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Diffstat (limited to 'src/jogl/native/libav/lavc54.lavf54.lavu52.lavr01/libavresample/avresample.h')
| -rw-r--r-- | src/jogl/native/libav/lavc54.lavf54.lavu52.lavr01/libavresample/avresample.h | 379 |
1 files changed, 0 insertions, 379 deletions
diff --git a/src/jogl/native/libav/lavc54.lavf54.lavu52.lavr01/libavresample/avresample.h b/src/jogl/native/libav/lavc54.lavf54.lavu52.lavr01/libavresample/avresample.h deleted file mode 100644 index 001278740..000000000 --- a/src/jogl/native/libav/lavc54.lavf54.lavu52.lavr01/libavresample/avresample.h +++ /dev/null @@ -1,379 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <[email protected]> - * - * This file is part of Libav. - * - * Libav 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. - * - * Libav 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 Libav; 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_available(avr) + - * av_rescale_rnd(avresample_get_delay(avr) + - * in_samples, 44100, 48000, AV_ROUND_UP); - * 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/log.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. - * - * @param avr audio resample context - * @return 0 on success, negative AVERROR code on failure - */ -int avresample_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 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); - -/** - * Convert input samples and write them to the output FIFO. - * - * The upper bound on the number of output samples is given by - * avresample_available() + (avresample_get_delay() + number of input samples) * - * output sample rate / input sample rate. - * - * 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_available() - * @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 **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); - -/** - * @} - */ - -#endif /* AVRESAMPLE_AVRESAMPLE_H */ |