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FFmpeg/libswresample/swresample.h
Martin Storsjö 2d368392a5 Keep including the full version.h when headers are included externally
This avoids unnecessary churn and build breakage for users, by
making sure the whole version.h is included like it has been so far,
while keeping the benefit of not needing to rebuild most files in
the ffmpeg tree on minor/micro bumps.

Signed-off-by: Martin Storsjö <martin@martin.st>
2022-03-19 00:01:57 +02:00

651 lines
25 KiB
C

/*
* Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
*
* 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,
AVChannelLayout *out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
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 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
*/
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 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 */