mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-02 03:06:28 +02:00
aedc908601
* qatar/master: (35 commits) flvdec: Do not call parse_keyframes_index with a NULL stream libspeexdec: include system headers before local headers libspeexdec: return meaningful error codes libspeexdec: cosmetics: reindent libspeexdec: decode one frame at a time. swscale: fix signed shift overflows in ff_yuv2rgb_c_init_tables() Move timefilter code from lavf to lavd. mov: add support for hdvd and pgapmetadata atoms mov: rename function _stik, some indentation cosmetics mov: rename function _int8 to remove ambiguity, some indentation cosmetics mov: parse the gnre atom mp3on4: check for allocation failures in decode_init_mp3on4() mp3on4: create a separate flush function for MP3onMP4. mp3on4: ensure that the frame channel count does not exceed the codec channel count. mp3on4: set channel layout mp3on4: fix the output channel order mp3on4: allocate temp buffer with av_malloc() instead of on the stack. mp3on4: copy MPADSPContext from first context to all contexts. fmtconvert: port float_to_int16_interleave() 2-channel x86 inline asm to yasm fmtconvert: port int32_to_float_fmul_scalar() x86 inline asm to yasm ... Conflicts: libavcodec/arm/h264dsp_init_arm.c libavcodec/h264.c libavcodec/h264.h libavcodec/h264_cabac.c libavcodec/h264_cavlc.c libavcodec/h264_ps.c libavcodec/h264dsp_template.c libavcodec/h264idct_template.c libavcodec/h264pred.c libavcodec/h264pred_template.c libavcodec/x86/h264dsp_mmx.c libavdevice/Makefile libavdevice/jack_audio.c libavformat/Makefile libavformat/flvdec.c libavformat/flvenc.c libavutil/pixfmt.h libswscale/utils.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
101 lines
3.1 KiB
C
101 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 "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);
|
|
|
|
#define ALSA_BUFFER_SIZE_MAX 65536
|
|
|
|
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 */
|