mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
5d35b279e2
The PTS for captured audio was measured using snd_pcm_htimestamp. snd_pcm_htimestamp hangs when the input is a dsnoop plugin. Furthermore, at some point, snd_pcm_htimestamp started returning monotonic timestamps rather than wall clock timestamps, in most but not all situations. Monotonic timestamps are fine, but ffmpeg uses wall clock timestamps everywhere else, and we have no API to inform the user which kind of timestamps it is. A separate snd_pcm_htimestamp is only slightly less accurate than snd_pcm_htimestamp: the standard deviation for the difference between two consecutive timestamps is (on my hardware): - ~13 µs with snd_pcm_htimestamp; - ~35 µs with av_gettime; - ~5 µs with av_gettime and a timefilter.
99 lines
3.1 KiB
C
99 lines
3.1 KiB
C
/*
|
|
* ALSA input and output
|
|
* Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
|
|
* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
|
|
*
|
|
* This file is part of FFmpeg.
|
|
*
|
|
* FFmpeg is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Lesser General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2.1 of the License, or (at your option) any later version.
|
|
*
|
|
* FFmpeg is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Lesser General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Lesser General Public
|
|
* License along with FFmpeg; if not, write to the Free Software
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
|
*/
|
|
|
|
/**
|
|
* @file
|
|
* ALSA input and output: definitions and structures
|
|
* @author Luca Abeni ( lucabe72 email it )
|
|
* @author Benoit Fouet ( benoit fouet free fr )
|
|
*/
|
|
|
|
#ifndef AVDEVICE_ALSA_AUDIO_H
|
|
#define AVDEVICE_ALSA_AUDIO_H
|
|
|
|
#include <alsa/asoundlib.h>
|
|
#include "config.h"
|
|
#include "libavutil/log.h"
|
|
#include "libavformat/timefilter.h"
|
|
#include "avdevice.h"
|
|
|
|
/* XXX: we make the assumption that the soundcard accepts this format */
|
|
/* XXX: find better solution with "preinit" method, needed also in
|
|
other formats */
|
|
#define DEFAULT_CODEC_ID AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE)
|
|
|
|
typedef void (*ff_reorder_func)(const void *, void *, int);
|
|
|
|
typedef struct {
|
|
AVClass *class;
|
|
snd_pcm_t *h;
|
|
int frame_size; ///< bytes per sample * channels
|
|
int period_size; ///< preferred size for reads and writes, in frames
|
|
int sample_rate; ///< sample rate set by user
|
|
int channels; ///< number of channels set by user
|
|
TimeFilter *timefilter;
|
|
void (*reorder_func)(const void *, void *, int);
|
|
void *reorder_buf;
|
|
int reorder_buf_size; ///< in frames
|
|
} AlsaData;
|
|
|
|
/**
|
|
* Open an ALSA PCM.
|
|
*
|
|
* @param s media file handle
|
|
* @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK
|
|
* @param sample_rate in: requested sample rate;
|
|
* out: actually selected sample rate
|
|
* @param channels number of channels
|
|
* @param codec_id in: requested CodecID or CODEC_ID_NONE;
|
|
* out: actually selected CodecID, changed only if
|
|
* CODEC_ID_NONE was requested
|
|
*
|
|
* @return 0 if OK, AVERROR_xxx on error
|
|
*/
|
|
int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode,
|
|
unsigned int *sample_rate,
|
|
int channels, enum CodecID *codec_id);
|
|
|
|
/**
|
|
* Close the ALSA PCM.
|
|
*
|
|
* @param s1 media file handle
|
|
*
|
|
* @return 0
|
|
*/
|
|
int ff_alsa_close(AVFormatContext *s1);
|
|
|
|
/**
|
|
* Try to recover from ALSA buffer underrun.
|
|
*
|
|
* @param s1 media file handle
|
|
* @param err error code reported by the previous ALSA call
|
|
*
|
|
* @return 0 if OK, AVERROR_xxx on error
|
|
*/
|
|
int ff_alsa_xrun_recover(AVFormatContext *s1, int err);
|
|
|
|
int ff_alsa_extend_reorder_buf(AlsaData *s, int size);
|
|
|
|
#endif /* AVDEVICE_ALSA_AUDIO_H */
|