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6524c8e3e6
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>
278 lines
7.1 KiB
C
278 lines
7.1 KiB
C
/*
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* sdp4430.c -- SoC audio for TI OMAP4430 SDP
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*
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* Author: Misael Lopez Cruz <x0052729@ti.com>
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* version 2 as published by the Free Software Foundation.
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*
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* This program is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
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* 02110-1301 USA
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*
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*/
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#include <linux/clk.h>
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#include <linux/platform_device.h>
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#include <linux/mfd/twl6040.h>
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#include <linux/module.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/soc.h>
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#include <sound/jack.h>
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#include <asm/mach-types.h>
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#include <plat/hardware.h>
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#include <plat/mux.h>
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#include "omap-dmic.h"
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#include "omap-mcpdm.h"
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#include "omap-pcm.h"
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#include "../codecs/twl6040.h"
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static int sdp4430_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_dai *codec_dai = rtd->codec_dai;
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int clk_id, freq;
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int ret;
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clk_id = twl6040_get_clk_id(rtd->codec);
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if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
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freq = 38400000;
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else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
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freq = 32768;
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else
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return -EINVAL;
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/* set the codec mclk */
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ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
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SND_SOC_CLOCK_IN);
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if (ret) {
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printk(KERN_ERR "can't set codec system clock\n");
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return ret;
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}
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return ret;
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}
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static struct snd_soc_ops sdp4430_ops = {
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.hw_params = sdp4430_hw_params,
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};
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static int sdp4430_dmic_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
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int ret = 0;
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ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
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19200000, SND_SOC_CLOCK_IN);
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if (ret < 0) {
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printk(KERN_ERR "can't set DMIC cpu system clock\n");
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return ret;
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}
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ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
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SND_SOC_CLOCK_OUT);
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if (ret < 0) {
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printk(KERN_ERR "can't set DMIC output clock\n");
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return ret;
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}
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return 0;
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}
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static struct snd_soc_ops sdp4430_dmic_ops = {
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.hw_params = sdp4430_dmic_hw_params,
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};
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/* Headset jack */
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static struct snd_soc_jack hs_jack;
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/*Headset jack detection DAPM pins */
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static struct snd_soc_jack_pin hs_jack_pins[] = {
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{
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.pin = "Headset Mic",
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.mask = SND_JACK_MICROPHONE,
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},
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{
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.pin = "Headset Stereophone",
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.mask = SND_JACK_HEADPHONE,
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},
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};
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/* SDP4430 machine DAPM */
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static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = {
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SND_SOC_DAPM_MIC("Ext Mic", NULL),
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SND_SOC_DAPM_SPK("Ext Spk", NULL),
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SND_SOC_DAPM_MIC("Headset Mic", NULL),
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SND_SOC_DAPM_HP("Headset Stereophone", NULL),
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SND_SOC_DAPM_SPK("Earphone Spk", NULL),
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SND_SOC_DAPM_INPUT("FM Stereo In"),
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};
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static const struct snd_soc_dapm_route audio_map[] = {
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/* External Mics: MAINMIC, SUBMIC with bias*/
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{"MAINMIC", NULL, "Main Mic Bias"},
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{"SUBMIC", NULL, "Main Mic Bias"},
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{"Main Mic Bias", NULL, "Ext Mic"},
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/* External Speakers: HFL, HFR */
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{"Ext Spk", NULL, "HFL"},
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{"Ext Spk", NULL, "HFR"},
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/* Headset Mic: HSMIC with bias */
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{"HSMIC", NULL, "Headset Mic Bias"},
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{"Headset Mic Bias", NULL, "Headset Mic"},
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/* Headset Stereophone (Headphone): HSOL, HSOR */
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{"Headset Stereophone", NULL, "HSOL"},
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{"Headset Stereophone", NULL, "HSOR"},
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/* Earphone speaker */
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{"Earphone Spk", NULL, "EP"},
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/* Aux/FM Stereo In: AFML, AFMR */
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{"AFML", NULL, "FM Stereo In"},
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{"AFMR", NULL, "FM Stereo In"},
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};
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static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd)
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{
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struct snd_soc_codec *codec = rtd->codec;
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int ret, hs_trim;
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/*
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* Configure McPDM offset cancellation based on the HSOTRIM value from
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* twl6040.
