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Diffstat (limited to 'thirdparties/mac/include/libswresample/swresample.h')
| -rw-r--r-- | thirdparties/mac/include/libswresample/swresample.h | 311 |
1 files changed, 311 insertions, 0 deletions
diff --git a/thirdparties/mac/include/libswresample/swresample.h b/thirdparties/mac/include/libswresample/swresample.h new file mode 100644 index 0000000..95e8a5a --- /dev/null +++ b/thirdparties/mac/include/libswresample/swresample.h @@ -0,0 +1,311 @@ +/* + * 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 + * @{ + * + * 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_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 + * + * 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" + +#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_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 */ +}; + +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); + +/** + * @} + */ + +#endif /* SWRESAMPLE_SWRESAMPLE_H */ |
