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"Official" Audyssey thread (FAQ in post #1) - Page 1990

post #59671 of 62237
Quote:
Originally Posted by Patrick Murphy View Post

Keith,

Do you get the Super Bowl broadcast over there and (to keep it Audyssey related) will you be listening in DSX?

 

We used to but not any more AFAIK. If we did, I listen with Dolby PLIIz for my height channels :)

post #59672 of 62237
Quote:
Originally Posted by batpig View Post

Keith hates DSX so I already know the answer to the second question tongue.gif

 

LOL!  'Not enamoured' is probably more accurate. But remember I don't have wides or even rear surrounds unfortunately, so my remarks are always only concerned with DSX as it relates to Height channels. (I think I always make that clear - hope so anyway).

post #59673 of 62237
Quote:
Originally Posted by habe View Post

I'm really torn between a Denon 2313CI and a Pioneer SC-1222-K. Both units have a 9 band manual EQ.

I'm trying to understand what Audyssey XT does to a sub below the crossover point..... Does it EQ a sub similar to using an multi-band EQ and if so, how many points is it EQ'ing in the sub range?

habe

 

 

No - it works nothing like multi-band EQ. Check this out:
 
post #59674 of 62237
Quote:
Originally Posted by RocShemp View Post

Quote:
Originally Posted by kbarnes701 View Post

Forgive me if this is an incorrect assumption on my part, but I am not sure that you fully understand how DV works. Have a look at this FAQ answer and see if this answers your questions:

g)1.   What is Dynamic Volume?
Quote:
Originally Posted by SoundofMind View Post

I would hope not. That's the point of turning DVol off first, setting MV and turning DVol back on. It will keep dialog at that level and all else will be compressed according to the DVol intensity setting; Med compression=evening, Strong compression=midnight.

IOW, the soft parts (dialog) are kept audible and the loud (music swells, explosions) are kept from getting too loud. Is that the effect you desire?

Over at the Ask Audyssey site, Chris Kyriakakis acknowledged that DV boost dialogue:
Quote:
The amount of boost depends on many variables including the master volume setting, the content in the center channel, and the content in the other channels. It is not a simple fixed boost for every volume setting.

So that tip in the FAQ seems incorrect. You can't just pick a volume level where dialogue sounds good with DV off and then simply turn DV on. The dialogue will be boosted (by an amount that varies from soundtrack to soundtrack) and will therefor be louder than where you found it comfortable with DV off.

 

No, the FAQ is correct. The bit I have bolded above has been edited so it now reflects what the FAQ says, and that info is correct. Prior to your edit you had it the other way around, which may explain why it isn't doing what you hoped it would do. DV does not attempt to match SPLs from one soundtrack to another - it isn't trying to do that. ~All that DV does is compress the dynamic range in the content - ie it makes the loud sounds less loud and the soft sounds less soft, centred around a level you choose with the MV. Usually people want to always hear dialogue, hence that tip. People are less concerned, late at night, with if an explosion sounds 'realistic' and more concerned with whether they can hear what the characters are saying.

post #59675 of 62237
Quote:
Originally Posted by batpig View Post

Quote:
Originally Posted by jlpowell84 View Post

Quote:
He's repeated his reasons about 100 times so all it would take is 2 minutes of your time an the "advanced search" feature to find one of his many statements on the matter

It's just funner for the sake of conversation to ask. I do some online gaming and we get new people often. We help them along even though it's a repetitive process so they can learn. cool.gif



So, the bottom line is that it creates a more in-your-face, front heavy presentation which Keith finds distasteful.

PLIIz on the other hand extracts ambiance (uncorrelated sounds) from the surround channels, and routes them to the height channels. The effect is more subtle and "ambient" and doesn't call quite as much attention to itself, and it also doesn't attempt to tinker with the front/rear balance of the content.

 

:)  Yes. Also DSX reduces the levels of the front left and right channels too, as well as that of the surrounds. This makes for a tremendously front-centric presentation, which is the precise opposite of what I have worked so hard for in my room - which is a surround 'bubble' that immerses me in the soundtrack of the movie (I only ever refer to movies - I ought to add that to my sig - I have no idea what happens when you use DSX for music). 

