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[dualmono]

heya all,

this is my first post, and i'm glad i came across this community.
i found the sticky 'DAW & Desks: Is ANYBODY actually still mixing on their desk?' and refering to post #65447, i wonder why i can't recreate Paul Frindle's findings.  

i noticed an improvement in the mix when keeping all the levels in the 'green' area (at least -6db on the channel fader) of SX's mixer, but i was wondering how close i could go to 0db on the master fader when using a brickwall limiter while mastering.
this has probably been beaten to deaf(th) already, but my main question was about Paul test setup.

i use mda TestTone as signal generator and Elemental Audio Equium as the lowpass filter. the host application is Cubase SX 3 on a Mac with 10.3.9 installed.

thanks for your ideas,
dualmono

edit: duh, i forgot to mention what isn't working. well, there is no change in signal level on the output meter, when the sg is giving white noise and the filter is switched off, bypassed or thrown out.

[dualmono]

ok, maybe i should quote Pauls instructions to shed some light, so you don't have to dig through the other thread.

as far as i understood the phenomenon described in his reply can be applied to all ITB mixing procedures?
could it be that the instructions are PT only?  

or did steinberg find a way to prevent this from happening in Cubase SX since 2005?

any ideas are greatly appreciated.

cheers,
dualmono

Quote:

Actually (at the risk of putting the cat amongst the pigeons) I can suggest a simple experiment people can do themselves to illustrate this in action in the most graphic way, which should dispel any lingering doubt that it's important. The aim is to show that what looks like a legal 'signal' way below any red light in your system can still represent something that cannot pass even remotely correctly out of your digital mixer at full level. And also to illustrate how this may affect your sound quality in practice when mixing ITB. It's a kind of worst case scenario - but it illustrates the problem. You need a W/S like ProTools, a signal generator plug-in that has a good filter section that actually goes flat to 20KHz and rolls off at 24dB/oct or so. - In Pro tools get a mono channel up, - stick the PT generator plug-in at the beginning of the channel and set it for sine at say 1-2KHz. - Follow this with a good filter plug-in set for the max slope at 20KHz. (For example the Oxford EQ plug-in has 36dB/oct at 20KHz and illustrates this well - any other good HF filter should work as well). - As an initial test set the channel fader at 0dB and note that the PT meters shows the sinewave signal at -6dBr and that putting the filter plug-in in and out using bypass has no effect. - OK now switch the signal genny to white noise and note that the level on PT is still -6dBr. - Now un-bypass the filter plug-in and watch the signal level rise dramatically!! In the case of the Oxford 36dB/oct filter the meter level will rise a full 5dBr to nearly flat out. Ok so what's happening - how is this possible? Well the digital genny plug-in produces sinewaves correctly - but when in noise setting it is just a random number generator driving the output. So although when set to -6dB peak value no sample ever gets to be greater than 50% modulation - a reconstruction of the undecoded SAMPLE VALUES produces nearly full level SIGNAL. Reconstruction means filtering and so the filter plug-in is acting like a partial reconstruction filter (much like a DAC) - which in turn is now feeding a more legitimate SIGNAL which the sample value only meter can read more correctly. Ok now if this SAMPLE train is passing out of your DAC it too is being reconstructed correctly - so this -6dBr noise from the genny would a produce nearly full modulation SIGNAL if you fed this to the DAC directly - even though no sample gets to be bigger than 50% and no reading say's it's bigger than -6dBr. If your filter is a good one you should be able to switch it in and out and hear no difference in the sound of the signal from your DAC - despite the PT meter reading wildly different. The filter has neither added nor taken anything significant out of the intended audio signal - but you have nearly doubled the sample values within the PT channel! Ok, now wind the genny level up to say -2 or -3dB (still less than only 75% full level) and do the same thing. What happens? Well it now clips when the filter is in (samples bigger than flat out) - now the sound definitely changes when you switch the filter in and out - because it is mathematically limited and in error when the filter is in - cos it cannot pass through TDM slot at the output of the filter!! That is what would be happening in your DAC, it would saturate if you sent this at only 3dB setting on the genny - reading -3dBr within the mixer itself, straight to the output!! Ok now what does this mean for a mix? Well with all those mixed signals, cymbal crashes, HF EQ and limiting etc.. how close do you imagine the output signal can get to being a bit like white noise in places within a real production - even if none of the contributing channels hit the red light? Is this not the exact register of what we term as 'air' and 'resolution'? And people are aiming at max possible mix output levels on meters that do not show SIGNAL. So why does an OTB mixer apparently sound better than an ITB mixer when you are modulating your digits close to 0dBr (sample value) all over the place? Well all those DACs (flawed as they may be) are acting to legitimately reconstruct your programme - before - you mix them all together and produce too many illegal signals that cannot pass out of your digital mixer! Paradoxically, the loss of sound quality due to all those converters is not as bad as the illegal signals created within the digital mixer by the 'too hot' signals you are trying in vain to pass out of the system. It is not a summing issue at all (the one thing digits CAN do is add up almost perfectly). It's an illegal output problem caused by the fact that there are no meters that display actual SIGNAL in your whole mixing environment - you simply never see it happening. So - go back and get your fav test mix back up on your W/S, re-mix the whole thing making sure that at every place in all chains (including between all plug-ins) never gets bigger than -6dBr. Make sure your final output after any limiting etc also never peaks beyond -6dBr. Now do the comparison between this ITB mix and a similar OTB mix. You might have a big surprise

[Tomas Danko]

I call for a stickie on this one, Mr. Frindle permitting.

