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Opus 1.1 released, For real this time
jmvalin
post Dec 6 2013, 03:56
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After more than two years of development, we have released Opus 1.1. This includes:
  • new analysis code and tuning that significantly improves encoding quality, especially for variable-bitrate (VBR)
  • automatic detection of speech or music to decide which encoding mode to use.
  • surround with good quality at 128 kbps for 5.1 and usable down to 48 kbps.
  • speed improvements on all architectures, especially ARM, where decoding uses around 40% less CPU and encoding uses around 30% less CPU.
These improvements are explained in more details in Monty's demo (updated from the 1.1 beta demo).
Of course, this new version is still fully compliant with the Opus specification (RFC 6716).
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IgorC
post Dec 6 2013, 06:15
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Thank You to all developers and people involved in development of Opus. smile.gif
This is a very impressive release.

How much time and effort have been put into it.

beer.gif
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themanintheshado...
post Dec 6 2013, 08:15
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I wonder how long it'll take to get a binary?
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eahm
post Dec 6 2013, 08:20
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QUOTE (themanintheshadows_2451 @ Dec 6 2013, 00:15) *
I wonder how long it'll take to get a binary?

It's already there: http://opus-codec.org/downloads/

opus-tools 0.1.8 has libopus 1.1.

This post has been edited by eahm: Dec 6 2013, 08:21


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RobertM
post Dec 6 2013, 09:19
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Big thanks to the Opus team. This is a fantastic release with great new features.
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darkbyte
post Dec 6 2013, 10:20
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Thank you! smile.gif Is there a list somewhere which details the fixed bugs, or my only source for this is the commit log right now?

I'm very angry now for the Poweramp guy to not include the decoder in it's player. We've asked for that more than a year ago. mad.gif

This post has been edited by darkbyte: Dec 6 2013, 10:21


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jmvalin
post Dec 6 2013, 10:32
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QUOTE (darkbyte @ Dec 6 2013, 04:20) *
Thank you! smile.gif Is there a list somewhere which details the fixed bugs, or my only source for this is the commit log right now?


For the bug fixes (as opposed to the new features in Monty's demo), I guess the shortlog is your only option.
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Anakunda
post Dec 6 2013, 11:22
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QUOTE
surround with good quality at 128 kbps for 5.1 and usable down to 48 kbps, and

I'm curious if there's upcoming directshow filter for opus, like updated OpenCodecs or another 3rd codec bundle?
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includemeout
post Dec 6 2013, 11:38
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QUOTE (jmvalin @ Dec 6 2013, 04:56) *
(...) These improvements are explained in more details in Monty's demo (updated from the 1.1 beta demo).
Of course, this new version is still fully compliant with the Opus specification (RFC 6716).


Really impressive work (along with the way you present those changes - which really caters to laymen such as myself). tongue.gif

Thank you!


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ChronoSphere
post Dec 6 2013, 13:44
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I wanted to post the results of rockbox runtime on the clip+ with the 1.1, then realized I forgot to start the battery benchmark... It does seem to have a longer battery time now though, compared to previous build.

QUOTE
Surround masking takes advantage of cross-channel masking between free-field loudspeakers. Obviously, we can't do that for stereo, as stereo is often listened to on headphones or nearfield monitors, but for surround encodings played on typical surround placements with listeners placed well within the soundfield, there's considerable savings to be had by assuming freefield masking.
Curious, what does this mean for the few real surround headphones? (3 speakers per side)
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TomasPin
post Dec 6 2013, 18:23
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Guess I'll finally give this a try, thanks for all the hard work. smile.gif


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IgorC
post Dec 6 2013, 21:39
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1.1 had a stable quality already starting from alpha build. Probably people should lose a prejudice to don't try anything that's not a 100% final.
Instead they could report some certain situations those could be improved. Everybody would win.
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LigH
post Dec 6 2013, 21:44
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Are there again public ABX listening tests, like these famous events by Roberto J. Amorim? Version 1.1 promises a severe quality-per-bitrate boost, I wonder how certain this is approved by the audience.


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IgorC
post Dec 6 2013, 21:54
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Last time we have organized test it was 2011 and that was CELT 0.11.2.

I've sent a few mails and talked with some other people who was involved into last public test.
Most probably there will be a test, this time I will be just a co-organizer (or even less than that). It should be a new person who will recolect the results among the other functions.

This post has been edited by IgorC: Dec 6 2013, 22:00
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Octocontrabass
post Dec 7 2013, 02:27
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QUOTE (Anakunda @ Dec 6 2013, 02:22) *
I'm curious if there's upcoming directshow filter for opus,
LAV filters has supported Opus for a while now.
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ChristianK
post Dec 7 2013, 14:33
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This new release is great!
Seems to perform just fine and now handles some of my previously problematic files like I expect it to do.
Can't wait till it can be used properly for surround sound in movies (VLC doesn't yet like it very much in an MKV but a dev said it was on its list).
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ChronoSphere
post Dec 7 2013, 21:48
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Looks like rockbox+opus now manages to squeeze out 30 minutes more of runtime on my clip+ (10:04:40 -> 10:38:05 @160kbps).
That's great, but still way behind FLAC (16:09:49@lv8) or MPC (16:13:47@192kbps). Logs for comparison here
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saratoga
post Dec 7 2013, 21:50
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QUOTE (ChronoSphere @ Dec 7 2013, 16:48) *
Looks like rockbox+opus now manages to squeeze out 30 minutes more of runtime on my clip+ (10:04:40 -> 10:38:05 @160kbps).
That's great, but still way behind FLAC (16:09:49@lv8) or MPC (16:13:47@192kbps). Logs for comparison here


Just an hour behind your mp3 scores, which isn't too bad. I'm surprised mp3/vorbis do so poorly though, I get longer out of MP3 then you do out of any format.
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ChronoSphere
post Dec 7 2013, 23:51
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I think it's because of the settings. MP3 one is -V0. Vorbis was q4.0.
I forgot to specify the bitrates for the rest of them.