 

It seems odd to me that Audyssey refer to the way our hearing works and how sounds from behind are less readily perceived by us as SPLs drop, and they invent Dynamic EQ to help with that - then they go and develop a tech that actually sticks everything up front and they even jimmy the surround channels to help achieve it!

 

That will be my 101st repetition I guess - LOL :)

post #59676 of 62237
Quote:
Originally Posted by djbluemax1 View Post

...To me, it sounded like turning Dvol on at -30db boosted the dialogue volume to about -15 to -17db levels. Turning the MV down to -50db from -30db sounded like the dialogue only turned down to -20 to -25db levels, but explosions were compressed so they don't get loud...Max
Max, I'm thinking it would be interesting if you had an SPL meter handy for your experiments.
post #59677 of 62237
Quote:
Originally Posted by janos666 View Post

I read the FAQ but I still have some questions regarding MultEQ XT on Denon AVR-2808 (EU model with firmware 19.04.2010 30.25).
My current speaker configuration is a bit skinny: 2x Chorus 726 fronts and 2x Dali Zensor 1 surrounds. There is no sub (but will be), center (I don't even want that) or surround backs (I absolutely don't want those).
The room is small and has relatively poor acoustic characteristics. I play to more the equipment to a bigger and caustically treated room in a few months, but until then...


1: Does Audyssey try to draw a 2D map of the speakers and apply any corrections based on the 2D speaker positions?

I see that it measures the time delays (and calculates 1D metric distances from that data to show me something I can understand more easily than milliseconds).
But does it calculates the angles and does it apply some kind of corrections based on this information?

I mean... If I place my surround speakers "anywhere behind my back", will the DSP know where they are and mix the 2D sound according to that information?

I want to know this, because I get better frequency-SPL spectrum in the main listening position when I use a single measurement point.
And may be that's only a coincidence or placebo, but the spatial placement of the sound feels better when I use more points.
This is a small room, so the sound (especially the low range) varies a lot based on the listening position. But the room being small means I use it alone, so I have one listening position I care about.
But I care about both frequency linearity and correct 2D surround sound positioning.

So, my dilemma is:
-> If the algorithm wants to calculate a 2D map (could be presented as angles on the OSD), then it absolutely needs multiple points across both horizontal axises, and thus I have to use multiple points.
-> If it doesn't care about the angles, then A: I can go with a single measuring point and enjoy the more linear frequency response when I am alone in the room + B: I need to be more careful with the speaker placement!

I didn't read a word about this, so I could assume there is no 2D mapping.
However, this would be a serious limitation from an otherwise very complex and hyped calibration system, so I must assume it does map the speaker positions in 2D and applies corrections for that.
What's the truth then...?


2: I don't have surround back or multiple front speakers, so I have various options to use the bi-amp capability of my front speakers
(The AVR has separated Front A and B outputs which can operate in A+B mode and the surround back outputs are the official choices from bi-amplification).

If the algorithms are smart enough, this can theoretically double the filter resolution of the front speakers because I assign two fully separate channels for the fronts which can be separately filtered by the DSP.
Does bi-amp really yields separately filtered low and high ranges for the front speakers? Or are the control points fixed around the frequency axis?

1. No. If you think about it, the microphone cannot see where the speakers are, and it cannot, using a single omnidirectional capture, identify direction. You'd need multiple spaced mics to triangulate the locations. So no. AUdyssey's position is that more information allows their system to work better.l Many use a very tight pattern around the main listening position because (a) being three-dimensional beings their ears occupy two different locations in space and (b) they move their heads sometimes.

2. The A and B outputs are different terminals for the front left and front right amplifiers. Except for being able to switch them, it's exactly like plugging two speakers into the left terminal and two apeakers into the right terminal.

There is lots of discussion about biamping on the boards and you could spend all day reading people's varying opinions if you wish. The bottom line for me is:

1 Even if you doubled power, you are talking about a 3 dB increase at max-before-clipping output, which most people would call one notch louder.

