// 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/device/faux.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_ret(const struct device *dev, int ret, const char *fmt, ...)
{
struct va_format vaf;
va_list args;
/* Positive, Zero values are not errors */
if (ret >= 0 )
return ret;
/* Negative values might be errors */
switch (ret) {
case -EPROBE_DEFER:
case -ENOTSUPP:
case -EOPNOTSUPP:
break ;
default :
va_start(args, fmt);
vaf.fmt = fmt;
vaf.va = &args;
dev_err(dev, "ASoC error (%d): %pV" , ret, &vaf);
}
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_ret);
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(const 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(const 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(const 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 = 4096 ,
.period_bytes_max = 4096 *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 = snd_soc_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_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 const 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 = SNDRV_PCM_RATE_CONTINUOUS,
.rate_min = 5512 ,
.rate_max = 768000 ,
.formats = STUB_FORMATS,
},
.capture = {
.stream_name = "Capture" ,
.channels_min = 1 ,
.channels_max = 384 ,
.rates = SNDRV_PCM_RATE_CONTINUOUS,
.rate_min = 5512 ,
.rate_max = 768000 ,
.formats = STUB_FORMATS,
},
.ops = &dummy_dai_ops,
};
int snd_soc_dai_is_dummy(const 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 snd_soc_dummy_dlc = {
.of_node = NULL,
.dai_name = "snd-soc-dummy-dai" ,
.name = "snd-soc-dummy" ,
};
EXPORT_SYMBOL_GPL(snd_soc_dummy_dlc);
int snd_soc_dlc_is_dummy(struct snd_soc_dai_link_component *dlc)
{
if (dlc == &snd_soc_dummy_dlc)
return true ;
if ((dlc->name && strcmp(dlc->name, snd_soc_dummy_dlc.name) == 0 ) ||
(dlc->dai_name && strcmp(dlc->dai_name, snd_soc_dummy_dlc.dai_name) == 0 ))
return true ;
return false ;
}
EXPORT_SYMBOL_GPL(snd_soc_dlc_is_dummy);
static int snd_soc_dummy_probe(struct faux_device *fdev)
{
int ret;
ret = devm_snd_soc_register_component(&fdev->dev,
&dummy_codec, &dummy_dai, 1 );
if (ret < 0 )
return ret;
ret = devm_snd_soc_register_component(&fdev->dev, &dummy_platform,
NULL, 0 );
return ret;
}
static struct faux_device_ops soc_dummy_ops = {
.probe = snd_soc_dummy_probe,
};
static struct faux_device *soc_dummy_dev;
int __init snd_soc_util_init(void )
{
soc_dummy_dev = faux_device_create("snd-soc-dummy" , NULL,
&soc_dummy_ops);
if (!soc_dummy_dev)
return -ENODEV;
return 0 ;
}
void snd_soc_util_exit(void )
{
faux_device_destroy(soc_dummy_dev);
}
Messung V0.5 in Prozent C=95 H=100 G=97
¤ Dauer der Verarbeitung: 0.14 Sekunden
(vorverarbeitet am 2026-09-29)
¤
*© Formatika GbR, Deutschland