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My Theory About Prosumer Audio Audio Interfaces
Old 23rd June 2006
  #121
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Quote:
As far as have I don't A/B test, none formally that I can remember. Have I had an opportunity to hear tracks recorded very low and hear them not sound as good? Yes, but it's been years since they were tracks that I cut that way.
I give up. No point in going on with this discussion.

If you have no interest in actually trying it to see if if makes any or no difference with this newer class of modern converters and the "resolution" differences between the tracking and mixing methods then this entire conversation is pointless.

Uncle.

Theory (about something that may have taken place "years ago") disproves and makes irrelevant real world experiences ("what happens to many every day, now") everytime on the Internet.

I'd actually be shocked if it was any other way. heh

Live long and prosper.

Lawrence
Old 23rd June 2006
  #122
The Distressor's "daddy"
 
Dave Derr's Avatar
 

A Little Protools Research

Ok, I did a little research and asking around from Protools Plug-in writers and I found something that could possibly explain some of the intermittent complaints about full scale recorded signals.

It seems that Protools LE uses floating point math, and Protools HD does not. This means that In-The-Box (ITB) summing/processing of full scale tracks in Protools LE doesnt have much chance of clipping, while Protools HD does! Screwed up, hunh? (I haven't done any research on other DAWS.)

The low end version of Protools (LE) can more gracefully employ processing & summing than the High End Protools version! Most HD plugins do accommodate full scale signals by having an input "trim" on them... but not always. From what I have gathered, a full scale signal in protools HD can clip a plug in that boosts the signal in any way, UNLESS YOU TRIM THE INPUT TO THE PLUG IN. Some plug in developers "pad" the input right off the bat so you have some headroom. The protools HD plugs are placed pre-fader (pre mix attenuation) which means even with a fader pulled down, a plug that boosts (such as an EQ) can clip ITB without the user knowing about it. Usually plugs have a clip indicator which should tip the user off... but not always.

This does not affect our discussion on getting the best quality from your ADC converters by using full scale... this is almost a seperate issue. You will still get the best resolution and achieve the specified performance of an ADC when you let the peaks go close to 0dBFS. This again is by definition, and by the uncontrovertable nature of digital audio. But if when mixing or processing ITB you aren't notified of clipping (thru metering) that occurs because an HD plug-in or mix buss "over-flows", this high res, full scale digitized track can become a nasty distorted ear-ripper.

I tend to trust peoples ears and comments, especially when they are corroborated by others. Even though they often don't make a bit of sense at first, one must always take them seriously and keep an eye out for possible explanations for them. This could possibly be one of several explanations why people have had problems with full scale digitized signals. Now... Im having thoughts about the ADC digital filters handling full scale transient edges again. If I have time, Im going to test some ADC's SOOOOON! At Eventide, I saw early digital filters clip overshoot on square waves, yet leave sine waves of the same peak to peak levels totally clean. NOT GOOD!

So with Protools, the LE version handles full scale tracks being mixed or processed much better because it uses floating point math processing. Floating point is a method which uses bipolar math with coefficients to scale what I think is a 32 bit audio buss, keeping the audio from "clipping" internally. Unless my recent research is flawed, HD uses a 24 bit buss and NON-floating point math to do processing. This does not easily accommodate any boost or summing of full scale digital signals, by default. You would have to trim the input to any plug-in that might provide digital gain.

Doesn't this seem ass-backwards? You would think the high end version of a product would be more bulletproof. It doesn't mean you can't safely accommodate full scale signals in HD, only that you must be aware that they can clip plug-ins and possibly summing busses more easily. The HD user should be ready to attenuate the input to a plug in with full scale or even lower signals, and keep his eye on all clip indicators.

Kind of makes ya wanna use a good ol analog desk with input trims, doesn't it?
Old 23rd June 2006
  #123
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Good information Dave, never heard that before.

I use Cubase SX3 or Nuendo 2 which are both 32-bit float.

Quote:
Unless my recent research is flawed, HD uses a 24 bit buss and NON-floating point math to do processing.
That's true, HD 's plugin bus is 24-bit fixed while most native daws are 32-bit float from input to bus output.

Lawrence
Old 23rd June 2006
  #124
Quote:
Originally Posted by Lawrence
Good information Dave, never heard that before.

I use Cubase SX3 or Nuendo 2 which are both 32-bit float.



That's true, HD 's plugin bus is 24-bit fixed while most native daws are 32-bit float from input to bus output.

