opus headers added

1 files changed, 981 insertions, 0 deletions

diff --git a/include/opus/opus.h b/include/opus/opus.h
new file mode 100644
index 0000000000..d282f21d25
--- /dev/null
+++ b/include/opus/opus.h
@@ -0,0 +1,981 @@
+/* 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 2 bytes or less, 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>opus_int16[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],
+   opus_int16 size[48],
+   int *payload_offset
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5);
+
+/** 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_BAD_ARG Insufficient data was passed to the function
+  * @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_BAD_ARG Insufficient data was passed to the function
+  * @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_BAD_ARG Insufficient data was passed to the function
+  * @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);
+
+/** Applies soft-clipping to bring a float signal within the [-1,1] range. If
+  * the signal is already in that range, nothing is done. If there are values
+  * outside of [-1,1], then the signal is clipped as smoothly as possible to
+  * both fit in the range and avoid creating excessive distortion in the
+  * process.
+  * @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
+  * @param [in] frame_size <tt>int</tt> Number of samples per channel to process
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
+  */
+OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
+
+
+/**@}*/
+
+/** @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 120 ms limit comes from the
+  * specification and limits decoder memory requirements at a point where
+  * framing overhead becomes negligible.
+  *
+  * 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);
+
+/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len);
+
+/** Remove all padding from a given Opus packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len);
+
+/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams);
+
+/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_H */