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92 lines
3.2 KiB
C
92 lines
3.2 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 FFMPEG_AUDIOCONVERT_H
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#define FFMPEG_AUDIOCONVERT_H
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/**
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* @file audioconvert.h
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* Audio format conversion routines
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*/
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#include "avcodec.h"
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/**
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* Generate string corresponding to the sample format with
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* number sample_fmt, or a header if sample_fmt is negative.
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*
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* @param[in] buf the buffer where to write the string
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* @param[in] buf_size the size of buf
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* @param[in] sample_fmt the number of the sample format to print the corresponding info string, or
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* a negative value to print the corresponding header.
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* Meaningful values for obtaining a sample format info vary from 0 to SAMPLE_FMT_NB -1.
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*/
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void avcodec_sample_fmt_string(char *buf, int buf_size, int sample_fmt);
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/**
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* @return NULL on error
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*/
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const char *avcodec_get_sample_fmt_name(int sample_fmt);
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/**
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* @return SAMPLE_FMT_NONE on error
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*/
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enum SampleFormat avcodec_get_sample_fmt(const char* name);
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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 out_channels Number of output channels
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* @param in_fmt Input sample format
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* @param in_channels Number of input channels
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* @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
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* @param flags See FF_MM_xx
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* @return NULL on error
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*/
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AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
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enum SampleFormat in_fmt, int in_channels,
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const float *matrix, int flags);
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/**
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* Free audio sample format converter context
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*/
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void av_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] out_stride distance between consecutive input samples (measured in bytes)
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* @param[in] in array of input buffers for each channel
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* @param[in] in_stride distance between consecutive output samples (measured in bytes)
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* @param len length of audio frame size (measured in samples)
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*/
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int av_audio_convert(AVAudioConvert *ctx,
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void * const out[6], const int out_stride[6],
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const void * const in[6], const int in_stride[6], int len);
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#endif /* FFMPEG_AUDIOCONVERT_H */
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