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Dialog-Normalization & accuracy issues on the HDM lossless codecs?

post #1 of 5
4/18/08 at 9:28am
Thread Starter 
Dialog-Normalization & accuracy issues on the HDM lossless codecs?

Purpose of this thread:
To strive for "bit-for-bit accuracy" on the HDM lossless codecs.

What is known about dial-norm and DD TrueHD:

From another post:
Quote:
the problem with DN isn't that it lowers the dynamic range (volume) by 4 (or more) db... the problem is HOW it does it.

It does it by digitally recalculating the audio data to digitally down-scale the waveform.

That means that you will NEVER get bit-for-bit accuracy from a TrueHD stream that's been flagged with dialog-normalization.

And it can't be bypassed because Dolby won't allow any in-spec consumer gear to let the user avoid it! It doesn't matter that it's just a meta-data instruction flag: if Dolby REQUIRED that it's processing instructions be followed by your decoder, then the fact that the original PCM lossless data was represented prior to DN processing is moot.

Known accuracy issues wtih DTS-HD MA:
Quote:
Originally Posted by FilmMixer:

Originally Posted by MSmith83:

"I just listened to Shoot 'Em Up. The first thing I noticed is that the MA extension is significantly lower in volume than the core. When cranked to volume match, it is further evident that it's like listening to an entirely different mix. The MA extension has even deeper bass, but the most significant difference was the increased spaciousness of the effects and seemingly better channel separation.

I listened to the best scenes in Transporter 2 and the difference was more subtle (with no real difference in volume in case you're wondering), but it was a noticeable difference nonetheless. The same goes for Live Free or Die Hard.

The good thing about the PS3 is that direct comparisons are easy, since you can quickly switch from internal decoding of MA to bit-streaming of the core."

Interesting comments....

One thing doesn't make sense, though.. the extension only contains information that differs from the PCM master and the core once decoded...

Because of the nature of the core+extension process, there cannot be a great deal of difference in volumes between the two decodes (core vs. lossless) unless there is other processing going on...

DTS-HD MA does allow for dialnorm, so it may be active on the New Line titles.
__________________
Re-Recording Mixer, Hollywood, CA.
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FilmMixer
Marc Fishman
Re-Recording Mixer, ToddAO Studios
Studio City, CA
Film Sound and Post Production Insider


My first question is to ask if dial-norm can be turned-off or "by-passed" when and if studios use it on DTS-HD MA titles?
We already know it can't be "by-passed" on DD TrueHD.

Paul
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post #2 of 5
4/18/08 at 11:42am

Re: Dialog-Normalization & accuracy issues on the HDM lossless codecs?

Originally Posted by MSmith83:

"I just listened to Shoot 'Em Up. The first thing I noticed is that the MA extension is significantly lower in volume than the core. When cranked to volume match, it is further evident that it's like listening to an entirely different mix. The MA extension has even deeper bass, but the most significant difference was the increased spaciousness of the effects and seemingly better channel separation.


This doesn't make sense. There should be no difference in amplitude between the core track of DTS_HD MA and the full file including the extensions.

Sounds to me like there is a possible set-up issue with the receiver.
Why would the core sound "significant difference was the increased spaciousness of the effects and seemingly better channel separation."

Better channel separation and spaciousness on the core track of DTS, than with the Lossless extensions included?

Something is being overlooked here.

I have found a bug in the DD TRU-HD encoder that under certain circumstances it's not exactly bit for bit. It only affects the surrounds and does not degrade the signal one bit, but it does make the file larger in size and the peak bit-rate higher than it needs to be.

Dolby is currently working on a fix for this.

It won't affect the consumer really, just the people mastering the audio and authoring the discs (IF they notice it).

Roogs
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post #3 of 5
4/18/08 at 10:11pm

Re: Dialog-Normalization & accuracy issues on the HDM lossless codecs?

Quote:
What is known about dial-norm and DD TrueHD:

From another post:
Author and link, please. On this particular topic (DD and dialnorm), there has been a great deal of misinformation masquerading as expertise.

M.
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post #4 of 5
4/19/08 at 10:23am
Thread Starter 

Re: Dialog-Normalization & accuracy issues on the HDM lossless codecs?

Quote:
Originally Posted by Michael Reuben

Originally Posted by Paul Hillenbrand:
What is known about dial-norm and DD TrueHD:

From another post:

Author and link, please. On this particular topic (DD and dialnorm), there has been a great deal of misinformation masquerading as expertise.

M.
Author is DaViD Boulet @ Link

Quote:
Originally Posted by DaViD Boulet
....Roger Dressler has confirmed what I've said is accurate in several discussions at AVS.

DN flag (set to anything other than -31) causes the digital filter in the DD decoding engine to then recalculate the audio data (which *is* bit-for-bit accurate from TrueHD to start with) to perform digital-level-reduction based on the value. You no longer have bit-for-bit accuracy to the original data, just like running your audio data through a sampling rate converter or noise-shaping filter would re-write data.

This how it works. Even Dolby doesn't challenge this. The only thing Roger Dressler had to say was that he felt personally that the re-writing of the data didn't cause any serious sonic degredation to the signal (a natural point of view for him to take). But he confirmed 100% that the process I've described above is exactly how DN is applied.
Link

Paul
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post #5 of 5
4/19/08 at 10:36am

Re: Dialog-Normalization & accuracy issues on the HDM lossless codecs?

Thanks, Paul. I thought so.

Until and unless Mr. Boulet supplies a link to this alleged admission by Roger Dressler, I would not take it as gospel. And I would want to know how much (if any) was directly from Dressler and how much was extrapolated from what he said (a phrase such as "just like running your audio data through a sampling rate converter or noise-shaping filter would re-write data" leaves room for all sorts of conceptual slippage).

Adam Barratt and Cees Alons addressed this argument in another thread where it was made about dialnorm and Dolby TrueHD:

Quote:
Originally Posted by Adam Barratt
This isn't rocket science we're talking about. The mathematics are black and white, cut and dried, and extremely basic wherever they occur: attentuation via subtraction. These mathematics aren't a problem for any DSP that can decode Dolby Digital (or any other compressed format). Transparency in this sense implies mathematical precision, and this sort of task can be done with total precision (given that DSPs are mathematical calculating devices I'm sure this is of no surprise to any observers).
http://www.hometheaterforum.com/htf/...ml#post3279880
Quote:
Originally Posted by Cees Alons
Changing the loudness of a digital audio signal doesn't involve determining a new "place" in the sampled signal, or the combining of two adjacent values: just the sampled value at hand is adjusted. All that's involved are individual signal values (more comparable to colour or brightness, video wise). If computations on these are done in the 24-bit domain (like in TrueHD decoders), it could affect the lowest bit(s) indeed. But affecting the lowest bits is hardly significant at all!

In fact, more than the 3 lowest bits of any 24-bit value will take no part in the waveform that will eventually reach the ears of a home HiFi-enthusiast (given a properly scaled signal as well as the noise and harmonic distortion levels of even the best non-professional HiFi-equipment on the market.)

Example: halving the value of the signal involves shifting the whole 24-bit word 1 bit down. The lowest bit will disappear, but that bit-value wasn't to be trusted in the first place! The resultant 23 bits (completed with an upper 0) will still represent all audible data.
http://www.hometheaterforum.com/htf/...ml#post3280031

M.
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