Upgrade opus 1.0.2 -> 1.1

1 files changed, 0 insertions, 903 deletions

diff --git a/src/opus-1.0.2/include/opus.h b/src/opus-1.0.2/include/opus.h
deleted file mode 100644
index 847a07c1..00000000
--- a/src/opus-1.0.2/include/opus.h
+++ /dev/null
@@ -1,903 +0,0 @@
-/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
-   Written by Jean-Marc Valin and Koen Vos */
-/*
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
-   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-/**
- * @file opus.h
- * @brief Opus reference implementation API
- */
-
-#ifndef OPUS_H
-#define OPUS_H
-
-#include "opus_types.h"
-#include "opus_defines.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/**
- * @mainpage Opus
- *
- * The Opus codec is designed for interactive speech and audio transmission over the Internet.
- * It is designed by the IETF Codec Working Group and incorporates technology from
- * Skype's SILK codec and Xiph.Org's CELT codec.
- *
- * The Opus codec is designed to handle a wide range of interactive audio applications,
- * including Voice over IP, videoconferencing, in-game chat, and even remote live music
- * performances. It can scale from low bit-rate narrowband speech to very high quality
- * stereo music. Its main features are:
-
- * @li Sampling rates from 8 to 48 kHz
- * @li Bit-rates from 6 kb/s to 510 kb/s
- * @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
- * @li Audio bandwidth from narrowband to full-band
- * @li Support for speech and music
- * @li Support for mono and stereo
- * @li Support for multichannel (up to 255 channels)
- * @li Frame sizes from 2.5 ms to 60 ms
- * @li Good loss robustness and packet loss concealment (PLC)
- * @li Floating point and fixed-point implementation
- *
- * Documentation sections:
- * @li @ref opus_encoder
- * @li @ref opus_decoder
- * @li @ref opus_repacketizer
- * @li @ref opus_multistream
- * @li @ref opus_libinfo
- * @li @ref opus_custom
- */
-
-/** @defgroup opus_encoder Opus Encoder
-  * @{
-  *
-  * @brief This page describes the process and functions used to encode Opus.
-  *
-  * Since Opus is a stateful codec, the encoding process starts with creating an encoder
-  * state. This can be done with:
-  *
-  * @code
-  * int          error;
-  * OpusEncoder *enc;
-  * enc = opus_encoder_create(Fs, channels, application, &error);
-  * @endcode
-  *
-  * From this point, @c enc can be used for encoding an audio stream. An encoder state
-  * @b must @b not be used for more than one stream at the same time. Similarly, the encoder
-  * state @b must @b not be re-initialized for each frame.
-  *
-  * While opus_encoder_create() allocates memory for the state, it's also possible
-  * to initialize pre-allocated memory:
-  *
-  * @code
-  * int          size;
-  * int          error;
-  * OpusEncoder *enc;
-  * size = opus_encoder_get_size(channels);
-  * enc = malloc(size);
-  * error = opus_encoder_init(enc, Fs, channels, application);
-  * @endcode
-  *
-  * where opus_encoder_get_size() returns the required size for the encoder state. Note that
-  * future versions of this code may change the size, so no assuptions should be made about it.
-  *
-  * The encoder state is always continuous in memory and only a shallow copy is sufficient
-  * to copy it (e.g. memcpy())
-  *
-  * It is possible to change some of the encoder's settings using the opus_encoder_ctl()
-  * interface. All these settings already default to the recommended value, so they should
-  * only be changed when necessary. The most common settings one may want to change are:
-  *
-  * @code
-  * opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
-  * opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
-  * opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
-  * @endcode
-  *
-  * where
-  *
-  * @arg bitrate is in bits per second (b/s)
-  * @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
-  * @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
-  *
-  * See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream.
-  *
-  * To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data:
-  * @code
-  * len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
-  * @endcode
-  *
-  * where
-  * <ul>
-  * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
-  * <li>frame_size is the duration of the frame in samples (per channel)</li>
-  * <li>packet is the byte array to which the compressed data is written</li>
-  * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended).
-  *     Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li>
-  * </ul>
-  *
-  * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
-  * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
-  * is 1 byte, then the packet does not need to be transmitted (DTX).
