forked from KolibriOS/kolibrios
97 lines
3.1 KiB
C
97 lines
3.1 KiB
C
/*
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* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#ifndef AVRESAMPLE_RESAMPLE_H
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#define AVRESAMPLE_RESAMPLE_H
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#include "avresample.h"
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#include "internal.h"
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#include "audio_data.h"
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struct ResampleContext {
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AVAudioResampleContext *avr;
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AudioData *buffer;
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uint8_t *filter_bank;
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int filter_length;
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int ideal_dst_incr;
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int dst_incr;
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unsigned int index;
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int frac;
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int src_incr;
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int compensation_distance;
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int phase_shift;
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int phase_mask;
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int linear;
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enum AVResampleFilterType filter_type;
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int kaiser_beta;
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void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
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void (*resample_one)(struct ResampleContext *c, void *dst0,
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int dst_index, const void *src0,
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unsigned int index, int frac);
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void (*resample_nearest)(void *dst0, int dst_index,
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const void *src0, unsigned int index);
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int padding_size;
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int initial_padding_filled;
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int initial_padding_samples;
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int final_padding_filled;
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int final_padding_samples;
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};
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/**
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* Allocate and initialize a ResampleContext.
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*
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* The parameters in the AVAudioResampleContext are used to initialize the
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* ResampleContext.
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*
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* @param avr AVAudioResampleContext
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* @return newly-allocated ResampleContext
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*/
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ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr);
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/**
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* Free a ResampleContext.
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*
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* @param c ResampleContext
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*/
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void ff_audio_resample_free(ResampleContext **c);
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/**
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* Resample audio data.
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*
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* Changes the sample rate.
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*
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* @par
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* All samples in the source data may not be consumed depending on the
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* resampling parameters and the size of the output buffer. The unconsumed
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* samples are automatically added to the start of the source in the next call.
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* If the destination data can be reallocated, that may be done in this function
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* in order to fit all available output. If it cannot be reallocated, fewer
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* input samples will be consumed in order to have the output fit in the
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* destination data buffers.
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*
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* @param c ResampleContext
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* @param dst destination audio data
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* @param src source audio data
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* @return 0 on success, negative AVERROR code on failure
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*/
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int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src);
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#endif /* AVRESAMPLE_RESAMPLE_H */
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