Mine too. And a couple popped up I didn't see before, so I'm now putting up the same things John already posted.Couple of my posts are missing... I'm being silenced!
So, when does your SDP-75 come and how many channels for you?
Official JBL Synthesis / Pro / Revel Home Theater Thread
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So, when does your SDP-75 come and how many channels for you?
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Lol, nothing happening anytime soon. If anything, I'm actually thinking less SDP and more the new 16 channel Trinnov, with remapping. I have the amps/tuning files already (for the M2's and 708 at least, and hopefully SCL's in the near future). The target curve is mildly interesting, but I'd end up making my own anyway (as thats largely salt-to-taste). No SFM with he plain vanilla Trinnov, but I could assign separate channels for each sub for trims and delays and then optimize the summed result.
There is a monumental difference in price - 15k vs 23k list, so its tough to argue PreEQ (versus Crown tuning files), SFM, JBL target curve, and expansion beyond 16 channels is worth the difference (because I'll never go past 16 channels)
Nonetheless, still very early in the thinking...
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Hopefully will ship to me on Friday. Not sure which unit - whichever Harman can spare! Asked for the 24 or 32.
I'm bummed that the remapping isn't an option, but it's hard for me to not trust Harman. The Altitude 16 ended up not being as great of a deal as I had hoped. It's a lot of money on a bologna budget for me. By the time I save, I'll be able to make the choice. It'll be a while.
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The new Altitude 16 is not shipping for a while, correct? So why assessed as not a great deal?
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I can't understand why anyone would hide your comments...
Couple of my posts are missing... I'm being silenced!
Cost has been quoted in the Altitude post. I would just buy the Altitude 32-8-16 for the cost. Sure it's a little more, but at least you have the peace of mind knowing it isn't maxed if you ever do decide to change. It does include the codecs, which I don't understand why they all don't. What good is a 32 channel processor without atmos?
I think it releases in a few months or less.
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Little more? I thought the new Altitude16 will be $15k or less. The upgradeable 16 is about $23k. Unless I misread something...
I'm presuming for a moment that the forthcoming $15k Altitude16 is all in - 3D codecs, mic/k, remapping and all other software features - but limited I/O per the back panel photos and not channel upgradeable....
This is inevitable if higher end companies want to survive. There’re only so many customers that will be seeking $20-$35,000 processors. They need to fish downstream a little bit more. I’m not saying they need to be in the $3-8000 range, but something in the $12-15 range opens up a whole new market to them.
I was hoping in in the 12 range. I guess the next step up was 22k, so that is a good jump from 15k, but, like I said, I'd hate to spend that and be stuck, so I would probably keep going, just in case. It's back to square one if you decide you want even one more channel.
You know floor speakers are next! jk
I've got a while, who knows what direction I'm going. But, I think I saw the mic was extra on the 16 as well. Atmos etc. included.
You know floor speakers are next! jk
I've got a while, who knows what direction I'm going. But, I think I saw the mic was extra on the 16 as well. Atmos etc. included.
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Personally, 15K to 23K is quite significant. And for me, I’m just not going past 16 channels. I don’t have room for it, and I’m sure my next space down the road would be the same or smaller, not larger.
But yes let’s see what CEDIA brings...
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Our SDP75 is here already, but have not calibrated it - still installing the SCL-3s and 4s. There are some default JBL profiles loaded in but haven't seen any specific target curve yet.
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I apologize for taking so long to reply to this.
Also, a minor detail I notice is that the attenuation in the 5-10 kHz region looks to be a bit more than should be expected for direct sound distance roll-off unless the rooms all had very low relative humidity. I wouldn't be surprised if there was just enough reflected energy in these rooms to make a very slight contribution to the magnitude-smoothed measured responses (MSMRs) that could account for the extra slope there.
The differences at high frequencies would probably be irrelevant if in-room EQ of high frequencies weren't necessary. However, in cinemas where sound absorbing screen material is present, in-room EQ of high frequencies *is* necessary unless the screen material has been tested separately. EQing high frequencies so that the MSMRs to the anticipated curve will get things close, but I think an approach that relies primarily on first-arrival sound and possibly some later arriving sound to look for narrower resonances induced by the screen would lead to more consistency.
Dealing with distance effects is trickier, especially in rooms with multiple rows of seats in which some seats may be much closer to the speakers than others. IMO, the apparent depth of most sounds should rarely reach beyond the screen or else listeners in the front rows will be assaulted with unnaturally forward sounds.
