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/*
* Copyright (C) 2011-2012 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
*/
/**
* @file
* libswresample public header
*/
#ifndef SWRESAMPLE_SWRESAMPLE_H
#define SWRESAMPLE_SWRESAMPLE_H
#include <inttypes.h>
#include "libavutil/samplefmt.h"
#include "libswresample/version.h"
#if LIBSWRESAMPLE_VERSION_MAJOR < 1
#define SWR_CH_MAX 32 ///< Maximum number of channels
#endif
#define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
//TODO use int resample ?
//long term TODO can we enable this dynamically?
enum SwrDitherType {
SWR_DITHER_NONE = 0,
SWR_DITHER_RECTANGULAR,
SWR_DITHER_TRIANGULAR,
SWR_DITHER_TRIANGULAR_HIGHPASS,
SWR_DITHER_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 */
};
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().
*/
const AVClass *swr_get_class(void);
/**
* 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
*/
struct SwrContext *swr_alloc(void);
/**
* Initialize context after user parameters have been set.
*
* @return AVERROR error code in case of failure.
*/
int swr_init(struct 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 Swr context, can be NULL
* @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
*/
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);
/**
* Free the given SwrContext and set the pointer to NULL.
*/
void swr_free(struct SwrContext **s);
/**
* 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 providing more output space than input.
* Convertion 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
* timestampe are in 1/(in_sample_rate * out_sample_rate) units.
*
* @note There are 2 slightly differently behaving modes.
* First is when automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
* in this case timestamps will be passed through with delays compensated
* Second is when automatic timestamp compensation is used, (min_compensation < FLT_MAX)
* in this case the output timestamps will match output sample numbers
*
* @param pts timstamp for the next input sample, INT64_MIN if unknown
* @returns the output timestamp for the next output sample
*/
int64_t swr_next_pts(struct SwrContext *s, int64_t pts);
/**
* Activate resampling compensation.
*/
int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance);
/**
* Set a customized input channel mapping.
*
* @param s allocated Swr context, not yet initialized
* @param channel_map customized input channel mapping (array of channel
* indexes, -1 for a muted channel)
* @return AVERROR error code in case of failure.
*/
int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map);
/**
* 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 AVERROR error code in case of failure.
*/
int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride);
/**
* Drops the specified number of output samples.
*/
int swr_drop_output(struct SwrContext *s, int count);
/**
* Injects the specified number of silence samples.
*/
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.
*
* @param s swr context
* @param base timebase in which the returned delay will be
* if its set to 1 the returned delay is in seconds
* if its set to 1000 the returned delay is in milli seconds
* if its set to the input sample rate then the returned delay is in input samples
* if its set to the output sample rate then the returned delay is in output samples
* an exact rounding free delay can be found by using LCM(in_sample_rate, out_sample_rate)
* @returns the delay in 1/base units.
*/
int64_t swr_get_delay(struct SwrContext *s, int64_t base);
/**
* Return the LIBSWRESAMPLE_VERSION_INT constant.
*/
unsigned swresample_version(void);
/**
* Return the swr build-time configuration.
*/
const char *swresample_configuration(void);
/**
* Return the swr license.
*/
const char *swresample_license(void);
#endif /* SWRESAMPLE_SWRESAMPLE_H */
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