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524 строки
21 KiB
C
524 строки
21 KiB
C
/* Copyright (c) 2010-2012 IETF Trust, Xiph.Org Foundation, Skype Limited. All rights reserved.
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Written by Jean-Marc Valin and Koen Vos */
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/*
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This file is extracted from RFC6716. Please see that RFC for additional
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information.
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Redistribution and use in source and binary forms, with or without
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modification, are permitted provided that the following conditions
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are met:
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- Redistributions of source code must retain the above copyright
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notice, this list of conditions and the following disclaimer.
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- Redistributions in binary form must reproduce the above copyright
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notice, this list of conditions and the following disclaimer in the
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documentation and/or other materials provided with the distribution.
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- Neither the name of Internet Society, IETF or IETF Trust, nor the
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names of specific contributors, may be used to endorse or promote
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products derived from this software without specific prior written
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permission.
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THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
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``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
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LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
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A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
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OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
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EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
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PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
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LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
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NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
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SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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/**
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* @file opus.h
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* @brief Opus reference implementation API
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*/
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#ifndef OPUS_H
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#define OPUS_H
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#include "opus_types.h"
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#include "opus_defines.h"
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#ifdef __cplusplus
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extern "C" {
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#endif
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/**
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* @mainpage Opus
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*
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* The Opus codec is designed for interactive speech and audio transmission over the Internet.
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* It is designed by the IETF Codec Working Group and incorporates technology from
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* Skype's SILK codec and Xiph.Org's CELT codec.
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*
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* The Opus codec is designed to handle a wide range of interactive audio applications,
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* including Voice over IP, videoconferencing, in-game chat, and even remote live music
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* performances. It can scale from low bit-rate narrowband speech to very high quality
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* stereo music. Its main features are:
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* @li Sampling rates from 8 to 48 kHz
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* @li Bit-rates from 6 kb/s 510 kb/s
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* @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
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* @li Audio bandwidth from narrowband to full-band
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* @li Support for speech and music
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* @li Support for mono and stereo
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* @li Frame sizes from 2.5 ms to 60 ms
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* @li Good loss robustness and packet loss concealment (PLC)
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* @li Floating point and fixed-point implementation
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*
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* Documentation sections:
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* @li @ref opusencoder
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* @li @ref opusdecoder
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* @li @ref repacketizer
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* @li @ref libinfo
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*/
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/** @defgroup opusencoder Opus Encoder
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* @{
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*
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* Since Opus is a stateful codec, the encoding process starts with creating an encoder
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* state. This can be done with:
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*
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* @code
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* int error;
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* OpusEncoder *enc;
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* enc = opus_encoder_create(Fs, channels, application, &error);
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* @endcode
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*
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* From this point, @c enc can be used for encoding an audio stream. An encoder state
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* @b must @b not be used for more than one stream at the same time. Similarly, the encoder
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* state @b must @b not be re-initialized for each frame.
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*
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* While opus_encoder_create() allocates memory for the state, it's also possible
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* to initialize pre-allocated memory:
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*
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* @code
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* int size;
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* int error;
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* OpusEncoder *enc;
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* size = opus_encoder_get_size(channels);
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* enc = malloc(size);
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* error = opus_encoder_init(enc, Fs, channels, application);
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* @endcode
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*
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* where opus_encoder_get_size() returns the required size for the encoder state. Note that
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* future versions of this code may change the size, so no assuptions should be made about it.
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*
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* The encoder state is always continuous in memory and only a shallow copy is sufficient
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* to copy it (e.g. memcpy())
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*
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* It is possible to change some of the encoder's settings using the opus_encoder_ctl()
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* interface. All these settings already default to the recommended value, so they should
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* only be changed when necessary. The most common settings one may want to change are:
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*
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* @code
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* opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
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* opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
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* opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
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* @endcode
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*
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* where
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*
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* @arg bitrate is in bits per second (b/s)
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* @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
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* @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
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*
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* 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.
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*
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* 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:
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* @code
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* len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
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* @endcode
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*
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* where
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* <ul>
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* <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
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* <li>frame_size is the duration of the frame in samples (per channel)</li>
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* <li>packet is the byte array to which the compressed data is written</li>
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* <li>max_packet is the maximum number of bytes that can be written in the packet (1276 bytes is recommended)</li>
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* </ul>
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*
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* opus_encode() and opus_encode_frame() return the number of bytes actually written to the packet.
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* The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
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* is 1 byte, then the packet does not need to be transmitted (DTX).
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*
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* Once the encoder state if no longer needed, it can be destroyed with
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*
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* @code
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* opus_encoder_destroy(enc);
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* @endcode
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*
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* If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
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* then no action is required aside from potentially freeing the memory that was manually
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* allocated for it (calling free(enc) for the example above)
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*
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*/
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/** Opus encoder state.
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* This contains the complete state of an Opus encoder.
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* It is position independent and can be freely copied.
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* @see opus_encoder_create,opus_encoder_init
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*/
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typedef struct OpusEncoder OpusEncoder;
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OPUS_EXPORT int opus_encoder_get_size(int channels);
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/**
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*/
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/** Allocates and initializes an encoder state.