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*/
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hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM);
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omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
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TWL6040_HSF_TRIM_RIGHT(hs_trim));
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/* Headset jack detection */
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ret = snd_soc_jack_new(codec, "Headset Jack",
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SND_JACK_HEADSET, &hs_jack);
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if (ret)
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return ret;
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ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
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hs_jack_pins);
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if (machine_is_omap_4430sdp())
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twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
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else
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snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET);
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return ret;
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}
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static const struct snd_soc_dapm_widget sdp4430_dmic_dapm_widgets[] = {
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SND_SOC_DAPM_MIC("Digital Mic", NULL),
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};
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static const struct snd_soc_dapm_route dmic_audio_map[] = {
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{"DMic", NULL, "Digital Mic1 Bias"},
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{"Digital Mic1 Bias", NULL, "Digital Mic"},
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};
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static int sdp4430_dmic_init(struct snd_soc_pcm_runtime *rtd)
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{
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struct snd_soc_codec *codec = rtd->codec;
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struct snd_soc_dapm_context *dapm = &codec->dapm;
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int ret;
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ret = snd_soc_dapm_new_controls(dapm, sdp4430_dmic_dapm_widgets,
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ARRAY_SIZE(sdp4430_dmic_dapm_widgets));
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if (ret)
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return ret;
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return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
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ARRAY_SIZE(dmic_audio_map));
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}
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/* Digital audio interface glue - connects codec <--> CPU */
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static struct snd_soc_dai_link sdp4430_dai[] = {
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{
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.name = "TWL6040",
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.stream_name = "TWL6040",
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.cpu_dai_name = "omap-mcpdm",
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.codec_dai_name = "twl6040-legacy",
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.platform_name = "omap-pcm-audio",
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.codec_name = "twl6040-codec",
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.init = sdp4430_twl6040_init,
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.ops = &sdp4430_ops,
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},
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{
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.name = "DMIC",
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.stream_name = "DMIC Capture",
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.cpu_dai_name = "omap-dmic",
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.codec_dai_name = "dmic-hifi",
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.platform_name = "omap-pcm-audio",
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.codec_name = "dmic-codec",
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.init = sdp4430_dmic_init,
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.ops = &sdp4430_dmic_ops,
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},
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};
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/* Audio machine driver */
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static struct snd_soc_card snd_soc_sdp4430 = {
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.name = "SDP4430",
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.dai_link = sdp4430_dai,
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.num_links = ARRAY_SIZE(sdp4430_dai),
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.dapm_widgets = sdp4430_twl6040_dapm_widgets,
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.num_dapm_widgets = ARRAY_SIZE(sdp4430_twl6040_dapm_widgets),
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.dapm_routes = audio_map,
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.num_dapm_routes = ARRAY_SIZE(audio_map),
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};
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static struct platform_device *sdp4430_snd_device;
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static int __init sdp4430_soc_init(void)
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{
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int ret;
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if (!machine_is_omap_4430sdp())
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return -ENODEV;
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printk(KERN_INFO "SDP4430 SoC init\n");
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sdp4430_snd_device = platform_device_alloc("soc-audio", -1);
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if (!sdp4430_snd_device) {
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printk(KERN_ERR "Platform device allocation failed\n");
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return -ENOMEM;
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}
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platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430);
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ret = platform_device_add(sdp4430_snd_device);
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if (ret)
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goto err;
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return 0;
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err:
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printk(KERN_ERR "Unable to add platform device\n");
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platform_device_put(sdp4430_snd_device);
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return ret;
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}
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module_init(sdp4430_soc_init);
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static void __exit sdp4430_soc_exit(void)
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{
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platform_device_unregister(sdp4430_snd_device);
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}
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module_exit(sdp4430_soc_exit);
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MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
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MODULE_DESCRIPTION("ALSA SoC SDP4430");
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MODULE_LICENSE("GPL");
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