For lower frequencies, the deviations are pretty big, IMO. Yes, things look better in Figure 19B than 19A, where the difference is primarily due to the lack of influence of the seats, but the deviations without the seats are still several dB in some cases. I don't see any justification in Figure 19B for EQing frequencies below 1000 Hz, 500 Hz, or whatever in MSMRs in every room to a common target.
That doesn't mean that EQ of low frequencies cannot provide benefits. Below some transition frequency, the wavelengths become long enough that the floor and boundaries near the speaker begin to alter the first arrival sound at various locations as well as the overall power response. This will occur well above the actual modal region.
In high quality cinemas, the front speakers are likely to be mounted flush to the wall, and this requires EQ to reverse the baffle step compensation that makes the speaker flat in an anechoic chamber. That compensation is fairly straightforward, especially if the anechoic chamber can be modified to allow for half-space measurements.
However, in home theaters and possibly some cinemas, placements very near but not flush with the front wall are common. Side walls may also be close enough to alter first arrival sound and overall power response in a home theater and possibly some cinemas in which the screen extends all the way to the walls. The optimal corrections for these things are not as straightforward as for flush-mounting, and it is necessary to distinguish these effects from other early reflections and room reverb build-up. Adjusting MSRMs to a common target cannot achieve this aim.
It remains an open question, however, what listening window is best to average within for this purpose. I am using a window that's approximately +/- 30 degree horizontal but includes off-axis data with different weighting, approximately based on the relative power response contribution in the higher frequencies. It sounds just about right. If I expand the window or assign more weight to off-axis measurements, then the low frequency parts of the measurement appear higher and will tend to be EQed down, leading to a thinner sound. If I EQ flat only on-axis, the sound is too full. Curiously, these trends are consistent across many different seats, and I hardly notice seat-to-seat variation.
For comparison, the M2's voicing appears to be optimized using equal weight throughout the listening window.
That measurement microphones are "deaf" to arrival time is not true at all if an impulse response measurement is being performed. Instead, it is the analysis method that is "deaf" to time of arrival. That critical information is lost as soon as the human operator engages "smoothing" of the measurements, which leads to MSRMs. The raw frequency response actually contains an enormous amount of information about the source and the acoustics within it. One just needs to use the right analytical techniques to extract that information, just as the ear and brain do the same.
This point is at the root of why I keep on hammering on the fact that MSRM targets are unproductive for room EQ. They are deaf to the differences that are needed to understand what needs to be corrected and what doesn't need to be correct. Sure, at high frequencies MSRMs don't differ much from first arrival response, but they can still differ enough to matter if EQ is being applied there for some reason.
And for low frequencies where interactions with room boundaries can literally change the sound of the speaker relative to anechoic conditions, but not so low that room modes dominate the first arrival sound, there is no way to distinguish the sound of the speaker from the reflections that don't alter the sound of the speaker in MSRMs. The anechoic chamber measurements are of limited use because some of the boundary interactions change the sound of the speaker in ways that no listener can adequately distinguish. It is as if these boundaries become part of the speaker itself.
What's needed is a way of extracting the first arrival sound as the ear/brain perceives it. This can be done in principle using impulse response data. The hard part, of course, is choosing a method is psycho-acoustically accurate. A time window is a crude method of separating parts of the impulse response in time, but it provides a mathematical building block from which a wide variety of more advanced methods can be constructed.
Even with perfectly anechoic flat speakers, the circle of confusion undoubtedly rules the subwoofer frequencies. Unlike the rest of the spectrum, which I can calibrate systematically apart from some very minor fine tuning, I find I need to adjust the shape of the sub frequencies by ear.
I am surprised by two things. First, the bass subjectively varies from seat-to-seat and in different parts of the room much less than my measurements suggest it should. Second, the bass balance subjectively varies much less between tracks than I'd expect.
By bass balance, I don't mean the levels of bass instruments vs. other instruments in the mix. Those differences are indeed pretty extreme to me. I'm not sure how often this is a matter of artistic preference and how much it is due to misjudgment during the mixing. Instead, what I mean is the balance of fundamental frequency to various overtones in the bass instruments themselves. With my subs configured as they are now, I find a very narrow range, within 0.25 dB or so, of subwoofer gain in which the bass instruments themselves sound balanced. What I do sometimes have to tweak is the center frequency of the shelf, but this is a relatively minor thing.