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* There are three coding modes:
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*
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* @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
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* signals. It enhances the input signal by high-pass filtering and
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* emphasizing formants and harmonics. Optionally it includes in-band
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* forward error correction to protect against packet loss. Use this
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* mode for typical VoIP applications. Because of the enhancement,
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* even at high bitrates the output may sound different from the input.
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*
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* @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
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* non-voice signals like music. Use this mode for music and mixed
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* (music/voice) content, broadcast, and applications requiring less
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* than 15 ms of coding delay.
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*
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* @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
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* disables the speech-optimized mode in exchange for slightly reduced delay.
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*
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* This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
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* @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
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* @param [in] channels <tt>int</tt>: Number of channels (1/2) in input signal
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* @param [in] application <tt>int</tt>: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
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* @param [out] error <tt>int*</tt>: @ref errorcodes
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* @note Regardless of the sampling rate and number channels selected, the Opus encoder
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* can switch to a lower audio audio bandwidth or number of channels if the bitrate
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* selected is too low. This also means that it is safe to always use 48 kHz stereo input
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* and let the encoder optimize the encoding.
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*/
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OPUS_EXPORT OpusEncoder *opus_encoder_create(
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opus_int32 Fs,
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int channels,
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int application,
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int *error
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);
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/** Initializes a previously allocated encoder state
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* The memory pointed to by st must be the size returned by opus_encoder_get_size.
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* This is intended for applications which use their own allocator instead of malloc.
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* @see opus_encoder_create(),opus_encoder_get_size()
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* To reset a previously initialized state use the OPUS_RESET_STATE CTL.
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* @param [in] st <tt>OpusEncoder*</tt>: Encoder state
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* @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
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* @param [in] channels <tt>int</tt>: Number of channels (1/2) in input signal
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* @param [in] application <tt>int</tt>: Coding mode (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY)
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* @retval OPUS_OK Success or @ref errorcodes
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*/
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OPUS_EXPORT int opus_encoder_init(
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OpusEncoder *st,
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opus_int32 Fs,
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int channels,
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int application
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);
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/** Encodes an Opus frame.
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* The passed frame_size must an opus frame size for the encoder's sampling rate.
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* For example, at 48kHz the permitted values are 120, 240, 480, or 960.
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* Passing in a duration of less than 10ms (480 samples at 48kHz) will
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* prevent the encoder from using the LPC or hybrid modes.
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* @param [in] st <tt>OpusEncoder*</tt>: Encoder state
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* @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
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* @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
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* @param [out] data <tt>char*</tt>: Output payload (at least max_data_bytes long)
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* @param [in] max_data_bytes <tt>int</tt>: Allocated memory for payload; don't use for controlling bitrate
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* @returns length of the data payload (in bytes) or @ref errorcodes
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*/
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OPUS_EXPORT int opus_encode(
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OpusEncoder *st,
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const opus_int16 *pcm,
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int frame_size,
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unsigned char *data,
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int max_data_bytes
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);
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/** Encodes an Opus frame from floating point input.
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* The passed frame_size must an opus frame size for the encoder's sampling rate.
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* For example, at 48kHz the permitted values are 120, 240, 480, or 960.
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* Passing in a duration of less than 10ms (480 samples at 48kHz) will
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* prevent the encoder from using the LPC or hybrid modes.
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* @param [in] st <tt>OpusEncoder*</tt>: Encoder state
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* @param [in] pcm <tt>float*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(float)
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* @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
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* @param [out] data <tt>char*</tt>: Output payload (at least max_data_bytes long)
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* @param [in] max_data_bytes <tt>int</tt>: Allocated memory for payload; don't use for controlling bitrate
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* @returns length of the data payload (in bytes) or @ref errorcodes
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*/
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OPUS_EXPORT int opus_encode_float(
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OpusEncoder *st,
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const float *pcm,
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int frame_size,
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unsigned char *data,
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int max_data_bytes
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);
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/** Frees an OpusEncoder allocated by opus_encoder_create.
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* @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
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*/
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OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
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/** Perform a CTL function on an Opus encoder.
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*
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* Generally the request and subsequent arguments are generated
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* by a convenience macro.
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* @see encoderctls
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*/
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OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...);
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/**@}*/
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/** @defgroup opusdecoder Opus Decoder
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* @{
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*
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*
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* The decoding process also starts with creating a decoder
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* state. This can be done with:
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* @code
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* int error;
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* OpusDecoder *dec;
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* dec = opus_decoder_create(Fs, channels, &error);
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* @endcode
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* where
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* @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
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* @li channels is the number of channels (1 or 2)
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* @li error will hold the error code in case or failure (or OPUS_OK on success)
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* @li the return value is a newly created decoder state to be used for decoding
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*
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* While opus_decoder_create() allocates memory for the state, it's also possible
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* to initialize pre-allocated memory:
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* @code
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* int size;
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* int error;
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* OpusDecoder *dec;
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* size = opus_decoder_get_size(channels);
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* dec = malloc(size);
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* error = opus_decoder_init(dec, Fs, channels);
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* @endcode
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* where opus_decoder_get_size() returns the required size for the decoder state. Note that
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* future versions of this code may change the size, so no assuptions should be made about it.
