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Author Topic: micro-timing in low sample rates  (Read 10509 times)

Graham Jordan

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Re: micro-timing in low sample rates
« Reply #30 on: October 04, 2005, 05:39:14 PM »

Pre-ringing doesn't tell you where it starts, it's simply a fact of 'brickwall' filtering. The less brick-wall, the less ringing. Also, if youuse a windowing function to 'smooth' the start of the tone you reduce ringing.

What I am saying is that the act of sampling has nothing to do with what you are trying to ask about - if sampling makes you 'lose' the accuracy of the location of the 'start' of the burst. The phase (alignment) of the sampling point makes NO difference to the final analog output of the incoming post-anti-alias filter signal. None!

You whole problem lies with the notion of defining the 'start' of the burst. You went on to talk about measuring the time between the starts of two successive bursts. The sampling process makes no difference to this.

The problem is that you're trying to line up the start of your 'ideal' burst signal with where you might say the 'start' is in a bandwith limited (i.e. real) system. A sampling system is a bandwidth limited system, the bandwith which is actually defined by the filtering before and often during the A/D conversion. Which has been (should have been) designed to perform within the noise limits, and the sampling rate, defined for the system.

The limit to pinning down the point at which the 'event' started (or ocurred) is actually being able to pin-point an indeal impulse, which is what I was meaning to get around to...
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Graham Jordan

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Re: micro-timing in low sample rates
« Reply #31 on: October 04, 2005, 06:14:35 PM »

OK, let's take the sine wave completely out of the problem, it's actually only obscuring the picture.

What we're looking at is placing a precise event in time. Since we started in 'theoretical land', let's keep it that way.

The 'event' to look for is a mathematical impulse. This is an infinitely thin spike, or pulse, that has a finite amount of energy. The energy is evenly spread across the whole spectrum (frequency). Real impulses have finite width, due to being bandwidth limited (by whatever physical situation they're in).

Let me first give a real world example (albeit electromagnetic - including radio frequencies): when you flick on a light switch, or otherwise create a spark, it is an impulse of current flow, which radiates out electro-magnetic waves over a broad band f frequencies. You radio (particularly AM) will pick this up pretty much whatever channel you're on and give you a click. Even if it's battery operated (no AC interference). This (somewhat) demonstates a real word impulse and it's broad band frequency content.

The thing about an impulse is that in time it is symmetrical so it is easier to see where it's center is - i.e. see where the actual event time is. To do this for the burst sine start, you have to remove the original signal to find the 'ringing', then find the center of that. The problem is you don't know where or how the sine starts - as that's what you're trying to find out.

Now I could do some more CoolEdit diagrams etc, but again, sampling has nothing to do with it. What you would do really is, pass this ideal impulse though what ever filter you like and see if you can locate the position of the original impulse.

I think what you'll find is that you can. Which I think ultimately answers the question - hopefully finally. I haven't looked into this much with real (analog) signals with real filters though, but the same argument should work.

If you can locate an impulse, you can locate the start of your 'ideal' burst.
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Graham Jordan

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Re: micro-timing in low sample rates
« Reply #32 on: October 04, 2005, 06:19:02 PM »

OK so I did some quick CoolEdit diagrams...

This first one is acting as the original 'analog' signal. These are two digital impulses, spaced 64 samples apart with 96kHz sample rate (sample rate is actually irrellevent).

index.php/fa/1664/0/

Note again the ringing on both sides of each impulse as this is a bandwidth limited signal.

Now for the next image....
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Graham Jordan

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Re: micro-timing in low sample rates
« Reply #33 on: October 04, 2005, 06:26:56 PM »

This image is the same signal and view, except it has been downsampled (with highest quality setting) to 44.1kHz. So none of the samples line up.

How far apart should we expect the impulses to be (going from peak to peak)? A quick bit of math leads you to the answer 29.4 samples.

index.php/fa/1665/0/

Looking closely in cool edit (can't see as accurately on this smaller posted image) it appears to be dead on 29.4 samples apart. Note that neither peak aligns with a sample point; both are between samples.

