kernel-fxtec-pro1x/sound/soc/samsung/goni_wm8994.c

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/*
* goni_wm8994.c
*
* Copyright (C) 2010 Samsung Electronics Co.Ltd
* Author: Chanwoo Choi <cw00.choi@samsung.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 <sound/soc.h>
#include <sound/jack.h>
#include <asm/mach-types.h>
#include <mach/gpio.h>
#include "../codecs/wm8994.h"
#define MACHINE_NAME 0
#define CPU_VOICE_DAI 1
static const char *aquila_str[] = {
[MACHINE_NAME] = "aquila",
[CPU_VOICE_DAI] = "aquila-voice-dai",
};
static struct snd_soc_card goni;
static struct platform_device *goni_snd_device;
/* 3.5 pie jack */
static struct snd_soc_jack jack;
/* 3.5 pie jack detection DAPM pins */
static struct snd_soc_jack_pin jack_pins[] = {
{
.pin = "Headset Mic",
.mask = SND_JACK_MICROPHONE,
}, {
.pin = "Headset Stereophone",
.mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL |
SND_JACK_AVOUT,
},
};
/* 3.5 pie jack detection gpios */
static struct snd_soc_jack_gpio jack_gpios[] = {
{
.gpio = S5PV210_GPH0(6),
.name = "DET_3.5",
.report = SND_JACK_HEADSET | SND_JACK_MECHANICAL |
SND_JACK_AVOUT,
.debounce_time = 200,
},
};
static const struct snd_soc_dapm_widget goni_dapm_widgets[] = {
SND_SOC_DAPM_SPK("Ext Left Spk", NULL),
SND_SOC_DAPM_SPK("Ext Right Spk", NULL),
SND_SOC_DAPM_SPK("Ext Rcv", NULL),
SND_SOC_DAPM_HP("Headset Stereophone", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("Main Mic", NULL),
SND_SOC_DAPM_MIC("2nd Mic", NULL),
SND_SOC_DAPM_LINE("Radio In", NULL),
};
static const struct snd_soc_dapm_route goni_dapm_routes[] = {
{"Ext Left Spk", NULL, "SPKOUTLP"},
{"Ext Left Spk", NULL, "SPKOUTLN"},
{"Ext Right Spk", NULL, "SPKOUTRP"},
{"Ext Right Spk", NULL, "SPKOUTRN"},
{"Ext Rcv", NULL, "HPOUT2N"},
{"Ext Rcv", NULL, "HPOUT2P"},
{"Headset Stereophone", NULL, "HPOUT1L"},
{"Headset Stereophone", NULL, "HPOUT1R"},
{"IN1RN", NULL, "Headset Mic"},
{"IN1RP", NULL, "Headset Mic"},
{"IN1RN", NULL, "2nd Mic"},
{"IN1RP", NULL, "2nd Mic"},
{"IN1LN", NULL, "Main Mic"},
{"IN1LP", NULL, "Main Mic"},
{"IN2LN", NULL, "Radio In"},
{"IN2RN", NULL, "Radio In"},
};
static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
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>
2010-11-05 07:53:46 -06:00
struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
/* add goni specific widgets */
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>
2010-11-05 07:53:46 -06:00
snd_soc_dapm_new_controls(dapm, goni_dapm_widgets,
ARRAY_SIZE(goni_dapm_widgets));
/* set up goni specific audio routes */
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>
2010-11-05 07:53:46 -06:00
snd_soc_dapm_add_routes(dapm, goni_dapm_routes,
ARRAY_SIZE(goni_dapm_routes));
/* set endpoints to not connected */
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>
2010-11-05 07:53:46 -06:00
snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
if (machine_is_aquila()) {
snd_soc_dapm_nc_pin(dapm, "SPKOUTRN");
snd_soc_dapm_nc_pin(dapm, "SPKOUTRP");
}
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>
2010-11-05 07:53:46 -06:00
snd_soc_dapm_sync(dapm);
/* Headset jack detection */
ret = snd_soc_jack_new(codec, "Headset Jack",
SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT,
&jack);
if (ret)
return ret;
ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins);
if (ret)
return ret;
ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios);
if (ret)
return ret;
return 0;
}
static int goni_hifi_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 pll_out = 24000000;
int ret = 0;
/* set the 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)
return 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)
return ret;
/* set the codec FLL */
ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out,
params_rate(params) * 256);
if (ret < 0)
return ret;
/* set the codec system clock */
ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1,
params_rate(params) * 256, SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
return 0;
}
static struct snd_soc_ops goni_hifi_ops = {
.hw_params = goni_hifi_hw_params,
};
static int goni_voice_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;
unsigned int pll_out = 24000000;
int ret = 0;
if (params_rate(params) != 8000)
return -EINVAL;
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J |
SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0)
return ret;
/* set the codec FLL */
ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out,
params_rate(params) * 256);
if (ret < 0)
return ret;
/* set the codec system clock */
ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2,
params_rate(params) * 256, SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
return 0;
}
static struct snd_soc_dai_driver voice_dai = {
.name = "goni-voice-dai",
.id = 0,
.playback = {
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.capture = {
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
};
static struct snd_soc_ops goni_voice_ops = {
.hw_params = goni_voice_hw_params,
};
static struct snd_soc_dai_link goni_dai[] = {
{
.name = "WM8994",
.stream_name = "WM8994 HiFi",
.cpu_dai_name = "samsung-i2s.0",
.codec_dai_name = "wm8994-aif1",
.platform_name = "samsung-audio",
.codec_name = "wm8994-codec.0-001a",
.init = goni_wm8994_init,
.ops = &goni_hifi_ops,
}, {
.name = "WM8994 Voice",
.stream_name = "Voice",
.cpu_dai_name = "goni-voice-dai",
.codec_dai_name = "wm8994-aif2",
.codec_name = "wm8994-codec.0-001a",
.ops = &goni_voice_ops,
},
};
static struct snd_soc_card goni = {
.name = "goni",
.dai_link = goni_dai,
.num_links = ARRAY_SIZE(goni_dai),
};
static int __init goni_init(void)
{
int ret;
if (machine_is_aquila()) {
voice_dai.name = aquila_str[CPU_VOICE_DAI];
goni_dai[1].cpu_dai_name = aquila_str[CPU_VOICE_DAI];
goni.name = aquila_str[MACHINE_NAME];
} else if (!machine_is_goni())
return -ENODEV;
goni_snd_device = platform_device_alloc("soc-audio", -1);
if (!goni_snd_device)
return -ENOMEM;
/* register voice DAI here */
ret = snd_soc_register_dai(&goni_snd_device->dev, &voice_dai);
if (ret) {
platform_device_put(goni_snd_device);
return ret;
}
platform_set_drvdata(goni_snd_device, &goni);
ret = platform_device_add(goni_snd_device);
if (ret) {
snd_soc_unregister_dai(&goni_snd_device->dev);
platform_device_put(goni_snd_device);
}
return ret;
}
static void __exit goni_exit(void)
{
snd_soc_unregister_dai(&goni_snd_device->dev);
platform_device_unregister(goni_snd_device);
}
module_init(goni_init);
module_exit(goni_exit);
/* Module information */
MODULE_DESCRIPTION("ALSA SoC WM8994 GONI(S5PV210)");
MODULE_AUTHOR("Chanwoo Choi <cw00.choi@samsung.com>");
MODULE_LICENSE("GPL");