Marantz AV 10 installed: - Early Review and Impressions.

[PaulBe]

So, you are using the Distribution(LFE) setting to get LFE to the speakers set to Large? At what level is it set?

Two channel signals do not contain LFE. When playing such signals, at what level are the lower frequencies copied to the subwoofers using the LFE+Main setting? Are you using the default 80Hz setting or have you adjusted it using the new Bass Extraction LPF setting to raise or lower it?

Do you use different settings for playback of two channel signals using the Dolby Surround up mixer compared to playing those same signals in Pure Direct/Direct mode or Stereo mode? Manual settings to Graphic EQ?

It seems the more questions I ask, the fewer answers I actually get in return. It is as simple as posting the settings used to adjust the sound to your liking without the reminders of the capabilities of the speakers in use.

Different DSP will give different results for sure but some signals cannot be crossed up mixed at all. Dolby Atmos and DTS:X signals cannot be cross up mixed. But, a blu-ray player set to output PCM will deliver the multichannel signals of both formats sans metadata and one can cross up mix those signals to one’s delight.

Here are my answers to this conversation because I want to know how other AV10 owners use the LFE Distribution function:

LFE Distribution –

Surround and Back: -14dB. At -14dB, it ‘appears’ to add just enough fill to even the LFE sound in my room without making me feel like I’m in a compression chamber. Adding any more volume makes me think of Neo in the movie Matrix, expanding and contracting the matrix when he breathes.

I don't know how the LFE distribution volume settings interact with the channel volume settings. You get a range of -20dB to 0dB. Pick a low setting and start experimenting.

I don’t add any LFE to LCR. LCR is already doing the heavy lifting. My 2 dedicated Subs are already in front – between L&C and C&R.

Using LFE Distribution gives me 6 distributed Subs in the room without the need for 6 dedicated Subs. Full range speakers are needed to use this function. Using LFE Distribution the way I do requires only a little more amp power to the Surrounds and Backs, and still much less than is used by LCR. All 7 floor channels use the same drivers.

LFE low pass filter is set to 250Hz.

I may experiment more in the future. It’s not on my priority list. The settings I use may or may not work for someone else.

Special thanks to @gene for convincing Marantz to add this function.

There is a 2ch Playback mode in the speaker settings for crossing over or adding 2 channel bass to the LFE channel. Haven’t used it yet. Nothing to say yet. See the manual.

 

[PaulBe]

You are at 0db of what exactly? Geez, don’t f#%kin’ make me beg for more like that poor ragamuffin Oliver!

I love that movie! The later version is out now on 4K UHD disc. I'll wait for a sale.

 

[TLS Guy]

You are at 0db of what exactly? Geez, don’t f#%kin’ make me beg for more like that poor ragamuffin Oliver!

Sorry I made a typo. Default is -20db, I have corrected it. I have figured out how to set the LFE to each speaker.

 

[PaulBe]

Sorry I made a typo. Default is -20db, I have corrected it. I have figured out how to set the LFE to each speaker.

There are off settings, and 4 individual ranges of -20dB to 0dB in 2dB steps for Front, Center, Surround, and Surround Back for me. There is no default volume. The number of full range floor speakers that can be used for distributed LFE, and are displayed in the menu, is determined by your speaker layout settings.

 

[TLS Guy]

There are off settings, and 4 individual ranges of -20dB to 0dB in 2dB steps for Front, Center, Surround, and Surround Back for me. There is no default volume. The number of full range floor speakers that can be used for distributed LFE, and are displayed in the menu, is determined by your speaker layout settings.

On mine when you bring up that menu, all the selected speakers show -20db. Anyhow I have set each individually now.

I am sitting back and listening at the moment. May be later I will get some measurements, but I like what I am hearing right now. Right now it is transporting me to the beautiful chapel at King's College Cambridge, which is a space I know. It has a wonderful acoustic, which is the stuff of legend and the rig is reproducing it very well.

I have to say that setting up this AV 10 is somewhat of a marathon.

 

[PaulBe]

On mine when you bring up that menu, all the selected speakers show -20db. Anyhow I have set each individually now.

I am sitting back and listening at the moment. May be later I will get some measurements, but I like what I am hearing right now. Right now it is transporting me to the beautiful chapel at King's College Cambridge, which is a space I know. It has a wonderful acoustic, which is the stuff of legend and the rig is reproducing it very well.

I have to say that setting up this AV 10 is somewhat of a marathon.

What kind of measurements will you make for distributed LFE?

