# IR Windows and Impulse graph



## Primare Knob (Dec 4, 2014)

I am working with the IR window and the Impulse graph, but I am not sure if my impulse graph should look like this. It is looking so messy.

Also, I can't find out what the left and right windows mean and is there perhaps some advise or info available on what these different type of "algorithms" are good for like turkey 0.25?


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## JohnM (Apr 11, 2006)

I've made this a separate thread since it doesn't relate specifically to the beta version.

It is difficult to tell a lot from the graph image since it only shows part of the response, if you attach the actual measurement file (the .mdat file) it will be much easier to check whether it is OK.

Different window shapes offer different trade-offs between resolution and artefacts arising from the truncating effect of the window, for most purposes you are best leaving the window types at REW's default settings. There is plenty of information about window functions in wikipedia, amongst other places.


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## Primare Knob (Dec 4, 2014)

My apologies for posting in the wrong thread.

View attachment 1 Sweep Surround A.mdat


Does the left Windows mean the mic left channel and the right the mic right channel?

Thank you for your help. I don't understand yet what all the controls do, but I am just beginning to learn.


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## JohnM (Apr 11, 2006)

Left and right are relative to the Impulse peak, left is before and right is after.

The measurement looks fine. There is some activity in the impulse response before the peak occurs, though not enough to be a concern. That can sometimes be caused by sample rate conversion in the computer (for example, REW sample rate set to 48k when the soundcard is running at 44.1k or vice versa) and sometimes by FIR equalisers (like Audyssey or Dirac).


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## Primare Knob (Dec 4, 2014)

How useable will my measurement be in building a FIR filter by using these USB audio devices. I am starting to believe that I have bought the wrong mic.


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## natehansen66 (Feb 20, 2011)

What is it you want to accomplish with an fir filter? The first image you posted looks like an impulse taken at the lp


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## Primare Knob (Dec 4, 2014)

I want to create a room acoustic filter for use in JRiver.

The measurement is that of a bookshelf speaker, on axis and at tweeter height, at 80 cm distance.


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## natehansen66 (Feb 20, 2011)

You want to create a room correction filter at that close distance? Is that actually the listening distance? A measurement for room correction is taken at the listening position. A measurement as close as you've taken it is for speaker design/correction with a gated impulse.


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## Primare Knob (Dec 4, 2014)

Do I not need to know my speaker response in order to find out what my room influence is?


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## natehansen66 (Feb 20, 2011)

Well that depends on who you ask. I don't do any rc over about 400hz but if I did I would take into consideration the anechoic response. That said your ir looks too chaotic to be useful for a quasi-anechoic measurement. You need to get the speaker on a stand as far away from all boundaries that you can, and have the mic halfway between the floor and ceiling (assuming a typical 8' ceiling). Then you gate the data before the first reflection.


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## Primare Knob (Dec 4, 2014)

That is what I did. I calculated the first room reflection, which is the floor and started gating my measurements. I cannot do multiple sweeps because this creates time alignment issues. I do have impulse activity before time = 0 which is not due to resampling or conversion, because I checked it.

My question is, if I can get reliable measurements by using USB devices. There seems to be to many factors involved that are beyond any control without a proper driver or software in place.


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## natehansen66 (Feb 20, 2011)

USB mics work fine as far as I've seen, though I've never used one. I use a usb interface with REW all the time with no issues. The IR in the mdat you posted looks fine to me other than the pre-ringing or whatever is going on before the first peak. Like John said, stick with the default window (Rectangular). The other windows basically "round off" the end the the gate in different ways. If you have your gate close to the first reflection then this reflection can be partially included with the other gate types.


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## Primare Knob (Dec 4, 2014)

Do I set the left window to 0, or where the impulse activity start before 0?


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## natehansen66 (Feb 20, 2011)

I'll start the gate 1ms before the first peak. In your IR it looks like there may be a reflection at 2ms? What kind of speaker is this? With a 2-3ms gate the data below 1khz is going to be quite smoothed and rolled off.


