Mark Brown
8 years ago
commit
0c2964cb38
@ -0,0 +1,283 @@ |
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/*
|
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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 "../../codecs/da7213.h" |
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#include "../atom/sst-atom-controls.h" |
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#include "../common/sst-acpi.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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|
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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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[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 = "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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|
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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[16]; /* i2c-<HID>:00 with HID being 8 chars */ |
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static int bytcht_da7213_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 snd_soc_card *card; |
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struct sst_acpi_mach *mach; |
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const char *i2c_name = NULL; |
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int dai_index = 0; |
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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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dai_index = MERR_DPCM_COMPR + 1; |
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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 = sst_acpi_find_name_from_hid(mach->id); |
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if (i2c_name != NULL) { |
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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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@ -0,0 +1,208 @@ |
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/*
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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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|
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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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/*
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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 unsigned int rates_48000[] = { |
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48000, |
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}; |
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static 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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|
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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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[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-LowSpeed Connector", |
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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 = "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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|
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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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|
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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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|
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/* register the soc card */ |
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bytcht_nocodec_card.dev = &pdev->dev; |
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|
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ret_val = devm_snd_soc_register_card(&pdev->dev, &bytcht_nocodec_card); |
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|
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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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|
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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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|
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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"); |
||||
MODULE_ALIAS("platform:bytcht_nocodec"); |
||||
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Reference in new issue