Marantz AV 10 installed: - Early Review and Impressions.

[PaulBe]

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.

 

[TLS Guy]

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.

 

[PENG]

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.

 

[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.

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.

 

[William Lemmerhirt]

I really hate it when daddy and daddy fight.

 

[PaulBe]

Got an e-mail from Marantz. A HEOS update:

HEOS Built-in

The smart way to connect your whole home with the sounds you love

www.marantz.com

 

[Trebdp83]

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.

I don’t recall a recommendation in either of the AV 10 manuals for selection of a filter concerning classical music. The manual has been revised and is a few pages off from a version put out in 2023.

The only mention of classical music in either manual concerns the
Dynamic EQ Reference Level Offset.

Old manual:

Revised Manual

The filter descriptions are sure to set some off. Why the need for filters at all and why would one be selected to enjoy the “Marantz sound” and one used for bench test measurements? What the f#%k are each of them actually doing in there?

Old manual:

Revised Manual:

Anyway, at least the Dolby Atmos presentation has been scientifically tested as dramatically improved. Yikes, my arms are getting tired from stirring this s#%t.

 

[ryanosaur]

Yikes, my arms are getting tired from stirring this s#%t.

Sure.

 

[PaulBe]

I don’t recall a recommendation in either of the AV 10 manuals for selection of a filter concerning classical music.

The filter descriptions are sure to set some off. Why the need for filters at all and why would one be selected to enjoy the “Marantz sound” and one used for bench test measurements? What the f#%k are each of them actually doing in there?

Yikes, my arms are getting tired from stirring this s#%t.

Your post is informative, and you are right that nothing in the manual describes filter 1 as the classical music filter. I prefer filter 2 for classical music... and rock music... and movies. But, perhaps what you are missing is the AV10 is a 'processor'. It's not just a decoder.

Ever since preamps had tone controls, a preamp is a processor. Ever since concert halls and great cathedrals were built, live sound has been 'processed' - example: people find that a certain amount of reverb tail added to vocals and music is pleasurable and enhances the listening experience. With electronic reverb, especially convolution reverb, I can add any kind of reverb I desire and even mimic great concert halls. It's not accurate.

Marantz says that filter 1 is the 'Marantz sound'. The OP of the thread declares that he likes filter 1 better than filter 2. Preference is ok. Since filter 1 is the 'Marantz sound' it is obviously Not more accurate. Marantz declares that filter 2 is better for measurements, so filter 2 IS more accurate.

Processing is production. Decoding is reproduction.

Perhaps sitting back and enjoying a favorite recording, the way you like to hear it, would be more useful than splitting hairs, stirring this s#%t, and tiring out your arm.

 

[ryanosaur]

I wonder what this thread would look like if we were to represent it as a pie chart... how many earnest posts contributing to Doc's thread, how many posts piling on Doc, how many posts arguing about Dacs and filters...



Hugs and kicks to all of you lovely folk!

PS: I still really like the AV20. *blows a kiss to all of you

 

[Trebdp83]

Your post is informative, and you are right that nothing in the manual describes filter 1 as the classical music filter. I prefer filter 2 for classical music... and rock music... and movies. But, perhaps what you are missing is the AV10 is a 'processor'. It's not just a decoder.

Ever since preamps had tone controls, a preamp is a processor. Ever since concert halls and great cathedrals were built, live sound has been 'processed' - example: people find that a certain amount of reverb tail added to vocals and music is pleasurable and enhances the listening experience. With electronic reverb, especially convolution reverb, I can add any kind of reverb I desire and even mimic great concert halls. It's not accurate.

Marantz says that filter 1 is the 'Marantz sound'. The OP of the thread declares that he likes filter 1 better than filter 2. Preference is ok. Since filter 1 is the 'Marantz sound' it is obviously Not more accurate. Marantz declares that filter 2 is better for measurements, so filter 2 IS more accurate.

Processing is production. Decoding is reproduction.

Perhaps sitting back and enjoying a favorite recording, the way you like to hear it, would be more useful than splitting hairs, stirring this s#%t, and tiring out your arm.

