| // SPDX-License-Identifier: GPL-2.0+ |
| // |
| // soc-util.c -- ALSA SoC Audio Layer utility functions |
| // |
| // Copyright 2009 Wolfson Microelectronics PLC. |
| // |
| // Author: Mark Brown <broonie@opensource.wolfsonmicro.com> |
| // Liam Girdwood <lrg@slimlogic.co.uk> |
| |
| #include <linux/platform_device.h> |
| #include <linux/export.h> |
| #include <linux/math.h> |
| #include <sound/core.h> |
| #include <sound/pcm.h> |
| #include <sound/pcm_params.h> |
| #include <sound/soc.h> |
| |
| int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots) |
| { |
| return sample_size * channels * tdm_slots; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_calc_frame_size); |
| |
| int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params) |
| { |
| int sample_size; |
| |
| sample_size = snd_pcm_format_width(params_format(params)); |
| if (sample_size < 0) |
| return sample_size; |
| |
| return snd_soc_calc_frame_size(sample_size, params_channels(params), |
| 1); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_params_to_frame_size); |
| |
| int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots) |
| { |
| return fs * snd_soc_calc_frame_size(sample_size, channels, tdm_slots); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_calc_bclk); |
| |
| int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params) |
| { |
| int ret; |
| |
| ret = snd_soc_params_to_frame_size(params); |
| |
| if (ret > 0) |
| return ret * params_rate(params); |
| else |
| return ret; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk); |
| |
| /** |
| * snd_soc_tdm_params_to_bclk - calculate bclk from params and tdm slot info. |
| * |
| * Calculate the bclk from the params sample rate and the tdm slot count and |
| * tdm slot width. Either or both of tdm_width and tdm_slots can be 0. |
| * |
| * If tdm_width == 0 and tdm_slots > 0: the params_width will be used. |
| * If tdm_width > 0 and tdm_slots == 0: the params_channels will be used |
| * as the slot count. |
| * Both tdm_width and tdm_slots are 0: this is equivalent to calling |
| * snd_soc_params_to_bclk(). |
| * |
| * If slot_multiple > 1 the slot count (or params_channels if tdm_slots == 0) |
| * will be rounded up to a multiple of this value. This is mainly useful for |
| * I2S mode, which has a left and right phase so the number of slots is always |
| * a multiple of 2. |
| * |
| * @params: Pointer to struct_pcm_hw_params. |
| * @tdm_width: Width in bits of the tdm slots. |
| * @tdm_slots: Number of tdm slots per frame. |
| * @slot_multiple: If >1 roundup slot count to a multiple of this value. |
| * |
| * Return: bclk frequency in Hz, else a negative error code if params format |
| * is invalid. |
| */ |
| int snd_soc_tdm_params_to_bclk(struct snd_pcm_hw_params *params, |
| int tdm_width, int tdm_slots, int slot_multiple) |
| { |
| if (!tdm_slots) |
| tdm_slots = params_channels(params); |
| |
| if (slot_multiple > 1) |
| tdm_slots = roundup(tdm_slots, slot_multiple); |
| |
| if (!tdm_width) { |
| tdm_width = snd_pcm_format_width(params_format(params)); |
| if (tdm_width < 0) |
| return tdm_width; |
| } |
| |
| return snd_soc_calc_bclk(params_rate(params), tdm_width, 1, tdm_slots); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_tdm_params_to_bclk); |
| |
| static const struct snd_pcm_hardware dummy_dma_hardware = { |
| /* Random values to keep userspace happy when checking constraints */ |
| .info = SNDRV_PCM_INFO_INTERLEAVED | |
| SNDRV_PCM_INFO_BLOCK_TRANSFER, |
| .buffer_bytes_max = 128*1024, |
| .period_bytes_min = PAGE_SIZE, |
| .period_bytes_max = PAGE_SIZE*2, |
| .periods_min = 2, |
| .periods_max = 128, |
| }; |
| |
| |
| static const struct snd_soc_component_driver dummy_platform; |
| |
| static int dummy_dma_open(struct snd_soc_component *component, |
| struct snd_pcm_substream *substream) |
| { |
| struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); |
| int i; |
| |
| /* |
| * If there are other components associated with rtd, we shouldn't |
| * override their hwparams |
| */ |
