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/*
* 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"
typedef void (conv_func_type)(uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end);
typedef void (simd_func_type)(uint8_t **dst, const uint8_t **src, int len);
typedef struct AudioConvert {
int channels;
conv_func_type *conv_f;
simd_func_type *simd_f;
const int *ch_map;
uint8_t silence[8]; ///< silence input sample
}AudioConvert;
/**
* 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
* @param ch_map list of the channels id to pick from the source stream, NULL
* if all channels must be selected
* @return NULL on error
*/
AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels, const int *ch_map,
int flags);
/**
* Free audio sample format converter context.
* and set the pointer to NULL
*/
void swri_audio_convert_free(AudioConvert **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 swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len);
#endif /* AUDIOCONVERT_H */
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