-  *
-  * Once the encoder state if no longer needed, it can be destroyed with
-  *
-  * @code
-  * opus_encoder_destroy(enc);
-  * @endcode
-  *
-  * If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
-  * then no action is required aside from potentially freeing the memory that was manually
-  * allocated for it (calling free(enc) for the example above)
-  *
-  */
-
-/** Opus encoder state.
-  * This contains the complete state of an Opus encoder.
-  * It is position independent and can be freely copied.
-  * @see opus_encoder_create,opus_encoder_init
-  */
-typedef struct OpusEncoder OpusEncoder;
-
-/** Gets the size of an <code>OpusEncoder</code> structure.
-  * @param[in] channels <tt>int</tt>: Number of channels.
-  *                                   This must be 1 or 2.
-  * @returns The size in bytes.
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
-
-/**
- */
-
-/** Allocates and initializes an encoder state.
- * There are three coding modes:
- *
- * @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
- *    signals. It enhances the  input signal by high-pass filtering and
- *    emphasizing formants and harmonics. Optionally  it includes in-band
- *    forward error correction to protect against packet loss. Use this
- *    mode for typical VoIP applications. Because of the enhancement,
- *    even at high bitrates the output may sound different from the input.
- *
- * @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
- *    non-voice signals like music. Use this mode for music and mixed
- *    (music/voice) content, broadcast, and applications requiring less
- *    than 15 ms of coding delay.
- *
- * @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
- *    disables the speech-optimized mode in exchange for slightly reduced delay.
- *    This mode can only be set on an newly initialized or freshly reset encoder
- *    because it changes the codec delay.
- *
- * This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
- * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
- *                                     This must be one of 8000, 12000, 16000,
- *                                     24000, or 48000.
- * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
- * @param [in] application <tt>int</tt>: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
- * @param [out] error <tt>int*</tt>: @ref opus_errorcodes
- * @note Regardless of the sampling rate and number channels selected, the Opus encoder
- * can switch to a lower audio bandwidth or number of channels if the bitrate
- * selected is too low. This also means that it is safe to always use 48 kHz stereo input
- * and let the encoder optimize the encoding.
- */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
-    opus_int32 Fs,
-    int channels,
-    int application,
-    int *error
-);
-
-/** Initializes a previously allocated encoder state
-  * The memory pointed to by st must be at least the size returned by opus_encoder_get_size().
-  * This is intended for applications which use their own allocator instead of malloc.
-  * @see opus_encoder_create(),opus_encoder_get_size()
-  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
-  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
-  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
- *                                      This must be one of 8000, 12000, 16000,
- *                                      24000, or 48000.
-  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
-  * @param [in] application <tt>int</tt>: Coding mode (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY)
-  * @retval #OPUS_OK Success or @ref opus_errorcodes
-  */
-OPUS_EXPORT int opus_encoder_init(
-    OpusEncoder *st,
-    opus_int32 Fs,
-    int channels,
-    int application
-) OPUS_ARG_NONNULL(1);
-
-/** Encodes an Opus frame.
-  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
-  * @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
-  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
-  *                                      input signal.
-  *                                      This must be an Opus frame size for
-  *                                      the encoder's sampling rate.
-  *                                      For example, at 48 kHz the permitted
-  *                                      values are 120, 240, 480, 960, 1920,
-  *                                      and 2880.
-  *                                      Passing in a duration of less than
-  *                                      10 ms (480 samples at 48 kHz) will
-  *                                      prevent the encoder from using the LPC
-  *                                      or hybrid modes.
-  * @param [out] data <tt>unsigned char*</tt>: Output payload.
-  *                                            This must contain storage for at
-  *                                            least \a max_data_bytes.
-  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
-  *                                                 memory for the output
-  *                                                 payload. This may be
-  *                                                 used to impose an upper limit on
-  *                                                 the instant bitrate, but should
-  *                                                 not be used as the only bitrate
-  *                                                 control. Use #OPUS_SET_BITRATE to
-  *                                                 control the bitrate.
-  * @returns The length of the encoded packet (in bytes) on success or a
-  *          negative error code (see @ref opus_errorcodes) on failure.
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
-    OpusEncoder *st,
-    const opus_int16 *pcm,
-    int frame_size,
-    unsigned char *data,
-    opus_int32 max_data_bytes
-) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
-
-/** Encodes an Opus frame from floating point input.
