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Old 5th October 2005
  #121
oh but I do want to read the paper ... it's just that the link does nothing when you try to save the file
Old 5th October 2005
  #122
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Quote:
Originally Posted by Mario-C.
oh but I do want to read the paper ... it's just that the link does nothing when you try to save the file
you have to log in for the link to work.

anyway, check your normal email. i sent it as an attachment.
Old 5th October 2005
  #123
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djui5's Avatar
 

Quote:
Originally Posted by Jay Kahrs

I'm totally with you until the very last sentence or two.

Sure, an analog meter will miss all kinds of peaks because it's ballistics aren't fast enough to catch 'em...but a good digital PPM should show all the peaks no matter how fast they are right? Or are we talking about the leading edge of something with a sharp transient like a snare drum...which a PPM meter should still show. And besides, once the signal has passed through the A/D and is in the digital domain then things like proper, realistic metering should be easy.

Right? (???)

You can only push things so far in either analog or digital before you run out of steam and it starts to sound like poo. The real difference is that when analog distorts it compresses a little and can sound "rock & roll" before it turns to total mush. When digital hits the wall...it's literally a wall and it's pretty damn unforgiving. Maybe what we need are real dummy lights, stuff that lights up red when the signal is 3dB away from the point of clipping. I changed the meter colors (if supported) on any software I own to reflect that and my all my gear, analog and digital is much happier.
Clipped from white paper.

"
The consequence of the way in which DAW’s treat waveforms is that the meter inside the DAW or other digital mixers inevitably shows inaccurate information. It is virtually a mathematical certainty that the waveform will actually exceed the amplitude of the samples in any sampling system. The samples themselves only represent a waveform, It is important to understand that the amplitude of the waveform will invariably exceed the sample values.
Manifestation
One may ask why this poses a problem. For various reasons, mostly having to do with marketing demands and industry trends, recordings made and mastered in today’s recording environment are mixed and mastered as ‘hot’ as is possible, pushing the levels up to the highest tolerable amount, supposedly just short of clipping. Sophisticated digital tools allow music to be highly compressed, then recompressed, compressed even more so with multi-band compressors, limited, normalized, and maximized to get the audio to play as loud as possible out of a consumer’s system. Hence, it is very common for popular music CDs to be full of digital samples that are at, or nearly at full scale.
The problem is realized in that while going through these digital gyrations and utilizing digital tools to amplify the signal as much as possible, both during mixing and during mastering, the ‘peak value’ of the sample points is closely watched to ensure that it does not get to full scale. Since, the peak meters in said DAW and digital mixing systems are inaccurate, and do not actually indicate the peak values of the resulting waveform, the result is that while the samples themselves do not exceed full scale, and are carefully monitored to insure this, the resulting waveforms represented by the samples may exceed full scale throughout any standard CD! July 2003 Page 5
While the digital mixing system is not clipping the music or distorting the music, the digital to analog converters that have the task of recreating the audio through digital reconstruction filters are clipping repeatedly throughout most CDs on the market. The result is that most CDs and DVDs end up distorting with regularity when they are asked to reconstruct and play back audio that appears to be completely ‘legal’ because not a single sample actually clipped.
"


I was guessing that the signal was faster than the meters could represent. I though PPM meters would represent everything, no matter how fast it was. It was just a thought I had.


One problem I have with this is if the waveform is hotter than the A/D's dynamic range on the way in, how is the converter going to "reconstruct" something it can't sample in the first place and will inaccurately represent inside the DAW.

Also, if your mixing ITB, and you process the signal at all, you're going to change the waveform. How is it going to reconstruct a signal that it didn't sample properly in the first place, and has been modified while mixing.


I'm not getting all of this....it dosen't make sense to me. Maybe I'm just a little tired tonight. I'll think it over and re-read the paper again tomorrow.
Old 5th October 2005
  #124
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djui5's Avatar
 

Quote:
Originally Posted by Mario-C.
oh but I do want to read the paper ... it's just that the link does nothing when you try to save the file

I attached the paper to a post on the previous page.
Old 5th October 2005
  #125
Lives for gear
Quote:
Originally Posted by djui5
One problem I have with this is if the waveform is hotter than the A/D's dynamic range on the way in, how is the converter going to "reconstruct" something it can't sample in the first place and will inaccurately represent inside the DAW.
if you didn't clip it on the way in, it's fine, and will play fine on the way back out. that's not where the problem lies ( i recorded something hot within reason, applied the TL meter to it to see how it would play back, and no problem at all). the problem is after processing with a plug.

