android_kernel_xiaomi_sm8350/sound/soc/intel/boards/bytcht_da7213.c
Thomas Gleixner 8e8e69d67e treewide: Replace GPLv2 boilerplate/reference with SPDX - rule 285
Based on 1 normalized pattern(s):

  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

extracted by the scancode license scanner the SPDX license identifier

  GPL-2.0-only

has been chosen to replace the boilerplate/reference in 100 file(s).

Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Reviewed-by: Alexios Zavras <alexios.zavras@intel.com>
Reviewed-by: Allison Randal <allison@lohutok.net>
Cc: linux-spdx@vger.kernel.org
Link: https://lkml.kernel.org/r/20190529141900.918357685@linutronix.de
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-06-05 17:36:37 +02:00

276 lines
7.1 KiB
C

// SPDX-License-Identifier: GPL-2.0-only
/*
* bytcht-da7213.c - ASoc Machine driver for Intel Baytrail and
* Cherrytrail-based platforms, with Dialog DA7213 codec
*
* Copyright (C) 2017 Intel Corporation
* Author: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*/
#include <linux/module.h>
#include <linux/acpi.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <asm/platform_sst_audio.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-acpi.h>
#include "../../codecs/da7213.h"
#include "../atom/sst-atom-controls.h"
static const struct snd_kcontrol_new controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone Jack"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
SOC_DAPM_PIN_SWITCH("Mic"),
SOC_DAPM_PIN_SWITCH("Aux In"),
};
static const struct snd_soc_dapm_widget dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("Mic", NULL),
SND_SOC_DAPM_LINE("Aux In", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
{"Headphone Jack", NULL, "HPL"},
{"Headphone Jack", NULL, "HPR"},
{"AUXL", NULL, "Aux In"},
{"AUXR", NULL, "Aux In"},
/* Assume Mic1 is linked to Headset and Mic2 to on-board mic */
{"MIC1", NULL, "Headset Mic"},
{"MIC2", NULL, "Mic"},
/* SOC-codec link */
{"ssp2 Tx", NULL, "codec_out0"},
{"ssp2 Tx", NULL, "codec_out1"},
{"codec_in0", NULL, "ssp2 Rx"},
{"codec_in1", NULL, "ssp2 Rx"},
{"Playback", NULL, "ssp2 Tx"},
{"ssp2 Rx", NULL, "Capture"},
};
static int codec_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
int ret;
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 convert 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);
/*
* 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 aif1_startup(struct snd_pcm_substream *substream)
{
return snd_pcm_hw_constraint_single(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE, 48000);
}
static int 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;
ret = snd_soc_dai_set_sysclk(codec_dai, DA7213_CLKSRC_MCLK,
19200000, SND_SOC_CLOCK_IN);
if (ret < 0)
dev_err(codec_dai->dev, "can't set codec sysclk configuration\n");
ret = snd_soc_dai_set_pll(codec_dai, 0,
DA7213_SYSCLK_PLL_SRM, 0, DA7213_PLL_FREQ_OUT_98304000);
if (ret < 0) {
dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret);
return -EIO;
}
return ret;
}
static int aif1_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret;
ret = snd_soc_dai_set_pll(codec_dai, 0,
DA7213_SYSCLK_MCLK, 0, 0);
if (ret < 0) {
dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret);
return -EIO;
}
return ret;
}
static const struct snd_soc_ops aif1_ops = {
.startup = aif1_startup,
};
static const struct snd_soc_ops ssp2_ops = {
.hw_params = aif1_hw_params,
.hw_free = aif1_hw_free,
};
static struct snd_soc_dai_link dailink[] = {
[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 = &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 = &aif1_ops,
},
/* CODEC<->CODEC link */
/* back ends */
{
.name = "SSP2-Codec",
.id = 0,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
.codec_dai_name = "da7213-hifi",
.codec_name = "i2c-DLGS7213:00",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBS_CFS,
.be_hw_params_fixup = codec_fixup,
.nonatomic = true,
.dpcm_playback = 1,
.dpcm_capture = 1,
.ops = &ssp2_ops,
},
};
/* SoC card */
static struct snd_soc_card bytcht_da7213_card = {
.name = "bytcht-da7213",
.owner = THIS_MODULE,
.dai_link = dailink,
.num_links = ARRAY_SIZE(dailink),
.controls = controls,
.num_controls = ARRAY_SIZE(controls),
.dapm_widgets = dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static char codec_name[SND_ACPI_I2C_ID_LEN];
static int bytcht_da7213_probe(struct platform_device *pdev)
{
struct snd_soc_card *card;
struct snd_soc_acpi_mach *mach;
const char *platform_name;
struct acpi_device *adev;
int dai_index = 0;
int ret_val = 0;
int i;
mach = (&pdev->dev)->platform_data;
card = &bytcht_da7213_card;
card->dev = &pdev->dev;
/* fix index of codec dai */
for (i = 0; i < ARRAY_SIZE(dailink); i++) {
if (!strcmp(dailink[i].codec_name, "i2c-DLGS7213:00")) {
dai_index = i;
break;
}
}
/* fixup codec name based on HID */
adev = acpi_dev_get_first_match_dev(mach->id, NULL, -1);
if (adev) {
snprintf(codec_name, sizeof(codec_name),
"i2c-%s", acpi_dev_name(adev));
put_device(&adev->dev);
dailink[dai_index].codec_name = codec_name;
}
/* override plaform name, if required */
platform_name = mach->mach_params.platform;
ret_val = snd_soc_fixup_dai_links_platform_name(card, platform_name);
if (ret_val)
return ret_val;
ret_val = devm_snd_soc_register_card(&pdev->dev, card);
if (ret_val) {
dev_err(&pdev->dev,
"snd_soc_register_card failed %d\n", ret_val);
return ret_val;
}
platform_set_drvdata(pdev, card);
return ret_val;
}
static struct platform_driver bytcht_da7213_driver = {
.driver = {
.name = "bytcht_da7213",
},
.probe = bytcht_da7213_probe,
};
module_platform_driver(bytcht_da7213_driver);
MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail+DA7213 Machine driver");
MODULE_AUTHOR("Pierre-Louis Bossart");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:bytcht_da7213");