2 You will not double power, because the tweeter never gets more than 20 or 25% of the total power. The amp going to the tweeter will add no more than 25% to total available power, about one dB, typically "just noticeable."

3 unless your amps are clipping you don't need more power. Alghouth I really wanted to believe otherwise, when my system is at 100 dB, using maybe 5 watts, those five watts sound no different if they come from a 10 watt amp or a 200 watt amp. Extra unused power simply keeps distortion levels lower than they would otherwise be at any given output (at least theoretically, all other things being equal). But if the tiny distortion of the amp is inaudible, making it even more inaudible pretty much be definition yields no audible benefits.

4 If you happended to have very inefficient speakers, or speakers that are very hard to drive because their impedance dips low in some range, a different amplifier (with a lower output impedance) might stay flatter in driving them, but that's not a problem that biamping with the receiver's internal amps could fix (and it appears to me that the deviations involved are a quarter of a decibel).

5 If you just want to biamp, go ahead. It won't hurt anyghint, but use the "real" (fake/passive) biamping option not the B terminals. You won't hurt anything. I had two different sets of speakers biamped in the past with no untoward effects . . .
post #59678 of 62237
Quote:
Originally Posted by SoundofMind View Post

Max, I'm thinking it would be interesting if you had an SPL meter handy for your experiments.

I can't speak for djbluemax1 but in my experience it seems to be around a 5 dB boost (though that varies depending on where I set the MV). That said, I've had the odd experience like say Fright Night (remake) and The Expendables 2 where everything seems to be boosted (even on Light).

I've not used an SPL meter to confirm my findings though I did use an SPL meter running test tones on the Disney Wow disc. I managed 85 dB with my MV around -1 (I've always been told such a minor discrepancy is negligible after having calibrated the system through Audyssey versus tones on a disc). With the same tones, I achieved a result of 75 dB with my MV at around -15 or -16. Finally, my SPL meter registered 65 dB when I had my MV around -30 to -32.

That's actually why I got it in my head that -30 should have been my go to volume with DV on (funny enough, -30 was my go to "safety" volume before I started playing with DV since some channels can be loud and I didn't want any unpleasant mornings). And it did work for the most part (even with really loud discs like Serenity and Battleship). But then I'd get anomalies like Fright Night (remake) and The Expendables 2 that weren't that loud to begin with but did sound loud with DV set to on. I never did think to try the SPL meter on the actual films since I simply assumed I was doing something wrong. Hence why I chose to post my questions here.
post #59679 of 62237
Quote:
Originally Posted by janos666 View Post

2: I don't have surround back or multiple front speakers, so I have various options to use the bi-amp capability of my front speakers
(The AVR has separated Front A and B outputs which can operate in A+B mode and the surround back outputs are the official choices from bi-amplification).

If the algorithms are smart enough, this can theoretically double the filter resolution of the front speakers because I assign two fully separate channels for the fronts which can be separately filtered by the DSP.
Does bi-amp really yields separately filtered low and high ranges for the front speakers? Or are the control points fixed around the frequency axis?

"Passive" biamping is a waste of an amp and some wire. Unless you physically disconnect the (passive) crossovers in your speakers and use "active" (powered) electronic crossovers such that the two amps truly drive each speaker separately, it's a total waste of time.

 

Here's a useful article on it:

 

Passive Biamping - AKA 'Fool's Biamping'

post #59680 of 62237
Quote:
Originally Posted by kbarnes701 View Post

"Passive" biamping is a waste of an amp and some wire. Unless you physically disconnect the (passive) crossovers in your speakers and use "active" (powered) electronic crossovers such that the two amps truly drive each speaker separately, it's a total waste of time.

Here's a useful article on it:

Passive Biamping - AKA 'Fool's Biamping'

Thanks for that link. I was, until very recently, one of those who wasted his time with passive bi-amping. I must admit I was also experiencing a placebo effect with it as well. That was confirmed when I finally put back the shorting bars on my RF-63's (and left only one cable per speaker) and didn't notice any difference from when I had them bi-amped.

