It’s Opus, it rocks and now it’s an audio codec standard

https://hacks.mozilla.org/2012/09/its-opus-it-rocks-and-now-its-an-audio-codec-standard/

by nullc • 14 years ago

167 65 14 years ago

hacks.mozilla.org

nvmc2• 14 years ago

MP3 is getting very long in the tooth, not to mention the fact that it's encumbered by patents. A new codec would be nice, but hardware adoption of the codec will take a while. Long enough, dare I say, to kill it dead in the water.

randallu2• 14 years ago

Why would you say that?

Most "hardware" decoders are DSPs that run software loaded onto them at runtime. This goes for anything with an OMAP, Tegra or Samsung (Samsung's own decode block) SoC in (that I know for sure) and probably almost everything else. Tegra3 and OMAP4 can decode WebM in hardware, for example.

wmf2• 14 years ago

The problem is that they never develop firmware updates so if those DSPs don't support yet another open codec on release day, they never will.

nvmc2• 14 years ago

I should have worded my comment better. I mean that adoption by hardware manufacturers will be slow. What's the incentive? Even for a firmware patch? MP3 is `good enough', and everyone already uses it.

pslam• 14 years ago

Almost no hardware audio decoders use DSPs, or at least actually need to, since about a decade or more now.

The manufacturers you mention use DSPs for bad reasons - usually the inertia of software already using a coprocessor and not being willing to license a new codec for the general purpose core. Audio decoding and encoding is an extremely low percentage of available CPU, and it's just not worth off-loading it.

Video is another matter, at least when talking about HD. It's still very much worth using special purpose DSPs for that.

For audio, the existence of hardware DSPs is simply not a barrier to entry.

kibwen3• 14 years ago

Before someone chimes in with that XKCD strip:

How does Opus compare to other codecs?

Opus is distinguished from most formats for high quality audio (AAC, Vorbis, MP3) by having low delay and it is distinguished from most low delay formats (G.711, GSM, Speex) by supporting high audio quality. It meets or exceeds existing codecs' quality across a wide range of bitrates, and it operates at lower delay than virtually any existing compressed format. Further, the Opus format itself and the reference implementation are available under liberal royalty-free licenses, making it easy to adopt, compatible with free software, and suitable for usage as part of the basic infrastructure of the Internet. See the Opus comparison page for more details.

http://wiki.xiph.org/OpusFAQ#How_does_Opus_compare_to_other_...

Which is to say, this isn't (okay, isn't just) a case of having "too many standards"; Opus is legitimately impressive from a technical standpoint. Oh, and it's unencumbered by patents. That's cool.

EDIT: And here's the coolest part of the article that completely took me by surprise:

"The IETF recently decided with “strong consensus” to adopt Opus as a mandatory-to-implement (MTI) codec for WebRTC, an upcoming standard for real-time communication on the web."

wissler10• 14 years ago

There is no such thing as "unencumbered by patents". Unless you have a technology that has already been tested thoroughly in the courts, which is the only way you can really determine whether something would be deemed to violate a patent or not, you simply can't know.

azakai5• 14 years ago

Even if it was tested by the courts, you can appeal. And even after the appeal, you can appeal to a higher court, all the way to the Supreme. And even if was tested there, congress can pass a law that allows it. And even if that happens the Supreme court can strike down the law. And even if that happens, the public can mount a revolution and tear down the legal system.

But, practically speaking, Opus has extremely good guarantees for being unencumbered, as much as is possible - the people working on it have given licenses to use it, and experts in the field have had a while to go over this stuff. Nothing is 100%, but this is as good as it gets.

shmerl8• 14 years ago

It means to the best of authors' knowledge.

derleth• 14 years ago

> There is no such thing as "unencumbered by patents".

When do the lawsuits over PNG begin?

smiddereens• 14 years ago

I don't know which XKCD strip you're referring to; link?

kibwen3• 14 years ago

http://xkcd.com/927/

Keep in mind that this strip is a strong contender for the title of "Most Frequently-Posted Link in HN Comment Threads."

That would be an interesting analysis, actually.

lloeki• 14 years ago

> And here's the coolest part of the article that completely took me by surprise: "The IETF recently decided with “strong consensus” to adopt Opus as a mandatory-to-implement (MTI) codec for WebRTC, an upcoming standard for real-time communication on the web."

Indeed. There may be interesting side consequences for that. The first article comment asks:

> Interesting! Any idea if this will become mandatory to implement for HTML5 audio as well? That’s another area in sore need of a single format that works everywhere.

This means that if a browser supports WebRTC, it has to support Opus, and it then happens that the codec will be available for <audio>. Now that does not mean that <audio> => Opus support, but it will have a (increasingly with time) high chance to be so.

AceJohnny2• 14 years ago

I'm not familiar with Opus. Can someone tell me if it is able to dynamically adapt quality based on available bandwidth? If it loses packets in the stream, do we get a 'skip' in the audio or does the quality go down?

tqs• 14 years ago

It is absolutely able to dynamically adapt based on bandwidth, and is robust to packet loss. It was designed for that :)

http://www.opus-codec.org/ https://wiki.xiph.org/OpusFAQ

tveita• 14 years ago

> Can someone tell me if it is able to dynamically adapt quality based on available bandwidth?

Depends on what you mean. It is a variable bitrate codec, and the encoder can easily adjust the bitrate of the stream. But you would need a way to discover the available bandwidth and communicate it to the encoder - Opus is just a codec, not a container or streaming protocol.

> If it loses packets in the stream, do we get a 'skip' in the audio or does the quality go down?

