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+/*
+ * Copyright (C) 2011-2013 Michael Niedermayer ([email protected])
+ *
+ * This file is part of libswresample
+ *
+ * libswresample is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * libswresample is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with libswresample; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef SWRESAMPLE_SWRESAMPLE_H
+#define SWRESAMPLE_SWRESAMPLE_H
+
+/**
+ * @file
+ * @ingroup lswr
+ * libswresample public header
+ */
+
+/**
+ * @defgroup lswr libswresample
+ * @{
+ *
+ * Audio resampling, sample format conversion and mixing library.
+ *
+ * Interaction with lswr is done through SwrContext, which is
+ * allocated with swr_alloc() or swr_alloc_set_opts2(). It is opaque, so all parameters
+ * must be set with the @ref avoptions API.
+ *
+ * The first thing you will need to do in order to use lswr is to allocate
+ * SwrContext. This can be done with swr_alloc() or swr_alloc_set_opts2(). If you
+ * are using the former, you must set options through the @ref avoptions API.
+ * The latter function provides the same feature, but it allows you to set some
+ * common options in the same statement.
+ *
+ * For example the following code will setup conversion from planar float sample
+ * format to interleaved signed 16-bit integer, downsampling from 48kHz to
+ * 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing
+ * matrix). This is using the swr_alloc() function.
+ * @code
+ * SwrContext *swr = swr_alloc();
+ * av_opt_set_channel_layout(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
+ * av_opt_set_channel_layout(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
+ * av_opt_set_int(swr, "in_sample_rate", 48000, 0);
+ * av_opt_set_int(swr, "out_sample_rate", 44100, 0);
+ * av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
+ * av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
+ * @endcode
+ *
+ * The same job can be done using swr_alloc_set_opts2() as well:
+ * @code
+ * SwrContext *swr = NULL;
+ * int ret = swr_alloc_set_opts2(&swr, // we're allocating a new context
+ * &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO, // out_ch_layout
+ * AV_SAMPLE_FMT_S16, // out_sample_fmt
+ * 44100, // out_sample_rate
+ * &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT1, // in_ch_layout
+ * AV_SAMPLE_FMT_FLTP, // in_sample_fmt
+ * 48000, // in_sample_rate
+ * 0, // log_offset
+ * NULL); // log_ctx
+ * @endcode
+ *
+ * Once all values have been set, it must be initialized with swr_init(). If
+ * you need to change the conversion parameters, you can change the parameters
+ * using @ref avoptions, as described above in the first example; or by using
+ * swr_alloc_set_opts2(), but with the first argument the allocated context.
+ * You must then call swr_init() again.
+ *
+ * The conversion itself is done by repeatedly calling swr_convert().
+ * Note that the samples may get buffered in swr if you provide insufficient
+ * output space or if sample rate conversion is done, which requires "future"
+ * samples. Samples that do not require future input can be retrieved at any
+ * time by using swr_convert() (in_count can be set to 0).
+ * At the end of conversion the resampling buffer can be flushed by calling
+ * swr_convert() with NULL in and 0 in_count.
+ *
+ * The samples used in the conversion process can be managed with the libavutil
+ * @ref lavu_sampmanip "samples manipulation" API, including av_samples_alloc()
+ * function used in the following example.
+ *
+ * The delay between input and output, can at any time be found by using
+ * swr_get_delay().
+ *
+ * The following code demonstrates the conversion loop assuming the parameters
+ * from above and caller-defined functions get_input() and handle_output():
+ * @code
+ * uint8_t **input;
+ * int in_samples;
+ *
+ * while (get_input(&input, &in_samples)) {
+ * uint8_t *output;
+ * int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) +
+ * in_samples, 44100, 48000, AV_ROUND_UP);
+ * av_samples_alloc(&output, NULL, 2, out_samples,
+ * AV_SAMPLE_FMT_S16, 0);
+ * out_samples = swr_convert(swr, &output, out_samples,
+ * input, in_samples);
+ * handle_output(output, out_samples);
+ * av_freep(&output);
+ * }
+ * @endcode
+ *
+ * When the conversion is finished, the conversion
+ * context and everything associated with it must be freed with swr_free().
+ * A swr_close() function is also available, but it exists mainly for
+ * compatibility with libavresample, and is not required to be called.