-  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
-  * @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
-  *          Samples with a range beyond +/-1.0 are supported but will
-  *          be clipped by decoders using the integer API and should
-  *          only be used if it is known that the far end supports
-  *          extended dynamic range.
-  *          length is frame_size*channels*sizeof(float)
-  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
-  *                                      input signal.
-  *                                      This must be an Opus frame size for
-  *                                      the encoder's sampling rate.
-  *                                      For example, at 48 kHz the permitted
-  *                                      values are 120, 240, 480, 960, 1920,
-  *                                      and 2880.
-  *                                      Passing in a duration of less than
-  *                                      10 ms (480 samples at 48 kHz) will
-  *                                      prevent the encoder from using the LPC
-  *                                      or hybrid modes.
-  * @param [out] data <tt>unsigned char*</tt>: Output payload.
-  *                                            This must contain storage for at
-  *                                            least \a max_data_bytes.
-  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
-  *                                                 memory for the output
-  *                                                 payload. This may be
-  *                                                 used to impose an upper limit on
-  *                                                 the instant bitrate, but should
-  *                                                 not be used as the only bitrate
-  *                                                 control. Use #OPUS_SET_BITRATE to
-  *                                                 control the bitrate.
-  * @returns The length of the encoded packet (in bytes) on success or a
-  *          negative error code (see @ref opus_errorcodes) on failure.
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float(
-    OpusEncoder *st,
-    const float *pcm,
-    int frame_size,
-    unsigned char *data,
-    opus_int32 max_data_bytes
-) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
-
-/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create().
-  * @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
-  */
-OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
-
-/** Perform a CTL function on an Opus encoder.
-  *
-  * Generally the request and subsequent arguments are generated
-  * by a convenience macro.
-  * @param st <tt>OpusEncoder*</tt>: Encoder state.
-  * @param request This and all remaining parameters should be replaced by one
-  *                of the convenience macros in @ref opus_genericctls or
-  *                @ref opus_encoderctls.
-  * @see opus_genericctls
-  * @see opus_encoderctls
-  */
-OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
-/**@}*/
-
-/** @defgroup opus_decoder Opus Decoder
-  * @{
-  *
-  * @brief This page describes the process and functions used to decode Opus.
-  *
-  * The decoding process also starts with creating a decoder
-  * state. This can be done with:
-  * @code
-  * int          error;
-  * OpusDecoder *dec;
-  * dec = opus_decoder_create(Fs, channels, &error);
-  * @endcode
-  * where
-  * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
-  * @li channels is the number of channels (1 or 2)
-  * @li error will hold the error code in case of failure (or #OPUS_OK on success)
-  * @li the return value is a newly created decoder state to be used for decoding
-  *
-  * While opus_decoder_create() allocates memory for the state, it's also possible
-  * to initialize pre-allocated memory:
-  * @code
-  * int          size;
-  * int          error;
-  * OpusDecoder *dec;
-  * size = opus_decoder_get_size(channels);
-  * dec = malloc(size);
-  * error = opus_decoder_init(dec, Fs, channels);
-  * @endcode
-  * where opus_decoder_get_size() returns the required size for the decoder state. Note that
-  * future versions of this code may change the size, so no assuptions should be made about it.
-  *
-  * The decoder state is always continuous in memory and only a shallow copy is sufficient
-  * to copy it (e.g. memcpy())
-  *
-  * To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
-  * @code
-  * frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
-  * @endcode
-  * where
-  *
-  * @li packet is the byte array containing the compressed data
-  * @li len is the exact number of bytes contained in the packet
-  * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
-  * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
-  *
-  * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
-  * If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio
-  * buffer is too small to hold the decoded audio.
-  *
-  * Opus is a stateful codec with overlapping blocks and as a result Opus
-  * packets are not coded independently of each other. Packets must be
-  * passed into the decoder serially and in the correct order for a correct
-  * decode. Lost packets can be replaced with loss concealment by calling
-  * the decoder with a null pointer and zero length for the missing packet.
-  *
-  * A single codec state may only be accessed from a single thread at
-  * a time and any required locking must be performed by the caller. Separate
-  * streams must be decoded with separate decoder states and can be decoded
-  * in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK
-  * defined.
-  *
-  */
-
-/** Opus decoder state.
-  * This contains the complete state of an Opus decoder.
