[AndreGB]

Orangeman, thank you for splitting the topics.

David, have you ever considered being a teacher? Thank you for your explanation. Like I said, I've done some basic encodings my own but I have never used any processing (beyond the ones required for lossy encodings) on my sources.

I understand now. Dial norm is basically what ReplayGain does to MP3. Some people were trying to make all MP3 sound the same, using a known "audiophile" base level. The problem is that this base level is sometimes too low, so it attenuates the MP3 too much. That is good, but unfortunately Windows sounds are not attenuated as well (and some Windows sounds are useful). And unfortunately ReplayGain doesn't restore the dynamic range killed by bad mastering.

Now, I do think we should be able to turn off this feature on our decoders. I mean, why force it? Anyway, consider that you would be feeding the DD streams right into your receiver/processor. In this case it does make sense to leave it on on your decoder/processor (you might use it for your DVD, TV, etc). Now, why do studios use it on DVDs? David, have you talked to any other audio engineer and asked him why does he use dial norm on his DVDs? I wish to know whether there is a good answer to that question too.

Now, can I help to guess why dial norm, even though it is a simple operation on the PCM code, makes the sound worse? Because "digital amplification" is not like "analog amplification". I mean, the gain you apply using your receiver's gain is not the same as making the PCM sound louder digitally. Your receiver's gain (the volume, that is) increases the power output of your amplifier to your speakers. "Digital volume" is just a trick to make the sound louder, but it is not capable of actually increasing the electrical energy output to your speakers. And this actual electrial energy (drawn from your power outlet) is the one that makes the sound stronger, bolder, that's the one that increases the sound wave energy to make your room shake. That's how dial norm ruins the experience. Not because it changes the decoded PCM quality, but because it doesn't change the actual electrical energy applied by your amplifiers onto the speakers.

Then I wonder that if you mess with your receiver/processor settings, to change the channels gain a bit, it will make the dial norm effect less worse, won't it?

[EDIT] Poor English

[AlexBC]

Great insights Andre. I'd like to hear what others have to say about it.

The fact that DTHD still has dial norm as default kind of proves my point that things usually aren't exactly what they're advertised as. There's always the fine print.

But if the defeat function becomes standard on all receivers or surround processors (or at least the high-end ones), then I guess all be fine with it.

Quote:

Now, why do studios use it on DVDs? David, have you talked to any other audio engineer and asked him why does he use dial norm on his DVDs? I wish to know whether there is a good answer to that question too.

It would be nice if he has talked to other audio engineers about it. Maybe FilmMixer from AVS can shed some light (he's currently following the 'We Were Soldiers' thread over there, as this is one of his works). But David has stated why he believe's such practice is far too common. I believe that's the reason as well, it's a shame this seems to be true, but too many years on this hobby have shown that many of the so called professionals are not so knowlegable as they should be, or maybe they just don't care much about it. Actually, I think this apply to all areas, where the really good professionals represent the 'great' minority.

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I'm not sure why the practice is so common... I think (and this is what I've been told by several recording engineers) the most common reason is that the folks mastering DD audio don't think twice about the default level setting. I'm sure if the default was 0 (off) we'd see far fewer DD soundtracks using dialog nor. In any case, you're right that the real issue lies with the parties doing the encoding.

[RobertR]

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That's how dial norm ruins the experience. Not because it changes the decoded PCM quality, but because it doesn't change the actual electrical energy applied by your amplifiers onto the speakers.

That makes no sense, Andre. It's the same as claiming that NO digital source can cause ANY amplifier to vary the electrical energy it sends to ANY speaker. That's obviously not true. The output of a D/A converter is a varying voltage, just like ANY analog source, and that voltage is determined by the digital numbers, which are always varying the volume regardless of the presence of dialnorm. The power amplifier stage sees NO difference between the output from an analog source and a digital one. Once things get to the amplifier, there is NO difference between "digital amplification" and "analog amplification". It's a meaningless distinction. Dialnorm doesn't change that process at ALL. And since you said there's no change in the PCM quality, it follows there's no change in the sound quality.

