John Curl's Blowtorch preamplifier part II

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um that TI DSP paper is way dated - current desktop PC processors run rings around that era's dedicated DSP chips

my nearly 6 year old desktop has Quad Core2, 2.4 GHz, 8 Mbyte L2 cache


http://www.nasoftware.co.uk/home/attachments/018_PPC_Intel_comparison_whitepaper.pdf

gives more recent Intel PC CPU DSP times, it is even a few years old - points to the current generation's PC CPU's doubling of DSP instruction bit width...

...got to buy me a new PC - its a shame only embarassingly garish gaming PCs use the better CPUs today off the shelf
 
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Well Scott, it seems to me that you have gotten to level 2. You know: It exists, but it's not important.
What Barrie's article, perhaps 15 years after Otala's AES preprint, did, was re-awaken interest in PIM.
What I would like people to understand is that people like Matti Otala are interested in WHY amplifiers sound the way they do, and Matti found that TIM was NOT the complete answer. New mechanisms had to be found to explain what was going on.
In 1978, Matti mentioned 'differential phase' as something to look into, and he wanted to pursue his series of AES papers in that direction.
You see, we, at that time, and even some of us, today, were dissatisfied with the sound quality of most amps and preamps at that time. Sometimes a really good tube amp seemed to fill the bill (as it sometimes does today), but solid state had a ways to go.
Both Matti and I (both together and independently) built very fast, wide open loop power amps in 1980. Matti did an amp for HK, and I did one for VMPS. Both were pretty good, but 'over the top' cost-wise, and neither was 'perfect' because we just did NOT know enough about materials (for instance) to make a 'perfect amp.
The IC of the day was the 5534, and I have to admit that it was a big improvement over previous IC's used for audio, BUT we still found that we could make discrete designs, better sounding.
We were always 'nagged' by: When we make low feedback, high speed amps, that they usually sounded 'better' than many stock amps with an IC in the front end and all kinds of negative feedback, making the MEASURED audio tests acceptable.
In all fairness, the PIM controversy fell away, until Barrie Gilbert's article. And I was amused that Barrie actually gave grudging acceptance to Matti Otala's work on TIM, as well. This opened up the controversy that continues today.

John,

The problem may be that those amplifiers sounded good for other reasons, not related to PIM. If you had measured their PIM, you might have found that they did not have any lower PIM than amplifiers with more NFB.

Barrie's paper did not really add anything to my paper on TIM, and did not contradict it. Barrie's paper just did not show explicitly that the PIM was dependent on movement of the closed loop gain pole, and he did not show experimental result from a real amplifier with and without large amounts of feedback.

My paper is here, and its explanations are true to this day:

http://www.cordellaudio.com/papers/phase_intermodulation_distortion.pdf

Any nonlinearity that moves the closed loop bandwidth around as a function of signal will generate PIM. When it involves NFB, usually it is amplitude modulation of the open loop gain that moves around the closed loop pole, thus effectively resulting and an amplitude-to-phase conversion (good old differential phase). The open loop bandwidth and the LF open-loop gain have virtually nothing to do with the magnitude of this effect.

Even in an amplifier with NO NFB, there are mechanisms that create signal-dependent bandwidth and phase.

PIM exists, it may be audible. It is virtually impossible for PIM to exist in the absence of other distortions, like THD20 and 19+20KHz CCIF IM.

Cheers,
Bob
 
This comment is not really appropriate except as a joke maybe? The time/frequency domain transform pair contains no thrown away or hidden variables and is perfectly reversible. They are EXACT representations of the same thing.

They certainly are, depends what you want to explore. Large FFT highly averaged is nothing to call home about in case you need to examine non-periodic phenomena with random occurence.
 
Once you've captured it in time domain (and the techniques for doing that are well known), it's trivial to convert to the frequency domain. And the reverse as well, of course.

Richard was faster.

And incorrect as well. Transformations from time to frequency domain do not have to be limited to harmonically related phenomena nor are they limited to periodic signals. That's been explained in this thread alone at least a dozen times.
 
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One point has been thoroughly made, over and over... the math of the FFT in all directions is correct. Thats a comfort to know [tongue in cheek]. Now can we get to what we actually have to test with? Amps (power) could use a transient stimulous and capture it and FFT it and all. Dont bother to tell me yet again that it can all be done with sine waves or deconvolved back and forth and broken down etc. Or that its done all the time on a super computer. Thats all fine and good for the FFT math and computer SIM guys doing R&D on a grand scale.

What can be done with a PC that is real time using 3D FFT's on a PC that is at least 10% accurate (+/- 1dB) to -100dB (or below)?
A transient waveform [and capture?] is the test stimulous for exciting a power amp.

Thx-RNMarsh
 
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Amps (power) could use a transient stimulous and capture it and FFT it and all.

It's 2013. We have marvelous things like impulses and MLS, even in cheap software. And way better than 1dB resolution at -100dB or below. No supercomputers needed.

Guess what? Amps act exactly like you'd expect them to, at least amps designed by real engineers (as opposed to former hifi salesmen, Japanese gurus, and 1970s era fossils with "philosophies").
 
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It's 2013. We have marvelous things like impulses and MLS, even in cheap software. And way better than 1dB resolution at -100dB or below. No supercomputers needed.

Guess what? Amps act exactly like you'd expect them to, at least amps designed by real engineers (as opposed to former hifi salesmen, Japanese gurus, and 1970s era fossils with "philosophies").

HAHAHA Cute.

I have all that stuff too. Even bought the first MLS made by DRA Labs. isnt good enough. Anything else?

Thx-RNMarsh

The new LTC2378-20 might be a start in the right direction? Or, what else?
 
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Oh, find something wrong with my amps, SY? Tell me more!

I've never used one of your amps. Odd that you would think that I was talking about them. What's the part of their performance that you think would show problems in a simple impulse test?

BTW, the amp impulse response is routinely taken during speaker measurements. So it isn't like this is an exotic measurement or that normal amplifiers show odd behavior.
 
Now FFT has come up again, it reminds me of the last time it surfaced on this thread.

On the issue of the perfect reversibility of the transform between time and frequency domains, I many posts ago made the rather contrived joke that this was only true if one would have the mathematical formula for random noise. I still can see it flying over all heads. The point was of course that the statement of perfect reversibility is only true for discrete FFT. This may have some relevance for audio measurements, since it requires sampling of the measured signal, with an unavoidable associated loss of information. Also, a discrete FFT only produces data, to get information out requires trade-offs.

I mention it again because of what Pavel and Dick are posting. Thought experiment: what is the FFT of a Kaiser Wilhelm sine? With this I mean a sine with a spike on the upper-half (not to be confused with its mirror image, the wineglass sine).

Edit: I remember Scott posting at the time of the last discussion on some measurements that were developped to, if I remember correctly, identify a peculiar kind of one sided crossover distortion. Pretty much a Kaiser Wilhelm situation if my memory doesn't play tricks on me, but I didn't have time to read it with sufficient attention at the time. Could you please post again?
 
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The new LTC2378-20 might be a start in the right direction? Or, what else?

A decent PC, something basic in a sound card (I use an M-Audio 192), basic software like AudioTester (you can get fancier signal generation capability if you want to create oddball stimuli, then use the FT analysis in Audiotester to measure the output). A simple analog interface box to get levels and impedances correct for what you're measuring. Really, that's all you need. This has enormously better capability than the ultra-expensive stuff I used to work with when I was at Nicolet.
 
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