ref: 4871fb9281bebddffaaabedae179c9ac7946355f
dir: /src/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,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
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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 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 */