"Official" Audyssey thread (FAQ in post #1) - Page 54

Thanks, I'll give that a try. I'm a little skeptical that the issue would be that the receiver would have incorrect wiring (Denon 4308ci), but the battery test would identify if its the speaker.

I used Audyssey for the first time yesterday on my new Integra DTC-9.8. I used a 7 point system and everything went very well. Audyssey picked up all the speakers (7.1) and distances were all very resonable. I tried playing a BD disk and a HD-DVD disk and both disks had DD+ as well as D TrueHD. My BD disk would only play stereo and My HDDVD disk would not play any sound at all. The picture dislpay was fine in both cases. I went through every menu I could find but no luck. I finally had to resort to factory defaults and everything went back to normal but I lost all my Audyssey settings. Anyone have clue why this happened?

Tony

Chris - I have a second set of surround speakers - identical to my LCR spkrs - located per ITU that I manually swap banana plugs with the rear surrounds for listening to DVD-A. Do I need to use Audyssey Pro to measure and store a second curve for them?

Tony, that doesn't sound like an Audyssey issue as Audyssey affects only speaker configuration and equalization. Before restoring factory defaults, had you poked around the 9.8 to look at other settings?

I would guess that any change in speaker position and/or type will require rerunning Audyssey as that will affect distance/level/EQ. No?

Right-o! Do I need to use Audyssey Pro to store settings? The thought of re-running it every time I switched is . . unappealing.

pepar,

I went through all the menus on the Integera and the players several times. I spent at least 2-3 hours doing so. I could not put my finger on it. I'm going to try again and make certain that I do not do something that caused my situation. Thanks for the response.

Tony

Connected via HDMI?

AFAIK, you cannot store and restore the data. Unappealing and unfortunate.

Ah, the Audyssey answer to pepar's dilemma....get a second Audyssey-capable unit, and swap a whole bunch of interconnects and speaker terminations anytime he wants to listen to DVD-A.

A quick update on people who have room correction for their sub. I'm using the Integra 9.8 with the Audyssey Pro kit, the sub is a Klipsch RT-12D.

Originally I had used the RT's room correction first and the bass was great. After adding 2 throw rugs I redid the calibration without using the subs room correction. Bass was uneven and didn't sound right.

I compared the graphs and noticed the one where I used the subs room correction extended way deeper and was basically flat to below 20 hz up to 120. Without room correction the Audyssey graph was only flat from about 40 hz to 80 hz.

I believe this is due to the room correction on the sub correcting 2 large peaks, flattening response before Audyssey got a hold of the signal. I did this multiple times to verify.

If I remember reading correctly, there are two pre-set curves available on the Pro, right? "Standard" or "flat"?

Nope. 3 but none are flat (unless you do other manipulations on the pre/pro).

Is there a way to store . . profiles . on the Pro software for e-a-s-y and q-u-i-c-k uploading to the pre/pro or AVR?

No. That is unfortunate since the Pro software stores the results for uploading to Audyssey but, afaik, there is no routine for re-installing it. Seems to me it would make sense to provide for this as it would make it easy for a dealer to re-install the Audyssey EQ in the event that a customer accidentally reset his AVR or had some other accident.

I'm a bit confused. The RT's room correction, with no Audyssey, gave you a good flat response. Then you say that Audyssey, with no RT room correction, gave you a limited flat response. Then in your last sentence you seem to say that you had the RT's correction and Audyssey on together?

Yes, and if it could be stored and re-uploaded, it could obviously be saved. And if it could be saved, then multiple "its" should be able to be generated and saved for uploading.

Yes. (sigh)

Hi Michael,

Our recommendation is to always try and cover an approximately oval shape that is centered at the main listening seat and extends out to about the width of the front L and R speakers. It's hard to tell you exactly without knowing your room arrangement, but I would think that following this general rule will give you good results. Frankly, the exact positioning of the mic is not so critical that it has to be measured with a ruler. The first position is the most critical because that is where the delays and levels are calculated from. But after that, just sample the listening area to the left, right, front, and perhaps rear of that first measurement. Try to avoid going too wide outside the main speakers and also avoid going too close to the back wall.

Filika,

Chris

Chris,

"Our recommendation is to always try and cover an approximately oval shape that is centered at the main listening seat and extends out to about the width of the front L and R speakers."

