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/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
*
* 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 AVRESAMPLE_AUDIO_CONVERT_H
#define AVRESAMPLE_AUDIO_CONVERT_H
#include "libavutil/samplefmt.h"
#include "avresample.h"
#include "internal.h"
#include "audio_data.h"
/**
* Set conversion function if the parameters match.
*
* This compares the parameters of the conversion function to the parameters
* in the AudioConvert context. If the parameters do not match, no changes are
* made to the active functions. If the parameters do match and the alignment
* is not constrained, the function is set as the generic conversion function.
* If the parameters match and the alignment is constrained, the function is
* set as the optimized conversion function.
*
* @param ac AudioConvert context
* @param out_fmt output sample format
* @param in_fmt input sample format
* @param channels number of channels, or 0 for any number of channels
* @param ptr_align buffer pointer alignment, in bytes
* @param samples_align buffer size alignment, in samples
* @param descr function type description (e.g. "C" or "SSE")
* @param conv conversion function pointer
*/
void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt, int channels,
int ptr_align, int samples_align,
const char *descr, void *conv);
/**
* Allocate and initialize AudioConvert context for sample format conversion.
*
* @param avr AVAudioResampleContext
* @param out_fmt output sample format
* @param in_fmt input sample format
* @param channels number of channels
* @param sample_rate sample rate (used for dithering)
* @param apply_map apply channel map during conversion
* @return newly-allocated AudioConvert context
*/
AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels, int sample_rate,
int apply_map);
/**
* Free AudioConvert.
*
* The AudioConvert must have been previously allocated with ff_audio_convert_alloc().
*
* @param ac AudioConvert struct
*/
void ff_audio_convert_free(AudioConvert **ac);
/**
* Convert audio data from one sample format to another.
*
* For each call, the alignment of the input and output AudioData buffers are
* examined to determine whether to use the generic or optimized conversion
* function (when available).
*
* The number of samples to convert is determined by in->nb_samples. The output
* buffer must be large enough to handle this many samples. out->nb_samples is
* set by this function before a successful return.
*
* @param ac AudioConvert context
* @param out output audio data
* @param in input audio data
* @return 0 on success, negative AVERROR code on failure
*/
int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in);
/* arch-specific initialization functions */
void ff_audio_convert_init_aarch64(AudioConvert *ac);
void ff_audio_convert_init_arm(AudioConvert *ac);
void ff_audio_convert_init_x86(AudioConvert *ac);
#endif /* AVRESAMPLE_AUDIO_CONVERT_H */
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