+ *
+ * There will be no memory leak if the data is not completely flushed before
+ * swr_free().
+ */
+
+#include <stdint.h>
+#include "libavutil/channel_layout.h"
+#include "libavutil/frame.h"
+#include "libavutil/samplefmt.h"
+
+#include "libswresample/version_major.h"
+#ifndef HAVE_AV_CONFIG_H
+/* When included as part of the ffmpeg build, only include the major version
+ * to avoid unnecessary rebuilds. When included externally, keep including
+ * the full version information. */
+#include "libswresample/version.h"
+#endif
+
+/**
+ * @name Option constants
+ * These constants are used for the @ref avoptions interface for lswr.
+ * @{
+ *
+ */
+
+#define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
+//TODO use int resample ?
+//long term TODO can we enable this dynamically?
+
+/** Dithering algorithms */
+enum SwrDitherType {
+ SWR_DITHER_NONE = 0,
+ SWR_DITHER_RECTANGULAR,
+ SWR_DITHER_TRIANGULAR,
+ SWR_DITHER_TRIANGULAR_HIGHPASS,
+
+ SWR_DITHER_NS = 64, ///< not part of API/ABI
+ SWR_DITHER_NS_LIPSHITZ,
+ SWR_DITHER_NS_F_WEIGHTED,
+ SWR_DITHER_NS_MODIFIED_E_WEIGHTED,
+ SWR_DITHER_NS_IMPROVED_E_WEIGHTED,
+ SWR_DITHER_NS_SHIBATA,
+ SWR_DITHER_NS_LOW_SHIBATA,
+ SWR_DITHER_NS_HIGH_SHIBATA,
+ SWR_DITHER_NB, ///< not part of API/ABI
+};
+
+/** Resampling Engines */
+enum SwrEngine {
+ SWR_ENGINE_SWR, /**< SW Resampler */
+ SWR_ENGINE_SOXR, /**< SoX Resampler */
+ SWR_ENGINE_NB, ///< not part of API/ABI
+};
+
+/** Resampling Filter Types */
+enum SwrFilterType {
+ SWR_FILTER_TYPE_CUBIC, /**< Cubic */
+ SWR_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall windowed sinc */
+ SWR_FILTER_TYPE_KAISER, /**< Kaiser windowed sinc */
+};
+
+/**
+ * @}
+ */
+
+/**
+ * The libswresample context. Unlike libavcodec and libavformat, this structure
+ * is opaque. This means that if you would like to set options, you must use
+ * the @ref avoptions API and cannot directly set values to members of the
+ * structure.
+ */
+typedef struct SwrContext SwrContext;
+
+/**
+ * Get the AVClass for SwrContext. It can be used in combination with
+ * AV_OPT_SEARCH_FAKE_OBJ for examining options.
+ *
+ * @see av_opt_find().
+ * @return the AVClass of SwrContext
+ */
+const AVClass *swr_get_class(void);
+
+/**
+ * @name SwrContext constructor functions
+ * @{
+ */
+
+/**
+ * Allocate SwrContext.
+ *
+ * If you use this function you will need to set the parameters (manually or
+ * with swr_alloc_set_opts2()) before calling swr_init().
+ *
+ * @see swr_alloc_set_opts2(), swr_init(), swr_free()
+ * @return NULL on error, allocated context otherwise
+ */
+struct SwrContext *swr_alloc(void);
+
+/**
+ * Initialize context after user parameters have been set.
+ * @note The context must be configured using the AVOption API.
+ *
+ * @see av_opt_set_int()
+ * @see av_opt_set_dict()
+ *
+ * @param[in,out] s Swr context to initialize
+ * @return AVERROR error code in case of failure.
+ */
+int swr_init(struct SwrContext *s);
+
+/**
+ * Check whether an swr context has been initialized or not.
+ *
+ * @param[in] s Swr context to check
+ * @see swr_init()
+ * @return positive if it has been initialized, 0 if not initialized
+ */
+int swr_is_initialized(struct SwrContext *s);
+
+#if FF_API_OLD_CHANNEL_LAYOUT
+/**
+ * Allocate SwrContext if needed and set/reset common parameters.
+ *
+ * This function does not require s to be allocated with swr_alloc(). On the
+ * other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters
+ * on the allocated context.