No worries. I do not take myself nearly as seriously as some others around this joint do themselves. Yes, the AV 10 is a processor and one of the best and most versatile out there. It is extremely customizable for those wanting different settings for their multichannel movie presentations vs their two channel music presentations. As for stirring the s#%, well, old habits die hard. I'll just switch arms.

 

[PaulBe]

No worries. I do not take myself nearly as seriously as some others around this joint do themselves. Yes, the AV 10 is a processor and one of the best and most versatile out there. It is extremely customizable for those wanting different settings for their multichannel movie presentations vs their two channel music presentations. As for stirring the s#%, well, old habits die hard. I'll just switch arms.

Lefties Rule...

 

[TLS Guy]

I wonder what this thread would look like if we were to represent it as a pie chart... how many earnest posts contributing to Doc's thread, how many posts piling on Doc, how many posts arguing about Dacs and filters...



Hugs and kicks to all of you lovely folk!

PS: I still really like the AV20. *blows a kiss to all of you

Well, I have spent most of the day working on the system. Integrating the system to spread the bass to the bed layer systems proved more difficult to integrate than I thought. I do think it was worth it. The system now seems very even throughout the room. The system is incredibly accurate now. The most pleasing aspect apart from the really high quality sound is the incredible depth of field. It seems you in a far bigger room than you are in terms of length width and height.

As is usually the case I thought speakers were the limiting factor, but with this new AVP and time spent in integration, it seem that was not so, and these speakers are even better than I ever thought or imagined.

The issue that has always set this system apart is the bass fidelity. There is no way any sealed or ported system can create such true to life realism in the last two octaves. There absolutely should be more commercial choices than there are, currently it is next to none. Unfortunately very few have ever heard a good TL based system.

 

[Trebdp83]

What exactly was done to improve the integration of the system to spread the bass to the bed layer? With every mystery tweak your speakers somehow continue to perform better and better without any explanation as to the settings used to accomplish the result.

I’m not doubting the improved presentation but the lack of information concerning the tweaks involved to achieve your miraculous results does not inspire confidence in your testimonial.

Does the speaker configuration still include all but height speakers set to Large? Any change to frequencies copied to the subwoofers when using LFE+Main setting with Large speakers? LFE Distribution being used? What is the level set for it if so?

Has the speaker configuration changed to all Small speakers? If so, what is the crossover setting of each channel?

With two connections to the subwoofer pre outs, is the subwoofer setting at the default “1 spkr” setting or changed to the 2 spkrs” setting? Is the “Directional” feature being used in the subwoofer settings?

Have you decided to use room correction? Have you made any tweaks to the Graphic EQ without Audyssey? Inquiring minds want to know. I want to know!

 

[PENG]

I don’t recall a recommendation in either of the AV 10 manuals for selection of a filter concerning classical music. The manual has been revised and is a few pages off from a version put out in 2023.

The only mention of classical music in either manual concerns the
Dynamic EQ Reference Level Offset.

Old manual:
View attachment 73830

Revised Manual
View attachment 73831

The filter descriptions are sure to set some off. Why the need for filters at all and why would one be selected to enjoy the “Marantz sound” and one used for bench test measurements? What the f#%k are each of them actually doing in there?

Old manual:
View attachment 73834

Revised Manual:
View attachment 73835

Anyway, at least the Dolby Atmos presentation has been scientifically tested as dramatically improved. Yikes, my arms are getting tired from stirring this s#%t.

You are right, but it doesn't matter what the manual or any user says, as Dr. Toole put it (in his book, article, and interviews too iirc), if the person knows which speaker he's listened to..., I don't care what the person think..." not quoting word for word but pretty close. I tend to agree with him. If he said that for loudspeakers, imagine what he would say about such AB comparison listening to DACs, or even preamps, power amps lol..

So, we are in the subjective territory, all bets are off. We already know Gene prefers filter 2, TLS Guy filter 1 and PaulBe filter 2, ryanosaur did not say but I get he prefers both.