Quote:
Also, if your mixing ITB, and you process the signal at all, you're going to change the waveform. How is it going to reconstruct a signal that it didn't sample properly in the first place, and has been modified while mixing.
again, if it didn't clip on the way in, it's fine. however after applying a plug, the waveform will obviously change as will the reconstruction of that waveform. that's why you need to trim the audio. after EQing for example (whether adding or subtracting), you can easily add another 6dB. with the meter you can see this clearly, so whatever little headroom you have after the initial recording, can instantly disappear after processing with just one plug!

Quote:
I'm not getting all of this....it dosen't make sense to me. Maybe I'm just a little tired tonight. I'll think it over and re-read the paper again tomorrow.
read it again fresh and you'll get it. it takes awhile to grasp the concept, but even i got it after awhile!
Old 5th October 2005
  #126
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Quote:
Originally Posted by djui5
Like stated earlier, this is the reason for the higher bitdepth mixing busses. To retain your 24 bit's of dynamic range, while allowing you to move the signal up or down in volume.


Imagine an ice cube in a glass of water. Whether that ice cube moves up or down in the glass, it's still the same size and displaces the same amount of water. 24bit's is the ice cube, the rest is the water.

The reason for the higher bit depth is the higher resolution of the calculations. This whole theory sounds interesting but can you back it up? Some links or texts confirming this?
Old 5th October 2005
  #127
Gear Maniac
 

Thanks to gwailoh, djui5, and Raal on very helpful explanations here. I wonder about some related issues here which I hadn't fully considered before:

1) How accurate is the digital meter on the ADC? If you track at the ADCs max sample rate, e.g. 96k or 192k, will your ADC meters indicate intersample peaks or only the recorded samples? If the latter, do the meters have a "cushion" (Bob Katz's term) of a few dB to avoid overs at reconstruction? I expect one's mileage will vary...

2) Even for legal and truly unclipped (when reconstructed) PCM streams, and even if your ADC inputs aren't padded, does your ADC really sound best when it is run hot?

3) To explore problem 1) I assume Inspector XL (or equivalent) should first be inserted and monitored right after the ADC as a safeguard, even when nothing precedes tracking to the DAC, yes?

Sorry if this post wanders from the topic.
Old 5th October 2005
  #128
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gsharp's Avatar
 

Quote:
Originally Posted by Fredrik
The reason for the higher bit depth is the higher resolution of the calculations. This whole theory sounds interesting but can you back it up? Some links or texts confirming this?
The reason for the higher bit depth of the mix bus is to preserve what you started with on each track while allowing you to accumulate tracks and apply gain to them. 1 24 bit track with no gain applied would only require a 24 bit mix bus. Doubling the # of tracks adds a bit required at the mixbus to calculate the result. So 2 tracks would need 25 bits, 4 tracks need 26, and so on. 128 tracks with no gain applied would sum into 31 bits.

The 48 bit mix bus allows gain to be applied upward and allows faders to be pulled down without losing data. The 48 bits is not there to apply some addition 'resolution' to your audio per se. Your audio is 24 bit (or 16 or whatever) and that's that. No adding resolution after the fact. The 48 bits are there to provide a cushion above and below the 24 bits of actual audio data while you manipulate (mix) the tracks together.
Old 5th October 2005
  #129
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gwailoh's Avatar
 

Quote:
Originally Posted by Samuel L. Lord
Thanks to gwailoh, djui5, and Raal on very helpful explanations here. I wonder about some related issues here which I hadn't fully considered before:

1) How accurate is the digital meter on the ADC? If you track at the ADCs max sample rate, e.g. 96k or 192k, will your ADC meters indicate intersample peaks or only the recorded samples? If the latter, do the meters have a "cushion" (Bob Katz's term) of a few dB to avoid overs at reconstruction? I expect one's mileage will vary...

2) Even for legal and truly unclipped (when reconstructed) PCM streams, and even if your ADC inputs aren't padded, does your ADC really sound best when it is run hot?

3) To explore problem 1) I assume Inspector XL (or equivalent) should first be inserted and monitored right after the ADC as a safeguard, even when nothing precedes tracking to the DAC, yes?

Sorry if this post wanders from the topic.