Now I'm using the other two amp channels to power side surrounds using Neo:X. smile.gif
post #59681 of 62237
Quote:
Originally Posted by JHAz View Post

1. No. If you think about it, the microphone cannot see where the speakers are, and it cannot, using a single omnidirectional capture, identify direction. You'd need multiple spaced mics to triangulate the locations. So no

It is actually very easy if you have >=3 measurement points and you can make the assumption that the microphone is always inside the poligon you get when you draw a line between the neighbouring speakers and the mic was always in the same horizontal plane.

I guess these are safe assumptions because Audyssey asks for fixed mic height and >3 points anyway. (Déjà vu?smile.gif) And it's a very bad idea to place the mic behind a speaker or too close a wall anyway. So, you will probably stay inside the poligon.

You can easily figure it out manually on a paper. I tried to draw you a possible solution:


If the Audyssey guys couldn't figure this out then I don't want to hear about them again. LOL.
I guess they were simply lazy to take this into account if they didn't. But I think they should.


Yes, I figured it out that Speaker A and B use the same DAC and amplifier outputs and they are only separated by an A/B/both switch which is only there for conveniece for some users.
Not that I could understand those who use the same multi-channel AVR with a different set of front speekers for different kind of inputs though.


Quote:
Originally Posted by kbarnes701 View Post

"Passive" biamping is a waste of an amp and some wire.

I thought so. I know it's redundant in general.
I just wondered if it can help when you use MultEQ because then you use two DAC output pins and every DAC output pins can have their own unique filters and every pins are limited to the same filter resolution but the sample points aren't fixed.
But I guess they didn't think about this and the AVR uses the same digital channel for both outputs in passive bi-amp modes.
Edited by janos666 - 2/5/13 at 7:17pm
post #59682 of 62237
Quote:
Originally Posted by janos666 View Post

It is actually very easy if you have >=3 measurement points and you can make the assumption that the microphone is always inside the poligon you get when you draw a line between the neighbouring speakers and the mic was always in the same horizontal plane.

I guess these are safe assumptions because Audyssey asks for fixed mic height and >3 points anyway. (Déjà vu?smile.gif) And it's a very bad idea to place the mic behind a speaker or too close a wall anyway. So, you will probably stay inside the poligon.

You can easily figure it out manually on a paper. I tried to draw you a possible solution:


If the Audyssey guys couldn't figure this out then I don't want to hear about them again. LOL.
I guess they were simply lazy to take this into account if they didn't. But I think they should.


Yes, I figured it out that Speaker A and B use the same DAC and amplifier outputs and they are only separated by an A/B/both switch which is only there for conveniece for some users.
Not that I could understand those who use the same multi-channel AVR with a different set of front speekers for different kind of inputs though.

I thought so. I know it's redundant in general.
I just wondered if it can help when you use MultEQ because then you use two DAC output pins and every DAC output pins can have their own unique filters and every pins are limited to the same filter resolution but the sample points aren't fixed.

But I guess they didn't think about this and the AVR uses the same digital channel for both outputs in passive bi-amp modes.

Lots of assumptions that we can't control in your multi-mic multi test locating concept and, based on the evidence here, are not aplicable to actual human beings. You haven't read this thread much if you think it's safe to assume folks won't place a mic close to a wall or outside the"polygon." People report doing so at least weekly and get pointed to the FAQ, etc. THat's just folks who are motivated enough to come here and ask questions. If one thought that was important they could use a fixed three mic array like Trinnov, so that you aren't leaving poeple who are trying to work around their furniture and may or may not have an appropriate stand for the mic to get everything right.

and of course Audyssey doesn't have access to elements in the mc mix to change them somehow. Of course it's unlikely that you could with significant accuracy get locations correct if somebody improperly placed their surrounds behind them instead of at the sides (why not assume they'll RTFM and put the surrounds at the sides where the surrounds belong if we can assume they'll place mic testing points appropriately?) The concept of using software to map a 3 D soundfield and recreate that soundfield using different numbers of differently located speakers is on its way, looks like, and will get to home playback within a reasonable amount of time. But that starts with a different way to mix the soundtrack in the first place and theoretically could yield far greater precision in location both on the mixing stage and at home. Which is to say it's a cool concept but not what Audyssey is intended to do.

