# Impulse response



## chinni123 (Jan 8, 2012)

I understood basics on graph on spl tab as well as waterfall tab. I am trying to understand impulse response. Here are the questions I have.

1. Is there any frequency range that I need to select while measuring for example 20 -200hz or 20-20khz?
2. I read that it is useful to find first reflections. How do I do that? In the second graph, I think I saw at 37msec, there is second peak at 78,
3. How is windowed and ETC are used. 
4. In the help, the pictures have spectrum spread more. But, I see very less spectrum. Not sure what I am missing.
5. What is the blue line. It appears and disappears by clicking on "window"

I am posting two of the graphs here.


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## JohnM (Apr 11, 2006)

To track down reflections you need to make full range measurements, e.g. 0 to 20kHz. If you measure over a narrower range, e.g. 20Hz to 200Hz, the reduced frequency range results in a more spread out impulse response (it lacks the high frequency content that is needed to have fast changes).

Before the frequency response is generated the impulse response is 'Windowed', that means it is multiplied by the response shape shown in the Window trace. The Window is used to make sure only the part of the impulse we are interested in is used to make the frequency response. For example, if the response is mostly noise after some point the end of the window could be set so that the noise part is cut off. Another use would be to exclude a reflection to see how the response looks without the effect of the reflection.

The ETC trace shows the envelope of the impulse response (a bit like showing the amplitude range the response covers). Reflections are easier to see using the ETC. It can also be easier to have the vertical axis set to dB rather than % FS.


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## chinni123 (Jan 8, 2012)

JohnM said:


> To track down reflections you need to make full range measurements, e.g. 0 to 20kHz. If you measure over a narrower range, e.g. 20Hz to 200Hz, the reduced frequency range results in a more spread out impulse response (it lacks the high frequency content that is needed to have fast changes).
> 
> Before the frequency response is generated the impulse response is 'Windowed', that means it is multiplied by the response shape shown in the Window trace. The Window is used to make sure only the part of the impulse we are interested in is used to make the frequency response. For example, if the response is mostly noise after some point the end of the window could be set so that the noise part is cut off. Another use would be to exclude a reflection to see how the response looks without the effect of the reflection.
> 
> The ETC trace shows the envelope of the impulse response (a bit like showing the amplitude range the response covers). Reflections are easier to see using the ETC. It can also be easier to have the vertical axis set to dB rather than % FS.


Thanks for the details. It is good to know that this graph is useful when measurement is done from 0 to 20khz.

I also observed and make sure the following.

1. The check boxes at the bottom says " Jul 17 22:56:40", "windowed", "etc" , "window", "step response". Even when I unselect "jul 17 22:56:40", I am assuming that all are referring to that sample only?
2. All these are overlapping
3. Clicking on *windowed *is causing to fill until 500ms. It seems content is same as "jul 17 22:56;40" until the blue line except little filling is missing.
4. size of window is controlled by "IR window"

I guess it takes time for me to understand. I will read again help. 
As a beginner I want to start understanding ETC first. I see second peak at 35msec. I guess reflection is coming at 35msec. How do I get how much more distance traveled and how to identify location on wall location reflection occurred?


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## chinni123 (Jan 8, 2012)

Since I find second spike happened at 38m sec, the distance traveled = 1.13*38=42.94 feet.* Is it right?*

My room is 21 feet long and 14 to 16 feet wide (two different length for back to middle and middle to front) and 8 feet high. How do I interpretate where is the reflection.

Appreciate any input. Please let me know if my impulse response is not right.

I am adding one more question. I took this measurement with both left speaker and sub and frequency from 20 to 20khz.* is it not right? *I wonder whether this test need to be done with subwoofer disconnected with just one speaker from 0 to 20khz. *is it true?*


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## chinni123 (Jan 8, 2012)

I think time range is not right for my ETC. Could you please provide what is the reasonable range for time and what should be entered (For example micro seconds or milli seconds etc).


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## Barleywater (Dec 11, 2011)

Simplify view of room; imagine as rectangular box with speaker and you located in the room where the microphone would be. As you look around you see images of the speaker from every wall. Looking closely you see images formed with a single mirror, two mirrors, and upwards. Some images just use two walls, or floor and ceiling, others involve three, four, or all six surfaces.

With sound it is much the same, but in the mirror analogy if the mirrors are slightly hazy, or tinted as such, each reflection looses some intensity; and so reflecting sound is scattering, being absorbed, and turned into heat.

Base on: http://www.hometheatershack.com/for...1342640537-test-my-home-theater-sukumar2.mdat using full range measurement I created two screen shots. An ETC picture and a spectrogram picture.

ETC has 0.1ms smoothing. Spectrogram uses 5.0ms window using Blackman-Harris 4 filtering for both sides of window. Good spectrogram views usually require tweaking all the settings a little.

