A reflection from the floor or ceiling is not going to give a sense of increased realism though. As Dave points out our binaural hearing can't easily separate the time delayed reflections that arrive from a different elevation but same azimuth, the result is comb filtering and sometimes a "stretching" of the apparent vertical size of the image.
The only hope we have to reproduce some actual height data in a stereo recording is HRTF related changes in the high treble frequencies, but most recordings made with normal mics would not encode such data, (unless added electronically as an HRTF effect during mixing) and floor/ceiling reflections are likely to corrupt any such HRTF encoded height data, not enhance it.
As far as I can see, floor and ceiling reflections are only harmful.
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Member
Joined 2009
http://www.diyaudio.com/forums/multi-way/211816-waveguie-cardioid-slim-baffle-5.html#post3035949
Post #50 describes the current state of the "enclosure". I decided to go open back and use "driver mufflers" to absorb some of the rear wave. I have trouble getting measurements from 90-160 degrees indoors but from what I was able to get I would say the response is supercardioid.
I was planing to do outside measurements this weekend but I found out the landlord is showing the upstairs apartment to potential tenants. It will not look good if they walk on me blasting sweeps in the backyard. It might be some time before I get reliable polars.
It might or it might not. Nobody has ever shown which reflections from what directions are the most relevant for increased realism. Any reflection creates comb filtering yet we don't perceive it as such. Time is very relevant to our hearing.
I had imagined that as the reflecting surfaces area grew with SPL intensity .. the surfaces become too close together to form complete (sort of) waves at our listening position.
... that the polar response (area produced by a given SPL) .. in air ... is as a conduit for subsequent sound waves. Sound waves travel right out to the polar edge very quickly .. (waves riding waves). As the SPL increases ... so does the area of the "conduit"? .. and therefore the shorter the distance becomes to another reflective area?
... that the polar response (area produced by a given SPL) .. in air ... is as a conduit for subsequent sound waves. Sound waves travel right out to the polar edge very quickly .. (waves riding waves). As the SPL increases ... so does the area of the "conduit"? .. and therefore the shorter the distance becomes to another reflective area?
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Yep, that's my experience too. An excessively reflective/reverberant room becomes confused and overwhelming at high SPL, while the same or even higher SPL in a more damped room sounds fine, and both sound fine at low SPL.
Why ? I don't know, my guess is there is some sort of threshold effect in our perception of reverberation level, as SPL increases we pass that threshold sooner in a reverberant room.
I also notice that my subjective opinion of how reverberant the room is and perceived direct to reflected ratio changes with SPL level.
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I understand.
... was thinking when I posted above was that SPL intensity was like a balloon where the physics for sound waves on the inside might differ slightly from the sound waves physics outside the balloon. A perfect environment inside this cloud (waves appear at the edge almost instantly in all directions) .. until the balloons edge encounters another independent surface (brought on by increased SPL intensity). Then the "perfect environment" inside the SPL structure is upset.
Well, physics stays the same regardless of SPL. What changes is our perception. There are little muscles in our ear that might have a bigger impact on sound perception than commonly thought.
There is a reason for this sensation. Yes, there is a relation to SPL, but its easier to ignore this for a moment to make a point - lets say there is a 'reference' listening level where muddiness happens, and we are not going to lower the listening level to solve the issue.
There are two different reasons - one for small rooms the other for large concert halls. First there is a image forming time span, Haas interval. Reflections don't change the apparent source position that is established by the direct signal - law of first reflection. This doesn't mean that high gain reflections are ignored! They can cause image smear, tonality errors, loss of intelligibility etc. Apart from flutter, usually this is the main reason a small room can sound muddy and .. small. (aren't generalizations nice!
Lets ignore modal activity; a 150Hz highpassed signal can also sound muddy)This is one reason why some control rooms (critical listening rooms) are designed to have an effectively anechoic response to ~20ms. This gives the brain enough time to form a 'unconfused' image while a strong reflection ending this period gives a nice spacious sensation to the presentation without giving the 'confused sound' effect.
