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Next-Gen Low-Latency Open Codec Beats HE-AAC
Aldenissin writes "From the Xiph.org developers, Opus is a non-patent encumbered codec designed for interactive usages, such as VoIP, telepresence, and remote jamming, that require very low latency. When they started working on Opus (then known as CELT), they used the slogan 'Why can't your telephone sound as good as your stereo?', and they weren't kidding. Now, test results demonstrate that Opus's performance against HE-AAC, one of the strongest (but highest-latency) codecs at this bitrate, bests the quality of two of the most popular and respected encoders for the format, on the majority of individual audio samples receiving a higher average score overall. Hydrogenaudio conducted a 64kbit/sec multiformat listening test including Opus, aoTuV Vorbis, two HE-AAC encoders, and a 48kbit/sec AAC-LC low anchor. Comparing 30 diverse samples using the highly sensitive ABC/HR methodology, Opus is running with 22.5ms of total latency but the codec can go as low as 5ms."
This will be perfect for my next level beats.
Expanding a vast wasteland since 1996.
and remote jamming
Took me a while to figure out they meant in a band. I was wondering how they were going to jam some sort of signal with this codec.
Sent from my PDP-11
As mentioned, it's needed for VoIP systems. With a full-duplex system, more than 150ms of lag is audible and noticeably uncomfortable, breaking the flow of conversation (As the apparent lag is doubled in a "conversation", with the delay at each end adding cumulatively). For simple half-duplex systems like gaming, more lag is not really noticeable.
Yes, 5 to 22.5 ms is the algorithmic delay of the codec. By comparison, codecs like AAC/MP3/Vorbis have more than 100 ms algorithmic delay (you need to give the encoder side more than 100 ms of audio before the decoder side gives you any audio back).
Opus: the Swiss army knife of audio codec
Lol what? You're crazy. I suppose it is never worth inventing a new codec ever, since everyone uses old codecs! /fail argument
To be exact, there *are* patents, but they will be available without fee in a way that is compatible with FOSS licences such as the GPL. The main idea behind these patents is that your license terminates if you sue someone by claiming Opus infringes your patents. Almost like a copyleft, but for patents (of course the details are different because copyright != patent).
Opus: the Swiss army knife of audio codec
While your rant appears informative if not insightful on its face, it is completely missing the point.
This is a test of audio codecs at low bitrates.
I don't know what this "LE-AAC" is you speak of (and rather suspect you don't either) but AAC-LC was actually in this test, as the low anchor.
At these bitrates (~64kbps) HE-AAC (despite its "low-accuracy" as you put it) is perceptually better sounding than AAC-LC. Lossy audio codecs (even the LE-AAC [sic] encoder in Apple's Core Audio framework you love) can only be judged by how they sound, not how they look. "Accuracy" is not a metric very worthy of discussion.
That's the whole idea behind any lossy codec. You're trading mathematical accuracy for psycho-acoustical accuracy; personally, I don't care if the root mean square error is higher, I just need it to sound like the original.
Anyway, if this really IS an improvement over HE-AAC, which uses some very techniques, I'll be extremely impressed, and quite pleased that it's patent free.
This is the license for the "old" SILK codec. The patent licenses for Opus has nothing to do with that. Please read them:
Xiph.Org IPR statement: https://datatracker.ietf.org/ipr/1524/
Broadcom IPR statement: https://datatracker.ietf.org/ipr/1526/
Skype IPR statement: https://datatracker.ietf.org/ipr/1525/
Opus: the Swiss army knife of audio codec
Skype will release their patents under a free software compatible license if the codec is standardized by the IETF: https://datatracker.ietf.org/ipr/1525/
The sad thing is it shouldn't be better than HE-AAC. Being low latency does tend to mean one is better at the kind of time-domain issues many find so objectionable, but outside that OPUS is really packing a MUCH smaller toolkit than HE-AAC.
This is really egg on AAC's face, IMHO, and quite the upset. OPUS is so immature the bitstream isn't even stable yet.
What makes you say that? If you find a real issue, please raise it -- either on the mailing list: codec@ietf.org, or to me privately (jmvalin@jmvalin.ca). Skype is on the good side on this one. The technology they have contributed is very useful and they're open about resolving any licensing issue.
Opus: the Swiss army knife of audio codec
If we were talking about a 96 kb/s test, I'd agree with you. But at 64 kb/s, HE-AAC sounds much better than AAC-LC. The guys who organized this test picked the best AAC implementation they could find at the rate the test was run at.
Opus: the Swiss army knife of audio codec
You do realize that most modern VoIP hardware / software supports out of band DTMF? In fact, the most modern software demands it.
He was discussing accuracy as being irrelevant because perception is more important in a medium designed to be perceived by a human. You've now apparently converted it to "because fewer people care about accuracy", which was in no way his point. Or: Straw man.