ETC shows the numerous reflections alluded to in my mirror analogy.



The reason the 38ms reflection stands out is that it has lots of coherent high frequency energy in it, as seen in spectrogram at 38ms, 8kHz region(In spectrogram reflections appear as horizontal bands). Most likely it is double reflection; sound from speaker passes microphone (direct sound t=0), bounces off wall behind microphone (first reflection), passes microphone again (one of the stronger early reflections), passes speaker, bounces off wall behind speaker(second reflection, and finally arrives back at microphone 38ms later.

Andrew


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## chinni123 (Jan 8, 2012)

Barleywater said:


> Simplify view of room; imagine as rectangular box with speaker and you located in the room where the microphone would be. As you look around you see images of the speaker from every wall. Looking closely you see images formed with a single mirror, two mirrors, and upwards. Some images just use two walls, or floor and ceiling, others involve three, four, or all six surfaces.
> 
> With sound it is much the same, but in the mirror analogy if the mirrors are slightly hazy, or tinted as such, each reflection looses some intensity; and so reflecting sound is scattering, being absorbed, and turned into heat.
> 
> ...


Thanks a lot for replying with detailed explanation. I never might have imagined the way sound is reflecting. First, I am trying to make my ETC curve to look like yours. Here is my understanding or questions.

1. You are trying to reduce white space on y axis by changing limits to %70 to -%20. It perfectly makes sense to observe rise/fall clearly.
2. I think the time range before was 500msec. I don't remember. I am able to change it to -.005sec to .05 sec implies -5misec to 50milli sec.

*What is suggested time range for ETC?*

3. I used controls to change smoothing to .1. Without changing time from -5 to 50msec, this setting is getting lost after tabing the control. Don't know why?

4. Now graph shows close details. What is first refection here. I don't know first reflection is first spike after t=0 or first highest spike after t=0 

a. I saw at 955 u second (approximately at 1misec) there is -19.1%
b. I saw at 2.86 msecond there is %32.8 (highest spike after t=0

Since the a condition is very close to t=0, do I need to ignore that?

5. If b is right, I need to multiple 2.86*1.13=3.23 feet. So first reflection traveled 18 feet (distance from speaker to listening position)+ 3.23 = 21.23 feet. Is it right?

How do I locate this position?

I am going to understand spectrogram. Appreciate all your help.


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## Barleywater (Dec 11, 2011)

First major peak of impulse response is usually taken as direct sound.

If microphone is 18ft from speaker, and first major reflected sound arrives 2.86ms later, then yes, reflected sound traveled about 3.2fr further than direct sound.

It becomes a matter of geometry as to which path was taken. for instance, if microphone is 3.2ft/2 from wall behind it from speaker, then that is an answer. Likewise if speaker is close to side wall then it could be a reflection from side wall.

Floor and ceiling first reflections are classic first reflections. Example: microphone is placed at about ear height for seated person, and tweeter/midrange drivers are about same height, lets call it 40". Assuming hard ceiling of 9ft, reflection will be strong, and occur at midpoint between equal height microphone and speaker. For 18ft microphone distance that is 9ft. Microphone is 68" from ceiling. Microphone to ceiling reflection point equals speaker to ceiling reflection point. (68"^2+108"^2)^0.5 /12" *2 = 21.27ft. Could be the answer. Corresponding path length of corresponding floor reflection is about 19.2ft, giving roughly 1ms delay between direct sound and floor reflection. 

Path length differences ultimately relate to summation and cancellation frequencies, and determine much of frequency response at microphone location.

Intersections of walls, floor, and ceilings make great reflectors, where all three meet reflect most strongly, at higher frequencies they reflect back in the direction of the source.

Andrew


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## Barleywater (Dec 11, 2011)

I review more impulse responses of your originally posted mdat. There are significant peaks in each about t= -17ms of main peak. They are highly suggestive of microphone input mixer putting part of what the microphone is picking up back into the sweep as it is being played.

Please check your sound device settings. Perform a loopback calibration measurement of your sound card, save and post the mdat. I would like to see it. Symptoms also include exaggerated base with poor decay, and a type of comb filtering. These need to be ruled out.

Thanks,

Andrew


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## chinni123 (Jan 8, 2012)

Barleywater said:


> I review more impulse responses of your originally posted mdat. There are significant peaks in each about t= -17ms of main peak. They are highly suggestive of microphone input mixer putting part of what the microphone is picking up back into the sweep as it is being played.
> 
> Please check your sound device settings. Perform a loopback calibration measurement of your sound card, save and post the mdat. I would like to see it. Symptoms also include exaggerated base with poor decay, and a type of comb filtering. These need to be ruled out.
> 
> ...


