OpenCloudOS-Kernel/sound/soc/samsung/rx1950_uda1380.c

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/*
* rx1950.c -- ALSA Soc Audio Layer
*
* Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com>
*
* Based on smdk2440.c and magician.c
*
* Authors: Graeme Gregory graeme.gregory@wolfsonmicro.com
* Philipp Zabel <philipp.zabel@gmail.com>
* Denis Grigoriev <dgreenday@gmail.com>
* Vasily Khoruzhick <anarsoul@gmail.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/platform_device.h>
#include <linux/i2c.h>
#include <linux/gpio.h>
#include <linux/clk.h>
#include <sound/soc.h>
#include <sound/uda1380.h>
#include <sound/jack.h>
#include <plat/regs-iis.h>
#include <mach/regs-clock.h>
#include <asm/mach-types.h>
#include "dma.h"
#include "s3c24xx-i2s.h"
#include "../codecs/uda1380.h"
static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd);
static int rx1950_startup(struct snd_pcm_substream *substream);
static int rx1950_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params);
static int rx1950_spk_power(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event);
static unsigned int rates[] = {
16000,
44100,
48000,
};
static struct snd_pcm_hw_constraint_list hw_rates = {
.count = ARRAY_SIZE(rates),
.list = rates,
.mask = 0,
};
static struct snd_soc_jack hp_jack;
static struct snd_soc_jack_pin hp_jack_pins[] = {
{
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
},
{
.pin = "Speaker",
.mask = SND_JACK_HEADPHONE,
.invert = 1,
},
};
static struct snd_soc_jack_gpio hp_jack_gpios[] = {
[0] = {
.gpio = S3C2410_GPG(12),
.name = "hp-gpio",
.report = SND_JACK_HEADPHONE,
.invert = 1,
.debounce_time = 200,
},
};
static struct snd_soc_ops rx1950_ops = {
.startup = rx1950_startup,
.hw_params = rx1950_hw_params,
};
/* s3c24xx digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link rx1950_uda1380_dai[] = {
{
.name = "uda1380",
.stream_name = "UDA1380 Duplex",
.cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "uda1380-hifi",
.init = rx1950_uda1380_init,
.platform_name = "samsung-audio",
.codec_name = "uda1380-codec.0-001a",
.ops = &rx1950_ops,
},
};
static struct snd_soc_card rx1950_asoc = {
.name = "rx1950",
.dai_link = rx1950_uda1380_dai,
.num_links = ARRAY_SIZE(rx1950_uda1380_dai),
};
/* rx1950 machine dapm widgets */
static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Mic Jack", NULL),
SND_SOC_DAPM_SPK("Speaker", rx1950_spk_power),
};
/* rx1950 machine audio_map */
static const struct snd_soc_dapm_route audio_map[] = {
/* headphone connected to VOUTLHP, VOUTRHP */
{"Headphone Jack", NULL, "VOUTLHP"},
{"Headphone Jack", NULL, "VOUTRHP"},
/* ext speaker connected to VOUTL, VOUTR */
{"Speaker", NULL, "VOUTL"},
{"Speaker", NULL, "VOUTR"},
/* mic is connected to VINM */
{"VINM", NULL, "Mic Jack"},
};
static struct platform_device *s3c24xx_snd_device;
static int rx1950_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
runtime->hw.rate_min = hw_rates.list[0];
runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
return snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&hw_rates);
}
static int rx1950_spk_power(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
if (SND_SOC_DAPM_EVENT_ON(event))
gpio_set_value(S3C2410_GPA(1), 1);
else
gpio_set_value(S3C2410_GPA(1), 0);
return 0;
}
static int rx1950_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 *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int div;
int ret;
unsigned int rate = params_rate(params);
int clk_source, fs_mode;
switch (rate) {
case 16000:
case 48000:
clk_source = S3C24XX_CLKSRC_PCLK;
fs_mode = S3C2410_IISMOD_256FS;
div = s3c24xx_i2s_get_clockrate() / (256 * rate);
if (s3c24xx_i2s_get_clockrate() % (256 * rate) > (128 * rate))
div++;
break;
case 44100:
case 88200:
clk_source = S3C24XX_CLKSRC_MPLL;
fs_mode = S3C2410_IISMOD_384FS;
div = 1;
break;
default:
printk(KERN_ERR "%s: rate %d is not supported\n",
__func__, rate);
return -EINVAL;
}
/* 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;
/* select clock source */
ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source, rate,
SND_SOC_CLOCK_OUT);
if (ret < 0)
return ret;
/* set MCLK division for sample rate */
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
fs_mode);
if (ret < 0)
return ret;
/* set BCLK division for sample rate */
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
S3C2410_IISMOD_32FS);
if (ret < 0)
return ret;
/* set prescaler division for sample rate */
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
S3C24XX_PRESCALE(div, div));
if (ret < 0)
return ret;
return 0;
}
static int rx1950_uda1380_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 21:53:46 +08:00
struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
/* Add rx1950 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 21:53:46 +08:00
err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
ARRAY_SIZE(uda1380_dapm_widgets));
if (err)
return err;
/* Set up rx1950 specific audio path audio_mapnects */
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 21:53:46 +08:00
err = snd_soc_dapm_add_routes(dapm, audio_map,
ARRAY_SIZE(audio_map));
if (err)
return err;
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 21:53:46 +08:00
snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
snd_soc_dapm_enable_pin(dapm, "Speaker");
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 21:53:46 +08:00
snd_soc_dapm_sync(dapm);
snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
&hp_jack);
snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
hp_jack_pins);
snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
hp_jack_gpios);
return 0;
}
static int __init rx1950_init(void)
{
int ret;
if (!machine_is_rx1950())
return -ENODEV;
/* configure some gpios */
ret = gpio_request(S3C2410_GPA(1), "speaker-power");
if (ret)
goto err_gpio;
ret = gpio_direction_output(S3C2410_GPA(1), 0);
if (ret)
goto err_gpio_conf;
s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
if (!s3c24xx_snd_device) {
ret = -ENOMEM;
goto err_plat_alloc;
}
platform_set_drvdata(s3c24xx_snd_device, &rx1950_asoc);
ret = platform_device_add(s3c24xx_snd_device);
if (ret) {
platform_device_put(s3c24xx_snd_device);
goto err_plat_add;
}
return 0;
err_plat_add:
err_plat_alloc:
err_gpio_conf:
gpio_free(S3C2410_GPA(1));
err_gpio:
return ret;
}
static void __exit rx1950_exit(void)
{
platform_device_unregister(s3c24xx_snd_device);
snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
hp_jack_gpios);
gpio_free(S3C2410_GPA(1));
}
module_init(rx1950_init);
module_exit(rx1950_exit);
/* Module information */
MODULE_AUTHOR("Vasily Khoruzhick");
MODULE_DESCRIPTION("ALSA SoC RX1950");
MODULE_LICENSE("GPL");