Fix libav/ffmpeg compilation across platforms: Move header back to 'stub_includes'

1 files changed, 379 insertions, 0 deletions

diff --git a/make/stub_includes/libav/lavc54.lavf54.lavu52.lavr01/libavresample/avresample.h b/make/stub_includes/libav/lavc54.lavf54.lavu52.lavr01/libavresample/avresample.h
new file mode 100644
index 000000000..001278740
--- /dev/null
+++ b/make/stub_includes/libav/lavc54.lavf54.lavu52.lavr01/libavresample/avresample.h
@@ -0,0 +1,379 @@
+/*
+ * 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 */