Thanks a lot Andrew. I read both your posts and now I understood better. I was told there is no need to calibration for TASCAM 144 II sound card. I never did calibration on it before. I will do calibration. I put monomix to computer. I will do test again and post it here.

I am in the middle of adding absorbers and bass traps. I just added three absorbers for 3 of windows. I need to see how it changes.

Thanks for detailed explanation with examples to map time delay with possible reflection points.


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## chinni123 (Jan 8, 2012)

Barleywater said:


> I review more impulse responses of your originally posted mdat. There are significant peaks in each about t= -17ms of main peak. They are highly suggestive of microphone input mixer putting part of what the microphone is picking up back into the sweep as it is being played.
> 
> Please check your sound device settings. Perform a loopback calibration measurement of your sound card, save and post the mdat. I would like to see it. Symptoms also include exaggerated base with poor decay, and a type of comb filtering. These need to be ruled out.
> 
> ...


I did loopback test quickly. Don't know it is right or not. I connected Left out to Left in using RCA cable and 1/4 inch adapter. I did click calibrate and while doing measurement, I made sure that there is no red bar for head room coming up. I am attaching screen shot and .dat file. Thanks for your help. I wan to make sure my TASCAM settings are right.


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## Barleywater (Dec 11, 2011)

Sound card cal looks great.

Looking forward to your next room measurements.

Andrew


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## chinni123 (Jan 8, 2012)

Barleywater said:


> Sound card cal looks great.
> 
> Looking forward to your next room measurements.
> 
> Andrew


Thanks for the feedback.

I am attaching new test .mdat file. In the home theater room, we replaced removing drapery over 3 large size windows with OC703 2 inch in each of them wrapped with cloth. That is only change so far and here is the picture.









Today, or tomorrow we are going put floor to ceiling bass traps. So, I took test before doing it.

Test scenarios all done with Anthem correction (ARC)
1 .Just subwoofer from 15 to 200hz
2. Just left speaker from 0 to 20hz
3. Both sub and Left speaker 0 to 20hz

I did not start exploring results yet. Appreciate feedback.


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## chinni123 (Jan 8, 2012)

I am trying to bring levels to around 80db while testing. If I increase volume of receiver, I get head room as red bar. I already set maximum at knob for Line in. How is done to get 80 db level without head room in red?


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## JohnM (Apr 11, 2006)

chinni123 said:


> I am trying to bring levels to around 80db while testing.


Why? The level does not change the shape of the response.



> How is done to get 80 db level without head room in red?


Turn down the line in volume, but after doing that you will need to recalibrate REW's SPL meter.


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## chinni123 (Jan 8, 2012)

JohnM said:


> Why? The level does not change the shape of the response.
> 
> Turn down the line in volume, but after doing that you will need to recalibrate REW's SPL meter.


I am getting around 75 as peak. I read that the signal should be high enough to have gap for noise floor.While testing, I remember unless I increase line volume to max, I got complaint that there is not enough input signal.

I am little confused on levels. I am doing following scearios assuming that this is how it need to tested.
1 .Just subwoofer from 15 to 200hz
2. Just left speaker from 0 to 20hz
3. Both sub and Left speaker 0 to 20hz

Do I need to test levels before doing each of them? Is REW SPL meter need to be calibrated each time? Since I am using microphone, do I still need calibration of REW SPL meter? I remember reading to freeze line output volume and receiver volume when it reaches 75db.

I understand basic concept that send enough signal above noise floor as well no clipping at the input of TASCAM. I also not really sure on line out knob, line in knob and receiver volume control and their relation.

Appreciate any help to understand concept what I am exactly doing. I guess final goal is enough head room and good s/n ratio.


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## Barleywater (Dec 11, 2011)

I review latest mdat. The early impulsive spikes at about -17ms to main peak are still present for Just Left Speaker and for Sub and Speaker. For Just Subwoofer, signal appears close to -40ms of main peak.

From pics of window treatment and posts it's obvious that your interest and commitment to tuning for performance are high.

(On safety and well being: Are windows second exit from fire standpoint? If so treatment should be easy to remove, and all household concerns should know how.)

In pursuit of performance: The current and very long thread: *First Measurement!* starting at about post #129 where jtalden comes on board demonstrates very good methodology concerning subwoofer (SW) integration. A good read of this may help this thread from becoming very long.:R

For setting delay distances, which is timing, a timing reference is needed. REW supports this with Preferences, Analysis tab, and checking "Use Loopback as Timing Reference". In Soundcard tab select Timing Reference Input and Timing Reference Output as channels not used for microphone and output to AVR. Connect them as direct loopback, as when doing soundcard calibration. Latency in measurement chain will then be tied relative to reference loop latency (software, converters, buffer sizes). Even very small movements of microphone or speakers, or AVR distance settings are then easy to detect.

