Marantz AV 10 installed: - Early Review and Impressions.

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PaulBe

Audioholic Intern
But if you do the Klippel tests (you said you did right?), that it done blind, then how small such EQ changes would that be for it to be perceived? Just curious, that's all.



On this point, we obviously have to agree because it is factual, 100% objective, but we also need to keep in mind that, there is accuracy/transparency, and preference. One being objective, the other subjective.

For accuracy/transparency, it is easy to agree that if the input signal waveform is the same as the output signal waveform, that accuracy is achieved, and if not then it is not 100% accurate by definition.

So, cutting out frequencies > 22,000 Hz should result in the output signal inherently more accurate for transparency than preserving it because any frequencies resulted back in the audible band of up to 20,000 Hz due to the frequencies in the >20,000 Hz ultrasonic range will be a result of distortions such as intermodulation distortions. Some people may be able to detect such distortions and prefer their presence, but it is doubtful, in the case of the likes of the high quality processors such as the AV10, because the magnitude of such distortions will be too loud for it to be in the so called audible threshold anyway.



Their are sites that I know of, conducted blind listening tests on those effects, and if I remember right, most couldn't tell a difference while some could, though I don't recall any conclusions have been made so far.
IIRC the Klippel test included an addition of overall distortion at different levels. More or less distortion doesn’t change the tone. How does the Klippel distortion test relate to an EQ change? It’s apples and oranges. An EQ change at low Q is a tone control. An octave band EQ has a Q of 1.4. 3rd octave PEQ has a Q of 4.4. 5th octave PEQ has a Q of 7. Useful low Q PEQ changes the pitch of the EQ’d sound, and it doesn’t take much to hear a change - .1dB can make an audible change in pitch. It’s much different than adjusting overall volume. It’s great for separating the different vocals in a mix. Think voicing.

High Q PEQ is useful for feedback suppression, other resonance problems, and is hard to discern when used properly. I addressed this in a previous post - #223. For reference see your 12th octave smoothed graph in post #222, and my unsmoothed tweeter response graph in post #245. If I fix responses in those graphs at a Q necessary to make them smooth without artificial smoothing in the graph, I may create more problems than help, and I will quickly run out of available PEQ’s.

Floyd Toole addresses this in his ‘Sound Reproduction’ book. To paraphrase – If you make a response perfectly flat with EQ, you are probably doing something wrong to the sound. At any rate, EQ is not the same issue as Klippel distortion tests.

