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Do you mean "narrow IR window" settings?

Mic distance needs to be close but far enough away for transducer integration - 0.5 m for small 2-way, 1 m for 3-way or small floor standing, 2 m for large multi-way floor standing - on tweeter axis or, if really close, on axis midway between tweeter and midrange driver.

In REW > IR Windows > Right Window: try values between 2 ms and 5 ms, then click on Apply Windows.

REW smoothing 1/24 oct or 1/48 oct.
 

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Discussion Starter #3
Do you mean "narrow IR window" settings?

Mic distance needs to be close but far enough away for transducer integration - 0.5 m for small 2-way, 1 m for 3-way or small floor standing, 2 m for large multi-way floor standing - on tweeter axis or, if really close, on axis midway between tweeter and midrange driver.

In REW > IR Windows > Right Window: try values between 2 ms and 5 ms, then click on Apply Windows.

REW smoothing 1/24 oct or 1/48 oct.
Thank you very much.

I am trying to get a near field measurement for my Energy Veritas 2.3i floor standers. The 2.4i were the biggest in the range.
I am trying to determine what the manufacturer intended them to sound like characteristics etc.

For this size speaker which would you recommend distance wise etc?

For the REW settings i am trying to setup the program to give me the near field results without room interference etc. I don't want to take the speakers outside.

Currently the IR window left and right is set to Tukey 0.25, is this ok?

Thanks again
 

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Unfortunately, no one can tell you what gate settings to use in your room.....you need to take a measurement with the speaker and mic in your desired position and look at the impulse to see where the first reflection happens. When you click on the IR Windows button, don't worry about the "Left" and "Right" window setting at the top. Leave that where it's set. What you need to change is the "Left Window (ms)" and "Right Window (ms)". In order to set those right you need to make a measurement.

In my picture is an impulse from my speaker, which is on a stand and the mic is about 6 feet from the speaker. You can see I have the first reflection marked. It will be the first major blip after the initial impulse from the speaker flattens out. This is a reflection from the room, and you don't want that. In my case shown, I set the left window to about .5ms before the initial rise of the impulse and the right window to a little less than 3ms. This will set the gate and window out the room influence. The only caveat here is that since we are truncating the impulse of the speaker we can't see all the information. A 3ms gate will give you info from about 300hz up. IMO the data below 1Khz is questionable because the frequency resolution is quite low to see the detail of what's actually going on, but it will give you a good idea of the basic trend. You can't set the gate long enough to get LF data without the room swamping the response.
 

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Discussion Starter #5
Unfortunately, no one can tell you what gate settings to use in your room.....you need to take a measurement with the speaker and mic in your desired position and look at the impulse to see where the first reflection happens. When you click on the IR Windows button, don't worry about the "Left" and "Right" window setting at the top. Leave that where it's set. What you need to change is the "Left Window (ms)" and "Right Window (ms)". In order to set those right you need to make a measurement.

In my picture is an impulse from my speaker, which is on a stand and the mic is about 6 feet from the speaker. You can see I have the first reflection marked. It will be the first major blip after the initial impulse from the speaker flattens out. This is a reflection from the room, and you don't want that. In my case shown, I set the left window to about .5ms before the initial rise of the impulse and the right window to a little less than 3ms. This will set the gate and window out the room influence. The only caveat here is that since we are truncating the impulse of the speaker we can't see all the information. A 3ms gate will give you info from about 300hz up. IMO the data below 1Khz is questionable because the frequency resolution is quite low to see the detail of what's actually going on, but it will give you a good idea of the basic trend. You can't set the gate long enough to get LF data without the room swamping the response.
Thank you

How did you determine the 6 feet for measurements, the same as above posts?

Did you position the mic between the tweeter and mid range?

Thanks again
 

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My speakers are Synergy horns, so the mid and tweeter radiate from the same "point" in the horn. I determined the 6' because that's about as far away as I feel I can get in room and still get good data. I'd like to go outside and go 12' or so ideally.

In your case positioning between the mid and tweet 1 meter from the speaker should do it if you're trying to see how well it matches up to the manufacturer's anechoic response. 1 meter distance is typical.
 

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My speakers are Synergy horns, so the mid and tweeter radiate from the same "point" in the horn. I determined the 6' because that's about as far away as I feel I can get in room and still get good data. I'd like to go outside and go 12' or so ideally.

