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f4c80d5a16
This time was again a relatively calm development cycle; most of updates are about drivers, and no radical changes are seen in any core code. Here are some highlights: ALSA core: - Continued hardening of ALSA hrtimer - A few leak fixes in timer interface - Fix poll error handling in PCM and compress - Add error propagation in compress API - Removal of dead rtctimer driver HD-audio: - Native ELD notify support for i915 HDMI - Realtek ALC234 & co support - Code refactoring to standardize chmap support - Continued development for SKL HDMI core support Firewire: - Apply delayed card registration to all drivers - Improved / stabilized the handling of PCM stream start / stop - Add tracepoints to dump a part of isochronous packet data - Fixed incoming/outgoing packet parameter usages - Add support for M-Audio profire series USB-audio: - Fixes for UAC2 clock source - SS+ support - Workaround for oft-seen repeated sample rate read errors ASoC: - Further slow progress on the topology code - Substantial updates and improvements for the da7219, es8328, fsl-ssi, Intel and rcar drivers. - Compress error handling in WM ADSP driver -----BEGIN PGP SIGNATURE----- Version: GnuPG v2 iQIcBAABCAAGBQJXPYgvAAoJEGwxgFQ9KSmka3IQAJfXxKYyL0mqOgUpFav2QprE j4nQFSQf2KMAHgod1iF4Pv5glRZ3T8CbWllu/+GT87ny4wwJH76D07VCZSnrA+cv NMxRMN8QiGWS+eNPDNqRbcpzQvgwRK17VAmvpIfZtdntq3IryPLyCnY+FJ6Xt5v7 CjgGjlKJQ8i6AJVtoKVlrCOTBPS8YezQ7o67v8+BNrHDyOr0pwLERhvqJBRjaCbj fKj+JNDsWyu4kX0nInKNGah+5Qiib68+UNK5M+/PnoWv9tEOBPNXeWqRkcRpwnrF t1BQLnKGdlcSIufXcvxHDdxLftJZ38w+EbnQ/2r+SYHYIwPqTWdvVeXZUiq70wW/ WBUEOHybaHTNc52nMpjo/PU72CHa29zvKq+QHMXMRmFfVrLepIgEpBRBUjENtCjM 3OUn1IhYiNI4FOfgLm5duuYSBVdS4C2qstBDMtGpP64l7AmBZMFtbGUP8pKhvpzF FR2VoQpBFLPo805lQBKYbxdpzUGqfR7M/O73WRMzB/ZPZa95VNCDoRDQBbYF4Wzy SByVcE56znxoS9AmbhU6LzCXxdyVp6YAXZNR0pHp+8QdrRoFQZwRhfNVN3FIeNub COV+0pCQ2GTYvVdfLjdh6VT4shXeg5ZrUVnE3akL+8OzXow9lKyhknvLHn71aTZi HT0vSirSdrEYf4zg6wtB =QsAc -----END PGP SIGNATURE----- Merge tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "This time was again a relatively calm development cycle; most of updates are about drivers, and no radical changes are seen in any core code. Here are some highlights: ALSA core: - Continued hardening of ALSA hrtimer - A few leak fixes in timer interface - Fix poll error handling in PCM and compress - Add error propagation in compress API - Removal of dead rtctimer driver HD-audio: - Native ELD notify support for i915 HDMI - Realtek ALC234 & co support - Code refactoring to standardize chmap support - Continued development for SKL HDMI core support Firewire: - Apply delayed card registration to all drivers - Improved / stabilized the handling of PCM stream start / stop - Add tracepoints to dump a part of isochronous packet data - Fixed incoming/outgoing packet parameter usages - Add support for M-Audio profire series USB-audio: - Fixes for UAC2 clock source - SS+ support - Workaround for oft-seen repeated sample rate read errors ASoC: - Further slow progress on the topology code - Substantial updates and improvements for the da7219, es8328, fsl-ssi, Intel and rcar drivers. - Compress error handling in WM ADSP driver" * tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (177 commits) ALSA: firewire-lib: change a member of event structure to suppress sparse wanings to bool type sound: oss: Use setup_timer and mod_timer. ASoC: hdac_hdmi: Remove the unused 'timeout' variable ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix samples being dropped at Playback startup ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: pcm5102a: Add support for PCM5102A codec ASoC: hdac_hdmi: add link management ASoC: Intel: Skylake: add link management ALSA: hdac: add link pm and ref counting ALSA: au88x0: Fix zero clear of stream->resources ASoC: rt298: Add DMI match for Broxton-P reference platform ASoC: rt298: fix null deref on acpi driver data ASoC: dapm: deprecate MICBIAS widget type ALSA: firewire-lib: drop skip argument from helper functions to queue a packet ALSA: firewire-lib: add context information to tracepoints ALSA: firewire-lib: permit to flush queued packets only in process context for better PCM period granularity ...