Muddiness in a large space is found to come from reflections arriving around 50-80ms. Thats why some of the best halls have dual slope decay - strong lateral reflections beginning ~20ms, fast(er) decay until 120ms, then a slowly decaying reverb giving good envelopment.
In small rooms and large rooms alike when a 'ff' moment comes in music there are 2 possibilities - either confusion/muddiness/fatigue or the sound 'magically' envelopes the listener. This doesn't explain why there is a level dependent threshold, but it is good to know the effect exists so that it can be 'engineered' around.
It's definitely a perceptual change with SPL, not something else.
The better you become as a "trained" listener, the more you start to notice the changes in your perception of sound which occur for everyone, but most people don't notice.
I often notice that my system sounds different from day to day, (since I'm always a critical listener, even when I just want to relax and enjoy the music...) and it used to drive me around the bend trying to figure out what was changing, until after lots of critical measuring and testing of equipment, checking connections and so on, I realised nothing in the system was changing, only my perception was changing
I notice changes in apparent tonal balance between too much bass and not enough bass from one day to another, changes in imaging between immersive and engrossing one day to somewhat flat and disappointing a few days later, changes in apparent room liveness from moderately well damped one day to overly reverberant another day, even changes in perceived volume - where the volume setting I'm using on the same music will be quite a lot higher or lower to achieve the same perceived "optimal" listening level...
Some factors that I've been able to correlate include time of day, (things seem louder to me early in the morning or late at night, making me set the volume level lower subconsciously) whether I'm tired or alert, (it always sounds worse when I'm tired) whether I've exercised / worked out in the last few days, (sounds much better if I have) mental state - happy and invigorated, or depressed - sounds worse if I'm feeling depressed, whether I've spent most of the day in a quiet serene place like staying at home indoors, or whether I've been subject to loud traffic/city/work noises all day.
All these (and probably more) affect my perception, so I've learnt that when I sit down and switch on some familiar music and it sounds a bit flat and un-involving not to go looking for whats wrong with the stereo and just accept that its not a good time for me to listen. Likewise I'll sometimes switch it on and listen and it can be an amazing experience where I'll listen for hours and can't stop because it sounds so good...
This is why as much as I place importance on how a speaker sounds, I never go by listening alone, I'll always back it up by measurements, and any changes to the response that I'm testing I'll typically evaluate over several weeks, to even out the daily "fluctuations" in my perception. Otherwise you just end up going around in circles tweaking never making any progress.
In regards to the original question about the effects of reverberation at different SPL - as I mentioned above I perceive not only differences in room reverberation on different days, but also different required volume levels for "normal" playback, and I think the different playback levels that seem "normal" on different days directly result in the difference in perceived room reverberation, and probably some of the perceived difference in tonal balance in the bass too.
Ears are very acute measuring devices in some ways but they are certainly not to be trusted on their own if you want any sort of consistency...they constantly go in and out of "calibration"
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Member
Joined 2009
OB amplification trouble
I am still working on this project and have some promising updates. I got into big trouble in the beginning of the summer when I realized that I need a huge amount of power to compensate the dipole loss in the woofer section. I had to reconsider my requirements or abandon the project.
After some thought I decided that a single channel of amplification for the 4 woofers is unrealistic. I added another amplifier channel to the woofer section and now each pair of woofers are driven in parallel by their own amp.
The speaker requires 8 channels of amplification total (+ an optional subwoofer). I am planning to use a high-end class D 7.1 receiver which has 7x110W channels. Additionally I plan to use a pair of miniDSP modules which will have 8 channels as well and that setup won't allow for an additional subwoofer.
I decided it will be best to convert the Mid-Tweeter crossover into a passive one. The speaker will be a hybrid passive-active 3-amped one.