At the same time, ultra high frequencies (UHF) in the top octave have a lot of influence on bass perception, in my experience. Too much UHF can mask this bass content, even including some of the the tactile aspects. The tone control I use the most by far is the adjustable UHF roll-off, without which, the bass goes completely missing in a lot of tracks.
You are correct. I definitely look forward to reading the book when it's out.
Reviewing Figure 19B in your 2015 JAES paper, I don't see anything that really surprises me or really changes my opinion. Yes, the curves are very close above 1 kHz (actually 2 kHz, really). I'm not sure I'd say they are "right on top of one another", being that I believe calibration to within 0.25 dB is ideal for high frequencies. The differences in distance roll-off definitely complicates the assessment quite a bit.
Also, a minor detail I notice is that the attenuation in the 5-10 kHz region looks to be a bit more than should be expected for direct sound distance roll-off unless the rooms all had very low relative humidity. I wouldn't be surprised if there was just enough reflected energy in these rooms to make a very slight contribution to the magnitude-smoothed measured responses (MSMRs) that could account for the extra slope there.
The differences at high frequencies would probably be irrelevant if in-room EQ of high frequencies weren't necessary. However, in cinemas where sound absorbing screen material is present, in-room EQ of high frequencies *is* necessary unless the screen material has been tested separately. EQing high frequencies so that the MSMRs to the anticipated curve will get things close, but I think an approach that relies primarily on first-arrival sound and possibly some later arriving sound to look for narrower resonances induced by the screen would lead to more consistency.
Dealing with distance effects is trickier, especially in rooms with multiple rows of seats in which some seats may be much closer to the speakers than others. IMO, the apparent depth of most sounds should rarely reach beyond the screen or else listeners in the front rows will be assaulted with unnaturally forward sounds.
For lower frequencies, the deviations are pretty big, IMO. Yes, things look better in Figure 19B than 19A, where the difference is primarily due to the lack of influence of the seats, but the deviations without the seats are still several dB in some cases. I don't see any justification in Figure 19B for EQing frequencies below 1000 Hz, 500 Hz, or whatever in MSMRs in every room to a common target.
That doesn't mean that EQ of low frequencies cannot provide benefits. Below some transition frequency, the wavelengths become long enough that the floor and boundaries near the speaker begin to alter the first arrival sound at various locations as well as the overall power response. This will occur well above the actual modal region.
In high quality cinemas, the front speakers are likely to be mounted flush to the wall, and this requires EQ to reverse the baffle step compensation that makes the speaker flat in an anechoic chamber. That compensation is fairly straightforward, especially if the anechoic chamber can be modified to allow for half-space measurements.
However, in home theaters and possibly some cinemas, placements very near but not flush with the front wall are common. Side walls may also be close enough to alter first arrival sound and overall power response in a home theater and possibly some cinemas in which the screen extends all the way to the walls. The optimal corrections for these things are not as straightforward as for flush-mounting, and it is necessary to distinguish these effects from other early reflections and room reverb build-up. Adjusting MSRMs to a common target cannot achieve this aim.
I agree with these considerations, and I would expand on them to note that early reflected sound does influence balance perception. In the process of voicing my speakers, I discovered that optimizing for flat first arrival sound averaged across a listening window provides superior sound to *flatter* first arrival sound at a single listening position, even at that listening position. In a way, this is kind of an elaboration on your secondary consideration of differentiating resonances from interference effects. But I have found this to be true even with regard to signal shaping at a very coarse scale.
It remains an open question, however, what listening window is best to average within for this purpose. I am using a window that's approximately +/- 30 degree horizontal but includes off-axis data with different weighting, approximately based on the relative power response contribution in the higher frequencies. It sounds just about right. If I expand the window or assign more weight to off-axis measurements, then the low frequency parts of the measurement appear higher and will tend to be EQed down, leading to a thinner sound. If I EQ flat only on-axis, the sound is too full. Curiously, these trends are consistent across many different seats, and I hardly notice seat-to-seat variation.
For comparison, the M2's voicing appears to be optimized using equal weight throughout the listening window.
A single measurement microphone at a single location is indeed "deaf" to incident angle. If that's a problem, it can be worked around using various tricks, but I don't believe it's that important for doing in-room EQ optimization unless one is working with multiple independent sound sources above the frequency at which sound can be localized. That's something I plan to study in the future.
