linux/sound/soc/omap/sdp4430.c
Peter Ujfalusi 6524c8e3e6 ASoC: sdp4430: Add support for digital microphones
OMAP4 SDP/Blaze boards have digital microphones.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-28 17:21:45 +00:00

279 lines
7.1 KiB
C

/*
* sdp4430.c -- SoC audio for TI OMAP4430 SDP
*
* 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 <linux/mfd/twl6040.h>
#include <linux/module.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include <asm/mach-types.h>
#include <plat/hardware.h>
#include <plat/mux.h>
#include "omap-dmic.h"
#include "omap-mcpdm.h"
#include "omap-pcm.h"
#include "../codecs/twl6040.h"
static int sdp4430_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;
int clk_id, freq;
int ret;
clk_id = twl6040_get_clk_id(rtd->codec);
if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
freq = 38400000;
else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
freq = 32768;
else
return -EINVAL;
/* set the codec mclk */
ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
SND_SOC_CLOCK_IN);
if (ret) {
printk(KERN_ERR "can't set codec system clock\n");
return ret;
}
return ret;
}
static struct snd_soc_ops sdp4430_ops = {
.hw_params = sdp4430_hw_params,
};
static int sdp4430_dmic_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;
int ret = 0;
ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
19200000, SND_SOC_CLOCK_IN);
if (ret < 0) {
printk(KERN_ERR "can't set DMIC cpu system clock\n");
return ret;
}
ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
SND_SOC_CLOCK_OUT);
if (ret < 0) {
printk(KERN_ERR "can't set DMIC output clock\n");
return ret;
}
return 0;
}
static struct snd_soc_ops sdp4430_dmic_ops = {
.hw_params = sdp4430_dmic_hw_params,
};
/* Headset jack */
static struct snd_soc_jack hs_jack;
/*Headset jack detection DAPM pins */
static struct snd_soc_jack_pin hs_jack_pins[] = {
{
.pin = "Headset Mic",
.mask = SND_JACK_MICROPHONE,
},
{
.pin = "Headset Stereophone",
.mask = SND_JACK_HEADPHONE,
},
};
/* SDP4430 machine DAPM */
static const struct snd_soc_dapm_widget sdp4430_twl6040_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_SPK("Earphone Spk", NULL),
SND_SOC_DAPM_INPUT("FM Stereo In"),
};
static const struct snd_soc_dapm_route audio_map[] = {
/* External Mics: MAINMIC, SUBMIC with bias*/
{"MAINMIC", NULL, "Main Mic Bias"},
{"SUBMIC", NULL, "Main Mic Bias"},
{"Main Mic Bias", NULL, "Ext Mic"},
/* External Speakers: HFL, HFR */
{"Ext Spk", NULL, "HFL"},
{"Ext Spk", NULL, "HFR"},
/* Headset Mic: HSMIC with bias */
{"HSMIC", NULL, "Headset Mic Bias"},
{"Headset Mic Bias", NULL, "Headset Mic"},
/* Headset Stereophone (Headphone): HSOL, HSOR */
{"Headset Stereophone", NULL, "HSOL"},
{"Headset Stereophone", NULL, "HSOR"},
/* Earphone speaker */
{"Earphone Spk", NULL, "EP"},
/* Aux/FM Stereo In: AFML, AFMR */
{"AFML", NULL, "FM Stereo In"},
{"AFMR", NULL, "FM Stereo In"},
};
static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
int ret, hs_trim;
/*
* Configure McPDM offset cancellation based on the HSOTRIM value from
* twl6040.
*/
hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM);
omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
TWL6040_HSF_TRIM_RIGHT(hs_trim));
/* Headset jack detection */
ret = snd_soc_jack_new(codec, "Headset Jack",
SND_JACK_HEADSET, &hs_jack);
if (ret)
return ret;
ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
hs_jack_pins);
if (machine_is_omap_4430sdp())
twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
else
snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET);
return ret;
}
static const struct snd_soc_dapm_widget sdp4430_dmic_dapm_widgets[] = {
SND_SOC_DAPM_MIC("Digital Mic", NULL),
};
static const struct snd_soc_dapm_route dmic_audio_map[] = {
{"DMic", NULL, "Digital Mic1 Bias"},
{"Digital Mic1 Bias", NULL, "Digital Mic"},
};
static int sdp4430_dmic_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
ret = snd_soc_dapm_new_controls(dapm, sdp4430_dmic_dapm_widgets,
ARRAY_SIZE(sdp4430_dmic_dapm_widgets));
if (ret)
return ret;
return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
ARRAY_SIZE(dmic_audio_map));
}
/* Digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link sdp4430_dai[] = {
{
.name = "TWL6040",
.stream_name = "TWL6040",
.cpu_dai_name = "omap-mcpdm",
.codec_dai_name = "twl6040-legacy",
.platform_name = "omap-pcm-audio",
.codec_name = "twl6040-codec",
.init = sdp4430_twl6040_init,
.ops = &sdp4430_ops,
},
{
.name = "DMIC",
.stream_name = "DMIC Capture",
.cpu_dai_name = "omap-dmic",
.codec_dai_name = "dmic-hifi",
.platform_name = "omap-pcm-audio",
.codec_name = "dmic-codec",
.init = sdp4430_dmic_init,
.ops = &sdp4430_dmic_ops,
},
};
/* Audio machine driver */
static struct snd_soc_card snd_soc_sdp4430 = {
.name = "SDP4430",
.dai_link = sdp4430_dai,
.num_links = ARRAY_SIZE(sdp4430_dai),
.dapm_widgets = sdp4430_twl6040_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(sdp4430_twl6040_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct platform_device *sdp4430_snd_device;
static int __init sdp4430_soc_init(void)
{
int ret;
if (!machine_is_omap_4430sdp())
return -ENODEV;
printk(KERN_INFO "SDP4430 SoC init\n");
sdp4430_snd_device = platform_device_alloc("soc-audio", -1);
if (!sdp4430_snd_device) {
printk(KERN_ERR "Platform device allocation failed\n");
return -ENOMEM;
}
platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430);
ret = platform_device_add(sdp4430_snd_device);
if (ret)
goto err;
return 0;
err:
printk(KERN_ERR "Unable to add platform device\n");
platform_device_put(sdp4430_snd_device);
return ret;
}
module_init(sdp4430_soc_init);
static void __exit sdp4430_soc_exit(void)
{
platform_device_unregister(sdp4430_snd_device);
}
module_exit(sdp4430_soc_exit);
MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
MODULE_DESCRIPTION("ALSA SoC SDP4430");
MODULE_LICENSE("GPL");