The AV10 has a lot of advanced features that take some time to determine their individual usefulness for a particular application and implementation. So many options... I will never use some of the features because of my own system limitations. The animated OSD setup screens could also be used by a 12 year old to do a basic setup.

I love good recordings that transport the listener. There are some in every venue.

 

[TLS Guy]

What kind of measurements will you make for distributed LFE?

The AV10 has a lot of advanced features that take some time to determine their individual usefulness for a particular application and implementation. So many options... I will never use some of the features because of my own system limitations. The animated OSD setup screens could also be used by a 12 year old to do a basic setup.

I love good recordings that transport the listener. There are some in every venue.

I can't do that one for sure. But I can get a good idea of the room response. Most of what I listen to does not have an LFE channel, so that is not a big issue.

 

[TLS Guy]

So, I have been tinkering with the levels and crossovers for the last several days. I have been taking multiple measurements with Omni mic at the MLP which is a long way from the front speakers. No Audyssey is engaged.

The measurements were made from the Omni mic disc. Now this has 5 bedlayer channels, but I used the Dolby Atmos upmixer, so that all 11 channels were driven including the two sub channels.

This is the room response at the MLP.

For an in room response with 1/24th octave smoothing. For an in room response at the MLP that is pretty good and the FR slope what you want for a realistic sound field at the MLP.

This is the response of the sub 1 and 2 outputs, and so this is the response of the long transmission lines of the main right and left speakers. The drivers are two 10" SEAS Excel drivers in each line, so four drivers total.

20 Hz is the lower limit of omnimic, but you can see the output from those lines is well maintained to 20 Hz. No wonder the things shake the floor and rattle the crockery downstairs when the grandkids are watching action movies. Actually large organs do it as well.

So, you don't need 18" or 24" drivers to get that response if you correctly harness natures physical laws.

I am very happy with the way it sounds now and will leave well alone.

I have to say that changing pre/pro in a large installation is a real PITA to say the least of it. So, I hope the AV 10 has long legs.

I bought the AV 10 because it was available and had a track record, which the AV 20 does not, but I anticipate it will do fine. I have not seen reports of an AV 10 failure, but maybe a member knows of one. Having the AVP7705 fail at 5 years and the 7706 fail at three months I don't think is bad luck. And honestly their SNR is not good enough. So, I think it reasonable the relegate those units to junk status.

 

[AcuDefTechGuy]

I bought the AV 10 because it was available and had a track record, which the AV 20 does not, but I anticipate it will do fine. I have not seen reports of an AV 10 failure, but maybe a member knows of one. Having the AVP7705 fail at 5 years and the 7706 fail at three months I don't think is bad luck.

Technology and engineering (at least with Sound United) have probably significantly improved over the past 6-7 years.

 

[Syphirioth]

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.

Yes personally I would say don't cut out on high end. My personal opinion is that I somehow hear or notice it with systems ending arround 25,000Hz. My Dali Lektor are rated 35,000 Hz if I remember correctly. And I always notice the difference. Don't ask me why or how. Maybe it hallicunations.

But ye the mosquito high frequency device also keeps me out of a room. I have heared LED light projectors no one heared. I heared the old bulb tvs. Also no one heared... Ye pretty annoying if you hear I can tell.

 

[PENG]

Yes personally I would say don't cut out on high end. My personal opinion is that I somehow hear or notice it with systems ending arround 25,000Hz. My Dali Lektor are rated 35,000 Hz if I remember correctly. And I always notice the difference. Don't ask me why or how. Maybe it hallicunations.

But ye the mosquito high frequency device also keeps me out of a room. I have heared LED light projectors no one heared. I heared the old bulb tvs. Also no one heared... Ye pretty annoying if you hear I can tell.

Of course, as long as people keep in mind that opinions do not always mean "facts". It is widely accepted that humans cannot hear frequencies above about 20,000 Hz, so the CD limit of about 22,000 Hz is transparent, except to a small % of gifted people.

As far as you hearing even the difference up to around 25,000 Hz or the Dali's rated 35 kHz making a difference (audibly??), then I can only refer to Dr. Toole's finding on the issue with people doing their AB comparisons without following protocols to eliminate the "bias" factor (ie the listener is aware of....).

I also prefer media contents (digital files) that have sampling rate at least 48 KHz, but it is more about recording/mastering quality. For recording, I would prefer those done with 96 KHz sampling rate or above.

Here's one of many article on this topic, I don't necessarily agree totally with any of them, but I do like reading some of them, those written by people who understands the theories.

