- 根目录:
- sound
- soc
- intel
- boards
- cht_bsw_rt5672.c
/*
* cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms
* Cherrytrail and Braswell, with RT5672 codec.
*
* Copyright (C) 2014 Intel Corp
* Author: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
* Mengdong Lin <mengdong.lin@intel.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; version 2 of the License.
*
* 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.
*/
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include "../../codecs/rt5670.h"
#include "../atom/sst-atom-controls.h"
/* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */
#define CHT_PLAT_CLK_3_HZ 19200000
#define CHT_CODEC_DAI "rt5670-aif1"
static struct snd_soc_jack cht_bsw_headset;
/* Headset jack detection DAPM pins */
static struct snd_soc_jack_pin cht_bsw_headset_pins[] = {
{
.pin = "Headset Mic",
.mask = SND_JACK_MICROPHONE,
},
{
.pin = "Headphone",
.mask = SND_JACK_HEADPHONE,
},
};
static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
{
int i;
for (i = 0; i < card->num_rtd; i++) {
struct snd_soc_pcm_runtime *rtd;
rtd = card->rtd + i;
if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
strlen(CHT_CODEC_DAI)))
return rtd->codec_dai;
}
return NULL;
}
static int platform_clock_control(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
struct snd_soc_dapm_context *dapm = w->dapm;
struct snd_soc_card *card = dapm->card;
struct snd_soc_dai *codec_dai;
int ret;
codec_dai = cht_get_codec_dai(card);
if (!codec_dai) {
dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
return -EIO;
}
if (SND_SOC_DAPM_EVENT_ON(event)) {
/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK,
CHT_PLAT_CLK_3_HZ, 48000 * 512);
if (ret < 0) {
dev_err(card->dev, "can't set codec pll: %d\n", ret);
return ret;
}
/* set codec sysclk source to PLL */
ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1,
48000 * 512, SND_SOC_CLOCK_IN);
if (ret < 0) {
dev_err(card->dev, "can't set codec sysclk: %d\n", ret);
return ret;
}
} else {
/* Set codec sysclk source to its internal clock because codec
* PLL will be off when idle and MCLK will also be off by ACPI
* when codec is runtime suspended. Codec needs clock for jack
* detection and button press.
*/
snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK,
48000 * 512, SND_SOC_CLOCK_IN);
}
return 0;
}
static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("Int Mic", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
platform_clock_control, SND_SOC_DAPM_PRE_PMU |
SND_SOC_DAPM_POST_PMD),
};
static const struct snd_soc_dapm_route cht_audio_map[] = {
{"IN1P", NULL, "Headset Mic"},
{"IN1N", NULL, "Headset Mic"},
{"DMIC L1", NULL, "Int Mic"},
{"DMIC R1", NULL, "Int Mic"},
{"Headphone", NULL, "HPOL"},
{"Headphone", NULL, "HPOR"},
{"Ext Spk", NULL, "SPOLP"},
{"Ext Spk", NULL, "SPOLN"},
{"Ext Spk", NULL, "SPORP"},
{"Ext Spk", NULL, "SPORN"},
{"AIF1 Playback", NULL, "ssp2 Tx"},
{"ssp2 Tx", NULL, "codec_out0"},
{"ssp2 Tx", NULL, "codec_out1"},
{"codec_in0", NULL, "ssp2 Rx"},
{"codec_in1", NULL, "ssp2 Rx"},
{"ssp2 Rx", NULL, "AIF1 Capture"},
{"Headphone", NULL, "Platform Clock"},
{"Headset Mic", NULL, "Platform Clock"},
{"Int Mic", NULL, "Platform Clock"},
{"Ext Spk", NULL, "Platform Clock"},
};
static const struct snd_kcontrol_new cht_mc_controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
SOC_DAPM_PIN_SWITCH("Int Mic"),
SOC_DAPM_PIN_SWITCH("Ext Spk"),
};
static int cht_aif1_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 ret;
/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK,
CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
if (ret < 0) {
dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
return ret;
}
/* set codec sysclk source to PLL */
ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1,
params_rate(params) * 512,
SND_SOC_CLOCK_IN);
if (ret < 0) {
dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
return ret;
}
return 0;
}
static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
{
int ret;
struct snd_soc_dai *codec_dai = runtime->codec_dai;
struct snd_soc_codec *codec = codec_dai->codec;
/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
if (ret < 0) {
dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
return ret;
}
/* Select codec ASRC clock source to track I2S1 clock, because codec
* is in slave mode and 100fs I2S format (BCLK = 100 * LRCLK) cannot
* be supported by RT5672. Otherwise, ASRC will be disabled and cause
* noise.
