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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 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 | /* * bytcht-da7213.c - ASoc Machine driver for Intel Baytrail and * Cherrytrail-based platforms, with Dialog DA7213 codec * * Copyright (C) 2017 Intel Corporation * Author: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> * * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; version 2 of the License. * * This program is distributed in the hope that it will be useful, but * WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * General Public License for more details. * * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ #include <linux/module.h> #include <linux/acpi.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <asm/platform_sst_audio.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-acpi.h> #include "../../codecs/da7213.h" #include "../atom/sst-atom-controls.h" static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Mic"), SOC_DAPM_PIN_SWITCH("Aux In"), }; static const struct snd_soc_dapm_widget dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_MIC("Mic", NULL), SND_SOC_DAPM_LINE("Aux In", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { {"Headphone Jack", NULL, "HPL"}, {"Headphone Jack", NULL, "HPR"}, {"AUXL", NULL, "Aux In"}, {"AUXR", NULL, "Aux In"}, /* Assume Mic1 is linked to Headset and Mic2 to on-board mic */ {"MIC1", NULL, "Headset Mic"}, {"MIC2", NULL, "Mic"}, /* SOC-codec link */ {"ssp2 Tx", NULL, "codec_out0"}, {"ssp2 Tx", NULL, "codec_out1"}, {"codec_in0", NULL, "ssp2 Rx"}, {"codec_in1", NULL, "ssp2 Rx"}, {"Playback", NULL, "ssp2 Tx"}, {"ssp2 Rx", NULL, "Capture"}, }; static int codec_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { int ret; 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); /* 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 int aif1_startup(struct snd_pcm_substream *substream) { return snd_pcm_hw_constraint_single(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, 48000); } static int aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; ret = snd_soc_dai_set_sysclk(codec_dai, DA7213_CLKSRC_MCLK, 19200000, SND_SOC_CLOCK_IN); if (ret < 0) dev_err(codec_dai->dev, "can't set codec sysclk configuration\n"); ret = snd_soc_dai_set_pll(codec_dai, 0, DA7213_SYSCLK_PLL_SRM, 0, DA7213_PLL_FREQ_OUT_98304000); if (ret < 0) { dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret); return -EIO; } return ret; } static int aif1_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; ret = snd_soc_dai_set_pll(codec_dai, 0, DA7213_SYSCLK_MCLK, 0, 0); if (ret < 0) { dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret); return -EIO; } return ret; } static const struct snd_soc_ops aif1_ops = { .startup = aif1_startup, }; static const struct snd_soc_ops ssp2_ops = { .hw_params = aif1_hw_params, .hw_free = aif1_hw_free, }; static struct snd_soc_dai_link dailink[] = { [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", .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", .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, .ops = &aif1_ops, }, /* CODEC<->CODEC link */ /* back ends */ { .name = "SSP2-Codec", .id = 0, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, .codec_dai_name = "da7213-hifi", .codec_name = "i2c-DLGS7213:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .be_hw_params_fixup = codec_fixup, .nonatomic = true, .dpcm_playback = 1, .dpcm_capture = 1, .ops = &ssp2_ops, }, }; /* SoC card */ static struct snd_soc_card bytcht_da7213_card = { .name = "bytcht-da7213", .owner = THIS_MODULE, .dai_link = dailink, .num_links = ARRAY_SIZE(dailink), .controls = controls, .num_controls = ARRAY_SIZE(controls), .dapm_widgets = dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(dapm_widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), }; static char codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */ static int bytcht_da7213_probe(struct platform_device *pdev) { struct snd_soc_card *card; struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; int dai_index = 0; int ret_val = 0; int i; mach = (&pdev->dev)->platform_data; card = &bytcht_da7213_card; card->dev = &pdev->dev; /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(dailink); i++) { if (!strcmp(dailink[i].codec_name, "i2c-DLGS7213:00")) { dai_index = i; break; } } /* fixup codec name based on HID */ i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); if (i2c_name) { snprintf(codec_name, sizeof(codec_name), "%s%s", "i2c-", i2c_name); dailink[dai_index].codec_name = codec_name; } 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 bytcht_da7213_driver = { .driver = { .name = "bytcht_da7213", }, .probe = bytcht_da7213_probe, }; module_platform_driver(bytcht_da7213_driver); MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail+DA7213 Machine driver"); MODULE_AUTHOR("Pierre-Louis Bossart"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:bytcht_da7213"); |