[calimark]

I stared a similar thread over on REW home forum, but I figured this place is a bit more lively and colorful .



1. Art USB PreAmp

2. EMC8000 Calibrated (cross spectrum).

3. 20ft mic cable

4. Older onkyo receiver ( will change later).

5. Klipsch SUB-12 ( Ported down firing sub)

6. Klipsch KLF-30 mains

7. Klipsch KLF C7 Center

8. Klipsch RS52-II Side Surrounds

9. Klipsch in wall rear surrounds


Room is 28L*19W*10.5H- Zero treatments at this time.


I've read a ton of threads about REW usage but could use a bit of confirmation that I am applying what I have read appropriately.


SUB is not smoothed. All others were smoothed to 1/6 as that is what I read on REW home forum as the preferred method. I've seen 1/12 posted as well.


So first off- my sound card cal and then SUB response.

For the sub I tried to get a SUB only response by placing the mic in the port hole- see attached. Is this good or is there a better way(without going overboard). I just wanted to see a baseline. I used the 0deg MIC cal for the measure.


Then I did multiple measures by moving the sub around the room, and I ended up with the current response shown. It has the least amount of peaks and valleys as compared to other spots. I do plan on using 2 subs eventually once I get the hang of things.


So how do my inital measures look as far as procedure?





RT front speaker as measured at main listening position along with ETC.





BAD ETC EXAMPLE!




Above is a bad example of ETC. The problem was not REW impulse calculation preference issue as implied.
http://www.avsforum.com/t/1415741/speaker-measurement-room-measurement-treatment#post_22135342

Quote:
Nyal Mellor, Acoustic Frontiers LLC & Jeff Hedback, HdAcoustics

ETC of the L & R speakers should:

• Be visually identical (with only minor deviations) from 0‐40ms

• Be down to 10dB

by 40ms to prevent breakdown of the precedence effect

• Clearly show a decrease in the amplitude of energy over 040ms.

The decay pattern may or may not be continuous.

Quote:
Nyal Mellor, Acoustic Frontiers LLC & Jeff Hedback, HdAcoustics

A popular approach is simply to analyze the level of reflections on an ETC and compare these to the

direct sound, setting a target for the reflections to be 10dB or more less than the direct sound. This

analysis is not sufficient since ETCs are spectrally blind (i.e. they contain no information as to the

spectral content of the reflected sound) and the auditory system is very discerning in its requirements

for spectral balance between the direct and reflected sounds in a room.

 

[calimark]

Overall on the ETCs I see some spikes on all the graphs which measure out to ~26' extra travel, so I assume this is all coming from the back wall.

If I am understanding this correctly, it is uncommon to treat the entire back wall (Absorption)?

 

[localhost127]

for generating ETCs,

you need to be utilizing the hardware loopback connection and such that the signal leaving the source (speaker) is T=0 , NOT the direct signal arrival at the receiver (mic) set to T=0.


there is also a checkbox to 'utilize hardware loopback' in REW Options/Analysis and also uncheck 'set direct signal to T=0' (or equivilant --- don't have REW in front of me at the moment).

 

[localhost127]

http://imgur.com/j2ChN.png%5B/IMG%5D

you can't just subtract the direct flight path from the indirect (to determine indirect total) without utilizing hardware loopback as it's not linear..

 

[calimark]



Quote:
Originally Posted by localhost127  /t/1415741/speaker-measurement-room-measurement-treatment#post_22134930


for generating ETCs,

you need to be utilizing the hardware loopback connection and such that the signal leaving the source (speaker) is T=0 , NOT the direct signal arrival at the receiver (mic) set to T=0. the geometric reflection path flight distances

there is also a checkbox to 'utilize hardware loopback' in REW Options/Analysis and also uncheck 'set direct signal to T=0' (or equivilant --- don't have REW in front of me at the moment).

And I am. I have read enough of your posts to know thats what you like to see done. You see something in my graphs that says I did not? (cuz i did)


The sub-sample and decimate IR are at their defaults.(checked)

 

[localhost127]

the ETCs you provided show the direct signal set to T=0. that is not what you want - you want T=0 set to when the signal leaves the source, and then the direct signal will be displayed at whatever time it takes for propagation to the receiver (mic) ... eg, straight vector distance from acoustic center of source (speaker) to receiver (mic) in feet * 1.126 = direct signal flight time in milliseconds.


eg, if your receiver is 15ft from your speaker, you should see the direct signal energy on the ETC at approx 17ms NOT 0ms.


once you have done this and thus have the total flight path of a particular signal (sparse, indirect specular reflection spike of energy as measured on the ETC) - then you can work backwards to identify the particular boundary or source of the indirect reflection. for the early-early (1-3ms) energy, look for sources of edge diffraction nearby the speaker (or the speaker cabinet itself?)