Some simple talk on the pros and cons:
48kHz vs 96kHz: Differences, Which To Use (And WHY)

and a more in depth one, that includes info about the negative effects of using sampling rate <48 kHz:

SoundStage! Hi-Fi | SoundStageHiFi.com - Understanding Digital Music -- What Bit Depth and Sample Rate Really Mean: Part Two

For music playback though, I am fine with 48 kHz but always prefer 192 kHz only because digital files I downloaded/purchased online tend to have more transparent quality, though there are no shortest of those at 48 or even 44.1 kHz that sound superb vs some of my 192 kHz or above files, that tells me the better SQ ones are due to the better recording/mastering process, not because of the final playback sampling rate.

Sorry about digressing, will stop right here so as to get back of the AV10 topic.

 

[PaulBe]

that tells me the better SQ ones are due to the better recording/mastering process, not because of the final playback sampling rate.

Yes. Once we get to a frequency range that covers the accepted audible range, 20-20K, the major 'bottleneck' is recording quality.

If possible, people should try to listen to their own first generation unprocessed recordings. It doesn't have be anything impressive, nor use expensive recording equipment. It could be a mic'd conversation in your own listening room. You will get a sense of what your system is really capable of doing, a sense of what your room does to reproduction, and show you the limits and attributes of microphones compared to human hearing. If you can, do this a couple of times with different microphones.

Note: Recording quality is only partially objective. It is also a major part of the art in recording.

See this reference about ambisonics:

Ambisonics - Wikipedia

en.wikipedia.org

See the animated gif under "Virtual microphones". Which mic pattern is the most accurate?

 

[Syphirioth]

Of course, as long as people keep in mind that opinions do not always mean "facts". It is widely accepted that humans cannot hear frequencies above about 20,000 Hz, so the CD limit of about 22,000 Hz is transparent, except to a small % of gifted people.

As far as you hearing even the difference up to around 25,000 Hz or the Dali's rated 35 kHz making a difference (audibly??), then I can only refer to Dr. Toole's finding on the issue with people doing their AB comparisons without following protocols to eliminate the "bias" factor (ie the listener is aware of....).

I also prefer media contents (digital files) that have sampling rate at least 48 KHz, but it is more about recording/mastering quality. For recording, I would prefer those done with 96 KHz sampling rate or above.

Here's one of many article on this topic, I don't necessarily agree totally with any of them, but I do like reading some of them, those written by people who understands the theories.

Some simple talk on the pros and cons:
48kHz vs 96kHz: Differences, Which To Use (And WHY)

and a more in depth one, that includes info about the negative effects of using sampling rate <48 kHz:

SoundStage! Hi-Fi | SoundStageHiFi.com - Understanding Digital Music -- What Bit Depth and Sample Rate Really Mean: Part Two

For music playback though, I am fine with 48 kHz but always prefer 192 kHz only because digital files I downloaded/purchased online tend to have more transparent quality, though there are no shortest of those at 48 or even 44.1 kHz that sound superb vs some of my 192 kHz or above files, that tells me the better SQ ones are due to the better recording/mastering process, not because of the final playback sampling rate.

Sorry about digressing, will stop right here so as to get back of the AV10 topic.

Maybe the high end is more open/clear with higher top frequency on speakers?

It not like a sound at all what I experience. It like penetrating my skull and it is realy annoying when it a long non stopping tone. I not hear it in my ear like sound but more like feel it between my ears. I have no idea what frequency range it exactly is. Noticing it might be the best description.

Best way how I can describe it is. Some sound systems lack so much detail in the high end somehow that it not the same as real sounds happening in my experience. My opinion. I not claiming I can hear up to 25k. I just noticed that the systems I hear and have higher frequencies seem to give me a better listening experience.

So can be many things what creates that experience. Maybe resonance in my head. I don't know. I just know that I like to keep frequency ranges beyond what the human can hear cause it adds to the experience in my opinion.

 

[PENG]

So can be many things what creates that experience. Maybe resonance in my head. I don't know. I just know that I like to keep frequency ranges beyond what the human can hear cause it adds to the experience in my opinion.

Exactly, too many things indeed.. But, the salient one is "bias", psychological I guess, that's why double blind tests were done when there's no other way to eliminate cognitive "biases".

Cognitive Bias List: 13 Common Types of Bias

There are always unknowns but on the known facts side of the equation:

- People can't (or maybe a small % could) hear frequency higher than 20-22 kHz

As to sampling rate above 44.1 kHz, that would be from 48 kHz and up, all information within the audio band of 20-22 kHz are captured, but sampling rate of higher, say 88.1 kHz would capture frequencies (if recorded in the content for playback) up to about 44 kHz.

