linux/sound/soc/soc-utils.c

// SPDX-License-Identifier: GPL-2.0+
//
// soc-util.c  --  ALSA SoC Audio Layer utility functions
//
// Copyright 2009 Wolfson Microelectronics PLC.
//
// Author: Mark Brown <[email protected]>
//         Liam Girdwood <[email protected]>

#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)
{ … }
EXPORT_SYMBOL_GPL(…);

int snd_soc_params_to_frame_size(const struct snd_pcm_hw_params *params)
{ … }
EXPORT_SYMBOL_GPL(…);

int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots)
{ … }
EXPORT_SYMBOL_GPL(…);

int snd_soc_params_to_bclk(const struct snd_pcm_hw_params *params)
{ … }
EXPORT_SYMBOL_GPL(…);

/**
 * 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(const struct snd_pcm_hw_params *params,
			       int tdm_width, int tdm_slots, int slot_multiple)
{ … }
EXPORT_SYMBOL_GPL(…);

static const struct snd_pcm_hardware dummy_dma_hardware = …;


static const struct snd_soc_component_driver dummy_platform;

static int dummy_dma_open(struct snd_soc_component *component,
			  struct snd_pcm_substream *substream)
{ … }

static const struct snd_soc_component_driver dummy_platform = …;

static const struct snd_soc_component_driver dummy_codec = …;

#define STUB_FORMATS …

/*
 * 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 const u64 dummy_dai_formats = …;

static const struct snd_soc_dai_ops dummy_dai_ops = …;

/*
 * 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 = …;

int snd_soc_dai_is_dummy(const struct snd_soc_dai *dai)
{ … }
EXPORT_SYMBOL_GPL(…);

int snd_soc_component_is_dummy(struct snd_soc_component *component)
{ … }

struct snd_soc_dai_link_component snd_soc_dummy_dlc = …;
EXPORT_SYMBOL_GPL(…);

static int snd_soc_dummy_probe(struct platform_device *pdev)
{ … }

static struct platform_driver soc_dummy_driver = …;

static struct platform_device *soc_dummy_dev;

int __init snd_soc_util_init(void)
{ … }

void snd_soc_util_exit(void)
{ … }