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I have just received files from a fairly reknowned studio that I will produce overdubs for and then mix. It was recorded with the new HD pro tools system at 48k and 24 bit. but the levels are amazing low. I remember someone telling me that when recording to DAT machines the lower the level also translates to a lower bit rate. The higher the level the better the bit rate. In other words, if the LEDs are barely moving the bit rate cab be as low as 8 bits. Is this correct for a Pro tools system???

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I believe it's 1 bit less for every 6 db down.

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Quote:
Originally posted by Allan Speers:
I believe it's 1 bit less for every 6 db down.
Correct,

And normalising afterwards doens't help.
The harm is done.


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This is what I was afraid of. What is funny is how long we went back and forth about wether to do this in 96k or not. The fore mentioned engineer thought it was imparitve to record in 96k since we want to mix to SACD. We still have some more tracks to cut so we can fix that at least. Thanks for your immediate replies.....

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No.

If you record to your DAW or MTHDR at -40dB @ 24bit/48kHz, and looked at the charactristics of that file, it would still be a 24bit file.

Recording at lower levels does not "lower" the bit rate. It (bit rate) stays unchanged. What you've done is not utilized the dynamic range provided by your system, and, you've recorded closer to the noise floor.

Upon inspection, the file would still be a "24" bit file.

Do you "lose" sample frequency if you limit your recordings to a maximum of 8kHz (ie - you cut evrything above 8kHz off with a 24dB per octave filter)?

No, the sample frequency is the same. Your bandwidth may become constrained.(edit: it may sound dull or muffled, or Lo-Fi).

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Thanks Drew, That is the way I understood it. It just surprises me how nay one could record such a low level. Maybe they were thinking that it is digital and the only reason to have hi levels to Analog was to cut down the noise ratio. I don't know. I will definetely sort this out the next time around.
Thanks again,
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where is nika when we need him? \:D

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Quote:
Originally posted by jindrich:
where is nika when we need him? \:D
Oh, he's probably off somewhere writing a book on the subject ;\)


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Argh

OK. Drew, I really like your statement there. It's an interesting way to shake-up the semantics on the subject. I like.

Fredo,

You say that Normalizing "doesn't help." Doesn't help with what, exactly?

Liefski,

I'm not going to let you off the hook easily. You'll have to think about this one. But for starters I'll agree that the lower you record the fewer bits you're actually using to encode your signal (while you may communicate it with 24 bits, you're only actually toggling 16 of them.)

Now, to think about this more thoroughly, a "bit" in digital recording gives you what? If you increase from 24 bits to 25 bits what do you get? It will be tempting for you to use the word "resolution" in your answer, but don't. There are a few characteristics of a waveform (frequency, amplitude, etc.) and "resolution" is not one of them. So what characteristic of a waveform do you improve when you add another "bit"?

BTW, SACD is not directly compatable to 96kS/s sample rates. You'd be better off going with 44.1kS/s or 88.2kS/s. Do the math - 2.8224MS/s (the sample rate of SACD) divided by 96kS/s equals what? Is it even math? How do we upsample something an uneven number of multiples? What are the sacrifices?

Nika.

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Leifski,

There's nothing really wrong with recording at fairly low levels. In fact it often makes for a better mix, if you're mixing inside the DAW. By the time you add a ton of EQ, a ton of compression, and other junk that everyone throws into a mix nowadays, you're slamming the 2-bus and that sounds like crap. Then if you lower the faders on the individual tracks to keep from slamming the 2-buss it also sounds like crap.

Recording at lower levels leaves you some headroom so you don't have this problem... at least, not as much... \:D

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Ok, you are right about me using the word "resolution" and I also agree that too high of levels on the output bus is not a good thing. But these tracks are recorded so low they are barely visible. There is a middle ground. I am also very new to SACD. Where can I learn more about it??? These tracks will be mixed down to half inch tape so it will leave the digital domain for a brief moment in time so the sample conversion process is bypassed...
Thanks again guys...

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Quote:
Originally posted by Lee Flier:
By the time you add a ton of EQ, a ton of compression, and other junk that everyone throws into a mix nowadays, you're slamming the 2-bus and that sounds like crap.
Whoah whoah whoah whoah whoah.

"Slamming the mix bus?" What in the world does that mean? Everything that goes to the "mix bus" is simply added together, like adding up the values in your checkbook. Why is it that adding together bigger numbers (louder levels) would result in a less accurate answer? Try it on the calculator and see if big numbers don't add together as well as small numbers. Remember that the "calculator" in a lot of these applications is 56 to 80 bits. You can send full scale levels from something like 64 channels at a time and still not overload the mix bus, so forget that idea. So why is it that hitting the mix bus hard is not as good as hitting it quietly?