In your case positioning between the mid and tweet 1 meter from the speaker should do it if you're trying to see how well it matches up to the manufacturer's anechoic response. 1 meter distance is typical.
Hello Nate,

Would you be willing to share a bit more info on Synergy horns? Are these DIY?

Above posted IR is very tantalizing. According to theory, you should be able to get good measurements even with microphone in plane of horn opening.

-----------------

With my DSP based clones of Linkwitz Pluto, driver setup allows referencing at 23cm. Inverse transfer functions of each driver become basis of EQ/crossover with convolution setup. First reflection becomes floor bounce that is essentially directly below speaker/microphone, with round trip of about 6ms.

Measured FR at 23cm, no smoothing and very big window:



Measured FR at 107cm, no smoothing very big window:




Using same 107cm measurement, with same very big window, but gating data to 3ms the response looks like this:



No additional smoothing is applied to above result. Dips in raw 107cm result are really what happens when reflections come into play, but human perception is much more like gated result. Only tones with significant sustain allow hearing the dips that are setup. The direct sound of speaker is single most important in playback. Equalizing for room gain and peaks/dips for listening setup then follow, if really needed/desired.

Normalized full scale view of 23cm measure of IR:

ir 23cm overview.png

And above zoomed in:

ir 23cm zoom.png


View of 107cm measure of IR:

ir 107cm zoom.png

In last picture reflection a bit past 1ms is seen. This is reflections from microphone body mounted to tripod. This reflection is present, but very hard to pick out in 23cm measurements, and do to geometry of closer setup occurs a little earlier.

Similarly, little peak in Nate's pic at about 0.8ms may partially be microphone related reflection.

Regards,

Andrew
 

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To keep from derailing the thread too far....I should have said "DIY" Synergy horns, and Paralines at that. These are no way indicative of anything from Danley. As you can see mine have their issues :rolleyes:
 

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Discussion Starter #9
In your case positioning between the mid and tweet 1 meter from the speaker should do it if you're trying to see how well it matches up to the manufacturer's anechoic response. 1 meter distance is typical.
Thank you

I can't find the manufacturer's anechoic response, i have emailed Energy but because of changes they haven't got records of this. Any ideas would be appreciated.

I will give this a go and post the results.

Thanks again
 

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The direct sound of speaker is single most important in playback. Equalizing for room gain and peaks/dips for listening setup then follow, if really needed/desired.

Thank you

Interesting, i have heard that equalizing for the direct sound can be preferred.

Also seen that equalizing using the Anechoic response as a target can give the listener the true response the manufacturer intended, or close as possible.
 

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No additional smoothing is applied to above result. Dips in raw 107cm result are really what happens when reflections come into play, but human perception is much more like gated result. Only tones with significant sustain allow hearing the dips that are setup. The direct sound of speaker is single most important in playback. Equalizing for room gain and peaks/dips for listening setup then follow, if really needed/desired.
Emphasis added.

Andrew:

I am enjoying your informative posts.

One of the psycho-acoustical concepts I have been trying to getting my arms around for years is the Haas effect (not the avocado:D), or Precedence effect. Your comments above got me thinking about it again as it relates to speaker and room measurements.

Quoting the Wikipedia "Precedence Effect" article, "When a sound is followed by another sound separated by a sufficiently short time delay," or less than about 40 to 50 mS, "listeners perceive a single fused auditory image; its perceived spatial location is dominated by the location of the first-arriving sound (the first wave front). The lagging sound also affects the perceived location. However, its effect is suppressed by the first-arriving sound."

So the way I understand it, if the listener hears a wavefront directly from the speaker, followed by an early reflection with, say, a 10 mS delay off a nearby wall, the listener will hear the sound from the speaker plus the early reflection as a single sound coming from the direction of the speaker. This does not mean that the early reflection is "tuned out" somehow, or ignored by the brain, only that is not heard as a separate sound (like one delayed 50 mS or more). Actually, it seems it would make the speaker sound "different," a simple reflection resulting usually in some comb effect filtering, and probably dulling the sharpness of imaging a bit. At least that is how I understand it to work.

Yet we talk all the time about using gating with speaker measurements in-room to get 1) a cleaner measurement free of room effects - that part I get - and 2) a measurement that is closer to what the ear would perceive - and that's the part that confuses me, the implication that the reflected sound gets ignored by the ear/brain.

An example that comes to mind is guitar doubling in a mix, where the delayed guitar sound is not heard as separate - it still sounds like one guitar, but it sure sounds different. Obviously, the doubling effect is a huge exaggeration of what we are talking about with speaker measurements.