I ran some simulations and I don't think that a phase-accurate system can accomplished with more than 1 le'Cleach crossovers. My initial idea of a phase-accurate system with IIR filters was probably unrealistic. At this point I'm looking into a passive acoustic 2nd order Linkwitz-Riley on the Midwoofer and Tweeter at around 2kHz, then an active 2nd order LR on the Mid and the Woofer array somewhere into the 200-400Hz region. As I mentioned earlier the woofer array will be driven by 2 amplifier channels.
The overall sensitivity should be 85-88dB @ 1m, not bad for OB.
I've been listening to the prototype and getting very excited. The bass remains very even throughout the room, it's quite a contrast from my boxed speaker.
I am still working on this project and have some promising updates. I got into big trouble in the beginning of the summer when I realized that I need a huge amount of power to compensate the dipole loss in the woofer section. I had to reconsider my requirements or abandon the project.
After some thought I decided that a single channel of amplification for the 4 woofers is unrealistic. I added another amplifier channel to the woofer section and now each pair of woofers are driven in parallel by their own amp.
The speaker requires 8 channels of amplification total (+ an optional subwoofer). I am planning to use a high-end class D 7.1 receiver which has 7x110W channels. Additionally I plan to use a pair of miniDSP modules which will have 8 channels as well and that setup won't allow for an additional subwoofer.
I decided it will be best to convert the Mid-Tweeter crossover into a passive one. The speaker will be a hybrid passive-active 3-amped one.
I ran some simulations and I don't think that a phase-accurate system can accomplished with more than 1 le'Cleach crossovers. My initial idea of a phase-accurate system with IIR filters was probably unrealistic. At this point I'm looking into a passive acoustic 2nd order Linkwitz-Riley on the Midwoofer and Tweeter at around 2kHz, then an active 2nd order LR on the Mid and the Woofer array somewhere into the 200-400Hz region. As I mentioned earlier the woofer array will be driven by 2 amplifier channels.
The overall sensitivity should be 85-88dB @ 1m, not bad for OB.
I've been listening to the prototype and getting very excited. The bass remains very even throughout the room, it's quite a contrast from my boxed speaker.
Member
Joined 2009
bass measurements
I have some measurements of the speaker outside.
Here are the unequalized drivers on the baffle.
woofer array = blue
midwoofer = red
tweeter = green
I want to equalize the speaker along the line denoted by -30dB in this graph. The woofer array is at that level around 400Hz and everything below is reduced at rate of 6dB/octave that is characteristic of open baffles. In order to extend down to 50Hz I would need to amplify by 18dB.
While that's doable, I tried it and I did not like the dynamics. I decided to dedicate 2 amplifier channels to the woofer array which will boost their output by 6dB. I'm also experimenting with small side wings which can boost the useful signal by another 3dB.
woofer array naked = red
woofer array with small wings = green
Wings introduce a problem however that is visible in the phase plot above. The top woofer pair reaches it's dipole peak as the bottom pair goes into a dipole dip.
woofer array with small wings combined = blue
woofer array with small wings top pair = red
woofer array with small wings bottom pair = green
I believe to have solved the problem by equalizing the peak on the bottom pair and thus improving the phase tracking. I took measurements of the equalized response at different distances.
equalized woofer array with small wings @1.5m = red
equalized woofer array with small wings @2m = blue
I think it looks very useful. I'm not sure yet what the woofers-to-mid crossover will be yet. I can probably pull off a 2nd order @ 200Hz or a 4th order at 400Hz. Below is a 180 degrees frequency response of the woofer array.
unequalized woofer array on a naked baffle
equalized woofer array with small wings
I have some measurements of the speaker outside.
Here are the unequalized drivers on the baffle.
woofer array = blue
midwoofer = red
tweeter = green
I want to equalize the speaker along the line denoted by -30dB in this graph. The woofer array is at that level around 400Hz and everything below is reduced at rate of 6dB/octave that is characteristic of open baffles. In order to extend down to 50Hz I would need to amplify by 18dB.
