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// 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 <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);
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);
}
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