#include <alsa/asoundlib.h>
#include "avdevice.h"
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "alsa.h"
Go to the source code of this file.
ALSA input and output: common code
- Author
- Luca Abeni ( lucabe72 email it )
-
Benoit Fouet ( benoit fouet free fr )
-
Nicolas George ( nicolas george normalesup org )
Definition in file alsa.c.
◆ MAKE_REORDER_FUNC
#define MAKE_REORDER_FUNC |
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NAME, |
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TYPE, |
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CHANNELS, |
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LAYOUT, |
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MAP |
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Value:static void alsa_reorder_ ## NAME ##
_ ## LAYOUT(
const void *in_v, \
void *out_v, \
{ \
\
MAP \
in += CHANNELS; \
out += CHANNELS; \
} \
}
Definition at line 65 of file alsa.c.
◆ MAKE_REORDER_FUNCS
#define MAKE_REORDER_FUNCS |
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CHANNELS, |
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LAYOUT, |
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Value:
MAKE_REORDER_FUNC(int16, int16_t, CHANNELS, LAYOUT,
MAP) \
MAKE_REORDER_FUNC(int32,
int32_t, CHANNELS, LAYOUT,
MAP) \
MAKE_REORDER_FUNC(f32,
float, CHANNELS, LAYOUT,
MAP)
Definition at line 80 of file alsa.c.
◆ FORMAT_I8
◆ FORMAT_I16
◆ FORMAT_I32
◆ FORMAT_F32
◆ PICK_REORDER
#define PICK_REORDER |
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layout | ) |
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◆ codec_id_to_pcm_format()
static av_cold snd_pcm_format_t codec_id_to_pcm_format |
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int |
codec_id | ) |
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◆ MAKE_REORDER_FUNCS()
◆ ff_alsa_open()
Open an ALSA PCM.
- Parameters
-
s | media file handle |
mode | either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK |
sample_rate | in: requested sample rate; out: actually selected sample rate |
channels | number of channels |
codec_id | in: requested AVCodecID or AV_CODEC_ID_NONE; out: actually selected AVCodecID, changed only if AV_CODEC_ID_NONE was requested |
- Returns
- 0 if OK, AVERROR_xxx on error
Definition at line 167 of file alsa.c.
Referenced by audio_read_header(), and audio_write_header().
◆ ff_alsa_close()
Close the ALSA PCM.
- Parameters
-
- Returns
- 0
Definition at line 299 of file alsa.c.
◆ ff_alsa_xrun_recover()
Try to recover from ALSA buffer underrun.
- Parameters
-
s1 | media file handle |
err | error code reported by the previous ALSA call |
- Returns
- 0 if OK, AVERROR_xxx on error
Definition at line 314 of file alsa.c.
Referenced by audio_read_packet(), and audio_write_packet().
◆ ff_alsa_extend_reorder_buf()
◆ ff_alsa_get_device_list()
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in