This is one of the main reasons howcome I run my levels low ITB, and the difference is clearly audible (sounding like harsh trebly distortion unless I'm way lower on the meters).

[dualmono]

are you working with PT as DAW?

though this is a perfectly logical test setup and the effect is explained thoroughly, i am not able to recreate this with Cubase SX3 ITB. how come?

thanks,
dual

[Tomas Danko]

Your noise generator is perhaps not totally random?

[dualmono]

ok, i'll try to find a different noise generator, and report back.

cheers,
dualmono

[johnR]

dualmono wrote on Sun, 22 April 2007 18:36

are you working with PT as DAW? though this is a perfectly logical test setup and the effect is explained thoroughly, i am not able to recreate this with Cubase SX3 ITB. how come? thanks, dual

Could be because PT uses fixed point math and native DAWs use floating point, which has a higher dynamic range for a given number of bits. PT uses more bits (48) to compensate for that, but it will still behave differently from a 32 bit floating point system.

[zmix]

johnR wrote on Sun, 22 April 2007 18:35

Could be because PT uses fixed point math and native DAWs use floating point, which has a higher dynamic range for a given number of bits. PT uses more bits (48) to compensate for that, but it will still behave differently from a 32 bit floating point system.

Yes, exactly right.

In this case, the behavior is the same (the LP filter will average the sample values and produce an output higher in level),
but the difference is that the Floating point DAW will NOT clip.

Score:
Pro Tools: 0
Every Floating Point DAW: 1

I tried this test and discovered that some third party noise generators,
such as the MDA test tone, will not produce this result.
I suspect that this is because these generators already have a LP filter on the output.

[dualmono]

thanks for pointing that out.

i took the noise generator in my native bundle's sonic destructor.

when i set the sine test tone to -6 and adjust the white noise generator to have -6 also, the channels and output meters peak gradually rises 1.5db from where is set it upwards.

anyway it is apparent that the signal shows a rise of 1 db when i switch in the eq.
why is the signal rising? i thought it is supposed to be sinking when switching the eq in?

btw, did i get it correct. the eq slope shouldn't start at 20k but have its lowest point at 20k, right?
just making sure i succeed in jumping the language barrier here.

if you say that the 32 float system would not clip in the given test setup, does that imply it doesn't matter if i drive the channels hot or conservative?
i could imagine, that when using lots of channels and plug-ins are used there still might be an advantage in not letting the channels peak above -6, would you agree?

but even if that would be so, what would i do when i would master on the 32bit float daw? how far would i be able to pull up the volume and still end up with a good signal?

cheers,
dualmono

[Dave Davis]

While I'm all for high-bit rate math, the comparison of "all native daws" vs PT is a real stretch because it assumes all native daws handle floats the same and rely on identical mantissas and overflow behavior.  And this simply is not the case.

The notion of 300+ dB of dynamic range began in the late 90s as a marketing come-on, relying on the same suggestions as the post above.  The problem in comparing float vs. fixed point maths is that the bits/integer values are most definitely NOT equal.  To achieve such high dynamic ranges, one must reduce the precision of large numbers, relative to the alternate system (ie quantization steps between values can get larger as the decimal point slides to the right, while the fixed point systems steps are inherently linear).  Modern floating point DAWs reserve enough decimal places to achieve good sound, but not surprisingly 64 bit systems are more linear across the same amplitude range than 32, and those steps really DO deliver greater performance than fixed point DAWs like PT.

At any rate, while many current native DAWs have greater dynamic range overall than PTHD, not all of them deliver more predictable, linear behavior near the limits.

-d-

[Etch-A-Sketch]

I was under the impression it was because of the way the Oxford does its computations (it accounts for intersample peaks).

What the test was trying to show is what happens AT THE CONVERTER when converting digital to analog.  So regardless of whether your DAW shows the result (32bitfloat vs 48bitfixed) the converters will still exhibit this behavior as the signal is converted to analog.

I vaguely remember Paul saying something to the effect that the test won't work with just any EQ, but it will work with the Sony and several others.  But it's been a while since I read through the whole thread.

[zmix]

These tests will work with any filter set. If the signal generator has a built in reconstruction filter, however, then these tests will show the indicated results.

The first test (white noise and LP filter) was demonstrating that a random signal will produce a higher peak sample value when reconstructed than indicated by the built in peak meters of a DAW.

The second test ( 200hz square wave and 20hz HP filter ) seemed to indicate that a filter can increase the word width value of the signal.

If you have processing downstream or an output converter that may clip, be aware.

[kappi]

I can approve the test on a PT-m-powered setup, and thats 32bit floating too.