From ace9429bb58fd418f0c81d4c2835699bddf6bde6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Thu, 11 Apr 2024 10:27:49 +0200 Subject: Adding upstream version 6.6.15. Signed-off-by: Daniel Baumann --- sound/soc/soc-utils.c | 279 ++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 279 insertions(+) create mode 100644 sound/soc/soc-utils.c (limited to 'sound/soc/soc-utils.c') diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c new file mode 100644 index 000000000..9c746e4ed --- /dev/null +++ b/sound/soc/soc-utils.c @@ -0,0 +1,279 @@ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-util.c -- ALSA SoC Audio Layer utility functions +// +// Copyright 2009 Wolfson Microelectronics PLC. +// +// Author: Mark Brown +// Liam Girdwood + +#include +#include +#include +#include +#include +#include +#include + +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, the tdm slot count and the + * tdm slot width. Optionally round-up the slot count to a given multiple. + * Either or both of tdm_width and tdm_slots can be 0. + * + * If tdm_width == 0: use params_width() as the slot width. + * If tdm_slots == 0: use params_channels() as the slot count. + * + * If slot_multiple > 1 the slot count (or params_channels() if tdm_slots == 0) + * will be rounded up to a multiple of slot_multiple. 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. + * + * If tdm_width == 0 && tdm_slots == 0 && slot_multiple < 2, this is equivalent + * to calling snd_soc_params_to_bclk(). + * + * @params: Pointer to struct_pcm_hw_params. + * @tdm_width: Width in bits of the tdm slots. Must be >= 0. + * @tdm_slots: Number of tdm slots per frame. Must be >= 0. + * @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, +}; + +#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; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_is_dummy); + +int snd_soc_component_is_dummy(struct snd_soc_component *component) +{ + return ((component->driver == &dummy_platform) || + (component->driver == &dummy_codec)); +} + +struct snd_soc_dai_link_component asoc_dummy_dlc = { + .of_node = NULL, + .dai_name = "snd-soc-dummy-dai", + .name = "snd-soc-dummy", +}; +EXPORT_SYMBOL_GPL(asoc_dummy_dlc); + +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 snd_soc_util_exit(void) +{ + platform_driver_unregister(&soc_dummy_driver); + platform_device_unregister(soc_dummy_dev); +} -- cgit v1.2.3