-  * It is position independent and can be freely copied.
-  * @see opus_decoder_create,opus_decoder_init
-  */
-typedef struct OpusDecoder OpusDecoder;
-
-/** Gets the size of an <code>OpusDecoder</code> structure.
-  * @param [in] channels <tt>int</tt>: Number of channels.
-  *                                    This must be 1 or 2.
-  * @returns The size in bytes.
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
-
-/** Allocates and initializes a decoder state.
-  * @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
-  *                                     This must be one of 8000, 12000, 16000,
-  *                                     24000, or 48000.
-  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
-  * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
-  *
-  * Internally Opus stores data at 48000 Hz, so that should be the default
-  * value for Fs. However, the decoder can efficiently decode to buffers
-  * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
-  * data at the full sample rate, or knows the compressed data doesn't
-  * use the full frequency range, it can request decoding at a reduced
-  * rate. Likewise, the decoder is capable of filling in either mono or
-  * interleaved stereo pcm buffers, at the caller's request.
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
-    opus_int32 Fs,
-    int channels,
-    int *error
-);
-
-/** Initializes a previously allocated decoder state.
-  * The state must be at least the size returned by opus_decoder_get_size().
-  * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
-  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
-  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
-  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
-  *                                     This must be one of 8000, 12000, 16000,
-  *                                     24000, or 48000.
-  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
-  * @retval #OPUS_OK Success or @ref opus_errorcodes
-  */
-OPUS_EXPORT int opus_decoder_init(
-    OpusDecoder *st,
-    opus_int32 Fs,
-    int channels
-) OPUS_ARG_NONNULL(1);
-
-/** Decode an Opus packet.
-  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
-  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
-  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
-  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
-  *  is frame_size*channels*sizeof(opus_int16)
-  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
-  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
-  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
-  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
-  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
-  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
-  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
-  *  decoded. If no such data is available, the frame is decoded as if it were lost.
-  * @returns Number of decoded samples or @ref opus_errorcodes
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
-    OpusDecoder *st,
-    const unsigned char *data,
-    opus_int32 len,
-    opus_int16 *pcm,
-    int frame_size,
-    int decode_fec
-) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
-
-/** Decode an Opus packet with floating point output.
-  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
-  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
-  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
-  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
-  *  is frame_size*channels*sizeof(float)
-  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
-  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
-  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
-  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
-  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
-  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
-  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
-  *  decoded. If no such data is available the frame is decoded as if it were lost.
-  * @returns Number of decoded samples or @ref opus_errorcodes
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
-    OpusDecoder *st,
-    const unsigned char *data,
-    opus_int32 len,
-    float *pcm,
-    int frame_size,
-    int decode_fec
-) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
-
-/** Perform a CTL function on an Opus decoder.
-  *
-  * Generally the request and subsequent arguments are generated
-  * by a convenience macro.
-  * @param st <tt>OpusDecoder*</tt>: Decoder state.
-  * @param request This and all remaining parameters should be replaced by one
-  *                of the convenience macros in @ref opus_genericctls or
-  *                @ref opus_decoderctls.
-  * @see opus_genericctls
-  * @see opus_decoderctls
-  */
-OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
-
-/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
-  * @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
-  */
-OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
-
-/** Parse an opus packet into one or more frames.
-  * Opus_decode will perform this operation internally so most applications do
-  * not need to use this function.
-  * This function does not copy the frames, the returned pointers are pointers into
-  * the input packet.
-  * @param [in] data <tt>char*</tt>: Opus packet to be parsed
-  * @param [in] len <tt>opus_int32</tt>: size of data
-  * @param [out] out_toc <tt>char*</tt>: TOC pointer
-  * @param [out] frames <tt>char*[48]</tt> encapsulated frames
-  * @param [out] size <tt>short[48]</tt> sizes of the encapsulated frames
-  * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
-  * @returns number of frames
-  */
-OPUS_EXPORT int opus_packet_parse(
-   const unsigned char *data,
-   opus_int32 len,
-   unsigned char *out_toc,
-   const unsigned char *frames[48],
-   short size[48],
-   int *payload_offset
-) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
-
-/** Gets the bandwidth of an Opus packet.
-  * @param [in] data <tt>char*</tt>: Opus packet
-  * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
-  * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
-  * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
-  * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
-  * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
-  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
-
-/** Gets the number of samples per frame from an Opus packet.
