mirror of https://git.ffmpeg.org/ffmpeg.git
286 lines
11 KiB
C
286 lines
11 KiB
C
/*
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* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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*
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* This file is part of Libav.
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*
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* Libav 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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* Libav 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 Libav; 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_AVRESAMPLE_H
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#define AVRESAMPLE_AVRESAMPLE_H
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/**
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* @file
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* external API header
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*/
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#include "libavutil/audioconvert.h"
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#include "libavutil/avutil.h"
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#include "libavutil/dict.h"
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#include "libavutil/log.h"
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#include "libavresample/version.h"
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#define AVRESAMPLE_MAX_CHANNELS 32
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typedef struct AVAudioResampleContext AVAudioResampleContext;
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/** Mixing Coefficient Types */
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enum AVMixCoeffType {
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AV_MIX_COEFF_TYPE_Q8, /** 16-bit 8.8 fixed-point */
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AV_MIX_COEFF_TYPE_Q15, /** 32-bit 17.15 fixed-point */
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AV_MIX_COEFF_TYPE_FLT, /** floating-point */
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AV_MIX_COEFF_TYPE_NB, /** Number of coeff types. Not part of ABI */
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};
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/**
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* Return the LIBAVRESAMPLE_VERSION_INT constant.
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*/
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unsigned avresample_version(void);
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/**
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* Return the libavresample build-time configuration.
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* @return configure string
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*/
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const char *avresample_configuration(void);
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/**
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* Return the libavresample license.
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*/
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const char *avresample_license(void);
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/**
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* Get the AVClass for AVAudioResampleContext.
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*
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* Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options
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* without allocating a context.
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*
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* @see av_opt_find().
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*
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* @return AVClass for AVAudioResampleContext
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*/
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const AVClass *avresample_get_class(void);
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/**
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* Allocate AVAudioResampleContext and set options.
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*
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* @return allocated audio resample context, or NULL on failure
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*/
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AVAudioResampleContext *avresample_alloc_context(void);
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/**
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* Initialize AVAudioResampleContext.
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*
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* @param avr audio resample context
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* @return 0 on success, negative AVERROR code on failure
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*/
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int avresample_open(AVAudioResampleContext *avr);
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/**
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* Close AVAudioResampleContext.
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*
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* This closes the context, but it does not change the parameters. The context
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* can be reopened with avresample_open(). It does, however, clear the output
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* FIFO and any remaining leftover samples in the resampling delay buffer. If
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* there was a custom matrix being used, that is also cleared.
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*
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* @see avresample_convert()
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* @see avresample_set_matrix()
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*
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* @param avr audio resample context
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*/
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void avresample_close(AVAudioResampleContext *avr);
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/**
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* Free AVAudioResampleContext and associated AVOption values.
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*
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* This also calls avresample_close() before freeing.
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*
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* @param avr audio resample context
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*/
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void avresample_free(AVAudioResampleContext **avr);
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/**
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* Generate a channel mixing matrix.
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*
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* This function is the one used internally by libavresample for building the
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* default mixing matrix. It is made public just as a utility function for
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* building custom matrices.
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*
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* @param in_layout input channel layout
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* @param out_layout output channel layout
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* @param center_mix_level mix level for the center channel
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* @param surround_mix_level mix level for the surround channel(s)
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* @param lfe_mix_level mix level for the low-frequency effects channel
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* @param normalize if 1, coefficients will be normalized to prevent
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* overflow. if 0, coefficients will not be
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* normalized.
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* @param[out] matrix mixing coefficients; matrix[i + stride * o] is
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* the weight of input channel i in output channel o.
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* @param stride distance between adjacent input channels in the
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* matrix array
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* @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
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* @return 0 on success, negative AVERROR code on failure
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*/
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int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
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double center_mix_level, double surround_mix_level,
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double lfe_mix_level, int normalize, double *matrix,
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int stride, enum AVMatrixEncoding matrix_encoding);
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/**
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* Get the current channel mixing matrix.
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*
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* @param avr audio resample context
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* @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
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* input channel i in output channel o.
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* @param stride distance between adjacent input channels in the matrix array
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* @return 0 on success, negative AVERROR code on failure
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*/
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int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
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int stride);
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/**
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* Set channel mixing matrix.
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*
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* Allows for setting a custom mixing matrix, overriding the default matrix
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* generated internally during avresample_open(). This function can be called
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* anytime on an allocated context, either before or after calling
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* avresample_open(). avresample_convert() always uses the current matrix.
