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linux-next/sound/soc/pxa/tavorevb3.c

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/*
* tavorevb3.c -- SoC audio for Tavor EVB3
*
* Copyright (C) 2010 Marvell International Ltd.
* Haojian Zhuang <haojian.zhuang@marvell.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.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/device.h>
#include <linux/clk.h>
#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include <asm/mach-types.h>
#include "../codecs/88pm860x-codec.h"
#include "pxa-ssp.h"
static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd);
static struct platform_device *evb3_snd_device;
static struct snd_soc_jack hs_jack, mic_jack;
static struct snd_soc_jack_pin hs_jack_pins[] = {
{ .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, },
};
static struct snd_soc_jack_pin mic_jack_pins[] = {
{ .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, },
};
/* tavorevb3 machine dapm widgets */
static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headset Stereophone", NULL),
SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
SND_SOC_DAPM_SPK("Ext Speaker", NULL),
SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
};
/* tavorevb3 machine audio map */
static const struct snd_soc_dapm_route audio_map[] = {
{"Headset Stereophone", NULL, "HS1"},
{"Headset Stereophone", NULL, "HS2"},
{"Ext Speaker", NULL, "LSP"},
{"Ext Speaker", NULL, "LSN"},
{"Lineout Out 1", NULL, "LINEOUT1"},
{"Lineout Out 2", NULL, "LINEOUT2"},
{"MIC1P", NULL, "Mic1 Bias"},
{"MIC1N", NULL, "Mic1 Bias"},
{"Mic1 Bias", NULL, "Ext Mic 1"},
{"MIC2P", NULL, "Mic1 Bias"},
{"MIC2N", NULL, "Mic1 Bias"},
{"Mic1 Bias", NULL, "Headset Mic 2"},
{"MIC3P", NULL, "Mic3 Bias"},
{"MIC3N", NULL, "Mic3 Bias"},
{"Mic3 Bias", NULL, "Ext Mic 3"},
};
static int evb3_i2s_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 width = snd_pcm_format_physical_width(params_format(params));
int ret;
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
PM860X_CLK_DIR_OUT);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
if (ret < 0)
return ret;
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;
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;
ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
return ret;
}
static struct snd_soc_ops evb3_i2s_ops = {
.hw_params = evb3_i2s_hw_params,
};
static struct snd_soc_dai_link evb3_dai[] = {
{
.name = "88PM860x I2S",
.stream_name = "I2S Audio",
.cpu_dai_name = "pxa-ssp-dai.1",
.codec_dai_name = "88pm860x-i2s",
.platform_name = "pxa-pcm-audio",
.codec_name = "88pm860x-codec",
.init = evb3_pm860x_init,
.ops = &evb3_i2s_ops,
},
};
static struct snd_soc_card snd_soc_card_evb3 = {
.name = "Tavor EVB3",
.dai_link = evb3_dai,
.num_links = ARRAY_SIZE(evb3_dai),
};
static int evb3_pm860x_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 ret;
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_new_controls(dapm, evb3_dapm_widgets,
ARRAY_SIZE(evb3_dapm_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
snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
/* connected pins */
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, "Ext Speaker");
snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
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
ret = snd_soc_dapm_sync(dapm);
if (ret)
return ret;
/* Headset jack detection */
snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
&hs_jack);
snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
hs_jack_pins);
snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
&mic_jack);
snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
mic_jack_pins);
/* headphone, microphone detection & headset short detection */
pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
return 0;
}
static int __init tavorevb3_init(void)
{
int ret;
if (!machine_is_tavorevb3())
return -ENODEV;
evb3_snd_device = platform_device_alloc("soc-audio", -1);
if (!evb3_snd_device)
return -ENOMEM;
platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3);
ret = platform_device_add(evb3_snd_device);
if (ret)
platform_device_put(evb3_snd_device);
return ret;
}
static void __exit tavorevb3_exit(void)
{
platform_device_unregister(evb3_snd_device);
}
module_init(tavorevb3_init);
module_exit(tavorevb3_exit);
MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3");
MODULE_LICENSE("GPL");