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Thread: JBL 4367 first listen

  1. #256
    Junior Member DallasJustice's Avatar
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    Quote Originally Posted by hsosdrum View Post
    *If you're not running the M2 DSP program you don't have M2's. Period. You can tweak that DSP plenty and still have what the designers intended, but you can't eliminate it.
    can you elaborate on this?

  2. #257
    Senior Member Ian Mackenzie's Avatar
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    Allowing 6 dB head room on the woofer the Crown I-T5000 is a smart decision in the studio.

  3. #258
    Senior Member pos's Avatar
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    Quote Originally Posted by DallasJustice View Post
    can you elaborate on this?
    That is what I tried to elaborate on several times in PMs with you:
    Quote Originally Posted by pos
    With the M2 the first part is already done by Harman: you just need to replicate as closely as possible the corrections done within the crown or bss. This has to be done using minimum-phase EQs and filters, and as precisely possible (no subtitute for a real minimum-phase LR here, this is part of the crossover and using any other type of filter will result in bad summation).
    Room correction can be done on top of this, typically for low frequencies (you should not need correction above Schroeder frequency with the M2, or maybe just to adapt the HF response but only gentle Q corrections).
    Quote Originally Posted by pos
    M2 settings are very precise and were done by JBL engineers with tons of measurements (LSR/spionara approach) and tons of listening sessions with all kind of audio professionals.
    Trying to redo the crossover and EQs yourself using in-room measurements will certainly not lead to the same result.
    But that is your call
    There are really only to ways with the M2: either buy a certified hardware (BSS, crown, etc.) or reproduce the correction as closely as possible (within typical driver-to-driver consistency tolerance).
    Once this is achieved you can play with room correction, and even swap drivers and/or replace the HF protection network (using measurement that exactly reproduce the intended response curve in close range for example), but there is not shortcut for the initial speaker EQ/filtering.

  4. #259
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    Quote Originally Posted by hsosdrum View Post
    I have no idea if those main engineers (Alan Devantier and Charles Sprinkle) use DCI amps in their home systems; in fact, I'm fairly confident that neither personally owns M2s.
    Can you elaborate on this?

  5. #260
    Senior Member baldrick's Avatar
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    Quote Originally Posted by sebackman View Post
    The BBS BLU50 will probably be a much better sounding DSP based upon topology and SW.
    If you are comparing BSS BLU50 to SDA4600, why do you think so? Doesn't they both use BSS DSP?

    Quote Originally Posted by hsosdrum View Post
    *If you're not running the M2 DSP program you don't have M2's. Period. You can tweak that DSP plenty and still have what the designers intended, but you can't eliminate it.
    I haven't seen my self but I was pretty sure that Crown DCI N did in fact have the M2 DSP preset and was identical to the DSP in iTech HD!?

    Another intersting thing:

    This dealer in Norway have both 4367 and M2 setup in the same room... M2 powered by SDA4600 and 4367 Powered by ML 585. I haven't listened myself but the owner says that the 4367 sound "way" better than the M2 setup.


  6. #261
    Junior Member DallasJustice's Avatar
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    Quote Originally Posted by pos View Post
    That is what I tried to elaborate on several times in PMs with you:


    There are really only to ways with the M2: either buy a certified hardware (BSS, crown, etc.) or reproduce the correction as closely as possible (within typical driver-to-driver consistency tolerance).
    Once this is achieved you can play with room correction, and even swap drivers and/or replace the HF protection network (using measurement that exactly reproduce the intended response curve in close range for example), but there is not shortcut for the initial speaker EQ/filtering.
    I understand that the factory EQ is based on anechoic measurement. However, EQ can be very effective in room as well.

    I think the argument that Harman’s DSP is perfect or ideal doesn’t hold up. The Harman crossovers are minimum phase. Linear phase crossovers are superior. Linear phase crossovers perform much better in time domain. In addition, Linear phase EQ can correct the phase shift due to passive network.

    Ultimately, a linear phase EQ with proper time domain correction will produce a better step response than using minimum phase filters.

