mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-29 10:52:20 +00:00
97 lines
3.1 KiB
C
97 lines
3.1 KiB
C
/*
|
|
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
|
|
*
|
|
* This file is part of Libav.
|
|
*
|
|
* Libav 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.
|
|
*
|
|
* Libav 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 Libav; 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 */
|