DAC Filtering - the "Rasmussen Effect"

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Please re-read the last two words of the below excerpt taken from my post #26:

"However, it once had a 2-pole LC filter, but it sounded better to my ears without any analog filter - until now."

OK, so we have 2 different filters that produce a different sound. Nothing really shocking here.

Sigma-delta DAC chips already feature an integrated low-pass reconstruction filter. It's called an oversampling digital interpolation filter, and has technical performance far superior to that of any practical analog reconstruction filter.
An oversampling digital filter shifts the HF noise and aliasing products to higher frequencies, making the job of the analog filter easier (you get buy with a much less steep analog filter), but you still need an analog filter.

rather than first seeking to debate sigma-delta conversion theory with me, may I suggest you take the simple step of trying the technique for yourself?
You may, but as I don't have any DACs available that don't have the required filter, it is rather pointless.
 
putting simple 6db filter ditto on output of chip isn't the same as putting filter on output of I/V stage

while later one is easy to understand /analyze , first one isn't easy , because we really don't know DAC chip output section in details
We don't need to. If the DAC is a current source, the pole will be dependent on the resistance of the equivalent resistance and the capacitance.

If a single pole filter is placed after the I/V, the results will be the same.

The only difference would be due to interactions with the DAC not behaving with a capacitive load. (If the I/V is perfect, the input impedance zero, the cap doesn't even have an effect!) That's going to be negligible unless we do something to upset the DAC by placing a load that it's not designed to drive on it.
 
This isn't blindly theorizing. This is 200 level electronics. You remove a filter, you alter frequency response. We've known this for longer than I've been alive. I suppose next you're going to suggest I use a yellow highlighter on my CDs, because I haven't tried it yet?

Here's what I will suggest. If you won't first try this rather simple experiment, that's your prerogative, but do not waste your time (or mine) commenting in this thread? There is a difference between healthy skepticism and closed-mindedness.
 
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OK, so we have 2 different filters that produce a different sound. Nothing really shocking here.

An oversampling digital filter shifts the HF noise and aliasing products to higher frequencies, making the job of the analog filter easier (you get buy with a much less steep analog filter), but you still need an analog filter.

No, you don't a specific analog filter circuit. The oversampled alias images are low-pass filtered by interconnect cable capacitance, by amplifier bandwidth limits, by loudspeaker bandwidth limits, and by human hearing bandwidth limits. The only potential technical problem due to high oversampled alias products is the potential for inter-modulation distortion, which would be quite low in level at any rate due to the zeroth-order hold operation of most DAC chips, which inherently overlays a SINC function shaped frequency suppression mask on top of the image products.[/QUOTE]

You may, but as I don't have any DACs available that don't have the required filter, it is rather pointless.

Julf, see my post #46 comment to FoMoCo, as it applies to you as well.
 
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... Why you are wasting your own time (and mine) by commenting in this thread.
Pardon me great master of all that is rubbish. For I did not mean to waste your time with simple textbook theory that infringes on your ability to conjure something from nothing and call it anything you please. You do have the right to call anything by any name you please. Science shall not trod upon that right.

There is a difference between healthy skepticism and closed-mindedness.
You are correct. Your closed mind refuses to accept that there is nothing special (beyond a simple low-pass filter) here, even though there are stacks of books and papers that will prove you wrong. Sit down and analyze the circuit presented and (Shazam!) you'll see nothing but a simple single-pole RC filter composed of the added capacitor and the parasitic elements of the circuit.

Of course, Zen Mod makes the point that we don't know the output stage, and that can indeed make a difference, but as long as we don't violate the constraints that the manufacturer of the DAC gives, we still have a simple low-pass filter. If we do violate those constraints, we most likely have a stability problem. Possibly some oscillations. But, most current source outputs of DACs behave just fine with a capacitive load.

Perhaps you should spend some time with basic circuit analysis before claiming that I shall not post simply because I won't perform an experiment that you feel is requisite. I've made RC filters before, no need to do it again to appease you.
 
Julf, see my post #46 comment to FoMoCo, as it applies to you as well.
Here's what I will suggest. If you won't first try this rather simple experiment, that's your prerogative, but do not waste your time (or mine) commenting in this thread? There is a difference between healthy skepticism and closed-mindedness.

I think it is you who are showing the closed-mindedness. This thread is supposedly about investigating a "new" effect, but you don't seem to want to hear a simple, rational explanation based on well-known principles. It is almost as if you are enthusiastic about having finally found an unicorn, albeit one without a horn, and get really offended when we tell you it is a horse.
 
No, you don't a specific analog filter circuit. The oversampled alias images are low-pass filtered by interconnect cable capacitance, by amplifier bandwidth limits, by loudspeaker bandwidth limits, and by human hearing bandwidth limits.

So clearly you agree we need some form of analog low-pass filter. How is the capacitor inserted by Joe Rasmussen any different than the cable capacitance? All he has done is ensured *sufficient* capacitance.
 
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We don't need to........

trying my own I/V stage with input impedance of 3R , balanced , is still on list to do ;

prior to that , I'm speaking from experience with xformer based I/V stages , and I wrote with which DAC chips I tried it .

sound wise - they're more then good enough , and mentioned 6db filter was necessary

however - dispute - is it plain filter or necessary patch (solution) - I don't care
 
It looks like that low pass filter creates a slight roll-off at the end (and not-so-near-the-end) of the audio spectrum, and it has been reported to sound "psychoacoustically better". I remember that Mr. Linkwitz mentioned it in the last section of his article about "A Three-Enclosure Loudspeaker System with Active Delay and Crossover".

Yes, while at the same time removing the aliasing products and high frequency noise from the digital-to-analog conversion process (as specified by the principles of sigma-delta conversion).
 
some comments on the scope traces posted earlier :

1) http://www.diyaudio.com/forums/atta...ods-using-working-configuration-cap-value.jpg

Is the right trace measured at the DAC output or at the opamp IV output ?

2) http://www.diyaudio.com/forums/atta...dac-processing-circuits-optimized-largest.jpg


Same question.
Could you zoom in ? I'd be very interested in the edge rate (V/µs) of these transitions. Something like 10ns/div should produce interesting results.

Also, lots of overshoot, noise, and clock feedthrough here, is it the circuit, the layout, or the probing ?

3) http://www.diyaudio.com/forums/atta...al-level-worse-case-scenario-degraded-aud.jpg

In this trace the opamp is probably oscillating (quite badly...) from capacitance overdose at the input.
 
I see you changed your post after I responded to it. Who gives you the right to tell me I cannot comment?

First of all, when I edit posts, it's either for greater clarity or for more effective tone. In short, for more effective communication. I do this regularly. Your response had nothing to do with my edit, as I hadn't read it prior. Frankly, I resent your implication. I would simply post a new comment if I had something further or difference to address to you.

Secondly, I did not tell you that you cannot comment. It's interesting how you mis-characterize my comment, and then overreact to your own mis-characterization.,You protest too much. The bottom line is this, if you want to maintain a closed-mind regarding something which is very easy to test for yourself, that's your free choice. Just as it's my free choice to stop wasting my time attempting to open that mind. Those who are open to Joe's discovery will benefit, while those who aren't open won't. Simple as that, and audiophile life goes on.
 
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