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// Targeted by JavaCPP version 1.4.2: DO NOT EDIT THIS FILE
package org.bytedeco.javacpp;
import java.nio.*;
import org.bytedeco.javacpp.*;
import org.bytedeco.javacpp.annotation.*;
import static org.bytedeco.javacpp.avutil.*;
public class swresample extends org.bytedeco.javacpp.presets.swresample {
static { Loader.load(); }
// Parsed from
/*
* 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_opts(). 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_opts(). 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.
*
*
* 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_opts(), 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():
*
*
* 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
// #include "libavutil/channel_layout.h"
// #include "libavutil/frame.h"
// #include "libavutil/samplefmt.h"
// #include "libswresample/version.h"
/**
* \name Option constants
* These constants are used for the \ref avoptions interface for lswr.
* \{
*
*/
/** Force resampling even if equal sample rate */
public static final int SWR_FLAG_RESAMPLE = 1;
//TODO use int resample ?
//long term TODO can we enable this dynamically?
/** Dithering algorithms */
/** enum SwrDitherType */
public static final int
SWR_DITHER_NONE = 0,
SWR_DITHER_RECTANGULAR = 1,
SWR_DITHER_TRIANGULAR = 2,
SWR_DITHER_TRIANGULAR_HIGHPASS = 3,
/** not part of API/ABI */
SWR_DITHER_NS = 64,
SWR_DITHER_NS_LIPSHITZ = 65,
SWR_DITHER_NS_F_WEIGHTED = 66,
SWR_DITHER_NS_MODIFIED_E_WEIGHTED = 67,
SWR_DITHER_NS_IMPROVED_E_WEIGHTED = 68,
SWR_DITHER_NS_SHIBATA = 69,
SWR_DITHER_NS_LOW_SHIBATA = 70,
SWR_DITHER_NS_HIGH_SHIBATA = 71,
/** not part of API/ABI */
SWR_DITHER_NB = 72;
/** Resampling Engines */
/** enum SwrEngine */
public static final int
/** SW Resampler */
SWR_ENGINE_SWR = 0,
/** SoX Resampler */
SWR_ENGINE_SOXR = 1,
/** not part of API/ABI */
SWR_ENGINE_NB = 2;
/** Resampling Filter Types */
/** enum SwrFilterType */
public static final int
/** Cubic */
SWR_FILTER_TYPE_CUBIC = 0,
/** Blackman Nuttall windowed sinc */
SWR_FILTER_TYPE_BLACKMAN_NUTTALL = 1,
/** Kaiser windowed sinc */
SWR_FILTER_TYPE_KAISER = 2;
/**
* \}
*/
/**
* 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.
*/
@Opaque public static class SwrContext extends Pointer {
/** Empty constructor. Calls {@code super((Pointer)null)}. */
public SwrContext() { super((Pointer)null); }
/** Pointer cast constructor. Invokes {@link Pointer#Pointer(Pointer)}. */
public SwrContext(Pointer p) { super(p); }
}
/**
* 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
*/
@NoException public static native @Const AVClass swr_get_class();
/**
* \name SwrContext constructor functions
* \{
*/
/**
* Allocate SwrContext.
*
* If you use this function you will need to set the parameters (manually or
* with swr_alloc_set_opts()) before calling swr_init().
*
* @see swr_alloc_set_opts(), swr_init(), swr_free()
* @return NULL on error, allocated context otherwise
*/
@NoException public static native SwrContext swr_alloc();
/**
* 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.