Liefsky, that one's for you, too.

Nika.

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Recording at lower digital levels (under -12 to -18dbfs) results in incomplete usage of the available bit rate, resulting in inferior frequency response. While the files still remain at the bit rate set for the session, they are in fact utilizing less than the session allows.

Hope this is helpful.

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Hi Nika,

I've slammed many a mix buss in the same way Lee describes in her post on an analog desk and in PT. I'm not sure technically why it happens but I've damn sure heard it!

Thanks,
Ted.

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I think the concept behind that little thingy (very discriptive, huh?) as far as the lower levels is that, if you have some number of tracks that peak at zero and have a high RMS, then you will, at least in my applications, overload your mix bus. Think about this - fly in a tune, copy it, then play the two together without lowering the volume. So, to compensate, you lower the volume of the tracks which results in the decrease of number of bits used per track. Unless your application dithers up to 24 or 32 or something (which makes little sense to me), then I can see where this could become an issue.

Of course, I could be completely wrong - but that would be the advantage of taking every digital track out individually into a good analog mixer.

Um, yeah. I think that's it.


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Quote:
Originally posted by where02190:
Recording at lower digital levels (under -12 to -18dbfs) results in incomplete usage of the available bit rate, resulting in inferior frequency response.
Where,

This is incorrect. Perhaps you meant to say something else? Inferior frequency response At 18dBfs? I can reassure you that your HDR records/samples at (pretty much) all levels (probably down to to the -80's (dbFS) with the same frequency response +/- a whole dB if that much.

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Quote:

"Slamming the mix bus?" What in the world does that mean? Everything that goes to the "mix bus" is simply added together, like adding up the values in your checkbook. Why is it that adding together bigger numbers (louder levels) would result in a less accurate answer? Try it on the calculator and see if big numbers don't add together as well as small numbers. Remember that the "calculator" in a lot of these applications is 56 to 80 bits. You can send full scale levels from something like 64 channels at a time and still not overload the mix bus, so forget that idea. So why is it that hitting the mix bus hard is not as good as hitting it quietly?

Liefsky, that one's for you, too.

Nika.[/QB]
Nika-san,

With absolutely no experience with Pro Tools whatsoever, I can't make any statements about slamming a 2 bit - err 2 mix bus. However it would appear that there are some systems (analog and digital) that do a horrible job of summing 30, 40 whatever number of signals together in a coherent fashion.

Is it phase shift? Or just poor summing - this is outside the scope of discussing calculators and how numbers are added within a given digital system.

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Quote:
Originally posted by NYC Drew:
Quote:
Originally posted by where02190:
Recording at lower digital levels (under -12 to -18dbfs) results in incomplete usage of the available bit rate, resulting in inferior frequency response.
Where,

This is incorrect. Perhaps you meant to say something else? Inferior frequency response At 18dBfs? I can reassure you that your HDR records/samples at (pretty much) all levels (probably down to to the -80's (dbFS) with the same frequency response +/- a whole dB if that much.
Right. We should know that bit rate = dynamic range and sample frequency = frequency range.


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Lower level D-A conversion does nto take advantage of all the available bits. Less bits, lower high frequency response.
Hope this is helpful.

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Explain yourself, young man!


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Quote:
Originally posted by where02190:
Lower level D-A conversion does nto take advantage of all the available bits. Less bits, lower high frequency response.
Hope this is helpful.
Wordlength has nothing to do with freq response. Bit rate dictates dynamic range only. Period.

Fs (sampling freq) dictates freq response.


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So my question is; How do the 'low level pro tools project' tracks sound anyway?

I have gone to accept Nika's lead on this. It goes directly to the question; Does the lower end of digital dynamic range have less fidelity than the top? If the resolution per db-range(if that's the right term)is the same at both ends, then it comes back to dynamic range, and how far down are the lowest details that can be heard.(Please, I pose this as a question and for my own understanding.)
I recorded some simple acoustic music at -70-80 once. It came back noisy as hell, but not muted or raspy.
Sorry about the ramble.


Part time -long time.

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Quote:
Originally posted by where02190:
Lower level D-A conversion does nto take advantage of all the available bits. Less bits, lower high frequency response...
Where (Nick),

Can you quote a text/white paper on this? Where did you learn this? I'd like to have a word with the head of the faculty there....

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Now I'm still confused. Sometimes when using direct outs to my ADAT during a live recording, I have to keep the input level down to avoid clipping. Thus, the track is recorded at a fairly low level. Usually the meter won't make it to the mid point. I have recorded those tracks to the DAW for mix and mastering and have not noticed a lack of fidelity.

