580 lines
21 KiB
C
580 lines
21 KiB
C
/*
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* Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
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*
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* This file is part of libswresample
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*
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* libswresample 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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* libswresample 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 libswresample; 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 SWRESAMPLE_SWRESAMPLE_H
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#define SWRESAMPLE_SWRESAMPLE_H
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/**
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* @file
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* @ingroup lswr
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* libswresample public header
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*/
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/**
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* @defgroup lswr libswresample
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* @{
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*
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* Audio resampling, sample format conversion and mixing library.
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*
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* Interaction with lswr is done through SwrContext, which is
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* allocated with swr_alloc() or swr_alloc_set_opts(). It is opaque, so all parameters
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* must be set with the @ref avoptions API.
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*
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* The first thing you will need to do in order to use lswr is to allocate
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* SwrContext. This can be done with swr_alloc() or swr_alloc_set_opts(). If you
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* are using the former, you must set options through the @ref avoptions API.
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* The latter function provides the same feature, but it allows you to set some
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* common options in the same statement.
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*
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* For example the following code will setup conversion from planar float sample
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* format to interleaved signed 16-bit integer, downsampling from 48kHz to
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* 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing
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* matrix). This is using the swr_alloc() function.
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* @code
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* SwrContext *swr = swr_alloc();
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* av_opt_set_channel_layout(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
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* av_opt_set_channel_layout(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
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* av_opt_set_int(swr, "in_sample_rate", 48000, 0);
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* av_opt_set_int(swr, "out_sample_rate", 44100, 0);
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* av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
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* av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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* @endcode
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*
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* The same job can be done using swr_alloc_set_opts() as well:
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* @code
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* SwrContext *swr = swr_alloc_set_opts(NULL, // we're allocating a new context
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* AV_CH_LAYOUT_STEREO, // out_ch_layout
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* AV_SAMPLE_FMT_S16, // out_sample_fmt
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* 44100, // out_sample_rate
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* AV_CH_LAYOUT_5POINT1, // in_ch_layout
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* AV_SAMPLE_FMT_FLTP, // in_sample_fmt
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* 48000, // in_sample_rate
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* 0, // log_offset
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* NULL); // log_ctx
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* @endcode
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*
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* Once all values have been set, it must be initialized with swr_init(). If
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* you need to change the conversion parameters, you can change the parameters
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* using @ref AVOptions, as described above in the first example; or by using
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* swr_alloc_set_opts(), but with the first argument the allocated context.
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* You must then call swr_init() again.
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*
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* The conversion itself is done by repeatedly calling swr_convert().
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* Note that the samples may get buffered in swr if you provide insufficient
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* output space or if sample rate conversion is done, which requires "future"
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* samples. Samples that do not require future input can be retrieved at any
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* time by using swr_convert() (in_count can be set to 0).
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* At the end of conversion the resampling buffer can be flushed by calling
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* swr_convert() with NULL in and 0 in_count.
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*
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* The samples used in the conversion process can be managed with the libavutil
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* @ref lavu_sampmanip "samples manipulation" API, including av_samples_alloc()
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* function used in the following example.
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*
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* The delay between input and output, can at any time be found by using
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* swr_get_delay().
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*
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* The following code demonstrates the conversion loop assuming the parameters
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* from above and caller-defined functions get_input() and handle_output():
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* @code
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* uint8_t **input;
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* int in_samples;
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*
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* while (get_input(&input, &in_samples)) {
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* uint8_t *output;
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* int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) +
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* in_samples, 44100, 48000, AV_ROUND_UP);
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* av_samples_alloc(&output, NULL, 2, out_samples,
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* AV_SAMPLE_FMT_S16, 0);
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* out_samples = swr_convert(swr, &output, out_samples,
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* input, in_samples);
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* handle_output(output, out_samples);
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* av_freep(&output);
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* }
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* @endcode
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*
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* When the conversion is finished, the conversion
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* context and everything associated with it must be freed with swr_free().
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* A swr_close() function is also available, but it exists mainly for
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* compatibility with libavresample, and is not required to be called.
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*
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* There will be no memory leak if the data is not completely flushed before
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* swr_free().
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*/
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#include <stdint.h>
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#include "libavutil/channel_layout.h"
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#include "libavutil/frame.h"
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#include "libavutil/samplefmt.h"
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#include "libswresample/version.h"
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/**
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* @name Option constants
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* These constants are used for the @ref avoptions interface for lswr.
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* @{
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*
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*/
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#define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
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//TODO use int resample ?
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//long term TODO can we enable this dynamically?