It might also be related to the CPU the clip+ has, or the fact I'm playing them off µSD, but that's something you know more about, I guess.

In any case, not a bad performance, but I'll still stick with FLAC for convenience of simply copying the files over, since my playlists are small and change frequently.

This post has been edited by ChronoSphere: Dec 7 2013, 23:53
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eahm
post Dec 8 2013, 03:07
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Can you test AAC? Particularly Apple's AAC?

This post has been edited by eahm: Dec 8 2013, 03:10


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Steve Forte Rio
post Dec 8 2013, 08:02
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Documentation (opusenc.html) claims that Opus supports bitrates beginning from 6 kbps per channel. But actually for bitrates <30 it performs downmixing to mono. At 30 kbps we have something (but very sick) on the side channel, and starting from 32 kbps we really have a stereo effect, but stereo pan is much narrower than original.

On other side FhG AAC's parametric stereo handles stereo at low bitrates very well. So which stereo encoding technology is used by the Opus? Looks like a simple Mid/Side encoding (but maybe separate for each frequency band, like in AAC).

P.S. Starting from about 40 kbps I get almost independent stereo encoding (I mean very low stereo crosstalk). However, I wonder what's this transition interval between 30 and 40 kbps.

This post has been edited by Steve Forte Rio: Dec 8 2013, 08:53
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jmvalin
post Dec 8 2013, 08:27
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QUOTE (Steve Forte Rio @ Dec 8 2013, 02:02) *
Documentation (opusenc.html) claims that Opus supports bitrates neginning from 6 kbps per channel. But actually for bitrates <30 it performs downmixing to mono. At 30 kbps we have something (but very sick) on the side channel, and starting from 32 kbps we really have a stereo effect, but stereo pan is much narrower than original.

Not all applications need stereo. At 6 kb/s, which is low bitrate even by VoIP standards, what you can have is narrowband mono.

QUOTE (Steve Forte Rio @ Dec 8 2013, 02:02) *
On other side FhG AAC's parametric stereo handles stereo at low bitrates very well. So which stereo encoding technology is used by the Opus? Looks like a simple Mid/Side encoding (but maybe separate for each frequency band, like in AAC).

Opus uses "normalized mid-side", which unlike MP3-type mid-side cannot introduce cross-talk (i.e. we could use it on every frame and still sound good). We also do a variant of intensity stereo. We do *not* do the kind of parametric stereo that HE-AAC v2 uses for multiple reasons. Obviously there'd be patent issues, but beyond that, parametric stereo introduces time-domain post-processing, which would increase latency (defeats one of the goals of Opus), increase complexity, and still not result in good quality (it's more of a race to the bottom). Personally, I find v2's PS extremely unpleasant to listen to (gives me motion sickness when listening with headphones) and I prefer mono. It makes nice demos, but that's about it.

QUOTE (Steve Forte Rio @ Dec 8 2013, 02:02) *
P.S. Starting from about 40 kbps I get almost independent stereo encoding (I mean very low stereo crosstalk). However, I wonder what's this transition interval between 30 and 40 kbps.

This is correct. The 1.1 encoder gradually transitions from mono to stereo between 30 and 38 kb/s. It's a tradeoff between coding artefacts and stereo image, but there's still some tuning left to do on this. If you're interested in fiddling with it, it's just a matter of changing two lines of code and recompiling.
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Steve Forte Rio
post Dec 8 2013, 08:57
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QUOTE (jmvalin @ Dec 8 2013, 05:27) *
Not all applications need stereo. At 6 kb/s, which is low bitrate even by VoIP standards, what you can have is narrowband mono.


Sure, but in fact that statement is incorrect and minimum 16 kbps per channel is needed to get stereo effects. Maybe you should make some corrections to the manual.
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jmvalin
post Dec 8 2013, 10:06
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QUOTE (Steve Forte Rio @ Dec 8 2013, 02:57) *
QUOTE (jmvalin @ Dec 8 2013, 05:27) *
Not all applications need stereo. At 6 kb/s, which is low bitrate even by VoIP standards, what you can have is narrowband mono.


Sure, but in fact that statement is incorrect and minimum 16 kbps per channel is needed to get stereo effects. Maybe you should make some corrections to the manual.


Euh, what exactly are you saying is incorrect in my statement? What does the manual say that you think should be changed?
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Steve Forte Rio
post Dec 8 2013, 11:17
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If the minimum bitrate is 6 kbps per channel (as stated in opusenc.html), then minimum bitrate for stereo encoding is 6*2=12 kbps. But in fact we get a mono encoding (one channel) at 12 kbps (and even up to 30 kbps).

This post has been edited by Steve Forte Rio: Dec 8 2013, 11:21
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