*/
rt5670_sel_asrc_clk_src(codec,
RT5670_DA_STEREO_FILTER
| RT5670_DA_MONO_L_FILTER
| RT5670_DA_MONO_R_FILTER
| RT5670_AD_STEREO_FILTER
| RT5670_AD_MONO_L_FILTER
| RT5670_AD_MONO_R_FILTER,
RT5670_CLK_SEL_I2S1_ASRC);
ret = snd_soc_card_jack_new(runtime->card, "Headset",
SND_JACK_HEADSET | SND_JACK_BTN_0 |
SND_JACK_BTN_1 | SND_JACK_BTN_2, &cht_bsw_headset,
cht_bsw_headset_pins, ARRAY_SIZE(cht_bsw_headset_pins));
if (ret)
return ret;
rt5670_set_jack_detect(codec, &cht_bsw_headset);
return 0;
}
static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
/* The DSP will covert the FE rate to 48k, stereo, 24bits */
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
/* set SSP2 to 24-bit */
params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
return 0;
}
static unsigned int rates_48000[] = {
48000,
};
static struct snd_pcm_hw_constraint_list constraints_48000 = {
.count = ARRAY_SIZE(rates_48000),
.list = rates_48000,
};
static int cht_aif1_startup(struct snd_pcm_substream *substream)
{
return snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&constraints_48000);
}
static struct snd_soc_ops cht_aif1_ops = {
.startup = cht_aif1_startup,
};
static struct snd_soc_ops cht_be_ssp2_ops = {
.hw_params = cht_aif1_hw_params,
};
static struct snd_soc_dai_link cht_dailink[] = {
/* Front End DAI links */
[MERR_DPCM_AUDIO] = {
.name = "Audio Port",
.stream_name = "Audio",
.cpu_dai_name = "media-cpu-dai",
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.platform_name = "sst-mfld-platform",
.nonatomic = true,
.dynamic = 1,
.dpcm_playback = 1,
.dpcm_capture = 1,
.ops = &cht_aif1_ops,
},
[MERR_DPCM_COMPR] = {
.name = "Compressed Port",
.stream_name = "Compress",
.cpu_dai_name = "compress-cpu-dai",
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.platform_name = "sst-mfld-platform",
},
/* Back End DAI links */
{
/* SSP2 - Codec */
.name = "SSP2-Codec",
.be_id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
.nonatomic = true,
.codec_dai_name = "rt5670-aif1",
.codec_name = "i2c-10EC5670:00",
.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
| SND_SOC_DAIFMT_CBS_CFS,
.init = cht_codec_init,
.be_hw_params_fixup = cht_codec_fixup,
.dpcm_playback = 1,
.dpcm_capture = 1,
.ops = &cht_be_ssp2_ops,
},
};
static int cht_suspend_pre(struct snd_soc_card *card)
{
struct snd_soc_codec *codec;
list_for_each_entry(codec, &card->codec_dev_list, card_list) {
if (!strcmp(codec->component.name, "i2c-10EC5670:00")) {
dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n");
rt5670_jack_suspend(codec);
break;
}
}
return 0;
}
static int cht_resume_post(struct snd_soc_card *card)
{
struct snd_soc_codec *codec;
list_for_each_entry(codec, &card->codec_dev_list, card_list) {
if (!strcmp(codec->component.name, "i2c-10EC5670:00")) {
dev_dbg(codec->dev, "enabling jack detect for resume.\n");
rt5670_jack_resume(codec);
break;
}
}
return 0;
}
/* SoC card */
static struct snd_soc_card snd_soc_card_cht = {
.name = "cherrytrailcraudio",
.dai_link = cht_dailink,
.num_links = ARRAY_SIZE(cht_dailink),
.dapm_widgets = cht_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
.dapm_routes = cht_audio_map,
.num_dapm_routes = ARRAY_SIZE(cht_audio_map),
.controls = cht_mc_controls,
.num_controls = ARRAY_SIZE(cht_mc_controls),
.suspend_pre = cht_suspend_pre,
.resume_post = cht_resume_post,
};
static int snd_cht_mc_probe(struct platform_device *pdev)
{
int ret_val = 0;
/* register the soc card */
snd_soc_card_cht.dev = &pdev->dev;
ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
if (ret_val) {
dev_err(&pdev->dev,
"snd_soc_register_card failed %d\n", ret_val);
return ret_val;
}
platform_set_drvdata(pdev, &snd_soc_card_cht);
return ret_val;
}
static struct platform_driver snd_cht_mc_driver = {
.driver = {
.name = "cht-bsw-rt5672",
},
.probe = snd_cht_mc_probe,
};
module_platform_driver(snd_cht_mc_driver);
MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver");
MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:cht-bsw-rt5672");