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https://github.com/FFmpeg/FFmpeg.git
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b5875b9111
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>
66 lines
2.1 KiB
C
66 lines
2.1 KiB
C
/*
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* audio conversion
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* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
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* Copyright (c) 2008 Peter Ross
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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 SWR_AUDIOCONVERT_H
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#define SWR_AUDIOCONVERT_H
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/**
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* @file
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* Audio format conversion routines
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*/
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#include "swresample_internal.h"
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#include "libavutil/cpu.h"
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#include "libavutil/audioconvert.h"
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struct AVAudioConvert;
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typedef struct AVAudioConvert AVAudioConvert;
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/**
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* Create an audio sample format converter context
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* @param out_fmt Output sample format
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* @param in_fmt Input sample format
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* @param channels Number of channels
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* @param flags See AV_CPU_FLAG_xx
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* @return NULL on error
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*/
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AVAudioConvert *swr_audio_convert_alloc(enum AVSampleFormat out_fmt,
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enum AVSampleFormat in_fmt,
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int channels, int flags);
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/**
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* Free audio sample format converter context.
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* and set the pointer to NULL
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*/
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void swr_audio_convert_free(AVAudioConvert **ctx);
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/**
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* Convert between audio sample formats
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* @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
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* @param[in] in array of input buffers for each channel
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* @param len length of audio frame size (measured in samples)
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*/
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int swr_audio_convert(AVAudioConvert *ctx, AudioData *out, AudioData *in, int len);
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#endif /* AVCODEC_AUDIOCONVERT_H */
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