opus/include/opus.h

/* 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 FOUNDATION OR
   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
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   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
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   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 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 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 opusencoder
 * @li @ref opusdecoder
 * @li @ref repacketizer
 * @li @ref libinfo
 */

/** @defgroup opusencoder Opus Encoder
  * @{
  *
  * 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 encoderctls and @ref 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 (1276 bytes is recommended)</li>
  * </ul>
  *
  * opus_encode() and opus_encode_frame() 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;

OPUS_EXPORT 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 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)
 * @param [in] channels <tt>int</tt>: Number of channels (1/2) 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 errorcodes
 * @note Regardless of the sampling rate and number channels selected, the Opus encoder
 * can switch to a lower audio 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 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 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)
  * @param [in] channels <tt>int</tt>: Number of channels (1/2) 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 errorcodes
  */
OPUS_EXPORT int opus_encoder_init(
    OpusEncoder *st,
    opus_int32 Fs,
    int channels,
    int application
);

/** Encodes an Opus frame.
  * The passed frame_size must an opus frame size for the encoder's sampling rate.
  * For example, at 48kHz the permitted values are 120, 240, 480, or 960.
  * Passing in a duration of less than 10ms (480 samples at 48kHz) will
  * prevent the encoder from using the LPC or hybrid modes.
  * @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 frame of input signal
  * @param [out] data <tt>char*</tt>: Output payload (at least max_data_bytes long)
  * @param [in] max_data_bytes <tt>int</tt>: Allocated memory for payload; don't use for controlling bitrate
  * @returns length of the data payload (in bytes) or @ref errorcodes
  */
OPUS_EXPORT int opus_encode(
    OpusEncoder *st,
    const opus_int16 *pcm,
    int frame_size,
    unsigned char *data,
    int max_data_bytes
);

/** Encodes an Opus frame from floating point input.
  * The passed frame_size must an opus frame size for the encoder's sampling rate.
  * For example, at 48kHz the permitted values are 120, 240, 480, or 960.
  * Passing in a duration of less than 10ms (480 samples at 48kHz) will
  * prevent the encoder from using the LPC or hybrid modes.
  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
  * @param [in] pcm <tt>float*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(float)
  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
  * @param [out] data <tt>char*</tt>: Output payload (at least max_data_bytes long)
  * @param [in] max_data_bytes <tt>int</tt>: Allocated memory for payload; don't use for controlling bitrate
  * @returns length of the data payload (in bytes) or @ref errorcodes
  */
OPUS_EXPORT int opus_encode_float(
    OpusEncoder *st,
    const float *pcm,
    int frame_size,
    unsigned char *data,
    int max_data_bytes
);

/** Frees an OpusEncoder 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.
  * @see encoderctls
  */
OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...);
/**@}*/

/** @defgroup opusdecoder Opus Decoder
  * @{
  *
  *
  * 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 or 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(enc, packet, len, decoded, max_size);
  * @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_frame() return the number of samples ()per channel) decoded from the packet.
  * If that value is negative, then an error has occured. This can occur if the packet is corrupted or if the audio
  * buffer is too small to hold the decoded audio.

*/

/** 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 OpusDecoder structure.
  * @param [in] channels <tt>int</tt>: Number of channels
  * @returns size
  */
OPUS_EXPORT int opus_decoder_get_size(int channels);

/** Allocates and initializes a decoder state.
  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
  * @param [in] channels <tt>int</tt>: Number of channels (1/2) in input signal
  * @param [out] error <tt>int*</tt>: OPUS_OK Success or @ref errorcodes
  */
OPUS_EXPORT OpusDecoder *opus_decoder_create(
    opus_int32 Fs,
    int channels,
    int *error
);

/** Initializes a previously allocated decoder state.
  * The state must be 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 of input signal (Hz)
  * @param [in] channels <tt>int</tt>: Number of channels (1/2) in input signal
  * @retval OPUS_OK Success or @ref errorcodes
  */
OPUS_EXPORT int opus_decoder_init(
    OpusDecoder *st,
    opus_int32 Fs,
    int channels
);

/** Decode an Opus frame
  * @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>int</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 *pcm,
  *  if less than the maximum frame size (120ms) some frames can not be decoded
  * @param [in] decode_fec <tt>int</tt>: Flag (0/1) 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 errorcodes
  */
OPUS_EXPORT int opus_decode(
    OpusDecoder *st,
    const unsigned char *data,
    int len,
    opus_int16 *pcm,
    int frame_size,
    int decode_fec
);

/** Decode an opus frame 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>int</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 *pcm,
  *  if less than the maximum frame size (120ms) some frames can not be decoded
  * @param [in] decode_fec <tt>int</tt>: Flag (0/1) 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 errorcodes
  */
OPUS_EXPORT int opus_decode_float(
    OpusDecoder *st,
    const unsigned char *data,
    int len,
    float *pcm,
    int frame_size,
    int decode_fec
);

/** Perform a CTL function on an Opus decoder.
  * @see decoderctls
  */
OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...);

/** Frees an OpusDecoder 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>int</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,
   int len,
   unsigned char *out_toc,
   const unsigned char *frames[48],
   short size[48],
   int *payload_offset
);

/** 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 int opus_packet_get_bandwidth(const unsigned char *data);

/** Gets the number of samples per frame from an Opus packet.
  * @param [in] data <tt>char*</tt>: Opus packet
  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz
  * @returns Number of samples per frame
  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
  */
OPUS_EXPORT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs);

/** 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 int opus_packet_get_nb_channels(const unsigned char *data);

/** Gets the number of frame in an Opus packet.
  * @param [in] packet <tt>char*</tt>: Opus packet
  * @param [in] len <tt>int</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 int opus_packet_get_nb_frames(const unsigned char packet[], int len);

/** 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>int</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 int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], int len);
/**@}*/

/** @defgroup 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.
  *
  */

typedef struct OpusRepacketizer OpusRepacketizer;

OPUS_EXPORT int opus_repacketizer_get_size(void);

OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp);

OPUS_EXPORT OpusRepacketizer *opus_repacketizer_create(void);

OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);

OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, int len);

OPUS_EXPORT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, int maxlen);

OPUS_EXPORT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp);

OPUS_EXPORT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, int maxlen);

/**@}*/

#ifdef __cplusplus
}
#endif

#endif /* OPUS_H */