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  #21  
Old 11-17-2008, 01:23 PM
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Default Re: Best Recording Levels

If your analog chain has an noise floor of -80 dB, you could be working at 16 bit without any problem (an analog cassete deck has a noise floor of -60 dB) My analog chain has a noise floor around -120dB, so recording at high levels at 24 bit has it's advantages... Btw, I did mean recording at -20 dBu, that is recording at 16 bit in a 24 bit resolution.
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  #22  
Old 11-17-2008, 01:32 PM
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Default Re: Best Recording Levels

Quote:
Originally Posted by Delta Music Belgium View Post
My analog chain has a noise floor around -120dB, so recording at high levels at 24 bit has it's advantages... Btw, I did mean recording at -20 dBu, that is recording at 16 bit in a 24 bit resolution.
Then you are the only person in existence to have such an ANALOG system... As the best analog systems I've worked on, seen or even read about are usually between 90dB and 100dB dynamic range.
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  #23  
Old 11-17-2008, 01:41 PM
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Default Re: Best Recording Levels

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Originally Posted by Delta Music Belgium View Post
Please read some basic material about digital recordings. If you record at 24 bit at a level of -20 dBFS, it is the same as recording at 16 bit and losing the resolution of the highest 8 bits.
I hate to break it to you, but I think YOu need to do some reading about basic sampling concepts. Because, if you chop the top 8 bits off of ANYTHING you lose 48.16 dB, not 20dB. I think THREE bits is more what you were going after...

But again...you aren't losing anything except analog noise by recording lower. If you're mic preamp has a dynamic range of 80dB. Then recording as hot as possible puts the noise at -80dB. What happens when you dither to 16bit for your final CD? Without ANY limiting or compression, your analog noise will REMAIN in the 16bit master... Why? 16bit files have a dynamic range of 96dB... When dithering from 24bit the bottom one or two bits will contain noise... so the noise floor of the 16bit system is at 84dB... while your analog noise sits above that at -80dB...

Basically you are making your end product noiser by recording louder. There is no gain in resolution, the only thing you are doing is limiting your headroom AND again, running your equipment out of spec.

In the end all music is VOLTAGE... and that's where we started this whole debate.
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  #24  
Old 11-17-2008, 02:28 PM
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Default Re: Best Recording Levels

A CD at 16 bit has a theoretical maximum resolution of 96 dB (all bits at 1), a SACD at 24 bit has a theoretical maximum resolution of 144 dB (all bits at 1). So one bit is 6 dB. In voltage both have the same output level, but SACD has a 48 dB more resolution, that is what I try to explain...

Btw, my analog input stage has a theoretical -120 dB resolution (Focusrite Red 1's + 192 I/O), not my analog monitoring output stage, that is indeed around 100 dB...
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  #25  
Old 11-17-2008, 06:22 PM
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Default Re: Best Recording Levels

hello,

"Analysis of the quantization error of low-amplitude signals reveals that the spectrum is a function of the input signal. The error is not noiselike (as with high-amplitude signals); it is correlated. At the system output, when the quantized sample values reconstruct the analog waveform, the in-band components of the error are contained in the output signal. Because quantization error is a function of the original signal, it cannot be described as noise; rather, it must be classified as distortion.

As noted, when quantization error is random from sample to sample, the rms quantization error E (sub)rms = Q(12) (sup)1/2. This equation demonstrates that the magnitude of the error is independent of the amplitude of the input signal, but depends on the size of the quantization interval; the greater the number of intervals, the lower the distortion. However, the relevant number of intervals is not only the number of intervals in the quantizer, but also the number intervals used to quantize a particular level. A maximum peak-to-peak signal (as used in the preceding analysis) presents the best case scenario because all the quantization intervals are exercised. However, as signal level decreases, fewer and fewer levels are exercised as shown in Fig. 2.8. For example, given a 15-bit quantizer, a half-amplitude signal would be mapped into half of the intervals. Instead of 65,536 levels, it would se 32,768 intervals. In other words, it would be quantized with 15-bit resolution.

The problem increases as the signal level decreases. A very low-level signal, for example, might receive only single-bit quantization or might not be quantized at all. In other words, as the signal level decreases, the percentage of distortion increases. Although the distortion percentage might be extremely small with a high level, ) 0 dBFS, its percentage increases significantly at low-amplitude levels.

