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Marantz AV7705 Home Theater Processor Review

Martin_320

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With a slow roll off filter you can get imaging above nyquist. A mirror imaged reflection of the below 22 khz stuff. If you had high level in the upper frequencies it can image at fairly high level just above 22 khz. Whether that can cause IM distortion back into the audible band in tweeters etc is dependent upon the particulars and wouldn't do so very often. I'm not sure that is what they meant prior to translation however.

I understood that regarding those "mirror imaged" reflections of which you speak, when oversampling is used, these images are pushed much further up the spectrum (compared with your 22kHz non-upsampled Nyquist pivot point). Eg, 2x oversampling would therefore push images up to 44kHz, and with 4x oversampling these images would occur at 88kHz; and so on...
And by doing so, you can use a less steep filter :) But I admit -- don't know the inner-workings of the AKM4458.
 

bigguyca

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- Omitted -

Gene's comments:
https://www.audioholics.com/av-receiver-reviews/marantz-sr8015/conclusion

"I've been asked by several audioholics to start including square wave response curves in our measurements. Ideally there'd be no overshoot or ringing like we are seeing in the curve above. BUT Marantz confirmed this was a deliberate design attribute due to the short delay, slow roll-off DAC filter curve employed on ALL Marantz products to help better preserve high frequency phase response."

Why did you post Gene's comments? Do you agree with the comments and with Marantz's claim?

Gene made a number of technical mistakes in the SR8015 review. The statement is yet another mistake, and a clear technical mistake. Gene evidently agrees with the statement since he offered no correction, he evidently doesn't understand how these filters work.

It is a clear technical mistake, not some sort of opinion. It is no wonder that Marantz makes odd design decisions if they don't understand the basics of these digital filters.

Linear phase filters with sharp cutoffs preserve phase. The is why they are called linear phase filters. Minimum filters, such as Marantz uses, cause phase distortion, that is, they delay the signal based on its frequency. This is how these filters actually work in the real world.

The filter that Marantz uses is a minimum phase filter and causes phase distortion. If preserving high frequency phase response is important then Marantz has chosen exactly the wrong filter and clearly doesn't understand what they have done.
 

bigguyca

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I understood that regarding those "mirror imaged" reflections of which you speak, when oversampling is used, these images are pushed much further up the spectrum (compared with your 22kHz non-upsampled Nyquist pivot point). Eg, 2x oversampling would therefore push images up to 44kHz, and with 4x oversampling these images would occur at 88kHz; and so on...
And by doing so, you can use a less steep filter :) But I admit -- don't know the inner-workings of the AKM4458.


Perhaps this will help you actually understand how digital sampling and filters work.

https://en.wikipedia.org/wiki/Reconstruction_filter
 

bigguyca

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I know it was by design. In fact I was the one who posted the response from Marantz Japan earlier this year about their rationale for choosing the slow roll-off filter. They never cited superior transient and impulse response as the reason. And regarding the pre-ringing issue that apparently is inherent from the use of fast roll off filter, of I understood it right, it would only apply when lower sampling rate such as 44.1/48 kHz are used, and even then it is not proven that people could hear the effect. At higher sampling freq, the ringing effect would be well outside the audible range. One of the site I read up on this was probably from the one linked below, time permitting I would read up on the topic again as I am now going by distant memory so I could be wrong and you may be right.:)

http://archimago.blogspot.com/2013/06/measurements-digital-filters-and.html

- Omitted -

The link above provides an incomplete view of the information presented on the Archimago site on the DAC filter issue. The following are links to some worthwhile reading on the specific subject.

https://archimago.blogspot.com/2017/

https://archimago.blogspot.com/2018/01/
 

bigguyca

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I have noted that Marantz likes to talk about fast slew rates and fast transient response performance in its products. That could explain their decision for setting the DAC filter to slow -- since a slow filter produces a faster impulse response and less ringing etc.. (that's even noted in AKM's own data sheet). Fast transient response could also be the reason they use HDAMs.

- Omitted -

:D

The quantity of ringing is dependent on the number of taps in a FIR filter.

Do keep in mind that single sample impulses don't appear in real content. The impulse response is a way to understand what type of filter is used, and some of its characteristics, without having access the details of the design, which are normally proprietary.

