John Curl's Blowtorch preamplifier part III

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...'Theory' says levels of harmonics are the same and that area under curves are the same....so they must sound the same !. Changes in timings of harmonics registers in the brain as a change in conditions and is an alert signal.....something wrong if we did not pick it up. Dan.

And before we go crazy with this, there must be a threshold of audibility of the phenomena which varies with amount of phase shift and fundamental frequency as well, so audibility is most likely a pretty complicated issue. I am unaware of a study which goes into the phenomena in this level of detail. The broadcast processing example can be extreme, as it is part of the commercial broadcast search for every last µdB of average level (bleccch), so it may be an extreme example.

Howie
 
Not what I meant an algorithm like the FFT does not lose any accuracy by eliminating computations it is not an "approximation" in that sense. Arbitrary ASRC in closed form is probably computationally too intensive.

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I´m sorry, i just misunderstood your previous line:
"Not the best analogy, there is only one answer and it is exact the speed enhancement is just that there is no loss of accuracy at all."

and thought it was related to the ASRC process.
 
Howie,
Have to agree that I have heard something at times that sounds like what might be referred to as 'smeared' sounding audio. In the past when people have made claims to that effect in forums such as this, sometimes the claim was dismissed as imaginary because nobody offered a convincing theoretical explanation consistent with prior auditory research, or new measurements and new research proving audibility.

While I agree that it is possible for audio to sound smeared, at this point its still not quite clear whether it is directly due to shifted phase, rate of change of phase vs frequency (aka group delay), quality of phase rotation processing DSP (i.e. somebody thought it was good enough to be inaudible), and or some other unstated factor(s). Do you have any further thoughts on the matter you would be willing to share?
 
I´m sorry, i just misunderstood your previous line:
"Not the best analogy, there is only one answer and it is exact the speed enhancement is just that there is no loss of accuracy at all."

and thought it was related to the ASRC process.

Right, now that I see some of these DAC's have 128 or 256 tap filters I can imagine there might be issues. I'm used to thousands of taps IIRC when I was playing with FIR RIAA 4 or 5 thousand taps got down to 1/2 LSB at 24 bits.
 
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Phase shift affects the transient sounds or dynamic signals the most to my ear.

Dynamically changing waveforms are a different matter from single sine tone phase shifts. In particular, it’s not only reasonable, but easy to demonstrate (at least under artificially produced conditions) that musical transients (pluck, ding, tap) can be severely damaged by phase shift. Many frequencies of short duration combine to produce a transient, and phase shift smears their time relationship, turning a “tock!” into a “thwock!”.



THx-RNMarsh
 
Typical anti scientific attitude.
I wonder what you see as typical pro-scientific attitude. Would it be never dismissed anything as imaginary because nobody offered a convincing theoretical explanation consistent with prior auditory research or new measurements and new research proving audibility in electronic audio replaying system?

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Phase shift affects the transient sounds or dynamic signals the most to my ear.

Dynamically changing waveforms are a different matter from single sine tone phase shifts. In particular, it’s not only reasonable, but easy to demonstrate (at least under artificially produced conditions) that musical transients (pluck, ding, tap) can be severely damaged by phase shift. Many frequencies of short duration combine to produce a transient, and phase shift smears their time relationship, turning a “tock!” into a “thwock!”.



THx-RNMarsh


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You have to draw the distinction here between changing the phase relationships of the component frequencies (a no-no) and phase shift through an amplifier.

Most amplifiers have 2-5 degrees of phase shift across the audio band ref DC (0 deg phase shift) of which most is in the upper octave and wont really cause a problem with music.
 
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--

You have to draw the distinction here between changing the phase relationships of the component frequencies (a no-no) and phase shift through an amplifier.

Most amplifiers have 2-5 degrees of phase shift across the audio band ref DC (0 deg phase shift) of which most is in the upper octave and wont really cause a problem with music.

Audio consists of more than a wide band amplifier. There are many filters involved.... more now with ADC/DAC , active cross-overs, loudspeakers and so forth. Not to mention, recreational filters from the older systems... tube era... with various tone controls, and filters and EQ. I am sure others can think of plenty of places where we have a filter.



THx-RNMarsh
 
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Audio consists of more than a wide band amplifier. There are many filters involved.... more now with ADC/DAC , active cross-overs, loudspeakers and so forth. Not to mention, recreational filters from the older systems... tube era... with various tone controls, and filters and EQ. I am sure others can think of plenty of places where we have a filter.