The error floor of a digital audio system differs from the noise floor of an analog system, because in a digital system the error is a function of the signal. the nature of quantization error varies with the amplitude and nature of the audio signal. For broadband, high amplitude input signals the quantization error is perceived similarly to white noise." [Pohlman, "Principle of Digital Audio"]

digilom
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  #26  
Old 11-17-2008, 06:57 PM
digilogin digilogin is offline
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Default Re: Best Recording Levels

Quote:
Originally Posted by O.G. Killa View Post

If you are recording into your DAW trying to get your peaks as close to 0dBFS as possible. That is the EXACT same thing as recording to an analog tape and trying to get your levels to tape up to +20dBVU.
Quote:
Originally Posted by digilogin View Post
hello,

that is incorrect. you do not get digital distortion per se under 0dBfs.


digilom
Quote:
Originally Posted by O.G. Killa View Post
Excuse me? Your comment doesn't in any way relate to what I was talking about... go back and reread my posts, and if you need me to, I'll explain after that... I'm not even talking about digital distortion, I don't know where you got that from?!?
hello,

you made [and appear to still be trying to make] a sweeping, overbroad statement that is erroneous. e.g. if you hit tape at +20 you will get a squashed, distorted recording, but an adc will not do that at 0dBFS. making a digital recording at 0dBFS is not at all like hitting tape at +20dBVU.

you misspoke.

perhaps you intended to limit the scope of your statement to apply only to some pre-defined, or previously specified, points of contention, but you did not do so.

taken on its face, and reasonably interpreted so as to be given the customary meaning found in ordinary usage, your statement is erroneous.



Quote:
Originally Posted by ogkila
This is a common misconception about sampling... it's completely urban legend. There is no more "accuracy" recording hot than recording quiet.

see "principles of digital audio" [excerpt above].


digilom
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  #27  
Old 11-18-2008, 02:57 AM
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Delta Music Belgium Delta Music Belgium is offline
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Default Re: Best Recording Levels

+1! I could not have said it better!
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Delta Music Belgium - Pro Tools | HDX Studio
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Mac Pro (Mid 2012) 3.33 GHz 6-core - 32 GB RAM - OS X 10.14.6 - Magma EB7 - PT HDX3 - PT Ultimate 2019.6 - HD I/O 16x16 - 192 I/O - 192D I/O (x3) - SYNC HD I/O - Apogee Big Ben - Rosetta 200 - Slate RAVEN MTi2 (x2) in a RAVEN CORE Station + 2 RAVEN CORE Station Sidecars - Almost all available AAX-DSP plug-ins
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  #28  
Old 11-18-2008, 10:53 AM
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Default Re: Best Recording Levels

Quote:
Originally Posted by digilogin View Post
hello,

"Analysis of the quantization error of low-amplitude signals reveals that the spectrum is a function of the input signal. The error is not noiselike (as with high-amplitude signals); it is correlated. At the system output, when the quantized sample values reconstruct the analog waveform, the in-band components of the error are contained in the output signal. Because quantization error is a function of the original signal, it cannot be described as noise; rather, it must be classified as distortion.

As noted, when quantization error is random from sample to sample, the rms quantization error E (sub)rms = Q(12) (sup)1/2. This equation demonstrates that the magnitude of the error is independent of the amplitude of the input signal, but depends on the size of the quantization interval; the greater the number of intervals, the lower the distortion. However, the relevant number of intervals is not only the number of intervals in the quantizer, but also the number intervals used to quantize a particular level. A maximum peak-to-peak signal (as used in the preceding analysis) presents the best case scenario because all the quantization intervals are exercised. However, as signal level decreases, fewer and fewer levels are exercised as shown in Fig. 2.8. For example, given a 15-bit quantizer, a half-amplitude signal would be mapped into half of the intervals. Instead of 65,536 levels, it would se 32,768 intervals. In other words, it would be quantized with 15-bit resolution.

The problem increases as the signal level decreases. A very low-level signal, for example, might receive only single-bit quantization or might not be quantized at all. In other words, as the signal level decreases, the percentage of distortion increases. Although the distortion percentage might be extremely small with a high level, ) 0 dBFS, its percentage increases significantly at low-amplitude levels.

The error floor of a digital audio system differs from the noise floor of an analog system, because in a digital system the error is a function of the signal. the nature of quantization error varies with the amplitude and nature of the audio signal. For broadband, high amplitude input signals the quantization error is perceived similarly to white noise." [Pohlman, "Principle of Digital Audio"]

digilom
PLEASE LOOK UP DITHER!!!! This is what happens when you read things out of context... you never get the whole picture.

At lower levels the quantizing error becomes dependent of the input signal, resulting in distortion. This distortion is created after the anti-aliasing filter, and if these distortions are above 1/2 the sample rate they will alias back into the audio band. In order to make the quantizing error independent of the input signal, noise with an amplitude of 1 quantization step is added to the signal. This slightly reduces signal to noise ratio, but completely eliminates the distortion. It is known as dither.

What is happening is you are looking at ONE little piece of the puzzle and not the whole picture. If we never dithered, then yes we would have quantization error.