The standard doesn't mandate a specific way to deal with packet loss, but the reference implementation tries to handle this gracefully. [1]

Some properties of the encoding facilitate this: for instance, packets can optionally include a reduced-bitrate encoding of the previous packet to provide redundancy. [2]

[1] http://tools.ietf.org/html/rfc6716#section-2.1.8

[2] http://tools.ietf.org/html/rfc6716#section-4.4

nullc5• 14 years ago

Yes, it's absolutely made for that, which is one of the reasons it has extreme bitrate and frame size agility. Though the decoder doesn't have a back channel to the encoder as part of the format (e.g. a duplex pair is not at all coupled by the codec and could have entirely different settings). So if you want to do rate adaptation the application will have to take care of telling the encoder that the decoder is struggling/losing frames/etc.

The codec also includes a pretty good packet loss concealer integrated, and can also use forward error correction to reduce the impact of loss.

mtgx• 14 years ago

I expect Firefox and Chrome to adopt this within months, but IE10 is coming out soon. Does it mean it won't have it? And then we'll have to wait another year or more for IE11 to implement it? Not to mention IE9 and IE8 will probably never get it.

I also assume WebM will replace Vorbis with Opus very soon?

bicknergseng• 14 years ago

While MS might not take full advantage of it, IE 10 will have silent updates. I would hope that would mean they drop the monolithic update strategy.

kibwen3• 14 years ago

The article mentions that Skype was a party to the standardization process, and hence Microsoft by association. So there's a small reason to hope.

azakai5• 14 years ago

> I expect Firefox and Chrome to adopt this within months,

Firefox already adopted it (it's in nightly builds). Chrome seems to have WontFixed it,

http://code.google.com/p/chromium/issues/detail?id=104241

nullc5• 14 years ago

It's actually in Firefox 15, not just nightly. People on Reddit are saying chrome's position has reversed but I don't have a link. Works in Opera already if you have gstreamer's opus plugin, and they've apparently said they'll ship it.

mcantor• 14 years ago

I'm not an audio guy. Can anyone explain like I'm five why this new codec was needed? How does it affect me as a consumer of audio? Who out there was saying "Gosh, I want to do X, but it's really tough because existing codecs force me to choose between high quality and low delay"? What is an example of "X" in that sentence?

azakai5• 14 years ago

There was no existing codec that is both good and not encumbered by patents.

Opus is good and not encumbered by patents, and it has just been standardized.

jmillikin• 14 years ago

X = {voip, videoconferencing}

Say you wanted to write a live audio chat app to talk to your friends. If you'd chosen the Speex codec, then voice would come over fine, but trying to show off your new guitar skills would fail because what they heard would sound horrible.

If you'd chosen the Vorbis codec, then everything would sound great, but there'd be a weird "lag" between participants, as if one of them were on the moon. Also, adding too many listeners would severely strain your internet connection's capacity.

Opus has latencies as good as Speex, and can be high-, medium-, or low-quality depending on bandwidth needs.

nnethercote• 14 years ago

Look at the graph in the article. Opus is as good as or slightly better than all the existing codecs at their strong points, and much better than them at their weak points.

YokoZar• 14 years ago

How does Opus compare to Celt, its direct predecessor?

rdtsc• 14 years ago

It actually includes code from Celt and merged with code from Silk (the Skype codec).

Silk is very good at speech and Celt is very good at music. It is like having gears in your car. You need low gears for one mode of operation and high gears for another. So they combined the two into one and made a single codec.

I imagine behind the scenes it has a transition mode from one codec to another depending on the configuration parameters.

derf_• 14 years ago

The final version of libcelt is bit-for-bit compatible with Opus. Unlike SILK, there was never a "stable" version of CELT (we routinely broke the bitstream on almost every commit), so CELT development continued after the two codecs were merged, right up until the Opus bitstream was frozen.

That makes Opus a strict superset of CELT, and we've stopped maintaining libcelt, so any existing users should switch to libopus.

mmphosis• 14 years ago

These comments answer some questions I have concerning FLAC...

starwed wrote on September 11th, 2012 at 1:29 pm:

Isn’t FLAC already* open and royalty-free?*

Dusk wrote on September 11th, 2012 at 1:40 pm:

FLAC is a lossless codec — it typically generates output at a bitrate orders of magnitude* higher than lossless codecs such as OPUS, and there’s no way to “dial it down”.*

natermer wrote on September 11th, 2012 at 2:30 pm:

You want to archive audio with 100% quality using Flac.

Opus is for were you are willing to give up some amount of audio quality for the ability to massively increase compression ratio. Using low bit rate Opus may yield a 3MB file for a song, were as FLAC may end up being 150MB for the same audio.

What I do for audio is to store it using flac, but encode it on the fly using other codecs for streaming for transfering to devices. That way I always get the best audio quality possible to go along with the best compression possible.

masklinn2• 14 years ago

> Using low bit rate Opus may yield a 3MB file for a song, were as FLAC may end up being 150MB for the same audio.

3MB will get you POTS-quality audio though. Lossless codecs usually compress audio to half the original size (so a 150MB FLAC file — for music — will correspond to ~300MB of raw bitstream). Lossy compression to something worth listening to will usually be around 10x compression, so ~30MB for an original 300MB bitstream.

A 35mn song compressed in 3MB probably won't be worth listening to.

ZoFreX2• 14 years ago

If you were using AAC (or anything else that's more advanced than MP3) a 3MB file would actually be listenable for a short song.

A 150MB FLAC file would be about 20-30 minutes long (based on a typical bitrate of 700-1000 kilobits per second).

sandGorgon• 14 years ago

Relevant - Hydrogenaudio listening tests (530 tests) scored Opus higher than HE-AAC

https://people.xiph.org/~greg/opus/ha2011/ http://listening-tests.hydrogenaudio.org/igorc/results.html

It’s Opus, it rocks and now it’s an audio codec standard

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