Most digital content has been subject to numerous linear phase FIR filters before it reaches the consumer.
 

peng

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Why did you post Gene's comments? Do you agree with the comments and with Marantz's claim?

Gene made a number of technical mistakes in the SR8015 review. The statement is yet another mistake, and a clear technical mistake. Gene evidently agrees with the statement since he offered no correction, he evidently doesn't understand how these filters work.

It is a clear technical mistake, not some sort of opinion. It is no wonder that Marantz makes odd design decisions if they don't understand the basics of these digital filters.

Linear phase filters with sharp cutoffs preserve phase. The is why they are called linear phase filters. Minimum filters, such as Marantz uses, cause phase distortion, that is, they delay the signal based on its frequency. This is how these filters actually work in the real world.

The filter that Marantz uses is a minimum phase filter and causes phase distortion. If preserving high frequency phase response is important then Marantz has chosen exactly the wrong filter and clearly doesn't understand what they have done.

I didn't really want to post Gene's comment as such, but Marantz's, but it was indirectly quoted by Gene so I thought I should post the whole thing, just to be fair.. I have no comments on Marantz's response. In my experience it is not always easy to interpret their responses as something could have lost in translation from their engineering in Japan.
 

bigguyca

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I didn't really want to post Gene's comment as such, but Marantz's, but it was indirectly quoted by Gene so I thought I should post the whole thing, just to be fair.. I have no comments on Marantz's response. In my experience it is not always easy to interpret their responses as something could have lost in translation from their engineering in Japan.

Gene posted the response so he certainly should have understood the comment as posted. He should also stand behind the quote since he provided it.

It is D/M's responsibility to make sure these quotes are correct. D/M is doing business in this case in the United States, their office in in Vista, CA, and so quotes in English should be correct. The quote has been available for several weeks and has not been corrected by D/M, so it should be taken as accurate.

Why are you offering excuses for D/M? D/M needs to provide accurate information. Are you claiming that every mistake D/M makes is a mistake in translation? Why is D/M not expected to provide accurate information? Is there some allowed handicap here as in golf or bowling?

It's hard to understand your position. These are technical questions that have actual factual answers.
 

Martin_320

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Gene posted the response so he certainly should have understood the comment as posted. He should also stand behind the quote since he provided it.

It is D/M's responsibility to make sure these quotes are correct. D/M is doing business in this case in the United States, their office in in Vista, CA, and so quotes in English should be correct. The quote has been available for several weeks and has not been corrected by D/M, so it should be taken as accurate.

Why are you offering excuses for D/M? D/M needs to provide accurate information. Are you claiming that every mistake D/M makes is a mistake in translation? Why is D/M not expected to provide accurate information? Is there some allowed handicap here as in golf or bowling?

It's hard to understand your position. These are technical questions that have actual factual answers.

If you take such issue with what Marantz says, via Gene, and here via Peng, and since I presume you are from CA (as implied by your user name ), then why don't you pop down to their office in Vista, CA discuss it with them directly and come back here with the neccessary clarification, and in addition call up Gene from Audioholics and provide him with your findings also? That would be more added-value than 'shooting the messenger' on forums like this one.
 

peng

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Gene posted the response so he certainly should have understood the comment as posted. He should also stand behind the quote since he provided it.

It is D/M's responsibility to make sure these quotes are correct. D/M is doing business in this case in the United States, their office in in Vista, CA, and so quotes in English should be correct. The quote has been available for several weeks and has not been corrected by D/M, so it should be taken as accurate.

Why are you offering excuses for D/M? D/M needs to provide accurate information. Are you claiming that every mistake D/M makes is a mistake in translation? Why is D/M not expected to provide accurate information? Is there some allowed handicap here as in golf or bowling?

It's hard to understand your position. These are technical questions that have actual factual answers.

There is no need to be confrontational, this is just a hobby forum, not an engineering forum!!

