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* 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
/*
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* ALSA input and output
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* Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
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* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* ALSA input and output: definitions and structures
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* @author Luca Abeni ( lucabe72 email it )
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* @author Benoit Fouet ( benoit fouet free fr )
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*/
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#ifndef AVDEVICE_ALSA_AUDIO_H
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#define AVDEVICE_ALSA_AUDIO_H
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#include <alsa/asoundlib.h>
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#include "config.h"
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#include "libavutil/log.h"
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#include "timefilter.h"
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#include "avdevice.h"
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/* XXX: we make the assumption that the soundcard accepts this format */
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/* XXX: find better solution with "preinit" method, needed also in
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other formats */
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#define DEFAULT_CODEC_ID AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE)
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typedef void (*ff_reorder_func)(const void *, void *, int);
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#define ALSA_BUFFER_SIZE_MAX 65536
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typedef struct {
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AVClass *class;
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snd_pcm_t *h;
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int frame_size; ///< bytes per sample * channels
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int period_size; ///< preferred size for reads and writes, in frames
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int sample_rate; ///< sample rate set by user
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int channels; ///< number of channels set by user
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TimeFilter *timefilter;
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void (*reorder_func)(const void *, void *, int);
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void *reorder_buf;
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int reorder_buf_size; ///< in frames
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} AlsaData;
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/**
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* Open an ALSA PCM.
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*
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* @param s media file handle
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* @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK
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* @param sample_rate in: requested sample rate;
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* out: actually selected sample rate
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* @param channels number of channels
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* @param codec_id in: requested CodecID or CODEC_ID_NONE;
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* out: actually selected CodecID, changed only if
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* CODEC_ID_NONE was requested
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*
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* @return 0 if OK, AVERROR_xxx on error
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*/
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int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode,
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unsigned int *sample_rate,
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int channels, enum CodecID *codec_id);
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/**
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* Close the ALSA PCM.
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*
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* @param s1 media file handle
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*
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* @return 0
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*/
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int ff_alsa_close(AVFormatContext *s1);
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/**
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* Try to recover from ALSA buffer underrun.
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*
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* @param s1 media file handle
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* @param err error code reported by the previous ALSA call
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*
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* @return 0 if OK, AVERROR_xxx on error
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*/
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int ff_alsa_xrun_recover(AVFormatContext *s1, int err);
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int ff_alsa_extend_reorder_buf(AlsaData *s, int size);
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#endif /* AVDEVICE_ALSA_AUDIO_H */
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