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1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 | // SPDX-License-Identifier: GPL-2.0-only /* * bytcht_nocodec.c - ASoc Machine driver for MinnowBoard Max and Up * to make I2S signals observable on the Low-Speed connector. Audio codec * is not managed by ASoC/DAPM * * Copyright (C) 2015-2017 Intel Corp * * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ * * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ #include <linux/module.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include "../atom/sst-atom-controls.h" static const struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_MIC("Mic", NULL), SND_SOC_DAPM_SPK("Speaker", NULL), }; static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Mic"), SOC_DAPM_PIN_SWITCH("Speaker"), }; static const struct snd_soc_dapm_route audio_map[] = { {"ssp2 Tx", NULL, "codec_out0"}, {"ssp2 Tx", NULL, "codec_out1"}, {"codec_in0", NULL, "ssp2 Rx"}, {"codec_in1", NULL, "ssp2 Rx"}, {"ssp2 Rx", NULL, "Mic"}, {"Speaker", NULL, "ssp2 Tx"}, }; static int codec_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); int ret; /* The DSP will convert the FE rate to 48k, stereo, 24bits */ rate->min = rate->max = 48000; channels->min = channels->max = 2; /* set SSP2 to 24-bit */ params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); /* * Default mode for SSP configuration is TDM 4 slot, override config * with explicit setting to I2S 2ch 24-bit. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ ret = snd_soc_dai_set_fmt(rtd->cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; } ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; } return 0; } static const unsigned int rates_48000[] = { 48000, }; static const struct snd_pcm_hw_constraint_list constraints_48000 = { .count = ARRAY_SIZE(rates_48000), .list = rates_48000, }; static int aif1_startup(struct snd_pcm_substream *substream) { return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_48000); } static struct snd_soc_ops aif1_ops = { .startup = aif1_startup, }; static struct snd_soc_dai_link dais[] = { [MERR_DPCM_AUDIO] = { .name = "Audio Port", .stream_name = "Audio", .cpu_dai_name = "media-cpu-dai", .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", .ignore_suspend = 1, .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ops = &aif1_ops, }, [MERR_DPCM_DEEP_BUFFER] = { .name = "Deep-Buffer Audio Port", .stream_name = "Deep-Buffer Audio", .cpu_dai_name = "deepbuffer-cpu-dai", .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", .ignore_suspend = 1, .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, .ops = &aif1_ops, }, /* CODEC<->CODEC link */ /* back ends */ { .name = "SSP2-LowSpeed Connector", .id = 0, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .be_hw_params_fixup = codec_fixup, .ignore_suspend = 1, .nonatomic = true, .dpcm_playback = 1, .dpcm_capture = 1, }, }; /* SoC card */ static struct snd_soc_card bytcht_nocodec_card = { .name = "bytcht-nocodec", .owner = THIS_MODULE, .dai_link = dais, .num_links = ARRAY_SIZE(dais), .dapm_widgets = widgets, .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), .controls = controls, .num_controls = ARRAY_SIZE(controls), .fully_routed = true, }; static int snd_bytcht_nocodec_mc_probe(struct platform_device *pdev) { int ret_val = 0; /* register the soc card */ bytcht_nocodec_card.dev = &pdev->dev; ret_val = devm_snd_soc_register_card(&pdev->dev, &bytcht_nocodec_card); if (ret_val) { dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n", ret_val); return ret_val; } platform_set_drvdata(pdev, &bytcht_nocodec_card); return ret_val; } static struct platform_driver snd_bytcht_nocodec_mc_driver = { .driver = { .name = "bytcht_nocodec", }, .probe = snd_bytcht_nocodec_mc_probe, }; module_platform_driver(snd_bytcht_nocodec_mc_driver); MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail Nocodec Machine driver"); MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart at linux.intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:bytcht_nocodec"); |