A little information theory: In an information system that exhibits linear, time invariant (LTI) behavior, the system's impulse response (IR) is the sum of its component impulse responses. So happens that air, and room reflections are highly linear and time invariant, and likewise so is synchronized measurement system. This means that the sum of the independently measured IR of SW and IR of another speaker equals the IR of the SW and speaker playing together.

Trace arithmetic in REW may be used to verify this, and also becomes powerful modeling tool and aid in setting up system.

Back to mystery peaks before main peak. Loopback timing reference, room dimensions, speaker to microphone distance, and locations of microphone and speakers may be useful in solving this. AVR likely has line outs mirroring speaker outputs for use with external power amplifiers. Direct measurement of these in conjunction with loopback timing reference reveals latency of AVR, and performance of its crossover functionality, LFE behavior, and resolution of its distance setting delays.


Here's my speculative theory though: Since your theater is under construction, the room is empty, and highly reflective. Dependent on microphone and speaker placement it is quite possible that early small peak is the direct sound, and main peak is highly coherent sum of secondary reflections. In two dimensions the extreme case is demonstrated with ellipse with microphone at one foci and speaker located at second foci. For this there is one direct path, and for every other point along the ellipse are reflected paths with exactly the same path length. The extended reflections converge to major axis through foci, and in case of room are multiple reflections from walls behind microphone and behind speaker. And all becomes much ado about nothing when room is furnished, and in your case additionally treated.

Look forward to room info, measurements with timing reference, and furnished room.

Also, by Anthem corrections do you mean LFE and SW to main speaker crossover?

And yes, if sweep is loud enough to be really disturbing, indeed turn it down.

Regards,

Andrew


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## chinni123 (Jan 8, 2012)

Thanks Andrew for quick reply and taking time to provide so much information. I am still reading, and it takes some time to understand. I am doing tests before installing bass traps and absorbers. My friend is helping to make bass traps and absorbers and he may install any time.

I am curious to compare tests before and after installing acoustic treatments. I am hoping that the measurements I did are enough.

I went beyond my budget to buy Paradigm Studio 100,690 , ADO 690 (surrounds), SUB 15 and Anthem receiver. Anthem correct is similar to Audyssey correction but more sophisticated. They provided calibrated microphone and I have to connect to receiver from computer and run their software. Here is the album of pictures for the graphs with Anthem.
https://plus.google.com/u/0/photos/107811778043676490320/albums/5696981087621441025

I will share photos of my home theater. It is complete except there is no acoustic treatments. 

I want to make sure I understand timing reference connection settings.

I am using left channel for testing. So, While measuring, I have to connect analog right out to analog Right in with 1/4 inch one end RCA at other end. So, Right is loop back wire connected all the time while measuring. The analog left in should be connected to microphone and left out to receiver. Following are setting in preferences. Is it right? 

After doing this, do you want me to do measure one more time assuming all the steps are same.


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## chinni123 (Jan 8, 2012)

Andrew,
I just started reading the link you sent (first measurement). I just found timing reference that you have posted. Thanks.


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## chinni123 (Jan 8, 2012)

I just did with loop back on unused Right channels. I am attaching .mdat. I am going to try understanding.


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## chinni123 (Jan 8, 2012)

What is wrong with this ETC that I took from just left speaker out. I see small signal after 40msec.


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## Barleywater (Dec 11, 2011)

Timing reference set up is correct. Sub + Left timing is consistent with Just Left Speaker. But Sub alone timing is not. Two more variables to eliminate: First is the Decimate IR setting in Preferences, Analysis tab. Decimation changes sample rate used internally by REW in handling data, based on bandwidth of measurement sweep. So for sweep to 200Hz, REW is doing sample rate conversion from 44100 samples per second to 2756 samples per second. 44100/16 = 2756.25, so REW isn't doing an integer value sample rate conversion. I have not studied how well REW converter works, and eliminating it as a variable is easier.

Yes more details... It will cause files to be slightly larger without the decimation.

The 2nd variable to eliminate is to use same full range sweep for all measurements, this will ensure timing characteristics across all measurements, simplifying analysis.

And, all that really matters is how system sounds to listener(s). Chances are if system doesn't sound right, or clearly isn't performing relative to other listening experiences, then an underlying measurable difference with remedy exists. Typically the smoother a sound systems response, the greater the agreement is that the system sounds good.

I'll look into Anthem correction.

Andrew


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## chinni123 (Jan 8, 2012)

Barleywater said:


> Timing reference set up is correct. Sub + Left timing is consistent with Just Left Speaker. But Sub alone timing is not. Two more variables to eliminate: First is the Decimate IR setting in Preferences, Analysis tab. Decimation changes sample rate used internally by REW in handling data, based on bandwidth of measurement sweep. So for sweep to 200Hz, REW is doing sample rate conversion from 44100 samples per second to 2756 samples per second. 44100/16 = 2756.25, so REW isn't doing an integer value sample rate conversion. I have not studied how well REW converter works, and eliminating it as a variable is easier.
> 
> Yes more details... It will cause files to be slightly larger without the decimation.
> 
> ...