Back to DSP sound such as Atmos, DTS:X upmixing etc., in those cases, I do think even sighted level match AB comparison listening would reveal the differences in overall sound effects/characteristics.

 

[TLS Guy]

What exactly was done to improve the integration of the system to spread the bass to the bed layer? With every mystery tweak your speakers somehow continue to perform better and better without any explanation as to the settings used to accomplish the result.

I’m not doubting the improved presentation but the lack of information concerning the tweaks involved to achieve your miraculous results does not inspire confidence in your testimonial.

Does the speaker configuration still include all but height speakers set to Large? Any change to frequencies copied to the subwoofers when using LFE+Main setting with Large speakers? LFE Distribution being used? What is the level set for it if so?

Has the speaker configuration changed to all Small speakers? If so, what is the crossover setting of each channel?

With two connections to the subwoofer pre outs, is the subwoofer setting at the default “1 spkr” setting or changed to the 2 spkrs” setting? Is the “Directional” feature being used in the subwoofer settings?

Have you decided to use room correction? Have you made any tweaks to the Graphic EQ without Audyssey? Inquiring minds want to know. I want to know!

Just careful attention to detail. I do not use room correction. I did activate the spread of LFE to all bed layer channels as outlined by Gene. All the bed layer speakers are set to large. The surrounds have four Dynaudio drivers in each, with 2.5" voice coils and take a lot of power. They are sealed and won't decouple from the box. F3 is 52 Hz, 12db. per octave roll off. The back surrounds have two KEF B 139s in each speaker these speakers are active bi-amp. Bass units are two KEF B139s in each speaker. These are phenomenal bass drivers are now back in production again. They are TL loaded, F3 is 27 Hz, 12db. per octave roll off, so plenty of bass output down below 20 Hz. The center uses two SEAS coaxial drivers, for active BSC in a TL. They handle a lot of power. F3 is 47 Hz roll off 12 db per octave.

The mains are dual TLs with lines tuned half an octave apart. F3 is around 20 Hz, this is a Tri-amped line using SEAS Excel drivers and is an integrated full range speaker fully able to handle the sub outputs. Total power to each main speaker is 550 watts. Center is 300 watts, side surrounds 250 watts each. Back surrounds 400 watts each. The four ceiling speakers are crossed at 120Hz. Power to each is 100 watts.

So, yesterday I carefully calibrated everything after making the changes. I delayed doing this so as to keep the system basically the same as it was previously.

The biggest change in the sound, is the depth and width of the sound stage. Choral music in old acoustic spaces is stunningly realistic with the Dolby upmixer. I listed to the monks of Buckfast Abbey last evening and the choir stalls seemed way beyond the end of the room. Their wonderful Ruffatti pipe organ was superbly reproduced. This was from a BBC stream on iPlayer. The BBC just do a superb job of recording this type of material. They use minimalist mic techniques at a distance. They have been doing these broadcasts weekly since 1923, and so they know who to record in these spaces better than anyone else on the planet.

So, for whatever reason this AV10 produces a much more realistic and natural sound field than the previous AVPs.

 

[highfigh]

I’m having trouble reconciling “perhaps we blame speakers for more then we should”, and “Speakers remain a big problem”. I’m not trying to test you.

I do realize that sound reproduction is not sound recreation. It’s an analogy and always will be an analogy. Accuracy with a speaker is a flawed concept. Thankfully, our ears were created to tolerate all kinds of natural and artificial distortions and still hear intelligible sound. Perhaps the closest concept of accuracy in a speaker is a Quad ESL-63.

I have heard mediocre department store speakers sound good with great electronics. I have never heard good speakers sound good with mediocre electronics.

Different kinds of speakers interact in different ways with any reasonably normal home listening space. Polar response vs frequency is a big factor. The list of things that are deemed important changes from time to time. It’s hard to be released from the ‘circle of confusion’.

Perhaps we do blame speakers far more than we should, and listening spaces remain a big problem. Everyone should try listening to their setup (if possible) in an outdoor environment. It’s a real ear-brain opener.

