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Opus — the Codec To End All Codecs
New submitter jmv writes "It's official. The Opus audio codec is now standardized by the IETF as RFC 6716. Opus is the first state-of-the-art, fully Free and Open audio codec ratified by a major standards organization. Better, Opus covers basically the entire audio-coding application space and manages to be as good or better than existing proprietary codecs over this whole space. Opus is the result of a collaboration between Xiph.Org, Mozilla, Microsoft (yes!), Broadcom, Octasic, and Google. See the Mozilla announcement and the Xiph.Org press release for more details."
It's true that Opus does better than AAC and Vorbis at CD-quality bitrates and thus would be an improvement for music players etc.
But the improvements there are fairly small- in fact, Opus wasn't originally targeted at that kind of use at all, and the authors were quite surprised that it outdid those kinds of high-latency codecs. Opus is a very low-latency codec, and it combines Skype's speech compression technology and more music-oriented technologies (those introduced in CELT) to allow interactive speech and music over the Web.
Gaining marketshare in the high-bitrate stored music market against dominant formats like MP3 and AAC is hard, even when you outperform them substantially. But there's not really any established players in low-latency Internet audio. Opus blows all the other low-latency and/or low-bitrate codecs out of the water when competing in those other codecs' bitrate-latency "sweet spots", is the only codec which can compete across that kind of a range, is now a standard, is royalty-free, and is already implemented in Firefox.
Those who are saying "meh, only audiophiles will care" or "this won't get marketshare against AAC" are missing the point. This codec will change the face of the Web.
It can support up to 255 channels. The two-channel maximum is per stream. Multiple streams can be packed into single frames, but for >2 channels the mapping and coupling has to be signaled at the container level.
The standard tools available at opus-codec.org use Ogg as a container for "at-rest files," and Firefox, foobar2000, and gstreamer-supporting apps (like Opera on Linux) all play Opus-in-Ogg files. VLC and Rockbox will soon release versions with playback support for these too. Though RTP etc is a primary focus, the "at-rest file" support is ahead of the interactive support at this stage of the game.
A Matroska mapping is still in progress. Most likely, for the time being, Opus files will be predominantly Ogg, while the Matroska mapping will be more important for using Opus with video streams (esp. vp8, improving on the webm vp8+vorbis+matroska combination).
See the "listening tests" part of our comparison page. These are all tests that were performed by other folks, independently from us.
Opus: the Swiss army knife of audio codec
To me the biggest difference is that Vorbis was competing head on with a strongly entrenched codec (MP3) and it's official successor (AAC). Opus on the other-hand fills niche in the audio encoding world that doesn't have an established winner; that is high-quality low-latency codecs. This area has largely been driven by cellphone market, and has focused on encoding voice signals at toll-quality, that is as good as an analog long-distance signal (8kHz mono). There really hasn't been much focus on creating a low-latency codec that can encode full-band (music signals), and Opus does that incredibly well. It also sounds much better encoding speech at the bitrates that are used for VoIP (rather than the lower ones used by cellphones).
The internet community has never really been happy with the performance of ITU specified codecs that have been primarily used for SIP and other VoIP applications in the past, and there is no good reason from them not to support Opus. The patent grants are there, the vender support is there, and there is no real competitor codec worth mentioning. I'm convinced this will make much deeper inroads than Vorbis did.
there is test code, but I didn't see sample code explicitly.
Well, we have code snippets for the encoder and the decoder. Otherwise, there's always the code for opus_demo.c and opusenc.c/opusdec.c. Let me know what you think is missing and we can try improving that.
Opus: the Swiss army knife of audio codec
When applied to a codec, "latency" (obviously) refers to stream latency, not network latency (the latter has nothing to do with a codec, obviously). The problem with codecs like MP3 for streaming purposes is that they encode fairly large "frames" of audio, and these frames must be recorded before they can be encoded, encoded before they can be transmitted, received before they can be decoded, and quite possibly also decoded (fully) before they can be played. It may be possible to begin playing before the decoding is complete, which would help a lot, but it also might not - it depends on the codec.
Suppose you've got a "high latency" codec (such as MP3) that uses a 250ms frame and requires full decoding (this is an example; I don't know the actual numbers for MP3). Then suppose you have a low-latency codec (like Opus) with a 15ms frame size. In both cases, your network latency is going to be the same (let's say 100ms). You want to stream audio over this connection. It's pretty high bandwidth and you've got a decent audio processor, so any codec can be encoded, transmitted, and decoded in 10ms or less.
At t=0, begin recording an audio (such as voice) segment.
Codec1 (high-latency):
At t=250ms, you've filled the frame and can begin compression.
At t=360ms, the frame has been encoded, transmitted, received, and decoded.
Total latency before playback can begin: almost 3/8 of a second after recording began.
Codec2 (low-latency):
At t=15ms, you've filled an audio frame and can begin compression.
At t=125ms, the frame has been recieved and decoded by the other end, and playback can begin.
Total latency: 1/8 of a second over the same network connection.
1/8 of a second can be perceived, but barely, and almost all of that is simply an inherent cost of the network transmission. 360ms is not only easy to perceive, it's quite enough to be annoying (think intercontinental call via satellite). There's tons of demand for low-latency codecs.
There's no place I could be, since I've found Serenity...
what's with the FUD, dude? pre-release Opus was included in the hydrogenaudio listening tests months ago alongside all the relevant (not shit) implementations of HE-AAC. just because it doesn't say "plus" doesn't mean it doesn't include SBR (or PS or any of the other bells and whistles).
SBR is good for perceptual "niceness", but as far as _sounding like the original_ goes, it's often quite harmful. all these things are accounted for with the hydrogenaudio listening tests - they're an extremely anal community and wouldn't dare let prejudice through without a long and protracted flamewar. they're actually very trustworthy.
aacplus was just the early CT-proprietary version of HE-AAC. They did test against the two best publicly available HE-AAC encoders, which have improved quite a bit since the aacplus days. Opus was better, by a statistically significant margin.
Opus has band folding, which is in some ways similar to SBR but considerably superior. Halfway down Monty's two-year-old CELT demo page there's some explanation and a visual of what this looks like on a spectrogram in low-bitrate situations. (Opus used technology from CELT but is considerably improved.)
If you really think HE-AAC type codecs sound like CD at 32kbps and so forth you are extremely insensitive to coding artifacts. Unless you meant mono for all of those.