+ *
+ * @param s existing Swr context if available, or NULL if not
+ * @param out_ch_layout output channel layout (AV_CH_LAYOUT_*)
+ * @param out_sample_fmt output sample format (AV_SAMPLE_FMT_*).
+ * @param out_sample_rate output sample rate (frequency in Hz)
+ * @param in_ch_layout input channel layout (AV_CH_LAYOUT_*)
+ * @param in_sample_fmt input sample format (AV_SAMPLE_FMT_*).
+ * @param in_sample_rate input sample rate (frequency in Hz)
+ * @param log_offset logging level offset
+ * @param log_ctx parent logging context, can be NULL
+ *
+ * @see swr_init(), swr_free()
+ * @return NULL on error, allocated context otherwise
+ * @deprecated use @ref swr_alloc_set_opts2()
+ */
+attribute_deprecated
+struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
+ int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
+ int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
+ int log_offset, void *log_ctx);
+#endif
+
+/**
+ * Allocate SwrContext if needed and set/reset common parameters.
+ *
+ * This function does not require *ps to be allocated with swr_alloc(). On the
+ * other hand, swr_alloc() can use swr_alloc_set_opts2() to set the parameters
+ * on the allocated context.
+ *
+ * @param ps Pointer to an existing Swr context if available, or to NULL if not.
+ * On success, *ps will be set to the allocated context.
+ * @param out_ch_layout output channel layout (e.g. AV_CHANNEL_LAYOUT_*)
+ * @param out_sample_fmt output sample format (AV_SAMPLE_FMT_*).
+ * @param out_sample_rate output sample rate (frequency in Hz)
+ * @param in_ch_layout input channel layout (e.g. AV_CHANNEL_LAYOUT_*)
+ * @param in_sample_fmt input sample format (AV_SAMPLE_FMT_*).
+ * @param in_sample_rate input sample rate (frequency in Hz)
+ * @param log_offset logging level offset
+ * @param log_ctx parent logging context, can be NULL
+ *
+ * @see swr_init(), swr_free()
+ * @return 0 on success, a negative AVERROR code on error.
+ * On error, the Swr context is freed and *ps set to NULL.
+ */
+int swr_alloc_set_opts2(struct SwrContext **ps,
+ const AVChannelLayout *out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
+ const AVChannelLayout *in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
+ int log_offset, void *log_ctx);
+/**
+ * @}
+ *
+ * @name SwrContext destructor functions
+ * @{
+ */
+
+/**
+ * Free the given SwrContext and set the pointer to NULL.
+ *
+ * @param[in] s a pointer to a pointer to Swr context
+ */
+void swr_free(struct SwrContext **s);
+
+/**
+ * Closes the context so that swr_is_initialized() returns 0.
+ *
+ * The context can be brought back to life by running swr_init(),
+ * swr_init() can also be used without swr_close().
+ * This function is mainly provided for simplifying the usecase
+ * where one tries to support libavresample and libswresample.
+ *
+ * @param[in,out] s Swr context to be closed
+ */
+void swr_close(struct SwrContext *s);
+
+/**
+ * @}
+ *
+ * @name Core conversion functions
+ * @{
+ */
+
+/** Convert audio.
+ *
+ * in and in_count can be set to 0 to flush the last few samples out at the
+ * end.
+ *
+ * If more input is provided than output space, then the input will be buffered.
+ * You can avoid this buffering by using swr_get_out_samples() to retrieve an
+ * upper bound on the required number of output samples for the given number of
+ * input samples. Conversion will run directly without copying whenever possible.
+ *
+ * @param s allocated Swr context, with parameters set
+ * @param out output buffers, only the first one need be set in case of packed audio
+ * @param out_count amount of space available for output in samples per channel
+ * @param in input buffers, only the first one need to be set in case of packed audio
+ * @param in_count number of input samples available in one channel
+ *
+ * @return number of samples output per channel, negative value on error
+ */
+int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,
+ const uint8_t **in , int in_count);
+
+/**
+ * Convert the next timestamp from input to output
+ * timestamps are in 1/(in_sample_rate * out_sample_rate) units.
+ *
+ * @note There are 2 slightly differently behaving modes.
+ * @li When automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
+ * in this case timestamps will be passed through with delays compensated
+ * @li When automatic timestamp compensation is used, (min_compensation < FLT_MAX)
+ * in this case the output timestamps will match output sample numbers.
