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Diffstat (limited to 'ffmpeg/libswresample/swresample.h')
| -rw-r--r-- | ffmpeg/libswresample/swresample.h | 311 |
1 files changed, 0 insertions, 311 deletions
diff --git a/ffmpeg/libswresample/swresample.h b/ffmpeg/libswresample/swresample.h deleted file mode 100644 index 3811301..0000000 --- a/ffmpeg/libswresample/swresample.h +++ /dev/null @@ -1,311 +0,0 @@ -/* - * 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_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 - * - * 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 */ |