-  * @param [in] data <tt>char*</tt>: Opus packet.
-  *                                  This must contain at least one byte of
-  *                                  data.
-  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
-  *                                     This must be a multiple of 400, or
-  *                                     inaccurate results will be returned.
-  * @returns Number of samples per frame.
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
-
-/** Gets the number of channels from an Opus packet.
-  * @param [in] data <tt>char*</tt>: Opus packet
-  * @returns Number of channels
-  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
-
-/** Gets the number of frames in an Opus packet.
-  * @param [in] packet <tt>char*</tt>: Opus packet
-  * @param [in] len <tt>opus_int32</tt>: Length of packet
-  * @returns Number of frames
-  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
-
-/** Gets the number of samples of an Opus packet.
-  * @param [in] packet <tt>char*</tt>: Opus packet
-  * @param [in] len <tt>opus_int32</tt>: Length of packet
-  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
-  *                                     This must be a multiple of 400, or
-  *                                     inaccurate results will be returned.
-  * @returns Number of samples
-  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
-
-/** Gets the number of samples of an Opus packet.
-  * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
-  * @param [in] packet <tt>char*</tt>: Opus packet
-  * @param [in] len <tt>opus_int32</tt>: Length of packet
-  * @returns Number of samples
-  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
-/**@}*/
-
-/** @defgroup opus_repacketizer Repacketizer
-  * @{
-  *
-  * The repacketizer can be used to merge multiple Opus packets into a single
-  * packet or alternatively to split Opus packets that have previously been
-  * merged. Splitting valid Opus packets is always guaranteed to succeed,
-  * whereas merging valid packets only succeeds if all frames have the same
-  * mode, bandwidth, and frame size, and when the total duration of the merged
-  * packet is no more than 120 ms.
-  * The repacketizer currently only operates on elementary Opus
-  * streams. It will not manipualte multistream packets successfully, except in
-  * the degenerate case where they consist of data from a single stream.
-  *
-  * The repacketizing process starts with creating a repacketizer state, either
-  * by calling opus_repacketizer_create() or by allocating the memory yourself,
-  * e.g.,
-  * @code
-  * OpusRepacketizer *rp;
-  * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
-  * if (rp != NULL)
-  *     opus_repacketizer_init(rp);
-  * @endcode
-  *
-  * Then the application should submit packets with opus_repacketizer_cat(),
-  * extract new packets with opus_repacketizer_out() or
-  * opus_repacketizer_out_range(), and then reset the state for the next set of
-  * input packets via opus_repacketizer_init().
-  *
-  * For example, to split a sequence of packets into individual frames:
-  * @code
-  * unsigned char *data;
-  * int len;
-  * while (get_next_packet(&data, &len))
-  * {
-  *   unsigned char out[1276];
-  *   opus_int32 out_len;
-  *   int nb_frames;
-  *   int err;
-  *   int i;
-  *   err = opus_repacketizer_cat(rp, data, len);
-  *   if (err != OPUS_OK)
-  *   {
-  *     release_packet(data);
-  *     return err;
-  *   }
-  *   nb_frames = opus_repacketizer_get_nb_frames(rp);
-  *   for (i = 0; i < nb_frames; i++)
-  *   {
-  *     out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
-  *     if (out_len < 0)
-  *     {
-  *        release_packet(data);
-  *        return (int)out_len;
-  *     }
-  *     output_next_packet(out, out_len);
-  *   }
-  *   opus_repacketizer_init(rp);
-  *   release_packet(data);
-  * }
-  * @endcode
-  *
-  * Alternatively, to combine a sequence of frames into packets that each
-  * contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
-  * @code
-  * // The maximum number of packets with duration TARGET_DURATION_MS occurs
-  * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
-  * // packets.
-  * unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
-  * opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
-  * int nb_packets;
-  * unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
-  * opus_int32 out_len;
-  * int prev_toc;
-  * nb_packets = 0;
-  * while (get_next_packet(data+nb_packets, len+nb_packets))
-  * {
-  *   int nb_frames;
-  *   int err;
-  *   nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
-  *   if (nb_frames < 1)
-  *   {
-  *     release_packets(data, nb_packets+1);
-  *     return nb_frames;
-  *   }
-  *   nb_frames += opus_repacketizer_get_nb_frames(rp);
-  *   // If adding the next packet would exceed our target, or it has an
-  *   // incompatible TOC sequence, output the packets we already have before
-  *   // submitting it.
