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Review and Measurements of Marantz AV8805 AV Processor

amirm

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I am not claiming that it is state-of the art in it’s DAC and pre-amplification, no way as it is using second level DACs (top ones are AK4497 and ESS9038PRO at the moment) and audio grade, but just ok op amps NJM8080.
You haven't read a fraction of the testing I do here to associate IC selection with fidelity. It is the implementation that by far determines the final performance of an audio device, not what IC is used internally. The best DAC chip in the wrong hand produces horrible performance as I have shown in countless reviews. Here is the Melokin 9.1 which you said you like: https://www.audiosciencereview.com/...n-da9-1-es9038pro-dac-and-headphone-amp.3618/



That ES9038PRO is not doing you a darn bit of good, is it? Or maybe you are a fan of it saturating so early?



The top performing DAC in THD+N measurements is the Okto DAC8. It uses ES9028 and gets a SINAD of 118 dB. The Melokin 9.1 despite using ES9038Pro stops at 100 dB for a whopping 18 dB higher noise+distortion.

As I said, if you want to ignore audio engineering/science, you are welcome. Here we go by what we can prove, not what we want to prove.
 
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You haven't read a fraction of the testing I do here to associate IC selection with fidelity. It is the implementation that by far determines the final performance of an audio device, not what IC is used internally. The best DAC chip in the wrong hand produces horrible performance as I have shown in countless reviews. Here is the Melokin 9.1 which you said you like:
Ok, it seems you are not tired anymore, I have rested as well, let’s continue. Melokin (let’s call him using his brand’s name, I don't remember the guy’s name) neither designs the USB to i2s convertor nor writes the drivers for it. He designed the hardware and provided an option to install second party’s USB to i2s convertor named Amanero. It is designed by an Italy based company Amanero Technologies, which also provides Windows drivers for their boards - for free. As you have figured out by yourself, the problem with linearity is not associated with the hardware design itself - using SPDIF port, which doesn’t require a driver, the linearity looks good enough for $300 DAC. It is obvious that the problem is related to two things - the USB input/Convertor (and the standard Windows driver), and also with your 0 d FS input level. I will talk about the input level a bit later, but tell me why didn’t you even try to use a native Amanero Windows driver for your test?
Second thing - your graphs can be used for YOUR Melokin DAC ONLY, they can not be true for my Melokin. The reason is simple - I don’t use either USB input or standard Amanero convertor, I use an advanced version of the convertor (if necessary) and SPDIF input (usually). There is another reason - 0 dB FS measurements can not be approximated to other similar devices, period. Why? I will tell you a bit later.

That ES9038PRO is not doing you a darn bit of good, is it? Or maybe you are a fan of it saturating so early?
There is nothing to do with ESS9038PRO - see above.

The top performing DAC in THD+N measurements is the Okto DAC8. It uses ES9028 and gets a SINAD of 118 dB. The Melokin 9.1 despite using ES9038Pro stops at 100 dB for a whopping 18 dB higher noise+distortion.
ONLY the devices you have tested, period. It can not be said about other Okto DAC8 or Melokin devices being tested at 0 dB FS. See below - why.
"Just acceptable" after spending $4,500. There are plenty of just acceptable multi-channel products out there at far lower prices.
First - I have spent about half of that price. Second - my level of demand is pretty high, and majority of stuff I have auditioned (and I auditioned a lot of devices at all levels) fits in the category “just acceptable”. Exceptional devices like dCS or EMM Labs are not acceptable for me due to their price level.

AES standards are created by the industry so at times you want to be careful in accommodations they make for companies. -1 dBFS to allow filter overruns and such is one. So in this regard, my standard is more rigorous.
This is Major misunderstanding of the DAC functionality and the standard. Look what happens when people try to test commercial products at 0 dB FS: https://service.tcgroup.tc/media/Level_paper_AES109(1).pdf
Check out those players measurements. 7 tested, none passed. Are they all “bad players”? Are those players which had been tested, bad ones?
Chris Tham published a perfect article regarding the issue: https://www.audioholics.com/audio-t...dbfs-levels-on-digital-audio-playback-systems
And finally, there is an approximation of 0 dB FS to analog level https://www.av-iq.com/avcat/images/documents/pdfs/measdigaudlvls.pdf
A 0 dBFS measurement unit is to be the highest audio level. Assuming this is to be the highest audio level before clipping occurs, this corresponds to an analog level of 24 dBu.
Not all professional products can output 24 dBu, just the best ones.