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## Primare Knob (Dec 4, 2014)

It is an AP Yara Monitor, 60Hz - 33kHz


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## jtalden (Mar 12, 2009)

I understand you want to create an FIR filter for EQ purposes. 

To determine an EQ filter to improve the sound quality in our listening area we are normally only concerned with the combined (speaker/room) response in that area. In that case there is no significant benefit to near field speaker measurements. I haven't seen any of the automated commercial systems that utilize a near field measurement in the filter creation process. People often refer to "room response" or "Room EQ", but it is normally intended to mean the combined response given a particular setup in a room.

For those that are DIY regarding the speaker XO or other speaker design considerations then quasi anechoic speaker measurements are helpful. Some of us are just curious; nothing wrong with that.

Regarding some of your other specific questions/concerns:
> natehanson66 his providing good direction regarding gating for quasi anechoic measurement, but, like he, I am not sure how this helps in designing EQ filters.

> Your comment regarding "impulse activity before time = 0": You are correct this is not due to resampling. This is a result of the anti-aliasing filter chosen for the DAC(s) in your measurement system. This is no concern to the sound quality in any practical sense. The anti-aliasing filter type applied can result in this IR appearance. It is not something that can be heard and there are pros and cons related to the choices made. A DAC that shows this type of "pre-ringing" is not "better" than one that doesn't it is a design tradeoff. Some DACs provide an option for the type of anti-aliasing filter to apply. While some audiophiles indicate they can hear a difference, I doubt they can agree on which type is "better".

There are other causes of pre-ringing that can be a problem, so not all pre-ringing is harmless. In this case it is. The DAC that caused this is operating at 48K. I can tell that because the period of the ringing is ~20.83µs. The envelope shape of this pre-ringing is also similar to that as found in other anti-aliasing filters.

> Single sweeps have no disadvantage in the development of an EQ filter; no worries there.


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## JohnM (Apr 11, 2006)

natehansen66 said:


> Like John said, stick with the default window (Rectangular).


The default window shape for the IR is Tukey 0.25, which is flat apart from the final 25% which is a Hann window. That reduces the artefacts from the discontinuity at the edge of a rectangular window.


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## natehansen66 (Feb 20, 2011)

John - thanks for the correction. I must have changed to rectangular so long ago that I just assumed it was the default!


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## Primare Knob (Dec 4, 2014)

jtalden said:


> I understand you want to create an FIR filter for EQ purposes.
> 
> To determine an EQ filter to improve the sound quality in our listening area we are normally only concerned with the combined (speaker/room) response in that area. In that case there is no significant benefit to near field speaker measurements. I haven't seen any of the automated commercial systems that utilize a near field measurement in the filter creation process. People often refer to "room response" or "Room EQ", but it is normally intended to mean the combined response given a particular setup in a room.
> 
> ...


Thank you for all the response guys, it is certainly helping me to understand better the relationship between things, and learning to trust my own measurements. I am certainly the curious kind of type, and messuring and under standing my speaker response is my way of learning, because I can compare this with a refference. 

How does the impulse response relate to samples? 1/48000 = 20.83us 

Do I need a work around for my in room measurements, 5 channels, since I have the Dayton calibration file which seems to be calibrated for on axis measurments. Could I take a gated on axis speaker measurment and off axis measurement and compare the differences?


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## jtalden (Mar 12, 2009)

It would have more correct and clear if I had indicated the expected period of the ringing to be at the Nyquist frequency (24kHz) or 41.7µs. Below is a zoom of your impulse so we can better see the pre ringing. Note that ringing is spaced uniformly and the time for 5 cycles is roughly measured as 207.8µs so; 1 / (207.8µs / 5cycles) = ~24,062Hz. This agrees closely with 24kHz Nyquist frequency for a 48kHz DAC. 

I really don't know if this represents a normal feature of using a linear phase anti-aliasing filter, or possibly something else. I only know that the 2 DACs I tested with this characteristic measured very well for SPL, phase, and THD characteristics. Since you are not setup for a loopback measurement on the DAC you will not be able to measure those items in your case. It may not be ideal for a DAC but it is not a fatal problem for REW purposes.

