Opus (audio format)
|Internet media type||
|Developed by||IETF codec working group|
|Initial release||September 11, 2012|
|Type of format||Audio|
|Contained by||Ogg, Matroska|
|Extended from||SILK, CELT|
|Initial release||August 26, 2012|
|Stable release||1.1 / December 5, 2013|
|Preview release||1.1-rc / November 26, 2013|
|Type||Audio codec, reference implementation|
|License||3-clause BSD license|
|Website||Opus codec downloads|
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. As an open format standardized through RFC 6716, a reference implementation audio codec called opus-tools is provided under the 3-clause BSD license. All known software patents which cover Opus are licensed under royalty-free terms.
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 (22.5 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. 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.
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 (22.5 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.
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, SILK has support for a longer delay 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.
Opus data can be encapsulated in Ogg containers. The content of such Ogg Opus streams should be specified as
audio/ogg; codecs=opus and for Ogg Opus files the
.opus filename extension is recommended. Support for encapsulation of Opus in Matroska containers has been completed. Opus is also allowed as a registered format in MPEG-TS.
|Abbreviation||Audio bandwidth||Effective sample rate|
|NB (narrowband)||4 kHz||8 kHz|
|MB (medium-band)||6 kHz||12 kHz|
|WB (wideband)||8 kHz||16 kHz|
|SWB (super-wideband)||12 kHz||24 kHz|
|FB (fullband)||20 kHz||48 kHz|
Opus was proposed for the standardization of a new audio format at the IETF, which was eventually accepted and granted by the codec working group. It is based on two initially separate standard proposals from the Xiph.Org Foundation and Skype Technologies S.A. (now Microsoft). Its main developers are Jean-Marc Valin (Xiph.Org, Octasic, Mozilla Corporation), Koen Vos (Skype), and Timothy B. Terriberry (Xiph.Org, Mozilla Corporation). Among others, Juin-Hwey (Raymond) Chen (Broadcom), Gregory Maxwell (Xiph.Org, Wikimedia), and Christopher Montgomery (Xiph.Org) were also involved.
The development of the CELT part of the format goes back to thoughts on a successor for Vorbis under the working name Ghost. As a newer speech codec from the Xiph.Org Foundation, Opus replaces Xiph's older speech codec Speex, an earlier project of Jean-Marc Valin. CELT has been worked on since November 2007.
The SILK part has been under development at Skype since January 2007 as the successor of their SVOPC, an internal project to make the company independent from third-party codecs like iSAC and iLBC and respective license payments.
In March 2009, Skype suggested the development and standardization of a wideband audio format within the IETF. Nearly a year passed with much debate on the formation of an appropriate working group. Representatives of several companies which were taking part in the standardization of patent-encumbered competing formats stated objections against the start of the standardization process for a royalty-free format: representatives of Polycom and Ericsson—the creators and licensors of G.719—as well as France Télécom, Huawei and the Orange Labs (department of France Télécom), which were involved in the creation of G.718. The working group finally formed in February 2010, and even the corresponding Study Group 16 from the ITU-T pledged to support its work.
In July 2010, a prototype of a hybrid format was presented that combined the two proposed format candidates SILK and CELT. In September 2010, Opus was submitted to the IETF as proposal for standardization. For a short time the format went under the name of Harmony before it got its present name in October 2010. At the beginning of February 2011, the bitstream format was tentatively frozen, subject to last changes. Near the end of July 2011, Jean-Marc Valin was hired by the Mozilla Corporation to continue working on Opus. In November 2011, the working group issued the last call for changes on the bitstream format. The bitstream has been frozen since January 8, 2012. On July 2, 2012, Opus was approved by the IETF for standardization. The reference software entered release candidate state on August 8. The final specification was released as RFC 6716 on September 10, 2012. and versions 1.0 and 1.0.1 of the reference implementation were released the day after.
On July 11, 2013, the first beta of version 1.1 was released incorporating significant encoder side improvements for quality and efficiency, and significant speed improvements for both encoding and decoding detailed in the release information. Surround sound encoding has better channel bitrate allocation, improved Low-Frequency Effects quality and efficiency, and use of cross-channel masking. Tonality estimation provides a bitrate boost to highly tonal sounds like harpsichord, significantly improving quality in Opus's weakest samples from previous public 64kbps tests in which it had otherwise performed well. Automated speech/music detection is enabled to improve quality. Other improvements include stereo bitrate saving when channels are closely correlated, dynamic allocation to improve quality for transients and high-energy bands adjacent to low-energy bands, DC rejection below 3 Hz to improve bass analysis. True unconstrained VBR improves quality consistency for same target bitrate, while Temporal VBR is a serendipitous discovery that improves quality on a number of samples.