Lawrence
Same for Samplitude as well I believe, that is what I use and I am pretty sure that is it 32 bit float. Wonder if it has something to do with Native vs. HD somehow? Something to do with the PCI buss on the mix cards?

Anyway wayyyy back about 35 pages ago I mentioned that I think we are talking about 2 different things here. Group a) is talking about converter resolution and group b) is talking about the DAW mix buss. While they are independent systems to some degree they are obviously related (hence the confusion).

I still maintain that in the end, just as Lawrence said, the benefits of lower levels at the converter overshadows the loss of resolution from not using the full scale.
Old 23rd June 2006
  #125
Dave you might be the best person to answer this but what about the headroom on the analog chain that was mentioned a page or so back. That seems valid to me but I wanted to get your take.

Quote:
Originally Posted by From Lawrence post on page 4 of this thread

Even more importantly, when you record this hot, I've got to ask - what did you do to your preamps, and analog chain to get this level? Most converters are set so that 0dbVU = -18dbFS. This means that if you set your preamp so that the output level shows 0 on a Studer 24 track's VU meter, then if you switched the preamp's output into your converter without changing anything, the level would read -18 below full scale.

That means that if you're getting -6 below full scale on your converter, that you're +12db over the 0VU point! Many analog electronics can crap out here, but almost all will sound different at least. Some times it may be "better" but usually, its a small accumulation of distortion that builds into a waxy fog that makes people blame "digital recording" for its pristine playback of their slightly distorted, but "pretty on the meter," tracks.

If you record with levels around your 0 point, some thing like -18dbFS or -14dbFS, depending on how your converter is calibrated, you'll have your analog electronics in their sweet spot, headroom for plug ins and summing, an appropriate analog friendly level if you use analog inserts later in the process, and on a modern 24bit converter with 110db S/N, the ability to faithfully record signals with a dynamic range of over 90db.
Old 23rd June 2006
  #126
The Distressor's "daddy"
 
Dave Derr's Avatar
 

Analog Signal Chain

Well I dont see how you could make rules on this. Most converters have input level trims, and who knows what the hell your analog chain is. I havent seen a converter that couldnt hit full scale with 20dBm or less, one way or the other. +20dBm is pretty easy for any pro analog gear to put out, especially since we are talking peaks, and not RMS.

Many people record thru chains that are custom such as NEVE1073, into Distressor , into Converter etc etc, and not even using the buss on analog mixing consoles. But still, most mixing consoles have busses that can put out +20dBM without any clipping or stress. And don't forget, most converters can be calibrated to clip well below a 20dBm input. Some have the old -10, +4dBm switch on them.

Lots of older vintage gear works great with reallllllly hot outputs. Most of Empirical Lab gear can put out +26dBm differential, the Lil FrEQ can put out +32dBm from its transformer output. Dan Kennedy designs Great River stuff to regularly put out +30dBm levels.... with a yawn. SSL Consoles put out +26dBm as most modern analog consoles do, differentially.

Shouldnt be any strain on analog gear to hit 0dBFS on the converters Ive seen. It would be interesting to do research on this tho. There ARE a lot of converters out there these days. Anyone seen a converter that couldnt go full scale with a +20dBm input?

Im still kind of shocked at the 24 bit HD buss thing. I wonder why they didnt use floating point, or provide for a wider buss. One must say that my ignorance was bliss.
Old 23rd June 2006
  #127
500 series nutjob
 
pan60's Avatar
 

dose anyone know how DP handles the math?
Old 24th June 2006
  #128
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Quote:
Now the great engineer using ADATs who printed with peaks at 0DBFS has got a better resolution than mr conservative using HD or Apogees or whatever.
Depends on which ADAT's...if he's got 20-bit ADAT's he's probably close. 16-bit and if he's recording the kick or snare you mentioned earlier, then if that 16-bit converter were perfect he'd be at exactly the same spot. But how many "usable" bits does a 16-bit converter have?

Quote:
In the end, I think the connection between dynamic range and bit resolution is completely irrelevant.

If it was, we would print a mix with a modern 3dB dynamic range to an 8bit converter and be fine and no one would debate that that's not true.
The connection between dynamic range and bit resolution (or bit depth) is not at all irrelevant...the bit depth is what defines the available dynamic range in a digital system. How could that be irrelevant?

If you really had a mix with a dynamic range of 3 dB, and you played it back on a system with an 8-bit converter, it would sound fine. Until the track ended, at least.