Like you, I build my own speakers. The 7 floor channels are all the same drivers, same crossovers, and similar cabinets. I have a lot of opinions, but, the most important thing to me is ‘match the channels’ – this is hard to almost impossible to do completely when including the height channels. Next is ‘integrate into the listening space’ – using a real room’s flaws to help you if possible. Reasonable room correction is much easier if you follow these two things. My goal is to make the room disappear when and where I can, and make the room part of the ‘instrument’ when and where I can’t make the room disappear.

WRT "Accuracy with a speaker is a flawed concept", if the input is compared with the output in a controlled environment, and it is if they use an anechoic chamber, it's possible for a speaker to be considered 'accurate', but not as accurate as a circuit like an amplifier since it's a reactive load, most amplifiers aren't able to handle some loads and most rooms affect the sound.

 

[Trebdp83]

Just careful attention to detail. I do not use room correction. I did activate the spread of LFE to all bed layer channels as outlined by Gene. All the bed layer speakers are set to large. The surrounds have four Dynaudio drivers in each, with 2.5" voice coils and take a lot of power. They are sealed and won't decouple from the box. F3 is 52 Hz, 12db. per octave roll off. The back surrounds have two KEF B 139s in each speaker these speakers are active bi-amp. Bass units are two KEF B139s in each speaker. These are phenomenal bass drivers are now back in production again. They are TL loaded, F3 is 27 Hz, 12db. per octave roll off, so plenty of bass output down below 20 Hz. The center uses two SEAS coaxial drivers, for active BSC in a TL. They handle a lot of power. F3 is 47 Hz roll off 12 db per octave.

The mains are dual TLs with lines tuned half an octave apart. F3 is around 20 Hz, this is a Tri-amped line using SEAS Excel drivers and is an integrated full range speaker fully able to handle the sub outputs. Total power to each main speaker is 550 watts. Center is 300 watts, side surrounds 250 watts each. Back surrounds 400 watts each. The four ceiling speakers are crossed at 120Hz. Power to each is 100 watts.

So, yesterday I carefully calibrated everything after making the changes. I delayed doing this so as to keep the system basically the same as it was previously.

The biggest change in the sound, is the depth and width of the sound stage. Choral music in old acoustic spaces is stunningly realistic with the Dolby upmixer. I listed to the monks of Buckfast Abbey last evening and the choir stalls seemed way beyond the end of the room. Their wonderful Ruffatti pipe organ was superbly reproduced. This was from a BBC stream on iPlayer. The BBC just do a superb job of recording this type of material. They use minimalist mic techniques at a distance. They have been doing these broadcasts weekly since 1923, and so they know who to record in these spaces better than anyone else on the planet.

So, for whatever reason this AV10 produces a much more realistic and natural sound field than the previous AVPs.

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.

 

[TLS Guy]

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.

I don't use the 2 channel setting very often. I normally use the Dolby Upmixer. The reason is that for classical music you really don't want all the ambient field coming from the front speakers. That is not natural at all. With multi miked pop music I can see why the up mixer may not be optimal. From the videos I watch I see that classical engineers are getting away from using multiple mics. I note that Mathew Pons who does the DSO streams is really getting away from spot mics, and relying on his classic Decca Tree. It sounds spectacular, but Orchestra Hall Detroit has a wonderful acoustic and has recorded well since the earliest days of stereo, with the recordings made under Paul Paray still sounding excellent.

So, in my view the upmixer has been a real boon to the classical music lover, if the recording is done well. It is especially good for large ensembles and especially opera and choral music. For cathedral sounds it is an absolute must as far as I am concerned. One caveat though is, I suspect that all the speakers have to sound very similar in overall sound balance. I would suspect it would not be good, if the different speakers had markedly different sonic signatures.

At the moment I am at the -20db setting which is the default. Now I have to figure out how to adjust it, which does not seem to be intuitive. I don't have many Dts-X discs, essentially only a few BDs, but if I play those I play them native.

 

[Trebdp83]

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