Also, I should point out that the quantization error noise in a 24bit signal is down around -144dB!!!!!! The ANALOG noise of the electrical circuits WITHIN a ADC bring the noise floor up to -120dB and essentially eliminates the quantization error.

Also, Pohlman's example is great IF YOU ARE ONLY DEALING WITH A SINGLE, SIMPLE SINE WAVE. With complex waveforms his point is completely out the window because you never have a CONSTANT amplitude!!! With a complex waveform your "15-bit" signal is only 15-bits for a fraction of a second, then it might be 12 bits, then 20bits, then 11bits...and so on...

Secondly... the reconstruction filter and the oversampling/averaging at the DAC also help to eliminate any quantization error that may have occurred during sampling... Did you just forget that or did you leave that part out on purpose? Or have you not read that far in the "Principles of Digital Audio" book?

Last edited by O.G. Killa; 11-18-2008 at 11:19 AM. Reason: grammer
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  #29  
Old 11-18-2008, 11:16 AM
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Default Re: Best Recording Levels

Quote:
Originally Posted by digilogin View Post
hello,

you made [and appear to still be trying to make] a sweeping, overbroad statement that is erroneous. e.g. if you hit tape at +20 you will get a squashed, distorted recording, but an adc will not do that at 0dBFS. making a digital recording at 0dBFS is not at all like hitting tape at +20dBVU.

you misspoke.

perhaps you intended to limit the scope of your statement to apply only to some pre-defined, or previously specified, points of contention, but you did not do so.

taken on its face, and reasonably interpreted so as to be given the customary meaning found in ordinary usage, your statement is erroneous.






see "principles of digital audio" [excerpt above].


digilom
I did not misspeak. Nor am I making a broad sweeping statement that is untrue. YOU, on the other hand though are just skimming posts and not really reading the whole thing... and so YOU are taking things out of context, similar to what you did with the Pohlman book...

I suggest you try recording at +20VU on a tape machine... it won't clip. Most tape machines are designed to handle up to +22~+26dB VU before clipping...

But here is my point... Since you choose not to read my previous posts...

ALL SOUND GETS CONVERTED TO ELECTRICITY IN THE ANALOG DOMAIN!!! Is that not correct? And at some point your signal from your DAW will be converted to electricity before it is converted into sound pressure, correct? Now unless you have a completely digital studio and are using the Neumann Digital Microphones, you are going to have your audio flowing as electricity into and out of your DAW.

So I ask you... if you were to turn on a test tone Generator, and turn that generator up until it was hitting +20 VU on your analog tape machine or your analog console, WHAT RMS VOLTAGE WOULD THAT BE?

Until you can answer that question, you are basically talking oranges while I'm talking apples.

Try it for yourself... Use a voltmeter, test the output and see what the voltage is... It will be around 9 or 10 Volts (RMS).

Now... using your DAW, Turn up the signal generator until your ouptut measures 9 or 10 VOLTS RMS. What is the level of the signal inside the DAW? I guarantee it will be 0dBFS.

In the end, VOLTAGE is what is important. +4dBu is 1.228 Volts (RMS). Most professional audio gear is calibrated to +4dBu. Your VU meters on your Console and on your analog tape machines are calibrated to 0 VU = +4dBu. The circuitry within ALL of your professional audio gear is designed around 1.228 volts as the optimal AVERAGE level.

All the specifications for the gear you use and all the numbers people spout off, only pertain to the unit if the signal is around 1.228 Volts. The farther away from that you get, the farther "OUT OF SPEC" the unit performs. Linearity goes out the window.

So you can sit here and try to argue with me that +20 VU in the analog world is not the same as 0dBFS and that you read a book on Digital Audio once so you are the expert... But in the end, we are talking about electricity here, not digital audio, or quantization error, or dither.

WHO THE HECK CARES IF YOU ARE USING ALL 24BITS OF SIGNAL IF YOUR ANALOG CHAIN IS DESTROYING THAT SIGNAL BEFORE AND AFTER IT IS IN YOUR DAW?!?!?!?!?!?!?!!?!?!?!?!? Not to mention what your plugins are doing to that full scale signal and what your DAC is doing to it. Try flipping ahead in the Pohlman book to "INTERSAMPLE PEAKS during Digital to Analog reconstruction" and see what he says about those? Does the book even mention them? Maybe you should try to find out more information on the subject?

The problem is, you've only learned enough so far to THINK you know what you are talking about... But there are SO MANY other factors involved in recording than just sampling theory... You just haven't read those books yet I guess...
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  #30  
Old 11-18-2008, 05:01 PM
digilogin digilogin is offline
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Default Re: Best Recording Levels

hello,

nice try. well, actually it was kind of juvenile. you are defensive and you are missing the point.

digilom
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