    So it is correct that not using Harman’s DSP will result in something different. The question is whether the something different is better or worse. I can’t say because I’m not planning on using minimum phase filters. From what I’ve heard so far, the M2 is a great speaker that can easily be EQ’d off the shelf using Audiolense or Acourate. The result is excellent.

    One thing to remember about this speaker is that EQ can not change the speaker’s off axis response. It is what it is. It’s one of the best speakers off axis out there; maybe the best. As long as the crossover is clean and set in the area Harman recommends (800hz), there should be no off axis variation with different EQ settings.

    Harman is a great company. They make the best loudspeakers in the world. Of course Harman will always say their crossovers and EQ is perfect and the user must use them. But that’s not the case, IMO. Harman isn’t known for their DSP technology. They are known for their loudspeaker science. For example, Harman frequently relies on third parties to implement their DSP in automobiles. They know others have better skill in that area.

    Just like all audio companies, Harman must sell boxes to make money. They aren’t in the software business. Small low power computer processors don’t have the resources to run FIR linear phase filters with a lot of taps. This is another practical limitation to the DSP-in-a-box solution across all manufacturers of similar boxes. The only examples I can think of where FIR linear phase XO is used in a box, the tap number must be greatly economized; Eg. Kiiaudio loudspeakers and open DRC. Finally, many pro audio customers wouldn’t accept an active speaker with long latency time. For the home user, this isn’t a problem.

    Based on what I’ve measured and heard so far, I’d encourage anyone who’s proficient with software like Acourate or Audiolense that the M2 is a great off the shelf active speaker.

    Dr. Uli Brueggemann wrote a nice paper showing why linear phase crossovers are better than minimum phase.
    http://files.computeraudiophile.com/...WhitePaper.pdf

  7. #262
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    You could also add phase-linearization only as an add-on to the JBL-settings, for the textbook LR36 phase-shift, I guess? If phase linearity is wanted.

    I would not underestimate the value of the measurement-facilities, blind listening-tests, the expertise of Devantier et al, etc. On the other hand, it is easy for you to test your own settings compared to the JBL ones in the end, and just choose what you prefer.

  8. #263
    Senior Member pos's Avatar
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    DallasJustice, I partly agree on some points, but IMHO there are three points you are missing here:

    1- linear-phase might be superior to minimum-phase, but the existing filters already "embedded" in the loudspeaker (compression driver and horn rolloff, which is 4th order, as well as the 1st order passive filter, as well as all the resonances, etc.) *are* minimum-phase, and need to be addressed if one targets a linear-phase acoustical response (ie not only a linear-phase electrical response). Regarding EQ the best way of addressing a (minimum-phase) resonances is to apply a minimum-phase EQ. Regarding filters (and rolloffs), you have two option to address them for a linear-phase acoustical result: you can either linearize (magnitude and phase, not so easy because it has to include a significant portion of the stop band) and then apply an electrical linear-phase filer, or -as suggested by bubbleboy76- target a given minimum-phase acoustical response (what the JBL M2 settings do) and then linearize its phase (very easy with the M2, it is a textbook LR crossover for both magnitude and phase, very nicely done).

    2- with a perfectly constant directivity what you say about on vs off axis response would be true, but it appears there is no such thing as a perfectly constant directivity loudspeakers (yet), even if you only consider the HF part. If you look at the spinorama measurement of the M2 you will see that the M2 settings try to address the "listening window" measurement, which is an averaged measurement over a small horizontal and vertical angle. The on-axis curve is not made as flat as possible, only the listening window curve is. And then they only address resonances that show on all the curves (on axis, listening window, early reflexions, and power response).
    I would suggest you read about this measurement and EQ technique in Floyd Toole "Sound Reproduction" book, it is very informative, and is a large part of what makes the M2 what it is.
    Today DSP technology can make almost any measurement look flat, but what is more important is discerning what should be corrected and what should be left alone, and the spinorama technique is a good tool to for this task.

    3- speaker EQ and room EQ are two different things, and should be done in that order.



