/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVRESAMPLE_AVRESAMPLE_H
#define AVRESAMPLE_AVRESAMPLE_H
/**
* @file
* @ingroup lavr
* external API header
*/
/**
* @defgroup lavr libavresample
* @{
*
* Libavresample (lavr) is a library that handles audio resampling, sample
* format conversion and mixing.
*
* Interaction with lavr is done through AVAudioResampleContext, which is
* allocated with avresample_alloc_context(). It is opaque, so all parameters
* must be set with the @ref avoptions API.
*
* 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):
* @code
* AVAudioResampleContext *avr = avresample_alloc_context();
* av_opt_set_int(avr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
* av_opt_set_int(avr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
* av_opt_set_int(avr, "in_sample_rate", 48000, 0);
* av_opt_set_int(avr, "out_sample_rate", 44100, 0);
* av_opt_set_int(avr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
* av_opt_set_int(avr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
* @endcode
*
* Once the context is initialized, it must be opened with avresample_open(). If
* you need to change the conversion parameters, you must close the context with
* avresample_close(), change the parameters as described above, then reopen it
* again.
*
* The conversion itself is done by repeatedly calling avresample_convert().
* Note that the samples may get buffered in two places in lavr. The first one
* is the output FIFO, where the samples end up if the output buffer is not
* large enough. The data stored in there may be retrieved at any time with
* avresample_read(). The second place is the resampling delay buffer,
* applicable only when resampling is done. The samples in it require more input
* before they can be processed. Their current amount is returned by
* avresample_get_delay(). At the end of conversion the resampling buffer can be
* flushed by calling avresample_convert() with NULL input.
*
* 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_linesize, in_samples;
*
* while (get_input(&input, &in_linesize, &in_samples)) {
* uint8_t *output
* int out_linesize;
* int out_samples = avresample_get_out_samples(avr, in_samples);
*
* av_samples_alloc(&output, &out_linesize, 2, out_samples,
* AV_SAMPLE_FMT_S16, 0);
* out_samples = avresample_convert(avr, &output, out_linesize, out_samples,
* input, in_linesize, in_samples);
* handle_output(output, out_linesize, out_samples);
* av_freep(&output);
* }
* @endcode
*
* When the conversion is finished and the FIFOs are flushed if required, the
* conversion context and everything associated with it must be freed with
* avresample_free().
*/
#include "libavutil/avutil.h"
#include "libavutil/channel_layout.h"
#include "libavutil/dict.h"
#include "libavutil/frame.h"
#include "libavutil/log.h"
#include "libavutil/mathematics.h"
#include "libavresample/version.h"
#define AVRESAMPLE_MAX_CHANNELS 32
typedef struct AVAudioResampleContext AVAudioResampleContext;
/** Mixing Coefficient Types */
enum AVMixCoeffType {
AV_MIX_COEFF_TYPE_Q8, /** 16-bit 8.8 fixed-point */
AV_MIX_COEFF_TYPE_Q15, /** 32-bit 17.15 fixed-point */
AV_MIX_COEFF_TYPE_FLT, /** floating-point */
AV_MIX_COEFF_TYPE_NB, /** Number of coeff types. Not part of ABI */
};
/** Resampling Filter Types */
enum AVResampleFilterType {
AV_RESAMPLE_FILTER_TYPE_CUBIC, /**< Cubic */
AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
AV_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
};
enum AVResampleDitherMethod {
AV_RESAMPLE_DITHER_NONE, /**< Do not use dithering */
AV_RESAMPLE_DITHER_RECTANGULAR, /**< Rectangular Dither */
AV_RESAMPLE_DITHER_TRIANGULAR, /**< Triangular Dither*/
AV_RESAMPLE_DITHER_TRIANGULAR_HP, /**< Triangular Dither with High Pass */
AV_RESAMPLE_DITHER_TRIANGULAR_NS, /**< Triangular Dither with Noise Shaping */
AV_RESAMPLE_DITHER_NB, /**< Number of dither types. Not part of ABI. */
};
/**
* Return the LIBAVRESAMPLE_VERSION_INT constant.
*/
unsigned avresample_version(void);
/**
* Return the libavresample build-time configuration.
* @return configure string
*/
const char *avresample_configuration(void);
/**
* Return the libavresample license.
*/
const char *avresample_license(void);
/**
* Get the AVClass for AVAudioResampleContext.
*
* Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options
* without allocating a context.
*
* @see av_opt_find().