As far as using a receiver to actively biamp speakers, it's an interesting concept that I suspect would never be implemented because of those dang human beings. How many folks would hook the low passed amp output to the tweeter element and blow their tweeters? How happy would speaker manufacturers be if the reciver manufactuers were advising people to take the speakers apart, remove the crossovers and use whatever they come up with in the receiver? While it would be simple enough, theoretically, to implement differing high and low pass slopes to mimic what was in the original crossover (assuming you can figure out what those are), but how would the home user know if his Thiels need a notch filter to control an oil can resonance that the shallow 6 dB per octave crossovers make the tweeter subject to? And how will they know what frequency and how deep hte notch should be? It would be a ton of fun to fiddle with, and I would have loved to have the ability especially with my Maggies (now long gone) but it'll never happen IMO.
post #59683 of 62237
Quote:
Originally Posted by JHAz View Post

You haven't read this thread much if you think it's safe to assume folks won't place a mic close to a wall or outside the"polygon." People report doing so at least weekly and get pointed to the FAQ, etc.

I guess those people who do such things will probably end up with improper calibration results anyway. This would be only an addition to their problems. And the angles could be manually editable, just like distances, so people could see if there is a problem.
Quote:
Originally Posted by JHAz View Post

and of course Audyssey doesn't have access to elements in the mc mix to change them somehow.

Same answer: this should work the same way as distances: passed to the manufacturer's software as measured parameters and let that figure out what to do with them.
Quote:
Originally Posted by JHAz View Post

(why not assume they'll RTFM and put the surrounds at the sides where the surrounds belong if we can assume they'll place mic testing points appropriately?)

Room limitations. That's what Audyssey is all about: "Usual living rooms are not studios and doesn't furnitured like studios but let's try to make make something about that.", isn't it?
But I thought about smaller things. Like somebody significantly (but not brutally) exceeded the recommended angles.
Or, if nothing else: warn people about their incorrect speaker and/or mic placement.
Quote:
Originally Posted by JHAz View Post

The concept of using software to map a 3 D soundfield and recreate that soundfield using different numbers of differently located speakers is on its way, looks like, and will get to home playback within a reasonable amount of time. But that starts with a different way to mix the soundtrack in the first place and theoretically could yield far greater precision in location both on the mixing stage and at home.

Sounds interesting. And very-very expensive if you need a lot of speakers, more DACs and fresh softwares with probably more processing power demand.
Quote:
Originally Posted by JHAz View Post

As far as using a receiver to actively biamp speakers

That's one thought. But I didn't go that far. I just assumed, that both channels would have a different MultEQ filter set. Two full-range inputs but with different MultEQ filters.
And since I need to externally connect the two + and two - inputs on my fronts, I thought they are wired separately inside. To be honest, I didn't try to remove those and hear what happens. biggrin.gif
Edited by janos666 - 2/5/13 at 9:27pm
post #59684 of 62237
Quick question! is recommended if one can remove the back of the seat at the main listening position to get correct distance setting? or will this cause improper calibrations? my Marantz 8801 w/ Audyssey is very accurate for calculating distances, it seems that placing the mic at 12" from the seat does indeed cause in actuate measurements and kinda defeats the point if Audyssey is measuring 7-8" away from my actual position. anyone try this?
post #59685 of 62237
Audyssey doesn't measure distances, it never did, but derives those parameters indirectly from delay measurements. Thus your mileage may vary depending upon the circumstances (arrival of first wave front, dominance and delay of reflections, internal processing etc.).
post #59686 of 62237
Quote:
Originally Posted by RocShemp View Post

Quote:
Originally Posted by kbarnes701 View Post

"Passive" biamping is a waste of an amp and some wire. Unless you physically disconnect the (passive) crossovers in your speakers and use "active" (powered) electronic crossovers such that the two amps truly drive each speaker separately, it's a total waste of time.

Here's a useful article on it:

Passive Biamping - AKA 'Fool's Biamping'

Thanks for that link. I was, until very recently, one of those who wasted his time with passive bi-amping. I must admit I was also experiencing a placebo effect with it as well. That was confirmed when I finally put back the shorting bars on my RF-63's (and left only one cable per speaker) and didn't notice any difference from when I had them bi-amped.

