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/*
* bytcht_es8316.c - ASoc Machine driver for Intel Baytrail/Cherrytrail
* platforms with Everest ES8316 SoC
*
* Copyright (C) 2017 Endless Mobile, Inc.
* Authors: David Yang <yangxiaohua@everest-semi.com>,
* Daniel Drake <drake@endlessm.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/init.h>
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/device.h>
#include <linux/slab.h>
#include <asm/platform_sst_audio.h>
#include <linux/clk.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include "../atom/sst-atom-controls.h"
#include "../common/sst-acpi.h"
#include "../common/sst-dsp.h"
struct byt_cht_es8316_private {
struct clk *mclk;
};
#define CODEC_DAI1 "ES8316 HiFi"
static inline struct snd_soc_dai *get_codec_dai(struct snd_soc_card *card)
{
struct snd_soc_pcm_runtime *rtd;
list_for_each_entry(rtd, &card->rtd_list, list) {
if (!strncmp(rtd->codec_dai->name, CODEC_DAI1,
strlen(CODEC_DAI1)))
return rtd->codec_dai;
}
return NULL;
}
static const struct snd_soc_dapm_widget byt_cht_es8316_widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
/*
* The codec supports two analog microphone inputs. I have only
* tested MIC1. A DMIC route could also potentially be added
* if such functionality is found on another platform.
*/
SND_SOC_DAPM_MIC("Microphone 1", NULL),
SND_SOC_DAPM_MIC("Microphone 2", NULL),
};
static const struct snd_soc_dapm_route byt_cht_es8316_audio_map[] = {
{"MIC1", NULL, "Microphone 1"},
{"MIC2", NULL, "Microphone 2"},
{"Headphone", NULL, "HPOL"},
{"Headphone", NULL, "HPOR"},
{"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, "Capture"},
};
static const struct snd_kcontrol_new byt_cht_es8316_controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone"),
SOC_DAPM_PIN_SWITCH("Microphone 1"),
SOC_DAPM_PIN_SWITCH("Microphone 2"),
};
static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime)
{
struct snd_soc_card *card = runtime->card;
struct byt_cht_es8316_private *priv = snd_soc_card_get_drvdata(card);
int ret;
card->dapm.idle_bias_off = true;
/*
* The firmware might enable the clock at boot (this information
* may or may not be reflected in the enable clock register).
* To change the rate we must disable the clock first to cover these
* cases. Due to common clock framework restrictions that do not allow
* to disable a clock that has not been enabled, we need to enable
* the clock first.
*/
ret = clk_prepare_enable(priv->mclk);
if (!ret)
clk_disable_unprepare(priv->mclk);
ret = clk_set_rate(priv->mclk, 19200000);
if (ret)
dev_err(card->dev, "unable to set MCLK rate\n");
ret = clk_prepare_enable(priv->mclk);
if (ret)
dev_err(card->dev, "unable to enable MCLK\n");
ret = snd_soc_dai_set_sysclk(runtime->codec_dai, 0, 19200000,
SND_SOC_CLOCK_IN);
if (ret < 0) {
dev_err(card->dev, "can't set codec clock %d\n", ret);
return ret;
}
return 0;
}
static const struct snd_soc_pcm_stream byt_cht_es8316_dai_params = {
.formats = SNDRV_PCM_FMTBIT_S24_LE,
.rate_min = 48000,
.rate_max = 48000,
.channels_min = 2,
.channels_max = 2,
};
static int byt_cht_es8316_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);
int ret;
/* The DSP will covert the FE rate to 48k, stereo */
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
/* set SSP2 to 24-bit */
params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
/*
* Default mode for SSP configuration is TDM 4 slot, override config
* with explicit setting to I2S 2ch 24-bit. The word length is set with
* dai_set_tdm_slot() since there is no other API exposed
*/
ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS
);
if (ret < 0) {
dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
if (ret < 0) {
dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
return ret;
}
return 0;
}
static int byt_cht_es8316_aif1_startup(struct snd_pcm_substream *substream)
{
return snd_pcm_hw_constraint_single(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE, 48000);
}
static const struct snd_soc_ops byt_cht_es8316_aif1_ops = {
.startup = byt_cht_es8316_aif1_startup,
};
static struct snd_soc_dai_link byt_cht_es8316_dais[] = {
[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 = &byt_cht_es8316_aif1_ops,
},
[MERR_DPCM_DEEP_BUFFER] = {
.name = "Deep-Buffer Audio Port",
.stream_name = "Deep-Buffer Audio",
.cpu_dai_name = "deepbuffer-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,
.ops = &byt_cht_es8316_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 ends */
{
/* Only SSP2 has been tested here, so BYT-CR platforms that
* require SSP0 will not work.
*/
.name = "SSP2-Codec",
.id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
.codec_dai_name = "ES8316 HiFi",
.codec_name = "i2c-ESSX8316:00",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBS_CFS,
.be_hw_params_fixup = byt_cht_es8316_codec_fixup,
.nonatomic = true,
.dpcm_playback = 1,
.dpcm_capture = 1,
.init = byt_cht_es8316_init,
},
};
/* SoC card */
static struct snd_soc_card byt_cht_es8316_card = {
.name = "bytcht-es8316",
.owner = THIS_MODULE,
.dai_link = byt_cht_es8316_dais,
.num_links = ARRAY_SIZE(byt_cht_es8316_dais),
.dapm_widgets = byt_cht_es8316_widgets,
.num_dapm_widgets = ARRAY_SIZE(byt_cht_es8316_widgets),
.dapm_routes = byt_cht_es8316_audio_map,
.num_dapm_routes = ARRAY_SIZE(byt_cht_es8316_audio_map),
.controls = byt_cht_es8316_controls,
.num_controls = ARRAY_SIZE(byt_cht_es8316_controls),
.fully_routed = true,
};
static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev)
{
int ret = 0;
struct byt_cht_es8316_private *priv;
priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC);
if (!priv)
return -ENOMEM;
/* register the soc card */
byt_cht_es8316_card.dev = &pdev->dev;
snd_soc_card_set_drvdata(&byt_cht_es8316_card, priv);
priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
if (IS_ERR(priv->mclk)) {
ret = PTR_ERR(priv->mclk);
dev_err(&pdev->dev,
"Failed to get MCLK from pmc_plt_clk_3: %d\n",
ret);
return ret;
}
ret = devm_snd_soc_register_card(&pdev->dev, &byt_cht_es8316_card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret);
return ret;
}
platform_set_drvdata(pdev, &byt_cht_es8316_card);
return ret;
}
static struct platform_driver snd_byt_cht_es8316_mc_driver = {
.driver = {
.name = "bytcht_es8316",
},
.probe = snd_byt_cht_es8316_mc_probe,
};
module_platform_driver(snd_byt_cht_es8316_mc_driver);
MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail Machine driver");
MODULE_AUTHOR("David Yang <yangxiaohua@everest-semi.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:bytcht_es8316");
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