  9. #264
    Member Fitero's Avatar
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    Quote Originally Posted by baldrick View Post
    owner says that the 4367 sound "way" better than the M2 setup.
    And therein lies the key point for me; is the perceived difference due to the amplifiers or the speakers themselves?

    There is so much speculation on the net concerning these speakers that it is hard to find the truth.

    I think the only opinions acceptable are the ones generated from actual experience with trial and error.

    In my own experience, the 4367's clearly revealed every cable and component change. It would seem logical to assume that the M2's will do the same. Following that same logic, any difference in performance between Crown's amplifiers and other manufacturer's amplifiers will be similarly revealed.

    I'll start off with one of JBL's SDA amplifiers and progress to an outboard BSS processor feeding signal to other amplifiers to experiment.

    I'm software challenged, unlike member POS who has aided many M2 owners with his knowledge and expertise.

  10. #265
    Member Mitchco's Avatar
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    Modern software DSP for loudspeakers in rooms is a huge topic (I wrote a book on it – see sig) and is also not well understood as there are many misconceptions about it.

    For me, accurate sound reproduction is ensuring the music arriving at my ears, at the listening position, matches as close as possible to what is encoded on the digital (or analog) media I am listening to. Technically, one is matching the acoustical waveforms arriving at ones ears to the waveforms stored on the digital media. Another way to say it mathematically is matching transfer functions.

    While this is relatively easy to do with DAC’s, preamps and amplifiers, decidedly not so with loudspeakers in rooms. Loudspeakers, even most excellent ones like the M2’s, are orders of magnitude less accurate than the electrical gear preceding it. Look at the “speaker” eq required to be applied to the M2 as an example. Also, if you measured a DAC or amplifier, aside from flat frequency response, the phase response is also flat, along with flat group delay. The Audiolense article linked below shows an example of this using a Lynx Hilo AD DA converter.

    But here is the thing about modern software DSP for audio. When a measurement is made at the listening position, the DSP software extracts the minimum phase response and eq is applied to that minimum phase response (i.e. minimum phase eq) to match the listeners preferred steady state frequency response which Olive and Toole have shown here in the Measurement and Calibration of Sound Reproducing Systems: http://www.aes.org/e-lib/browse.cfm?elib=17839 See Figure 14.

    The excess phase that is left over is corrected so that the sound arriving at one’s ears is both flat phase and flat group delay. Just like a DAC or amplifiers response. This includes correcting for standing waves or low frequency room reflections as the Audiolense article linked below clearly demonstrates.

    Linear phase XO enables this scenario as not only does the electrical response sum correctly in both the frequency and time domain, it does so also in the acoustic domain. How so? Typically by using steep-ish crossovers slopes so that effectively the acoustic response of the driver is "convolved" with the electrical (or digital) response of the XO. Ergo the acoustic response is the same as the electrical response and therefore the perfect summing, again the in the frequency and time domain.

    Finally, the drive units are time aligned so that all direct sound frequencies arrive at the same time at the ears at the LP. Note that the M2’s are not time aligned out of the box, and linearizing the phase does not change that.

    The concepts described above are applied in a real life example to biamping a JBL 4722 cinema loudspeaker:
    https://www.computeraudiophile.com/c...kthrough-r682/

    Using different DSP software and more detail here: https://www.computeraudiophile.com/c...n-Walkthrough/ This one shows speaker eq first and then room eq. But in the end, the result is the same as applying so called room correction, as it only matters what is arriving at ones ears.

    Wrt to constant directivity, as shown in detail in my eBook using constant directivity 3-way loudspeakers, and not as good as the M2’s, moving the measurement mic across a 6’x 2’ listening area at the LP, shows virtually the same frequency response across the listening area, along with flat phase and flat group delay and no change to the driver time alignment.

    There is more than one way to great sound at the LP… I do have a set of measures at the LP for the M2 “system” with all the right components as supplied by a member from the AVSFroum. If DallasJustice is up for it, perhaps he can supply me with his REW results and we can overlay the two responses to see which approach is more accurate :-)

    Enjoy the music!

  11. #266
    Senior Member Ian Mackenzie's Avatar
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    That was a very informative post.

