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Topic: Opus 1.1.1 is out (Read 26979 times) previous topic - next topic
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Opus 1.1.1 is out

Reply #50
After carefully switching between the files at the Dire Straits part, I've noticed slightly less annoying artifacts in comp_is1. The stereo doesn't sound as wide as some of the other files, but I prefer comp_is1.
Hope this helps.

Opus 1.1.1 is out

Reply #51
I couldn't spot differences between is1 and is2. Maybe somewhere better somewhere worse though all 6 concatenated samples sound practically the same for me.
Now will try t12 and t14.

Opus 1.1.1 is out

Reply #52
I quickly ABXed comp_is1.wav and comp_is2.wav to make sure that I wasn't just imagining differences:

Code: [Select]
foo_abx 2.0.1 report
foobar2000 v1.3.9
2015-12-31 23:23:12

File A: comp_is1.wav
SHA1: 7309290e4bfb738eaa9c1cc92bc02e0978ad568e
File B: comp_is2.wav
SHA1: c4913af74913d7dacb4a8c719338da92441410ff

Output:
DS : Primary Sound Driver
Crossfading: NO

23:23:12 : Test started.
23:23:58 : 01/01
23:24:38 : 02/02
23:24:55 : 03/03
23:25:17 : 04/04
23:25:45 : 05/05
23:27:05 : 06/06
23:27:21 : 07/07
23:27:40 : 08/08
23:28:08 : 09/09
23:28:52 : 10/10
23:31:05 : 11/11
23:31:24 : 12/12
23:36:34 : 13/13
23:36:54 : 14/14
23:37:30 : 15/15
23:40:14 : 16/16
23:40:14 : Test finished.

----------
Total: 16/16
Probability that you were guessing: 0.0%

-- signature --
f3bbef890ada779ab9231f141e7eff2112517574

Opus 1.1.1 is out

Reply #53
Happy New Year everyone! Jean-Marc, I finally got around listening to your 32/38-kbps tuning attempts.

Thanks. So it looks like you like less intensity stereo...
...
Well, the t14 file you said you preferred is actually the one with the least amount of intensity stereo and other stereo tricks.

I think I actually like the t14 version the least. Yes, the stereo image is wider, but I also hear more birdies on some transients (due to more spectral holes or folding, I guess, esp. Dave Matthews Band, woodblocks, a bit also on Dire Straits). No preference on the other versions. Is comp_orig the Opus 1.1.1 coded one?

Quote
Parametric stereo includes intensity stereo, but also controlling inter-channel phase difference (time of arrival), and inter-channel coherence (diffuse vs directional sound). I think the phase difference part is a stupid idea...

What makes you say that? For items like CantWait with its near-90-degrees inter-channel phase difference I would guess a respective stereo parameter would be very useful to be able to lower the intensity/parametric stereo start frequency.

And regarding Igor's post here: by "Fraunhofer HE-AAC" I guess you mean the encoder in Winamp (not fdk), right? (Edit: note to myself and anyone interested: some news about current status of Winamp)

Chris
If I don't reply to your reply, it means I agree with you.

Opus 1.1.1 is out

Reply #54
And regarding Igor's post here: by "Fraunhofer HE-AAC" I guess you mean the encoder in Winamp (not fdk), right?

Yes,  it was the latest version of FhG encoder from the latest Winamp.

Happy New Year, guys! 

Opus 1.1.1 is out

Reply #55
Quote
Parametric stereo includes intensity stereo, but also controlling inter-channel phase difference (time of arrival), and inter-channel coherence (diffuse vs directional sound). I think the phase difference part is a stupid idea...

What makes you say that? For items like CantWait with its near-90-degrees inter-channel phase difference I would guess a respective stereo parameter would be very useful to be able to lower the intensity/parametric stereo start frequency.

90-degree out of phase generally means an uncorrelated, diffuse field, which AFAIK is handled through inter-channel coherence (IC), rather than with inter-channel phase difference (IPD). IC is indeed useful. OTOH, IPD is related to time delay of arrival, which is a cue that we only perceive below about 1-2 kHz. The problem is that in that range the ear is very sensitive to phase, and messing it up causes really bad artefacts. I believe it's the reason why HE-AACv2 makes me motion sick when I listen to it with headphones.