This is different than what I had understood you to recommend for two rows of seats, where the centroid of the measurement positions would be between the two rows.

So how about these 8 positions:

L spk--------------C spk--------------R spk

<----------------- 11 ft ----------------->









---------------X------------X


-----------X---------1------------X


-----------X---------X------------X

I have a similar experience with my Velodyne SMS-1. After consulting the matter with Chris (Audyssey), I determined that the proper way to use the SMS-1 sub equalizer with Audyssey is as follows:

1. Optimize sub/speaker placement before any calibration/equalization by eye-balling the RTA curve and choosing the position with the flattest response.

2. Use SMS-1 to correct the peaks.

3. Run Audyssey only after doing #1 and #2.

The result is excellent and I'm very grateful to Chris for his advice.

What you DON'T want to do is:

1. Don't use the sub equalizer after you use Audyssey because you will throw off the Audyssey calculations in the time domain.

2. Don't use the sub equalizer to boost frequencies because Audyssey might likewise boost those and you may get over-boosting and distortion.

Yep..24p mostof the time.

I have an Onkyo Pro 885. When I first ran Audyssey, I ran it for the minimum 3 locations only, Primary, and Left and Right. Now, I have read in the last few pages that it is best to run multiple measurements in an "oval" around the central primary listening position. I haven't tried that yet, and before I do, I have several questions:

With a standard parametric equalizer, one can EQ the primary listening position to be as "flat" as possible. This will optimize one listening position, but it may actually make other LP's worse. OR... one can take a series of measurements at multiple locations and average them, and EQ to improve most locations at least somewhat. However, one then gives up the "best" possible EQ for the primary listening position.

How is Audyssey different from this? If the system measures at multiple positions, does it also attempt to correct at multiple positions? If so, doesn't this compromise the correction at the primary LP, at least somewhat? After all, Audyssey can only output one global correction curve, and if it's optimized for an area, it doesn't seem like it could be as optimized as for a single point.

If Audyssey is correcting for an "area", how does it know how big the area is, since you don't tell it the relative positions of the mic? Does it "triangulate" the positions of the mic based on arrival times from different speakers, much like a navigation system triangulates a vehicle's location relative to multiple satellites?

I also have another, more general question: I am having a difficult time conceptualizing equalization in the time domain. (Is "equalization" even the proper term for what is occurring, or is it some other type of manipulation?) The only "time domain" method I can think of to reduce mechanical ringing in a room is to truncate (or shorten the length of) the note that is causing the ringing. Is this what Audyssey is doing? IOW, if a certain frequency has a long decay time in a room, due to a resonance or ringing at that frequency, does Audyssey reduce the initial level of the frequency, (EQ), and then shorten the length of time it plays for, (truncating in the time domain)?

I don't need trade secrets here, nor do I want to know all the math involved, (I'm certain I wouldn't understand it anyway). I'm just looking for a basic description of what Audyssey does to manipulate the "time domain". This is a new and foreign concept for me, and I would like to understand it a little better.

Thanks.

Craig

First, the answers to all your questions are in this thread as well as on the Audyssey website, so I will answer them very briefly.

The problem is that the first approach is there is no way to make a "flat" response at one listening spot since movements as small as the distance between one's ears will result in quite different responses, EQ-ed or not. The problem with the second lies in the averaging rather than the intelligent weighting and analysis to determine what are the common or prevalent problems requiring correction.


Because the major modal problems of a room are determined by the room's dimensions. If one corrects these common problems, the effect is salutory for all positions. Audyssey groups the measurements by their common parameters and weights them by, as they call it, "fuzzy logic" to determine the selected filters.


Nope. Distances and levels are calculated only from the first mic position.


Sorta. All filters have amplitude and time parameters. The trick is matching those to the amplitude and time problems of the room.


From here, you might look to the Audyssey website. There are others that deal with the issues in a more general manner but those tend to be rather mathematical.

Thanks, Kal. I have checked the Audyssey website and didn't see the answers. I'll re-read it more closely. I have also read the last 5 or 6 pages of this thread, (although admittedly not the entire 54 pages), and did not see the answers.

Well, yes, but the problems change with different listening positions. For example, if a room has a resonance at 60 Hz, at one position it will result in a peak at 60 Hz and in another position, it will result in a trough. If you reduce the energy introduced at 60 Hz, you reduce the peak, but you don't improve the trough, (I suppose it would be "less canceled", but any cancellation will still result in a loss.) Even so, this is all frequency domain equalization and I still can't conceptualize "time domain correction".