-  *   // N.B., The nb_packets > 0 check ensures we've submitted at least one
-  *   // packet since the last call to opus_repacketizer_init(). Otherwise a
-  *   // single packet longer than TARGET_DURATION_MS would cause us to try to
-  *   // output an (invalid) empty packet. It also ensures that prev_toc has
-  *   // been set to a valid value. Additionally, len[nb_packets] > 0 is
-  *   // guaranteed by the call to opus_packet_get_nb_frames() above, so the
-  *   // reference to data[nb_packets][0] should be valid.
-  *   if (nb_packets > 0 && (
-  *       ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
-  *       opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
-  *       TARGET_DURATION_MS*48))
-  *   {
-  *     out_len = opus_repacketizer_out(rp, out, sizeof(out));
-  *     if (out_len < 0)
-  *     {
-  *        release_packets(data, nb_packets+1);
-  *        return (int)out_len;
-  *     }
-  *     output_next_packet(out, out_len);
-  *     opus_repacketizer_init(rp);
-  *     release_packets(data, nb_packets);
-  *     data[0] = data[nb_packets];
-  *     len[0] = len[nb_packets];
-  *     nb_packets = 0;
-  *   }
-  *   err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
-  *   if (err != OPUS_OK)
-  *   {
-  *     release_packets(data, nb_packets+1);
-  *     return err;
-  *   }
-  *   prev_toc = data[nb_packets][0];
-  *   nb_packets++;
-  * }
-  * // Output the final, partial packet.
-  * if (nb_packets > 0)
-  * {
-  *   out_len = opus_repacketizer_out(rp, out, sizeof(out));
-  *   release_packets(data, nb_packets);
-  *   if (out_len < 0)
-  *     return (int)out_len;
-  *   output_next_packet(out, out_len);
-  * }
-  * @endcode
-  *
-  * An alternate way of merging packets is to simply call opus_repacketizer_cat()
-  * unconditionally until it fails. At that point, the merged packet can be
-  * obtained with opus_repacketizer_out() and the input packet for which
-  * opus_repacketizer_cat() needs to be re-added to a newly reinitialized
-  * repacketizer state.
-  */
-
-typedef struct OpusRepacketizer OpusRepacketizer;
-
-/** Gets the size of an <code>OpusRepacketizer</code> structure.
-  * @returns The size in bytes.
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
-
-/** (Re)initializes a previously allocated repacketizer state.
-  * The state must be at least the size returned by opus_repacketizer_get_size().
-  * This can be used for applications which use their own allocator instead of
-  * malloc().
-  * It must also be called to reset the queue of packets waiting to be
-  * repacketized, which is necessary if the maximum packet duration of 120 ms
-  * is reached or if you wish to submit packets with a different Opus
-  * configuration (coding mode, audio bandwidth, frame size, or channel count).
-  * Failure to do so will prevent a new packet from being added with
-  * opus_repacketizer_cat().
-  * @see opus_repacketizer_create
-  * @see opus_repacketizer_get_size
-  * @see opus_repacketizer_cat
-  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
-  *                                       (re)initialize.
-  * @returns A pointer to the same repacketizer state that was passed in.
-  */
-OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
-
-/** Allocates memory and initializes the new repacketizer with
- * opus_repacketizer_init().
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
-
-/** Frees an <code>OpusRepacketizer</code> allocated by
-  * opus_repacketizer_create().
-  * @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
-  */
-OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
-
-/** Add a packet to the current repacketizer state.
-  * This packet must match the configuration of any packets already submitted
-  * for repacketization since the last call to opus_repacketizer_init().
-  * This means that it must have the same coding mode, audio bandwidth, frame
-  * size, and channel count.
-  * This can be checked in advance by examining the top 6 bits of the first
-  * byte of the packet, and ensuring they match the top 6 bits of the first
-  * byte of any previously submitted packet.
-  * The total duration of audio in the repacketizer state also must not exceed
-  * 120 ms, the maximum duration of a single packet, after adding this packet.
-  *
-  * The contents of the current repacketizer state can be extracted into new
-  * packets using opus_repacketizer_out() or opus_repacketizer_out_range().