| for_each_rtd_components(rtd, i, component) { |
| if (component->driver == &dummy_platform) |
| return 0; |
| } |
| |
| /* BE's dont need dummy params */ |
| if (!rtd->dai_link->no_pcm) |
| snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware); |
| |
| return 0; |
| } |
| |
| static const struct snd_soc_component_driver dummy_platform = { |
| .open = dummy_dma_open, |
| }; |
| |
| static const struct snd_soc_component_driver dummy_codec = { |
| .idle_bias_on = 1, |
| .use_pmdown_time = 1, |
| .endianness = 1, |
| .non_legacy_dai_naming = 1, |
| }; |
| |
| #define STUB_RATES SNDRV_PCM_RATE_8000_384000 |
| #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ |
| SNDRV_PCM_FMTBIT_U8 | \ |
| SNDRV_PCM_FMTBIT_S16_LE | \ |
| SNDRV_PCM_FMTBIT_U16_LE | \ |
| SNDRV_PCM_FMTBIT_S24_LE | \ |
| SNDRV_PCM_FMTBIT_S24_3LE | \ |
| SNDRV_PCM_FMTBIT_U24_LE | \ |
| SNDRV_PCM_FMTBIT_S32_LE | \ |
| SNDRV_PCM_FMTBIT_U32_LE | \ |
| SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) |
| |
| /* |
| * Select these from Sound Card Manually |
| * SND_SOC_POSSIBLE_DAIFMT_CBP_CFP |
| * SND_SOC_POSSIBLE_DAIFMT_CBP_CFC |
| * SND_SOC_POSSIBLE_DAIFMT_CBC_CFP |
| * SND_SOC_POSSIBLE_DAIFMT_CBC_CFC |
| */ |
| static u64 dummy_dai_formats = |
| SND_SOC_POSSIBLE_DAIFMT_I2S | |
| SND_SOC_POSSIBLE_DAIFMT_RIGHT_J | |
| SND_SOC_POSSIBLE_DAIFMT_LEFT_J | |
| SND_SOC_POSSIBLE_DAIFMT_DSP_A | |
| SND_SOC_POSSIBLE_DAIFMT_DSP_B | |
| SND_SOC_POSSIBLE_DAIFMT_AC97 | |
| SND_SOC_POSSIBLE_DAIFMT_PDM | |
| SND_SOC_POSSIBLE_DAIFMT_GATED | |
| SND_SOC_POSSIBLE_DAIFMT_CONT | |
| SND_SOC_POSSIBLE_DAIFMT_NB_NF | |
| SND_SOC_POSSIBLE_DAIFMT_NB_IF | |
| SND_SOC_POSSIBLE_DAIFMT_IB_NF | |
| SND_SOC_POSSIBLE_DAIFMT_IB_IF; |
| |
| static const struct snd_soc_dai_ops dummy_dai_ops = { |
| .auto_selectable_formats = &dummy_dai_formats, |
| .num_auto_selectable_formats = 1, |
| }; |
| |
| /* |
| * The dummy CODEC is only meant to be used in situations where there is no |
| * actual hardware. |
| * |
| * If there is actual hardware even if it does not have a control bus |
| * the hardware will still have constraints like supported samplerates, etc. |
| * which should be modelled. And the data flow graph also should be modelled |
| * using DAPM. |
| */ |
| static struct snd_soc_dai_driver dummy_dai = { |
| .name = "snd-soc-dummy-dai", |
| .playback = { |
| .stream_name = "Playback", |
| .channels_min = 1, |
| .channels_max = 384, |
| .rates = STUB_RATES, |
| .formats = STUB_FORMATS, |
| }, |
| .capture = { |
| .stream_name = "Capture", |
| .channels_min = 1, |
| .channels_max = 384, |
| .rates = STUB_RATES, |
| .formats = STUB_FORMATS, |
| }, |
| .ops = &dummy_dai_ops, |
| }; |
| |
| int snd_soc_dai_is_dummy(struct snd_soc_dai *dai) |
| { |
| if (dai->driver == &dummy_dai) |
| return 1; |
| return 0; |
| } |
| |
| int snd_soc_component_is_dummy(struct snd_soc_component *component) |
| { |
| return ((component->driver == &dummy_platform) || |
| (component->driver == &dummy_codec)); |
| } |
| |
| static int snd_soc_dummy_probe(struct platform_device *pdev) |
| { |
| int ret; |
| |
| ret = devm_snd_soc_register_component(&pdev->dev, |
| &dummy_codec, &dummy_dai, 1); |
| if (ret < 0) |
| return ret; |
| |
| ret = devm_snd_soc_register_component(&pdev->dev, &dummy_platform, |
| NULL, 0); |
| |
| return ret; |
| } |
| |
| static struct platform_driver soc_dummy_driver = { |
| .driver = { |
| .name = "snd-soc-dummy", |
| }, |
| .probe = snd_soc_dummy_probe, |
| }; |
| |
| static struct platform_device *soc_dummy_dev; |
| |
| int __init snd_soc_util_init(void) |
| { |
| int ret; |
| |
| soc_dummy_dev = |
| platform_device_register_simple("snd-soc-dummy", -1, NULL, 0); |
| if (IS_ERR(soc_dummy_dev)) |
| return PTR_ERR(soc_dummy_dev); |
| |
| ret = platform_driver_register(&soc_dummy_driver); |
| if (ret != 0) |
| platform_device_unregister(soc_dummy_dev); |
| |
| return ret; |
| } |
| |
| void __exit snd_soc_util_exit(void) |
| { |
| platform_driver_unregister(&soc_dummy_driver); |
| platform_device_unregister(soc_dummy_dev); |
| } |