Final words about increase in bit depth/sample rate?

skippy said:
Atomictoyz wrote: ...Blah Blah...
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While at work on a Monday that started bad and got worse. Im left handed and dyslexic, so ignore everything I typed that day... Heh heh...no excuses though... I have no idea what Im talking about.

Also Originally posted by skippy
Regrettably, this is false.......
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Reaallly!.........no regrets? I regret posting it....When I read your response Skippy I didn't recognize what I had typed as my own...This BBS needs some clutter nulling DSP...Im sure sonusman would approve :0)


Peace,
Dennis
 
No, don't regret it: none of us was born knowing this stuff, and it is a hard topic- usually clothed in nerdy verbiage like the stuff I spewed above. But if we all play this right, we can all learn something new: this is a developing field, and the stuff I learned in the 70's and '80s has been expanded on, and in some cases obsoleted outright.

Remember when I first showed up here about a year and a half ago, talking about A/D conversion? Hell, I used to design DSP signal paths for audio *for a living* back in the mid-80s, and thought I had a very good handle on it. But then I got completely out of the pro audio industry for 10 years. And while I was asleep, the industry moved forwards, and the introduction of cost-effective oversampling/digital filtering converters changed the nature of the conversion process is a very fundamental way (with the demise of brickwall antialiasing filters, for example). So all my cherished old hard-won knowledge, while not exactly _wrong_, was instantly revealed to be just a historic curiosity. Shoot, we don't _do_ stuff that way anymore. Coulda knocked me over with a feather. I'm still amazed every time I walk into the studio.

That's why I said "regrettably". I don't want to attack the messenger, because *I've* been wrong before, and I will be again, and getting attacked ain't fun. I do want to attack the _concepts_ that are wrong, though, so we can all improve the breed...

No sweat! I'm looking forward to the collaboration that Ed proposed: a "readable-by-the-non-nerd explanation" would be a true service to the industry.
 
It's the processing stupid!

Simply mixing a bunch of 16 bit tracks down in 24 bit does have some minor signal to noise ratio benefit, but processing is where the true power lays.

Here is a VERY SIMPLE example to illustrate the point:

Take a 16 bit recording of a hand clap peaking at 0dB and apply a simple delay echo to it. Each echo has worse and worse signal to noise ratio as the volume drops off. The echo decays to about -92dB until it gets completely washed out in the noise floor.

Now take that same 16 bit recording and convert it to 24 bit, then apply the delay. The echo drops off -48dB before it sees any degradation in signal to noise (its original 92dB) then finally gets lost in the noise at about -140 dB.

Now if that's not a good example of the power of extra bits, I don't know what is.

In a 24 bit mix of 16 bit tracks there is obviously always noise down around -92dB. But other signals like delay and reverb tails don't get lost just because they are mixing with noise. If that were true you wouldn't be able to understand your wife/girlfriend saying "Turn that off!" when the TV cable goes out. Just by adding the very basics, delay or reverb, you dramatically increase the signal to noise ratio of your final mix.

Hope this helps. :)


barefoot
 
The attached image shows a graphical representation of what a delay or reverb tail from a 16 bit track does in a 16 bit mix versus a 24 bit mix.

barefoot

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Alright, 'nough words and pictures. Let's hear sum!

The noise floor of 8bit is clearly audible, so it makes for a good listening example of the point I tried to make earlier. And 16bit has 256 times the resolution of 8bit, just as 24bit has 256 time the resolution of 16bit. So, 16bit is to 8bit as 24bit is to 16 bit.
Make sense?

I took an 8bit hand clap file and added delay. Here is the result:

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Now I took the same 8bit file, converted it 16bit first, then added the same delay processing. Any questions?

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Finally, here is the processed 16bit file dithered back down to 8bit.

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There wasn't much difference between the 8bit or 16bit but the 8 bit did seem to trail off faster at the 3rd echo. The dithered file noticably had less quality with a definite hiss throughout. Does dithering always add this hiss? Is it less noticable dithering from 24bit to 16bit?
 
