mirror of
https://github.com/AuxXxilium/linux_dsm_epyc7002.git
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3a1479599a
Instead of home grown snd_soc_acpi_find_name_from_hid() use acpi_dev_get_first_match_name(). Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
275 lines
7.3 KiB
C
275 lines
7.3 KiB
C
/*
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* bytcht-da7213.c - ASoc Machine driver for Intel Baytrail and
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* Cherrytrail-based platforms, with Dialog DA7213 codec
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*
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* Copyright (C) 2017 Intel Corporation
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* Author: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
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*
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* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; version 2 of the License.
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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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* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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*/
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#include <linux/module.h>
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#include <linux/acpi.h>
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#include <linux/platform_device.h>
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#include <linux/slab.h>
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#include <asm/platform_sst_audio.h>
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#include <sound/pcm.h>
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#include <sound/pcm_params.h>
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#include <sound/soc.h>
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#include <sound/soc-acpi.h>
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#include "../../codecs/da7213.h"
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#include "../atom/sst-atom-controls.h"
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static const struct snd_kcontrol_new controls[] = {
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SOC_DAPM_PIN_SWITCH("Headphone Jack"),
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SOC_DAPM_PIN_SWITCH("Headset Mic"),
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SOC_DAPM_PIN_SWITCH("Mic"),
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SOC_DAPM_PIN_SWITCH("Aux In"),
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};
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static const struct snd_soc_dapm_widget dapm_widgets[] = {
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SND_SOC_DAPM_HP("Headphone Jack", NULL),
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SND_SOC_DAPM_MIC("Headset Mic", NULL),
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SND_SOC_DAPM_MIC("Mic", NULL),
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SND_SOC_DAPM_LINE("Aux In", NULL),
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};
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static const struct snd_soc_dapm_route audio_map[] = {
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{"Headphone Jack", NULL, "HPL"},
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{"Headphone Jack", NULL, "HPR"},
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{"AUXL", NULL, "Aux In"},
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{"AUXR", NULL, "Aux In"},
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/* Assume Mic1 is linked to Headset and Mic2 to on-board mic */
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{"MIC1", NULL, "Headset Mic"},
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{"MIC2", NULL, "Mic"},
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/* SOC-codec link */
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{"ssp2 Tx", NULL, "codec_out0"},
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{"ssp2 Tx", NULL, "codec_out1"},
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{"codec_in0", NULL, "ssp2 Rx"},
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{"codec_in1", NULL, "ssp2 Rx"},
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{"Playback", NULL, "ssp2 Tx"},
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{"ssp2 Rx", NULL, "Capture"},
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};
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static int codec_fixup(struct snd_soc_pcm_runtime *rtd,
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struct snd_pcm_hw_params *params)
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{
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int ret;
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struct snd_interval *rate = hw_param_interval(params,
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SNDRV_PCM_HW_PARAM_RATE);
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struct snd_interval *channels = hw_param_interval(params,
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SNDRV_PCM_HW_PARAM_CHANNELS);
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/* The DSP will convert the FE rate to 48k, stereo, 24bits */
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rate->min = rate->max = 48000;
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channels->min = channels->max = 2;
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/* set SSP2 to 24-bit */
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params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
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/*
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* Default mode for SSP configuration is TDM 4 slot, override config
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* with explicit setting to I2S 2ch 24-bit. The word length is set with
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* dai_set_tdm_slot() since there is no other API exposed
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*/
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ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
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SND_SOC_DAIFMT_I2S |
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SND_SOC_DAIFMT_NB_NF |
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SND_SOC_DAIFMT_CBS_CFS);
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if (ret < 0) {
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dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
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return ret;
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}
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ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
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if (ret < 0) {
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dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
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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 int aif1_startup(struct snd_pcm_substream *substream)
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{
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return snd_pcm_hw_constraint_single(substream->runtime,
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SNDRV_PCM_HW_PARAM_RATE, 48000);
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}
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static int aif1_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 ret;
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ret = snd_soc_dai_set_sysclk(codec_dai, DA7213_CLKSRC_MCLK,
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19200000, SND_SOC_CLOCK_IN);
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if (ret < 0)
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dev_err(codec_dai->dev, "can't set codec sysclk configuration\n");
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ret = snd_soc_dai_set_pll(codec_dai, 0,
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DA7213_SYSCLK_PLL_SRM, 0, DA7213_PLL_FREQ_OUT_98304000);
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if (ret < 0) {
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dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret);
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return -EIO;
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}
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return ret;
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}
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static int aif1_hw_free(struct snd_pcm_substream *substream)
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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 ret;
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ret = snd_soc_dai_set_pll(codec_dai, 0,
