Help with strong left image but weak right

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could you post the one at 5 ms?
Let me know if you meant something else. I assume you are asking for a before/after treating the 5ms reflection. If so, the 'before' would the left_1_forward_wall and the after would be the left_2_side_wall, but I went ahead and did before/after for both walls and speakers:



For the left and right, I have an impulse response for



0 = baseline (no absorption)
1 = baseline + absorption on forward wall (calculated reflection ~10ms)
2 = 1 + absorption on side (right) wall (calculated reflection ~5ms for left speaker and ~10 ms for right speaker)
3 = 'final' is the same as 2 but a few days later
 

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Hm, a bummer from me. I did read you have a port firing out from the backside of the speaker. Still, I forgot that, -sorry. The port needs some breathing space of course, a distance like the port diameter away from the wall should be OK but preferably larger. Consideration to be taken though are in what range the sub will / can play so you avoid the null you have around 50-60 Hz. That means either close to the wall behind or very far into the room. Close to the wall: If it plays up to say 80 Hz, no further than 344/80/4 = 1,09 m, so make that about 80-90 cm or closer. Out into the room and it is capable down to say 35 Hz: 344/35/4 = 2,45 from the wall behind, make that more than 2,5 m. Increased distance means a dip at lower frequency. For a sub playing 35-80 Hz the ”no go zone” would be roughly 90-250 cm from the wall behind it. If it is placed in the ”no go zone”, you can expect a more or less large dip in the frequncy curve. The same idea applies for the main speakers and the range they play. The crossover frequency determines how far away one should place the main speakers from the sub. Within 1/8 to about ¼ of the crossover wave length is OK, preferably not further away.

As for pinpointing instruments in the soundstage, if that is the important thing and increased soundstage width is less important. If the engineer in the control room has done a good job, it is there in the recording, panned between the 2 stereo speakers. In a room each speaker has several mirror images though, = reflection points in the walls, ceiling and floor. Many sound sources, around at different places, are of no help for pinpointing what is on the recording.

The Haas effect / precendence effect locks your hearing to the direction of the first arriving sound = direct sound. Delayed (reflected) sound arriving after about 5 ms has to be about 10 dB louder than the direct sound to make a change of that first heard direction. The 0-5 ms is a bit of a zone of confusion where less dB-strength is needed. If these reflection come close enough in time, are of sufficient strength and from a certain angle, you may receive double messages about the pinpointing and it is also depending on frequency.

From ”Acoustics and Psychacoustics” by David Howard and James Angus:
”Thus the interaural intensity difference is a cue for direction at high frequencies whereas the interaural time difference is a cue for direction at low frequencies. Note that the cross-over between the two techniques starts at about 700 Hz and would be complete at about four times this frequency at 2.8 kHz. In between these two frequencies the ability of our ears to resolve direction is not as good as at other frequencies.” and:

”The main compromise in stereo sound reproduction is the presence of spurious direction cues in the listening environment because the loudspeakers and environment will all contribute cues about their position in the room, which have nothing to do with the original recording.”

It is a couple of years ago since I used REW last time but I found the ETC (Energy Time Curve) to be of better help than the Impulse Response. Tick the box for Schroeder integral and you get the overall decay time in your room. If the Schroeder Integral is a more or less straight down sloping line, fine! If it is not constantly decreasing and has large bumps, it shows you have a decay that varies, modes go up and down in time. Often because of irregular L-shaped rooms or the room has some heavy rigid walls and some light weight stud walls which low frequncies passes through and gets reflected back from another room with heavier walls. -You may get a kind of pulsating sound, this can also be seen in the waterfall diagram. You can also filter the ETC to show certain frequencies so you get an idea how the decay rate is different depending on lower or higher frequencies, one can also get a better idea which frequency causes the impulse peak you see at a certain time.
 
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Thanks, adhoc. I'm still absorbing this and your last post since I haven't had much time to look at this the last week. The article from last post and what you wrote here are very helpful but take some time to sink in.



Pano, were you able to make any use of my measurements?
 
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Yes, a few days back I ran them and found a few things, then got busy and forgot about it. :(

Anyway, was looking at the mod response and there are some definite differences between the two sides. I couldn't tell much from the ETC, but decay and waterfall show the differences well enough. Try looking at those, using the other channel as an overlay. It can be hard to visually separate the amplitude differences from the time differences, but both are there. I can post the graphics I got, if you like.
 