Not trying to put you on the spot, actually hoping you can help me understand the principle a little better.

Thanks.
 

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Survival value of stereo vision and hearing existed long before speech and music.

In hearing, long before higher functions of brain have assigned advanced attributes to sound source, the size and direction of a source are deduced. High frequency components for assessing direction are processed faster than for lower frequencies. The lower the frequency the source, the less deterministic is its location. Two sources cannot be resolved as separate when closer than a 1/4 wavelength apart from a fixed point of observation.

Perceptual mechanism makes good use of timing difference in sound arrival at ears in about 500Hz-4kHz band, and additionally makes use of head shadow effects from within this bandwidth to higher frequencies.

Early reflections <1ms cannot be directly detected, and are integrated into the source. Reflections <3ms effect apparent direction. In context of stereo sound, delaying same sound in this range between speakers shifts virtual image location. Reflections >3ms up in time to perception of echo type effects increase perceived loudness, and direction of reflections builds perception of spaciousness in which source is occurring.

Many manifestations are possible, details may be lost or enhanced, including increase or loss of depth.

In typical domestic living/listening spaces that are highly reverberant, the initial source of a sound in a recording is still bouncing around at levels that interfere with the recorded source's reverberant tail, typically leading to masking of intended spaciousness of recording.

Overly dead rooms place great demands of speakers/amplifiers to get realistic playback levels.

The hearing mechanism does a good job at assigning a reflection to its specific source, and assigns less attention to the direction of the reflection, leaving more processing power to new events. When spectral content of reflections differ significantly from direct sound, the brain is constantly assessing if reflection is new source requiring detailed attention, and becomes fatiguing.

Longer measurement are required to assess balance of low frequency sound in room. The room becomes part of the speaker. Luckily this is in region where imaging is not a possibility.
 

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In referencing a new design, it is best to see first what basic performance is in reflection free environment. This is effectively carried out in an anechoic chamber, outdoors on a mast (with no air movement, very unlikely), in large warehouse on a mast with gating and windowing (still mighty inconvenient)....

A good commercial designer will anticipate application, and in case of product targeted for domestic living environments, speaker placement close to walls and room corners is highly anticipated. Often manufacturer recommends fairly specific setup to achieve targeted results. Design at this level anticipates some level of furnishing that absorbs and diffuses sound.

Designing for speaker on floor, a moderate distance from a front wall, and from a side wall can readily anticipate the characteristic room gain at low frequencies, roughly those for which boundary distances are <1/3 of a wavelength.

It is easy to anticipate that average consumer doesn't want floor space dominated by speakers. Less easy is anticipating listener location. Most speaker manufacturers pitching realistic hi-fi, will not only make stipulations about speaker placement, but also recommend listener location. Most that do this make clear that listener should be >3ft from back wall. In context that speakers are >=3ft from side walls and each equally spaced from front wall, symmetric relationship makes it highly likely that listener will be >3ft from side walls.

A lot goes on in 3ft. From back wall this is about 6ms difference in direct v reflected time, giving brain ample time for processing primary imaging cues. Direct to reflected intensity ratio is close, and remains so at low frequency even with extensive application of absorbent materials on back walls. Peaks and dips forming at listener's head shift dramatically with just a few inches of head shift fore and aft.

These effects are present in live listening as well. In large concert halls, small proportion of seats are effected by this, and tend to be the least desirable seats. In small intimate venues it is problematic, as more seats/tables tend to be along walls. One only needs get up and walk about during show in small club with attentive, quiet audience to hear continuously changing character of sound of bass, and lower registers of voice.

In typical domestic space with 8ft ceiling, room rapidly becomes transmission line, and modal behavior dominates. Equalization becomes control of ringing type behaviors by selecting peak or dip frequency and choosing Q and gain to suppress effect. This can be done at higher frequency, but becomes more location dependent. This becomes so much the case, that once again head shifts of a few inches results in totally different response.

Below about 60Hz in smallish rooms, modes are few and equalization is highly effective over fairly broad listening zone. Extending sub woofer duties higher, and using multiple subs may lead to more locations with smoother responses over wider range of measurement conditions, but transient attack becomes smeared.

In the end, compromise rules.

Personally for me this translates to: Maximize performance of direct sound, and season to needs of situation at lower frequencies.


Phillips;
I am guessing that you will find near field type measurements reveal very nice performance of your speakers >125Hz.