My simple answer to your interrogation:D is, no I am not offering excused for D+M at all. I already said I had no comments on the response itself. No comments does not equal offering excuses does it? When offering comments, or excuses (not offered), It would be easy to take things out of context too, even just unintentionally, when only one sentence was used in the quoted response in this case. I thought I was clear, that, my quoting of that indirect quote via Gene meant to point out it was "by design" according to Marantz. I thought that part seems clear.

Again, I don't want to get into debating the underlying reasons and/or details that I do not feel enough information was given in that one liner. If you feel right about making a conclusion as you apparently have done, that's you, not me. The "lost in translation" was mentioned by Martin, specifically referred to Marantz response of their rational in choosing that slow roll off filter and I was agreeing with Martin. I think my position has been simple and clear, you might have just overreacted, to something that isn't there..:D
 

Sal1950

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The "lost in translation" was mentioned by Martin, specifically referred to Marantz response of their rational in choosing that slow roll off filter and I was agreeing with Martin.
I'm not entering into this discussion and have zero technical digital knowledge but I would simply like to make short comment.
As I see it the issue with Marantz in this situation could have two possible answers.
1 They made a subjective choice from the beginning to use the filter set up the way it is, just like they say.
2. They have been caught out by ASR and others who have measured their products and revealed some shortcomings. After being outed, the tech and marketing departments came together to whip up a face saving answer and found this one. "although the numbers suggest a problem, after subjective listening it just sounded better this way."
An excuse that will certainly fly with the subjective crowd, which is by far the vast majority of audiophiles today.
Only problem from our point of view is they have now painted themselves into a corner and will have to find more creative marketing words to make the "believers" happy to change to a better measuring setup.
 

Matthew J Poes

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I don’t intend to get in the middle of this discussion, nor am I talking sides. But thought I would clarify based on what I know.

there is no Marantz engineering in California or any part of the United States. That is just marketing and sales, logistics, etc. all engineering is in Japan.

the comment associated with Marantz Japan does appear to have come from the main engineer in Japan. English is not his first language and so certainly it can be a little broken or hard to follow.

Marantz believes that slow filters, while having some measurement faults, sound better. It’s a design choice. I don’t personally agree with that design choice and wish they would at least make it selectable.

Marantz is not alone in this view. Yamaha engineering in Japan, Cherry Amps here in the US, and others have all told me that they prefer slow rolloff. Yamaha at least makes it selectable.

I own a few AKM based dac/headphone amps with selectable filters. I’ll be totally honest in saying I can’t hear a difference between them. I couldn’t hear a difference on the last Yamaha product I reviewed with that feature either.

Sean Olive has commented to me before that they did some research into the use of FIR phase correction filters. He told me that when people could hear a difference, it was the Pre-ringing introduced by the correction. Suggesting he felt the pre-ringing is audible. Is that a knock against sharp filters with a lot of pre-ringing I leave for you to decide. The research into any of this sucks, so I am not sure we have much to look at. As I said, I can’t hear a difference.
 

Matthew J Poes

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Ah good one -- I didn't see your answer from Marantz in Japan before.

However, when I read the statement, he hasn't really been very clear, or these are the words of a comms spokesperson and something got 'lost in translation'.
For example, I don't follow how he can be talking "aliasing" and "quantisation" noise -- as I've always understood that anti-aliasing is process you do in the PCM sampling (A-to-D) stage, rather than in the PCM decoding (D-to-A) process.
eg. You'd get aliasing in the encoded waveform if you hadn't previously low-passed the source analog signal (eg from a vocal microphone or guitar interface etc. ) at just below half the sample rate. (Nyquist/Shannon) .
Here I don't see how that can apply to a reconstruction filter situation, which is there to take out DAC noise (and there is no re-sampling going on -- as the fs rate stays exactly the same as what was used during the recording/sampling phase -- so no further aliasing is would happen anyway).
https://en.m.wikipedia.org/wiki/Reconstruction_filter

the concept is explained here if it helps. The filter is needed to stop aliasing into the DAC output as well. A slow filter does a poorer job allows some aliasing and distortion.

I have a measurement example of what this looks like with music somewhere. I created it by accident decoding a bunch of music and it’s spectrum with no filter at all. I was trying to see the entire bandwidth and inadvertently introduced aliasing. It was textbook perfect. Cool to look at.

I’ll post it if I can find it on my laptop.
 

