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#1
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Techie Geek Question
I am studying how A/D's work and I have a 192 and a 96i, and was wondering about how the antialiasing filters work for those units.
a basic question: is the lowpass filter different for every sample rate you record / playback at? Meaning if I am at 44.1, the filter is a little lower than the nyquist, and as I go higher in sample rates, is it always a little lower than the nyquist, or is the antialiasing filter always just over the human hearing threshhold? I know there is a big difference with the different sample rates, but I guess what I am trying to figure out is, what is the fundamental differences between recording at 88.2 vs 96k, outside of the obvious mathematic difference when ultimately going to 16/44.1 --- And a side question, If anyone is using an api 4032 console, are you using the +4 or the -10 inputs? I was in a studio the other day where the board was clipping before protools and am wondering if thats what the problem was. Keny
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just to toot my own horn, I am in the process of relocating to Los Angeles and lookign for a job, so if anybody knows someone who can help me get established as an audio engineer and producer, check out my creds at http://www.kenyruyter.com and call me 310-909-9115 myspace page http://www.kenyruyter.com |
#2
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Re: Techie Geek Question
I can't say for sure how the converters actually work electronically, but digital theory says that the antialiasing filter should change as you change sampling rates. Maybe you can conduct an experiment with a 35 kHz sine wave created with signal generator, out of one converter and into another at 96kHz sampling rate. it should survive the trip, but not a SRC down to 44.1kHz or 48kHz.
Regarding gain staging, try to remember that 0dbVU should equal -18db FS on your converters (if that's how they're set up) I expect the API to be able to handle that much headroom, so maybe you're right about the -10/+4 issue, however, in normal practice, you shouldn't be anywhere near the peak of a Pro Tools meter anyway (excpet the master fader after the last plug in?), so the board channles should have been able to deal. |
#3
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Re: Techie Geek Question
From www.digidesign.com:
Frequency Response (192 I/O A/D): ±0.05 dB @ +2 dBu, 20 Hz – 20 kHz Frequency Response (192 I/O D/A): ±0.05 dB, -20 ', 20 Hz – 20 kHz; Frequency Response (96i I/O A/D): +0.1 / -0.2 dB @ -2 dBu, 20 Hz – 20 kHz Frequency Response (96i I/O D/A): +0.1 / -0.4 dB; -20 dBFS, 20 Hz – 20 kHz This would seem to indicate that the frequency response is independent of the sample frequency. If higher sample rates used higher frequency antialiasing filters, I expect the frequency response would also change. I would then expect Digi marketing to advertise that as a feature. So, I'm voting for a fixed frequency anitaliasing filter. |
#4
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Re: Techie Geek Question
It may be that the analog electronics in the converter can't reproduce those frequencies well, or at all.
Nonetheless, I'd be very surpised if the Nyquist filter was the same for all. This is a very intriguing question - any comment from Digi? Personally, I never record with rates over 48kHz anyway. |
#5
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Re: Techie Geek Question
*i'm currently editing this post, my conclusions from the first pdf are inaccurate
Ok well I did some more thought on this and I came up with this pdf. This is a texas Instruments data sheet for a 24 bit 192 khz A/D http://focus.ti.com/lit/ds/symlink/pcm1804.pdf In the internal circuitry of this a/d there is a lowpass filter. I am beginning to believe that earlier technology had the filters outside of the chips, but now chips are coming with the filters built in. But then this book I'm reading suggests they could also be recording a signal at twice (or a multiple of) the "recording sample rate" with a less drastic lowpass pre filter, then applying a digital post a/d filter, then reducing the sample rate to what is sent to the computer. HUH. Im reading this book called "Principles of Digital Audio" by Ken Pohlman. Its pretty damn intense, and I understand it when Im reading it, but after a couple days, i find it difficult to remember all 800 details of digital concepts per page. Geez. He mentions several types of filter circuits, and the whole oversampling and using digital filters. Wow. Intense.
__________________
just to toot my own horn, I am in the process of relocating to Los Angeles and lookign for a job, so if anybody knows someone who can help me get established as an audio engineer and producer, check out my creds at http://www.kenyruyter.com and call me 310-909-9115 myspace page http://www.kenyruyter.com |
#6
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Re: Techie Geek Question
hehehe.. That Pohlman book does get pretty heady. Another good old reference is Atkinson's "The Art of Digital Audio" which is now in its third edition, I believe. Basically, the filter should most definitely change. The whole point of having higher sample rates is that you can relax the low-pass filter and eliminate some of the side-effects of steeper filters, such as phase shifting. It would be completely possible to create a 192 kHz signal sampled through the same low-pass as a 44.1 kHz signal, but it would be completely pointless. By going to 192 kHz, you get a Nyquist frequency of 96 kHz, which is miles above what any animal could possibly hear. By doing so, you can place a gentler filter (less phase-shift) at a way higher frequency (whatever phase-shift remains is above any audible frequency). I did a quick search and could find nothing about such specs for Digidesign hardware. As stated above, the filtering is now done inside the converter, so it may be in the specs for the specific converters used, I didn't get there.
The filter frequency and steepness should change when switching sample-rates. If they don't, higher sample rates are worth nothing at all. |
#7
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Re: Techie Geek Question
I tried searching for "Atkinson's "The Art of Digital Audio"?
Is the author actually "John Watkinson"? I could find his book... http://www.amazon.com/Art-Digital-Au.../dp/0240513207 |
#8
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Re: Techie Geek Question
Quote:
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#9
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Re: Techie Geek Question
wouldnt that make the 192's frequency response range go to a little below 96k? and in that case i'm guessing they are doing the oversampling then using a digital filter, to make more headroom. (or something...)
__________________
just to toot my own horn, I am in the process of relocating to Los Angeles and lookign for a job, so if anybody knows someone who can help me get established as an audio engineer and producer, check out my creds at http://www.kenyruyter.com and call me 310-909-9115 myspace page http://www.kenyruyter.com |
#10
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Re: Techie Geek Question
Quote:
[image]http://www.eastcoastbands.com/img/snow07/fang.jpg[/image] Keny
__________________
just to toot my own horn, I am in the process of relocating to Los Angeles and lookign for a job, so if anybody knows someone who can help me get established as an audio engineer and producer, check out my creds at http://www.kenyruyter.com and call me 310-909-9115 myspace page http://www.kenyruyter.com |
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