040ad30bc8
* release/0.7: (296 commits) Update version numbers for 0.7.5 vp6: partially propagate huffman tree building errors during coeff model parsing and fix misspelling Check for huffman tree building error in vp6 decoder. Release old pictures after a resolution change in vp5/6 decoder Check for missing reference in vp5/6 decoder. Check for invalid slices offsets in RV30/40 decoder. Check output buffer size in nellymoser decoder. Hack around gcc 4.6 breaking asm using call. Hack around gcc 4.6 breaking asm using call. Fix dxva2 decoding for some H264 samples. (cherry picked from commit bf7dc6b29d785f149f18c39db021413e08735546) Fix dxva2 decoding for some H264 samples. mp3demux: pass on error code on packet read. Check for invalid slice offsets in real decoder. rmdec: Reject invalid deinterleaving parameters Use deinterleavers for demangling audio packets in RealMedia. rv10: Reject slices that does not have the same type as the first one rmdec: use the deinterleaving mode and not the codec when creating audio packets. MAINTAINERS: add my GPG fingerprint. (cherry picked from commit 7882dc10f871bf25a848fe62a152f63814f9c7d1) Support 3IVD in isom, produced by 3ivx DivX Doctor. mpegpsdec: fix reading first mpegps packet (cherry picked from commit b2f230e23dd61112ac090b0c059d87b5f6bcb307) ... Conflicts: Changelog Doxyfile Makefile RELEASE configure doc/general.texi ffmpeg.c ffplay.c libavcodec/dxva2_h264.c libavcodec/h264.c libavcodec/h264_loopfilter.c libavcodec/h264idct_template.c libavcodec/kgv1dec.c libavcodec/mpegvideo.c libavcodec/tableprint.h libavcodec/vp3.c libavdevice/alsa-audio.h libavformat/gxf.c libavformat/mpegts.c libavformat/segafilm.c libavformat/utils.c libavutil/dict.h 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 "libavformat/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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