Quote:
Sample rate is like the frame rate in film. The higher the rate, the smoother the reproduction - for example, movement in film, the roundness of a sine wave in audio.
Higher sampling rates don't make a "rounder" sine wave. A sine wave is by definition "round". Higher sampling rates allow us to capture and reproduce smaller sine waves (higher frequencies) but what comes out of our D/A converter will always be perfecty "round" sine waves. No "steps" that are so often used to visualize digital audio.

Quote:
The concept of targeting ODBFS for peaks does not mean that your levels are all smashed up within 1-2dB of 0DBFS for the entire song, it means you've got your average levels, the stuff that counts because you actually hear it, up to a level where it's getting more 1s and 0s to describe it's detail.
As long as your signal is high enough above the noise floor that you don't hear it and far enough below it that your'e not clipping, you're using the same number of 1s and 0s to describe the detail. In one case you have more "always on" bits (when you're peaking closer to 0 dBFS) and in the other more "always off" (when you're peaking lower) but either way you're using the same number of them to capture and reproduce the signal. Again, there are other reasons you may want to peak higher or lower, but "using more bits" is not one of them.

Quote:
If we needed bits to describe the dynamic range we actually use we could all uses blackface ADATs for converters.
There are reasons that the blackface converters sound inferior aside from the fact that they were sixteen-bit converters. If you take your favorite 24-bit converter and properly dither it and record a signal with a dynamic range that can be contained within that range you can get great results with a sixteen-bit system. In that case, though, peaking closer to 0 dBFS will be more important since your noise floor will be higher.

Quote:
Unless my recent research is flawed, HD uses a 24 bit buss and NON-floating point math to do processing. This does not easily accommodate any boost or summing of full scale digital signals, by default
The mix buss if 48-point fixed. I do believe that the signal has to be dithered down to 24 bits as it passes between DSPs, though, so you do have to be careful about gain staging at different points along the way.

-Duardo
Old 24th June 2006
  #129
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Nordenstam's Avatar
 

Quote:
Originally Posted by Dave Derr
This does not affect our discussion on getting the best quality from your ADC converters by using full scale... this is almost a seperate issue. You will still get the best resolution and achieve the specified performance of an ADC when you let the peaks go close to 0dBFS. This again is by definition, and by the uncontrovertable nature of digital audio.
Please explain this: if bit depth is the quantization or dithering noise floor, as it undoubtly is, how can it alter the 'resolution' of the upper bits? It's not the image on the screen that counts, it's what we hear out of the DA.

Smaller increments between the steps on the volume axis equals less noise. Having more detail doesn't do anything else! Not less distortion, less of some frequencies, less phase lag or less anything. But the noise floor. By which other mechanism is it supposed to alter the audible outcome for the better in the upper bits?


Avoiding near zero troubles is one of the blessings of modern conversion. As loud as possible is a waste of energy. Modern converters are in a different world than the stuff of years long gone. They're usually the most precise gear in the studio, surpassing many measuring instruments. They have no problem making perfectly fine readings of a sliding 60 to 90 dB real world signal within their 110-120dB range. There is no need to blow the input!


Quote:
As a side note it is appropriate at this point to emphasize that
all analog signals have some form of noise corruption. If for
example an input signal has a finite signal-to-noise ratio of
40dB it would be superfluous to select an A/D converter with
a high number of bits. It may be realized that the use of a
large number of bits does not give the digitized signal a
higher signal-to-noise ratio than that of the original analog
input signal. As a supportive argument one may say that
though the quantization steps q are very small with respect
to the peak input signal the lower order bits of the A/D
converter merely provide a more accurate representation of
the noise inherent in the analog input signal.

National Semiconductor Application Note 236 - An introduction to the sampling theorem
Wikipedia: Quantization (sound processing)

Bob Katz: "It is a common myth that audible signal-to-noise ratio will deteriorate if a recording does not reach full scale digital." http://www.digido.com/portal/pmodule...er_page_id=59-


Related:

Nika Aldrich: The Consequences of Traditional Digital Peak Meters

Long thread at PSW with loads of good info.



With no personal flames to anyone and all best intentions for a better sounding era of digital audio,

Andreas Nordenstam
Old 24th June 2006
  #130
Quote:
Originally Posted by RainbowStorm
With the highest possible resolution you get the best possible description of the sound,
Exactly. I don't understand why people view it interms of dynamics.
Old 24th June 2006
  #131
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John The Cut's Avatar
 

Quote:
Originally Posted by synthoid
There's no reason why converters have to be designed so that they are most accurate right at the edge of clipping.

