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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.
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
...in your opinion.
Is that 5~22.5ms of latency on top of network latency?
[Fuck Beta]
o0t!
Even back when I used to play games online with voice chat
Imagine Rock Band with voice chat. Or imagine actually making real music with voice chat.
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.
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..
Surely you mean AAC-LC not LE-AAC?
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?
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.
...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)
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.
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/
In something like an actual telephone conversation it creates awkward pauses when each person is finished speaking of a length equal to twice the latency. High latency codecs also greatly encumber echo cancellation algorithms and hardware, which is extremely important in VoIP as anyone who has had to deal with it would know.
"Low-Latency" is in the summary title, and that's really the best you can do for a first post?
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.
i rather am ok with the sound of AAC-HE opposed to MP3 (and MP3Pro), WMA9Prowhatever, and older OGG. haven't heard OGG in a while, tho.
however, my first test with Opus/CELT blew me away...
http://www.multiupload.com/HTIGW82UD0
i suppose in hindsight i could provide the original lossless clip (cut after the fact) to play with... http://www.multiupload.com/XVRXX256WO
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
Yes, but for digitally re-un-non-illossless compression I would go with the Foobar Audio Framework.
We always knew Comcast was corrupt, here's the proof: http://tech.slashdot.org/comments.pl?sid=1909890&cid=34545432
Ah damnit, I was gonna say it was powered by The Power of Greyskull.
Of course parent doesn't have anything substantial to support his opinion.
This is slashdot, where opinions bear more credibility than point-of-fact
"Suppose you were an idiot...and suppose you were a member of Congress...but I repeat myself." Mark Twain
Not sure what AC is talking about, seems to be what the site says it does. I downloaded a sound file. This is the first time I have heard the codec, and it does sound extremely good. I don't know much about these matters, but I liked that it was "open", and could be relevant to my interests somehow.
Like a city whose walls are broken down is a man who lacks self-control.
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.
what kind of latency do you get with AAC? Do you know? (I'm trying to find out now via google)
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.
Jesus dude, did you just get done reading a high school debate book or something? None of what you posted makes any sense. I call your method of argument "Non Sequitur Appeal to Misused Logical Fallacies", I just haven't added it to the Wiki yet so dunces like you can misapply and mangle it.
For simple half-duplex systems like gaming, more lag is not really noticeable.
The only practical difference between gaming VOIP and Skype is having to hit a push-to-talk button. Latency issues like people stepping on each other crop up in gaming VOIP in much the same way that they pop up in high-latency cell phone or Skype conversations.
Make me a friend and I'll mod you up
For simple half-duplex systems like gaming, more lag is not really noticeable.
The only practical difference between gaming VOIP and Skype is having to hit a push-to-talk button. Latency issues like people stepping on each other crop up in gaming VOIP in much the same way that they pop up in high-latency cell phone or Skype conversations.
Not really. You're not (typically) having a back-and-forth conversation while gaming, just announcing your information and clearing the channel. So there is little difference, conversationally speaking, if your burst is delayed by half a second or so. It's not a conversation, it's a series of announcements. With noticeable lag in a phone call, however, you'll find yourself (and the caller/callee) tripping over each other's sentence beginnings as you both play the "no, after you" as the lag causes you (and your train of thought) to be interrupted. Add to that the highly distracting nature of hearing your own words back after a half-second delay (try it, it's very confusing and distracting) that a lack of echo cancellation can cause, and you have a recipe for conversational disaster.
The problem arises when two people have an announcement to make at the same time, usually when they're both waiting for another person to finish making their own announcement. Also don't forget that gaming VOIP software is quite often used for social purposes (VOIP use in public server TF2 is very very rarely related to the game at hand), and occasionally used by casters for commentating as well. It absolutely needs to live up to the same demands that "conversational" VOIP software needs to live up to.
Make me a friend and I'll mod you up
Parent post is complete bullshit. HE-AAC greatly outperforms LC-AAC at 64kbps. This can be seen in several previous listening tests, including the ITU ones that standardized the format itself.
Holy snap!
Rampant carbon sequestration destroyed the Dinosaurs' tropical paradise. I'm here to help repair the damage.
I don't know who you are, sir, but I like you.
Rampant carbon sequestration destroyed the Dinosaurs' tropical paradise. I'm here to help repair the damage.
Are you Daniel French, aka nirvgorilla?
Doesn't the test linked in the summary put Vorbis almost up to par with Nero's encoder? With both of them smoking AAC-LC?
Analogies don't equal equalities, they are merely somewhat analogous.
More evidence that FLOSS only copies and can't innovate!
Analogies don't equal equalities, they are merely somewhat analogous.
If you would have RTFA, you would have seen that the actual p value was smaller than 0.000 (99.99%) , not smaller than 0.050 (95%). This even accounts for all comparisons being performed, so the famous xkcd green jelly beans comic does not apply.
Opus is better than Vorbis (p=0.000)
Opus is better than Nero_HE-AAC (p=0.000)
Opus is better than Apple_HE-AAC (p=0.000)
You would also have seen that your "higher bitrate" comment makes no sense, because all codecs were run with settings that average 64kbps on a large corpus of music. The fact that some codecs ended up with slightly higher or lower bitrates on the (much smaller) selected sample does not change that. The full reasoning is again explicitly explained in TFA.
Vorbis predates HE-AAC by several years, so staying competitive with what used to be one of the best HE-AAC encoders before Apple started working on theirs ain't bad at all.
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.
Appealing to Wikipedia's authority?
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.
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Is that you NBCA?
Like a city whose walls are broken down is a man who lacks self-control.
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