third_party_ffmpeg/libswresample/swresample.h

/*
 * 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 */