Thanks Andrew. I am attaching preferenes->Analysys. The decimate checkbox is already checked. Next time I will do full range sweep of 0 to 20khz for subwoofer alone , left speaker alone and sub+ left speaker. I will continue researching to understand. 
Appreciate taking time to analyzing .mdat files.

Edit: sorry. I guess you want without decimation next time. I will do following test.
decimiation unchecked. Loop back connected right to right and following sweep range
1. sub 0 to 20k
2 . left 0 to 20k
3. sub+left 0 to 20k


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## Barleywater (Dec 11, 2011)

I see now post #22. Levels appear maybe a bit low, in Controls set Plot Responses Normalized. It will make Sub measurements bloated in display, but easy to see.

And yes, with timing reference in place, impulse responses of Left Speaker and Sub + Left start at 47.3ms and Sub impulse response starts at 43.6ms. This is additional time it takes signal to travel from soundcard, through AVR, out speakers, through air as sound, to microphone, and back to soundcard. Speed of sound is roughly 1.13ft/ms, so if microphone is 18ft from speaker, flight time is about 16ms, which means it takes signal 31ms to get through AVR. Much of this is probably related to speaker distances chosen by Anthem correction, operator input, or default settings.

Without experimentation it is difficult to rate how well Anthem correction optimizes system in terms of equalization and speaker distance settings. A baseline is formed by switching all equalization out, setting all distances to zero, setting Sub to full range and zero phase (assuming it has this feature), turning AVR crossover off, and measuring full range response of Sub alone, and Left alone or Right alone. Based off of Left or Right speakers natural low frequency response, and natural response of Sub, it is then possible to select crossover frequency, and if AVR is flexible the type of crossover filter to use.

It is easy to see why AVR manufactures seek automated tuning function. The actual step by step process becomes quite involved....

And to post #23: Yes, decimation unchecked.

1. sub 0 to 20k
2 . left 0 to 20k
3. sub+left 0 to 20k

and again with AVR corrections, equalization, crossover off as described above would be great. Might need to post as two posts due to size of mdat files. I imagine Anthem allows storing settings as presets, which is very useful.

Andrew


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## chinni123 (Jan 8, 2012)

Thanks. Andrew. I am still digesting to understand the concepts. I will do more testing as you requested. The impulse graph I complained is beyond the time range. Since I specified -5msec to 50msec, I was not able to see.

I did change time range to -5 5o 150msec and increased level to 100% FS, now I got proper response as attached here. I guess with loop back, I have to change time range.

"Plot response is normalized" was already checked in controls of impulse tab. I see peak at 64.9msec and second peak at 67.9msec. I guess it took 64.9msec for sound to take path for example TASCAM->Receiver->Speaker->air->microphone->sound card


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## chinni123 (Jan 8, 2012)

I took measurements. I am going to post in two replies. I need to update these problems that I did face cause any incorrect measurement.

1. I have to increase/decrease volume from receiver during measurement. For example, just sub vs sub & left connected etc makes REW complain input levels are low or head room is in red after disconncect/connect combinations of speakers/sub.

2. I did REW SPL calibration one time. after that I did not touch either line out or line in. However, I changed volume of receiver couple of times. Not sure if REW SPL will affect measurement.

3. I don't know if the level is 75db. I just try to make sure input levels ok. No red head room.

Some times head room shows red, but it does not pop up "do you want to cancel". Don't know if this kind of measurement is allowed. I changed receiver volume to make sure head room is not red.


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## chinni123 (Jan 8, 2012)

*First measurment with decimation unchecked and 20 to 20k*

I did this measurement with the following changes.

1. Anthem off
2. phase on sub changed from 90 to 0 (sub is located 2 feet from LP close to back and side wall, facing other side wall)
3. Cross over on sub -bypass (Not changed)
4. distance set to 0 for sub, left, right
5. calibration levels set to 0 for sub, left, right
6. left and right set to full

Attaching two parts


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## chinni123 (Jan 8, 2012)

I did measure with decimiation unchecked and 20 to 20k. Some how I lost SW and Left +SW. I only have Left measurement. I don't know "remove all" deleted. I thought it only removes from the screen. I am attaching here.
Basically, this is just before anthem off, distances set to 0, phase of SW 0 etc.


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## Barleywater (Dec 11, 2011)

Having established timing reference, it becomes necessary to remove excess delay/phase, bringing peaks back to near zero, in part because REW sets left and right windowing functions relative to t = 0. Procedure is to look at measurement set and select measurement with peak closest to t =0. Viewing the Impulses tab with Controls panel open click Estimate IR Delay. Note the values. Clicking the Shift IR button will apply calculated shift to current measurement. The rest of the measurements need shifting by the same amount. Switch to each remaining measurement and in the “t=0 (ms):” box enter the previously calculated delay and click “apply” once. This shifts all measurements by the same amount, and REW windowing function will work as intended.