Martin_320

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Hi, there's been lots of discussions in another thread.
In actual fact, to talk about "aliasing" in a DAC situation (when there is no resampling) is not the correct terminology.
The artefacts in the spectrum are actually due to "imaging", which are a by-product of the sample-and-hold process.
 

peng

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I don’t intend to get in the middle of this discussion, nor am I talking sides. But thought I would clarify based on what I know.

there is no Marantz engineering in California or any part of the United States. That is just marketing and sales, logistics, etc. all engineering is in Japan.

the comment associated with Marantz Japan does appear to have come from the main engineer in Japan. English is not his first language and so certainly it can be a little broken or hard to follow.

Marantz believes that slow filters, while having some measurement faults, sound better. It’s a design choice. I don’t personally agree with that design choice and wish they would at least make it selectable.

Marantz is not alone in this view. Yamaha engineering in Japan, Cherry Amps here in the US, and others have all told me that they prefer slow rolloff. Yamaha at least makes it selectable.

I own a few AKM based dac/headphone amps with selectable filters. I’ll be totally honest in saying I can’t hear a difference between them. I couldn’t hear a difference on the last Yamaha product I reviewed with that feature either.

Sean Olive has commented to me before that they did some research into the use of FIR phase correction filters. He told me that when people could hear a difference, it was the Pre-ringing introduced by the correction. Suggesting he felt the pre-ringing is audible. Is that a knock against sharp filters with a lot of pre-ringing I leave for you to decide. The research into any of this sucks, so I am not sure we have much to look at. As I said, I can’t hear a difference.

Fully agreed, and I also have experience the different filters on one that have them selectable, couldn't hear a difference that people claimed they had, not even my my HD650 headphone..

On the theory side, I understood the "pre-ringing" thing, but if that's the issue, I would think people with extraordinary hearing may be able to hear the effects when using sampling frequencies 44.1 and 48 kHz when the Nyguist frequency would be at or slightly higher than 22,000 Hz. I couldn't imagine how they could hear the effects at high sampling frequencies. So to me, once again Placebo came to mind! Dr. Toole also said, if you know which one you are listening to......, it doesn't matter what you think...., so like speaker comparison listening (what Dr. Toole referred to with that statement), those who compare the pre-ringing effects between the use of slow and fast roll off filters need to do so in DBT, assuming all else (e.g. using the same DAC such as the AK4490 being one..) being equal.
 

Matthew J Poes

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Hi, there's been lots of discussions in another thread.
In actual fact, to talk about "aliasing" in a DAC situation (when there is no resampling) is not the correct terminology.
The artefacts in the spectrum are actually due to "imaging", which are a by-product of the sample-and-hold process.

can you post to both the thread and a source for this information?

min very simple terms, aliasing is when you see a mirror below the artifacts. Imaging is when you see it fold up. This is aliasing and you absolutely can have aliasing without resampling. I don’t understand where this came from.

Ive shared your comment with an engineer at TI who works in their chip development group and he wants to see the source of this. He doesn’t understand where it’s coming from.

if I am wrong (and be is wrong) then so are a number of text books used in college courses to teach digital systems.

unless we aren’t talking about the same thing.
 

Matthew J Poes

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Fully agreed, and I also have experience the different filters on one that have them selectable, couldn't hear a difference that people claimed they had, not even my my HD650 headphone..

On the theory side, I understood the "pre-ringing" thing, but if that's the issue, I would think people with extraordinary hearing may be able to hear the effects when using sampling frequencies 44.1 and 48 kHz when the Nyguist frequency would be at or slightly higher than 22,000 Hz. I couldn't imagine how they could hear the effects at high sampling frequencies. So to me, once again Placebo came to mind! Dr. Toole also said, if you know which one you are listening to......, it doesn't matter what you think...., so like speaker comparison listening (what Dr. Toole referred to with that statement), those who compare the pre-ringing effects between the use of slow and fast roll off filters need to do so in DBT, assuming all else (e.g. using the same DAC such as the AK4490 being one..) being equal.