If I understand Paul Frindle correctly -- his posts on PSW are the original source of all this killer info -- you don't need to worry about your ADC causing reconstruction errors later in the DAC. I think that's because by definition the ADC creates a legal sample representation of an unclipped waveform. At that stage, there will be no intersample overs, provided you haven't clipped the ADC itself. (I suppose this assumes that the ADC is designed properly!) The only issue at input is whether you clip the ADC. If you don't, you should be able to work with the resulting samples with no reconstruction errors provided you do no more processing on those samples. I hope this is clear!

Re InspectorXL, sadly enough it's not an oversampling meter. There's a previous message in this thread which relays my conversation with them. Please do join me in requesting that they implement oversampling. I like Inspector XL very much and it would be one-stop-shopping for me if they'd do oversampling.
Old 5th October 2005
  #130
Gear Guru
 
u b k's Avatar
 

my understanding from the reading i've done is that, absent an oversampling meter, if you simply keep the level going into and out of a plugin to -8db, the odds of ever having an intersample peak is next to nil.

which comports with my own experience that the more conservative i play it in the daw, the more open things stay.


gregoire
del ubik
Old 5th October 2005
  #131
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RKrizman's Avatar
 

Quote:
Originally Posted by JoeyB
Also how do the "high end" converters (as mentioned in the white paper) handle the illegal data? If you have a converter that can handle it nearly transparently, couldn't you just do a D/A A/D straight wire at unity and allow the converter to "legalize" the data on the round trip?

Does this even make sense?
I had the same thought but couldn't figure out how to esplain it.

Right, bring it back in and then don't mess with it anymore. you could do this to individual tracks or the whole mix.

-R
Old 5th October 2005
  #132
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Quote:
Originally Posted by gsharp
The reason for the higher bit depth of the mix bus is to preserve what you started with on each track while allowing you to accumulate tracks and apply gain to them.
I Agree


Quote:
Originally Posted by gsharp
1 24 bit track with no gain applied would only require a 24 bit mix bus. Doubling the # of tracks adds a bit required at the mixbus to calculate the result. So 2 tracks would need 25 bits, 4 tracks need 26, and so on. 128 tracks with no gain applied would sum into 31 bits.

The 48 bit mix bus allows gain to be applied upward and allows faders to be pulled down without losing data.
This is not true, as soon as anything is done in a digital mixer something is lost, sorry.

[QUOTE=gsharp] The 48 bits is not there to apply some addition 'resolution' to your audio per se. Your audio is 24 bit (or 16 or whatever) and that's that.QUOTE]

Aha, now thats not true. I´m not saying that any aditional info is to be gained in the data itself, but once you start to manipulate the data it can be done with higher precision if the "grid" is twice as big so to speek.

Quote:
Originally Posted by gsharp
No adding resolution after the fact. The 48 bits are there to provide a cushion above and below the 24 bits of actual audio data while you manipulate (mix) the tracks together.

This sounds sweet, but like a metaphore. Perhaps we are speakin of the same thing?
Old 5th October 2005
  #133
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gsharp's Avatar
 

[QUOTE=Fredrik]I Agree




This is not true, as soon as anything is done in a digital mixer something is lost, sorry.

Quote:
Originally Posted by gsharp
The 48 bits is not there to apply some addition 'resolution' to your audio per se. Your audio is 24 bit (or 16 or whatever) and that's that.QUOTE]

Aha, now thats not true. I´m not saying that any aditional info is to be gained in the data itself, but once you start to manipulate the data it can be done with higher precision if the "grid" is twice as big so to speek.




This sounds sweet, but like a metaphore. Perhaps we are speakin of the same thing?
You agree with the first point but disagree with the 2nd point? What exactly is "lost" when you do something in a digital mixer. Is this same 'loss' not present in an analog mixer?

Just for fun I took a track with a song ripped from a cd. I turned the gain on the fader down 3dB and turned the gain on the Master up 3dB. I bounced a piece, pulled it back in, lined it up, inverted the polarity and they null to -90dB (this is a 16bit file so 96dB is max dyn range). I'm guessing the 6dB is 1 bit of dither at the mixer output (I'm running the PT dithered mixer). What other 'loss' should I be looking for?
Old 5th October 2005
  #134
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[QUOTE=gsharp]
Quote:
Originally Posted by Fredrik
I Agree




This is not true, as soon as anything is done in a digital mixer something is lost, sorry.



You agree with the first point but disagree with the 2nd point? What exactly is "lost" when you do something in a digital mixer. Is this same 'loss' not present in an analog mixer?