Now let me tell you what is happening in real life devices at 0 dBFS. Let’s say DAC outputs this level properly to the analog stage, but such as it is the last step before clipping occurs, we have to take in consideration the tolerance of analog components used in that stage. Usually it is +/-5% for commercial devices, so some devices will start clipping and we will get Nielsen’s results. Some of them with no tolerance (or negative tolerance) will pass. Can we say that all measured devices are good or bad? No we can not. We can say it only about the tested device.

But there is another problem with a DAC. They all use a reference voltage to create analog output. What if that voltage also has a tolerance (either the reference voltage source or a resistive divider)? We will also get the clipping ruining the test results, and some people after such test even start blaming the tested devices ...
Now let’s get back to Marantz 8805. Per data sheet AK4490 DAC has maximum analog output voltage min +/-2.65V, typical +/-2.8V, and max +/-2.95V, according to note 10 it is p-p value. Those levels equal to 1.87V RMS, 1.97V RMS and 2.08V RMS accordingly. Let’s check what is used after the DAC:
https://hometheaterhifi.com/reviews/receiver-processor/processors/marantz-av8802-processor-review/
The AV8801 output stage had a gain of two. The AV8802 output stage is now unity gain. ... The signal at the output of the DAC, which is before the volume control, is doubled in the AV8802 to compensate for the gain reduction at the output.
OK, it is doubled, but it is still within the max output limits, where is 4V RMS?
The design is based on the output capabilities of the DAC chip, connected to the unity gain op amp. Is it wrong? No, the signal part from the DAC to the output is maximally simplified to avoid added distortions, like for example in Denon 5803, which had 7!!! Op amps after DAC per channel.
Now to the voltage reference level for the DACs, it is formed by PQ050DNA1ZPH chip, per it’s data sheet the output voltage can be min 4.875V DC, typ 5.0 V DC, and max 5.125V DC, it is 2.5% tolerance. What if that chip will have a bit higher voltage, but still within the tolerance limit? This DAC will accept 0 dB FS signal, but due to the 2% DC reference level shift the output stage will start clipping. Is it a design problem? No, it it 0 dB FS problem / tolerance of the components problem:
  • 0dBFS+ levels cannot be generated by recording from an analog source. No matter how high the recording gain level is set at, the digital recording will never contain 0dBFS+ levels. Of course, if the recording gain is set too high, digital "clipping" will occur, but this is not the same thing as 0dBFS+ levels.
  • 0dBFS+ levels can be created through subsequent manipulation or processing of a digital recording. For example, if a digital recording is "amplified" (by multiplying each digital sample by a constant amount) it may then subsequently lead to 0dBFS+ levels.
  • Therefore, 0dBFS+ levels are technically "illegal" states and represent anomalies or artifacts created by processing signals in the digital domain.
    [*]If a player does not handle 0dBFS+ levels it is NOT strictly speaking the fault of the player. There is no "requirement" in the CD specification for players to be able to handle 0dBFS+ levels and arguably the ability to handle 0dBFS+ levels are a violation of specifications.
Ref. Chris Tham
Now what exactly is -1 dB FS level per AES 17 standard? It is enough to cover all component’s tolerances. (-1dB=-12% decrease). Is it “too old” and 0 dB FS testing is “more reliable”?
May be, but not for commercial products. 24 dBu is used for professional production, as well as 1% components tolerance. Thus if you test a commercial product be ready that one of them will fail, and another one will pass. It doesn’t give a right to either blame or praise such product basing “conclusions” just on one test. Make them statistically correct and then call such test results “a science”, otherwise they are equal to some subjective opinion and will always be different.
 
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You lost that battle anyway
Hmm, I was under impression that it was a technical discussion. Battles happen between enemies ... do you consider all forum subscribers who dare to argue with you as enemies and fight with them winning your “battles”, or just me?
 