-synthoid

well... cost? High end converters are already massive bucks for a full 16 channels of AD, DA

What you talk about basically amounts to another converter circuit just to deal with the headroom. Its just a lot easier to turn sh!t down
Old 24th June 2006
  #132
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Quote:
Originally Posted by RainbowStorm
Some more thoughts on this topic... I must admit that besides the op amp quality I think digital jitter (noise artifacts due to an inaccurate internal crystal reference clock in the converter)
Strictly, this is not what jitter is.
Old 24th June 2006
  #133
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James Lehmann's Avatar
 

Reading through this excellent thread from a position of comparative ignorance it seems to me different people are talking about different things here - there are several parallel but inter-related discussions going on:

1) Audible differences in utilising every last dBfs in a 24-bit recording
2) Audible differences in overloading mix busses
3) Audible differences in overloading plug-ins
4) Audible differences between floating-point and fixed-point systems

Some A/B test WAV's would be listened to with interest.
Old 24th June 2006
  #134
Gear Guru
 
lucey's Avatar
Silver Sonya I agree 100% ... gain staging is foreign to millions!



Lawrence the d8b handles data and does conversion so poorly that I wouldn't trust much of anything you learn through it.



I dont like DAWs much but enjoy the 002s bussing ... it almost never craps out, and now does 96 channels.
Old 24th June 2006
  #135
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Quote:
Originally Posted by mogWai
Strictly, this is not what jitter is.
What jitter is and not is, that's a subject of discussion, as some even go as far as saying it's the reason for audible problems when there are no other explanations. Sctrictly speaking though it's random short-term variations of the significant instants of a digital signal from their ideal positions in time. There are a number of different things contributing to jitter, clock-jitter (which I already mentioned) is one important source to jitter, but there are other sources as well, such as power supply fluctuations. I won't go deeper into this subject as it is very deep in itself.
Old 24th June 2006
  #136
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>>With the highest possible resolution you get the best possible description of the sound
>Exactly. I don't understand why people view it in terms of dynamics.

There's two major factors in sampling. The sample rate sets the frequency content from zero to half the sample rate. The bit depth sets the noise level(quantization error).

For each additional bit, the noise floor decreases by about 6dB. It doesn't do anything magical to the describe the sound better in the part that is above the noise floor! What end result should that bring? A change in frequency balance? Less distortion on the peaks of the signal? Phase changes? Noo.. It gives less quantization distortion on the lowest levels of the signal - the noise floor.

Better description gives less noise.
Less noise gives more dynamic range.
Old 24th June 2006
  #137
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Quote:
Originally Posted by Mike Caffrey
Exactly. I don't understand why people view it interms of dynamics.

People view it in terms of dynamics because until the noise floor reaches the S/N of your signal, it simply doesn't matter. The waveform will be represented perfectly anywhere in that range, until noise begins to affect it.

If you stay above the noise floor of the dynamic range the signal should be exactly the same at -10 or -2 except for amplitude.

As was stated if your record anything with 70db of dynamic range you're doing good. You don't need the full 24-bits to represent that.

That's the myth that so many people seem to be holding on to.

Lawrence
Old 24th June 2006
  #138
Gear Maniac
 

I'm going to offer a couple of conclusions I've come to from this thread, and see if the experts here can groove with it.

First, if you're mixing analog, you should get as high a level as possible, in order to send the most level to your mixer and use its dynamic range to the fullest, whether you record to tape or to hard drive. But if you mix digitally, or plan to use digital effects, you want to leave some space so as not to overload your automated counting box. It appears that it would be enough simply to reduce the level of the recorded tracks after recording, so that you could record with high levels if you didn't know whether something would be mixed in or outside of the count-box.

Second, it seems that on an absolute basis, your record levels, again whether you use tape or hard drive, need to be concerned with the noise floor of your source and with your requirements in terms of perceived noise floor in the final product. If you have a source with little noise floor, or again if you need an imperceptibly low noise floor (for example, if you are recording someone playing the drums with telekinesis, or maybe the government has some needs like this), then you would want to record at the highest level possible without clipping. A noisy source or a product in which noise is unimportant (say, a dictaphone, or maybe some music with no dynamic range and huge amounts of distortion) would require you only to record it at a level which puts its noise floor above that of the recorder. Since digital recorders have noise floors below tape machines, you could record it even lower on them.

Now, what interesting is that it seems that not the level but the dynamic range of a recording is what determines the "number of bits" used to record it.