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/** Dithering algorithms */
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enum SwrDitherType {
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SWR_DITHER_NONE = 0,
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SWR_DITHER_RECTANGULAR,
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SWR_DITHER_TRIANGULAR,
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SWR_DITHER_TRIANGULAR_HIGHPASS,
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SWR_DITHER_NS = 64, ///< not part of API/ABI
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SWR_DITHER_NS_LIPSHITZ,
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SWR_DITHER_NS_F_WEIGHTED,
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SWR_DITHER_NS_MODIFIED_E_WEIGHTED,
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SWR_DITHER_NS_IMPROVED_E_WEIGHTED,
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SWR_DITHER_NS_SHIBATA,
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SWR_DITHER_NS_LOW_SHIBATA,
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SWR_DITHER_NS_HIGH_SHIBATA,
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SWR_DITHER_NB, ///< not part of API/ABI
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};
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/** Resampling Engines */
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enum SwrEngine {
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SWR_ENGINE_SWR, /**< SW Resampler */
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SWR_ENGINE_SOXR, /**< SoX Resampler */
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SWR_ENGINE_NB, ///< not part of API/ABI
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};
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/** Resampling Filter Types */
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enum SwrFilterType {
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SWR_FILTER_TYPE_CUBIC, /**< Cubic */
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SWR_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall windowed sinc */
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SWR_FILTER_TYPE_KAISER, /**< Kaiser windowed sinc */
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};
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/**
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* @}
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*/
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/**
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* The libswresample context. Unlike libavcodec and libavformat, this structure
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* is opaque. This means that if you would like to set options, you must use
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* the @ref avoptions API and cannot directly set values to members of the
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* structure.
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*/
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typedef struct SwrContext SwrContext;
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/**
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* Get the AVClass for SwrContext. It can be used in combination with
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* AV_OPT_SEARCH_FAKE_OBJ for examining options.
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*
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* @see av_opt_find().
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* @return the AVClass of SwrContext
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*/
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const AVClass *swr_get_class(void);
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/**
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* @name SwrContext constructor functions
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* @{
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*/
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/**
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* Allocate SwrContext.
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*
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* If you use this function you will need to set the parameters (manually or
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* with swr_alloc_set_opts()) before calling swr_init().
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*
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* @see swr_alloc_set_opts(), swr_init(), swr_free()
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* @return NULL on error, allocated context otherwise
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*/
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struct SwrContext *swr_alloc(void);
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/**
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* Initialize context after user parameters have been set.
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* @note The context must be configured using the AVOption API.
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*
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* @see av_opt_set_int()
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* @see av_opt_set_dict()
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*
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* @param[in,out] s Swr context to initialize
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* @return AVERROR error code in case of failure.
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*/
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int swr_init(struct SwrContext *s);
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/**
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* Check whether an swr context has been initialized or not.
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*
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* @param[in] s Swr context to check
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* @see swr_init()
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* @return positive if it has been initialized, 0 if not initialized
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*/
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int swr_is_initialized(struct SwrContext *s);
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/**
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* Allocate SwrContext if needed and set/reset common parameters.
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*
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* This function does not require s to be allocated with swr_alloc(). On the
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* other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters
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* on the allocated context.
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*
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* @param s existing Swr context if available, or NULL if not
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* @param out_ch_layout output channel layout (AV_CH_LAYOUT_*)
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* @param out_sample_fmt output sample format (AV_SAMPLE_FMT_*).
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* @param out_sample_rate output sample rate (frequency in Hz)
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* @param in_ch_layout input channel layout (AV_CH_LAYOUT_*)
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* @param in_sample_fmt input sample format (AV_SAMPLE_FMT_*).
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* @param in_sample_rate input sample rate (frequency in Hz)
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* @param log_offset logging level offset
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* @param log_ctx parent logging context, can be NULL
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*
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* @see swr_init(), swr_free()
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* @return NULL on error, allocated context otherwise
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*/
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struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
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int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
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int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
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int log_offset, void *log_ctx);
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/**
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* @}
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*
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* @name SwrContext destructor functions
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* @{
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*/
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/**
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* Free the given SwrContext and set the pointer to NULL.
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*
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* @param[in] s a pointer to a pointer to Swr context
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*/
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void swr_free(struct SwrContext **s);
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/**
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* Closes the context so that swr_is_initialized() returns 0.
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*
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* The context can be brought back to life by running swr_init(),
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* swr_init() can also be used without swr_close().
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* This function is mainly provided for simplifying the usecase
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* where one tries to support libavresample and libswresample.
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*
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* @param[in,out] s Swr context to be closed
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*/
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void swr_close(struct SwrContext *s);
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/**
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* @}
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*
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* @name Core conversion functions
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* @{
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*/
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/** Convert audio.
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*
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* in and in_count can be set to 0 to flush the last few samples out at the
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* end.
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*
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* If more input is provided than output space, then the input will be buffered.