Having applied shifts to the “Anthem off” measurements, and the “Left Anthem on” based on “Right only” for calculating shift, it is seen that Anthem Room Correction is creating 17ms additional propagation delay, and adding artifact. Artifact may simply be some form of channel cross talk, or processing artifact of Anthem’s DSP engine. The artifacts under some types of program material may be audible as a pre echo, but hopefully not. This can be problematic when using digital room correction (DRC) techniques.

Visually, comparison of Anthem correction v off suggests that Anthem correction may be effective for equalizing the system. It’s all about how it sounds….. Based on limited measurements, a dip in response exists in 200Hz region. Pinning down absolute source would require yet more measurements. More important perhaps is apparent channel imbalance of Left and Right speakers, both in general amplitude, and in frequency response, assuming that levels were not changed during measurement set. Impulse responses indicate microphone was several inches closer to right speaker. This could also impact 200Hz response dip to some degree, but is more likely do to asymmetry in placement of Left and Right speakers in distance from front wall, and in distances from left wall and right wall; or if left and right front corners of room have large feature differences such as furniture, proximity to doorways or closets. These impacts are greatest upon front speaker sound stage width, depth, detail, and coherence, with music being most affected. Small tweaks in speaker location can have big effects, as can fairly small movements in head position of listeners.

So, now I have some more questions and suggestions.

How far in front of back wall is microphone? What are Left and Right front speaker locations? More important, what are your listening impressions of system, especially with Anthem Correction?

How many measurements, and at what microphone locations, were done for making Anthem Correction?

I tried to reach anthemav.com website without luck this weekend with no luck. I did follow some reviews, which not surprisingly, mostly thumbs up for ARC.

Following THX Ultra II thread, jtalden raised question of user modification of settings when Audyssey is active being limited, or causing Audyssey to disengage when user modified.

This is legitimate concern with Antherm product as well. Given that heavy part of your outlay in system is for Anthem badge, and ARC feature, it make sense to work with these first to find settings that work well for music and movies in your new system. 

I recommend completing your room treatment program and furnishing of your theater first.

Then without sub and speakers set full range I would work on tweaking speaker location and listening position for music, aiming for good sound stage depth, imaging detail, and stability of image to small head movements sideways.

Next I would do series of Anthem room correction measurements with Anthem suggested number of measurements and microphone locations for music, and storing the preset.

Following this a series of Anthem room correction measurements with Anthem suggested number of measurements and microphone locations for movies, and store the preset.

My expectation is that ARC allows 5+ measurements for calculating settings. For music, listening is focused on single seat or small area, and series of fairly close microphone placements are likely used. For movies, listening is typically for multiple seats, or a larger area, and a series of more widely spaced microphone placements are likely used.

ARC selection of crossover frequencies in your current set up appears to be 60Hz high pass for fronts, 100Hz high pass for surrounds, and 120Hz low pass for sub. These settings strike me as good movie settings for your speaker array. Initial REW measurements with ARC on support this view. Sub range overlap with fronts looks like it yields bass boost suited to movies.

Given ARC off measurements of you fronts shows your 100’s to be fairly capable in your theater room. For music I would keep 60Hz high pass for fronts, and find way to convince ARC to use 60Hz or 80Hz low pass for sub while getting ARC equalization to work.

In all cases I would set up ARC to perform its corrections from as low a frequency as possible to 5kHz, which I am led to understand is the high frequency max for ARC correction.

In testing the ARC music and movies settings I would suggest a fair amount of listening to wide range of music, and of course movies with LFE, action, music and scenes with multiple people talking. For analytical testing of ARC settings on your system using REW, I recommend full range sweeps, 10Hz to 20kHz length 256k. When assessing low frequency behavior, be it peaks, dips, or apparently really flat, I suggest at least three measurements with a change in microphone placement of 18”-30” that includes change along all three axes, before jumping to conclusions.

If unhappy with ARC, or just curious, I would rerun ARC setup using same microphone placements first, to check repeatability, and then different microphone placements (and or number of measurements) to see what happens.

Once well versed by the above process, if you are still curious or unhappy with results, then go for complete manual alignment and equalization with aid of REW.

I hope you get great enjoyment out of your new theater.

Best regards,

Andrew


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## chinni123 (Jan 8, 2012)

Andrew,

Thanks for detailed reply. I am still digesting. I will answer some of the questions now. Both front speakers are around 17 to 18 feet from LP. Not sure if I did not unset distance of Right speaker to 0ft like left one. During measurement, I increased/decreased receiver volume particularly before/after sub is tested alone and with combination. I did got knack on avoiding either headroom is not enough to avoid clipping or input level(I guess sound card) not enough. 

