John Atkinson: Yes, MQA IS Elegant...

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6 hours ago, Shadders said:

Have you analysed the ringing ?.

 

Often. Analysed, and studied the literature about it.

 

And guess what. Even when the frequency of ringing is audible to the test person, as in the 2kHz example, this in itself does not mean that the ringing is audible. The audibility depends on how the temporal spread compares to the inverse of the cochlea's critical band width at that frequency.

 

 

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It is not applicable, if you like that more.

 

For some people 'appear' seems to hold a 50% chance.

 

For others it is more like 90-99%, and words like 'appear' are used to avoid

an absolute.

 

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12 hours ago, John_Atkinson said:

You can see from the article of mine that triggered this thread an example where a perfectly legal, band-limited impulse nevertheless excites the DAC reconstruction filter's sinc-function ringing.

 

 

John, seriously: if it does do that, you have done something wrong in preparing the stimulus and/or in capturing the DAC's output.

 

If you want conceptual proof then a safer and cleaner approach is this:

 

1) work only in the digital domain

 

2) prepare the stimulus as an impulse at a very high sample rate, downsampled to a much lower Fs by using a blameless minimum phase SRC, like iZotope.

This gives you a truly band-limited 'impulse' with a clean leading edge, but obviously with the post-ringing of the MP SRC.

 

3) send this through a linear-phase reconstructor (i.e. the DAC, or rather a model of it).

You will see that no pre-ringing will be added to the output signal if the original filtering was done to a frequency not higher than slightly below Fs/2.

 

4) do so at high numerical accuracy, as any truncation or rounding will generate out of band signal.

 

 

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16 hours ago, Shadders said:

I said earlier that ringing occurs in every waveform that has a transient change - the transient does NOT have to be at the ringing frequency -

 

 

Take a  'ringing' Sinc-like low-pass filter with a transition frequency F.

 

If the spectrum of the stimulus is non-zero at F, then the output signal will visually exhibit filter ringing. If the spectrum of the stimulus is zero at F, then the output signal will not show ringing.

 

It is as simple as that.

 

 

As for the ringing envelope occupying frequencies below F ... consider that a filter is a linear operator, and think this through.

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13 hours ago, John_Atkinson said:

You can see from the article of mine that triggered this thread an example where a perfectly legal, band-limited impulse nevertheless excites the DAC reconstruction filter's sinc-function ringing.

 

This here, on page 2 of your article:

" To answer this question, I took the 96kHz pulse captured by the Ayre QA-9 in Listen mode and used the highest-quality sample-rate converter in BIAS Peak to downsample it to 44.1kHz.

...

Note that the digital data has already been band-limited to half the 44.1kHz sample rate, thus is a "legal" signal."

 

Did you verify the nature of the BIAS Peak SRC? The two versions documented at http://src.infinitewave.ca/ are clearly half-band, allow aliasing, and therefore do not provide legal band-limiting.

 

 

Isn't it the audio press's task to be thorough in its search for the truth? To avoid publishing half-truths and lies? And does this not imply a certain level of understanding of the subject? Is this not your responsibility?

 

Once more Stereophile is letting us down.

 

 

 

As an aside, the entire article, once you see through the confusing style and presentation, and with ignoring the more farcical parts, brings us nothing new at all.

 

 

 

 

 

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3 minutes ago, Shadders said:

Not sure what you are saying here.

 

That is abundantly clear.

 

Once more ... you claim that you investigate the ringing in isolation, and you claim that the ringing envelope emcompasses frequencies down in the passband. In other words, you claim that the ringing itself has audible content all over the passband. This means that the filter creates new spectral content, well below the filter's transition frequency.

 

Now tell us the main properties of a linear operator.

 

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A linear operator cannot create new spectral content.

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A linear, time invariant system, ... like a filter.

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I was not asking you what a linear system is.

 

I asked the question for you to ponder what this means in the face of your claim

 

" Separate issue, the envelope of ringing does have energy within the passband. "

 

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Moreover, mentioning the envelope in this story is quite irrelevant.

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11 minutes ago, Shadders said:

It is not irrelevant. If the envelope was different, then the frequency spectrum would be different.

 

If the envelope were different then the filters would be different. And in discussing filters we talk about transition frequency, transition band(width), steepness/order, ... There is no need to single out the envelope of the (impulse) response.

 

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Deep sigh ...

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2 hours ago, Shadders said:

I agree, that you DO NOT require the input signal into the filter to have any out of band frequencies to cause ringing. Here is the signal analysed – this is the output of the 512tap linear phase filter. The input signal is a 1kHz sine wave, sampled at 192kHz.

 

 

You claim something, and then you try to demonstrate it with a signal that happens to have a lot of 'out of band frequencies' ...

 

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1 hour ago, Shadders said:

Can you explain why it does have out of band energy ?.

 

Your function is the multiplication of a sine and a rectangle function. The spectrum of the

result is the convolution of the part spectra. The spectrum of the rectangle function, while

falling with frequency, still extends to infinity.

 

Something to ponder: the ringing imparted by a low-pass filter is not something added to the

signal. Rather, it is the result of something being taken away from the signal.

 

 

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6 hours ago, John_Atkinson said:

Because there is no spectral content above the new Nyquist frequency of 22.05kHz, due to the SRC's high-order low-pass filter.

