ffmpeg/libavcodec/audioconvert.h
Michael Niedermayer e37f161e66 Merge remote-tracking branch 'qatar/master'
* 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>
2012-01-28 07:53:34 +01:00

80 lines
2.8 KiB
C

/*
* audio conversion
* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
* Copyright (c) 2008 Peter Ross
*
* 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
*/
#ifndef AVCODEC_AUDIOCONVERT_H
#define AVCODEC_AUDIOCONVERT_H
/**
* @file
* Audio format conversion routines
*/
#include "libavutil/cpu.h"
#include "avcodec.h"
#include "libavutil/audioconvert.h"
/**
* Guess the channel layout
* @param nb_channels
* @param codec_id Codec identifier, or CODEC_ID_NONE if unknown
* @param fmt_name Format name, or NULL if unknown
* @return Channel layout mask
*/
uint64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name);
struct AVAudioConvert;
typedef struct AVAudioConvert AVAudioConvert;
/**
* Create an audio sample format converter context
* @param out_fmt Output sample format
* @param out_channels Number of output channels
* @param in_fmt Input sample format
* @param in_channels Number of input channels
* @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
* @param flags See AV_CPU_FLAG_xx
* @return NULL on error
*/
AVAudioConvert *av_audio_convert_alloc(enum AVSampleFormat out_fmt, int out_channels,
enum AVSampleFormat in_fmt, int in_channels,
const float *matrix, int flags);
/**
* Free audio sample format converter context
*/
void av_audio_convert_free(AVAudioConvert *ctx);
/**
* Convert between audio sample formats
* @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
* @param[in] out_stride distance between consecutive output samples (measured in bytes)
* @param[in] in array of input buffers for each channel
* @param[in] in_stride distance between consecutive input samples (measured in bytes)
* @param len length of audio frame size (measured in samples)
*/
int av_audio_convert(AVAudioConvert *ctx,
void * const out[6], const int out_stride[6],
const void * const in[6], const int in_stride[6], int len);
#endif /* AVCODEC_AUDIOCONVERT_H */