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* You can avoid this buffering by using swr_get_out_samples() to retrieve an
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* upper bound on the required number of output samples for the given number of
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* input samples. Conversion will run directly without copying whenever possible.
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*
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* @param s allocated Swr context, with parameters set
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* @param out output buffers, only the first one need be set in case of packed audio
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* @param out_count amount of space available for output in samples per channel
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* @param in input buffers, only the first one need to be set in case of packed audio
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* @param in_count number of input samples available in one channel
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*
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* @return number of samples output per channel, negative value on error
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*/
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int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,
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const uint8_t **in , int in_count);
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/**
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* Convert the next timestamp from input to output
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* timestamps are in 1/(in_sample_rate * out_sample_rate) units.
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*
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* @note There are 2 slightly differently behaving modes.
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* @li When automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
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* in this case timestamps will be passed through with delays compensated
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* @li When automatic timestamp compensation is used, (min_compensation < FLT_MAX)
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* in this case the output timestamps will match output sample numbers.
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* See ffmpeg-resampler(1) for the two modes of compensation.
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*
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* @param s[in] initialized Swr context
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* @param pts[in] timestamp for the next input sample, INT64_MIN if unknown
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* @see swr_set_compensation(), swr_drop_output(), and swr_inject_silence() are
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* function used internally for timestamp compensation.
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* @return the output timestamp for the next output sample
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*/
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int64_t swr_next_pts(struct SwrContext *s, int64_t pts);
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/**
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* @}
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*
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* @name Low-level option setting functions
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* These functons provide a means to set low-level options that is not possible
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* with the AVOption API.
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* @{
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*/
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/**
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* Activate resampling compensation ("soft" compensation). This function is
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* internally called when needed in swr_next_pts().
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*
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* @param[in,out] s allocated Swr context. If it is not initialized,
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* or SWR_FLAG_RESAMPLE is not set, swr_init() is
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* called with the flag set.
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* @param[in] sample_delta delta in PTS per sample
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* @param[in] compensation_distance number of samples to compensate for
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* @return >= 0 on success, AVERROR error codes if:
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* @li @c s is NULL,
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* @li @c compensation_distance is less than 0,
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* @li @c compensation_distance is 0 but sample_delta is not,
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* @li compensation unsupported by resampler, or
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* @li swr_init() fails when called.
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*/
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int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance);
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/**
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* Set a customized input channel mapping.
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*
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* @param[in,out] s allocated Swr context, not yet initialized
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* @param[in] channel_map customized input channel mapping (array of channel
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* indexes, -1 for a muted channel)
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* @return >= 0 on success, or AVERROR error code in case of failure.
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*/
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int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map);
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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 libswresample 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 rematrix_maxval if 1.0, coefficients will be normalized to prevent
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* overflow. if INT_MAX, 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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* @param log_ctx parent logging context, can be NULL
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* @return 0 on success, negative AVERROR code on failure
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*/
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int swr_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, double rematrix_maxval,
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double rematrix_volume, double *matrix,
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int stride, enum AVMatrixEncoding matrix_encoding,
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void *log_ctx);
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/**
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* Set a customized remix matrix.
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*
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* @param s allocated Swr context, not yet initialized
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* @param matrix remix 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 offset between lines of the matrix
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* @return >= 0 on success, or AVERROR error code in case of failure.
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*/
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int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride);
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/**
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* @}
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*
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* @name Sample handling functions
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* @{
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*/
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/**
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* Drops the specified number of output samples.
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*
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* This function, along with swr_inject_silence(), is called by swr_next_pts()
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* if needed for "hard" compensation.
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*
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* @param s allocated Swr context
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* @param count number of samples to be dropped
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*
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* @return >= 0 on success, or a negative AVERROR code on failure
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*/
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int swr_drop_output(struct SwrContext *s, int count);
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/**
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* Injects the specified number of silence samples.
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*
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* This function, along with swr_drop_output(), is called by swr_next_pts()
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* if needed for "hard" compensation.
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*
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* @param s allocated Swr context
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* @param count number of samples to be dropped
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*
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* @return >= 0 on success, or a negative AVERROR code on failure
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*/
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int swr_inject_silence(struct SwrContext *s, int count);
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/**
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* Gets the delay the next input sample will experience relative to the next output sample.
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*
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* Swresample can buffer data if more input has been provided than available
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* output space, also converting between sample rates needs a delay.
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* This function returns the sum of all such delays.
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* The exact delay is not necessarily an integer value in either input or
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* output sample rate. Especially when downsampling by a large value, the
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* output sample rate may be a poor choice to represent the delay, similarly
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* for upsampling and the input sample rate.