THx-RNMarsh

Ok - I was specifically referring to power amps. But every system has to deal with this issue. Let’s start at the speakers for instance . . .
 
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Ok - I was specifically referring to power amps. But every system has to deal with this issue. Let’s start at the speakers for instance . . .

Mark has been listening to several filters in the digital side of things.

Lets start with those. While your at it, throw in group-delay of the filters.

?? Identify the weak places where there is the greatest possible neg issues.



-RNM


BTW -- Has anyone here measured the group delay of their entire system?
 
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Hmm... Not sure where this is going. But, I will kick it off I guess: Sabre dacs have seven built-in PCM interpolation filter choices, plus ability for one set of custom coefficients to be loaded so as to give 'a unique sound signature' to your dac product. AKM refers to their interpolation filter choices as 'sound color' choices. The different filters are all audible in a well implemented dac design and using a low distortion power amp. AK4137 external ASRC has four filters to choose from and they all sound different. Naturally, not all the filters can be right in the sense of accurately reproducing digital music content, at most only one could be right since they all sound different. Pretty sure there are 0 right. The dac filters all look flat in the passband, so nothing to show in Stereophile measurements to explain differences in filter sound if passband frequency response graphs were shown. Exciting the filters with illegal digital signals at least gives different pictures to look at.

Also interestingly, although filter passbands look flat for both AKM and ESS dacs, Sabre dacs have a well earned reputation for sounding like they have 'weak bass.' By way of contrast, AKM's reputation is for deep, satisfying bass. Don't know if the tendency of Sabre dacs to sound 'bright' at all frequencies makes the bass sound weak relative the brightness, of if there is some dynamic effect with music that makes bass sound weak despite flat FR shown with measurements. Also, interesting is that DAC-3 lacks that artificial brightness instead and has a wonderfully balanced (or voiced) sound (at least in comparison to other Sabre dacs), which tends to make it sound much more analog and or vinyl-like.

Before anyone asks, I can't explain it all. I haven't been working on dacs long enough yet, and what I have done with them so far hasn't really focused on trying to correlate measurements with perceptual experiences.
 
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Doesn't the Red Book brickwall dominate any possible system "transfer curve"? Or, would it be possible, knowing the recording LPF characteristics in detail, to "back out" of them in playback?


Otherwise, it seems like all of these playback remedies are general answers to specific questions. A naive view of the system would look at both ends together. Folks playing (and especially, transfering) old shellacs need to know a lot about the original processes - like that.


All good fortune,
Chris
 
Hmm... Not sure where this is going. But, I will kick it off I guess: Sabre dacs have seven built-in PCM interpolation filter choices, plus ability for one set of custom coefficients to be loaded so as to give 'a unique sound signature' to your dac product. AKM refers to their interpolation filter choices as 'sound color' choices. The different filters are all audible in a well implemented dac design and using a low distortion power amp. AK4137 external ASRC has four filters to choose from and they all sound different. Naturally, not all the filters can be right in the sense of accurately reproducing digital music content, at most only one could be right since they all sound different. Pretty sure there are 0 right. The dac filters all look flat in the passband, so nothing to show in Stereophile measurements to explain differences in filter sound if passband frequency response graphs were shown. Exciting the filters with illegal digital signals at least gives different pictures to look at.

Also interestingly, although filter passbands look flat for both AKM and ESS dacs, Sabre dacs have a well earned reputation for sounding like they have 'weak bass.' By way of contrast, AKM's reputation is for deep, satisfying bass. Don't know if the tendency of Sabre dacs to sound 'bright' at all frequencies makes the bass sound weak relative the brightness, of if there is some dynamic effect with music that makes bass sound weak despite flat FR shown with measurements. Also, interesting is that DAC-3 lacks that artificial brightness instead and has a wonderfully balanced (or voiced) sound (at least in comparison to other Sabre dacs), which tends to make it sound much more analog and or vinyl-like.

Before anyone asks, I can't explain it all. I haven't been working on dacs long enough yet, and what I have done with them so far hasn't really focused on trying to correlate measurements with perceptual experiences.

This is great contrast to all the zealot jerks on AVS forum that will go all out pitch-fork hunt on you for telling them DACs sound different. They claim the measurements mean they sound the same. Hmm do I trust a bunch of biased people or the manufacturers offering solutions on different filter tastes... hmm
 
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