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https://github.com/xenia-project/FFmpeg.git
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c8af852b97
This is a new library for audio sample format, channel layout, and sample rate conversion.
71 lines
2.3 KiB
C
71 lines
2.3 KiB
C
/*
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* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
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*
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* This file is part of Libav.
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*
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* Libav 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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* Libav 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 Libav; 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 AVRESAMPLE_RESAMPLE_H
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#define AVRESAMPLE_RESAMPLE_H
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#include "avresample.h"
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#include "audio_data.h"
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typedef struct ResampleContext ResampleContext;
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/**
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* Allocate and initialize a ResampleContext.
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*
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* The parameters in the AVAudioResampleContext are used to initialize the
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* ResampleContext.
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*
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* @param avr AVAudioResampleContext
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* @return newly-allocated ResampleContext
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*/
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ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr);
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/**
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* Free a ResampleContext.
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*
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* @param c ResampleContext
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*/
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void ff_audio_resample_free(ResampleContext **c);
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/**
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* Resample audio data.
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*
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* Changes the sample rate.
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*
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* @par
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* All samples in the source data may not be consumed depending on the
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* resampling parameters and the size of the output buffer. The unconsumed
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* samples are automatically added to the start of the source in the next call.
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* If the destination data can be reallocated, that may be done in this function
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* in order to fit all available output. If it cannot be reallocated, fewer
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* input samples will be consumed in order to have the output fit in the
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* destination data buffers.
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*
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* @param c ResampleContext
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* @param dst destination audio data
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* @param src source audio data
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* @param consumed number of samples consumed from the source
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* @return number of samples written to the destination
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*/
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int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
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int *consumed);
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#endif /* AVRESAMPLE_RESAMPLE_H */
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