March Audio

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I am tired of arguing with you. You clearly are not listening. I could have done three other reviews for the amount of time it has taken to explain all this to you. :(

So do this: tell us what measurements from wherever shows this processor to be state-of-the-art in its DAC and pre-amplification.
@amirm sometimes the ignore button is the only option. @ProFan does not have adequate technical understanding to enable a reasoned debate.

Whatever way you slice it this unit does not provide the high end performance implied from its position in the manufacturers range, its marketing or price.
 

Blumlein 88

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@amirm sometimes the ignore button is the only option. @ProFan does not have adequate technical understanding to enable a reasoned debate.

Whatever way you slice it this unit does not provide the high end performance implied from its position in the manufacturers range, its marketing or price.
There has been some indication that Marantz is using similar designs for several models and years. And over their mid as well as top of the market processors. The specs in their manuals remain nearly completely identical.

If I look at testing I did with the older 7701 Marantz, it is within a couple db of what Amir has gotten on this newer far more expensive model. And for @ProFan I tested with - 1dbFS signals and with gain settings for -1 db peaks on the ADC. I also test some signals over a slow sweep from -20 to -1 dbFS. So the poor showing isn't some accident of variable tolerances due to using a 0 dbFS signal. I had results right in line with Amir's using completely different test signals, methodology, and using a good recording interface instead of an AP unit. So whatever critique of Amir's methods you may have they weren't a one off result that unfairly paints the performance of this processor. Do you think the small change in signal level will cause this processor to jump up 10 db in most performance metrics? It would be far beyond Marantz's claims for the unit.
 
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I have one of the "slim" models, the 1508. I had space limitations and only needed stereo + sub. It works fine for my purposes. As I wrote before in the thread, I also have a DAC with an AK chip in the same series (this one, https://www.audiophonics.fr/fr/dac-...-384khz-asynchrone-usb-xmos-noir-p-11961.html, AK4495 chip vs AK4490 in the Marantz).

As explained in my earlier post all I can do is subjective testing, i.e based on what I hear, but the difference between the two for music listening isn't even subtle, with the separate DAC winning by a long, long mile, contrary to what an earlier comment seemed to imply. The sound is downright engrossing and I could listen to it for hours, while I would use the Marantz only for background listening, radio, podcasts and such. The Audiophonics DAC obviously is optimized, looks well built from the pictures and uses an "R-Core transformer" although I only have a foggy notion of what this implies. For TV and movies the Marantz is fine and everybody loves it. I'm curious though about comparing the sound on movies with the TV sound routed through the separate DAC's optical input. I came to do this testing with no preconceived notions, in fact I had been planning to remove the DAC for use in a different room, and would have preferred the Marantz to perform equally well for music, and now I'll have to buy a second DAC.
 

edechamps

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Besides, there is a STANDARD in stone, defined in the AES17 document:
8.5.1 Total harmonic distortion and noise versus frequency
The measurement should be conducted with a sine wave at – 1,0 dB FS and repeated with a sine wave at – 20 dB FS.
What version of AES17 are you using? I have the latest version (2015). It doesn't have that specific text, and the section numbers are different. The text in AES17-2015 is:

6.3.2 THD+N ratio vs frequency
[...] The test signal level shall be a sine wave at -1 dB relative to the maximum input level (see 6.2.1). [...]
Note that this is not quite the same as -1 dBFS. Here's what §6.2.1 states:

6.2.1 Maximum input level
[...] One of the following methods shall be used to determine the maximum input level: [...] THD+N ratio method. The THD+N ratio at the EUT output is measured (see 6.3.1). The level of the test signal is increased until the THD+N ratio reaches -40 dB. [...] The THD+N ratio method is recommended for EUTs that hard clip [...]
At this point it's now quite obvious why they specify -1 dB in the THD+N measurement: using 0 dB would not provide interesting data, as 0 dB is, by definition, -40 dB THD+N at 997 Hz (and so is likely to have very bad THD+N across the spectrum). There's not much point in measuring at that level. So allow me to express scepticism when you say:

Now what exactly is -1 dB FS level per AES 17 standard? It is enough to cover all component’s tolerances. (-1dB=-12% decrease).
I did a quick test just now to determine what at what input level a pure digital chain would reach -40 dB THD+N (997 Hz sine wave, 48 kHz SR, 0-20 kHz measurement). The number I arrived at is approx. +0.35 dBFS. (Someone should check me on this - I measured this the quick and dirty way.) Therefore, if one were to follow the letter of AES2017-2015, the "maximum input level" is 0.35 dBFS (assuming input digital clipping happens first), and the signal level for THD+N vs. frequency measurements is -0.65 dBFS, not -1 dBFS. Granted, that's still not the level @amirm is using, but it's closer.