 

[calimark]



Quote:
Originally Posted by localhost127  /t/1415741/speaker-measurement-room-measurement-treatment#post_22135206


the ETCs you provided show the direct signal set to T=0. that is not what you want - you want T=0 set to when the signal leaves the source, and then the direct signal will be displayed at whatever time it takes for propagation to the receiver (mic) ... eg, straight vector distance from acoustic center of source (speaker) to receiver (mic) in feet * 1.126 = direct signal flight time in milliseconds.

eg, if your receiver is 15ft from your speaker, you should see the direct signal energy on the ETC at approx 17ms NOT 0ms.

once you have done this and thus have the total flight path of a particular signal (sparse, indirect specular reflection spike of energy as measured on the ETC) - then you can work backwards to identify the particular boundary or source of the indirect reflection. for the early-early (1-3ms) energy, look for sources of edge diffraction nearby the speaker (or the speaker cabinet itself?)

I dont know what to do here- I assure you they are unchecked. Is there some other button somewhere that would cause this?

I'll go back and do a measure right now for sanity.


Also, the loopback is used on the 2nd channel right? No other magic necessary?

 

[localhost127]

 http://www.avsforum.com/t/1415741/speaker-measurement-room-measurement-treatment#post_22135051


"use loopback as timing reference"

 

[calimark]



Quote:
Originally Posted by localhost127  /t/1415741/speaker-measurement-room-measurement-treatment#post_22135285

http://www.avsforum.com/t/1415741/speaker-measurement-room-measurement-treatment#post_22135051

"use loopback as timing reference"

And thats how it has been for 3 days now.


After tinkering a bit more I have found the problem and its not related at all to that preferences tab.

Problem was two fold:

1. Physical loopback was on the wrong channel.(doh!)

2. Once above was rectified, I saw severe clipping on the loopback channel, even with input gain 0.

The only way I was able to control this is by using windows control to reduce the loopback channel's output to near 0- while maintaining the level within 6db of the output signal.

 

[localhost127]



Quote:
Originally Posted by calimark  /t/1415741/speaker-measurement-room-measurement-treatment#post_22133767


So they recommend being down >=10db by 40ms, which my ETC is, but what about those spikes at ~1,~2.5 &~6.5 ? Should I care?

if you care about maintaining accuracy with respect to intelligibility, localization, and imaging - then yes. you will want to utilize the ETC to identify and attenuate any destructive high-gain early arriving indirect signals (reflections) that arrive within the haas interval of which the ear-brain lacks the resolution to identify them as discrete (separate) signals and thus, fuses them with the direct signal into a single auditory event. richard heyser referred to this as "time smear distortion". attenuating these high-gain early arriving signals gives the ear-brain adequate time to process the direct signal only - an effectively anechoic time-period referred to as the InterSignalDelay (ISD) gap. no high-gain indirect signals (that the ear-brain keys on for localization, imaging, etc) are allowed to impede the listening position within this time period - the direct signal is all that is processed. you naturally have a larger ISD-gap as the room's dimensions are increased (boundaries are further away and thus, indirect boundary reflections take longer in time to impede the listening position). if you have personal tastes for a completely damped room, then this effective anechoic time period (within specular region) would be infinite and no later arriving specular energy would be reintroduced to the listening position. you would utilize the ETC to identify any boundaries incident of specular indirect energies and attenuate (eg, absorb). if you do not want a completely damped room, then you can reintroduce specular energy back to the listening position at a certain time period (corresponding to a specific indirect flight path in distance), effectively terminating (sharply delineating) the ISD-gap - preferably as diffused as possible (NOT as later arriving high-gain sparse reflections) for spaciousness. eg, an exponentially decaying laterally arriving (semi)diffuse-field. unfortunately, in a small acoustical space there lacks a statistically developed reverberant sound-field and instead we deal with focused specular reflections that reflect within the acoustical space like laser beams of sound - and you can see these sparse (focused) spikes of energy within your ETC. so this is why complex geometric or reflection phase grating diffusers (or scattering surfaces) are used to break up these sparse reflections into highly complex/mixed returns. utilizing the ETC here in such a scenario will detail this as well - you can identify a boundary incident of a high-gain sparse specular reflection, place a diffuser (eg, a reflection phase grating diffuser that also offers temporal dispersion) - and see the energy on the ETC be changed from a sparse spike, to many spikes lower in gain AND spread out in time.


the ETC displays how ALL of the specular energy impedes the listening position. from the direct signal, to the sparse, high-gain early reflections, to the later reflections, to the specular room decay until the last of the energy is damped and below the ambient noise floor. you can even utilize it to identify coupling issues - as you may come across a scenario where you see measured energy arriving before the direct signal reaches the receiver (mic) --- which may confuse you at first until you realize transmission speed is much faster via solid than air medium.