Opus 1.1.1 is out

Reply #56
90-degree out of phase generally means an uncorrelated, diffuse field, which AFAIK is handled through inter-channel coherence (IC), rather than with inter-channel phase difference (IPD).
I have to disagree with the first part of your sentence. Attached are three WAV files I created with the Octave/Matlab script below:
  • [attachment=8480:noise90.wav]fully correlated (i.e. identical) noise sources in both channels, left channel created using forward MDCT + inverse MDCT, right channel created using forward MDCT + inverse MDST to achieve a 90-degree phase shift,
  • [attachment=8481:noiseUC.wav]fully uncorrelated (i.e. independent) noise sources for the two channels, both left and right channel created using forward MDCT + inverse MDCT,
  • [attachment=8482:noiseUC90.wav]same as noiseUC, but right channel created using forward MDCT + inverse MDST (which doesn't matter since the noise sources are uncorrelated).
I find the noise90 file to sound much more like a (somewhat distorted) point source than the (truly diffuse sounding) noiseUC* files. By the way, I could have used a FFT instead of the MDCT/MDST here as well, the results sound the same.
I haven't checked, but a parametric stereo codec without IPD support would probably turn noise90 either into true mono or into something which sounds like the noiseUC* versions, by handling the IPD by way of the IC parameter, as you say.
Of course, this is an extreme demonstration. I have only observed such stable long-term IPDs in some very few Dolby Surround encoded tracks. And the CantWait sample.

Quote
OTOH, IPD is related to time delay of arrival, which is a cue that we only perceive below about 1-2 kHz. The problem is that in that range the ear is very sensitive to phase, and messing it up causes really bad artefacts. I believe it's the reason why HE-AACv2 makes me motion sick when I listen to it with headphones.
Indeed. I don't like the sound of HE-AAC's Parametric Stereo either. In fact, its phase coding is so expensive and unintuitive that I'm not even sure it's actually used. Which might be the reason why HE-AAC v2 sounds quite bad on "phasy" items like the abovementioned CantWait. Extended HE-AAC (USAC) sounds much better in such cases.

Code: [Select]
N = 1024; % frame length
F = 94;  % number of frames
% mdct4() and imdct4() are from www.ee.columbia.edu/~marios/mdct/mdct_giraffe.html
w_sin = sin(pi*(0.5:2*N)'/(2*N));
outUC = zeros(F*N, 2);
out90 = zeros(F*N, 2);
outUC90 = zeros(F*N, 2);
rand1 = rand(F*N, 1) - rand(F*N, 1) + rand(F*N, 1) - rand(F*N, 1);
rand2 = rand(F*N, 1) - rand(F*N, 1) + rand(F*N, 1) - rand(F*N, 1);
for i = 0 : F-2
  spec = mdct4(w_sin .* rand1(i*N+1 : i*N+2*N));
  out90(i*N+1 : i*N+2*N, 1)  = out90(i*N+1 : i*N+2*N, 1)  + w_sin .* imdct4(spec);
  outUC(i*N+1 : i*N+2*N, 1)  = outUC(i*N+1 : i*N+2*N, 1)  + w_sin .* imdct4(spec);
  outUC90(i*N+1 : i*N+2*N, 1) = outUC90(i*N+1 : i*N+2*N, 1) + w_sin .* imdct4(spec);
  spec(1:2:end) = -spec(1:2:end);
  temp = w_sin .* imdct4(spec); % IMDST via IMDCT
  out90(i*N+1 : i*N+2*N, 2)  = out90(i*N+1 : i*N+2*N, 2)  + temp(end:-1:1);

  spec = mdct4(w_sin .* rand2(i*N+1 : i*N+2*N)); % uncorrelated second channel
  outUC(i*N+1 : i*N+2*N, 2)  = outUC(i*N+1 : i*N+2*N, 2)  + w_sin .* imdct4(spec);
  spec(1:2:end) = -spec(1:2:end);
  temp = w_sin .* imdct4(spec); % IMDST via IMDCT
  outUC90(i*N+1 : i*N+2*N, 2) = outUC90(i*N+1 : i*N+2*N, 2) + temp(end:-1:1);
end
wavwrite(out90*0.25,  48000, 16, 'noise90.wav');
wavwrite(outUC*0.25,  48000, 16, 'noiseUC.wav');
wavwrite(outUC90*0.25, 48000, 16, 'noiseUC90.wav');
Chris
If I don't reply to your reply, it means I agree with you.

Opus 1.1.1 is out

Reply #57
I find the noise90 file to sound much more like a (somewhat distorted) point source than the (truly diffuse sounding) noiseUC* files. By the way, I could have used a FFT instead of the MDCT/MDST here as well, the results sound the same.
I haven't checked, but a parametric stereo codec without IPD support would probably turn noise90 either into true mono or into something which sounds like the noiseUC* versions, by handling the IPD by way of the IC parameter, as you say.
Of course, this is an extreme demonstration. I have only observed such stable long-term IPDs in some very few Dolby Surround encoded tracks. And the CantWait sample.