Adjustment to EQ <125Hz is heavily impacted by listening level, engineering of recording, and state of mind. State of mind ranges from general mood, to cumulative listening is past minutes, hours, and even days. Tweaking and listening hard at realistic levels for an hour, leaving all settings the same including volume, coming back the next morning at hitting play can prove quite distasteful. Kind of like jumping out of bed and starting the day with a full on sprint without warming up.
 

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A familiar question; what are your perceptions of your speakers that you would like to change?

I review above mdat files; importing to one, adjusting all windows to 12ms Blackmann-Harris 7, smoothing 1/6 octave, and adjusting levels to get overlay:

spl overlay 12ms window.png

Speaker as measured could be described as having flat response ±3dB over significant portion of spectrum. A trained eye sees ripple due to diffraction across width of front baffle. Location of peaks/dips >1kHz will likely shift as measurement point is moved off axis. Some of response is inherent to driver spacing, crossover points, and crossover slopes.

Very little that can be done, even with very powerful processing techniques. Bass in 100Hz region could be broadly boosted; but also needs to take in considerations of subwoofer. If peaks/dips >1kHz remain fairly constant as microphone is moved off axis, some EQ is possible with equalizer with very narrow bands of control. Results are not likely to be dramatic, even if feasible.
 

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A familiar question; what are your perceptions of your speakers that you would like to change?
Thank you

Basically the high above 5 kHz falls off at listening position.

I review above mdat files; importing to one, adjusting all windows to 12ms Blackmann-Harris 7, smoothing 1/6 octave, and adjusting levels to get overlay:

View attachment 42568
Those dips and peaks are pretty much followed the same, seen in the listening position graphs, other than the 5 khz drop off.

Can you please show your adjustments in the IR window and Impulse graph to get this result.

Speaker as measured could be described as having flat response ±3dB over significant portion of spectrum. A trained eye sees ripple due to diffraction across width of front baffle. Location of peaks/dips >1kHz will likely shift as measurement point is moved off axis. Some of response is inherent to driver spacing, crossover points, and crossover slopes.
They are bi wired would this make a difference.
Does bi wiring bypass the crossover?

Very little that can be done, even with very powerful processing techniques. Bass in 100Hz region could be broadly boosted; but also needs to take in considerations of subwoofer. If peaks/dips >1kHz remain fairly constant as microphone is moved off axis, some EQ is possible with equalizer with very narrow bands of control. Results are not likely to be dramatic, even if feasible.
Currently i am running 2 x REL Strata 5s high level.

These measurements were taken with the Dspeaker Antimode Dual Core 2.0 disabled (with a push of a button).

Thanks again
 

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Ok,

What convinces you that you are having a problem >5k, and what have you done to change this? And what happened when you attempted to change this?


You seek measurement techniques, yet something tells you where the problem is? Or has previous measurement convinced you that a problem exists in one place, yet experiments in this place yield no fruit?

As I recall, three-way speaker with dome mid and dome tweeter. These, once again in conjunction with baffle width, and these driver's distances from top edge too, and in conjunction with likely high crossover point between them lead to ripple, and comb filtering. Additionally comb filtering exists when both speaker are running, and none of this noted in literature to cause any audible effect with normal program material. Sure, sweeps, or continuous noise reveals all this with measurements, and just slowly moving about listening space.
 

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Ok,

What convinces you that you are having a problem >5k,
Thank you

With REW test sweeps for each speaker and then combined.

and what have you done to change this?
Used Treble Tone Tilt and PEQ.

And what happened when you attempted to change this?
It improved bringing up the treble, but still remains. I tried not to use too much boost.

You seek measurement techniques, yet something tells you where the problem is? Or has previous measurement convinced you that a problem exists in one place, yet experiments in this place yield no fruit?
Correct in the latter comment.

As I recall, three-way speaker with dome mid and dome tweeter. These, once again in conjunction with baffle width, and these driver's distances from top edge too, and in conjunction with likely high crossover point between them lead to ripple, and comb filtering. Additionally comb filtering exists when both speaker are running, and none of this noted in literature to cause any audible effect with normal program material. Sure, sweeps, or continuous noise reveals all this with measurements, and just slowly moving about listening space
I first thought my ears where either blocked or decreasing in the higher frequencies until i started to use REW, and realized my ears were not bad after all.

The main reason for the near field measurements were to see the characteristics of the speakers, and see if this was a characteristic of the Veritas.

When i get home i will post a mdat file of my Veritas in the listening position.

Thanks again
 
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