Now I'm using the other two amp channels to power side surrounds using Neo:X. smile.gif

 

Now that is a sensible use of the amps and wire!

post #59687 of 62237
Quote:
Originally Posted by JHAz View Post

As far as using a receiver to actively biamp speakers, it's an interesting concept that I suspect would never be implemented because of those dang human beings. How many folks would hook the low passed amp output to the tweeter element and blow their tweeters? How happy would speaker manufacturers be if the reciver manufactuers were advising people to take the speakers apart, remove the crossovers and use whatever they come up with in the receiver? While it would be simple enough, theoretically, to implement differing high and low pass slopes to mimic what was in the original crossover (assuming you can figure out what those are), but how would the home user know if his Thiels need a notch filter to control an oil can resonance that the shallow 6 dB per octave crossovers make the tweeter subject to? And how will they know what frequency and how deep hte notch should be? It would be a ton of fun to fiddle with, and I would have loved to have the ability especially with my Maggies (now long gone) but it'll never happen IMO.

 

Great points there. I am leaning towards active loudspeakers these days. All the problems you mention melt away when the speaker designer knows the intimate performance details of the amps used, and when the amps are purpose-matched to the drivers they are working with. If I was starting from scratch today, but with what I know now, that is the route I would go. Forget AVRs, forget external amps - hook my prepro to the speakers and be done with it.

post #59688 of 62237
Quote:
Originally Posted by audiofan1 View Post

Quick question! is recommended if one can remove the back of the seat at the main listening position to get correct distance setting? or will this cause improper calibrations? my Marantz 8801 w/ Audyssey is very accurate for calculating distances, it seems that placing the mic at 12" from the seat does indeed cause in actuate measurements and kinda defeats the point if Audyssey is measuring 7-8" away from my actual position. anyone try this?

 

It's OK to remove the back of the seat for measuring if it is also removed for listening. As that is unlikely the answer is 'no'. If the back of the seat is getting between the mic and the surround or rear surround speakers, raise the mic a little until it clears the seat back and has line of sight to the speakers. There will never be a problem with Audyssey measuring the distance to any speaker - even speakers the mic can't 'see' - because Audyssey doesn't measure the distance. It measures the delay between the time of the input signal and the time the output is received by the mic, and then converts this into distance using the speed of sound in air for the conversion. Personally I have never understood why they don't just leave it in msecs instead of converting it to feet and inches, as this seems to cause a lot of confusion. What is important is that the sound arrives from each speaker at the MLP at the appropriate time - and that is what Audyssey sets out to do when you calibrate. The result is a cohesive soundstage with better imaging.

post #59689 of 62237
Quote:
Originally Posted by gurkey View Post

Audyssey doesn't measure distances, it never did, but derives those parameters indirectly from delay measurements. Thus your mileage may vary depending upon the circumstances (arrival of first wave front, dominance and delay of reflections, internal processing etc.).

I know but I don't see how that changes anything about the 2D speaker mapping .. well, I proposed ... it seems?
These 2D positions should be calculated from delay measurements even if you have a survey station in your room.
The OSD would show you angles for diagnostic reasons the same way it shows the distances now. But they should work the same way as distances: not real, directly measured distances -> not real angles, but 2D positions calculated from signal delays. Distances and angel would only be calculated for human readability.
post #59690 of 62237
Quote:
Originally Posted by kbarnes701 View Post

It's OK to remove the back of the seat for measuring if it is also removed for listening. As that is unlikely the answer is 'no'. If the back of the seat is getting between the mic and the surround or rear surround speakers, raise the mic a little until it clears the seat back and has line of sight to the speakers. There will never be a problem with Audyssey measuring the distance to any speaker - even speakers the mic can't 'see' - because Audyssey doesn't measure the distance. It measures the delay between the time of the input signal and the time the output is received by the mic, and then converts this into distance using the speed of sound in air for the conversion. Personally I have never understood why they don't just leave it in msecs instead of converting it to feet and inches, as this seems to cause a lot of confusion. What is important is that the sound arrives from each speaker at the MLP at the appropriate time - and that is what Audyssey sets out to do when you calibrate. The result is a cohesive soundstage with better imaging.