400 lines
11 KiB
C
400 lines
11 KiB
C
/*
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* cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms
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* Cherrytrail and Braswell, with RT5645 codec.
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*
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* Copyright (C) 2015 Intel Corp
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* Author: Fang, Yang A <yang.a.fang@intel.com>
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* N,Harshapriya <harshapriya.n@intel.com>
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* This file is modified from cht_bsw_rt5672.c
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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 <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/jack.h>
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#include "../../codecs/rt5645.h"
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#include "../atom/sst-atom-controls.h"
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#define CHT_PLAT_CLK_3_HZ 19200000
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#define CHT_CODEC_DAI "rt5645-aif1"
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struct cht_acpi_card {
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char *codec_id;
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int codec_type;
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struct snd_soc_card *soc_card;
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};
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struct cht_mc_private {
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struct snd_soc_jack jack;
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struct cht_acpi_card *acpi_card;
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};
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static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
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{
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struct snd_soc_pcm_runtime *rtd;
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list_for_each_entry(rtd, &card->rtd_list, list) {
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if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
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strlen(CHT_CODEC_DAI)))
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return rtd->codec_dai;
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}
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return NULL;
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}
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static int platform_clock_control(struct snd_soc_dapm_widget *w,
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struct snd_kcontrol *k, int event)
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{
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struct snd_soc_dapm_context *dapm = w->dapm;
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struct snd_soc_card *card = dapm->card;
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struct snd_soc_dai *codec_dai;
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int ret;
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codec_dai = cht_get_codec_dai(card);
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if (!codec_dai) {
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dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
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return -EIO;
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}
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if (!SND_SOC_DAPM_EVENT_OFF(event))
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return 0;
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/* Set codec sysclk source to its internal clock because codec PLL will
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* be off when idle and MCLK will also be off by ACPI when codec is
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* runtime suspended. Codec needs clock for jack detection and button
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* press.