Note that the AV10 does not have native PEQ. It has octave band GEQ, which is a PEQ with a Q of 1.4, and fixed frequency points. @gene wants to see PEQ on the AV10. I think the octave band GEQ is good enough for most situations but perhaps could use some frequency adjustability. PEQ on the AV10 is part of Audessey, and can be manually adjusted after the fact.

~~~

Our ear/brain connection and DAC filters - I remember Archimago’s Musings having a discussion about filters. Oppo agreed with Archimago. The discussion was several years ago. There might be more at his site. Oppo referred me to Archimago when I inquired about filter settings on my player. I don’t remember what they said.

I just looked at my device settings to see what I’m currently using. I forgot what I set the filter at. It’s Linear Phase Slow Roll-off. I haven’t changed it for a long time. I heard differences and they are subtle. Perhaps I picked one that has more distortion.

Correction – I misspoke when saying the delay may be heard. Bad Grammar. Bad sentence structure. Bad PaulBe.
 
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TLS Guy

TLS Guy

Audioholic Jedi
IMO this is a real winner rated post! It explains a lot, though never totally I supposed, why "all amps sound the same" and "all amps don't sound the same" are both a matter of "facts".

As mentioned (or alluded to), by the time our ears and brains get to even start working, perceiving, the signal, since created in the recording and mastering process, has gone through so many electronic parts, connectors, cables, wires, solder joints etc etc etc...., so from one of the numerous online dictionaries:

when trying to do a bottleneck analysis on the audio signal chain, from the recording mics through the loudspeakers or headphones, then to the ears and brains, one must keep the following in mind, unless one is able to ignore what I call "logic":
  • A Chain is As Strong As The Weakest Link
  • A Chain is No Stronger Than Its Weakest Link
  • A Chain is As Strong As Its Weakest Link
  • You’re Only As Strong As Your Weakest Link
Now apply to the bs claims of HDAMs, let me repeat my logic:

a) Marantz AVR and AVPs: HDAMs cannot improve slew rate (see definition on Sweetwater website):



So if the op amp before and after the HDAMs are the same as those used in the audio signal chain of the corresponding AVRs, as well as Marantz own AVRs but the low profile series, then even if the slew rate of the HDAMs are "infinitely" fast, it will not mater become it ends up going through the same opamps that have much slower slew rate (assuming Marantz if right about the much faster slew rate of the HDAM opamps (yes HDAMs are opamps, just discrete, not IC type)

-b) Marantz integrated amps: HDAMs in those will more likely realize Marantz claim on faster slew rate, because they are used in quite a few more stages, vs the AVRs/AVPs, where they are only used in the stage just prior to the final opamp buffer (again, yes, I saw it in the schematics/block diagrams of the SR6014 that I purchased when it was available).

c) as PaulBe pointed out, HDAMs, even just for one single stage in the AVRs/AVPs, will likely have different output impedance than units without HDAMs.

d) HDAMs, should in theory, if Marantz had chosen to, improve the pre out voltage SINAD at higher voltage when driving power amps that has relatively lower input impedance, that is, they could be more effective in terms of "buffering".

Overall, HDAMs unlikely has audible effects, that's probably why Marantz, since the AV10, and the Cinema 30 has included the selectable dac reconstruction filter, so that the user could select the default or the optional filter as follow (Marantz website):


This particular claim while just a claim because "audible" or not is debatable and it certainly depends on the individual's hearing ability, or discernibility in the high frequency range as well as being subjective in nature, it is however, at least 100% logical, even when bottleneck analysis is correctly applied.

Bottom line, @TLS Guy has already confirmed the AV10 has superior surround sound performance, and is very audible to him, I hope he will try "Filter 1" and "Filter 2" and report back on whether he could easily hear a difference and then which one he prefers.

If I remember right, Gene said he could, or he thought he could? hear a difference and he prefers Filter 2 (NOT 100% SURE IF I INTERPRET THAT CORRECTLY FROM HIS TALK), that means he would have been just as happier with the Denon (A1H, A10H) sound too.;)

Finally, @ryanosaur , can you please also try Filter 1 and Filter 2 and gives us some feedback, thank in advance.

NOTE: Gene told us his preference between Filter 1 and 2 but he used the term that he prefers the one with the "wider bandwidth" so while I am somewhat confident he meant filter 2, there is an outside chance he might mean filter 1 because filter 1 will result in a roll off of about 2 dB by 20 kHz and about 2.2 dB at around 22 kHz, filter 2 will drop more at 22 kHz as it would act more like a brick wall at that point (ref: Nyquist criteria)

So at up to 44.2 kHz sampling, one can consider either filter as having wider bandwidth, I lean on believing Gene meant he prefers filter 2 because a) at 88.1, 96 kHz or above sampling, filter 2 definitely and clearly has wider bandwidth, and Gene mentioned it looked better on the test bench, since most bench test measure 20-20 kHz, and on that one, Filter 2's FR looks like a straight line vs Filer 1's roll off from about 12 kHz, so Gene most like meant Filter 2 look better on measurements. Marantz basically has the same narrative on that too:

View attachment 73794

Regardless, if @gene happens to notice my post, please clarify that your preference is in fact filter 2, or 1 (edit again: just realize in his review on this website he did make clear the one he preferred was the Filter 2), and thank you again for such in depth review plus a even more in depth review in the Youtube follow up video.

