/* * Copyright (c) 2004 Michael Niedermayer * * 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 */