Just for fun I took a track with a song ripped from a cd. I turned the gain on the fader down 3dB and turned the gain on the Master up 3dB. I bounced a piece, pulled it back in, lined it up, inverted the polarity and they null to -90dB (this is a 16bit file so 96dB is max dyn range). I'm guessing the 6dB is 1 bit of dither at the mixer output (I'm running the PT dithered mixer). What other 'loss' should I be looking for?

I think that an extreme fader down say -60db would be audiable as loss in resolution dont you agree? So is this loss introduced abruptly or gradually? I think gradually but you might not be able to easily detect it until it´s´done to the extreme.

I agree with the fact that double precision arithmatic will mantain the original data intact in software, calculations can be done with double precision. But a -60db signal in the mixer wont give a 24bit signal after the dac, it´s ónly 24bit after all.
Old 5th October 2005
  #135
One with big hooves
 
Jay Kahrs's Avatar
Quote:
Originally Posted by djui5
Clipped from white paper.

"
The consequence of the way in which DAW’s treat waveforms is that the meter inside the DAW or other digital mixers inevitably shows inaccurate information. It is virtually a mathematical certainty that the waveform will actually exceed the amplitude of the samples in any sampling system. The samples themselves only represent a waveform, It is important to understand that the amplitude of the waveform will invariably exceed the sample values.
Manifestation
One may ask why this poses a problem. For various reasons, mostly having to do with marketing demands and industry trends, recordings made and mastered in today’s recording environment are mixed and mastered as ‘hot’ as is possible, pushing the levels up to the highest tolerable amount, supposedly just short of clipping. Sophisticated digital tools allow music to be highly compressed, then recompressed, compressed even more so with multi-band compressors, limited, normalized, and maximized to get the audio to play as loud as possible out of a consumer’s system. Hence, it is very common for popular music CDs to be full of digital samples that are at, or nearly at full scale.
The problem is realized in that while going through these digital gyrations and utilizing digital tools to amplify the signal as much as possible, both during mixing and during mastering, the ‘peak value’ of the sample points is closely watched to ensure that it does not get to full scale. Since, the peak meters in said DAW and digital mixing systems are inaccurate, and do not actually indicate the peak values of the resulting waveform, the result is that while the samples themselves do not exceed full scale, and are carefully monitored to insure this, the resulting waveforms represented by the samples may exceed full scale throughout any standard CD! July 2003 Page 5
While the digital mixing system is not clipping the music or distorting the music, the digital to analog converters that have the task of recreating the audio through digital reconstruction filters are clipping repeatedly throughout most CDs on the market. The result is that most CDs and DVDs end up distorting with regularity when they are asked to reconstruct and play back audio that appears to be completely ‘legal’ because not a single sample actually clipped.
"


I was guessing that the signal was faster than the meters could represent. I though PPM meters would represent everything, no matter how fast it was. It was just a thought I had.
Thanks for posting that little section, I *think* I get it now!

Yeah, originally I was with you...a PPM meter should show everything right? But, what about peak vs. RMS levels? PPM meters don't show RMS which is much higher and a truer indication of the actual volume.

Let's say we have some audio that's running below clipping, say -2dBfs is our highest peak. It's not just entirely possible, but highly plausible that the average level, or RMS value is actually much higher and that's where the problems with distortion are occuring. Even though none of the audio is anywhere near the point of clipping, the average level is higher then Redman at the Source awards and the D/A on the backend is being run into both digital and analog distortion.

Quote:
One problem I have with this is if the waveform is hotter than the A/D's dynamic range on the way in, how is the converter going to "reconstruct" something it can't sample in the first place and will inaccurately represent inside the DAW.

Also, if your mixing ITB, and you process the signal at all, you're going to change the waveform. How is it going to reconstruct a signal that it didn't sample properly in the first place, and has been modified while mixing.
It can't. Once it's sampled, it's sampled. But when you apply a plug-in or reset your faders the DAW is going to crunch numbers again and it's going to have to 'resample' (for lack of a better word) the audio and maybe that's where the meters are dropping the ball.
Old 5th October 2005
  #136
One with big hooves
 
Jay Kahrs's Avatar
Quote:
Originally Posted by gsharp
What exactly is "lost" when you do something in a digital mixer. Is this same 'loss' not present in an analog mixer?
When you reduce gain or pull faders down on an analog mixer, the audio will start to blend in with the noise floor but the overall signal stays pretty much intact. Drop a fader to -30 or -40dB in the digital domain and that same signal will lose resolution, clarity and start to sound grainy & pixelated.
Old 5th October 2005
  #137
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gsharp's Avatar
 

[QUOTE=Fredrik]
Quote:
Originally Posted by gsharp


I think that an extreme fader down say -60db would be audiable as loss in resolution dont you agree? So is this loss introduced abruptly or gradually? I think gradually but you might not be able to easily detect it until it´s´done to the extreme.

