J.J. Blair wrote on Thu, 19 October 2006 21:58 |
Jimmy, you keep disagreeing about things we agree on. You keep missing my point that a DAW's visual rendering of a waveform is not showing you what an oscilloscope would show you. I'm talking about the information that a DAW, or in my case, ProTools, is capabale of showing you about a waveform.
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JJ, I'm with Jimmy here, but I think I see where you're coming from. This is a bit long, and it'll either clear things up or stir the mud some more ...
A DAW's rendering of the waveform should show the same thing one would see on an oscilloscope. Zoom in on the DAW so you can see individual samples and it's obvious.
(Aside: I have a .WAV file of the closest we can get to the Kronecker delta function. The delta function is infinite at time zero and zero at every other time. Since we can't have an infinite pulse height, the sample at time zero is 0x7FFFFF, the most-positive value we can represent with 24 bits. If I view this file in CoolEdit, and zoom in on the samples around time zero, you see what you expect: the sample at full scale, followed by the samples at zero. Furthermore, you see what you expect if you're familiar with sampling theory: rather than straight lines doing connect-the-dots, the samples are connected with a sinc (that is, sin(x)/x) waveform. That's right: CoolEdit gets it right. And if I wasn't lazy, I'd see how ProTools, SAWPro and GarageBand display this file.
And if you play this waveform and look at the result on the 'scope, you see a sinc response. Life is good when theory and practice agree.)
There's another rub, though: as you zoom out on the DAW, you get to a point where you have more samples than pixels in the waveform window. The DAW authors have to decide how to handle this: simply drop samples, or average samples, or show only the peak sample within the interval to display, or show the envelope, or whatever. So the crux of this biscuit is that when you zoom out, you
can't display the waveform exactly. And an oscilloscope, whether digital or analog, has to handle the same situation. If the DAW doesn't handle this scenario in the same way that your 'scope does, then of course there will be a difference in the display!
Try playing with a 'scope, varying the horizontal scale (time per division) to match the DAW's time display. See if they're similar or different as you zoom out.
Basically, if you zoom in enough so that you can see the actual samples, you can time-align your tracks. It's real easy to see a snare hit or click and line things up to the peak.
It's worth noting that neither the DAW nor the 'scope show phase information, because without a reference, phase information is meaningless. What's useful is when you compare two signals to get the relative phase between them, which Jimmy mentioned.
Now, a thought about the original topic, which is can one mic (say one that's close up to an amp's speaker be delayed to match the time-of-arrival to a room mic, and will they be time-aligned? The answer is yes, of course, and in fact that's exactly what we do when we're using a transfer-function measurement tool (like Smaart) to measure the response of a sound system. You have the reference signal, a direct feed of the test signal, and you have the measurement signal from the mic in the room, and you do an impulse response to learn the delay between the two. You then insert that delay on the reference input, and the result is that the two signals line up in time.
So, yeah, you could measure the distance between the two mics, insert a delay on the close mic, and they'd be time-aligned. But (and this is the preaching-to-the-choir part of our show) the two mics will still not sound the same, and the reason is obvious and doesn't need advanced math to understand: the two mics aren't picking up the same signal source! It should be obvious that the close mic picks up pretty much only the direct signal (high S/N) and the distant mic picks up the direct signal plus room reflections. When you delay the close-up signal, you just move that signal back to line up with the strongest signal at the distant mic (which should be the direct signal), but you haven't added any of the multipath that gives the distant mic its character (and it's this multipath that is the cause of comb filtering). Yeah, you've reinforced the direct signal, which may or may not be what you want.
If you were in an anechoic chamber, where the reflections are nil, then delaying the doesn't buy you anything. But there's no point in putting a room mic in an anechoice chamber ...
-a