I really hope this answers this thread, at lesat for the most part. It should also but to bed any lingering doubts that the sampled data cannot know what happens between samples.
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Graham Jordan

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Re: micro-timing in low sample rates
« Reply #34 on: October 04, 2005, 06:29:16 PM »

Oh, incase you're wondering why the second impulse is smaller then the first, it's because the bandwith has been reduced (due to the anti-aliasing filter to go down to 44.1K), so some of the energy has been taken out - so is a smaller signal.
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blueboy

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Re: micro-timing in low sample rates
« Reply #35 on: October 04, 2005, 07:00:50 PM »

Thank you very much Graham.

This is very helpful for those of us that need to "visualize" an otherwise very complex subject.

Regards,

JL
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Yannick Willox

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Re: micro-timing in low sample rates
« Reply #36 on: October 05, 2005, 05:01:51 AM »

Graham,

great pics indeed.
While you're at it, maybe you could post a comparison between an ideal spike, the 44.1K sample of it, the 96K sample, the DSD sample - BUT in an non idealised situation meaning WITH the 40-50KHz filter on playback !!! - and a true analog scenario bandwith limited to 25 KHz ?
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Yannick Willox
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Graham Jordan

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Re: micro-timing in low sample rates
« Reply #37 on: October 05, 2005, 01:02:57 PM »

My time on this topic is done Smile

An ideal spike is basically infinitely thin (short) and infinitely steep, but has energy. Imagine the thinest vertical line rising from the base line . No matter how much you zoom in, it never looks any fatter.

I've already posted the 96kHz and 44.1kHz sample versions. they are basically the same, but the 44.1Khz version looks wider. Doens't matter what sampling system you're using, it's width is defined by the bandwidth available. It's actual shape for a real filter/system is defined by (and defines) the exact frequency response (as I said before). This is known as the impulse response. The impulse response of a filter is exactly interchangable with the frequency response. So for an analog filter, you can do this mathematically. Also if you know the complete frequency/phase responbse you know the impulse response. I'm not going to go into how different filters differ in impulse response, that a very large mathematical/engineering topic. Maybe someone else you add something, or look up a book on filter theory. There are even websites and software programs on filter design that will show you the impulse response of a particular filter.
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blueboy

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Re: micro-timing in low sample rates
« Reply #38 on: October 05, 2005, 04:57:06 PM »

Graham...to go slightly off topic for a minute...

Do you (or anybody else) know if the display mode you are using to visually reconstruct the sampled wave is exclusive to Cool Edit/Adobe Audtition? Which other editors have this capability?

The reason I am asking is that there was a thread recently about the importance of setting levels in DAWs. The problem as I understand it is that level meters in these programs do not fully reconstruct the wave when indicating levels, so even though the meters were not peaking, distortion was occuring during the D/A conversion when the "hot" signals were reconstructed in the converter.

It would be interesting to "see" if this peaking shows up in an audio editor.

Regards,

JL
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Graham Jordan

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Re: micro-timing in low sample rates
« Reply #39 on: October 05, 2005, 05:20:12 PM »

It does show up in CoolEdit. Don't know about other editors. Well designed A/Ds should be able to handle this. To avoid this possibility, keep peak levels at least 2dB below 0dBFS.
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C-J

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Re: micro-timing in low sample rates
« Reply #40 on: October 05, 2005, 05:23:51 PM »

Graham Jordan wrote on Wed, 05 October 2005 01:26

Note that neither peak aligns with a sample point; both are between samples.
I really hope this answers this thread, at lesat for the most part. It should also but to bed any lingering doubts that the sampled data cannot know what happens between samples.

Yeah, that was just what I showed in my first graph with the two sine waves. The first wave had sample points on both sides of peaks and troughs, while the second one had points exactly at peaks and troughs. So anything in between is possible...

My best,
C.J.
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C-J

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Re: micro-timing in low sample rates
« Reply #41 on: October 05, 2005, 05:26:29 PM »

Graham Jordan wrote on Thu, 06 October 2005 00:20

It does show up in CoolEdit. Don't know about other editors. Well designed A/Ds should be able to handle this. To avoid this possibility, keep peak levels at least 2dB below 0dBFS.