*
* @return AVClass for AVAudioResampleContext
*/
const AVClass *avresample_get_class(void);
/**
* Allocate AVAudioResampleContext and set options.
*
* @return allocated audio resample context, or NULL on failure
*/
AVAudioResampleContext *avresample_alloc_context(void);
/**
* Initialize AVAudioResampleContext.
* @note The context must be configured using the AVOption API.
* @note The fields "in_channel_layout", "out_channel_layout",
* "in_sample_rate", "out_sample_rate", "in_sample_fmt",
* "out_sample_fmt" must be set.
*
* @see av_opt_set_int()
* @see av_opt_set_dict()
* @see av_get_default_channel_layout()
*
* @param avr audio resample context
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_open(AVAudioResampleContext *avr);
/**
* Check whether an AVAudioResampleContext is open or closed.
*
* @param avr AVAudioResampleContext to check
* @return 1 if avr is open, 0 if avr is closed.
*/
int avresample_is_open(AVAudioResampleContext *avr);
/**
* Close AVAudioResampleContext.
*
* This closes the context, but it does not change the parameters. The context
* can be reopened with avresample_open(). It does, however, clear the output
* FIFO and any remaining leftover samples in the resampling delay buffer. If
* there was a custom matrix being used, that is also cleared.
*
* @see avresample_convert()
* @see avresample_set_matrix()
*
* @param avr audio resample context
*/
void avresample_close(AVAudioResampleContext *avr);
/**
* Free AVAudioResampleContext and associated AVOption values.
*
* This also calls avresample_close() before freeing.
*
* @param avr audio resample context
*/
void avresample_free(AVAudioResampleContext **avr);
/**
* Generate a channel mixing matrix.
*
* This function is the one used internally by libavresample 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 normalize if 1, coefficients will be normalized to prevent
* overflow. if 0, 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)
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, int normalize, double *matrix,
int stride, enum AVMatrixEncoding matrix_encoding);
/**
* Get the current channel mixing matrix.
*
* If no custom matrix has been previously set or the AVAudioResampleContext is
* not open, an error is returned.
*
* @param avr audio resample context
* @param 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
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
int stride);
/**
* Set channel mixing matrix.
*
* Allows for setting a custom mixing matrix, overriding the default matrix
* generated internally during avresample_open(). This function can be called
* anytime on an allocated context, either before or after calling
* avresample_open(), as long as the channel layouts have been set.
* avresample_convert() always uses the current matrix.
* Calling avresample_close() on the context will clear the current matrix.
*
* @see avresample_close()
*
* @param avr audio resample context
* @param 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
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
int stride);
/**
* Set a customized input channel mapping.
*
* This function can only be called when the allocated context is not open.
* Also, the input channel layout must have already been set.
*
* Calling avresample_close() on the context will clear the channel mapping.
*
* The map for each input channel specifies the channel index in the source to
* use for that particular channel, or -1 to mute the channel. Source channels
* can be duplicated by using the same index for multiple input channels.
*
* Examples:
*
* Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to FFmpeg order (L,R,C,LFE,Ls,Rs):
* { 1, 2, 0, 5, 3, 4 }
*
* Muting the 3rd channel in 4-channel input:
* { 0, 1, -1, 3 }
*
* Duplicating the left channel of stereo input:
* { 0, 0 }
*
* @param avr audio resample context
* @param channel_map customized input channel mapping
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_set_channel_mapping(AVAudioResampleContext *avr,
const int *channel_map);
/**
* Set compensation for resampling.
*
* This can be called anytime after avresample_open(). If resampling is not
* automatically enabled because of a sample rate conversion, the
* "force_resampling" option must have been set to 1 when opening the context
* in order to use resampling compensation.
*
* @param avr audio resample context
* @param sample_delta compensation delta, in samples
* @param compensation_distance compensation distance, in samples
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
int compensation_distance);
/**
* Provide the upper bound on the number of samples the configured
* conversion would output.
*
* @param avr audio resample context
* @param in_nb_samples number of input samples
*
* @return number of samples or AVERROR(EINVAL) if the value
* would exceed INT_MAX
*/
int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples);
/**
* Convert input samples and write them to the output FIFO.
*
* The upper bound on the number of output samples can be obtained through
* avresample_get_out_samples().
*
* The output data 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 avresample_read().
*
* If converting sample rate, there may be data remaining in the internal
* resampling delay buffer. avresample_get_delay() tells the number of remaining
* samples. To get this data as output, call avresample_convert() with NULL
* input.
