Contributors: 16
| Author |
Tokens |
Token Proportion |
Commits |
Commit Proportion |
| Mark Brown |
382 |
45.10% |
8 |
21.05% |
| Kuninori Morimoto |
162 |
19.13% |
11 |
28.95% |
| Liam Girdwood |
84 |
9.92% |
4 |
10.53% |
| Richard Fitzgerald |
79 |
9.33% |
2 |
5.26% |
| Stas Sergeev |
61 |
7.20% |
1 |
2.63% |
| Jerome Brunet |
22 |
2.60% |
1 |
2.63% |
| Amadeusz Sławiński |
21 |
2.48% |
1 |
2.63% |
| Richard Purdie |
8 |
0.94% |
1 |
2.63% |
| Uwe Kleine-König |
8 |
0.94% |
1 |
2.63% |
| Krzysztof Kozlowski |
5 |
0.59% |
2 |
5.26% |
| Sameer Pujar |
5 |
0.59% |
1 |
2.63% |
| Frank Mandarino |
3 |
0.35% |
1 |
2.63% |
| Paul Gortmaker |
3 |
0.35% |
1 |
2.63% |
| Julia Lawall |
2 |
0.24% |
1 |
2.63% |
| Anatol Pomozov |
1 |
0.12% |
1 |
2.63% |
| Rohit kumar |
1 |
0.12% |
1 |
2.63% |
| Total |
847 |
|
38 |
|
// 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(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 = 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 = 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);
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);
}