Also, keep in mind that AES17-2015 only mandates this level specifically for THD+N vs. frequency measurements. For THD+N vs. level measurements, AES17-2015 goes all the way to the maximum input/output level, which, as I explained above, is actually above 0 dBFS!
 
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There has been some indication that Marantz is using similar designs for several models and years. And over their mid as well as top of the market processors. The specs in their manuals remain nearly completely identical.
It is not true. See the link above - even 8801 is different, it uses +6 dB amplification after the DAC chips. 7701 per it’s service manual uses AK4458 DAC chips, which are different than used in 8802/8802A/8805 DAC chips AK4490 (all three use unity gain after the DAC chip). Also 7701 uses +6 dB amplification after the DAC, so it can output 4V RMS without any problem taking in consideration max output voltage levels of AK4458 DAC chip per it’s data sheet.
It is doubtful that measurement results for 7701 can be approximated and used for 8805 due to the difference in hardware and software designs.
 
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For THD+N vs. level measurements, AES17-2015 goes all the way to the maximum input/output level, which, as I explained above, is actually above 0 dBFS!
Regarding testing at and above 0dB FS ref the links above - Nielsen’s document, and Chris Tham’s article. Also ref Rane’s article - regarding 0 dB FS:
https://rane.com/note169.html
Such as 0 dB FS is just at the limit of clipping, it us so easy to pass it and get that clipping just due to the tolerance in components besides other factors like drift of parameters due to the temperature change, etc.
And I agree with you, I shouldn’t use -1 dB =-12% analog level comparing it to -1 dB FS digital level, it is not correct.
 
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edechamps

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Regarding testing above 0dB FS ref the links above - Nielsen’s document, and Chris Tham’s article. Also ref Rane’s article regarding 0 dB FS:
https://rane.com/note169.html
I don't understand what you're trying to say here. I am literally reading this off AES17-2015 directly. (Just to clear things up, when I say +0.35 dBFS in this context, I of course mean a sine wave 0.35 dB above full scale that is hard clipped to make it fit into digital samples. I am not suggesting the use of an actual +0.35 dBFS sine wave, which I don't think can be done at the specified 997 Hz test frequency anyway.)
 
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edechamps

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Again, I still have no idea what point you're trying to make. Please explicitly state your argument, and how it contradicts (or augments) what I said, instead of just throwing links around.

I don't care whether 0+ dBFS signals should be "illegal" or not. (I wish they were, but, sadly, they are plentiful in real world content, so we can't really ignore them.) I'm merely reporting what AES17-2015 says. If you believe that my interpretation of AES17-2015 is wrong, please state explicitly which one of my statements you take issue with and why. If you believe that AES17-2015 itself is wrong, well, that's your prerogative, but it might be more productive to debate this with the people at the AES in charge of drafting such standards. (There is also the more philosophical issue of whether it even makes sense for a standard to be "wrong" in the first place.)
 
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My point us simple - we can not rely on measurements, performed at 0 dB FS. They are too close to the level of clipping and even a good “passing” unit can give us bad results due to different type of fluctuations. We also can not approximate such results for the same product, because tolerance in components can make it either passing or not. Nielsen proved that 7 tested by him players perform really badly being tested at 0 dB FS. Are they all “bad players”? What if the same player will have enough room (again due to tolerance in components, for example) and pass?
The difference in THD+N measurements vs frequency in clipping and non-clipping mode is huge. Clipping can also make additional artifacts- significant raise of THD+N at certain frequencies, way more artifacts in multi-tone tests, etc.
Thus, I would prefer to see -1 dB FS testing results, they are very close to 0 dB FS in level, but at least provide some headroom before clipping occurs. I believe such headroom is enough to avoid clipping due to tolerance in components and drifting of parameters due to temperature or other factors. The results of measurements would be more reliable and will provide similar results for the same products. Such requirement is defined in AES17 version I had, and it looks logical to me.
 
