Slashdot Mirror
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.
HE-AAC uses SBR to reduce its data footprint. This results in worse reproduction of the source audio than LE-AAC at same bitrate (and often even lower bitrate). The whole deal with HE is that it can maintain good quality at very low bitrate, by giving up accuracy. So far, Apple's LE-AAC encoder in their Core Audio framework is the best choice for digitally non-lossless compression.
Patent free? Or royalty free?
For justice, we must go to Don Corleone
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
Who really would need such low latency? Even back when I used to play games online with voice chat, I could call out "incoming enemy!" and the other players on my team would have plenty of time to react.
Is that 5~22.5ms of latency on top of network latency?
[Fuck Beta]
o0t!
Who cares what codec is being used for my VoIP phone at home or on my desk, when anyone I call is still most likely to be connected over the PSTN with g.711 or g.723, or (far worse) a cell phone?
And don't get me wrong: I want to care; I really do. And maybe I did care, at one point. I was going to build an Asterisk system for home -- I even collected some of the hardware to make it work.
But I stopped caring when the boy got old enough to properly want a cell phone, the wife got a cell phone, and I had a cell phone. After that, I dropped the home phone line altogether, since it was just a waste of money.
I have no interest, at this moment, in having any sort of telephony tied to my premises.
And while I could, I suppose, run some manner of VoIP client on my Droid over cellular, I think that's a complete non-starter at the moment: I had trouble earlier today getting a 64kbps MP3 to stream correctly over 3G Verizon (even though I controlled both ends of the stream), but that was just an inconvenience.
It'd be a lot more than simply inconvenient if my phone calls were that spotty. I don't care how good it sounds if it doesn't work.
Is there any good and practical use for this new codec?
Kid-proof tablet..
I think HE-MAN is better than HE-ACC.
There's no -1 for "I don't get it."
if the codec cant reliably do dtmf detection, then its no good -- i'll stick with ulaw disallow=all allow=ulaw
To be honest, I didn't click most of those links in the summary, but I did check out the codec's website, and it made me wonder where I can find an app that actually uses this codec. I would be really interested in trying this out or participating in any kind of testing they might be doing since I live in China, Skype is uber-slow here and I do enjoy jamming from time to time. Anyone know how to put this codec to use yet?
...it can't have been "then known as CELT" since it is a merge of two codecs of which CELT is one and SILK is the other. It's good that it's an IETF standard as that will help some with adoption. It will also help some with getting other implementations. (Hell, Dirac is a great codec for video but because it's not a recognized standard for anything it's not getting used.)
It's a small world and it smells funny; I'd buy another if it wasn't for the money; Take back what I paid (SoM)
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/
"Low-Latency" is in the summary title, and that's really the best you can do for a first post?
4.56×10^6 milliseconds of latency and that's the best you can do?
The end user has no reason to care about this, right? It's just an implementation thing?
Unity? Screw that: XFCE. Slashdot Beta? Screw that: SoylentNews. Australis? Screw that: Pale Moon. UX developers DIAF
When you are dealing with audio signals in the home, low latency can be needed too. If you are doing something like playing prerecorded video then no, the system can find out the delays of the screen, audio, codecs, etc and insert delays as needed to sync it all up. However not if you are doing something live, like games. That's the reason for stuff like Dolby Digital Live and DTS Interactive. They are made so that you can get low latency encoding so the sound from a game console syncs up with the video.
It is also important for mobile phones. There's only so much latency you can tolerate in a conversation before things start to sound strange to the people using it. Of course there's already latency from the phone network, so codec latency matters. That is part of the reason why new phone standards aren't using something like AAC to get better sounding audio out of the bandwidth available.
As such this project is has a lot of really cool potential. If it not only offers better per-bit perceptual sound but also is extremely low latency, it can be used in situations the others can't.
...is at the top of the first Opus/CELT demo page:
http://people.xiph.org/~xiphmont/demo/celt/demo.html
The low latency makes more interactive applications possible. By way of illustration, the total algorithmic delay of an Opus or CELT stream is approximately equivalent to the time it takes sound to travel from you to someone standing five feet away.
