/* * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> * * 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_RESAMPLE_H#define AVRESAMPLE_RESAMPLE_H#include "avresample.h"#include "internal.h"#include "audio_data.h"struct ResampleContext { AVAudioResampleContext *avr; AudioData *buffer; uint8_t *filter_bank; int filter_length; int ideal_dst_incr; int dst_incr; unsigned int index; int frac; int src_incr; int compensation_distance; int phase_shift; int phase_mask; int linear; enum AVResampleFilterType filter_type; int kaiser_beta; void (*set_filter)(void *filter, double *tab, int phase, int tap_count); void (*resample_one)(struct ResampleContext *c, void *dst0, int dst_index, const void *src0, unsigned int index, int frac); void (*resample_nearest)(void *dst0, int dst_index, const void *src0, unsigned int index); int padding_size; int initial_padding_filled; int initial_padding_samples; int final_padding_filled; int final_padding_samples;};/** * Allocate and initialize a ResampleContext. * * The parameters in the AVAudioResampleContext are used to initialize the * ResampleContext. * * @param avr AVAudioResampleContext * @return newly-allocated ResampleContext */ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr);/** * Free a ResampleContext. * * @param c ResampleContext */void ff_audio_resample_free(ResampleContext **c);/** * Resample audio data. * * Changes the sample rate. * * @par * All samples in the source data may not be consumed depending on the * resampling parameters and the size of the output buffer. The unconsumed * samples are automatically added to the start of the source in the next call. * If the destination data can be reallocated, that may be done in this function * in order to fit all available output. If it cannot be reallocated, fewer * input samples will be consumed in order to have the output fit in the * destination data buffers. * * @param c ResampleContext * @param dst destination audio data * @param src source audio data * @return 0 on success, negative AVERROR code on failure */int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src);#endif /* AVRESAMPLE_RESAMPLE_H */
/*
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
*
* 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_RESAMPLE_H
#define AVRESAMPLE_RESAMPLE_H
#include "avresample.h"
#include "internal.h"
#include "audio_data.h"
struct ResampleContext {
AVAudioResampleContext *avr;
AudioData *buffer;
uint8_t *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
unsigned int index;
int frac;
int src_incr;
int compensation_distance;
int phase_shift;
int phase_mask;
int linear;
enum AVResampleFilterType filter_type;
int kaiser_beta;
void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
void (*resample_one)(struct ResampleContext *c, void *dst0,
int dst_index, const void *src0,
unsigned int index, int frac);
void (*resample_nearest)(void *dst0, int dst_index,
const void *src0, unsigned int index);
int padding_size;
int initial_padding_filled;
int initial_padding_samples;
int final_padding_filled;
int final_padding_samples;
};
/**
* Allocate and initialize a ResampleContext.
*
* The parameters in the AVAudioResampleContext are used to initialize the
* ResampleContext.
*
* @param avr AVAudioResampleContext
* @return newly-allocated ResampleContext
*/
ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr);
/**
* Free a ResampleContext.
*
* @param c ResampleContext
*/
void ff_audio_resample_free(ResampleContext **c);
/**
* Resample audio data.
*
* Changes the sample rate.
*
* @par
* All samples in the source data may not be consumed depending on the
* resampling parameters and the size of the output buffer. The unconsumed
* samples are automatically added to the start of the source in the next call.
* If the destination data can be reallocated, that may be done in this function
* in order to fit all available output. If it cannot be reallocated, fewer
* input samples will be consumed in order to have the output fit in the
* destination data buffers.
*
* @param c ResampleContext
* @param dst destination audio data
* @param src source audio data
* @return 0 on success, negative AVERROR code on failure
*/
int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src);
#endif /* AVRESAMPLE_RESAMPLE_H */