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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."
Obligatory.
What's wrong with Bill the Cat?
A feeling of having made the same mistake before: Deja Foobar
Cue MPEG-LA calling for a patent portfolio to be created and licensed for hard cash, under their gracious auspices, of course.
Ezekiel 23:20
They also left out n-channel support. You get mono or stereo, but that's it. No 7.1 surround encoding. That would have made it the one true codec for everything. That and lossless encoding. The two true codecs for everything.
Oh, and support in the iPhone. That would have made three true codecs. Among the many true codecs... oh bugger, I'll start again.
John
Opus has support for up to 255 channels. Indeed, lossless was the most glaring omission, but considering the obsolescence of MP3HD, I think they must had good reasons to leave it out.
If I clone myself, can I call it a thread?
If a girl winks to us, can I call it a race condition?
About 9 months ago, I implemented Opus in our VoIP products, replacing G722 and Speex. It kicks a whole lot of ass. Compared to speex, It's far better coded, uses far fewer CPU cycles, and sounds vastly better (even to me, and I have shitty hearing). Similarly, we replaced all our old audio DSP pipeline, based on the Speex library (thanks Xiph.org, etc) with the low-level components from WebRTC (thanks Google!) and things have never sounded better.
Today's weirdness is tomorrow's reason why. -- Hunter S. Thompson
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.
Audiophiles? Really? The only format they care about is original wax drums rubbed with a diamond and amplified by analog equipment connected by gold cables soaked in unicorn tears. They want nothing to do with digital audio codecs.
Kudos to the folks working on this. We were all rooting for ogg/vorbis/xiph, but they had some lessons to learn. Positives that I see for Opus:
still could use some love:
Still, an order of magnitude better than the last attempt at gaining industry acceptance of free codecs. This one might just work out!
My God, it's Full of Source!
OUTSIDE_IP=$(dig +short my.ip @outsideip.net)
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).
Despite what the wiki page says, the RFC page states mono or stereo, and indeed the reference source code checks for channels equal to 1 or 2.
if((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000)||(channels!=1&&channels!=2)||
(application != OPUS_APPLICATION_VOIP && application != OPUS_APPLICATION_AUDIO
&& application != OPUS_APPLICATION_RESTRICTED_LOWDELAY))
{
if (error)
*error = OPUS_BAD_ARG;
return NULL;
}
I think the 255 reference was probably a left over from the Vorbis definition. It's too bad that multi-channel isn't naively supported, as multiplexing multiple mono or stereo streams will be a bit of a pain.
Bitrates below 16 kb/s are irrelevant on the Internet. Just the overhead (IP+UDP+RTP headers) of sending packets every 20 ms is 16 kb/s. At that point, you might as well transmit some real quality.
Opus: the Swiss army knife of audio codec
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But... the future refused to change.
the standard does not specify a recommended container format
See the Opus Ogg mapping for more details. Of course, if people want to use other containers, we're not a container police.
Opus: the Swiss army knife of audio codec
A quick look at the graph shows that they stop at 128kbps, which would mean it's a great codec for high-quality real-time audio telephony rather than as a codec to span the spectrum of low end real time to lossless audio.
The reason the graph stops at 128 kb/s is that things become uninteresting at that point -- because nobody's able to actually tell the difference. With VBR, we've never had anyone report audio not being transparent above 200 kb/s. There's a reason people don't want to organize listening tests at 128 kb/s and (especially) above. It's indeed the case that we don't support lossless. That one is already covered very well by FLAC and there was no point adding completely different algorithms to handle that. Otherwise, Opus can replace MP3/AAC/Vorbis at rates above 128 kb/s too.
Opus: the Swiss army knife of audio codec
none other than Bruce Perens (Open Source champion) points us to these: https://datatracker.ietf.org/ipr/1520/ https://datatracker.ietf.org/ipr/1741/ wherein we learn that Opus is "possible royalty/fee". this is not consistent with "Fully Free" to any patent troll waiting for broad adoption before jumping.
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.
One Codec to rule them all
Once Codec to find them
Once Codec to bring them all
And in the RIAA's darkness bind them
General Relativity: Space-time tells matter where to go; Matter tells space-time what shape to be.
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.
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Pffft, you say ordinary diamonds. real audiophiles only listen to BLUE diamonds that were polished on the thighs of virgins.
Do not look at laser with remaining good eye.
Actually generic data compression algorithms don't work very well on audio files. Lossless audio codecs like ALAC and FLAC do waveform analysis and "know more" about the nature of audio files so that they can be compressed more than your typical compression algorithm like gzip would do.
jmv and I both work for Mozilla and are on the Opus team.
I don't think silicon support for audio codecs is really useful anymore. Audio codecs have such a low complexity compared to video that modern smartphones can run them really easily.I haven't measured exactly, but I'd say you can probably decode an Opus stream with about 2% CPU on the latest Android phone. Not worth paying for extra silicon.
Opus: the Swiss army knife of audio codec
I asked Qualcom if they'd consider changing their declaration to royalty-free. I don't know anyone at Huawei.
Bruce Perens.
What Monty mostly regretted were some decisions that caused a large increase in the amount of *RAM* required. We were actually careful about that. Another improvement compared to Vorbis is that we don't have these codebooks that you need to carry in the headers. This means you can join an ongoing Opus stream with no signalling whatsoever.
Opus: the Swiss army knife of audio codec