Does anyone else think compression drivers sound bad?

It's hard enough getting it correct in only one iteration. I don't believe you can simply turn a speaker a little towards a wall and say yes, I like more reflections. Other issues creep in.Have you confirmed the 2k dip, what causes it? Can you measure it?

Toole covers it in Chapter 7.1.1

The stereo phantom center will suffer from a significant dip in the
spectrum around 2 kHz because the sound from both loudspeakers reaches
both ears at different times. This interaural crosstalk cancellation is the first
dip in an acoustical interference comb filter as explained in Figure 7.2.

Figure 7.2 Anechoic frequency-response measurements (c) made at oneear of a KEMAR mannequin for sounds arriving from a real center
loudspeaker, shown in (a), and from a stereo pair, shown in (b). The curves
contain the axial frequency response of the loudspeaker used in the test, and
the head-related transfer functions (HRTFs) for the relevant incident angles
for that particular anthropometric mannequin. The important information,
therefore, is in the difference between the curves that, around 2 kHz, is
substantial. The smoothed difference is shown in (d). Nothing is shown
above about 5 kHz because it is difficult to separate the effects of this
specific acoustical interference from those of other acoustical effects. The
dashed curve is the first interference dip (the first “tooth” of the comb
filter) calculated for an interaural delay of 0.27 ms (appropriate for a
loudspeaker at 30° left or right of center) and for an attenuation of the
delayed sound of 6 dB. (e) is the same kind of measurement done in a
normally reflective room, showing that early reflections within the room
reduce the depth of the interference dip.
Data from Shirley et al. (2007).

Figure 7.2d shows the difference between the curves, revealing the result
of acoustical interference. This can be confirmed by a simple calculation.
The time differential between the ears for a sound source at 30° away from
the frontal axis is about 0.27 ms for an average head. A destructive
acoustical interference will occur at the frequency at which this is one-half
of a period: 1.85 kHz. It won’t be a perfect cancellation, because of a tiny
propagation loss, and a significant diffraction effect. The wavelength is just
over 7 inches (178 mm) that, because it is similar in dimension to the head,
will experience a substantial head-shadowing effect at the ear opposite to
the sound source.

But I'm not worried about contending with the dip or improving a phantom center image. I'm thinking in terms of how the human brain uses reflections to process audio. Since we can't localize in the 1,000Hz to 3,000Hz range maybe it's beneficial to increase reflections in that spectrum.
 

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No, no serious mixing or mastering engineer would claim to mix on headphones
- I've heard this argument before, it holds no weight. There are famous engineers who have mastered albums using headphones....I don't care to try and find the names to prove the point........I have waves nx and a pair of hd600, I have used them myself to do the same.
 
Headphones, I said that jokingly in response to the comment made about not being able to hear all the fine details with the supposed crappy systems that the industry has let creep into acceptance, Low quality, low inefficient stuff like this -->
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How ever good horns are headphones aren't going to struggle keeping up with the "micro details" or "hyper resolution" that we are claiming horns are responsible for. Headphones have no room issues, no multi driver summing, no crossovers, etc etc...technology and mathematics has allowed us to emulate real acoustical spaces as well as create rooms that physics would make near impossible. So even if those crappy speakers above....masked the stuff that was being missed in the studio, as claimed in the post I was responding to...they should be able to hear it on headphones(they have headphones at the studio......)...So the passionate post about the industry has truth in it, but lets try keep the picture being painted as accurate as possible is the point....the blame was being pointed at the sound engineers, when from what I can see they are victims like the rest of us lol.

I was apart of a thread where a person actually purchased a set of these lol, he had a pair of Focals(2.5ways) already but they didn't have "it"....I told him to save his money and just add stereo subs and active system to cross it over and eq etc etc....he laughed at me...but apparently, SQ is directly tied to sensitivity and there is no other way.....

There is another part of the picture that is not being discussed, apart of sound engineering can be tied to polar response, in the sense that the signal has a need to be monitored within a polar that represents the consumer.... my 4" two way serves for that purpose(for me)....the average consumer isn't likely listening on a full horn system....so

I'm building a horn system so stop yelling at me lol! We've had this type of discussion before...the blame isn't the recording industry. Has something to do with the consumer I believe? Just like the loudness war....its money driven.
 

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I'm thinking in terms of how the human brain uses reflections to process audio.
I can't say that I find it a problem, I don't generally set 2kHz louder. For one thing the measurements don't take into account how well we can hear this. Can we detect and process that the secondary part of what is at each ear matches the primary part at the other ear, and account for this... and does the pinna detect direction and maintain this separation to somehow reduce the cancellation. Then there's the way that a delayed copy of a wave will cancel only during the overlap with the other component.

I'd say that extra reflected energy may bring it up a bit, but at what cost. It certainly isn't going to be a correct replacement. You need to go the other way, unintuitive as it is.

Make it more narrow in a band. Turn it up and you get more direct energy to compensate, and the room power is also restored.
 
Been there, done that.
Headphones need a special miking to produce truthfull sounds. Otherwise, the so called "In head localisation" takes place. This is called "artificialheadstereophonic recording", or, in germany, "Kunstkopfstereophonie". Those recordings are especially made for headphone listening.
You can't interchange them with recordings done for loudspeaker listening, it won't work successfull.