So the extra high frequencies captured are really for other animals, not humans, but there will be distortions resulted back in the audio band for various reasons, one being intermodulation distortions that are subject to some known facts also such as:

- Such IMD resulted in the audio band would be of very small magnitude so typically would be below the threshold of audibility even if it's there.

- If such IMD are significant enough (not likely if bench test results did not show..), it would make perceptible sound quality worse because IMD are supposedly more objectionable than harmonic distortions for the obvious reason, though it is always possible some people might like certain kind of distortions, if they could hear/detect their presence.

Bottom line, anyone can say they heard better sound quality (to them) if the ultra sonic frequencies are in the playback contents, and the playback devices including their speakers can reproduce them accurately, but there's no proof, and no known science that supports the scenarios.

Also, if there are unknown facts that would otherwise explain the claimed "better sound", then the whole thing would be random, at least in the electronic devices side. For example, if the designers, engineers etc., don't really know what are the factors totally unknow to them could improve sound quality, then they would have no way to design/implement their gear. Marantz claims they use their "sound masters" to fine tune the sound with their designers, but that's illogical, hilarious actually.. As Peter Walker mentioned in the long past, engineers don't just tune their gear by ears, in fact, their gear, namely, the legendary Quad amplifiers were designed and implemented without relying on "listening". If those Marantz sound masters tell the engineers/designers to tune for the sound they like, then how come on the test bench, they still ended up with flat frequency response 20-20,000 Hz and distortions below the threshold of audibility such as 0.001% THD.

Even if the engineers/designers/implementers did manage to tune to please their sound masters at Marantz, and yet the changed parameters (electronically, examples: resister, capacitance values, bias, or whatever things they adjusted, would somehow not changed the final measured distortions, noises, cross talks, damping factors, slew rates etc. etc., that is whatever measured, yet would in fact change the sound quality, then the logically question would remain, that is, how do they know the potential customers, at least the vast majority would prefer what the sound master prefer?

Note: glad to say preemptively that I am not digressing as I am still on topic, citing Marantz as example, on the perceived SQ topic.

 

[dlaloum]

Technology and engineering (at least with Sound United) have probably significantly improved over the past 6-7 years.

I have yet to be convinced on the hardware side.... but on the software side, yes - I think there have been improvements in Atmos (and probably other competing software as well) - which means we can look forward to seeing this improvement spread quickly up and down the line to all AVR's/AVP's.

Tracking it will be difficult though, as we get very little information about software versions embedded within firmware!

 

[AcuDefTechGuy]

I have yet to be convinced on the hardware side....

You don’t believe engineering can improve the layout, parts quality, less heat, etc., that could improve both performance and reliability?

@PENG you’re actually an electrical engineer. What do you think?

 

[TLS Guy]

I have yet to be convinced on the hardware side.... but on the software side, yes - I think there have been improvements in Atmos (and probably other competing software as well) - which means we can look forward to seeing this improvement spread quickly up and down the line to all AVR's/AVP's.

Tracking it will be difficult though, as we get very little information about software versions embedded within firmware!

Good components and a good neat layout are vital. There is too much gear around now thrown together and built with components of suspect provenance.

The reason there is so much Quad gear, a lot of it very old, fetching high prices on the used market, is because people know not only of its reliability, but excellent sound quality and that it is still able to be serviced, with a significant number offering service for vintage Quad gear.

The reason is that Peter Walker was very careful about his component sources, good design and very neat layout. That gives servicing a big leg up. If it is thrown together like so much gear, there are multiple problems.

1). Components can be crowded, not ventilate properly and operate above their design temperature.

2). In providing service there is excess expense on dismantling and reassembly.

3). It makes it next to impossible to insert test probes where you need to.

4). This can make serving next to impossible and make for more problems due to damage and stress on disassembly and reassembly.

I don't know if you remember that in the early days of electronic engine control car breakdowns were common due to failed electronics. This also caused hazards with injuries. So laws were passed to make it mandatory for all electronic components be autograde. That was very effective at increasing the reliability of cars and trucks.

Planes have to have military grade electronic components.

So auto grade components cost significantly more than domestic grade. And military grade a lot more.

I would bet a lot of higher end audio products do use auto grade components.

Then we get into black market and counterfeit components especially in Asia.

The way my AV 10 is constructed I think any service tech would have a good chance at repairing it without a huge hassle. For a star the thing is much bigger than any receiver even though it contains no amps.