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*
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* @param s swr context
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* @param base timebase in which the returned delay will be:
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* @li if it's set to 1 the returned delay is in seconds
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* @li if it's set to 1000 the returned delay is in milliseconds
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* @li if it's set to the input sample rate then the returned
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* delay is in input samples
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* @li if it's set to the output sample rate then the returned
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* delay is in output samples
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* @li if it's the least common multiple of in_sample_rate and
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* out_sample_rate then an exact rounding-free delay will be
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* returned
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* @returns the delay in 1 / @c base units.
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*/
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int64_t swr_get_delay(struct SwrContext *s, int64_t base);
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/**
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* Find an upper bound on the number of samples that the next swr_convert
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* call will output, if called with in_samples of input samples. This
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* depends on the internal state, and anything changing the internal state
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* (like further swr_convert() calls) will may change the number of samples
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* swr_get_out_samples() returns for the same number of input samples.
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*
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* @param in_samples number of input samples.
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* @note any call to swr_inject_silence(), swr_convert(), swr_next_pts()
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* or swr_set_compensation() invalidates this limit
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* @note it is recommended to pass the correct available buffer size
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* to all functions like swr_convert() even if swr_get_out_samples()
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* indicates that less would be used.
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* @returns an upper bound on the number of samples that the next swr_convert
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* will output or a negative value to indicate an error
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*/
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int swr_get_out_samples(struct SwrContext *s, int in_samples);
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/**
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* @}
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*
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* @name Configuration accessors
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* @{
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|
*/
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|
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|
/**
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* Return the @ref LIBSWRESAMPLE_VERSION_INT constant.
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*
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* This is useful to check if the build-time libswresample has the same version
|
|
* as the run-time one.
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|
*
|
|
* @returns the unsigned int-typed version
|
|
*/
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|
unsigned swresample_version(void);
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|
|
|
/**
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|
* Return the swr build-time configuration.
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|
*
|
|
* @returns the build-time @c ./configure flags
|
|
*/
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|
const char *swresample_configuration(void);
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|
|
/**
|
|
* Return the swr license.
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|
*
|
|
* @returns the license of libswresample, determined at build-time
|
|
*/
|
|
const char *swresample_license(void);
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|
|
|
/**
|
|
* @}
|
|
*
|
|
* @name AVFrame based API
|
|
* @{
|
|
*/
|
|
|
|
/**
|
|
* Convert the samples in the input AVFrame and write them to the output AVFrame.
|
|
*
|
|
* Input and output AVFrames must have channel_layout, sample_rate and format set.
|
|
*
|
|
* If the output AVFrame does not have the data pointers allocated the nb_samples
|
|
* field will be set using av_frame_get_buffer()
|
|
* is called to allocate the frame.
|
|
*
|
|
* The output AVFrame can be NULL or have fewer allocated samples than required.
|
|
* In this case, any remaining samples not written to the output will be added
|
|
* to an internal FIFO buffer, to be returned at the next call to this function
|
|
* or to swr_convert().
|
|
*
|
|
* If converting sample rate, there may be data remaining in the internal
|
|
* resampling delay buffer. swr_get_delay() tells the number of
|
|
* remaining samples. To get this data as output, call this function or
|
|
* swr_convert() with NULL input.
|
|
*
|
|
* If the SwrContext configuration does not match the output and
|
|
* input AVFrame settings the conversion does not take place and depending on
|
|
* which AVFrame is not matching AVERROR_OUTPUT_CHANGED, AVERROR_INPUT_CHANGED
|
|
* or the result of a bitwise-OR of them is returned.
|
|
*
|
|
* @see swr_delay()
|
|
* @see swr_convert()
|
|
* @see swr_get_delay()
|
|
*
|
|
* @param swr audio resample context
|
|
* @param output output AVFrame
|
|
* @param input input AVFrame
|
|
* @return 0 on success, AVERROR on failure or nonmatching
|
|
* configuration.
|
|
*/
|
|
int swr_convert_frame(SwrContext *swr,
|
|
AVFrame *output, const AVFrame *input);
|
|
|
|
/**
|
|
* Configure or reconfigure the SwrContext using the information
|
|
* provided by the AVFrames.
|
|
*
|
|
* The original resampling context is reset even on failure.
|
|
* The function calls swr_close() internally if the context is open.
|
|
*
|
|
* @see swr_close();
|
|
*
|
|
* @param swr audio resample context
|
|
* @param output output AVFrame
|
|
* @param input input AVFrame
|
|
* @return 0 on success, AVERROR on failure.
|
|
*/
|
|
int swr_config_frame(SwrContext *swr, const AVFrame *out, const AVFrame *in);
|
|
|
|
/**
|
|
* @}
|
|
* @}
|
|
*/
|
|
|
|
#endif /* SWRESAMPLE_SWRESAMPLE_H */
|