Is there any suggestion to avoid changing receiver volume with out getting complaints from software either headroom or input level?

I did not notice any difference with Anthem on/off. It could be my inability to not find difference. In AVS forum everybody is so happy to use their anthem correction. I need to find clip and what they are seeing difference. I took 5 measurements for ARC.

I started putting 4 bass traps today (4 corners from floor to ceiling) and adding two more on the front/black wall on the floor from left side wall to right side wall. Unfortunately, I have front speakers close to corner walls. With new bass traps I am moving little front but very close to side wall. I have lot of bass traps behind it, can add more as well. Don't know if it is problem. Since I did not know all these things at the time of screen selection (120 inch wide, 142 inch diagnol), there is not enough space.

First I am trying to understand IR and still digesting. I will post reply to confirm my understand on IR in next reply. Thanks again for all your help.


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## Barleywater (Dec 11, 2011)

Can you show me floor plan with seating and speaker layout? For 21 x 15 floor and 18ft distance from front speaker to LP, puts front speakers really close to both side and front wall, and LP really close to back wall. Angles to fronts, and surrounds from LP are far from standards for 5.1 sound mixing, and far from equilateral triangle for stereo. I'm guessing that this is not a projection system?

Andrew


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## chinni123 (Jan 8, 2012)

It is projector system. The room is 21 feet long. However at the screen side it is 13 feet 9 inch and width changes suddenly at around 7 feet from front wall to 15 feet 6 inches. Seating is close to back wall (2 or 3 feet). I am attaching photos of home theater. All my test that I did so far does not include any bass traps. One more is thing speakers were not so close to side wall. They were placed diagnose to current bass traps. Now they are moved close to side wall. My friend thinks that is how tweeter is exactly pointing to center seating.

I like this idea, since moving to as side as possible will reduce distraction while watching on the screen.If needed I can place more absorbers.

Thanks for your help Andrew. You can see slide show or see each photo with caption.
https://plus.google.com/u/0/photos/107811778043676490320/albums/5771003652663651985


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## chinni123 (Jan 8, 2012)

Andrew,
I am understanding post #29. I did follow the instructions you provided which are listed here.
1. Viewing first impulse on left speaker measurement, clicked control->Estimate delay and wrote down the value (47.915) and did click Shift IR
2. Next for all measurements, clicked control-> set value for t=0 off set (msec) =47.915 and click apply. I did all the steps including second left speaker measure that was choose when anthem correction was there. I see following with Anthem, but all others have peak at t=0.

In the following picture, the peak is happening after 17msec extra time with Anthem. This is what you are referring right? You mentioned following.

_Having applied shifts to the “Anthem off” measurements, and the “Left Anthem on” based on “Right only” for calculating shift, it is seen that Anthem Room Correction is creating 17ms additional propagation delay, and adding artifact. Artifact may simply be some form of channel cross talk, or processing artifact of Anthem’s DSP engine. The artifacts under some types of program material may be audible as a pre echo, but hopefully not. This can be problematic when using digital room correction (DRC) techniques._

*What is artifact you are referring to? Is it signal before the peak?* I some times saw some signal before the peak. Not sure what is this? How did you infer microphone is close of serveral inches from right speaker? Also, how do you see comparison of with anthem vs without showing equalizing the system?


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## jtalden (Mar 12, 2009)

I believe the Anthem ARC is applying a convolution EQ that explains the unusual IR.


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## Barleywater (Dec 11, 2011)

jtalden:

All DSP is applied by convolution operation, but yes ARC appears to utilize IR with inversion process in generating EQ, with multiple measurement averaging. Without great care in windowing and envelope smoothing of raw inverse, readily apparent, at least visually, pre causal artifacts occur, some are well masked with continuous time convolution with program material. Others are unmasked by certain types of material.

chinni123: post#33, Yes that is result of process I describe. If you display a left/right pair of as Left alone with Right alone or pair with subs, and zoom in close you will see peeks are separated by about 700u (that's 0.7 milliseconds) which works out to about 9.5 inches. Left peak is trailing, so real center line between speakers was about 4.75 inches left of microphone.

Anthem on v Anthem off: My response is based on observations from looking at info from all the mdats thus far, and lots of experience. Yes, some definitive measurements would clarify if my opinion is justified. Anthemav.com was available tonight, so I've read through MRX manual, and ARC usage. I'll cook up a measurement request after some more thought.

