1 // SPDX-License-Identifier: GPL-2.0+ 2 // 3 // soc-util.c -- ALSA SoC Audio Layer utility functions 4 // 5 // Copyright 2009 Wolfson Microelectronics PLC. 6 // 7 // Author: Mark Brown <broonie@opensource.wolfsonmicro.com> 8 // Liam Girdwood <lrg@slimlogic.co.uk> 9 10 #include <linux/platform_device.h> 11 #include <linux/export.h> 12 #include <sound/core.h> 13 #include <sound/pcm.h> 14 #include <sound/pcm_params.h> 15 #include <sound/soc.h> 16 17 int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots) 18 { 19 return sample_size * channels * tdm_slots; 20 } 21 EXPORT_SYMBOL_GPL(snd_soc_calc_frame_size); 22 23 int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params) 24 { 25 int sample_size; 26 27 sample_size = snd_pcm_format_width(params_format(params)); 28 if (sample_size < 0) 29 return sample_size; 30 31 return snd_soc_calc_frame_size(sample_size, params_channels(params), 32 1); 33 } 34 EXPORT_SYMBOL_GPL(snd_soc_params_to_frame_size); 35 36 int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots) 37 { 38 return fs * snd_soc_calc_frame_size(sample_size, channels, tdm_slots); 39 } 40 EXPORT_SYMBOL_GPL(snd_soc_calc_bclk); 41 42 int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params) 43 { 44 int ret; 45 46 ret = snd_soc_params_to_frame_size(params); 47 48 if (ret > 0) 49 return ret * params_rate(params); 50 else 51 return ret; 52 } 53 EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk); 54 55 static const struct snd_pcm_hardware dummy_dma_hardware = { 56 /* Random values to keep userspace happy when checking constraints */ 57 .info = SNDRV_PCM_INFO_INTERLEAVED | 58 SNDRV_PCM_INFO_BLOCK_TRANSFER, 59 .buffer_bytes_max = 128*1024, 60 .period_bytes_min = PAGE_SIZE, 61 .period_bytes_max = PAGE_SIZE*2, 62 .periods_min = 2, 63 .periods_max = 128, 64 }; 65 66 static int dummy_dma_open(struct snd_soc_component *component, 67 struct snd_pcm_substream *substream) 68 { 69 struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); 70 71 /* BE's dont need dummy params */ 72 if (!rtd->dai_link->no_pcm) 73 snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware); 74 75 return 0; 76 } 77 78 static const struct snd_soc_component_driver dummy_platform = { 79 .open = dummy_dma_open, 80 }; 81 82 static const struct snd_soc_component_driver dummy_codec = { 83 .idle_bias_on = 1, 84 .use_pmdown_time = 1, 85 .endianness = 1, 86 .non_legacy_dai_naming = 1, 87 }; 88 89 #define STUB_RATES SNDRV_PCM_RATE_8000_384000 90 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ 91 SNDRV_PCM_FMTBIT_U8 | \ 92 SNDRV_PCM_FMTBIT_S16_LE | \ 93 SNDRV_PCM_FMTBIT_U16_LE | \ 94 SNDRV_PCM_FMTBIT_S24_LE | \ 95 SNDRV_PCM_FMTBIT_S24_3LE | \ 96 SNDRV_PCM_FMTBIT_U24_LE | \ 97 SNDRV_PCM_FMTBIT_S32_LE | \ 98 SNDRV_PCM_FMTBIT_U32_LE | \ 99 SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) 100 101 /* 102 * Select these from Sound Card Manually 103 * SND_SOC_POSSIBLE_DAIFMT_CBP_CFP 104 * SND_SOC_POSSIBLE_DAIFMT_CBP_CFC 105 * SND_SOC_POSSIBLE_DAIFMT_CBC_CFP 106 * SND_SOC_POSSIBLE_DAIFMT_CBC_CFC 107 */ 108 static u64 dummy_dai_formats = 109 SND_SOC_POSSIBLE_DAIFMT_I2S | 110 SND_SOC_POSSIBLE_DAIFMT_RIGHT_J | 111 SND_SOC_POSSIBLE_DAIFMT_LEFT_J | 112 SND_SOC_POSSIBLE_DAIFMT_DSP_A | 113 SND_SOC_POSSIBLE_DAIFMT_DSP_B | 114 SND_SOC_POSSIBLE_DAIFMT_AC97 | 115 SND_SOC_POSSIBLE_DAIFMT_PDM | 116 SND_SOC_POSSIBLE_DAIFMT_GATED | 117 SND_SOC_POSSIBLE_DAIFMT_CONT | 118 SND_SOC_POSSIBLE_DAIFMT_NB_NF | 119 SND_SOC_POSSIBLE_DAIFMT_NB_IF | 120 SND_SOC_POSSIBLE_DAIFMT_IB_NF | 121 SND_SOC_POSSIBLE_DAIFMT_IB_IF; 122 123 static const struct snd_soc_dai_ops dummy_dai_ops = { 124 .auto_selectable_formats = &dummy_dai_formats, 125 .num_auto_selectable_formats = 1, 126 }; 127 128 /* 129 * The dummy CODEC is only meant to be used in situations where there is no 130 * actual hardware. 131 * 132 * If there is actual hardware even if it does not have a control bus 133 * the hardware will still have constraints like supported samplerates, etc. 134 * which should be modelled. And the data flow graph also should be modelled 135 * using DAPM. 136 */ 137 static struct snd_soc_dai_driver dummy_dai = { 138 .name = "snd-soc-dummy-dai", 139 .playback = { 140 .stream_name = "Playback", 141 .channels_min = 1, 142 .channels_max = 384, 143 .rates = STUB_RATES, 144 .formats = STUB_FORMATS, 145 }, 146 .capture = { 147 .stream_name = "Capture", 148 .channels_min = 1, 149 .channels_max = 384, 150 .rates = STUB_RATES, 151 .formats = STUB_FORMATS, 152 }, 153 .ops = &dummy_dai_ops, 154 }; 155 156 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai) 157 { 158 if (dai->driver == &dummy_dai) 159 return 1; 160 return 0; 161 } 162 163 int snd_soc_component_is_dummy(struct snd_soc_component *component) 164 { 165 return ((component->driver == &dummy_platform) || 166 (component->driver == &dummy_codec)); 167 } 168 169 static int snd_soc_dummy_probe(struct platform_device *pdev) 170 { 171 int ret; 172 173 ret = devm_snd_soc_register_component(&pdev->dev, 174 &dummy_codec, &dummy_dai, 1); 175 if (ret < 0) 176 return ret; 177 178 ret = devm_snd_soc_register_component(&pdev->dev, &dummy_platform, 179 NULL, 0); 180 181 return ret; 182 } 183 184 static struct platform_driver soc_dummy_driver = { 185 .driver = { 186 .name = "snd-soc-dummy", 187 }, 188 .probe = snd_soc_dummy_probe, 189 }; 190 191 static struct platform_device *soc_dummy_dev; 192 193 int __init snd_soc_util_init(void) 194 { 195 int ret; 196 197 soc_dummy_dev = 198 platform_device_register_simple("snd-soc-dummy", -1, NULL, 0); 199 if (IS_ERR(soc_dummy_dev)) 200 return PTR_ERR(soc_dummy_dev); 201 202 ret = platform_driver_register(&soc_dummy_driver); 203 if (ret != 0) 204 platform_device_unregister(soc_dummy_dev); 205 206 return ret; 207 } 208 209 void __exit snd_soc_util_exit(void) 210 { 211 platform_driver_unregister(&soc_dummy_driver); 212 platform_device_unregister(soc_dummy_dev); 213 } 214