OK, back to it folks.

b
Old 24th June 2006
  #139
The Distressor's "daddy"
 
Dave Derr's Avatar
 

I GIVE

Old 24th June 2006
  #140
500 series nutjob
 
pan60's Avatar
 

ok if no one knows, how is the best way to find out how DP handles the math?
Old 24th June 2006
  #141
The Distressor's "daddy"
 
Dave Derr's Avatar
 

Digital Performer

Pan, this is from their website:
Real Time 32-bit effects processing

"Enjoy Digital Performer's real-time, 32-bit native effects processing on your Pro Tools Project (PCI), Project II, Pro Tools III (PCI) or even Pro Tools|24 system. Digital Performer's MAS-native effects include up to 40 (!) bands of parametric EQ per track, dynamics processing (compressor/expander/limiter/gate), chorus, echo, flange, phaser, tremolo, our sparkling eVerb, PreAmp-1 tube warm/distortion, the utterly unique Sonic Modulator - and even the new MasterWorks Compressor/Limiter plug-ins. You can also run world-class third-party plug-ins such as Waves and DUY in real time. All of these plug-ins sound fantastic, thanks to the MOTU Audio System's internal 32-bit floating point processing resolution."
Old 24th June 2006
  #142
500 series nutjob
 
pan60's Avatar
 

thanks Dave i looked on their site but i missed that somehow.
Old 25th June 2006
  #143
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danasti's Avatar
 

A digital clip is the loudest most annoying noise that will consume an entire mix.

If they can't hear it they shouldn't be in audio.
Old 25th June 2006
  #144
Quote:
Originally Posted by Lawrence
I give up. No point in going on with this discussion.

If you have no interest in actually trying it to see if if makes any or no difference with this newer class of modern converters and the "resolution" differences between the tracking and mixing methods then this entire conversation is pointless.

Uncle.

Theory (about something that may have taken place "years ago") disproves and makes irrelevant real world experiences ("what happens to many every day, now") everytime on the Internet.

I'd actually be shocked if it was any other way. heh

Live long and prosper.

Lawrence
What an odd tone to take in a public discussion that so far has been civilized, interesting and informative.

It's also especially odd that you're putting words in my mouth when everyone else is seeing the same posts and there were none where I said "I have have no interest in trying....."

I've easily logged 30,000 hours as a professional engineer in NYC. If you think I waited until the beginning of this thread to begin thinking about bit rate, you're wrong. That's something I took care of early on.

If you think I remember exact details of how I came to my conclusions about bit rate some time around 20,000 hours of sessions ago, hell even 1,000 hours ago, you give me far too much credit and seriously overestimate the quality of my memory. But thanks for the optimism!

Instead, since you've obviously done the A/B (properly, of course), why not offer to send me the files you made during your A/B test? I'm sure your real world experience has shown you that that would be more effective. Who knows, maybe I'm incapable of properly setting up the A/B or maybe I wouldn't have known that I need to to the same A/B test on multiple sources, like individual drums/percussion, bass, guitars, keys, vocals, full mixes, as I'm sure you did when you did your A/B.

So, yes you're probably right, we should discontinue our discussion, as it's clearly taken a turn for pointless.
Old 25th June 2006
  #145
Quote:
Originally Posted by Duardo
Depends on which ADAT's...if he's got 20-bit ADAT's he's probably close. 16-bit and if he's recording the kick or snare you mentioned earlier, then if that 16-bit converter were perfect he'd be at exactly the same spot. But how many "usable" bits does a 16-bit converter have?
-Duardo

Whether there at the same spot or not is all dependent on how many bit got used in each scenario.
Quote:

The connection between dynamic range and bit resolution (or bit depth) is not at all irrelevant...the bit depth is what defines the available dynamic range in a digital system. How could that be irrelevant?
Becuase we don't need anywhere near the dynamic range that digital is giving us. The reason to care about bit rate is not so that we can have a song that goes from the acutal volume of a whisper to the acutal volume of a jet enginer from 6 inches away, the reason is that is sample is described more acurately with a longer digital word. If you could do a freeze frame on a single sample of audiio, a 64 bit description would sound far better than a 16 or 24 bit description.

