Fader Math test please sign in software users

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lol, that's what I was thinking, but at 2:00 a.m. my mind is saying "don't even fucking think about it!" :D

32 bit float should be able to retain much of the information, but a fixed precision is probably going to lose enough information on a complex waveform to cause rounding errors.

We do need to go through a mathematical example, but think about it this way: if you start with 24 bits and reduce the gain to the point where you only have 16 bits, then your waveform has to fit in those sixteen bits. You throw away 8 bits worth of information. When you then scale back up to 24 bits of dynamic range, I don't see how you could expect to reclaim the information that was lost. Sure, some rounding will come back to the right value, but I don't see how all of it could. In fact, such a drastic scaling would likely have errors "everywhere."
 
ok my theory might be flawed but lets say we had a mix buss with 24 bits and 8 bits after the decimal point, so 32 bits total...I realize I am already in error but just theorizing

lets say we got a single 24 bit sample, reading 123456789101112131415161 and we want to turn it down by 6 dB, so
123456789101112131415161 x 0.5

uh oh using the windows calculator hope its got enough bits :)

= 61728394550556065707580.5

that comes out correct easily as the eight extra bits give us up to 256 separate values after the decimal point ( boy thats not that much!! I can see where this could go wrong!) at least, .5 fits ( it does right?)

turn it back up 6 dB

61728394550556065707580.5 X 2
=123456789101112131415161

no problem

how about 333333333333333333333333 down 6.1 dB?

someone figure that one :) and put it back up by 6 db and see if it works
 
Uh-oh. Here comes the 'M' word.
and those little voices
"Look away Dorothy! Don't go near the light!
Stay away from the light!"
:eek:
Ok. I'm a wussy
 
Yeah pipeline, I did some math in a dream last night. :)

I think a signal recorded at 16 or 24 bit fixed resolution, and then converted to 32 bit float should have no problem. Since all the decimal places are zero's to begin with there's quite a bit of resolution for any scaling down and back up again. If you were to add 32 or 64 bit reverb though (for example) to the sample and scale down and back up, you would probably lose a tiny bit of information due to the zero's containing real information as a result of applying the reverb. However, as soon as you convert back to say a 24 bit fixed format the original and scaled up/down versions should still round to the same integer values, since all of the scaling error should be well past the decimal point.

The story is different though if your hardware/software only allows using fixed (integer) bit depths. Any fader here that permanently alters the sample will likely throw away information.

At least, that's what the goat in my dream told me. :D
 
Just tested a 10sec. white noise sample in Logic Audio Platinum 4.7.2. The first time around I used the Gainer Plugin that comes with Logic in Inv mode. Result: Noise@-55.2 dB.
Then I inverted the bounced file in Logic's Sample editor and didn't use the Plugin (thought it could be the plugin's fault). Result: Noise @-55.2 dB. Same thing.

Mind you, I used a quite old version. As far as I know, the newer 5.x versions all have an internal 32bit floating point architecture, so I suppose that problem might be solved in the newer releases.
 
"if you start with 24 bits and reduce the gain to the point where you only have 16 bits, then your waveform has to fit in those sixteen bits. You throw away 8 bits worth of information. When you then scale back up to 24 bits of dynamic range, I don't see how you could expect to reclaim the information that was lost"

If I understand the test properly, the information won't be lost if the software is using enough precision on the various busses, ie some kind of floating point number. With a 32 bit float you have 25 bits of precision. I'm not sure of all the exact details of float storage, but you should be able to scale a 24 bit wave down then back up without the waveform being compromised.
 
I would be really interested in the result of a newer Logic release. If someone could test and post the result, I would be very grateful. I mean, not THAT grateful, but I would appreciate it.
 
FYI it should cancel out in all the apps that work at 32 bit floating point precision internally. i forgot the reason why but if u want a complete mathematical explanation visit www.prosoundweb.com and go to recpit and visit NIKA's forum, he knows all about that stuff

The reason PT TDM fucks shit up is because with each change in audio it processes it at 48 bit internally then dithers back to 24..and like someone else said dither is random...
 
JuSumPilgrim said:
In order to make sure everything is lined up in CEP, right click on the wave block and go to wave properties and check the offsets. Just put the same offsets on both waves and theyll be lined up. Its pretty straightforward. Make sure your routing is exactly the same for both waves.

...I did that Pilgrim. :) Just like you mentioned. It still brings me those -55dB noises at the result. But hey.. glad to know I'm not the only one here living with those -55dB noises at the end... :D see... sumthin' wrong here. Hey Teacher... NICE LINK !!!
;)
James
 
Yes, after thinking about it 32 bit float should be capable of total phase cancellation unless you scale from something like 24 bits to 12 bits and back, or use an extreme amount of processing (I'm guessing it would take ridiculous amounts before the quantization error would approach the integer value range).

If you're just using 24 bit fixed processing though, this could be a realistic issue.
 
I rendered all wave forms in Magix Samplitude v6.0 Total cancellation.

I decided to take the test a bit further.

With multiple eq settings
With generated white noise, I applied +6db 1Khz and -6db at 5Khz, bounced that track and then applied -6db at 1khz and +6db 5Khz, then inverted. I did get a little noise but it was very low, which can probably be attributed to the parametric aspects of the equalizer. But for my test purposes it seemed the software was extremely accurate.

I also did single fader eq settings (-6db to +6db settings at 1khz) on the white noise with almost imperceptible results, I had to zoom in on the wave about 10 times to even see anything and it was definitely not audible <-60db.

But overall I think Sampitude did a excellent job of the single fader test.

Thanks pipeline for the suggested experiments:)
 
gatorhaus, you outdid yourself! Have you tried sam 7? some neat new stuff...the plugin delay compensation doesnt always work so hot, I hope they are working on it
 
gatorhaus, you outdid yourself! Have you tried sam 7? some neat new stuff...the plugin delay compensation doesnt always work so hot, I hope they are working on it

No not yet,

we just got 6.0 about 6 months ago and don't think I'm ready to shell out the bucks for the upgrade just yet, still trying to finish construction.

But I think I will like the direct VST support though, that will be a big plus.

larry
 
Vegas four: passed

did it using the track volume to turn it down then the buss volume to turn it back up...gonna see how it is with the fx sends
 
Although it's probably too late now, I would like to know if those who got a 'flat line' also tried playing it. Some of my test looked flat but still registered with the meters at maximum scale.

Am I correct in assuming that at this point, we think a total cancel would be good but if it doesn't, we still don't know what it means? I'm particularly interested in light of my 'residue' on music at -70 not being distorted.
Wayne
 
:confused:
...I tried again in other machine with CEP 2. using Delta 66. Result : Total phase cancellation. Shit, now I think it's my machine... Hey... I'll never know if my P!!! 866 machine was so stupid without this test... I found it slept durring the summer class. I'll send this stupid bastard to math class... :D REINSTALL ALL THE APPS... !!! :rolleyes:
 
I just did the test with n-track at 24 bit. I took a wave file of white, pink, and some other test tones. Turned it down 6 db on the fader and back up with the master channel gain, rendered the file and etc. when I rendered the two tracks and scaned it I had -137 db on the wave file. I guess this would be considered total cancelation. Does this mean I can turn down the channel faders without introducing noise into my overall mix?

Larrye
 
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