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e37f161e66
* qatar/master: (71 commits) movenc: Allow writing to a non-seekable output if using empty moov movenc: Support adding isml (smooth streaming live) metadata libavcodec: Don't crash in avcodec_encode_audio if time_base isn't set sunrast: Document the different Sun Raster file format types. sunrast: Add a check for experimental type. libspeexenc: use AVSampleFormat instead of deprecated/removed SampleFormat lavf: remove disabled FF_API_SET_PTS_INFO cruft lavf: remove disabled FF_API_OLD_INTERRUPT_CB cruft lavf: remove disabled FF_API_REORDER_PRIVATE cruft lavf: remove disabled FF_API_SEEK_PUBLIC cruft lavf: remove disabled FF_API_STREAM_COPY cruft lavf: remove disabled FF_API_PRELOAD cruft lavf: remove disabled FF_API_NEW_STREAM cruft lavf: remove disabled FF_API_RTSP_URL_OPTIONS cruft lavf: remove disabled FF_API_MUXRATE cruft lavf: remove disabled FF_API_FILESIZE cruft lavf: remove disabled FF_API_TIMESTAMP cruft lavf: remove disabled FF_API_LOOP_OUTPUT cruft lavf: remove disabled FF_API_LOOP_INPUT cruft lavf: remove disabled FF_API_AVSTREAM_QUALITY cruft ... Conflicts: doc/APIchanges libavcodec/8bps.c libavcodec/avcodec.h libavcodec/libx264.c libavcodec/mjpegbdec.c libavcodec/options.c libavcodec/sunrast.c libavcodec/utils.c libavcodec/version.h libavcodec/x86/h264_deblock.asm libavdevice/libdc1394.c libavdevice/v4l2.c libavformat/avformat.h libavformat/avio.c libavformat/avio.h libavformat/aviobuf.c libavformat/dv.c libavformat/mov.c libavformat/utils.c libavformat/version.h libavformat/wtv.c libavutil/Makefile libavutil/file.c libswscale/x86/input.asm libswscale/x86/swscale_mmx.c libswscale/x86/swscale_template.c tests/ref/lavf/ffm Merged-by: Michael Niedermayer <michaelni@gmx.at>
80 lines
2.8 KiB
C
80 lines
2.8 KiB
C
/*
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* audio conversion
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* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
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* Copyright (c) 2008 Peter Ross
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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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#ifndef AVCODEC_AUDIOCONVERT_H
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#define AVCODEC_AUDIOCONVERT_H
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/**
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* @file
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* Audio format conversion routines
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*/
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#include "libavutil/cpu.h"
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#include "avcodec.h"
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#include "libavutil/audioconvert.h"
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/**
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* Guess the channel layout
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* @param nb_channels
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* @param codec_id Codec identifier, or CODEC_ID_NONE if unknown
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* @param fmt_name Format name, or NULL if unknown
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* @return Channel layout mask
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*/
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uint64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name);
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struct AVAudioConvert;
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typedef struct AVAudioConvert AVAudioConvert;
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/**
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* Create an audio sample format converter context
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* @param out_fmt Output sample format
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* @param out_channels Number of output channels
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* @param in_fmt Input sample format
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* @param in_channels Number of input channels
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* @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
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* @param flags See AV_CPU_FLAG_xx
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* @return NULL on error
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*/
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AVAudioConvert *av_audio_convert_alloc(enum AVSampleFormat out_fmt, int out_channels,
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enum AVSampleFormat in_fmt, int in_channels,
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const float *matrix, int flags);
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/**
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* Free audio sample format converter context
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*/
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void av_audio_convert_free(AVAudioConvert *ctx);
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/**
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* Convert between audio sample formats
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* @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
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* @param[in] out_stride distance between consecutive output samples (measured in bytes)
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* @param[in] in array of input buffers for each channel
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* @param[in] in_stride distance between consecutive input samples (measured in bytes)
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* @param len length of audio frame size (measured in samples)
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*/
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int av_audio_convert(AVAudioConvert *ctx,
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void * const out[6], const int out_stride[6],
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const void * const in[6], const int in_stride[6], int len);
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#endif /* AVCODEC_AUDIOCONVERT_H */
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