Andrew


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## chinni123 (Jan 8, 2012)

Barleywater said:


> chinni123: post#33, Yes that is result of process I describe. If you display a left/right pair of as Left alone with Right alone or pair with subs, and zoom in close you will see peeks are separated by about 700u (that's 0.7 milliseconds) which works out to about 9.5 inches. Left peak is trailing, so real center line between speakers was about 4.75 inches left of microphone.
> Andrew





Barleywater said:


> Based on limited measurements, a dip in response exists in 200Hz region. Pinning down absolute source would require yet more measurements. More important perhaps is apparent channel imbalance of Left and Right speakers, both in general amplitude, and in frequency response, assuming that levels were not changed during measurement set. Impulse responses indicate microphone was several inches closer to right speaker. This could also impact 200Hz response dip to some degree, but is more likely do to asymmetry in placement of Left and Right speakers in distance from front wall, and in distances from left wall and right wall; or if left and right front corners of room have large feature differences such as furniture, proximity to doorways or closets. These impacts are greatest upon front speaker sound stage width, depth, detail, and coherence, with music being most affected. Small tweaks in speaker location can have big effects, as can fairly small movements in head position of listeners.


Andrew,

I am trying to digest understanding on the above statements for couple of hours. Appreciate pointers to try it myself. Speakers are not perfectly placed to align exactly to microphone. Also, I might have increased receiver volume since what works for speaker may not have worked for subwoofer or combination of them. For example, I got complaint on head room or input level is too low. I need to find way to test them without changing levels next time.

To understand 700 micro second gap between Left and right speaker alone graphs, how do I verify in REW for my education? Since you mentioned time, I assume ETC is the one to compare. Do I need to use overlay?

I generated following graph. I don't know if it is right because I don't see difference of 700 micro seconds. Thanks for your time.
Chinni


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## chinni123 (Jan 8, 2012)

I want to mention that the microphone is in horizontal position facing front wall all the time. For Anthem correction, I was always pointing up. Don't know if there is any problem placing microphone in horizontal position.


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## Barleywater (Dec 11, 2011)

Getting peak performance from great stereo recordings demands much from playback system. Linkwitzlab.com is extensive resource covering this topic. Excellent speakers set up in reasonable listening space, that typically doesn't require extensive treatments, will produce sound image where both the speakers, and the room itself disappear. The basic trick is getting the spatial cues contained in the recording into the mind, and adding minimal distractions such as reflections differing in spectral content compared to direct sound. This is where highly symmetrical room and speaker placement come into play.

Anyway, when looking at overlay of left and right speaker impulse responses after offsetting back to t=0 as you did for post #33, zoom in real close and you see right speaker with peak at t=0 and left with peak at 0.749m, which is 749 microseconds (749u):









Plot results are not normalized, and it is easy to see that peaks are two very different levels. Same conclusion if viewing ETC.

Microphone primarily has different high frequency response when on axis (0 degrees) and pointing straight up (90 degrees). Calibrations for both positions are possible. I don't know about Anthem's microphone. I understand it is USB microphone. When plugged in does it show up in windows as sound device? Can you see it in REW preferences?

Andrew


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## chinni123 (Jan 8, 2012)

Barleywater said:


> Getting peak performance from great stereo recordings demands much from playback system. Linkwitzlab.com is extensive resource covering this topic. Excellent speakers set up in reasonable listening space, that typically doesn't require extensive treatments, will produce sound image where both the speakers, and the room itself disappear. The basic trick is getting the spatial cues contained in the recording into the mind, and adding minimal distractions such as reflections differing in spectral content compared to direct sound. This is where highly symmetrical room and speaker placement come into play.
> 
> Anyway, when looking at overlay of left and right speaker impulse responses after offsetting back to t=0 as you did for post #33, zoom in real close and you see right speaker with peak at t=0 and left with peak at 0.749m, which is 749 microseconds (749u):
> 
> ...


Andrew, I did use 0 degree calibration file for ECM 8000 microphone. I did not use Anthem microphone since calibration is their propriety.

Finally, after spending lot of time, I am able to verify the time delay around 750 micro. I am attaching graph. when I uncheck normalized, the graph is collapsing to tiny. I am having hard time to scroll left and right to go to the graph starting point.

I see legend in the picture you posted to have 3.0%fs for left and .1%fs for right and %fs to end at 4 on upper side. How you determine the values and apply them. Curious to know. zoom in/out and scrolling to search for 0 time is taking for ever. Thanks again for your time.


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## floddern (Dec 29, 2011)

Hi,

Can anybody help me interpret this IR?
I m reading in the manual but cannot say anything about this one, other measurements I do get to understand.

Its a measurement of both speakers and room has treatment


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## Barleywater (Dec 11, 2011)

Hello floddern,

Be more than happy to take a closer look.

Please post to your thread, and post .mdat (saved measurement) of what you've got going.

Regards,

Andrew


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## floddern (Dec 29, 2011)

Absolutely I am sorry to have highjacked here, my impression was this thread was for multile contributions.


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