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Opus is a lossy audio coding format developed by the Internet Engineering Task Force (IETF) that is particularly suitable for interactive real-time applications over the Internet. Opus incorporates technology from two other audio coding formats: the speech-oriented SILK and the low-latency CELT.〔 Opus can be adjusted seamlessly between high and low bitrates, and internally, it transitions between linear predictive coding at lower bitrates and transform coding at higher bitrates (as well as a hybrid for a short overlap). Opus has a very low algorithmic delay (26.5 ms by default), which is a necessity for use as part of a low audio latency communication link, which can permit natural conversation, networked music performances, or lip sync at live events. Opus permits trading-off quality or bitrate to achieve an even smaller algorithmic delay, down to 5 ms. Its delay is very low compared to well over 100 ms for popular music formats such as MP3, Ogg Vorbis and HE-AAC; yet Opus performs very competitively with these formats in terms of quality per bitrate.〔Raymond Chen et al. (Opus Testing ). IETF 80〕 Unlike Ogg Vorbis, Opus does not require the definition of large codebooks for each individual file, making it preferable to Vorbis for short clips of audio. As an open format standardized through RFC 6716, a reference implementation audio codec called opus-tools is available under the New BSD License. All known software patents which cover Opus are licensed under royalty-free terms. ==Features== Opus supports constant and variable bitrate encoding from 6 kbit/s to 510 kbit/s, frame sizes from 2.5 ms to 60 ms, and certain sampling rates from 8 kHz (with 4 kHz bandwidth) to 48 kHz (with 20 kHz bandwidth, where the entire hearing range of the human auditory system can be reproduced). An Opus stream can support up to 255 audio channels, and it allows channel coupling between channels in groups of two using mid-side coding. The inherently low delay in Opus (26.5 ms by default) makes it possible to be used in the same real-time applications required by telephony, Voice over IP and videoconferencing. For high quality audio, the ability to maintain low latency at higher bitrates is made possible by research done by the Xiph.Org Foundation on the CELT codec. During any Opus stream (live or in a file), the bitrate, bandwidth, and delay can be changed quickly and seamlessly without introducing any distortion or discontinuity in the audio. As an open standard, the algorithms are openly documented, and a reference implementation (including the source code) is published. Broadcom and the Xiph.Org Foundation own software patents on some of the CELT algorithms, and Skype Technologies S.A./Microsoft own some on the SILK algorithms; but each has pledged to make them available royalty-free for use with Opus once the format was accepted as an IETF standard. They also reserve the right to make use of their patents to defend against infringement suits of third parties. The applicability of non royalty-free patent claims from Qualcomm and Huawei to Opus is disputed.〔(【引用サイトリンク】title=It's Opus, it rocks and now it's an audio codec standard! )〕 The Opus format is based on the low-latency CELT format and the speech-oriented SILK format (both of which have been heavily modified, rendering them incompatible with their original formats). The transform layer (CELT) is based on the modified discrete cosine transform (MDCT) with approaches from CELP (codebooks for excitation, although in the frequency domain). CELT was modified and among other things, support for 20 ms frames was added. The SILK layer that specializes in speech signals is based on linear predictive coding (LPC) and an optional Long-Term Prediction filter. SILK was modified and among other things, support for 10 ms frames was added. To minimize packet overhead at low bitrates, SILK has support for larger frames of 60 ms (versus CELT's 20 ms). The shared range encoding of both parts of a hybrid stream was taken from CELT. The format has three different modes, two being for pure speech signals, and a third for general audio (including music and speech). One of the speech modes is capable of reproducing the full spectrum of the human hearing range. In this mode, CELT is used for the upper part of the frequency range (from 8 kHz upwards), and SILK is used for the lower part. For low bitrates (below about 30 kbit/s), the upper frequencies can be cut off and the CELT layer left out. For audio at higher bitrates, the SILK layer that specializes in speech signals is left out, and the non-specialized CELT layer is used. The reference implementation is written in C and compiles on hardware architectures with or without a floating-point unit. 抄文引用元・出典: フリー百科事典『 ウィキペディア（Wikipedia）』 ■ウィキペディアで「Opus (audio format)」の詳細全文を読む スポンサード リンク
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