I would however, urge you to do another one when you have tried out DLBC, as I am quite sure you will get even better, smoother bass response with DLBC, with some minor tweaking. I am saying this based on my own extensive trial comparing Audyssey, ARC G and DLBC.
Since you are a good fellow PENG, I carried out your request.

I first did a hearing test, not audiologist grade, as that takes a patient and an operator. Anyhow I connected my Sens headphones to my oscillator and did a rough hearing test. I have not done that for 10 years at least. When I was in med school on the ENT rotation we had to do Audiograms on each other. Then I could hear out to 16KHz and have done I think pretty much most of my life. I think no more than a couple could hear to 20K and they were all women. 16 to 18K was the peak scatter, as far as I remember. Anyhow last time I checked I could still hear to 16K. However, in 18 months I will be 80, if I have not had my half day out with the undertaker before that. Anyhow I think I could just detect 16K, but to all intents and purposes my upper limit is now 15K, which for my age is very good. I could hear 15K clearly.

So then I did the test. I used the BPO BD Atmos disc and played the last movement of the Tchaikovsky fifth symphony, as that has plenty of action.
I have to say that during setup, I was skeptical and suspected BS.

However I have to report there is an easily discernable difference. I had a strong preference for filter one and did not like filter 2 at all. The biggest problem was the it collapsed the front to back sound stage. On filter 1 the orchestra had a natural perspective and seemed beyond the front wall where the screen is. The sound seemed very well balanced. On Filter 2 the perspective came forward and there was a slight harshness to the sound. The difference of the two settings was easily discernible. So, I have been in for rather a lot of surprises with this AV 10.

Now, I built and designed the speakers and the room. I design my systems for classical music, orchestras, chamber music, choral music, opera, piano, organ and other instrumental recitals. I do not design for rock and music which is what might be referred to as music in the pop domain. Having said that rock music engineers seem to like the system. That however was not what it was designed for. So, that has dealt with that disclaimer. After 70 years of speaker building I intuitively know how to get the sound stage I prefer.

So, I have to agree with the advice from Marantz in the user manual to select filter 1 if your preference is classical music. They don't particularly state what filter 2, is for, but my take it likely to put nasty rackets even more in your face.

So, to me in my system the two filters are easily discernible and I have a strong preference for filter 1. I have the impression that this filter is changing more than just FR, as I would not have thought I would have detected that change in FR.

I would surmise you should select Filter 1 for classical music, movie and TV watching, and instrumental jazz. Not sure what filter 2 is best for, but I did not like it.
 
P

PENG

Audioholic Slumlord
IIRC the Klippel test included an addition of overall distortion at different levels. More or less distortion doesn’t change the tone. How does the Klippel distortion test relate to an EQ change? It’s apples and oranges. An EQ change at low Q is a tone control. An octave band EQ has a Q of 1.4. 3rd octave PEQ has a Q of 4.4. 5th octave PEQ has a Q of 7. Useful low Q PEQ changes the pitch of the EQ’d sound, and it doesn’t take much to hear a change - .1dB can make an audible change in pitch. It’s much different than adjusting overall volume. It’s great for separating the different vocals in a mix. Think voicing.

High Q PEQ is useful for feedback suppression, other resonance problems, and is hard to discern when used properly. I addressed this in a previous post - #223. For reference see your 12th octave smoothed graph in post #222, and my unsmoothed tweeter response graph in post #245. If I fix responses in those graphs at a Q necessary to make them smooth without artificial smoothing in the graph, I may create more problems than help, and I will quickly run out of available PEQ’s.

Floyd Toole addresses this in his ‘Sound Reproduction’ book. To paraphrase – If you make a response perfectly flat with EQ, you are probably doing something wrong to the sound. At any rate, EQ is not the same issue as Klippel distortion tests.

Note that the AV10 does not have native PEQ. It has octave band GEQ, which is a PEQ with a Q of 1.4, and fixed frequency points. @gene wants to see PEQ on the AV10. I think the octave band GEQ is good enough for most situations but perhaps could use some frequency adjustability. PEQ on the AV10 is part of Audessey, and can be manually adjusted after the fact.

~~~

Our ear/brain connection and DAC filters - I remember Archimago’s Musings having a discussion about filters. Oppo agreed with Archimago. The discussion was several years ago. There might be more at his site. Oppo referred me to Archimago when I inquired about filter settings on my player. I don’t remember what they said.

I just looked at my device settings to see what I’m currently using. I forgot what I set the filter at. It’s Linear Phase Slow Roll-off. I haven’t changed it for a long time. I heard differences and they are subtle. Perhaps I picked one that has more distortion.

Correction – I misspoke when saying the delay may be heard. Bad Grammar. Bad sentence structure. Bad PaulBe.
My mistake, change in distortions should not be compared to change in EQ regarding discernibility. I don’t know why I posted the way I did at that moment.:D
 
P

PaulBe

Audioholic Intern
My mistake, change in distortions should not be compared to change in EQ regarding discernibility. I don’t know why I posted the way I did at that moment.:D
No problem. None of us gets it right all the time. If I got $10 for every time I had a brain burp, I'd be a millionaire. :)
 
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