On a loopback test you can see the ringing continue after the IR peak also. When measuring the room with a mic the post ringing is buried in the measured IR response and thus not noticeable.

I just wanted you know that it is a DAC issue not a speaker/room issue and also that it isn't a rare situation.

I don't understand the premise of your question about needing a workaround so I have no comment there.


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## Primare Knob (Dec 4, 2014)

Thanks jtalden. I am amased how much people are able to read out of one measurement. I will certainly have to dig in deeper to get the broader picture. 

The calibrated mic file for the Dayton UMM-6 is an on axis measurement only. When I am going to measure my room response the measurment will be off axis, and technically the calibration file could be off. Just something that I have read on this forum. Could I make an off axis calibration file by doing the same measurment with a calibrated on axis aim, and an off axis aim, and then extract one off the other to create an off axis calibration file?

When I am going to do the room response with my 5.1 channel setup, am I supposed to be doing a measurment for every single speaker at listen position with the mic in the same postion, which the recommended postion seems to be aiming at the ceiling or at the corner of the front wall and the ceiling.

My apologies for asking simple questions. I haven't read that far ahead yet.


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## jtalden (Mar 12, 2009)

Measurements for EQ done at the LP do not need to be at the same position. The mic can be pointed in the general direction of the speaker using a 0° cal file. Pointing it within 30° of the direction of the speaker has almost no influence in my room so I don't think small errors in this regard would normally be expected to be a problem. 

It is even preferable to average several measurement at different positions around the listening position. If we are interested in optimizing for one seat we can stay within 0.5m of that LP. If we spread the measurement area to accommodate 2 or more seats, it will be at some detriment to the main seat. The amount of detriment is related the particular room setup. 

The advantage of averaging is in smoothing out the SPL measurement in the midrange where reflections occur. This makes it slightly less likely we would be tempted to over EQ for a particular SPL aberration. The advantage can be small and unnecessary depending on the room setup however. It also can help slightly above 1k particularly in case where there are strong early reflections/diffraction from the speaker box. 

There is also the "Moving Mic Measurement" (MMM) method that does the same thing by using "forever" averaging within the RTA feature. A browser search will detail this. *Here* is one link.

The point is that small mic position changes can influence the measurement and averaging can improve the repeatability of the measurement while finding the central tendency. Averaging is not required for good results but it does help explain why the mic can be in slightly different positions for different speaker measurements.

See *here* regarding the other question about creating a 90° cal file. It can be done very satisfactorily if we use good techniques. I have compared my results to a supplied 90° cal for my mic with close agreement. My method was basically similar to this reference although I used other methods for smoothing the results.


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## Primare Knob (Dec 4, 2014)

Thank you all. I will read through all the information and do a complete system measurement once I have everything complete again. At some stage I will probably get a different mic when I understand all of this better.

One last question, does smoothing the curve have any influence on frequency limit like with gating? I remember reading somewhere where they gated and smoothed the curve, which pushed the frequency limit way up into the midrange.


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## jtalden (Mar 12, 2009)

Primare Knob said:


> One last question...


We've heard that before. 



> ...does smoothing the curve have any influence on frequency limit like with gating? I remember reading somewhere where they gated and smoothed the curve, which pushed the frequency limit way up into the midrange.


That's correct, gating the IR will limit the lower frequencies. Assuming left window is very close to the start of the IR REW will properly truncate the SPL and Phase chart at the lowest freq that is usable. The first octave or 2 displayed is overly smoothed so its best to keep that in mind.

If the left window is set large then chart shows more low freq than there is data for. REW can't tell if you are including meaningless (zeroed) data before the IR starts. In your Post 3 measurement with windows set to 1 and 3 REW will show the low frequency limit as 250 Hz. If 0.1 and 3 are used then the low limit is calculated as 323Hz. With either setting I would suggest the low limit is 323Hz and the data below 2x323 (646Hz) is overly smoothed. As natehansen66 suggested it is convenient to use 1ms for left window for several reasons. I use that setting most often.


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