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ec7f0b527c
* khirnov/release/0.7: (64 commits) rv34: Check for invalid slice offsets rv34: Fix potential overreads rv34: Avoid NULL dereference on corrupted bitstream rv10: Reject slices that does not have the same type as the first one lavf: Fix context pointer in av_open_input_stream when avformat_open_input fails oggdec: fix out of bound write in the ogg demuxer Fixed size given to init_get_bits(). smacker: fix a few off by 1 errors Check for invalid VLC value in smacker decoder. Check and propagate errors when VLC trees cannot be built in smacker decoder. Fixed off by one packet size allocation in the smacker demuxer. Check for invalid packet size in the smacker demuxer. ape demuxer: fix segfault on memory allocation failure. xan: Add some buffer checks (cherry picked from commit0872bb23b4) Fixed size given to init_get_bits() in xan decoder. (cherry picked from commit393d5031c6) smacker demuxer: handle possible av_realloc() failure. Fixed segfault with wavpack decoder on corrupted decorrelation terms sub-blocks. cljr: init_get_bits size in bits instead of bytes (cherry picked from commit0c1f5b93d9) indeo2: fail if input buffer too small (cherry picked from commitb7ce4f1d1c) indeo2: init_get_bits size in bits instead of bytes (cherry picked from commit68ca330cbd) ... Conflicts: ffmpeg.c libavdevice/alsa-audio.h libavformat/gxf.c libswscale/x86/swscale_template.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
99 lines
3.0 KiB
C
99 lines
3.0 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 "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; ///< preferred size for reads and writes
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int period_size; ///< bytes per sample * channels
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ff_reorder_func reorder_func;
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void *reorder_buf;
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int reorder_buf_size; ///< 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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} 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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