 

But if the SRC is half-band then there is content exactly at 22.05k (and aliased content below that), and that makes the signal illegal in the sense that it will trigger the ringing of any later reconstruction filter.

 

Now if you had commanded the SRC to stop at 22.04k all would be well, but most SRCs do not support this.

 

This is Bias Peak Pro 6.03:

 

 

This is iZotope:

 

 

The difference, and its implications, is significant.

 

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14 hours ago, Shadders said:

 it is a bank holiday here in the UK - so it is eat, drink and be merry.

 

Any Sunday will do for that. In our case a Roter Veltliner with salmon and green asparagi.

 

14 hours ago, Shadders said:

 are you are stating that at the sin(phi)=0 (start of the sine wave in the example) that there is a discontinuity ?

 

is the multiplication of the sine which is continuous (-infinity to +infinity) is gated by the rectangular function, and this gating function of the rectangular pulse causes the ringing ?

 

There certainly is a discontinuity. Switching something on or off in our ideal mathematical world implies a discontinuity. Your particular test signal is simple enough to be separated into its component functions, namely an eternal sine, a rectangle (keying the sine on and off), and a step (keeping the signal at high after the sine went off). For all of these functions we have closed-form Fourier transforms, two of which stretch to infinity. This allows us to see/know quickly that the gated sine also stretches to infinity in the frequency domain.

 

 

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11 hours ago, Jud said:

 Minimum phase filtering is widely used in audio, including by nearly all room equalization software.

 

That is because room equalisation aims at correcting (mostly) minimum phase problems. You don't want to use linear phase for that.

 

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12 hours ago, Shadders said:

In terms of a rooms equalisation, where there are many peaks and troughs, and as per the text in their website - they use multiple filters.

 

But that does not mean that band-splitting is going on. It also does not mean that any of these filters need to be of very high steepness.

 

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Wrong

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13 hours ago, John_Atkinson said:

As I have said before, spectral analysis shows that the "band-limited impulse" I used has no content above 22.05kHz.

 

I had to be brief yesterday, but now I am back.

 

You used Bias Peak for the sample rate conversion, i.e. anti-alias filtering followed with downsampling. If you did spectral analysis on the result you could only have done this after the downsampling, which is wrong. You should have inspected the naked output of Bias Peak's anti-alias filter, prior to downsampling.

It is known that the majority of commercial tools (and hardware) use half-band filters, including Bias Peak. These are only 6dB (*) down at Fs/2 and thus do not provide correct band-limiting, violating the sampling theorem.

 

You should have used a SRC that guarantees that the stopband is reached before Fs/2.

 

(* Or a similar value, as many filters are a cascade of elementary half-band and non--HB filters.)

 

 

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10 hours ago, vl said:

Wouldn't the use of a gentler, non brick wall filter to reduce the signal bandwidth to LESS THAN the Nyquist frequency make more sense?  This will avoid excitation of the filters at the Nyquest frequency.

 

For base rate (1x) signal you always need a close-to-brick-wall filter, but you can take a transition band wide enough to make its ringing of the same order as the inherent time constant of the upper band of the cochlea of a healthy teenager (*). For 2x rate and higher you can of course roll off much smoother.

 

These are the filter configurations I use, always reaching full stop-band at Fs/2:

-44.1 kHz: 4 kHz transition band width (i.e. starting at 18kHz)

-48 kHz: 6 kHz transition

-96 kHz: 18 kHz transition

 

Recordings made like this do not trigger any DAC ringing, ever.

 

(* If this were important, which I do not know because I am no longer a healthy teenager. But neither is anyone else in this discussion, or, for the matter, in the entire audio industry.)

 

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35 minutes ago, adamdea said:

Conversely the point which your article seems to skate over is that if there is ringing in the 16/44 data then a slow/lazy/non  filter in the dac will just pass the (data) ringing and there is no (apparent) time domain advantage in such a filter. So the time domain blather about such filters is pure bull.  (edited, F)

 

It has been, since the early days of Wadia. And the entire audio press bought it.

 

35 minutes ago, adamdea said:

And who says your made up shape is a musical transient?

 

To be fair, JA's example does not have to be real. We can always invent a new musical instrument and write a score for it, to be played in an anechoic room with all noise below 10 dB SPL. Recording practices have to be prepared for any conceivable sound.

 

But that still does not make ringing a real problem ...

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3 minutes ago, John_Atkinson said:

I resampled the 44.1kHz file to 96kHz, in order to examine the content above 22.05kHz in the digital domain

 

An utterly pointless exercise. Of course there is no content above 22.05k, provided the upsampler did its job well.

 

Do you have any idea what you are doing?

 

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33 minutes ago, mansr said:

Where did he use an SRC?

 

Page 2 of the article:

 

"To answer this question, I took the 96kHz pulse captured by the Ayre QA-9 in Listen mode and used the highest-quality sample-rate converter in BIAS Peak to downsample it to 44.1kHz. "

 

...

 

"I therefore repeated the test using the Brooklyn's FR (Fast Rolloff) filter. This will attenuate the data's ringing at 22.05kHz, but as you can see from fig.15, it appears to have substituted its own acausal ringing. (Note that the digital data has already been band-limited to half the 44.1kHz sample rate, thus is a "legal" signal.) "

 

The issue is with JA's assertion that the digital stimulus is a 'legal' signal. It is not, since at the root it was generated with the (extremely likely half-band) AA filter of Bias Peak.

 

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Not 'above', but 'at'.