I agree with the fact that double precision arithmatic will mantain the original data intact in software, calculations can be done with double precision. But a -60db signal in the mixer wont give a 24bit signal after the dac, it´s ónly 24bit after all.
We're not talking about -60 out the dac, we're talking -60 within the mixer. A mixer with basically 288dB of dynamic range. You can pull a fader down to -96 before you start losing anything. The PT mixer is set up as follows:

Bits 48-40 for headroom
Bits 39-16 for your audio
Bits 15-0 for maintaining resolution as you pull faders down

You have 96db of 'headroom' to the downside. If you need to pull a fader down below -96 you might as well just mute the track. heh
Old 5th October 2005
  #138
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RKrizman's Avatar
 

Quote:
Originally Posted by Jay Kahrs
When you reduce gain or pull faders down on an analog mixer, the audio will start to blend in with the noise floor but the overall signal stays pretty much intact. Drop a fader to -30 or -40dB in the digital domain and that same signal will lose resolution, clarity and start to sound grainy & pixelated.
Jay, I think you're on shaky ground with this one. There's really no reason why it should, other than the challenge to the analog components to reamplify the signal loud enough to where you can evaluate it.

Although softer, the signal stays "24-bit wide" until it butts up against the lower limit of the headroom window.

At least, that's how Protools works. Can't say about the others.

-R
Old 5th October 2005
  #139
One with big hooves
 
Jay Kahrs's Avatar
Quote:
Originally Posted by RKrizman
Jay, I think you're on shaky ground with this one. There's really no reason why it should, other than the challenge to the analog components to reamplify the signal loud enough to where you can evaluate it.
That's entirely possible.

I'm not a digital theorist, nor am I going to pretend to be...so I probably have enough knowledge to be dangerous. heh I have a grasp on the basic concepts and the why's & hows but really, I'd rather spend my time turning knobs and making noises then being a digital audio mathematician. I want to know enough to know that I don't know enough. At the end of the day I make records and either people like 'em or they don't like 'em regardless of what tools were used in the process.
Old 5th October 2005
  #140
Gear Maniac
 

Quote:
Originally Posted by gwailoh
If I understand Paul Frindle correctly -- his posts on PSW are the original source of all this killer info -- you don't need to worry about your ADC causing reconstruction errors later in the DAC. I think that's because by definition the ADC creates a legal sample representation of an unclipped waveform... ...I hope this is clear!
Yes, on rethinking this, that makes perfect sense.

Quote:
Originally Posted by gwailoh
Re InspectorXL, sadly enough it's not an oversampling meter. There's a previous message in this thread which relays my conversation with them. Please do join me in requesting that they implement oversampling. I like Inspector XL very much and it would be one-stop-shopping for me if they'd do oversampling.
I should have read that. Thanks again gwailoh for your help!
Old 6th October 2005
  #141
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djui5's Avatar
 

It makes sense. Jay's RMS comments helped.


What I don't understand is how the reconstruction filters are going to create something outside of the range of the DAW's amplitude threshold. That's what's confusing me. It's like saying "Ok, paint me a picture of a mountain with a lake, but you can't see the mountain and lake, you have to imagine it. Make it perfect".

BTW, I don't have papers to back up my ice cube theory. I had read that somewhere (about the 48 bit thing, I made up the ice cube thing), but can't remember where.

It makes sense, and even if it's not true, it surely sounds good don't it? heh
Old 6th October 2005
  #142
One with big hooves
 
Jay Kahrs's Avatar
Quote:
Originally Posted by djui5
What I don't understand is how the reconstruction filters are going to create something outside of the range of the DAW's amplitude threshold. That's what's confusing me. It's like saying "Ok, paint me a picture of a mountain with a lake, but you can't see the mountain and lake, you have to imagine it. Make it perfect".
Imagination is something that people unlearn in school.

Such a shame really.

If you can imagine it then you can make it real. You have to have it all in your head to get it out right?