+ * See ffmpeg-resampler(1) for the two modes of compensation.
+ *
+ * @param[in] s initialized Swr context
+ * @param[in] pts timestamp for the next input sample, INT64_MIN if unknown
+ * @see swr_set_compensation(), swr_drop_output(), and swr_inject_silence() are
+ * function used internally for timestamp compensation.
+ * @return the output timestamp for the next output sample
+ */
+int64_t swr_next_pts(struct SwrContext *s, int64_t pts);
+
+/**
+ * @}
+ *
+ * @name Low-level option setting functions
+ * These functons provide a means to set low-level options that is not possible
+ * with the AVOption API.
+ * @{
+ */
+
+/**
+ * Activate resampling compensation ("soft" compensation). This function is
+ * internally called when needed in swr_next_pts().
+ *
+ * @param[in,out] s allocated Swr context. If it is not initialized,
+ * or SWR_FLAG_RESAMPLE is not set, swr_init() is
+ * called with the flag set.
+ * @param[in] sample_delta delta in PTS per sample
+ * @param[in] compensation_distance number of samples to compensate for
+ * @return >= 0 on success, AVERROR error codes if:
+ * @li @c s is NULL,
+ * @li @c compensation_distance is less than 0,
+ * @li @c compensation_distance is 0 but sample_delta is not,
+ * @li compensation unsupported by resampler, or
+ * @li swr_init() fails when called.
+ */
+int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance);
+
+/**
+ * Set a customized input channel mapping.
+ *
+ * @param[in,out] s allocated Swr context, not yet initialized
+ * @param[in] channel_map customized input channel mapping (array of channel
+ * indexes, -1 for a muted channel)
+ * @return >= 0 on success, or AVERROR error code in case of failure.
+ */
+int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map);
+
+#if FF_API_OLD_CHANNEL_LAYOUT
+/**
+ * Generate a channel mixing matrix.
+ *
+ * This function is the one used internally by libswresample for building the
+ * default mixing matrix. It is made public just as a utility function for
+ * building custom matrices.
+ *
+ * @param in_layout input channel layout
+ * @param out_layout output channel layout
+ * @param center_mix_level mix level for the center channel
+ * @param surround_mix_level mix level for the surround channel(s)
+ * @param lfe_mix_level mix level for the low-frequency effects channel
+ * @param rematrix_maxval if 1.0, coefficients will be normalized to prevent
+ * overflow. if INT_MAX, coefficients will not be
+ * normalized.
+ * @param[out] matrix mixing coefficients; matrix[i + stride * o] is
+ * the weight of input channel i in output channel o.
+ * @param stride distance between adjacent input channels in the
+ * matrix array
+ * @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
+ * @param log_ctx parent logging context, can be NULL
+ * @return 0 on success, negative AVERROR code on failure
+ * @deprecated use @ref swr_build_matrix2()
+ */
+attribute_deprecated
+int swr_build_matrix(uint64_t in_layout, uint64_t out_layout,
+ double center_mix_level, double surround_mix_level,
+ double lfe_mix_level, double rematrix_maxval,
+ double rematrix_volume, double *matrix,
+ int stride, enum AVMatrixEncoding matrix_encoding,
+ void *log_ctx);
+#endif
+
+/**
+ * Generate a channel mixing matrix.
+ *
+ * This function is the one used internally by libswresample for building the
+ * default mixing matrix. It is made public just as a utility function for
+ * building custom matrices.
+ *
+ * @param in_layout input channel layout
+ * @param out_layout output channel layout
+ * @param center_mix_level mix level for the center channel
+ * @param surround_mix_level mix level for the surround channel(s)
+ * @param lfe_mix_level mix level for the low-frequency effects channel
+ * @param rematrix_maxval if 1.0, coefficients will be normalized to prevent
+ * overflow. if INT_MAX, coefficients will not be
+ * normalized.
+ * @param[out] matrix mixing coefficients; matrix[i + stride * o] is
+ * the weight of input channel i in output channel o.
+ * @param stride distance between adjacent input channels in the
+ * matrix array
+ * @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
+ * @param log_ctx parent logging context, can be NULL
+ * @return 0 on success, negative AVERROR code on failure
+ */
+int swr_build_matrix2(const AVChannelLayout *in_layout, const AVChannelLayout *out_layout,
+ double center_mix_level, double surround_mix_level,
+ double lfe_mix_level, double maxval,
+ double rematrix_volume, double *matrix,
+ ptrdiff_t stride, enum AVMatrixEncoding matrix_encoding,
+ void *log_context);
+
+/**
+ * Set a customized remix matrix.