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* Calling avresample_close() on the context will clear the current matrix.
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*
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* @see avresample_close()
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*
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* @param avr audio resample context
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* @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
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* input channel i in output channel o.
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* @param stride distance between adjacent input channels in the matrix array
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* @return 0 on success, negative AVERROR code on failure
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*/
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int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
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int stride);
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/**
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* Set compensation for resampling.
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*
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* This can be called anytime after avresample_open(). If resampling was not
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* being done previously, the AVAudioResampleContext is closed and reopened
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* with resampling enabled. In this case, any samples remaining in the output
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* FIFO and the current channel mixing matrix will be restored after reopening
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* the context.
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*
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* @param avr audio resample context
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* @param sample_delta compensation delta, in samples
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* @param compensation_distance compensation distance, in samples
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* @return 0 on success, negative AVERROR code on failure
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*/
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int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
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int compensation_distance);
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/**
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* Convert input samples and write them to the output FIFO.
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*
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* The output data can be NULL or have fewer allocated samples than required.
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* In this case, any remaining samples not written to the output will be added
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* to an internal FIFO buffer, to be returned at the next call to this function
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* or to avresample_read().
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*
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* If converting sample rate, there may be data remaining in the internal
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* resampling delay buffer. avresample_get_delay() tells the number of remaining
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* samples. To get this data as output, call avresample_convert() with NULL
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* input.
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*
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* At the end of the conversion process, there may be data remaining in the
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* internal FIFO buffer. avresample_available() tells the number of remaining
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* samples. To get this data as output, either call avresample_convert() with
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* NULL input or call avresample_read().
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*
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* @see avresample_available()
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* @see avresample_read()
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* @see avresample_get_delay()
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*
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* @param avr audio resample context
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* @param output output data pointers
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* @param out_plane_size output plane size, in bytes.
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* This can be 0 if unknown, but that will lead to
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* optimized functions not being used directly on the
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* output, which could slow down some conversions.
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* @param out_samples maximum number of samples that the output buffer can hold
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* @param input input data pointers
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* @param in_plane_size input plane size, in bytes
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* This can be 0 if unknown, but that will lead to
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* optimized functions not being used directly on the
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* input, which could slow down some conversions.
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* @param in_samples number of input samples to convert
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* @return number of samples written to the output buffer,
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* not including converted samples added to the internal
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* output FIFO
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*/
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int avresample_convert(AVAudioResampleContext *avr, void **output,
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int out_plane_size, int out_samples, void **input,
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int in_plane_size, int in_samples);
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/**
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* Return the number of samples currently in the resampling delay buffer.
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*
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* When resampling, there may be a delay between the input and output. Any
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* unconverted samples in each call are stored internally in a delay buffer.
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* This function allows the user to determine the current number of samples in
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* the delay buffer, which can be useful for synchronization.
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*
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* @see avresample_convert()
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*
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* @param avr audio resample context
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* @return number of samples currently in the resampling delay buffer
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*/
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int avresample_get_delay(AVAudioResampleContext *avr);
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/**
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* Return the number of available samples in the output FIFO.
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*
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* During conversion, if the user does not specify an output buffer or
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* specifies an output buffer that is smaller than what is needed, remaining
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* samples that are not written to the output are stored to an internal FIFO
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* buffer. The samples in the FIFO can be read with avresample_read() or
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* avresample_convert().
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*
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* @see avresample_read()
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* @see avresample_convert()
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*
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* @param avr audio resample context
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* @return number of samples available for reading
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*/
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int avresample_available(AVAudioResampleContext *avr);
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/**
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* Read samples from the output FIFO.
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*
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* During conversion, if the user does not specify an output buffer or
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* specifies an output buffer that is smaller than what is needed, remaining
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* samples that are not written to the output are stored to an internal FIFO
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* buffer. This function can be used to read samples from that internal FIFO.
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*
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* @see avresample_available()
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* @see avresample_convert()
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*
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* @param avr audio resample context
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* @param output output data pointers. May be NULL, in which case
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* nb_samples of data is discarded from output FIFO.
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* @param nb_samples number of samples to read from the FIFO
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* @return the number of samples written to output
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*/
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int avresample_read(AVAudioResampleContext *avr, void **output, int nb_samples);
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#endif /* AVRESAMPLE_AVRESAMPLE_H */
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