forked from Minki/linux
2193eb9631
The hard-coded compressed dailinks are not supported using publicly-available firmwares, which creates unnecessary user confusion [1]. Even if the firmware was available, the mainline code does not have the required .dynamic=1 and .dpcm_playback=1 fields so probably never worked as is, and last and they conflict with topology-defined streams. Remove them and move on. This can be re-enabled with SOF later in a more flexible manner. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/124868.html Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
201 lines
5.4 KiB
C
201 lines
5.4 KiB
C
/*
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* bytcht_nocodec.c - ASoc Machine driver for MinnowBoard Max and Up
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* to make I2S signals observable on the Low-Speed connector. Audio codec
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* is not managed by ASoC/DAPM
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*
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* Copyright (C) 2015-2017 Intel Corp
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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 <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 "../atom/sst-atom-controls.h"
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static const struct snd_soc_dapm_widget widgets[] = {
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SND_SOC_DAPM_MIC("Mic", NULL),
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SND_SOC_DAPM_SPK("Speaker", NULL),
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};
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static const struct snd_kcontrol_new controls[] = {
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SOC_DAPM_PIN_SWITCH("Mic"),
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SOC_DAPM_PIN_SWITCH("Speaker"),
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};
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static const struct snd_soc_dapm_route audio_map[] = {
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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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{"ssp2 Rx", NULL, "Mic"},
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{"Speaker", NULL, "ssp2 Tx"},
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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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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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int ret;
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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 const unsigned int rates_48000[] = {
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48000,
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};
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static const struct snd_pcm_hw_constraint_list constraints_48000 = {
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.count = ARRAY_SIZE(rates_48000),
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.list = rates_48000,
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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_list(substream->runtime, 0,
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SNDRV_PCM_HW_PARAM_RATE,
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&constraints_48000);
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}
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static struct snd_soc_ops aif1_ops = {
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.startup = aif1_startup,
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};
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static struct snd_soc_dai_link dais[] = {
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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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.ignore_suspend = 1,
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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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.ignore_suspend = 1,
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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-LowSpeed Connector",
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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 = "snd-soc-dummy-dai",
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.codec_name = "snd-soc-dummy",
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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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.ignore_suspend = 1,
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.nonatomic = true,
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.dpcm_playback = 1,
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.dpcm_capture = 1,
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},
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};
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/* SoC card */
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static struct snd_soc_card bytcht_nocodec_card = {
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.name = "bytcht-nocodec",
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.owner = THIS_MODULE,
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.dai_link = dais,
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.num_links = ARRAY_SIZE(dais),
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.dapm_widgets = widgets,
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.num_dapm_widgets = ARRAY_SIZE(widgets),
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.dapm_routes = audio_map,
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.num_dapm_routes = ARRAY_SIZE(audio_map),
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.controls = controls,
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.num_controls = ARRAY_SIZE(controls),
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.fully_routed = true,
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};
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static int snd_bytcht_nocodec_mc_probe(struct platform_device *pdev)
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{
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int ret_val = 0;
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/* register the soc card */
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bytcht_nocodec_card.dev = &pdev->dev;
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ret_val = devm_snd_soc_register_card(&pdev->dev, &bytcht_nocodec_card);
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if (ret_val) {
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dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n",
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ret_val);
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return ret_val;
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}
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platform_set_drvdata(pdev, &bytcht_nocodec_card);
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return ret_val;
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}
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static struct platform_driver snd_bytcht_nocodec_mc_driver = {
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.driver = {
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.name = "bytcht_nocodec",
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},
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.probe = snd_bytcht_nocodec_mc_probe,
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};
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module_platform_driver(snd_bytcht_nocodec_mc_driver);
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MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail Nocodec Machine driver");
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MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart at linux.intel.com>");
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MODULE_LICENSE("GPL v2");
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MODULE_ALIAS("platform:bytcht_nocodec");
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