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*
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* The decoder state is always continuous in memory and only a shallow copy is sufficient
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* to copy it (e.g. memcpy())
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*
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* To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
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* @code
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* frame_size = opus_decode(enc, packet, len, decoded, max_size);
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* @endcode
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* where
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*
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* @li packet is the byte array containing the compressed data
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* @li len is the exact number of bytes contained in the packet
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* @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
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* @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
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*
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* opus_decode() and opus_decode_frame() return the number of samples ()per channel) decoded from the packet.
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* If that value is negative, then an error has occured. This can occur if the packet is corrupted or if the audio
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* buffer is too small to hold the decoded audio.
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*/
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/** Opus decoder state.
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* This contains the complete state of an Opus decoder.
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* It is position independent and can be freely copied.
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* @see opus_decoder_create,opus_decoder_init
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*/
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typedef struct OpusDecoder OpusDecoder;
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/** Gets the size of an OpusDecoder structure.
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* @param [in] channels <tt>int</tt>: Number of channels
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* @returns size
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*/
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OPUS_EXPORT int opus_decoder_get_size(int channels);
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/** Allocates and initializes a decoder state.
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* @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
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* @param [in] channels <tt>int</tt>: Number of channels (1/2) in input signal
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* @param [out] error <tt>int*</tt>: OPUS_OK Success or @ref errorcodes
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*/
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OPUS_EXPORT OpusDecoder *opus_decoder_create(
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opus_int32 Fs,
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int channels,
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int *error
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);
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/** Initializes a previously allocated decoder state.
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* The state must be the size returned by opus_decoder_get_size.
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* This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
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* To reset a previously initialized state use the OPUS_RESET_STATE CTL.
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* @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
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* @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
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* @param [in] channels <tt>int</tt>: Number of channels (1/2) in input signal
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* @retval OPUS_OK Success or @ref errorcodes
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*/
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OPUS_EXPORT int opus_decoder_init(
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OpusDecoder *st,
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opus_int32 Fs,
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int channels
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);
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/** Decode an Opus frame
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* @param [in] st <tt>OpusDecoder*</tt>: Decoder state
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* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
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* @param [in] len <tt>int</tt>: Number of bytes in payload*
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* @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
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* is frame_size*channels*sizeof(opus_int16)
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* @param [in] frame_size Number of samples per channel of available space in *pcm,
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* if less than the maximum frame size (120ms) some frames can not be decoded
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* @param [in] decode_fec <tt>int</tt>: Flag (0/1) to request that any in-band forward error correction data be
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* decoded. If no such data is available the frame is decoded as if it were lost.
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* @returns Number of decoded samples or @ref errorcodes
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*/
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OPUS_EXPORT int opus_decode(
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OpusDecoder *st,
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const unsigned char *data,
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int len,
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opus_int16 *pcm,
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int frame_size,
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int decode_fec
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);
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/** Decode an opus frame with floating point output
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* @param [in] st <tt>OpusDecoder*</tt>: Decoder state
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* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
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* @param [in] len <tt>int</tt>: Number of bytes in payload
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* @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
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* is frame_size*channels*sizeof(float)
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* @param [in] frame_size Number of samples per channel of available space in *pcm,
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* if less than the maximum frame size (120ms) some frames can not be decoded
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* @param [in] decode_fec <tt>int</tt>: Flag (0/1) to request that any in-band forward error correction data be
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* decoded. If no such data is available the frame is decoded as if it were lost.
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* @returns Number of decoded samples or @ref errorcodes
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*/
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OPUS_EXPORT int opus_decode_float(
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OpusDecoder *st,
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const unsigned char *data,
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int len,
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float *pcm,
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int frame_size,
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int decode_fec
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);
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/** Perform a CTL function on an Opus decoder.
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*
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* Generally the request and subsequent arguments are generated
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* by a convenience macro.
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* @see genericctls
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*/
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OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...);
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/** Frees an OpusDecoder allocated by opus_decoder_create.
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* @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
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*/
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OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
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/** Parse an opus packet into one or more frames.
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* Opus_decode will perform this operation internally so most applications do
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* not need to use this function.
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* This function does not copy the frames, the returned pointers are pointers into
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* the input packet.
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* @param [in] data <tt>char*</tt>: Opus packet to be parsed
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* @param [in] len <tt>int</tt>: size of data
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* @param [out] out_toc <tt>char*</tt>: TOC pointer
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* @param [out] frames <tt>char*[48]</tt> encapsulated frames
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|
* @param [out] size <tt>short[48]</tt> sizes of the encapsulated frames
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|
* @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
|
|
* @returns number of frames
|
|
*/
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|
OPUS_EXPORT int opus_packet_parse(
|
|
const unsigned char *data,
|
|
int len,
|
|
unsigned char *out_toc,
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|
const unsigned char *frames[48],
|
|
short size[48],
|
|
int *payload_offset
|
|
);
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|
|
|
/** 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 frames 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 */
|