Like I said before I'm not a digital theory dude. My guess is that if the reconstruction filters are creating something that's outside the DAW's 'amplitude threshold' as you say then it's because of things accumulating to the final result and producing an additional tone.

Imagine taking a bunch of paints and mixing them together. What happens when you mix red & blue or green & red?

You get a new color.
Old 6th October 2005
  #143
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A test we can do to illustrate the concept of "Intersample Peakes" or the "Gibb Effect".

This has to be one of the greatest threads in GS to date.

Shane
Old 6th October 2005
  #144
Quote:
Originally Posted by raal
you have to log in for the link to work.

anyway, check your normal email. i sent it as an attachment.

thumbsup cool, thank you very much Raal !
Old 6th October 2005
  #145
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[QUOTE=gsharp]
Quote:
Originally Posted by Fredrik
We're not talking about -60 out the dac, we're talking -60 within the mixer. A mixer with basically 288dB of dynamic range. You can pull a fader down to -96 before you start losing anything. The PT mixer is set up as follows:

Bits 48-40 for headroom
Bits 39-16 for your audio
Bits 15-0 for maintaining resolution as you pull faders down

You have 96db of 'headroom' to the downside. If you need to pull a fader down below -96 you might as well just mute the track. heh
My point is that what is happening inside the PT mixer is all ideal! Sooner or later it all has to pass trough the DAC stage ans thats when you loose resolution.
Old 6th October 2005
  #146
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[QUOTE=Fredrik]
Quote:
Originally Posted by gsharp

My point is that what is happening inside the PT mixer is all ideal! Sooner or later it all has to pass trough the DAC stage ans thats when you loose resolution.
if i may correct, it's all ideal 'til you start using fixed point plugs...
Old 6th October 2005
  #147
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djui5's Avatar
 

Quote:
Originally Posted by Shan
This has to be one of the greatest threads in GS to date.

Shane




Quote:
Originally Posted by Paul Frindle on REP
SO if you use the system as a straight recorder (with well designed ADCs and DACs) it shouldn't be possible to produce an error of this kind and the sample value metering - although not perfect - will give an acceptable indication of peak programme level.
This is good news. Using PT's as a tape machine I've done more than a few times, mixing analog.


Quote:
Originally Posted by Paul Frindle on REP
But in practice it's risky since any contribution that gets processed after the ADC recording that introduces phase shifts, non-linearities or accentuates distortions that existed in the recording at upper mids or HF could result in a reconstruction error at higher contribution gains. In other words raising the levels of a 'troubled signal' may push it into the reconstruction error zone, where previously it was admissible.
I've experienced this before. I was working with some guitars that were recorded with the tubes going out and feeding back into the signal. I put EQIII on the signal and worked it a litte. It was still a bit harsh on the top end, so I turned on the lowpass filter just for kicks. I set it to 20Khz and noticed a HUGE difference in the sound. It didn't offend me anymore. I always thought it was the guitar sound giving me the problem. Apparently not.


Now I understand it all. It's amazing that it's happening. I really like Paul's "white noise" test. Can't wait to fire up the rig and see for myself.


"Now you know, and knowing is half the battle"
Old 6th October 2005
  #148
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djui5's Avatar
 

[QUOTE=raal]
Quote:
Originally Posted by Fredrik
if i may correct, it's all ideal 'til you start using fixed point plugs...

According to Paul, it dosen't matter. It's entirely a level issue.


to quote Terry

"Now everyone turn it all down a bit..."
Old 6th October 2005
  #149
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s0nguy's Avatar
 

OK... I'm anxious to try this... But I dont know how to trim the audio going in... I looked in my list of HD plugins and I dont see trim...

tell me now.... smack me later for being a dumbass.

Thanks,
S0nguy
Old 6th October 2005
  #150
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djui5's Avatar
 

Quote:
Originally Posted by s0nguy
OK... I'm anxious to try this... But I dont know how to trim the audio going in... I looked in my list of HD plugins and I dont see trim...

tell me now.... smack me later for being a dumbass.

Thanks,
S0nguy
Going into what? If it's going into the A/D's, turn down the pre-amp.
Or if you have a pre-eq type of thing, turn down the output. Or turn down your buss sends from the console, etc etc etc.


If you mean going into the plug-in, you'll have to bring down the fader. A lot of plug-in's don't have an input trim. If your sending to an fx plug on an aux track, turn down the send from the source channel.
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