I can understand amplitude filters, but how do you filter time? Yeah, I just re-read the website about "pattern recognition and fuzzy logic". Now I can conceptualize why I can't conceptualize time domain correction. I'm going to bed now...

Craig

The depth of the trough and the height of the peak are due to the same energy and a reduction in that energy will improve both. This, however, does not come with a pure amplitude reduction (even though this is impossible, anyway) but with a reduction from the initiation through the decay time of the signal. Remember, it takes time for the interaction, that creates the peaks/nulls, to occur.


Basic filter theory but you do not "filter time" but, rather, filter over time. Think about implementing the filter with physical inductors and capacitors. Both lead to phase shifts which, in essence, are also shifts in time. In other words, the effects are not instantaneous in the time it takes for the amplitude change to start and stop. These are time-domain effects.


Good night.

Would I do acoustical measurements - FR and waterfall - in my room as I would have before with the signal generator running to a pre/pro input?

Filters can be described in the frequency domain by their amplitude and phase response, or in the time domain by their impulse response. The room is a filter which has certain characteristics because it is an acoustical space: the phase response is related to the amplitude response. MultEQ uses that relationship to construct a filter that compensates as much as possible for the room effect, and so produces a EQ'd room response that is closer to the ideal impulse response, i.e. ringing is much reduced (time domain correction) and frequency response is flatter (frequency domain correction).

Hi Chris,

First of all, thank you very much for your reply! The problem with my setup is that, due to the way the room is set up, at this point in time the L and R speakers are positioned on either side of the TV. In other words they have a width of about 80cm.

Here is a crude diagram:
The 'X's are the couch, '1' is the main listening position. L, C, R, and S are the position of the left, center, right, and surround speakers. The L, C, and R speakers are in front of a corner of the room.


S
-------------L
---------------C
-----------------R
X
X
X
X
1XXXXXXX--------S



Obvously this arrangement in far from ideal but at the moment its the only option . So, given this scheme, I would be very grateful for your suggestions with regard to possible micing positions for proper Audyssey MultEQ XT calibration.

Thanks kindly in advance,
Filika,
Michael

I have seen a bit of confusion around this subject, and I think I am confused myself.
Therefore I am trying to resume here the info that can be gathered in the forum in order to have them checked buy the gurus and be, after beeing edited with the correction indicated by the gurus, a reference for everyone.

1. Audyssey setup verifies at which frequence the speakers roll off -3 dB in SPL and informs the AVR consequently. In some AVRs these xover are shown, in others are not.

2. The AVR sets a xover for the frequence found in point 1 for each speaker.

3. If the frequencies found by Audyssey are above a certain value set by the AVR's producer, the speakers are indicated as Small or a xover value is shown, else the speakers are indicated as Large or Full Band and the xover value found in point 1 can be shown or not like said in point 1.

4. If the FL and FR are indicated or set Large by the user, the AVR allows the double bass function while the xover value set in point 2 is disabled, if they are indicated Small or a xover is set by the user, double bass is disabled.
If the FL and FR are found Small or a xover value is set by the user, even if other speakers are found Large or so set by the user, the double bass function remains disabled.

5. if the Double bass function is enabled, content below xover set in point 2 will be sent to the subwoofer and also continue to be sent to the speaker if the LF and LR are set large or found large.
The AVR sends to the sub the frequency, below the xover set in point 2, directed to FL and FR if a xover value is set by the user or found small.

6. Frequencies, below the xover set in point 2, directed to C SrL SrR SrBL and SrBR are sent to the sub only if the said speakers are not found, or set Large by the user. If a xover value is set by the user for said speakers all frequencies below that value are sent to the sub.
7. LFE is directed to the subwoofer only, unless the subwoofer is not there or set to not be used. In this case the LFE are sent to the FL and FR which are considered large by default. The LPF of LFE can still be adjusted.

8. In all case when a speaker is set large automatically, it may simply be able to not start roll off -3dB at frequency around 80 Hz, so the bass may not be reproduced correctly. It is always better, unless having speakers very good in handling bass, to set a xover frequency so allowing the bas to be sent to the sub.

Please let me have your comments, so I can edit the mistakes I am sure I made.

ciao