+ *
+ * @param s allocated Swr context, not yet initialized
+ * @param matrix remix coefficients; matrix[i + stride * o] is
+ * the weight of input channel i in output channel o
+ * @param stride offset between lines of the matrix
+ * @return >= 0 on success, or AVERROR error code in case of failure.
+ */
+int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride);
+
+/**
+ * @}
+ *
+ * @name Sample handling functions
+ * @{
+ */
+
+/**
+ * Drops the specified number of output samples.
+ *
+ * This function, along with swr_inject_silence(), is called by swr_next_pts()
+ * if needed for "hard" compensation.
+ *
+ * @param s allocated Swr context
+ * @param count number of samples to be dropped
+ *
+ * @return >= 0 on success, or a negative AVERROR code on failure
+ */
+int swr_drop_output(struct SwrContext *s, int count);
+
+/**
+ * Injects the specified number of silence samples.
+ *
+ * This function, along with swr_drop_output(), is called by swr_next_pts()
+ * if needed for "hard" compensation.
+ *
+ * @param s allocated Swr context
+ * @param count number of samples to be dropped
+ *
+ * @return >= 0 on success, or a negative AVERROR code on failure
+ */
+int swr_inject_silence(struct SwrContext *s, int count);
+
+/**
+ * Gets the delay the next input sample will experience relative to the next output sample.
+ *
+ * Swresample can buffer data if more input has been provided than available
+ * output space, also converting between sample rates needs a delay.
+ * This function returns the sum of all such delays.
+ * The exact delay is not necessarily an integer value in either input or
+ * output sample rate. Especially when downsampling by a large value, the
+ * output sample rate may be a poor choice to represent the delay, similarly
+ * for upsampling and the input sample rate.
+ *
+ * @param s swr context
+ * @param base timebase in which the returned delay will be:
+ * @li if it's set to 1 the returned delay is in seconds
+ * @li if it's set to 1000 the returned delay is in milliseconds
+ * @li if it's set to the input sample rate then the returned
+ * delay is in input samples
+ * @li if it's set to the output sample rate then the returned
+ * delay is in output samples
+ * @li if it's the least common multiple of in_sample_rate and
+ * out_sample_rate then an exact rounding-free delay will be
+ * returned
+ * @returns the delay in 1 / @c base units.
+ */
+int64_t swr_get_delay(struct SwrContext *s, int64_t base);
+
+/**
+ * Find an upper bound on the number of samples that the next swr_convert
+ * call will output, if called with in_samples of input samples. This
+ * depends on the internal state, and anything changing the internal state
+ * (like further swr_convert() calls) will may change the number of samples
+ * swr_get_out_samples() returns for the same number of input samples.
+ *
+ * @param in_samples number of input samples.
+ * @note any call to swr_inject_silence(), swr_convert(), swr_next_pts()
+ * or swr_set_compensation() invalidates this limit
+ * @note it is recommended to pass the correct available buffer size
+ * to all functions like swr_convert() even if swr_get_out_samples()
+ * indicates that less would be used.
+ * @returns an upper bound on the number of samples that the next swr_convert
+ * will output or a negative value to indicate an error
+ */
+int swr_get_out_samples(struct SwrContext *s, int in_samples);
+
+/**
+ * @}
+ *
+ * @name Configuration accessors
+ * @{
+ */
+
+/**
+ * Return the @ref LIBSWRESAMPLE_VERSION_INT constant.
+ *
+ * This is useful to check if the build-time libswresample has the same version
+ * as the run-time one.
+ *
+ * @returns the unsigned int-typed version
+ */
+unsigned swresample_version(void);
+
+/**
+ * Return the swr build-time configuration.
+ *
+ * @returns the build-time @c ./configure flags
+ */
+const char *swresample_configuration(void);
+
+/**
+ * Return the swr license.
+ *
+ * @returns the license of libswresample, determined at build-time
+ */
+const char *swresample_license(void);
+
+/**
+ * @}
+ *
+ * @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 out output AVFrame
+ * @param in 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 */