Quote:

Higher sampling rates don't make a "rounder" sine wave. A sine wave is by definition "round". Higher sampling rates allow us to capture and reproduce smaller sine waves (higher frequencies) but what comes out of our D/A converter will always be perfecty "round" sine waves. No "steps" that are so often used to visualize digital audio.
I think the term for what I'm describing is aliasing. If you sample a sine wave at fewer point in the wave, there are gaps. The more samples the fewer gaps and since the wave is curved, making it smoother makes it more round, but we can describe it as smoother or more accurate if you want. Now, if your tlaking about wnat comes out and saying that the D/A converter inerpolates or extrapolates and fills in the gaps that that there aren't any gaps, fine. But which would you prefer gaps filled in by math or more an more samples to minimize the gaps even further?
Old 25th June 2006
  #146
Quote:
Originally Posted by James Lehmann
Reading through this excellent thread from a position of comparative ignorance it seems to me different people are talking about different things here - there are several parallel but inter-related discussions going on:

1) Audible differences in utilising every last dBfs in a 24-bit recording
2) Audible differences in overloading mix busses
3) Audible differences in overloading plug-ins
4) Audible differences between floating-point and fixed-point systems

Some A/B test WAV's would be listened to with interest.

That's a good observation.

I think there's also the point being made that the bottle neck happens at the plugins, so your levels should be set not based on what sounds best in the raw audio file, but what sounds best in the final mix and that the plugin bottle next has a greater impact than what you've labeled discussion #1.
Old 25th June 2006
  #147
Here for the gear
 

Quote:
Originally Posted by Mike Caffrey
I think the term for what I'm describing is aliasing. If you sample a sine wave at fewer point in the wave, there are gaps. The more samples the fewer gaps and since the wave is curved, making it smoother makes it more round, but we can describe it as smoother or more accurate if you want. Now, if your tlaking about wnat comes out and saying that the D/A converter inerpolates or extrapolates and fills in the gaps that that there aren't any gaps, fine. But which would you prefer gaps filled in by math or more an more samples to minimize the gaps even further?
The above argument reflects a common misunderstanding about digital audio. Beneath the Nyquist frequency (sampling rate/2), more samples per sine wave do not create a more "accurate" reproduction of a sine wave. For instance, if you sample a 10 kHz sine wave at 44.1 kS/s, you'll only manage to get 4 samples per wave cycle; however, as Nyquist demonstrated, that is enough information to perfectly represent the 10 kHz sine wave. Sampling the 10 kHz sine wave at a higher sampling rate does not provide a more "accurate" representation of the sine wave. I know this is not intuitive, but that's the story. Blame Nyquist if you don't like it!

Some good resources to get a handle on this stuff: Dan Lavry's white papers at www.lavryengineering.com, Nika Aldrich's Digital Audio Explained for the Audio Engineer, Bob Katz's Mastering Audio.

Miles
Old 25th June 2006
  #148
Gear Nut
 
Ricky's Avatar
 

Quote:
Originally Posted by Silver Sonya
I have a radical concept: what if Protools/Logic/Nuendo, etc. had a bult in alarm that would sound whenever the user overloaded the i/o of a channel, buss, or plug-in within the system? Or better yet: prevented them from doing it at all! Like maybe the fader just won't go as high as you want it to... or maybe at some point, turning the fader up on one channel simply results in all other channels being lowered?
Like Protools (AN) Audio Natzi
Old 25th June 2006
  #149
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midnightsun's Avatar
 

Quote:
Originally Posted by ObnoxiousTyrant
That's exactly right. Anytime you lift a fader ITB or amplify a signal ITB, it increases the harshness or brittleness of the mix. Use a preamp to amplify and if a signal is weak, reprocess it through an OTB preamp. Let the amplifier do the amplification and keep the digital fader at unity or below. Cut frequencies with digital eq's and compressors, don't amplify them.
hummmm.... so a fader should be used for "fading."
Old 25th June 2006
  #150
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Quote:
Originally Posted by Bob Olhsson
Instead of jawing about theory, I'd recommend that people do some actual testing for themselves.

My epiphany came when I set up a songwriter friend with a digi 001 system and he turned up a few weeks later with bounced mix files that peaked at -25 because he couldn't figure out how to get any higher levels.

I was working on a mastering project that had been mixed by a great engineer on an SSL 9k using Lavry Blue converters. My friend's demos using the 001 with just an old U-87, a 414 and a 57 sounded bigger, ballsier and more like analog than 90% of what you hear today including the high-end project I was working on.

It's important to never think of digital audio in analog terms. It's numbers. Period. Digital clipping doesn't sound that bad provided you never try to calculate any additional signal processing math using those numbers. At that point you'd have more resolution with a 12 bit signal than the mathematical gibberish that digital clipping produces.
Bob... this makes a ton of sense. Do you think that it follows that if a person is prone to record to close to 0 that an analog console might mitigate some of the mathematical gibberish?
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DeadPoet / So much gear, so little time
1

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