#include <alsa/asoundlib.h>
#include "config.h"
#include "libavformat/avformat.h"
Go to the source code of this file.
Data Structures | |
| struct | AlsaData |
Defines | |
| #define | DEFAULT_CODEC_ID CODEC_ID_PCM_S16LE |
Functions | |
| int | ff_alsa_open (AVFormatContext *s, snd_pcm_stream_t mode, unsigned int *sample_rate, int channels, enum CodecID *codec_id) |
| Opens an ALSA PCM. | |
| int | ff_alsa_close (AVFormatContext *s1) |
| Closes the ALSA PCM. | |
| int | ff_alsa_xrun_recover (AVFormatContext *s1, int err) |
| Tries to recover from ALSA buffer underrun. | |
Definition in file alsa-audio.h.
| #define DEFAULT_CODEC_ID CODEC_ID_PCM_S16LE |
| int ff_alsa_close | ( | AVFormatContext * | s1 | ) |
Closes the ALSA PCM.
| s1 | media file handle |
Definition at line 161 of file alsa-audio-common.c.
| int ff_alsa_open | ( | AVFormatContext * | s, | |
| snd_pcm_stream_t | mode, | |||
| unsigned int * | sample_rate, | |||
| int | channels, | |||
| enum CodecID * | codec_id | |||
| ) |
Opens an ALSA PCM.
| 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 CodecID or CODEC_ID_NONE; out: actually selected CodecID, changed only if CODEC_ID_NONE was requested |
Definition at line 46 of file alsa-audio-common.c.
Referenced by audio_read_header(), and audio_write_header().
| int ff_alsa_xrun_recover | ( | AVFormatContext * | s1, | |
| int | err | |||
| ) |
Tries to recover from ALSA buffer underrun.
| s1 | media file handle | |
| err | error code reported by the previous ALSA call |
Definition at line 169 of file alsa-audio-common.c.
Referenced by audio_read_packet(), and audio_write_packet().
1.5.8