I completely understand what it does, perhaps I better get use to using the term "delay" I gave it a shot and the kick Audyssey gave upper mids and highs is gone:D and and overall a lot smoother presentation I'll see what it sounds like over a couple of days to see if I still like it.
post #59691 of 62237
The only thing Audyssey or the mike actually "sees" / hears is the resulting complex sound field at the current measuring position compared to the original signal generated.
No angles, no nothing, just delays, frequency and phase differences. Each and every speaker is measured on its own, no triangulation or anything.
Thus wall, ceiling, floor and surface reflections from all surrounding objects are been detected indirectly and compared by algorithms to the direct first wave front/sound, if any.
Even small disturbances, like resonances from furniture, sound traveling (faster) through solid objects, vibrations from the microphone stand, reflections from nearby surfaces will influence and cause alterations of the results. The same holds true for sound processing through all those onboard DSP chips etc., which will induce additional delay times into the propagated sound and which will influence this outcome one way or the other. Thus the calculated "distances" from those measurements are just theoretical and not real.
The whole thing is just been used to visualize in common terms for the average user, what the measuring system has taken into account somewhat somehow.
Edited by gurkey - 2/6/13 at 3:08am
post #59692 of 62237
sometime todayish I'm going to see if you can really triangulate actual distance and angle using unknown mic positions and unknown speaker positions. In my head I see each indiviual speaker's delay (at each mic position if the software chose to look at it) as yielding a series of concentric circles: the speaker would be known, for sure, only to lie somewhere on the circle that represents the calculated distance. I have to assess (to the extent my dyslexia allows me to get there) the extent to which results form differeing unknown locations of mic and the same unknown locations of speakers actually constrain the point(s) along the circle that a speaker must occupy.
post #59693 of 62237
I have a Marantz NR 1403 with Audyssey MultEQ, Dynamic EQ, Dynamic Volume. The receiver only puts out 50 watts per channel and my front floor standing polk tsi 400's require 20-200w. I would like the add a two channel amp to power just these two speakers in my 5.1 system. Will adding an amplifier like Emotiva UPA-200 2-Channel Amplifier mess up my Audyssey calibration?

Thanks,

Dan
post #59694 of 62237
Quote:
Originally Posted by kbarnes701 View Post

Great points there. I am leaning towards active loudspeakers these days. All the problems you mention melt away when the speaker designer knows the intimate performance details of the amps used, and when the amps are purpose-matched to the drivers they are working with. If I was starting from scratch today, but with what I know now, that is the route I would go. Forget AVRs, forget external amps - hook my prepro to the speakers and be done with it.

If I had semiunlimited funds, I'd be eyeing Mark Seaton's work in that direction. I love, for example, the way my little QSC K10 makes my synthetic pianos and organs sound both at home and abroad. Although I think the conceit of having a 500 watt amp on the tweeter is more about manufacturing simplicity 9just put in a stereo amp and be done with it) than any real need for that kind of power when the woofer also gets 500 watts. Although my not-too-closely assessed sense is that in the home powered studio monitor world, matched amps for tweeter and woofer have become more or less de rigeur, even though patently unnecessary . . .
post #59695 of 62237
I have a Denon 2808. Anyone know if changeing the main crossover point affects the Auddyssey tune or does it self adjust everything, i.e. if it crossover over the mains at 150hz, and I change it to 120hz, will to cause a lot of issues? Will it readjust the Audyssey and Audyssey flat EQ values? Just trying to figure out what I can and can't adjust without affect the Audyssey tune much.
post #59696 of 62237
Quote:
Originally Posted by kbarnes701 View Post

Now that is a sensible use of the amps and wire!