"Open-Source Vorbis codec uses higher bitrates than a proprietry HE-AAC to provide worse quality"
What's more, that headline is actually backed up by statistically significant results. Those bars are 95% confidence -- 2 sigma. So we're saying that Opus is near as damn it 2 sigma better than Apple's HE-AAC (and at lower bitrates, which I'd have expected the summary to comment if, well, this wasn't Slashdot). That's good and they should certainly be pleased, but it's not really *that* statistically significant. 2 sigma results crop up all the time in science and are typically rejected as fluctuations. The thing that surprised me, since I've always liked Vorbis, is how it's about four sigma worse than Opus, and perhaps two and a half three or so worse than HE-AAC. At a higher bitrate. That's a fair old fail for Vorbis there, to go along with at least a nice result for Opus. I don't think they'll take much joy from the fact that they've been demonstrated to be as shit as Nero's HE-AAC implementation. While using a higher bitrate.
Am I misunderstanding, or is the headline "open codec designed for voip is slightly better for voip than closed codec designed for music"? How does it compare to the other voip codecs?
I mod down anyone who says "I will be modded down for this", regardless of the rest of their comment
Your point was to stand on top of your soapbox and proclaim "why should I care" -- as if the program was designed specifically for you. It's a surefire way to get modded up on slashdot: rush in and be the first to say "this is useless", even though it may be extremely useful and desirable for other people.
I think it's time to face the ugly truth: you're not the only person in the world, and your wants and needs aren't the same wants and needs as other people.
Game announcements have lag IRL. 200ms lag would be a distance of 60m between people. Easily possible on a battlefield (for example). Therefore a lag here is no problem in a simulated reality.
However, a phone conversation has you and the other person represented by the phone handset and are ~10cm away. A real conversation would have you two ~60cm apart. 2ms lag. Lag is therefore an introduced problem.
I'm curious what's the problem with Speex for voice transmission? (A non-rhetorical question.)
What about lower bitrates? HE-AAC is designed for low-bitrate audio, and 64 Kbps is right on the outside edge of where HE-AAC is useful. 24-32 Kbps is where HE-AAC really shines, and that's where stuff really gets impressive.
Cell phones, ISDN, and all the like operate at 64kbps.
Most users DSL lines have plenty more than 64kbps both directions, so 64kbps is also a safe bet for VoIP applications.
If hydrogene audio want to prove that this codec is a good replacement for the codecs currently used in phone, it has to be tested on the bandwith usually associated with phones.
"Sufficiently advanced satire is indistinguishable from reality." - [Tips: 1DrYakQDKCQ6y52z6QbnkxHXAocMZJE61o ]
I tried to comment on this a while ago but the latency messed me up.
I Need someone to rebuild a Digitech Digital Delay pedal for me....for me...for me...for me.
Seeing it is efficient, with low latency, it would be delightful if the codec could be enhanced to allow stereo music listening, without requiring MP3 as the only playback software. I am sure that hardware manufacturers would appreciate the elimination of licensing fees for MP3 support.
Leslie Satenstein Montreal Quebec Canada
It's not clear why you're asking about if Opus can handle stereo music when the test has shown that new codec handles it very well.
Opus has 4 (of max 5) points on music test at 64 kbps. It will be even better at 96 kbps.
Sorry, do we look at the same results? http://listening-tests.hydrogenaudio.org/igorc/results.html
The slashdot article described it's use as for cellphones. It did not state that it was a generalized codec that could replace mp3 or flac or other.
I am glad to read in your reply, that it does, and does it well.
Just reading the license and regarding silk licensing:
"Skype Limited hereby separately grants to the contributors a license
under its patents to the software contributed by Skype for internal
evaluation and testing purposes only. This license expressly excludes
use of the software for distribution or use in any commercial product or
any commercial or production use whatsoever."
I haven't really dug around or figured out how much silk goes into opus, but if there is any, I think you need a silk developers license.