With recordings for speakers, the listener always will always feel like he is in a false headroom-space located in the situation of the recording. That artificial headroom has to be added electronically or with a special set of headphones (Stax had those headphones some 30 years ago, but they had the electronic added versions, too).
At this time, I have heard only with headphones from Stax and they were the best and most expensive at that time. Its true, those electrostatics reveal every detail and they first wow the listener with their clarity, sound volume and absolute ease of reproduction. I never auditioned this with ordinary cone speakers, but with my Martin Logan CLS electrostats.
But those CLS prooved the electrostatic principle to have other problems when used in speaker panels and not being able to fully satisfy.
 
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Sounds great compared to what? My tapedecks sounded great, too.

R2R, even with low speed, has blown them away.
So always when making a subjective statement, compare it with something better to have a real relationship, a benchmark.
Otherwise, I could say "My portable radio sounds quite good, too. But what does that mean, compared to any serious gear?

Compared to FLACs, same song.

On various headphones & speakers over time....including these still owned.

Sony MDR 7506
Stax Lambda Pro with SRM-1/Mk2 solid state drive unit
Stax Lambda Pro Signature with SRM-T1 triode drive unit
Martin Logan CLS
Acoustat X with servo amp
Nearly a dozen high quality CD/horn combos in DIY speaker designs, using both commercial horns and DIY synergy horns.
A few systems have been completely horn loaded, top to bottom.

Most often on any of those the compared difference can be large,
but sometimes not at all...heck, sometimes the dang MP3 sounds better.

And I don't mean to turn this into another of the inane threads about source material resolution, or quality.

I just took exception to your statement that a hi rez horn system makes all MP3s sound like crap.
I also take exception to the idea that it takes a CD/horn system to achieve any kind of remarkable high rez.

It's all about horses-for-courses imo.

I've never heard a speaker system deliver the resolution of any of my headphones, lowly Sony's included. (btw, renowned John Dunlavy used to say those are the phones he used to try to match his speakers to)

The Acoustat-X has great bass and dynamics for an electrotstat. (Much better than CLS)
But it is miles below any of my CD/horn DIY builds in term of dynamics and bass impact.

I've never quite gotten any of my CD/horn builds to quite match the resolution of the electrostat speakers.
But they are close enough; the overall sound impresses is much more engaging for me. (Synergies appear to be closing in on the electrostats
resolution.)

Anyway, horses-for-courses, speakers for rooms, speakers for music tastes / sonic preferences. No such thing as a do all superior system imho, or even a do-all source format. ;)
 
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Hi Mark100,


What is the cut-off of your Unity horn please ? Is it really the compression drivesr that handle or help this feeling of bass impact according to you ? Or simply because you use more higher efficienty mid and bass cones drivers below a 1 k he CD ?


I know tweeter are needed for the feeling of sna and bass transcients, but here the square wave of an esl is better than a compression driver ? Not sure I understand why a cd performs better than say an ESL hybrid system...
 
- I've heard this argument before, it holds no weight. There are famous engineers who have mastered albums using headphones....I don't care to try and find the names to prove the point........I have waves nx and a pair of hd600, I have used them myself to do the same.

I actually re-mixed low budget pop song with HD580 20 years ago, and the mastering engineer told me my master is over compressed. I thought it's bullsxxt, but later I confirmed he was right when I heard published 12" on speakers.

It's hard to hear the correct compression level with headphones, but today's pop music is so much more compressed than 20 years ago, plus, today's digital metering is extremely helpful to determine the compression level, so I guess it would not be a serious issue any more for mixing. Mastering? I think it's still risky doing it. We human probably can't hear compression over headphones (and car speakers) as much as as home speakers, and it would be one of the reasons for the loudness war. It's probably because the sound from headphones and car speakers is somewhat acoustically compressed due to tiny space, if I'm correct...
 
Let’s assume the artist primarily listens on headphones and the intended audience does as well. Why *shouldnt* such material be mixed/mastered using that playback format?
So true. It'll be up to the consumer to demand better quality as the industry transition to digital becomes more complete. I even find myself listening more to a streaming service nowadays. It's just more convenient. All in all the feeds sound fine ... most at least. The opportunity to listen to things you'd not have been exposed to in earlier times is a big plus in my book. If this is the modern tradeoff, I think it's a win win, for the artists and listener.
 
hmm, I don't know anything about crossfeed. I'm not really serious about headphones, because their weight make my neck hurt after 20 min. LOL.

Anyway, I think another possible reason why we can hear more detail over headphone is, the sound of headphone is compressed (low level signal is louder at the same RMS level), if my hypotheses about headphone compression is correct.
 
Hi diyiggy,

My synergy/unity's are all designed to go down to 100Hz, albeit using a steep hpf. So not much contribution below 100Hz.
I use coax CD's to be able to reach down to 450-650Hz. This allows using one type driver to go from 100Hz to CD xover.
I've had success with builds using pairs of 12s", 10s" and 8s" along with the CD.
So I'm not really sure, given those volume displacement differences of the cones, how much efficiency matters.
I think efficiency and power handling are just part of the "desired SPL specs".

I don't believe the square wave of an esl is better than a CD. I think a CD system can be the same for all practical purposes.

A CD system can have more SPL and more dynamic capability than any esl i know of, by far, even in the range esl's have strength....a range that doesn't include bass.
And it's very hard to integrate a sub with an esl.
My own belief is that with higher SPL, hybrid bass units that match the esl's levels can't help but resonate an esl's panels given the panels large size. The panels become a big passive resonator so to speak. (I'm pretty sure this is true from measuring how subs effect other non-driven subs in the same room.)

Ime, bass impact, transient impact, is simply about full range SPL without loss of headroom anywhere in the spectrum....of course properly tuned.