While that's doable, I tried it and I did not like the dynamics. I decided to dedicate 2 amplifier channels to the woofer array which will boost their output by 6dB. I'm also experimenting with small side wings which can boost the useful signal by another 3dB.
woofer array naked = red
woofer array with small wings = green
Wings introduce a problem however that is visible in the phase plot above. The top woofer pair reaches it's dipole peak as the bottom pair goes into a dipole dip.
woofer array with small wings combined = blue
woofer array with small wings top pair = red
woofer array with small wings bottom pair = green
I believe to have solved the problem by equalizing the peak on the bottom pair and thus improving the phase tracking. I took measurements of the equalized response at different distances.
equalized woofer array with small wings @1.5m = red
equalized woofer array with small wings @2m = blue
I think it looks very useful. I'm not sure yet what the woofers-to-mid crossover will be yet. I can probably pull off a 2nd order @ 200Hz or a 4th order at 400Hz. Below is a 180 degrees frequency response of the woofer array.
unequalized woofer array on a naked baffle
equalized woofer array with small wings
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Did you equalise the woofer array by applying some bandpass filters? Your last diagram looks like that.
I've learned from SL and MJK that dipole woofers should be EQed by applying a 6 dB low pass at the point where the 6 dB dipole roll-off starts:
For your woofer array on a naked baffle this point would be 60 Hz.
Rudolf
I've learned from SL and MJK that dipole woofers should be EQed by applying a 6 dB low pass at the point where the 6 dB dipole roll-off starts:
For your woofer array on a naked baffle this point would be 60 Hz.
Rudolf
boris81 ... could you explain what's going on with the two different baffle arrangements you have depicted? I noticed the first photo shows the baffle as a solid 1-1/2" ... second photo (with the shortened wings) seems to have a space between the 1-1/2" sandwich ... with an insulation? wrap around the drivers?
I'd listen to the speaker playing music (in room) before settling on a xo point ..(your 200 vs 400hz proposal). While the response might look good on paper ... it might not sound right. This area, to me, seems to be pretty critical for getting vocals to sound natural. ... and there is the room resonance that might show up in the 300hz range. How severe that is will determine how much of a notch a guy needs to implement.
Because of the way my system is developing and the rooms 310hz resonance (pretty strong) , I chose to deal with that notch in the mid/woofer. ... but my bass woofers are separate from the mains ... so the xo is low to begin with. Anyway ... might be something to consider.
I'd listen to the speaker playing music (in room) before settling on a xo point ..(your 200 vs 400hz proposal). While the response might look good on paper ... it might not sound right. This area, to me, seems to be pretty critical for getting vocals to sound natural. ... and there is the room resonance that might show up in the 300hz range. How severe that is will determine how much of a notch a guy needs to implement.
Because of the way my system is developing and the rooms 310hz resonance (pretty strong) , I chose to deal with that notch in the mid/woofer. ... but my bass woofers are separate from the mains ... so the xo is low to begin with. Anyway ... might be something to consider.
Member
Joined 2009
Woofer EQ
I should not have put all those different graphs in one post, I'm getting confused looking at it myself. The 2 directivity plots don't belong together because the top one is of a Naked baffle and the bottom one is of an Equalized Baffle with Small Wings. It's a different baffle.
You are correct, the equalization of the Naked baffle should be done with a 6dB/oct low pass.
On the Baffle with the Small Wings the trouble is that the top woofer pair has it's dipole peak at a different frequency than the bottom pair. I equalized the signal by attenuating the dipole peak on each woofer pair individually.
I should not have put all those different graphs in one post, I'm getting confused looking at it myself. The 2 directivity plots don't belong together because the top one is of a Naked baffle and the bottom one is of an Equalized Baffle with Small Wings. It's a different baffle.
You are correct, the equalization of the Naked baffle should be done with a 6dB/oct low pass.
On the Baffle with the Small Wings the trouble is that the top woofer pair has it's dipole peak at a different frequency than the bottom pair. I equalized the signal by attenuating the dipole peak on each woofer pair individually.
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