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* Copyright (C) 2011-2012 Michael Niedermayer (
* 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
* 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
* @defgroup lswr Libswresample
* @{
* Libswresample (lswr) is a library that handles audio resampling, sample
* format conversion and mixing.
* 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.
* 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
* SwrContext *swr = swr_alloc();
* av_opt_set_int(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
* av_opt_set_int(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_int(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
* av_opt_set_int(swr, "out_sample_fmt, AV_SAMPLE_FMT_S16, 0);
* @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
* as described above, or by using swr_alloc_set_opts(), 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 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().
* There will be no memory leak if the data is not completely flushed before
* swr_free().
#include <stdint.h>
#include "libavutil/samplefmt.h"
#include "libswresample/version.h"
#define SWR_CH_MAX 32 ///< Maximum number of channels
#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_NB, ///< not part of API/ABI
/** Resampling Filter Types */
enum SwrFilterType {
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
* timestamps 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 timestamp for the next input sample, INT64_MIN if unknown
* @return 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.
* 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
* 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);
* @}
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