-  *
-  * In order to add a packet with a different configuration or to add more
-  * audio beyond 120 ms, you must clear the repacketizer state by calling
-  * opus_repacketizer_init().
-  * If a packet is too large to add to the current repacketizer state, no part
-  * of it is added, even if it contains multiple frames, some of which might
-  * fit.
-  * If you wish to be able to add parts of such packets, you should first use
-  * another repacketizer to split the packet into pieces and add them
-  * individually.
-  * @see opus_repacketizer_out_range
-  * @see opus_repacketizer_out
-  * @see opus_repacketizer_init
-  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
-  *                                       add the packet.
-  * @param[in] data <tt>const unsigned char*</tt>: The packet data.
-  *                                                The application must ensure
-  *                                                this pointer remains valid
-  *                                                until the next call to
-  *                                                opus_repacketizer_init() or
-  *                                                opus_repacketizer_destroy().
-  * @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
-  * @returns An error code indicating whether or not the operation succeeded.
-  * @retval #OPUS_OK The packet's contents have been added to the repacketizer
-  *                  state.
-  * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
-  *                              the packet's TOC sequence was not compatible
-  *                              with previously submitted packets (because
-  *                              the coding mode, audio bandwidth, frame size,
-  *                              or channel count did not match), or adding
-  *                              this packet would increase the total amount of
-  *                              audio stored in the repacketizer state to more
-  *                              than 120 ms.
-  */
-OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
-
-
-/** Construct a new packet from data previously submitted to the repacketizer
-  * state via opus_repacketizer_cat().
-  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
-  *                                       construct the new packet.
-  * @param begin <tt>int</tt>: The index of the first frame in the current
-  *                            repacketizer state to include in the output.
-  * @param end <tt>int</tt>: One past the index of the last frame in the
-  *                          current repacketizer state to include in the
-  *                          output.
-  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
-  *                                                 store the output packet.
-  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
-  *                                    the output buffer. In order to guarantee
-  *                                    success, this should be at least
-  *                                    <code>1276</code> for a single frame,
-  *                                    or for multiple frames,
-  *                                    <code>1277*(end-begin)</code>.
-  *                                    However, <code>1*(end-begin)</code> plus
-  *                                    the size of all packet data submitted to
-  *                                    the repacketizer since the last call to
-  *                                    opus_repacketizer_init() or
-  *                                    opus_repacketizer_create() is also
-  *                                    sufficient, and possibly much smaller.
-  * @returns The total size of the output packet on success, or an error code
-  *          on failure.
-  * @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
-  *                       frames (begin < 0, begin >= end, or end >
-  *                       opus_repacketizer_get_nb_frames()).
-  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
-  *                                complete output packet.
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
-
-/** Return the total number of frames contained in packet data submitted to
-  * the repacketizer state so far via opus_repacketizer_cat() since the last
-  * call to opus_repacketizer_init() or opus_repacketizer_create().
-  * This defines the valid range of packets that can be extracted with
-  * opus_repacketizer_out_range() or opus_repacketizer_out().
-  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
-  *                                       frames.
-  * @returns The total number of frames contained in the packet data submitted
-  *          to the repacketizer state.
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
-
-/** Construct a new packet from data previously submitted to the repacketizer
-  * state via opus_repacketizer_cat().
-  * This is a convenience routine that returns all the data submitted so far
-  * in a single packet.
-  * It is equivalent to calling
-  * @code
-  * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
-  *                             data, maxlen)
-  * @endcode
-  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
-  *                                       construct the new packet.
-  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
-  *                                                 store the output packet.
-  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
-  *                                    the output buffer. In order to guarantee
-  *                                    success, this should be at least
-  *                                    <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
-  *                                    However,
-  *                                    <code>1*opus_repacketizer_get_nb_frames(rp)</code>
-  *                                    plus the size of all packet data
-  *                                    submitted to the repacketizer since the
-  *                                    last call to opus_repacketizer_init() or
-  *                                    opus_repacketizer_create() is also
-  *                                    sufficient, and possibly much smaller.
-  * @returns The total size of the output packet on success, or an error code
-  *          on failure.
-  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
-  *                                complete output packet.
-  */
-OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
-
-/**@}*/
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif /* OPUS_H */