That measurement microphones are "deaf" to arrival time is not true at all if an impulse response measurement is being performed. Instead, it is the analysis method that is "deaf" to time of arrival. That critical information is lost as soon as the human operator engages "smoothing" of the measurements, which leads to MSRMs. The raw frequency response actually contains an enormous amount of information about the source and the acoustics within it. One just needs to use the right analytical techniques to extract that information, just as the ear and brain do the same.
This point is at the root of why I keep on hammering on the fact that MSRM targets are unproductive for room EQ. They are deaf to the differences that are needed to understand what needs to be corrected and what doesn't need to be correct. Sure, at high frequencies MSRMs don't differ much from first arrival response, but they can still differ enough to matter if EQ is being applied there for some reason.
And for low frequencies where interactions with room boundaries can literally change the sound of the speaker relative to anechoic conditions, but not so low that room modes dominate the first arrival sound, there is no way to distinguish the sound of the speaker from the reflections that don't alter the sound of the speaker in MSRMs. The anechoic chamber measurements are of limited use because some of the boundary interactions change the sound of the speaker in ways that no listener can adequately distinguish. It is as if these boundaries become part of the speaker itself.
What's needed is a way of extracting the first arrival sound as the ear/brain perceives it. This can be done in principle using impulse response data. The hard part, of course, is choosing a method is psycho-acoustically accurate. A time window is a crude method of separating parts of the impulse response in time, but it provides a mathematical building block from which a wide variety of more advanced methods can be constructed.
Bass: the final frontier.
Even with perfectly anechoic flat speakers, the circle of confusion undoubtedly rules the subwoofer frequencies. Unlike the rest of the spectrum, which I can calibrate systematically apart from some very minor fine tuning, I find I need to adjust the shape of the sub frequencies by ear.
I am surprised by two things. First, the bass subjectively varies from seat-to-seat and in different parts of the room much less than my measurements suggest it should. Second, the bass balance subjectively varies much less between tracks than I'd expect.
By bass balance, I don't mean the levels of bass instruments vs. other instruments in the mix. Those differences are indeed pretty extreme to me. I'm not sure how often this is a matter of artistic preference and how much it is due to misjudgment during the mixing. Instead, what I mean is the balance of fundamental frequency to various overtones in the bass instruments themselves. With my subs configured as they are now, I find a very narrow range, within 0.25 dB or so, of subwoofer gain in which the bass instruments themselves sound balanced. What I do sometimes have to tweak is the center frequency of the shelf, but this is a relatively minor thing.
At the same time, ultra high frequencies (UHF) in the top octave have a lot of influence on bass perception, in my experience. Too much UHF can mask this bass content, even including some of the the tactile aspects. The tone control I use the most by far is the adjustable UHF roll-off, without which, the bass goes completely missing in a lot of tracks.
You're very welcome. And thank you likewise, both for your responses and your participation on this forum. I never imagined I would have the privilege of corresponding with a preeminent expert in this field.
Looks wise, the datasat has it beat. I'm curious to see how it sounds though.Our SDP75 is here already, but have not calibrated it - still installing the SCL-3s and 4s. There are some default JBL profiles loaded in but haven't seen any specific target curve yet.
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When you hook it up and go through the speaker drop downs, are subs listed in the drop downs?
For me, the datasat has the look of a big bowtie with a little screen in the middle
Since the goal of SFM is to make the bass response as similar as possible across all seats, there has been some research into how best to prepare for SFM (i.e., how best to arrange your subs to get the bass response as similar as possible across all seats, in order to give SFM a head start rather than cripple it).
https://www.harman.com/sites/default/files/multsubs_0.pdf
Best results were four subs in four corners or four subs at the midpoints of the four walls or two subs at the midpoint of opposite walls. No nearfield locations were used, will almost all subwoofer locations at the boundary. There's nothing in the SFM algorithm that prevents it from being used with nearfield subs, but it hasn't been tested, so there's no definitive answer to your question.
Thanks for this great study Sanjay, right up my alley.
I'm interested in a room without a rear wall.
However, two is almost as good as four, and with better LF response than 4! was an eyeopener.
Many threads go with 6 or more, but this likely depends on room size/volume.
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Personally speaking, I liked the 5000 subwoofer results.
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Just to answer the other part of your question: SFM2 is coming to the SDP75. It's not in there yet. Right now the only way to get SFM is in the JBL SDEC, and that's "original recipe" SFM.
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What's been described to me is a general situation where 2 subs will help smooth out seat to seat response either front to back in a room or side to side, where four will smooth out response front to back and side to side. There are variables, but that's the general idea I have taken away.
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