• SONAR
  • What algorithms up-/down-/re-sampling in Sonar? (p.3)
2015/03/29 23:39:04
lfm
drewfx1
 
Yes. In theory, a DAC should allow for a few dB of headroom above 0dBFS to allow for intersample peaks. In practice that might or might not be the case. 




Thanks for your input.
I read an article by some that promote doing all mixing at -20dB peak levels. They have arguments like that - DAC seldom sound good at full 0dBFs. Noise floor is so low anyway - so you would not loose anything doing that - just benefits.
 
Haven't really checked this, but there might be something to it.
2015/03/30 10:37:14
Anderton
lfm
Let's assume one track is inaudible to hear differences.
Import 20 tracks and listen to sum of these on master - then there might be even disturbing artifacts as a total. And make it 40-50 tracks and it's even more obvious.
 
I read an article on preamps, why studios spend $3000 on a preamp when most of us feel that $1000 preamps sound as good as anything.
But this tiny extra bit of clarity on each recorded track makes quite a difference when coming to the total mix of it all.



I agree with this thinking 100%. I first noticed it with hiss and acoustic projects. Preamp hiss would not be audible with individual tracks, but add together 24 of them, and you could hear the difference. If I applied noise reduction to the tracks, each track went from "inaudible hiss" to "even more inaudible hiss" but the final mix was like removing a layer of dust from a painting.
 
This also happens with amp sims and layering guitars. The resonances that are "baked" into the sims become additive and the more guitars you add, the worse it sounds. I worked hard to avoid this effect in the sims I did for Cakewalk. Hopefully you can layer them without fear 
 
Part of the reason for debates about SRC is that before the days of 64-bit calculations, there were audible differences among SRC algorithms. (In fact the 48 kHz sampling rate was chosen for DAT because the record industry felt conversion between 48 and 44.1 kHz was sufficiently difficult that it would discourage digital copying.) This is one reason why those who first started experimenting with higher sample rates chose 88.2 kHz because the conversion to 44.1 kHz was simpler. However these days, it is possible to do the math extremely precisely when converting from one sample rate to another. 
 
So saying "I won't argue with 'sounds exactly the same' - but maybe consider that's theory, not real life" has validity. However, thanks to improved calculation engines and algorithms, real life is becoming if not identical to theory, then really really really really close.
 
2015/03/30 13:20:06
drewfx1
AndertonI agree with this thinking 100%. I first noticed it with hiss and acoustic projects. Preamp hiss would not be audible with individual tracks, but add together 24 of them, and you could hear the difference. If I applied noise reduction to the tracks, each track went from "inaudible hiss" to "even more inaudible hiss" but the final mix was like removing a layer of dust from a painting.

 
You would hear the same difference just by turning up the volume of an individual track. Adding tracks together cannot reduce signal to noise ratio unless there is more cancellation between the actual signals on each track than the noise. In the real world, the reverse is arguably more likely.
 

This also happens with amp sims and layering guitars. The resonances that are "baked" into the sims become additive and the more guitars you add, the worse it sounds. I worked hard to avoid this effect in the sims I did for Cakewalk. Hopefully you can layer them without fear 

 
If by "resonances" you are talking about frequency response, frequency response cannot change by adding things together. It can, however become much more apparent after adding tracks together because they cover a wider range of frequencies.
 
 
This is one reason why those who first started experimenting with higher sample rates chose 88.2 kHz because the conversion to 44.1 kHz was simpler. However these days, it is possible to do the math extremely precisely when converting from one sample rate to another. 

 
Agreed, but it wasn't just precision - available CPU power to do the calculations was a huge issue compared to recent years.
 
 
So saying "I won't argue with 'sounds exactly the same' - but maybe consider that's theory, not real life" has validity. However, thanks to improved calculation engines and algorithms, real life is becoming if not identical to theory, then really really really really close.




In digital audio, anywhere where you're between the converters it's all just math. There can of course be bugs, HW problems and people can always mess up the theory or leave stuff out. But when applied to digital audio, well, let's just say those "theory vs. real life" arguments sound nice in theory. 
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