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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.

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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:

Text Line Parallel Font Number


And above zoomed in:

Text Red Line Font Plot



View of 107cm measure of IR:

Text Line Yellow Font Design


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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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:

Text Blue Line Plot Wave


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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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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Blackman type windows are smooth and highly symmetrical. They introduce very little ripple in FR display relative to other types. "7" refers to number of terms in formula. This has smooth performance over very large dynamic range, four terms is really quite sufficient for 16bit data on CD; even exceptional speaker and measurement circumstance is typically <12bits of useful dynamic range.


In near field measurements, ratio of direct to reflected sound is very high. For Blackman-Harris 7 12ms window, weighting is 50% at 3ms, and falls off rapidly. Combination leads to very good overview. Exact shape of curve going down to 60Hz may be overly smooth, but likely captures realistic -6dB point of speaker with minimal interaction of floor and walls. This low frequency curve fall off as microphone moves closer due to relative spacing of dual woofers and port.

In summary: You don't like the low end of your speakers, so you got two subs, and apparently you are not happy with the high end either. I'd seriously consider a nice pair of active two-ways with sealed cabinets so they integrate well with subs.
 

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12ms is good for general overview; shorter windows more fully limit room interactions, and require little if any additional smoothing when concentrating on HF performance; even 1ms for working above 1kHz.

Try running some near field measures (<1m) at small angles off axis, vertical and horizontal and see how peaks and dips shift. Also do at listening position. Sometimes small changes to speaker location, and toe-in can bring significant changes to perceptions at listening location. Even though speakers are floor standers, sometimes feet or small stands that add small amount of tilt can be used for focusing alignment at listening position too.

I used to have floor standing speakers with four drivers that where very difficult to get right. Getting a friend to tweak toe-in while listening always worked better than using a tape measure and repeatedly getting up to bump a speaker a fraction of an inch this way or that.
 
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