On the subject of "ringing"

[mansr]

1 hour ago, semente said:

What if you can't move the transition frequency higher up i.e. low-passing a mid-woofer with a digital crossover?

Then you need to use a slower roll-off.

[mansr]

14 minutes ago, semente said:

How slower is that?

"Hard" cones need steep filters.

Choose a driver for next band up (midrange/tweeter) with a useful response starting below the cone breakup frequency. This overlap determines the minimum steepness of the filter. Or conversely, the desired filter roll-off determines the low end of the midrange/tweeter response.

 

Crossover filters are never of the "brick-wall" variety (that would make a smooth transition impossible), so there's not going to be all that much ringing.

[mansr]

6 minutes ago, John_Atkinson said:

Incidentally, I have recently been examining the time-behavior of A/D converter antialiasing filters and have found just one which captures a band-limited impulse without any ringing before or after: the Listen filter on Ayre's QA-9 converter, designed by Charley Hansen and Ariel Brown.

Could you describe your "band-limited impulse" a bit more? How is it generated? What does the waveform look like?

[mansr]

4 minutes ago, John_Atkinson said:

In brief I created a "digital black " file sampled at 38kHz and drew the shape of the diagnostic waveform I needed with BIAS Peak's pencil tool. I then modified it  until I got the desired spectral content.

Well, that's rather vague.

 

4 minutes ago, John_Atkinson said:

To create an analog signal to feed to the ADCs under test, I decoded the signal with a DAC capable of handing 384k PCM data without downsampling. As the ADCs to be tested were all set to 96kHz sampling, the DAC's own ringing at Nyquist would be an octave above the ADC's output passband and would be rejected. Apologies but people will have to wait for a forthcoming article in Stereophile for more detail.

When you publish the results, will you include at least a plot of this "impulse" signal? If you made the file available for download, others could perform the same test on their equipment and get results meaningfully comparable to yours. Just something to consider.

[mansr]

1 hour ago, Ralf11 said:

unusual results require unusual proof -- and anything appearing to contravene known physical laws is indeed unusual

New physics is pretty much confined to high-energy accelerators these days.

[mansr]

9 hours ago, Spacehound said:

If pre-ringing  actually exists in 'reality' then the filter must be made out of the electrical equivalent of thiotimoline.

Failing that it is merely an artefact of the measurement methods.

 

There is only one guy on this entire forum that might be able to convince me otherwise.

First of all, pre-ringing isn't a well-defined term. In fact, it is never used outside of audio. Let us thus define it for the sake of this discussion as an impulse response having at least one negative excursion prior to the (positive) peak.

 

Symmetrical impulse response plots are often centred around time zero. On the face of it, this amounts to (part of) the response preceding the input. Such a system is termed non-causal and cannot be physically realised. We nevertheless use this representation because it is mathematically convenient. Now remember, we are dealing with a time-invariant system, which means a time-shift of the input results in a time-shift of the output. Thus, provided the impulse response is finite, we can make it causal simply by shifting it such that it becomes zero for all negative time values. The only effect of this is the addition of a constant delay to the output. Importantly, the frequency response is not affected.

 

In practice, a digital filter incorporating such a time-delay is trivially constructed. An all-analogue realisation is trickier, although it can be accomplished through the use of various delay elements.

 

Have I convinced you yet?

[mansr]

13 hours ago, John_Atkinson said:

The high-frequency rolloff of the Ayre QA-9's Listen filter seems pretty benign. You can find my measurements at https://www.stereophile.com/content/ayre-acoustics-qa-9-usb-ad-converter-measurements

 

Below is fig.2  on that page, showing the frequency response at –1dBFS with the Listen filter,  analyzed in the digital domain, and data sampled at 192kHz (left channel blue, right red), 96kHz (left green, right gray), 48kHz (left cyan, right magenta)  The vertical scale is 1dB/div. The response at 20kHz is down by just a fraction of a dB at all 3 sample rates, though the rolloff is slow, which will potentially allow some aliasing at single Fs rates with some music.

 

John Atkinson

Editor, Stereophile

 

Something is off here. A 48 kHz sampled signal by definition has no content above 24 kHz. What is that graph actually showing?

[mansr]

3 minutes ago, John_Atkinson said:

Look more closely at the graph using the color coding for the traces I supplied in my earlier posting.

Oh, I see now the traces stop at different places. That's far too many overlapping traces in one graph if you ask me.

 

Now that's settled, I'm anything but impressed. An anti-aliasing filter that is down only a few dB at Nyquist will result in severe aliasing if the input has any content above this frequency. For the higher sample rates, this might not be a problem, but for 44.1/48 kHz, a lot of music has content extending far enough above Nyquist that aliases could easily fold back well into the audible range.

[mansr]

3 minutes ago, Shadders said:

I only have Linux OS at the moment - i checked the SoX page - seems to be Windows or Mac only ?

As de facto co-maintainer of Sox, I can state with certainty that it runs on Linux.

[mansr]

1 minute ago, Shadders said:

I have mode NO mention of downsampling. Why you persist in repeating this, despite i am not talking about it, and my example does NOT state anything about downsampling, shows you intentionally lie about others to state that they are wrong.

The downsampling in Fokus' example is incidental and irrelevant. Use the 'sinc' effect in Sox to apply filters without resampling. You'll get the same effect.

[mansr]

6 minutes ago, Shadders said:

Happens in the analogue world too, but without the pre ringing.

I'm tempted to build an analogue filter with "pre-ringing" just because.

[mansr]

6 minutes ago, psjug said:

Ah - OK.  I thought this was a recording.  If you started and stopped the sine wave components abruptly, then these are not band limited, right?

Of course not. Any deviation from a pure sine wave creates additional frequency components.

[mansr]

9 hours ago, anwaypasible said:

pre ringing to me is when the quantified airspace of what the convolver frames has a gap with the frame of normal allocated space.

that can only happen with say 16bit audio going through a 24bit convolver, because thanks to the size difference between the two there is room for a gap (doesn't work/happen when staying within the same bitdepth).

WTF?

 

9 hours ago, anwaypasible said:

imagine you've got an audio track with phase that varies from 1 through 180 degrees.

What is that supposed to mean?