So..does recording to digital media at low levels have an effect on the recording quality? And what is considered a low level? KISS...


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Quote:
Originally posted by GZsound@hotmail.com:
Now I'm still confused. Sometimes when using direct outs to my ADAT during a live recording, I have to keep the input level down to avoid clipping. Thus, the track is recorded at a fairly low level. Usually the meter won't make it to the mid point. I have recorded those tracks to the DAW for mix and mastering and have not noticed a lack of fidelity.

So..does recording to digital media at low levels have an effect on the recording quality? And what is considered a low level? KISS...
What you are experiencing is most likely impedance mismatching, and or level mismatching.

Check your equipment (read the specs, and run tests with tones and a volt meter) to ensure that level "X" on your mixer corresponds to level "X" on your ADATs.

At the risk of sounding like a goddamn elitist gear snob, do you guys read the fucking manuals and have a basic understanding of electronics/levels etc?

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Quote:
Originally posted by GZsound@hotmail.com:
So..does recording to digital media at low levels have an effect on the recording quality? And what is considered a low level? KISS...
Think of it like this - you can record with peaks at -24 dbfs and yer still recording a 20 bit word...

If it sounds good, don't sweat it. Done well, 16 bit can sound amazing.


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This is my understanding

If you record at low levels at a fixed point bit rate, the lower the level, the less amount of points that can be used to describe the amplitude of a sine wave.

For instance, at full 16 bit resolution, you have 32768 points to describe the amplitude, and at 24 bit, its around 16 million. But if you record at a really low level, you USE a fraction of those points. IF you use 8 points to describe a sine wave, you get jaggies along that sine wave. When you normalize that file then, it'll still have those jaggies, but will take up the full amount of points your bit rate has to offer. Those jaggies will crate high frequency content because of the sharp edges (like square waves have infinite high frequency harmonics because of the sharp edges) THis high frequency content is called Quantenization noise (i think i spelled that wrong)

This has nothing to do with the theoretical frequency range of a sound file. The theoretical frequency range is determined by the sampling rate that you recorded in... 44.100KHZ will give you a theoretical frquency range of 22.05KHZ. I say theoretical because all AD and DA converters have lowpass filters in them to stop higher frequencies then the Nyquest rate from passing. If there wasn't any lowpass filters and a frequency passed higher then the Nyquest rate (1/2 the sampling rate) you would get what is called Foldback. Foldback is basically when a frquency gets to high for a sampling rate and cannot be described. The points are then recognized as a lower frequency then it actually is and is rendered as that lower frequency (there is a formula for it, but i can't remember it off the top of my head)

BY the way, if you mix and record in floating point instead of fixed point, alot of these problems with BIT RATE goes away.

I hope this helped.
Marshall

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btw--- the jaggies i was describing are called aliasing
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"Think of it like this - you can record with peaks at -24 dbfs and yer still recording a 20 bit word..."

Brad, if Marshall is right about the "jaggies" (whatever you actually call them) then there would indeed be an advantage to recording fairly hot, even with a 24-bit system. No matter what the peak level, the level of the upper harmonics, and of delicate room ambience is always significantly lower. That's the audio we need to be concerned with.

Some have reported that recording a 24-bit signal at very low levels sounds fine. Why then does a track that is converted to 8 bit (even 12 bit) sound so edgy and, um, lacking in resolution (did I use the "R" word? Oops) Why did the early 12-bit samplers sound the same? Is it possible that the accepted "1 bit lost for every 6db down" only refers to dynamic range, and not to ALL the deficiencies of lower bit-rates?

Surely there is SOMETHING gained from higher bit rates than just dynamic range?

The question then becomes, how many db's down ARE the harmonics and room ambience from the peaks in a typical track, and just how many bits down can you go before there is audible degradation?

My gut feeling is to agree with you that a -24dbfs peak @ 24-bit is completely acceptable, but I wish i had some hard numbers to look at.
====================

BTW: Nika, with a PT Mix system, you most definitely CAN overload the "mix bus." I used to do it constantly, and it sounds pretty bad. Digi has always maintained that you can simply pull-down your master (2-mix) fader to solve this. This is not true. When you do that, the distortion goes away, but "something" is not right with the sound. I don't care what Digi claims, I did very careful tests and am sure. You must lower the individual faders, much like with an analog console.

On the HD system, however, there has been a SERIOUS improvement in the mixer headroom. This is a major reason that I upgraded to HD.

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Quote:
Originally posted by where02190:
Lower level D-A conversion does nto take advantage of all the available bits. Less bits, lower high frequency response.
Hope this is helpful.
OK. Let's get started. This one just plain isn't right. It's not close to right. It's really, really not right. Moving on...

Nika.

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