 

[calimark]

Now that I believe I have rectified the loopback issue (pls see updated first post).


I have generated a new graph.The first spike is at 23mswhich is ~26ft. Its barely 7 feet physical distance.

AVR does not have any distance compensation applied.





And here is the etc zoomed in. Can you please have a look?


Do the measures look ok now?

If they are, please help with the following:


1. Identify where within the window I should be addressing ( I want to make sure I thoroughly understand this before going any further).

2. Why should the be addressed ( purely based on data shown- again u might repeat your self with what is posted above but please do).

3. What is the defined interval I need to focus on? My reading says 30ms, 40ms and 10db down- which one?


In addition, I should have been a bit clearer in this post:

Quote:
So they recommend being down >=10db by 40ms, which my ETC is, but what about those spikes at ~1,~2.5 &~6.5 ? Should I care?

If, there are no reflections up to 40ms after the initial signal that are less than 10db down, should I even care about them?


Please use this graph in this post for futher explanation.

 

[localhost127]



Quote:
Originally Posted by calimark  /t/1415741/speaker-measurement-room-measurement-treatment#post_22135801


Now that I believe I have rectified the loopback issue (pls see updated first post).

I have generated a new graph.The first spike is at 23mswhich is ~26ft. Its barely 7 feet physical distance.

AVR does not have any distance compensation applied.

do you run audyssey or equiv?

 

[calimark]

Current receiver is old school it doesnt have all that. Basic level cals, SUB XOVER, sub/main multiplex setting, Large/small speaker settings and speaker distance setting. (set to the min 1ft allowed in the measures).

 

[localhost127]

is the test signal routed directly to the channel of interest? seems there is some processing going on somewhere in the chain.


would you mind detailing your USB DUALPRE physical cabling setup?

 

[calimark]



Quote:
Originally Posted by localhost127  /t/1415741/speaker-measurement-room-measurement-treatment#post_22136214


is the test signal routed directly to the channel of interest? seems there is some processing going on somewhere in the chain.

would you mind detailing your USB DUALPRE physical cabling setup?

EMC8000 20' cable into DUAL PRE

LEFT ouput into receiver AV3 input, via 1/4 connector that splits signal into 2.

RIGHT output looped back to RIGHT input.


AVR is set in mono mode, using onlyfront LEFT AVR out to drive speaker(s) one at a time.


I even swapped channels, same result

 

[calimark]

Having rerun these measures even at 1m i am still seeeing this huge dealy.

Per this post http://www.avsforum.com/t/1411590/we-built-it-weve-measured-it-help-us-tweak-it-acoustics-of-the-black-cat/60#post_22072024


Its pointing back to receiver delay. Is there anything else to check before I chalk this up to avr delay?


I will take a measure from my other receiver which is a 2011 Onkyo model. If the timing is closer to actual I guess it will be more evidence to chalk up as avr delay.


Also re t=0, why cant it be used to reference reflections?? It a reflection is 2 ms after t= 0, why cant the distance between t=0 and reflection be calculated using t=0 as the base?

 

[Dennis Erskine]

Place your microphone a known distance from the speaker. Assuming you have an accurate test kit, the first arrival of sound to the microphone will be the direct sound. Take that time. Subtract from that time, the time sound takes to travel the known distance to the microphone. That will be your total processor related delay. If your kit allows you to adjust the gate, you can make that adjustment in your kit. A more simple approach is to take the delta-T between the first arrival and the second, third, etc, arrival. That delta represents the difference in the path length between the first and subsequent arrivals.


Example: If your speaker/microhphone distance is 8' and the second arrival took path of 8'-6", stretch a string 8'6" long to the mic position and the speaker. Any surface that string can touch would represent your reflection point. What if you cannot find such a surface? Look for a port on the rear of the speaker. Consider the number of drivers in the speaker...if there are three of them, you can easily have 3 first arrival times. The face of the speaker is also a reflection point.

 

[dragonfyr]



Quote:
Originally Posted by Dennis Erskine  /t/1415741/speaker-measurement-room-measurement-treatment#post_22149816


A more simple approach is to take the delta-T between the first arrival and the second, third, etc, arrival. That delta represents the difference in the path length between the first and subsequent arrivals.

