1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-21 10:55:51 +02:00
FFmpeg/libavresample/resample.h
Justin Ruggles c8af852b97 Add libavresample
This is a new library for audio sample format, channel layout, and sample rate
conversion.
2012-04-24 21:28:27 -04:00

71 lines
2.3 KiB
C

/*
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVRESAMPLE_RESAMPLE_H
#define AVRESAMPLE_RESAMPLE_H
#include "avresample.h"
#include "audio_data.h"
typedef struct ResampleContext ResampleContext;
/**
* Allocate and initialize a ResampleContext.
*
* The parameters in the AVAudioResampleContext are used to initialize the
* ResampleContext.
*
* @param avr AVAudioResampleContext
* @return newly-allocated ResampleContext
*/
ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr);
/**
* Free a ResampleContext.
*
* @param c ResampleContext
*/
void ff_audio_resample_free(ResampleContext **c);
/**
* Resample audio data.
*
* Changes the sample rate.
*
* @par
* All samples in the source data may not be consumed depending on the
* resampling parameters and the size of the output buffer. The unconsumed
* samples are automatically added to the start of the source in the next call.
* If the destination data can be reallocated, that may be done in this function
* in order to fit all available output. If it cannot be reallocated, fewer
* input samples will be consumed in order to have the output fit in the
* destination data buffers.
*
* @param c ResampleContext
* @param dst destination audio data
* @param src source audio data
* @param consumed number of samples consumed from the source
* @return number of samples written to the destination
*/
int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
int *consumed);
#endif /* AVRESAMPLE_RESAMPLE_H */