*
* At the end of the conversion process, there may be data remaining in the
* internal FIFO buffer. avresample_available() tells the number of remaining
* samples. To get this data as output, either call avresample_convert() with
* NULL input or call avresample_read().
*
* @see avresample_get_out_samples()
* @see avresample_read()
* @see avresample_get_delay()
*
* @param avr audio resample context
* @param output output data pointers
* @param out_plane_size output plane size, in bytes.
* This can be 0 if unknown, but that will lead to
* optimized functions not being used directly on the
* output, which could slow down some conversions.
* @param out_samples maximum number of samples that the output buffer can hold
* @param input input data pointers
* @param in_plane_size input plane size, in bytes
* This can be 0 if unknown, but that will lead to
* optimized functions not being used directly on the
* input, which could slow down some conversions.
* @param in_samples number of input samples to convert
* @return number of samples written to the output buffer,
* not including converted samples added to the internal
* output FIFO
*/
int avresample_convert(AVAudioResampleContext *avr, uint8_t **output,
int out_plane_size, int out_samples,
uint8_t * const *input, int in_plane_size,
int in_samples);
/**
* Return the number of samples currently in the resampling delay buffer.
*
* When resampling, there may be a delay between the input and output. Any
* unconverted samples in each call are stored internally in a delay buffer.
* This function allows the user to determine the current number of samples in
* the delay buffer, which can be useful for synchronization.
*
* @see avresample_convert()
*
* @param avr audio resample context
* @return number of samples currently in the resampling delay buffer
*/
int avresample_get_delay(AVAudioResampleContext *avr);
/**
* Return the number of available samples in the output FIFO.
*
* During conversion, if the user does not specify an output buffer or
* specifies an output buffer that is smaller than what is needed, remaining
* samples that are not written to the output are stored to an internal FIFO
* buffer. The samples in the FIFO can be read with avresample_read() or
* avresample_convert().
*
* @see avresample_read()
* @see avresample_convert()
*
* @param avr audio resample context
* @return number of samples available for reading
*/
int avresample_available(AVAudioResampleContext *avr);
/**
* Read samples from the output FIFO.
*
* During conversion, if the user does not specify an output buffer or
* specifies an output buffer that is smaller than what is needed, remaining
* samples that are not written to the output are stored to an internal FIFO
* buffer. This function can be used to read samples from that internal FIFO.
*
* @see avresample_available()
* @see avresample_convert()
*
* @param avr audio resample context
* @param output output data pointers. May be NULL, in which case
* nb_samples of data is discarded from output FIFO.
* @param nb_samples number of samples to read from the FIFO
* @return the number of samples written to output
*/
int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
/**
* 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.
*
* The upper bound on the number of output samples is obtained through
* avresample_get_out_samples().
*
* If the output AVFrame does not have the data pointers allocated the nb_samples
* field will be set using avresample_get_out_samples() and 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 avresample_convert() or to avresample_read().
*
* If converting sample rate, there may be data remaining in the internal
* resampling delay buffer. avresample_get_delay() tells the number of
* remaining samples. To get this data as output, call this function or
* avresample_convert() with NULL input.
*
* At the end of the conversion process, there may be data remaining in the
* internal FIFO buffer. avresample_available() tells the number of remaining
* samples. To get this data as output, either call this function or
* avresample_convert() with NULL input or call avresample_read().
*
* If the AVAudioResampleContext 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 AVERROR_OUTPUT_CHANGED|AVERROR_INPUT_CHANGED is returned.
*
* @see avresample_get_out_samples()
* @see avresample_available()
* @see avresample_convert()
* @see avresample_read()
* @see avresample_get_delay()
*
* @param avr audio resample context
* @param output output AVFrame
* @param input input AVFrame
* @return 0 on success, AVERROR on failure or nonmatching
* configuration.
*/
int avresample_convert_frame(AVAudioResampleContext *avr,
AVFrame *output, AVFrame *input);
/**
* Configure or reconfigure the AVAudioResampleContext using the information
* provided by the AVFrames.
*
* The original resampling context is reset even on failure.
* The function calls avresample_close() internally if the context is open.
*
* @see avresample_open();
* @see avresample_close();
*
* @param avr audio resample context
* @param out output AVFrame
* @param in input AVFrame
* @return 0 on success, AVERROR on failure.
*/
int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in);
/**
* @}
*/
#endif /* AVRESAMPLE_AVRESAMPLE_H */