I appreciate you taking a look. I'll look at both of those displays, but do you any thoughts on what I should be looking for? I have yet to read up on what the decay shows and how to interpret it, and I have only seen the waterfall plot discussed for low end issues. Did you look at the full spectrum here?


And adhoc, I moved the speakers about 5" from the wall (3" diameter) and certainly heard (and felt) a low end improvement, but it also produced some rather boomy, one-note bass. (I suppose that's where the super chunks would help?) I also moved them way out into the room, beyond the point of generating the null at 38Hz (speaker's lower limit). Since that put me closer to the middle of the room, I had to sit further away from the speakers, but I did like it. There wasn't as much bass, but it added some depth to the image which was pretty neat to hear.
 
As your room is not a typical shoebox room (seems so anyway) and your setup of speakers and LP is not symmetrical versus the inner walls and large furniture, take any advices with a grain of salt. There are many things to consider, like if all walls are of the same or different materials which will then reflect energy differently.

Regarding boomy bass, it is probably from an axial mode and most disturbing at a certain LP. If it is a length mode and a low tone, superchunks will not be extremly efficient. Why? Lets say the room is a shoebox one with massive stiff walls and room size is W4,5 x L7,8 x H2,5 m. 1st length mode would be 22 Hz, 2nd 44 Hz and 3rd one 66 Hz. If LP is at a peak for 44 Hz it will certainly sound boomy.

Now, measure the SPL for 44 Hz close to and along the short wall of 4,5x2,5 m. Does the length mode vary a lot up-down or from side to side? Probably not. A 44 Hz wave length is 344/44 = 7,8 m (22,5’) so it covers the whole short wall! A superchunk from floor to ceiling with a base of 60 cm / 2 feet and 0,85 m diagonal covers only 0,6 / 4,5 = 13,3% of the wall area. 2 superchunks about 26% of the total wall area. The depth of that superchunk goes from 0,6 m along the side wall to 0 m towards middle of the room, average depth = 0,3 m. If one dives into books like Acoustic Absorbers and Diffusers (by Cox and D’Antonio) one will find that to start to be significantly effective a resistive absorber needs to be at least 1/10 of a wave length. The max thickness 0,6 m compared to 7,8 m of 44 Hz is less than 1/10 and it is 0,6 m thick only at a very short width … The rest of the 4,5 m wide wall has no effective absorbtion at all versus 44 Hz. So, superchunks are rather worthless for low frequency bass and length modes due to their small width versus (common) room width. You need to cover the whole wall. Very deep resitive absorbers. A membrane absorber or helmholtz absorber is a better choice as they require less depth to be effective at very low frequencies. One can add resitive asborbers for upper bass / lower midrange on the outside of them if needed.

To figure out if your peak / dip in a frequency curve comes from SBIR or is mode related, this link has a pretty easy to use calculator for SBIR: SBIR calculator .
 
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Thanks, adhoc. Would you recommend "Absorbers and Diffusers"? Part of my problem is I have read a lot of various sources but most simply discuss a rough procedure for setting up and treating your room. Without a lot of understanding of why the procedures work and since many are often contradictory, I wind up confused. A reference that starts off assuming you know nothing and working up to an understanding of why you do certain things would certainly help me.

In fact, I've decided to just leave it for now and focus on other things with the idea that I'll come back to it at some point down the line. I've had quite a bit of fun recently pulling out the crossovers and going active...
 
It is a good book but it will set you back US$ 150-200 at Amazon, depending on which edition you go for. (I have the 3rd edition). It contains a lot of good easy to read info but also heavy theory and math about absorbers and diffusers. If your main interest is ”how do I fix / build my room”, you have to read between the lines and draw your own conclusions from what your situation is. It is then also beneficial if you are good in math (university degree).

A good ”down to earth” book with minimal math required is Rod Gervais ”Build it like the Pros”. You could also check out John H Brandts homepage. Under resources he has some good advices and drawings: Resources - John H. Brandt Acoustic Designs
 
How is the math presented? Is it something very specific to the issues in the book and not much more than equations, or does it provide/develop a mathematical framework to discuss what it is doing? If it's closer to the latter, the price may be worth it since I want to learn more general signal processing. (I am assuming there is strong overlap with room acoustics and both analogue and digital audio since both are essentially taking an input signal source, applying something to it, and producing an output signal.)
 
If one is able to follow the math, I believe it is quite straight forward and down to the point. (I certainly can't. Too many years ago and no practical use in every day work with "higher math", so whatever stuck for some time is long gone by now.)

I include some pages of what to me is now mostly gibberish.
 

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