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*/
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ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK,
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0, SND_SOC_CLOCK_IN);
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if (ret < 0) {
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dev_err(card->dev, "can't set codec sysclk: %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 struct snd_soc_dapm_widget cht_dapm_widgets[] = {
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SND_SOC_DAPM_HP("Headphone", NULL),
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SND_SOC_DAPM_MIC("Headset Mic", NULL),
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SND_SOC_DAPM_MIC("Int Mic", NULL),
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SND_SOC_DAPM_SPK("Ext Spk", NULL),
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SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
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platform_clock_control, SND_SOC_DAPM_POST_PMD),
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};
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static const struct snd_soc_dapm_route cht_rt5645_audio_map[] = {
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{"IN1P", NULL, "Headset Mic"},
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{"IN1N", NULL, "Headset Mic"},
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{"DMIC L1", NULL, "Int Mic"},
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{"DMIC R1", NULL, "Int Mic"},
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{"Headphone", NULL, "HPOL"},
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{"Headphone", NULL, "HPOR"},
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{"Ext Spk", NULL, "SPOL"},
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{"Ext Spk", NULL, "SPOR"},
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{"AIF1 Playback", NULL, "ssp2 Tx"},
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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, "AIF1 Capture"},
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{"Headphone", NULL, "Platform Clock"},
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{"Headset Mic", NULL, "Platform Clock"},
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{"Int Mic", NULL, "Platform Clock"},
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{"Ext Spk", NULL, "Platform Clock"},
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};
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static const struct snd_soc_dapm_route cht_rt5650_audio_map[] = {
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{"IN1P", NULL, "Headset Mic"},
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{"IN1N", NULL, "Headset Mic"},
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{"DMIC L2", NULL, "Int Mic"},
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{"DMIC R2", NULL, "Int Mic"},
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{"Headphone", NULL, "HPOL"},
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{"Headphone", NULL, "HPOR"},
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{"Ext Spk", NULL, "SPOL"},
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{"Ext Spk", NULL, "SPOR"},
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{"AIF1 Playback", NULL, "ssp2 Tx"},
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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, "AIF1 Capture"},
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{"Headphone", NULL, "Platform Clock"},
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{"Headset Mic", NULL, "Platform Clock"},
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{"Int Mic", NULL, "Platform Clock"},
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{"Ext Spk", NULL, "Platform Clock"},
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};
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static const struct snd_kcontrol_new cht_mc_controls[] = {
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SOC_DAPM_PIN_SWITCH("Headphone"),
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SOC_DAPM_PIN_SWITCH("Headset Mic"),
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SOC_DAPM_PIN_SWITCH("Int Mic"),
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SOC_DAPM_PIN_SWITCH("Ext Spk"),
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};
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static struct snd_soc_jack_pin cht_bsw_jack_pins[] = {
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{
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.pin = "Headphone",
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.mask = SND_JACK_HEADPHONE,
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},
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{
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.pin = "Headset Mic",
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.mask = SND_JACK_MICROPHONE,
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},
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};
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static int cht_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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/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
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ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
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CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
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if (ret < 0) {
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dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
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return ret;
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}
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ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1,
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params_rate(params) * 512, SND_SOC_CLOCK_IN);
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if (ret < 0) {
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dev_err(rtd->dev, "can't set codec sysclk: %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 cht_codec_init(struct snd_soc_pcm_runtime *runtime)
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{
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int ret;
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int jack_type;
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struct snd_soc_codec *codec = runtime->codec;
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struct snd_soc_dai *codec_dai = runtime->codec_dai;
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struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
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/* Select clk_i2s1_asrc as ASRC clock source */
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rt5645_sel_asrc_clk_src(codec,
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RT5645_DA_STEREO_FILTER |
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RT5645_DA_MONO_L_FILTER |
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RT5645_DA_MONO_R_FILTER |
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RT5645_AD_STEREO_FILTER,
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RT5645_CLK_SEL_I2S1_ASRC);
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/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
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ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
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if (ret < 0) {
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dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
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return ret;
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}
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if (ctx->acpi_card->codec_type == CODEC_TYPE_RT5650)
