mastering audio by Bob Katz

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IIRs wrote:
The Chase wrote: Well Bob Katz said that the digital audio coming from a CD is different than that of a streaming hard drive...Just shows that people talented in a certain feild can sometimes be a bit clueless in the technical aspects of such a feild.
Where does he say that?
In most cases, the final CDR sounds better than the source, as auditioned direct off the hard disk!
http://www.digido.com/modules.php?name= ... cle&sid=15
Please note that I perform all my listening tests at Digital Domain through the same D/A converter, and that converter is preceded by an extremely powerful jitter-reduction device. Surprisingly, I can still hear some variation in source quality, depending on whether I am listening to hard disk, CDR, 20-bit tape, or DAT. The ear is an incredibly powerful "jitter detector"!
Bob Katz is great but at times he's as much as a $9000-wooden-volume-knob-audiophile as possibly imaginable.
Last edited by The Chase on Mon Apr 16, 2007 7:53 am, edited 1 time in total.

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Bob Katz wrote:A properly dithered 16-bit recording can have over 120 dB of dynamic range;
:hihi: :hihi: :hihi: Orly ?

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jupiter8 wrote:
Bob Katz wrote:A properly dithered 16-bit recording can have over 120 dB of dynamic range;
:hihi: :hihi: :hihi: Orly ?
You mean...

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:hihi:
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The Chase wrote:
IIRs wrote:
The Chase wrote: Well Bob Katz said that the digital audio coming from a CD is different than that of a streaming hard drive...Just shows that people talented in a certain feild can sometimes be a bit clueless in the technical aspects of such a feild.
Where does he say that?
In most cases, the final CDR sounds better than the source, as auditioned direct off the hard disk!
You have taken that quote out of context and/or misunderstood it completely!

He is refering specifically to hardware digital systems that use jittery AES/EBU connections for monitoring... if you work "in the box" (as I suspect 99% of KVRers do) then this does not apply to you. *

* The statement might still be true, but for other reasons: if you monitor through an interface with a jittery clock but master the CD entirely ITB you may well find it sounds better on a good CD player than from your HD. :shrug:

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beej wrote: You *can* also clip output channels *even if* your signal *does not reach 0dBFS*, say, you've chucked on a limiter and set it to limit and normalise to -0.3dBFS.

This is because even though the individual sample values are less than 0dBFS, the reconstructed waveform made from those sample values *can* go over the 0dBFS point - while you won't generally hear this as distortion, it isn't desirable and may well be the cause why some people feel their mixes start to sound muddy or dead or cluttered when approaching the 0dBFS point, which doesn't happen when mixing with plenty of headroom. It will also cause your mixes to be rejected from any reputable mastering house if you sent your stuff out to be mastered.
I'm trying to understand this -- once a wave is reconstructed, isn't that placing it in the analog realm where being a little over 0 dBFS is OK?

Edit -- Hmm, does a D-A convertor make its output calcuations in digital or analog? I can see how the signal would clip if the only analog part is the final output.
Last edited by xcomp on Mon Apr 16, 2007 10:59 pm, edited 1 time in total.
"This sentence is true"

(Take that, Epimenides!)

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The Chase wrote:
IIRs wrote:
The Chase wrote: Well Bob Katz said that the digital audio coming from a CD is different than that of a streaming hard drive...Just shows that people talented in a certain feild can sometimes be a bit clueless in the technical aspects of such a feild.
Where does he say that?
In most cases, the final CDR sounds better than the source, as auditioned direct off the hard disk!
http://www.digido.com/modules.php?name= ... cle&sid=15
Please note that I perform all my listening tests at Digital Domain through the same D/A converter, and that converter is preceded by an extremely powerful jitter-reduction device. Surprisingly, I can still hear some variation in source quality, depending on whether I am listening to hard disk, CDR, 20-bit tape, or DAT. The ear is an incredibly powerful "jitter detector"!
Bob Katz is great but at times he's as much as a $9000-wooden-volume-knob-audiophile as possibly imaginable.
:lol: Good one Chase. :-o

Those Bob Katz quotes do have a fishy similarity to the crap on the Vestman site, but unlike Vestman, Bob Katz arguments are actually rooted in real science. They may be silly "golden ear" audiophile type arguments, and I kinda doubt he's actually hearing some of these differences, but at least he seems to have his facts straight about the technology. Bob Katz argument that these things are actually audiblle may be on the fringe in the first place, but then Vestman runs with the concept taking it to a whole new level of pseudoscience that has little to do with reality.

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xcomp wrote:I'm trying to understand this -- once a wave is reconstructed, isn't that placing it in the analog realm where being a little over 0 dBFS is OK?
It means that potentially you are asking the D-to-A convertor to output illegally high values, with unknown results (ie how the specific convertor handles out of range values may well vary - I'm not an expert in that stuff by any means).

Think about it - normally when clipping the main output channel on your DAW, any clipping distortion is happening inside the DAW - over FS samples are clipped to 0dBFS, and you are only ever sending sample values of 0dBFS at the highest to the convertor - the convertor can therefore never clip, as such, because we cannot send sample values of higher than 0dBFS.

But once you add the potential for overloads of the reconstructed wave, you now may be asking the convertor to output +3dBFS. Which it kinda isn't designed to do - nothing analog about it (yes, ultimately it's producing an analog signal, but it still has digital electronics to convert the incoming sample values into an analog waveform, and if those sample values have inter-sample peaks above 0dBFS, the DAC doesn't necessarily know what to do, and thus can potentially output an incorrect wave in those instances.

If you really want to understand what goes on *inside* the DAC, you'll need to do some research - I'm not that hardcore I'm afraid, and don't have the specific knowledge to know exactly how these things behave internally, or whether some convertors can handle it and others cannot.

More specialist audio boards likes ProSoundWeb or Gearslutz would probably be a better place to ask...
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@ beej -- Thanks for the explanation!
"This sentence is true"

(Take that, Epimenides!)

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In mine and many others opinion the best mastering engineer http://www.stevehoffman.tv/
my music: http://www.alexcooperusa.com
"It's hard to be humble, when you're as great as I am." Muhammad Ali

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That book is one of favs. If I got nothing else out of it the idea of mixing at -3db on the master bus has been the most valuable. Doing trip-hop and downtempo it doesn't really doesn't sound that great to have a clipped, distorted, brickwalled song.

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