Sorry, meant, to say "wide speakers" not "side surrounds". I always had side surrounds. I just have no real space to put back surrounds. frown.gif

But movies sounds sweet with the wide speakers up front. Now I just gotta see if I can fit in some height speakers as well. smile.gif
post #59697 of 62237
Quote:
Originally Posted by truwarrior22 View Post

I have a Denon 2808. Anyone know if changeing the main crossover point affects the Auddyssey tune or does it self adjust everything, i.e. if it crossover over the mains at 150hz, and I change it to 120hz, will to cause a lot of issues? Will it readjust the Audyssey and Audyssey flat EQ values? Just trying to figure out what I can and can't adjust without affect the Audyssey tune much.

The "basic" speaker settings like size, distance, crossover, etc. are totally independent of the EQ filter.

If you haven't yet, start reading the Audyssey 101 laboriously and lovingly compiled by our own Keith Barnes, starting here: [URL=http://www.avsforum.com/t/795421/official-audyssey-thread-faq-in-post-1/51750#user_c4]c)4. Is it OK to change the Crossovers from Audyssey's recommendation?[/URL]
post #59698 of 62237
Quote:
Originally Posted by kbarnes701 View Post

Great points there. I am leaning towards active loudspeakers these days. All the problems you mention melt away when the speaker designer knows the intimate performance details of the amps used, and when the amps are purpose-matched to the drivers they are working with. If I was starting from scratch today, but with what I know now, that is the route I would go. Forget AVRs, forget external amps - hook my prepro to the speakers and be done with it.

I know you are just getting antsy, staring around the room looking for SOMETHING to upgrade tongue.gif Must be hard to scratch the itch when you have already taken your system to such high levels.

That said, I have heard the M&K 2510P's (the active versions of your S150's) in a setup I helped configure for a friend, powered by a 4311CI. They sounded pretty freakin' sweet wink.gif It ended up topping out as a 9ch DSX setup, with the guy's older S-100B's as wide speakers, and in-wall M&K surrounds along with dual M&K subs. Pretty rockin' setup that could hit reference levels without breaking a sweat.

But not sure honestly how much of a difference you'd notice from that swap, and whether it would be worth the $$$.
post #59699 of 62237
Quote:
Originally Posted by Whiteout007 View Post

I have a Marantz NR 1403 with Audyssey MultEQ, Dynamic EQ, Dynamic Volume. The receiver only puts out 50 watts per channel and my front floor standing polk tsi 400's require 20-200w. I would like the add a two channel amp to power just these two speakers in my 5.1 system. Will adding an amplifier like Emotiva UPA-200 2-Channel Amplifier mess up my Audyssey calibration?

Thanks,

Dan

 

No - an external amp won't 'mess Audyssey up' but it would be prudent to run the calibration level again as the external amp's gain may be different from the AVR amps' gain and this could throw the relative levels of the speakers off.

 

What are you using as a centre channel speaker?  It may be worth considering the XPA-3 to power the R, L & C. The centre channel is the most important in a 5.1 system.  Just a thought.


Edited by kbarnes701 - 2/6/13 at 2:01pm
post #59700 of 62237
Quote:
Originally Posted by JHAz View Post

Quote:
Originally Posted by kbarnes701 View Post

Great points there. I am leaning towards active loudspeakers these days. All the problems you mention melt away when the speaker designer knows the intimate performance details of the amps used, and when the amps are purpose-matched to the drivers they are working with. If I was starting from scratch today, but with what I know now, that is the route I would go. Forget AVRs, forget external amps - hook my prepro to the speakers and be done with it.

If I had semiunlimited funds, I'd be eyeing Mark Seaton's work in that direction. I love, for example, the way my little QSC K10 makes my synthetic pianos and organs sound both at home and abroad. Although I think the conceit of having a 500 watt amp on the tweeter is more about manufacturing simplicity 9just put in a stereo amp and be done with it) than any real need for that kind of power when the woofer also gets 500 watts. Although my not-too-closely assessed sense is that in the home powered studio monitor world, matched amps for tweeter and woofer have become more or less de rigeur, even though patently unnecessary . . .

 

Yup. I would definitely eye Mark's powered speakers. And the powered versions of my MK S150s. I doubt if I will upgrade in this direction (I am happy with the current system) but if I was starting again, or ever moved house and started a new HT from scratch, this is the way I would definitely go.

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