    Part 1

    Not wanting to be a fly in the ointment but the term dsp makes a lot of people nervous.

    On the weekend l had 8 experienced listeners over from our local audio club to audition a new phono preamp project we have been working on for the past 6 months.

    Just prior to that l used a CD player as a source to make sure the amp and loudspeaker worked okay.

    We played numerous vinyl disks over the course of the afternoon.

    The feedback I got on the system playing vinyl was surprisingly good.

    I generally l use a digital source (CD player - dac) for convenience.

    The subjective audible difference between the vinyl source and digital source wasn’t subtle it was significant.

    I’m not going to elaborate further because it’s a well beaten road.

    The issue is with all this stuff is what you don’t know you don’t know until the opportunity arises for a reality check.

  12. #267
    Senior Member pos's Avatar
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    Hello Mitch,

    Quote Originally Posted by Mitchco View Post
    Note that the M2’s are not time aligned out of the box, and linearizing the phase does not change that.
    What do you mean exactly?


    Regarding in-room measurements for automatically doing speaker+room correction, I must say I am not a "automated correction" type of guy (and rephase reflects that).
    I have no doubt automated corrections can lead to good results, but the spinorama approach (which relies an automated measurement tools) associated with manual corrections (experienced human brain) and listening session (pairs of hears) is a tough one to beat, and the M2 is the perfect example of that approach.

    I think JBL is embracing DSP concepts the right way: automated measurement process, automated LF room corrections (including multiple subs arrangements), but manual corrections (and listening tests) for the speaker itself because it only has to be done once (independently of the room the speaker will be installed in, if it is well behaved) and has to be done well.
    Manual corrections take more efforts and time, but most of the burden is on the measurement side (very time consuming).
    They have now integrated Trinnov technology in their to synthesis processor, so we will see what happens, but I doubt they plan to use it for crossover duty.

    I am also not a big fan of steep crossovers, but that is another subject

  13. #268
    Senior Member hsosdrum's Avatar
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    Originally Posted by hsosdrum *If you're not running the M2 DSP program you don't have M2's. Period. You can tweak that DSP plenty and still have what the designers intended, but you can't eliminate it.


    Quote Originally Posted by DallasJustice View Post
    can you elaborate on this?
    I think that POS has done a fine job explaining the specifics. I'll only add that the M2 was designed from the outset as a system, with DSP speaker correction as a very important part of that system. If you don't use the Harman-developed M2 DSP (or a perfect copy of it) you're not listening to M2s, the same as if you removed the 2216Nd woofers and replaced them with some other driver. It would not be the speaker that Alan and Charles designed (using drivers designed by Jerry Morro and Alex Voishvilo), it would be some other speaker. You may like it better, but it wouldn't be an M2.

    The same goes for the home-built versions that have been talked about around this forum. If you build your own M2 at home you have to copy every aspect of its physical design if your speaker is to perform identically to a Harman-built M2. For example, the "real" M2 has a stepped baffle panel that puts the vent openings proud of the woofer baffle. I asked Alan about this before he'd even finalized the speaker's physical design. My concern was that the baffle step would create audible diffraction effects that would affect the speaker's performance. He told me that on the contrary, Spinorama measurements had revealed a response anomaly in the midrange, and that adding the baffle step eliminated this anomaly. So the baffle step was retained in the M2's design right through to the production version. It wasn't an aesthetic choice, it was an engineering solution. I've seen photos here of beautiful M2 clones that eliminate this baffle step (I'm sure the step makes the speaker more difficult to manufacture). Beautiful those speakers may be, but M2s they're not.

    FYI, the M2 DSP was not developed solely from measurements (Spinorama and otherwise) taken in anechoic chambers. During development we listened to the prototypes in several different listening rooms (two of which were built according to IEC specs, others of which were not). Sometimes the DSP was even tweaked via computer during the listening test, allowing us to compare different approaches. Developing and finalizing the DSP took months. (The speaker's physical aspects — enclosure and waveguide design, driver selection and integration — were locked-down much earlier in the design process.)