The pre-ringing of the phase linearization filters references by Olive must have been tested with DBT’s. Maybe he was speaking in hyperbole here. We were not discussing DAC reconstruction filters. We were discussing if there is any merit in zeroing out the phase. His view is that they have studied it and found it inaudible other than the pre-ringing artifacts it introduces. That implies pre-ringing is audible as some kind of distortion in the signal. I can’t find a lot of research doing DBT or filters that introduce pre-ringing vs not and hold all else equal.
 

peng

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The pre-ringing of the phase linearization filters references by Olive must have been tested with DBT’s. Maybe he was speaking in hyperbole here. We were not discussing DAC reconstruction filters. We were discussing if there is any merit in zeroing out the phase. His view is that they have studied it and found it inaudible other than the pre-ringing artifacts it introduces. That implies pre-ringing is audible as some kind of distortion in the signal. I can’t find a lot of research doing DBT or filters that introduce pre-ringing vs not and hold all else equal.

I thought we were talking about the reconstruction filter.

Can you please post a link to his comments/studies on this? I am always interested in he's opinions/findings.
 

Matthew J Poes

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I thought we were talking about the reconstruction filter.

Can you please post a link to his comments/studies on this? I am always interested in he's opinions/findings.

I have no clue where he has posted this publicly. It was in a private conversation. We were disagreeing on the advantages/disadvantages of very steep filters and merit of different approaches to phase linearization. He was making his case that he thinks both are a waste of time.

he has cited research to me before that was never published. I’ve asked his team about this before and the answer I got was that most groups and most researchers are uninterested in publishing negative findings. As a researcher myself, this is actually a know problem with the field. It’s sometimes known as the file drawer effect. Basically, non-significant findings go unpublished and their “information” doesn’t go into the collective knowledge on a topic. Only the positive studies.

this is now totally off topic. However I can’t stop myself since I suspect most people don’t fully understand how research publications work, as most don’t conduct research that ever needs to be published.

there is a known bias amongst academic journals toward positive findings. Studies that fail to find anything are rarely published by the journals unless there is something really compelling in the study. This is more true today than in the past, so we find more negative published studies in the past too. This creates a systematic bias in the research base.

researchers also don’t tend to submit negative findings (I.e. those for which you failed to reject the null hypothesis or even those with findings contrary to your hypothesis) because researchers are unlikely to make tenure based on a body of failed research. Again, this introduced a bias. If we think in Bayesian terms, our priors don’t accurately reflect our true universe. There is now lost knowledge.

we can see this as some grand indictment of the scientific community. It is in fact a problem. But it isn’t so simple either. Remember the common research concept, the absence of evidence is not evidence of absence. This is more true than most realize.

Statistics, used in these studies, are both wonderful tools and horrifically misleading. In the hands of the inexperienced and poorly educated (on the topic of statistics) they get horribly misused. A hypothesis test, as used in these studies, is trying to test just one main hypothesis. But we aren’t. We are really testing 100’s or 1000’s of hypothesis in this one experiment. We only care about one, but the others are there. These are all the confounding factors in the study that could explain a change (or lack of change) in the estimate. In the absence of a lot of knowledge about the problem being tested and a sound theory for how it all works, both positive and negative findings are actually suspect.

mad a result of these statistical complexities, a negative findings can actually be explained by many factors. One of which is that no difference exists. However, we can’t say that because we don’t have enough information to know that. No matter how good the study seemed. Failing to find something (like an audible difference from linear phase filters) is not a finding itself. Hence why journals normally don’t publish them. Hence why researchers normally don’t try to publish them. Hence why institutions normally don’t promote researchers or give tenure when publishing large bodies of non-evidence. Remember, a failure to reject the null hypothesis is not a finding. It’s not evidence. It’s a non-finding and this non-evidence.

However there is always this distinct possibility that you have disproven a theory and this we have learned something important. Because we can’t prove the theory wrong (we can only fail to prove it right repeatedly) we end up with an academic bias.

because of type 1-2, M, and S errors, and because a lot of researchers are actually quite lousy staticians, simply publishing all null findings would be just as problematic. A lot of studies fail because the researchers kind of suck at their job, for lack of a better way of putting it.

all to say, I can’t find any published literature on the inaudibility of lots of things. When I talk to the experts about why they didn’t publish the null findings, this is essentially what they tell me.
 

