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/*
* 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 FFMPEG_AUDIOCONVERT_H
#define FFMPEG_AUDIOCONVERT_H
/**
* @file audioconvert.h
* Audio format conversion routines
*/
#include "avcodec.h"
/**
* Generate string corresponding to the sample format with
* number sample_fmt, or a header if sample_fmt is negative.
*
* @param[in] buf the buffer where to write the string
* @param[in] buf_size the size of buf
* @param[in] sample_fmt the number of the sample format to print the corresponding info string, or
* a negative value to print the corresponding header.
* Meaningful values for obtaining a sample format info vary from 0 to SAMPLE_FMT_NB -1.
*/
void avcodec_sample_fmt_string(char *buf, int buf_size, int sample_fmt);
/**
* @return NULL on error
*/
const char *avcodec_get_sample_fmt_name(int sample_fmt);
/**
* @return SAMPLE_FMT_NONE on error
*/
enum SampleFormat avcodec_get_sample_fmt(const char* 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 FF_MM_xx
* @return NULL on error
*/
AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
enum SampleFormat 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 input samples (measured in bytes)
* @param[in] in array of input buffers for each channel
* @param[in] in_stride distance between consecutive output 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 /* FFMPEG_AUDIOCONVERT_H */
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