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FFmpeg/libswresample/audioconvert.h
Michael Niedermayer b5875b9111 Add libswresample.
Similar to libswscale this does resampling and format convertion, just for audio
instead of video.
changing sampling rate, sample formats, channel layouts and sample packing all
in one with a very simple public interface.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-19 07:04:17 +02:00

66 lines
2.1 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 SWR_AUDIOCONVERT_H
#define SWR_AUDIOCONVERT_H
/**
* @file
* Audio format conversion routines
*/
#include "swresample_internal.h"
#include "libavutil/cpu.h"
#include "libavutil/audioconvert.h"
struct AVAudioConvert;
typedef struct AVAudioConvert AVAudioConvert;
/**
* Create an audio sample format converter context
* @param out_fmt Output sample format
* @param in_fmt Input sample format
* @param channels Number of channels
* @param flags See AV_CPU_FLAG_xx
* @return NULL on error
*/
AVAudioConvert *swr_audio_convert_alloc(enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels, int flags);
/**
* Free audio sample format converter context.
* and set the pointer to NULL
*/
void swr_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] in array of input buffers for each channel
* @param len length of audio frame size (measured in samples)
*/
int swr_audio_convert(AVAudioConvert *ctx, AudioData *out, AudioData *in, int len);
#endif /* AVCODEC_AUDIOCONVERT_H */