--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/lib/libswresample/swresample.h	Tue Dec 31 14:52:14 2013 +0000
@@ -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 "lib/libavutil/samplefmt.h"
+
+#include "lib/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 */