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jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
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SND_JACK_BTN_0 | SND_JACK_BTN_1 |
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SND_JACK_BTN_2 | SND_JACK_BTN_3;
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else
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jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE;
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ret = snd_soc_card_jack_new(runtime->card, "Headset",
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jack_type, &ctx->jack,
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cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins));
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if (ret) {
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dev_err(runtime->dev, "Headset jack creation failed %d\n", ret);
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return ret;
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}
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rt5645_set_jack_detect(codec, &ctx->jack, &ctx->jack, &ctx->jack);
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return ret;
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}
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static int cht_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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/* The DSP will covert 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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return 0;
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}
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static int cht_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 struct snd_soc_ops cht_aif1_ops = {
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.startup = cht_aif1_startup,
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};
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static struct snd_soc_ops cht_be_ssp2_ops = {
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.hw_params = cht_aif1_hw_params,
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};
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static struct snd_soc_dai_link cht_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 = &cht_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 = &cht_aif1_ops,
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},
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[MERR_DPCM_COMPR] = {
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.name = "Compressed Port",
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.stream_name = "Compress",
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.cpu_dai_name = "compress-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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},
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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 = 1,
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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 = "rt5645-aif1",
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.codec_name = "i2c-10EC5645:00",
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.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
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| SND_SOC_DAIFMT_CBS_CFS,
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.init = cht_codec_init,
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.be_hw_params_fixup = cht_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 = &cht_be_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 snd_soc_card_chtrt5645 = {
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.name = "chtrt5645",
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.owner = THIS_MODULE,
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.dai_link = cht_dailink,
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.num_links = ARRAY_SIZE(cht_dailink),
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.dapm_widgets = cht_dapm_widgets,
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.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
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.dapm_routes = cht_rt5645_audio_map,
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.num_dapm_routes = ARRAY_SIZE(cht_rt5645_audio_map),
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.controls = cht_mc_controls,
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.num_controls = ARRAY_SIZE(cht_mc_controls),
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};
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static struct snd_soc_card snd_soc_card_chtrt5650 = {
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.name = "chtrt5650",
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.owner = THIS_MODULE,
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.dai_link = cht_dailink,
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.num_links = ARRAY_SIZE(cht_dailink),
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.dapm_widgets = cht_dapm_widgets,
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.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
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.dapm_routes = cht_rt5650_audio_map,
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.num_dapm_routes = ARRAY_SIZE(cht_rt5650_audio_map),
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.controls = cht_mc_controls,
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.num_controls = ARRAY_SIZE(cht_mc_controls),
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};
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static struct cht_acpi_card snd_soc_cards[] = {
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{"10EC5645", CODEC_TYPE_RT5645, &snd_soc_card_chtrt5645},
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{"10EC5650", CODEC_TYPE_RT5650, &snd_soc_card_chtrt5650},
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};
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static int snd_cht_mc_probe(struct platform_device *pdev)
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{
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int ret_val = 0;
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int i;
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struct cht_mc_private *drv;
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struct snd_soc_card *card = snd_soc_cards[0].soc_card;
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char codec_name[16];
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drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC);
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if (!drv)
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return -ENOMEM;
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for (i = 0; i < ARRAY_SIZE(snd_soc_cards); i++) {
|
|
if (acpi_dev_found(snd_soc_cards[i].codec_id)) {
|
|
dev_dbg(&pdev->dev,
|
|
"found codec %s\n", snd_soc_cards[i].codec_id);
|
|
card = snd_soc_cards[i].soc_card;
|
|
drv->acpi_card = &snd_soc_cards[i];
|
|
break;
|
|
}
|
|
}
|
|
card->dev = &pdev->dev;
|
|
sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id);
|
|
|
|
/* set correct codec name */
|
|
for (i = 0; i < ARRAY_SIZE(cht_dailink); i++)
|
|
if (!strcmp(card->dai_link[i].codec_name, "i2c-10EC5645:00"))
|
|
card->dai_link[i].codec_name = kstrdup(codec_name, GFP_KERNEL);
|
|
|
|
snd_soc_card_set_drvdata(card, drv);
|
|
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 snd_cht_mc_driver = {
|
|
.driver = {
|
|
.name = "cht-bsw-rt5645",
|
|
},
|
|
.probe = snd_cht_mc_probe,
|
|
};
|
|
|
|
module_platform_driver(snd_cht_mc_driver)
|
|
|
|
MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver");
|
|
MODULE_AUTHOR("Fang, Yang A,N,Harshapriya");
|
|
MODULE_LICENSE("GPL v2");
|
|
MODULE_ALIAS("platform:cht-bsw-rt5645");
|