Stereo and higher sampling rates - "time domain" question

[j_j]

But this is the argument that says reproducing inaudible frequencies can actually reduce audible quality.

Since the inaudible frequencies are inaudible - they can't improve the sound. But they can inter-modulate with other frequencies (especially in the output transducer) to create audible distortion that otherwise wouldn't exist.

I'm afraid that such IM distortion is part of how a rimshot, for instance (or a glockenspiel hit, some other stuff) sounds, where the IM comes from the air transmission. So is it distortion or signal properties? I think that's a semantic issue that is going to be very hard to settle.

 

[antcollinet]

I'm afraid that such IM distortion is part of how a rimshot, for instance (or a glockenspiel hit, some other stuff) sounds, where the IM comes from the air transmission. So is it distortion or signal properties? I think that's a semantic issue that is going to be very hard to settle.

But isn't that only going to be the case for the IM distortion that is captured in the recording - which can also be captured at 44.1kHz since it consists of audible frequencies (Not considering edge case frequencies between 20 and 22kHz)? Any additional intermodulation taking place in (for example) the speakers as part of reproduction - is not part of that sound? It is just distortion.

 

[voodooless]

This is not a large problem at sane levels,

Well, obviously not. I'm just reaching out to the high-sample-rate proponents to hand them some straws to grasp at

 

[j_j]

But isn't that only going to be the case for the IM distortion that is captured in the recording - which can also be captured at 44.1kHz since it consists of audible frequencies (Not considering edge case frequencies between 20 and 22kHz)? Any additional intermodulation taking place in (for example) the speakers as part of reproduction - is not part of that sound? It is just distortion.

I would suggest you get a wideband mike (instrument mike would work) and capture a couple of glockenspiel hits. Keep the level down. You may wince at those peaks. No, much equipment can't handle them. Ribbon tweeters can, little else, and not all ribbons.

Furthermore, you seem to not quite grasp the issue of anti-aliasing filters. The pre-echo thing is real.

 

[antcollinet]

I would suggest you get a wideband mike (instrument mike would work) and capture a couple of glockenspiel hits. Keep the level down. You may wince at those peaks. No, much equipment can't handle them. Ribbon tweeters can, little else, and not all ribbons.

Furthermore, you seem to not quite grasp the issue of anti-aliasing filters. The pre-echo thing is real.

I think I am not explaining myself properly - or I am just being dense. Either way - I'll leave it there.

 

[pma]

I would suggest you get a wideband mike (instrument mike would work) and capture a couple of glockenspiel hits.

There we go ... /resonances between the hits/

 

[j_j]

There we go ... /resonances between the hits/

View attachment 470111

It's on the hits that I would expect problems with antialiasing filters, although, I am assuming they are at least competent.

If they weren't, I don't want to think about what happens to all that HF stuff.

 

[pma]

It's on the hits that I would expect problems with antialiasing filters,

There is a file (192/24) to check it, with the hits:
/but you wanted the 44.1 I would guess/

.... and there is a downsampled version to 44.1/24 as well. A chance to make an ABX, though it does not test ADC anti-aliasing filter, but rather the downsampling algorithm.

 

[HalSF]

My layman’s sense is that ambient FUD around digital music timing is one of the most common ways to create an opening for uncertainty around the “problem” and then going on to sell various exotic high-end audio solutions. Words like “smearing” and “jitter,” and pitches for stuff like “the incredible sense of space and clarity brought by MQA” (vs. the supposedly inherently disfigured audio produced by bog-standard digital playback) are some of the ways you can claim that digital audio timing is an unsolved challenge requiring a dramatic proprietary intervention.

 

[j_j]

My layman’s sense is that ambient FUD around digital music timing is one of the most common ways to create an opening for uncertainty around the “problem” and then going on to sell various exotic high-end audio solutions. Words like “smearing” and “jitter,” and pitches for stuff like “the incredible sense of space and clarity brought by MQA” (vs. the supposedly inherently disfigured audio produced by bog-standard digital playback) are some of the ways you can claim that digital audio timing is an unsolved challenge requiring a dramatic proprietary intervention.

This informed person thinks the "timing" stuff is poppycock. Jitter is a real thing but only if your DAC is poorly designed, of course. Or ADC on capture. Both of these were licked in the 1950's in telecom land.

I have no comment on MQA beyond the comment I have none.

As to exotic cables, special DAC's, etc, I'm pretty much in your court.

There are a few POSSIBLE nits for the most sensitive listener on the most critical material, yes. Also, 18 bits would have been better, but 16 is enough in most rooms.

 

[pma]

The pre-echo thing is real

As the ADC anti-aliasing filters are always compromised, why not to design own, Bessel-Thomson LPF filter and put it in front of the ADC? Some penalty in amplitude response might not be audible.

 

[j_j]

As the ADC anti-aliasing filters are always compromised, why not to design own, Bessel-Thomson LPF filter and put it in front of the ADC? Some penalty in amplitude response might not be audible.

The right input will make the amplitude response audible if it's over a dB or so.

 

[pma]

The right input will make the amplitude response audible if it's over a dB or so.

Yeah.