[DaViD Boulet]

RobertR is correct about voltage etc. The way digital volume control (like dialog nor) works is by recalculating the data so the volume change will be heard directly of the D/A converter (voltage).

but this comment is concluding one error from another:

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And since you said there's no change in the PCM quality

Because the LPCM data *does* change. That's how "digital volume control" works... it recalculates each data point with a new value. Any non-zero change value will cause every single bit-word to take on a different value than what it previously had. Aside from the question of how much degradation to the final sound would occur from data-recalculation of this kind, as ChristopherDac explained, dropping just 3db in level constitutes a loss of an entire bit of resolution at the 16-bit level... the native resolution of most DD encodings. Most tracks with dialog-norm seem to drop between 3 and 6 db (1-2 bits at the 16-bit level... dropping to 15 or 14 effective bits of resolution). This doesn't change even if 24 bit filters are used (but it would change if the source data was 20 or 24 bit resolution to begin with).

[RobertR]

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the 16-bit level... the native resolution of most DD encodings

That won't be the case with Dolby TrueHD.

[PeterTHX]

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That won't be the case with Dolby TrueHD

Unfortunately, many studio masters are still 16/48. So any DTHD tracks would be limited likewise.

[AndreGB]

Quote:

Originally Posted by AlexBC

The fact that DTHD still has dial norm as default kind of proves my point that things usually aren't exactly what they're advertised as. There's always the fine print.

Wait Alex, having dial norm is good for the reasons David posted before. Default mean whether it is "turned on" on the encoder then? Well, I hope not. This depends solely on Dolby.

Robert and David, you are right, but probably I didn't make myself so clear, sorry. The DAC itself is not an amplifier, and that's the whole point I was trying to make. So even if you increase (or decrease) the perceived loudness digitally (i.e, before the DAC) you don't change the amount of energy the amplifier is sending to the speakers. I know PCM translates to voltages, but the "number of voltages" a PCM can handle is defined by the amount of bits devoted to each sample (16-bits for CDs, DVDs, etc). So digital amplification has a logical limit. You can make the PCM describe the highest amplitide wave possible, but the energy this wave is going to have in the speaker is limited by the amplifier.

As a good example you can use your PC. If you have any speakers connected to it you can do this "test". Windows has a master volume control. So let's say you set it to maximum. Windows will be recalculating every sound on your computer to be the loudest it can be and sending it to the DAC on your sound board. So you can listen to this sound at a certain volume by changing the volume on your speaker. Now, you can set Window's master volume to 50%. You will have to increase the volume on your speakers to listen to the same sound at the same volume as before.

If digital amplification had any effect on the wave energy then the two situations I described above wouldn't be different. And they are. I can hear them both differently on my THX certified PC speakers (oh yeah - http://www.logitech.com/index.cfm/pr...CONTENTID=9586 ).

I hope it is clearer now.

[AndreGB]

Another post since it is another "topic". Going back to PCM x lossless quality, here is a nice document I found from Meridian.

-> http://www.meridian-audio.com/w_paper/mlp_jap_new.PDF

I believe the whole document is interesting for this discussion, but I recommend you read at least until chapter 4 (MLP Encoder) on page 2. Great read.

[RobertR]

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You can make the PCM describe the highest amplitide wave possible, but the energy this wave is going to have in the speaker is limited by the amplifier.

It's true that the digital data is not the limiting factor (thermal noise, amplifier power, and loudspeaker physics make a dynamic range of 144 dB impossible, which is why all the talk about "you gotta have 24 bit resolution" is silly), but the volume adjustment caused by dialnorm is too small to cause such a limitation in anything after the D/A converter.

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many studio masters are still 16/48. So any DTHD tracks would be limited likewise.

So would LPCM, DTSHD, and any other format, making talk of 24/96 etc. pretty meaningless.

[DaViD Boulet]

Despite the noise-floor of typical electronics, 20 bit and above tends to sound more natural to the ear versus digital audio at the 16-bit level. Whether it has to do with easing the task of d/a conversion or whether the smoother more "sine like" wave produced is perceived to sound more natural is a common topic of discussion among audiophiles. But anyone who's spent any time with SACD or DVD-A knows that even on "mid fi" gear the improvements are noticible for an audiophile listener. I would agree that the improvement from 16/44.1 to 20/48 is probably more noticable than the benefit of 20/48 to 24/96 etc.

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many studio masters are still 16/48. So any DTHD tracks would be limited likewise.

I've heard different reports on this. Some engineers contend that they have many 16 bit masters and other engineers report doing almost all mixing/mastering at 20/48 with only the final downres to 16 for consumer media.

Certainly as time marches on we'll get more and more 20/48 and above-res sources to encode!