My point is that actual recordings are highly unlikely to contain a constant 90-degree inter-channel difference. Either the phase is random (averages to 90, but not constant) as is the case in a diffuse noise field, or else it's a linear phase caused by a different delay of arrival between the channels. And as I stated earlier, the ear is only sensitive to such delay of arrival below 1-2 kHz, so it's useless to model it at high frequency. And for low frequencies, the ear is just *too* sensitive to phase, that trying parametric tricks is bound to cause motion sickness. That's why I say it's pointless to use parametric phase. For the record, here's some previous experimentation I did with phase.

Opus 1.1.1 is out

Reply #58
And as I stated earlier, the ear is only sensitive to such delay of arrival below 1-2 kHz, so it's useless to model it at high frequency.

Yes, of course, I'm not questioning that. Or rather, above ~2 kHz delay of arrival should be modeled using inter-channel intensity differences with (if necessary) high temporal resolution.

Quote
And for low frequencies, the ear is just *too* sensitive to phase, that trying parametric tricks is bound to cause motion sickness. That's why I say it's pointless to use parametric phase.

Fair enough. But if the bit-rate is so low that you can't even afford coding a band-limited second (residual) channel, what would you do? Surely, using an intensity-stereo approach below 2 kHz would not be a good idea... regardless of whether or not you (can) model the IC.

Chris
If I don't reply to your reply, it means I agree with you.

Opus 1.1.1 is out

Reply #59
And as I stated earlier, the ear is only sensitive to such delay of arrival below 1-2 kHz, so it's useless to model it at high frequency.

Yes, of course, I'm not questioning that. Or rather, above ~2 kHz delay of arrival should be modeled using inter-channel intensity differences with (if necessary) high temporal resolution.

No need for high temporal resolution. We're deaf to time delay of arrival above 2 kHz. It's only the intensity that matters.

Quote
Quote
And for low frequencies, the ear is just *too* sensitive to phase, that trying parametric tricks is bound to cause motion sickness. That's why I say it's pointless to use parametric phase.

Fair enough. But if the bit-rate is so low that you can't even afford coding a band-limited second (residual) channel, what would you do? Surely, using an intensity-stereo approach below 2 kHz would not be a good idea... regardless of whether or not you (can) model the IC.

I'm not sure what I would do, but I know what I *wouldn't* do... mess around with the phase (and that includes IC below 1 kHz). At least intensity stereo doesn't make you motion sick :-)

Opus 1.1.1 is out

Reply #60
Does any one know if Opus has been selected for VoLTE yet? Or Will that be EVS?

How does EVS compared to Opus in VoLTE use case?





Re: Opus 1.1.1 is out

Reply #65
Does any one know if Opus has been selected for VoLTE yet? Or Will that be EVS?

How does EVS compared to Opus in VoLTE use case?

It seems they have chosen EVS instead, anyone know why?


Re: Opus 1.1.1 is out

Reply #67
Looking at the EVS DMOS chart, at 16 and 24 kbps Opus is rated as being equivalent to whichever EVS codec is giving the same bandwidth, and I wonder to what extent the ratings it gets at lower rates are due to it going to mediumband and narrowband. Where would e.g. LP3.5, LP7 fit on this chart? And many of the recent Opus changes have been focused on increasing bandwidth at low bit rates.

Honestly, I find it odd how much emphasis increasing the audio bandwidth gets. Especially for speech. I wonder whether test conditions in most listening tests exaggerate the importance of high frequencies, especially for speech. Silent listening environment, and crystal-clear test samples usually recorded in studio conditions and given some processing. Rating via comparison to original - even with a hidden reference people are going to try to focus on figuring out which is the original and compare to that - rather than assessing intelligibility or pleasantness by other means.

I mean, the difference between narrowband and mediumband is huge- narrowband misses critical information that we always use to tell different consonants apart and it's painful to listen to. But I would think that the returns diminish steeply from there.

For speech samples I worked with (academic conference recordings done with TASCAM DR-05 built in mics rather than controlled environments and high-end equipment) I had people tell me in blind testing that they preferred wideband-lowpassed versions to fullband. (They found it more intelligible and less distracting without the high frequencies which even after intelligent postprocessing were still mostly noise.) And SWB was not readily ABXable from fullband.

Re: Opus 1.1.1 is out

Reply #68
Why no one is talking about 1.1.2 release?

I wanted to notify Peter as well for the next foobar2000 beta but he probably knows by now and I don't want to start a new thread just for this.

 

Re: Opus 1.1.1 is out

Reply #69
Just saw someone posted the release sorry guys I missed for few months and I'm slow to see things with the new forum as well.

 
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