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221 lines
7.8 KiB
C
221 lines
7.8 KiB
C
/*
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* Copyright (C) 2011-2012 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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/**
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* @file
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* libswresample public header
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*/
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#ifndef SWRESAMPLE_SWRESAMPLE_H
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#define SWRESAMPLE_SWRESAMPLE_H
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#include <inttypes.h>
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#include "libavutil/samplefmt.h"
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#include "libswresample/version.h"
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#if LIBSWRESAMPLE_VERSION_MAJOR < 1
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#define SWR_CH_MAX 32 ///< Maximum number of channels
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#endif
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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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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_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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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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*/
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const AVClass *swr_get_class(void);
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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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*
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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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* 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 Swr context, can be NULL
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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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* Free the given SwrContext and set the pointer to NULL.
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*/
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void swr_free(struct SwrContext **s);
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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 providing more output space than input.
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* Convertion 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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* timestampe 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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* First is 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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* Second is 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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*
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* @param pts timstamp for the next input sample, INT64_MIN if unknown
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* @returns 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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* Activate resampling compensation.
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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 s allocated Swr context, not yet initialized
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* @param channel_map customized input channel mapping (array of channel
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* indexes, -1 for a muted channel)
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* @return 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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* 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 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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* Drops the specified number of output samples.
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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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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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*
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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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* if its set to 1 the returned delay is in seconds
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* if its set to 1000 the returned delay is in milli seconds
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* if its set to the input sample rate then the returned delay is in input samples
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* if its set to the output sample rate then the returned delay is in output samples
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* an exact rounding free delay can be found by using LCM(in_sample_rate, out_sample_rate)
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* @returns the delay in 1/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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* Return the LIBSWRESAMPLE_VERSION_INT constant.
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*/
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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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*/
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const char *swresample_configuration(void);
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/**
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* Return the swr license.
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*/
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const char *swresample_license(void);
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#endif /* SWRESAMPLE_SWRESAMPLE_H */
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