NYMorningstar said:
There wasn't much difference between the 8bit or 16bit but the 8 bit did seem to trail off faster at the 3rd echo. The dithered file noticably had less quality with a definite hiss throughout. Does dithering always add this hiss? Is it less noticable dithering from 24bit to 16bit?
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NYMorningstar,
You must be listening through the cheapest computer speakers known to man. :eek:

barefoot
 
Not the cheapest, they musta cost at least 10 pesos. Sorry using just cheap puter speakers but it supports what he is saying, even on cheap speakers you can hear the difference. I guess it's a good idea to keep in mind most music is listened to on crap speakers anyway.
 
NYMorningstar,

Have a listen with some descent speakers or headphones. I'm sure you'll hear a world of difference between the 8bit and 16bit versions.

Dithering does add noise, but it also helps retain some of the extra low level detail from the 16bit mix. This is easily recognizable in the dithered version. 8bit is an extreme example, so the dithering noise is very noticeable. Dithering noise going from 24bit to 16bit is nowhere near this apparent.

The important thing to remember is that moving to a higher bit depth doesn't improve the quality of the original sound. It can't do that. But, it does cause far less degradation to the waveform when it's processed, so the final processed sound winds up being higher quality.

barefoot
 
Good point barefoot about higher bit depths not "improving the original sound". I suspect that people think that is the case sometimes...:eek:

Higher bit depth just means that any "hash" from quantiniztion errors that WILL happen from even a .1dB of boost/cut in level from the original data.

Think of it this way. The data is "stacked" so to speak. Digital only reflects a % of all volumes that are possible, so there has to be known volume levels to store the digital data. Digital is that way. I wants to KNOW what to do with something. It is not a living breathing thing! :)

So, when you record, the volume steps are defined. Whether the converter did it well or not, with dithering or not, what you have is what you have. Cool.

The problem starts with boosting/cutting levels in ANY way. EQ tweeks are a form of level adjustment because in fact you ARE boosting/cutting the overall volume of a recorded track when you apply eq to it. Makes sense right? Cool. But with a digital audio, this means that when you create the "new" digital audio, which is "post DSP", how the volume steps needs to be redone. Now your audio is subject to coding of the software application. It may or may not do such a hot job of created new "good" data. But it is what it is. The real kicker though is that the resulting "post DSP" data will not fit neatly into the fixed values that can be stored for the bit resolution you are working at, thus, you will be creating MORE quantinization errors from the recalibration of the levels. This adversely effects the last bit or two or three, while it usually isn't too noticeable at the most significant bits. So, your low level information starts to become corrupt. Okay, most realize this is the case.

What higher bit resolution gives you is the opportunity to move those quantinization errors down to bit levels that our hears are not sensitive enough to hear. Our hearing range has a rough "bit equivelent" of about 21 bits. So, if quantinization errors are only obvious to the least 2 or 3 significant bits, then those errors would be low enough so that our ears don't HEAR them, thus, DSP free of Q errors!!!! Very cool!!!

Now, how dithering comes into play is that dithering adds a bit of noise at a certain level. Waaaaaaaaaaaaaaaaaay down there in volume. It applies it to the least significant bit. It keeps that least significant bit to ON! Thus, a few more bits of data is possible to store to 16 bit, because the audio data is COMBINED with the dithering data. So even though the audio is lower in volume than the dithering noise, the dithering noise is SHAPED NOISE and you can hear or sounds that are lower in volume than it, because the frequency that shaped noise is at is out of the ears ultra sensitivity range ( around 2-7KHz....) So, lower level stuff STILL retains dynamics because it CAN, because it doesn't require bit depth to represent values, because the lowest bit is always on and there is some noise there to blend with it. Sound weird, but that is what happens, and it WORKS.

Ed
 
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