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Frank Dernie

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I don’t listen at max anyway, this is true. If your DAC has a level adjustment (like most DAC do), there will be no problem, but why do we need that amount of headroom if it is not used? If the DAC doesn’t have a volume adjustment, it could be a problem - overloading of the input stage of that pre-amp (not all of them will absorb that level).
True I haven’t been in the market for a DAC for a while but I still own several DACs and NONE of them have a variable output as standard. I have one DAC/preamp which (obviously) does and a DAC/headphone amp which equally obviously needs one but all 6 DACs I own have a fixed output. So your are certainly wrong there
 

Blumlein 88

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Having seen measures of three Marantz units, I think the situation is that Marantz is using the same analog output/preamp circuitry across their line of products. I've not seen the schematics of their circuits. I believe that is the bottleneck to their performance. With the 7701 distortion was .002% at 3.4 volts. It rose to .005% at 4.5 volts output. Amir measured .0027% at 4.13 volts. All of those are in the same ballpark.

The 7701 gets to a bit over 1% at 12 volts output. Performance isn't being misrepresented by some difference in -1 and 0 dbFS.

You can feed it a lowered digital level like - 12 dbFS and turn up the volume 12 db and you'll get identical results in distortion vs 0 dbFS without the gain. The gain raises the noise floor just a little. So I think the dominant factor is the distortion and possibly noise of the analog output circuit. That is why you don't see much difference even if they use different DAC chips.

Now I'm not bothered by -90 db of THD (which is primarily 2nd and 3rd harmonic) at full scale of what I'll use. The noise floor is low enough. Then again it seems that Marantz could develop a new and better circuit and use it for several years. It is certainly possible to have better analog performance without it costing a huge amount of money. It also bothers me a bit they have what is supposed to be their flagship processor at a very high cost which has its audio performance hampered by this so that it performs the same as their least expensive stand alone pre/pro. It seems likely it has the same output circuitry. Marantz claims the same specs across all of these products and have for several years. They actually slightly outperform claimed performance. .005% doesn't look too bad, but it is only -86 db which is hardly incredible performance.
 

Blumlein 88

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Wrong in what? I have described both types of DACs, so what is your point?
“Like most DACs do?”? Ok, no problem, I can take it back.
I'd like to know what your point of contention is. Do you think the slight change in level would boost the 8805 test results? If not, then what is the point? As indicated in my last post, it doesn't appear the limit to performance is the DAC chip.
 

Blumlein 88

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My point us simple - we can not rely on measurements, performed at 0 dB FS. They are too close to the level of clipping and even a good “passing” unit can give us bad results due to different type of fluctuations. We also can not approximate such results for the same product, because tolerance in components can make it either passing or not. Nielsen proved that 7 tested by him players perform really badly being tested at 0 dB FS. Are they all “bad players”? What if the same player will have enough room (again due to tolerance in components, for example) and pass?
The difference in THD+N measurements vs frequency in clipping and non-clipping mode is huge. Clipping can also make additional artifacts- significant raise of THD+N at certain frequencies, way more artifacts in multi-tone tests, etc.
Thus, I would prefer to see -1 dB FS testing results, they are very close to 0 dB FS in level, but at least provide some headroom before clipping occurs. I believe such headroom is enough to avoid clipping due to tolerance in components and drifting of parameters due to temperature or other factors. The results of measurements would be more reliable and will provide similar results for the same products. Such requirement is defined in AES17 version I had, and it looks logical to me.
Well at the very least some of the products tested using 0 dbFS have performed wonderfully. And cost less while doing it. Its the analog performance that is hurting the Marantz anyway, not the digital portion. I test with -1 dbFS myself, but I've tried a few things at 0 and haven't seen anything fall completely apart when I did. Doesn't seem a problem with most designs.

You can see all kinds of weird bordline cases. I've one DAC which goes effectively crazy if fed max level above 21 khz while in 44.1 mode or above 22.5 khz in 48 mode. But reduce it .1 db and it is fine. It shouldn't do that, but then again there shouldn't be that level of signal in any normal music file.
 
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