Martin_320

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can you post to both the thread and a source for this information?

min very simple terms, aliasing is when you see a mirror below the artifacts. Imaging is when you see it fold up. This is aliasing and you absolutely can have aliasing without resampling. I don’t understand where this came from.
Ive shared your comment with an engineer at TI who works in their chip development group and he wants to see the source of this. He doesn’t understand where it’s coming from.
if I am wrong (and be is wrong) then so are a number of text books used in college courses to teach digital systems.
unless we aren’t talking about the same thing.

Ok, here's what I mean:
For an audio signal sampled say at 48kHz, then the spectrum of the 'played back' version -- if there is no reconstruction filter -- would include the original spectrum 0-24kHz, then a mirrored (ultrasonic) copy of that spectrum from 24-48kHz (ie greater than the Nyquist frequency), then a regular copy of the 0-24kHz spectrum at 48-72kHz, then a mirrored copy of that spectrum at 72-96kHz, and so on, stretching out to infinity...

ps. I've seen such mirrored images myself, when taking a spectrum snapshot of music in Audacity from the analog outputs of my AVP. In my case these mirror images were not stretching out to infinity because the 'slow' anti-imaging DAC "reconstruction" filter of my Marantz AVP reduced them post Nyquist.

Are we now talking apples-to-apples, or have I described something different to what you were thinking?
 

Matthew J Poes

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Ok, here's what I mean:
For an audio signal sampled say at 48kHz, then the spectrum of the 'played back' version -- if there is no reconstruction filter -- would include the original spectrum 0-24kHz, then a mirrored (ultrasonic) copy of that spectrum from 24-48kHz (ie greater than the Nyquist frequency), then a regular copy of the 0-24kHz spectrum at 48-72kHz, then a mirrored copy of that spectrum at 72-96kHz, and so on, stretching out to infinity...

ps. I've seen such mirrored images myself, when taking a spectrum snapshot of music in Audacity from the analog outputs of my AVP. In my case these mirror images were not stretching out to infinity because the 'slow' anti-imaging DAC "reconstruction" filter of my Marantz AVP reduced them post Nyquist.

Are we now talking apples-to-apples, or have I described something different to what you were thinking?

Yes I think we are on the same page now.

you are describing the fold up so that is imaging.

the professor I had when I took all of one course on this topic felt strongly that the fold down that then happens from the fold up should be called aliasing and so that might be pedantic semantics.

there is also noise in a digital to analog system that exists above the content. In the absence of a filter you get true aliasing. That aliasing introduces some distortion in the high frequencies. In the example I was referring to, the musical content was from a 44.1khz source. It was placed inside a 192khz container but not upsampled. It’s spectrum was analyzed in order to examine the recording side filter behavior of the engineer and A/D portion. It was never converted to analog on my side. The recording was taken directly from the playback software into Matlab. We originally thought it was imaging, but when we removed the source content and examined the spectrum we found it was a mirror image of noise that was somehow introduced. The individual helping me is a PhD digital systems researcher at Berkeley, so I trust he knew what he was doing, and he claimed the process introduced some random ultrasonic noise that normally is filtered out, but because my process was done filter free, allowed for aliasing. In Matlab we convolved the recording with a 96khz low pass filter and the artifacts went away so what he said seems to be right.
I have never bothered to measure the output of a Marantz product to see what imaging artifacts exist. I wouldn’t even think to do it if not for Amir. As he noted, most DACs and preamps are flat to 20khz. An AP uses no filter on the A/D which is an unusual arrangement. What I have used an off the shelf A/D chip from AKM which has an antialiasing filter. That filter can be moved up to 384khz but it can’t. They claim the chop can have up to 192khz of bandwidth but it’s just a digital trick, it’s a normal 192khz chip like all the rest and so it’s accurate bandwidth is only up to about 96khz. As such I don’t have the bandwidth to properly look at this like he does.

since I work for Gene at Audioholics and will soon live within 2 hours of him, I plan to be able to drive to his place and test on his AP myself in the future. Gene doesn’t measure frequency response the way Amir does and so had not noticed this problem.
 
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