Also on July 11, 2013, stable version 1.0.3 was introduced implementing a backport of the new 1.1 surround API mentioned in beta 1.1 above as well as some minor bug fixes.
Opus 1.1 was officially released on December 5, 2013.
Quality comparison and low latency performance
In listening tests at around 64 kbit/s Opus shows superior quality compared to HE-AAC codecs, which have so far been dominant in that area because of their use of the proprietary spectral band replication (SBR) technology.
Opus has very low algorithmic delay, 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. Total algorithmic delay for an audio format is the sum of delays that must be incurred in the encoder and the decoder of a live audio stream regardless of processing speed and transmission speed, such as buffering audio samples into blocks or frames, allowing for window overlap and possibly allowing for noise-shaping look-ahead in a decoder and any other forms of look-ahead, or for an MP3 encoder, the use of bit reservoir.
Total one-way latency below 150 ms is the preferred target of most VoIP systems, to enable natural conversation with turn-taking little affected by delay. Musicians typically feel in-time with up to around 30 ms audio latency, roughly in accord with the fusion time of the Haas effect, though matching playback delay of each user's own instrument to the round-trip latency can also help. It is suggested for lip sync that around 45-100 ms audio latency may be acceptable.
Opus permits trading-off quality or bitrate to achieve an even smaller algorithmic delay (5.0 ms minimum). While the default Opus frame is 20 ms long, a further 2.5 ms lookahead is required for window overlap in the CELT layer and noise shaping in the SILK layer, typically giving a delay of 22.5 ms.[disputed ] The SILK layer's minimum is 10 ms frames (12.5 ms delay), while the CELT layer's minimum is 2.5 ms (5.0 ms delay).
The format and algorithms are openly documented and the reference implementation is published as free software. Xiph's reference implementation is called opus-tools, and it consists of a separate encoder and decoder; it is published under the terms of a BSD-like license. It is written in C and can be compiled for hardware architectures with or without a floating point unit. The accompanying diagnostic tool opusinfo reports detailed technical information about Opus files, including information on the standard compliance of the bitstream format. It is based on ogginfo from the vorbis-tools and therefore — unlike the encoder and decoder — is available under the terms of version 2 of the GPL.
Most end-user software relies on multimedia frameworks provided by the operating system. Native Opus codec support is implemented in most major multimedia frameworks for Unix-like operating systems, including GStreamer, FFmpeg and Libav libraries. Although Microsoft Windows DirectShow framework lacks an Opus encoder and decoder, such capabilities may be acquired by installing third-party codecs like LAV Filters or Xiph.org Foundation's OpenCodecs.
While support in multimedia frameworks automatically enables Opus support in software which is built on top of such frameworks, several applications developers made additional efforts for supporting the Opus audio format in their software. Such support was added to AIMP, Amarok, cmus, foobar2000, Mpxplay, MusicBee, SMplayer, VLC media player, Winamp and Xmplay audio players; Icecast, Liquidsoap and Airtime audio streaming software; and Asunder audio CD ripper, CDBurnerXP CD burner, FFmpeg, Libav and MediaCoder media encoding tools. Streaming Icecast radio trials are live since September 2012 and January 2013. SteamOS uses Opus or Vorbis for streaming audio.
Opus support is mandatory for WebRTC implementations. Opus is supported in Mozilla Firefox, Chromium and Google Chrome, Blink-based Opera, as well as all browsers for Unix-like systems relying on GStreamer for multimedia formats support. Although Internet Explorer doesn't provide Opus playback natively, support for the format may be added by installing OpenCodecs by Xiph.Org Foundation.
Due to its abilities, Opus gained early interest from VoIP software vendors. Several SIP clients, including Acrobits Softphone, Bria X-Lite, CiceroSupra, CSipSimple (via additional plug-in), Empathy (via GStreamer), Jitsi, Line2 (currently only on iOS), Phoner and PhonerLite, SFLphone, Mumble and TeamSpeak 3 voice chat software also support Opus. TrueConf supports Opus in its VoIP products. As Asterisk lacks builtin Opus support for legal reasons, third-party patch is available for download.
Since version 3.13 Rockbox enables Opus playback on supported portable media players, including products from the iPod series by Apple, devices made by iriver, Archos and Sandisk, and on Android devices using "Rockbox as an Application" (RaaA). Chromecast supports Opus decoding.
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