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DA7213_SYSCLK_MCLK, 0, 0);
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if (ret < 0) {
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dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret);
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return -EIO;
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}
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return ret;
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}
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static const struct snd_soc_ops aif1_ops = {
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.startup = aif1_startup,
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};
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static const struct snd_soc_ops ssp2_ops = {
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.hw_params = aif1_hw_params,
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.hw_free = aif1_hw_free,
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};
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static struct snd_soc_dai_link dailink[] = {
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[MERR_DPCM_AUDIO] = {
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.name = "Audio Port",
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.stream_name = "Audio",
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.cpu_dai_name = "media-cpu-dai",
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.codec_dai_name = "snd-soc-dummy-dai",
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.codec_name = "snd-soc-dummy",
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.platform_name = "sst-mfld-platform",
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.nonatomic = true,
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.dynamic = 1,
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.dpcm_playback = 1,
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.dpcm_capture = 1,
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.ops = &aif1_ops,
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},
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[MERR_DPCM_DEEP_BUFFER] = {
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.name = "Deep-Buffer Audio Port",
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.stream_name = "Deep-Buffer Audio",
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.cpu_dai_name = "deepbuffer-cpu-dai",
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.codec_dai_name = "snd-soc-dummy-dai",
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.codec_name = "snd-soc-dummy",
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.platform_name = "sst-mfld-platform",
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.nonatomic = true,
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.dynamic = 1,
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.dpcm_playback = 1,
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.ops = &aif1_ops,
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},
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/* CODEC<->CODEC link */
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/* back ends */
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{
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.name = "SSP2-Codec",
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.id = 0,
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.cpu_dai_name = "ssp2-port",
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.platform_name = "sst-mfld-platform",
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.no_pcm = 1,
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.codec_dai_name = "da7213-hifi",
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.codec_name = "i2c-DLGS7213:00",
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.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
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| SND_SOC_DAIFMT_CBS_CFS,
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.be_hw_params_fixup = codec_fixup,
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.nonatomic = true,
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.dpcm_playback = 1,
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.dpcm_capture = 1,
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.ops = &ssp2_ops,
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},
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};
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/* SoC card */
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static struct snd_soc_card bytcht_da7213_card = {
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.name = "bytcht-da7213",
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.owner = THIS_MODULE,
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.dai_link = dailink,
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.num_links = ARRAY_SIZE(dailink),
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.controls = controls,
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.num_controls = ARRAY_SIZE(controls),
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.dapm_widgets = dapm_widgets,
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.num_dapm_widgets = ARRAY_SIZE(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 char codec_name[SND_ACPI_I2C_ID_LEN];
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static int bytcht_da7213_probe(struct platform_device *pdev)
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{
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struct snd_soc_card *card;
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struct snd_soc_acpi_mach *mach;
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const char *i2c_name = NULL;
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int dai_index = 0;
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int ret_val = 0;
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int i;
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mach = (&pdev->dev)->platform_data;
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card = &bytcht_da7213_card;
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card->dev = &pdev->dev;
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/* fix index of codec dai */
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for (i = 0; i < ARRAY_SIZE(dailink); i++) {
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if (!strcmp(dailink[i].codec_name, "i2c-DLGS7213:00")) {
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dai_index = i;
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break;
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}
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}
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/* fixup codec name based on HID */
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i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1);
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if (i2c_name) {
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snprintf(codec_name, sizeof(codec_name),
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"%s%s", "i2c-", i2c_name);
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dailink[dai_index].codec_name = codec_name;
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}
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ret_val = devm_snd_soc_register_card(&pdev->dev, card);
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if (ret_val) {
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dev_err(&pdev->dev,
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"snd_soc_register_card failed %d\n", ret_val);
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return ret_val;
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}
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platform_set_drvdata(pdev, card);
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return ret_val;
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}
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static struct platform_driver bytcht_da7213_driver = {
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.driver = {
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.name = "bytcht_da7213",
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},
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.probe = bytcht_da7213_probe,
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};
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module_platform_driver(bytcht_da7213_driver);
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MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail+DA7213 Machine driver");
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MODULE_AUTHOR("Pierre-Louis Bossart");
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MODULE_LICENSE("GPL v2");
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MODULE_ALIAS("platform:bytcht_da7213");
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