As an open format standardised through Request for Comments RFC , c a high quality reference implementation is provided under the 3-clause BSD license a which compiles and runs on the vast majority of general purpose and embedded fixed point processors. Many Software patents which cover Opus are licensed under royalty-free terms. Also unlike Vorbis, Opus does not require the definition of large codebooks for each individual file, making it also preferable for short clips of audio, such as those often used by game developers, a field where patent-free Vorbis is commonly used.
Considerably more details of the history and potential applications for Opus are included in the Wikipedia page for Opus audio format. Opus supports bitrates from 6 kbps to kbps for typical stereo audio sources and a maximum of around kbps per channel for multichannel audio , with the 'sweet spot' for music and general audio around 30 kbps mono and 40— kbps stereo. It is intrinsically variable bitrate , though constrained VBR and constant bitrate modes are possible where required.
In the case of the reference release, libopus, the target bitrate is calibrated against the internal constant quality targets so that over a typical music collection, something very close to the target bitrate will be achieved.
This bitrate-calibrated approach differs from most VBR encoders e. LAME, helix mp3, qaac, Nero aacenc, Ogg Vorbis, Musepack where a setting on some 'constant quality' scale which differs between encoders is used and the bitrate will fall where it may. Improved future versions can be expected to offer improved quality at the same setting. Independent implementations may adopt a different approach. Opus is able to seamlessly adapt its mode of operation without glitches or sound interruption an illustrative demonstration of bitrate scalability is on the Opus Examples page , which can be particularly useful for mixed-content audio or varying network conditions, making the unified Opus codec superior to a suite of different codecs that might otherwise cover the same range of bitrate and quality settings and would require out-of-band signalling to instigate codec switching.
The switching includes the choice of mono, stereo and other channel mappings, the use of the speech-oriented SILK layer, the general-purpose CELT layer or the hybrid of both, and the use of different audio bandwidths 4, 6, 8, 12, or 20 kHz as well as the quality adjustments within the same operating mode that are available in most VBR-capable codecs.
Of importance mainly to interactive uses, but potentially useful in time-delayed audio streaming also, Opus includes packet loss concealment PLC in all modes and, in the speech-oriented modes where the SILK layer is active it also supports Forward Error Correction FEC where the expected rate of packet loss can be indicated to the encoder by the user or by application software and critical frames e.
This technique likewise increases coding efficiency at bitrates targetting transparent music reproduction. Short blocks 2. Short blocks can also be used exclusively, if very low algorithmic delay 5. CELT uses a number of additional techniques and provides additional advanced tools to enable encoder tuning. Opus natively supports gapless playback though poor player design might itself induce interruptions during playback. Playback gain is also required, making some form of ReplayGain or similar volume control possible in any compliant player.
For mono speech, Opus ranges from intelligible narrowband speech reproduction starting at 6 kbps to medium-band, wideband and superwideband speech, reaching full-band speech by around 14 kbps in encoder version 1. The hybrid mode is adopted as bitrate increases, extending bandwidth first to 12 kHz comparable with compact cassette then to the full 20 kHz and CELT then takes over.
Assuming the source is stereo, the transition from mono to stereo typically happens between the transition from 12 kHz to 20 kHz. Encoder version 1. Version 1. The tables below give illustrative, indicative quality guidance based on typical modes used internally by Opus and a range of listening tests.
In encoder version 1. These tables are likely to require updates as the encoder is improved, especially in low-bitrate regions. The default 20ms frame size Note that the selection of VOIP mode will deliberately modify the sound with a High Pass Filter and emphasis of formants and harmonics to improve intelligibility of speech especially in noisy environments much as telephones do.
Auto mode will not modify the sound prior to encoding so is usually better for high quality speech recordings or mixed speech and music. This table assumes a stereophonic source sampled at CD quality or above typ 48 kHz sampling rate. Opus will automatically use mono at very low bitrates, though a certain amount of stereo encoding can still be used content dependent even when mono is specified as the typical stereo mode in the table below.
For interactive use on the Internet or other packet-based networks, total bandwidth used will be subject to packet overhead. The more packet headers that are transmitted every second, the greater will be the overhead that is required.
For this reason, Opus, while defaulting to 20 ms frames, supports 60 ms frames to reduce overhead when transporting low-bitrate SILK frames at the expense of greater latency, which may still be acceptable for speech, and also supports 10 ms SILK frames to reduce latency somewhat at the expense of packet overhead. Loudgain usage screenshot. Opus logo1. Opus logo2. Opus quality comparison colorblind compatible fr.
Opus quality comparison colorblind compatible hu. Opus quality comparison colorblind compatible. Opus quality comparison.
Test mp3 opus 16kbps. YouTube AV1 video with Opus audio stat screenshot. YouTube H video with Opus audio stat screenshot. YouTube VP9 video with Opus audio stat screenshot. Categories : Audio coding formats Audio data compression Xiph. Namespaces Category Discussion. Views View Edit History.
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