Example: If your speaker/microhphone distance is 8' and the second arrival took path of 8'-6", stretch a string 8'6" long to the mic position and the speaker. Any surface that string can touch would represent your reflection point. What if you cannot find such a surface? Look for a port on the rear of the speaker. Consider the number of drivers in the speaker...if there are three of them, you can easily have 3 first arrival times. The face of the speaker is also a reflection point.

Note, that the deltas between subsequent arrival times are not linear differences as most interpret them. In other words, to use a simple example, if the difference is 1ms between arrivals, it means that the 'triangulated' path from source, to incident boundary to mic is 1.13 feet longer.


As local has observed, something is wonky in your configuration and you are not employing a 'direct' path through the AVR where the signal is only being amplified. Assuming things are connected and configured correctly ('use loopback as timing reference' is set and the 'set IR to t=0' is NOT engaged), additional latency due to 'something' in the AVR is in play.


Either this needs to be corrected such that the arrival time correlates to the actual distance from source to mic, or the AVR itself needs to be included in the hardware propagation delay compensation loopback, by taking the output of the AVR and looping that back to the input of he other channel in the mic pre.



Also, if you have a non signal aligned source (a good reason we typically utilize aligned signal sources with a unity acoustic origin), the driver offsets can be determined as well with an ETC, including any diffracted virtual sources such as the baffle. But note that a driver offset (delay) of 6inches would correspond to a time offset of ~.44ms. This is typically part of the initial grunge associated with many direct arrivals (which typically includes diffraction sources from speaker mountings which must be physically corrected, as well as driver offset that should ideally be corrected with active crossover delay adjustments) that must be corrected - as this initial grunge is very detrimental.


As should be becoming apparent, the ETC can be used to evaluate multiple 'scales' of behavior, from the gross levels of speaker to mic, to the finer degrees of driver-driver signal alignment, to even finer scales of internal driver reflection issues such as the behavior of sound within the throat of a horn. What we are proposing to do here simply scratches the surface of the capabilities of the ETC response.

 

[calimark]



Quote:
Originally Posted by Dennis Erskine  /t/1415741/speaker-measurement-room-measurement-treatment#post_22149816


Place your microphone a known distance from the speaker. Assuming you have an accurate test kit, the first arrival of sound to the microphone will be the direct sound. Take that time. Subtract from that time, the time sound takes to travel the known distance to the microphone. That will be your total processor related delay. If your kit allows you to adjust the gate, you can make that adjustment in your kit. A more simple approach is to take the delta-T between the first arrival and the second, third, etc, arrival. That delta represents the difference in the path length between the first and subsequent arrivals.

Example: If your speaker/microhphone distance is 8' and the second arrival took path of 8'-6", stretch a string 8'6" long to the mic position and the speaker. Any surface that string can touch would represent your reflection point. What if you cannot find such a surface? Look for a port on the rear of the speaker. Consider the number of drivers in the speaker...if there are three of them, you can easily have 3 first arrival times. The face of the speaker is also a reflection point.

Thank you so much for that clarification. I dont think I really care to know what my delay is coming from as this receiver is going to be replaced, but that being said, I think its a ridiculous delay ( I dont have anything to ref against).


So with your explanation above, what is the problem with using t=0 in REW and just using it as the reference for tracking down subsequent reflections?


I'm inferring that the 'gate' you referenced here is adjusting the graphs for the processor delay [which REW can do by setting t=0, but this will not show the initial time of flight] or you can offset manually to show graphically the time of flight for the initial peak while removing the processing delay.


I'm just trying to understand why its perpetually preached not to use t=0 as the reference.

 

[Dennis Erskine]

You can do that with REW; but, understand processor delay is just what it is. If you know what it is, you can still determine where year early reflection points are simply by taking that delay into account. There's also no point in locating these surfaces to an silly level of precision (your panels will be larger than any non-linear error). That level of accuracy is useful in driver/cross over alignments...also, take into account that ol' angle of incidence = angle of reflection thingy. There are potentially several reflective surfaces which are not affecting listening areas (useful to consider AFTER the low hanging fruit has been picked).


We use digital input signals for calibration, so we'll have processor delay 100% of the time. In your case, the bigger issue is determining why "direct" doesn't appear to be "direct". If you cannot get everything except DA conversion out of the loop, all your calibration measurements are going to be "cooked".


One other thing you ought to do is measure the noise floor in your room. You don't really care about reflections/sound which is below the noise floor ... no sense chasing crud down in the weeds which is masked by the noise floor.