUTE_GPIO);
platform_device_unregister(zoom2_snd_device);
}
module_exit(zoom2_soc_exit);
MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
MODULE_DESCRIPTION("ALSA SoC Zoom2");
MODULE_LICENSE("GPL");