Or even better, use an oversampling peak meter! Wink

C.J.
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blueboy

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Re: micro-timing in low sample rates
« Reply #42 on: October 05, 2005, 05:58:36 PM »

C-J wrote on Wed, 05 October 2005 14:26

Graham Jordan wrote on Thu, 06 October 2005 00:20

It does show up in CoolEdit. Don't know about other editors. Well designed A/Ds should be able to handle this. To avoid this possibility, keep peak levels at least 2dB below 0dBFS.

Or even better, use an oversampling peak meter! Wink

C.J.



I'd rather just keep my levels in check than use the extra DSP required (or spend the money) for an oversampling meter (i.e. TL Mastermeter).

I'm not crazy about Cool Edit / Audition as an editor either, but the antialiased waveform view is cool. Apparently neither Wavelab nor Soundforge offer this display option. Sad

Maybe Audacity does??

Regards,

JL
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C-J

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Re: micro-timing in low sample rates
« Reply #43 on: October 05, 2005, 06:41:50 PM »

blueboy wrote on Thu, 06 October 2005 00:58

I'd rather just keep my levels in check than use the extra DSP required (or spend the money) for an oversampling meter (i.e. TL Mastermeter).

I use RME's Digicheck software, which taxes the DSP of the RME card only.

Quote:

I'm not crazy about Cool Edit / Audition as an editor either, but the antialiased waveform view is cool. Apparently neither Wavelab nor Soundforge offer this display option. Sad
Maybe Audacity does??

I don't know about the others, but no, Sound Forge does not have it. I would however rather call it a reconstructed waveform view, than an anti-aliased.
BTW, have you read Nika's great pdf about digital peak meters? Here:
http://www.cadenzarecording.com/papers.html

C.J.
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blueboy

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Re: micro-timing in low sample rates
« Reply #44 on: October 05, 2005, 07:55:47 PM »

C-J wrote on Wed, 05 October 2005 15:41


BTW, have you read Nika's great pdf about digital peak meters? Here:
http://www.cadenzarecording.com/papers.html

C.J.

No I hadn't, thanks very much C-J. Looks like some interesting reading...

To get back on topic...and maybe I shouldn't do this but...

Just to clarify my understanding of this topic as discussed so far:

The question was whether or not a sound (or pulse) that occurred between sample periods at a given sampling rate would be shifted in time to accommodate that sampling rate and therefore sound "out of time" after being sampled and then played back.

The answer is that any pulse that was "manufactured" to start and rise to full amplitude within that given sample period would have to contain high frequencies, and those frequencies would exceed Nyquist and therefore would be filtered out anyway. In addition, real world sounds don't have such clearly defined "start times" so the whole exercise is essentially theoretical.

Graham's diagrams also showed that even after sample rate conversion, the reconstruction of waveforms was correct, even though the sample points had shifted their relative positions on the waveform. The reconstruction filters simply "knew what to do" to fill in between the original sample points.

Although I may have grossly over-simplified things...is this the conclusion, or am I misunderstanding anything?

What I'm not sure about then is this:

I was reading this older article by Julio Alvarez & Richard Elen from Apogee Electronics regarding sample rates and came across this statement regarding our perception of a sounds arrival time.

Quote:


Having established that higher sampling rates are a good idea – or at least a fact of modern life – there is question as to what the sample rate should actually be in studio environment. On the face of it, 96kHz takes care of capturing any audio that might ever happen, and 24 bits offer quite enough quantization steps. Is that enough?

Yes, in theory – more than enough. But there are some potential problems, real or imaginary, to having a production environment that has no better resolution than the consumer distribution format, and the emerging DVD-Audio standard offers not just 24-bit, 96kHz sampling: It even goes beyond that to support 192 kHz sampling in stereo.

[On the face of it this is quite absurd. Do we need to capture “audio” signals at up to 96 kHz? Obviously not – such signals don ’t exist. However, some recent research suggests that the human brain can discern a difference in a sound's arrival time between the two ears of better than 15 microseconds – around the time between samples at 96 kHz sampling – and some people can even discern a 5
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