*/
@NoException public static native int swr_init(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
*/
@NoException public static native int swr_is_initialized(SwrContext s);
/**
* 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
*/
@NoException public static native SwrContext swr_alloc_set_opts(SwrContext s,
@Cast("int64_t") long out_ch_layout, @Cast("AVSampleFormat") int out_sample_fmt, int out_sample_rate,
@Cast("int64_t") long in_ch_layout, @Cast("AVSampleFormat") int in_sample_fmt, int in_sample_rate,
int log_offset, Pointer 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
*/
@NoException public static native void swr_free(@Cast("SwrContext**") PointerPointer s);
@NoException public static native void swr_free(@ByPtrPtr 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
*/
@NoException public static native void swr_close(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
*/
@NoException public static native int swr_convert(SwrContext s, @Cast("uint8_t**") PointerPointer out, int out_count,
@Cast("const uint8_t**") PointerPointer in, int in_count);
@NoException public static native int swr_convert(SwrContext s, @Cast("uint8_t**") @ByPtrPtr BytePointer out, int out_count,
@Cast("const uint8_t**") @ByPtrPtr BytePointer in, int in_count);
@NoException public static native int swr_convert(SwrContext s, @Cast("uint8_t**") @ByPtrPtr ByteBuffer out, int out_count,
@Cast("const uint8_t**") @ByPtrPtr ByteBuffer in, int in_count);
@NoException public static native int swr_convert(SwrContext s, @Cast("uint8_t**") @ByPtrPtr byte[] out, int out_count,
@Cast("const uint8_t**") @ByPtrPtr byte[] 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 s[in] initialized Swr context
* @param pts[in] 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
*/
@NoException public static native @Cast("int64_t") long swr_next_pts(SwrContext s, @Cast("int64_t") long 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.
*/
@NoException public static native int swr_set_compensation(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.
*/
@NoException public static native int swr_set_channel_mapping(SwrContext s, @Const IntPointer channel_map);
@NoException public static native int swr_set_channel_mapping(SwrContext s, @Const IntBuffer channel_map);
@NoException public static native int swr_set_channel_mapping(SwrContext s, @Const int[] channel_map);
/**
* 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
*/
@NoException public static native int swr_build_matrix(@Cast("uint64_t") long in_layout, @Cast("uint64_t") long out_layout,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, double rematrix_maxval,
double rematrix_volume, DoublePointer matrix,
int stride, @Cast("AVMatrixEncoding") int matrix_encoding,
Pointer log_ctx);
@NoException public static native int swr_build_matrix(@Cast("uint64_t") long in_layout, @Cast("uint64_t") long out_layout,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, double rematrix_maxval,
double rematrix_volume, DoubleBuffer matrix,
int stride, @Cast("AVMatrixEncoding") int matrix_encoding,
Pointer log_ctx);
@NoException public static native int swr_build_matrix(@Cast("uint64_t") long in_layout, @Cast("uint64_t") long out_layout,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, double rematrix_maxval,
double rematrix_volume, double[] matrix,
int stride, @Cast("AVMatrixEncoding") int matrix_encoding,
Pointer log_ctx);
/**
* 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.
*/
@NoException public static native int swr_set_matrix(SwrContext s, @Const DoublePointer matrix, int stride);
@NoException public static native int swr_set_matrix(SwrContext s, @Const DoubleBuffer matrix, int stride);
@NoException public static native int swr_set_matrix(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
*/
@NoException public static native int swr_drop_output(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
*/
@NoException public static native int swr_inject_silence(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
* @return the delay in 1 / \c base units.
*/
@NoException public static native @Cast("int64_t") long swr_get_delay(SwrContext s, @Cast("int64_t") long 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.
* @return an upper bound on the number of samples that the next swr_convert
* will output or a negative value to indicate an error
*/
@NoException public static native int swr_get_out_samples(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.
*
* @return the unsigned int-typed version
*/
@NoException public static native @Cast("unsigned") int swresample_version();
/**
* Return the swr build-time configuration.
*
* @return the build-time \c ./configure flags
*/
@NoException public static native @Cast("const char*") BytePointer swresample_configuration();
/**
* Return the swr license.
*
* @return the license of libswresample, determined at build-time
*/
@NoException public static native @Cast("const char*") BytePointer swresample_license();
/**
* \}
*
* \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.
*/
@NoException public static native 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.
*/
@NoException public static native int swr_config_frame(SwrContext swr, @Const AVFrame out, @Const AVFrame in);
/**
* \}
* \}
*/
// #endif /* SWRESAMPLE_SWRESAMPLE_H */
}