    I personally found the speakers unlistenable with the first couple of iterations of the DSP: Very bright and fatiguing, with an overabundance of midrange detail and clarity and a much-too-forward soundstage. The speaker's performance improved with each subsequent version of the DSP: The brightness was tamed, the soundstage bloomed and the speaker itself began to disappear. Listening to the final production version of the M2 in Harman's large IEC listening room was the most uncanny experience I've ever had listening to reproduced music (I've been in the industry since 1977 and have heard literally thousands of different speakers in hundreds of different rooms). No matter what program material I played on the speakers (I auditioned them for over an hour and was able to use music I recorded in my own studio as well as commercially-available recordings and some of the bit-for-bit digital transfers that JBL engineers used), the M2s simply weren't there. I'd never before heard speakers that imposed so little of themselves on the resulting sound. Each recording sounded completely different — there were virtually no sonic characteristics common on multiple recordings that I could ascribe to the speakers. They were showing me more of what were on those recordings (including the ones I myself had made) than any other speaker I'd ever heard before (or since).

    Alan and Charles had indeed created a tool that would tell people involved in the creation of music recordings exactly what they were putting on their recordings. Being a musician and recording enthusiast I think this is no small achievement. Others here (and elsewhere) are welcome to second-guess the choices made by Alan and Charles (and a host of others like myself, who were peripherally involved in the M2's development), but my money is on the real thing.

  14. #269
    Junior Member DallasJustice's Avatar
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    Quote Originally Posted by hsosdrum View Post
    Originally Posted by hsosdrum *If you're not running the M2 DSP program you don't have M2's. Period. You can tweak that DSP plenty and still have what the designers intended, but you can't eliminate it.




    I think that POS has done a fine job explaining the specifics. I'll only add that the M2 was designed from the outset as a system, with DSP speaker correction as a very important part of that system. If you don't use the Harman-developed M2 DSP (or a perfect copy of it) you're not listening to M2s, the same as if you removed the 2216Nd woofers and replaced them with some other driver. It would not be the speaker that Alan and Charles designed (using drivers designed by Jerry Morro and Alex Voishvilo), it would be some other speaker. You may like it better, but it wouldn't be an M2.

    The same goes for the home-built versions that have been talked about around this forum. If you build your own M2 at home you have to copy every aspect of its physical design if your speaker is to perform identically to a Harman-built M2. For example, the "real" M2 has a stepped baffle panel that puts the vent openings proud of the woofer baffle. I asked Alan about this before he'd even finalized the speaker's physical design. My concern was that the baffle step would create audible diffraction effects that would affect the speaker's performance. He told me that on the contrary, Spinorama measurements had revealed a response anomaly in the midrange, and that adding the baffle step eliminated this anomaly. So the baffle step was retained in the M2's design right through to the production version. It wasn't an aesthetic choice, it was an engineering solution. I've seen photos here of beautiful M2 clones that eliminate this baffle step (I'm sure the step makes the speaker more difficult to manufacture). Beautiful those speakers may be, but M2s they're not.

    FYI, the M2 DSP was not developed solely from measurements (Spinorama and otherwise) taken in anechoic chambers. During development we listened to the prototypes in several different listening rooms (two of which were built according to IEC specs, others of which were not). Sometimes the DSP was even tweaked via computer during the listening test, allowing us to compare different approaches. Developing and finalizing the DSP took months. (The speaker's physical aspects — enclosure and waveguide design, driver selection and integration — were locked-down much earlier in the design process.)

    I personally found the speakers unlistenable with the first couple of iterations of the DSP: Very bright and fatiguing, with an overabundance of midrange detail and clarity and a much-too-forward soundstage. The speaker's performance improved with each subsequent version of the DSP: The brightness was tamed, the soundstage bloomed and the speaker itself began to disappear. Listening to the final production version of the M2 in Harman's large IEC listening room was the most uncanny experience I've ever had listening to reproduced music (I've been in the industry since 1977 and have heard literally thousands of different speakers in hundreds of different rooms). No matter what program material I played on the speakers (I auditioned them for over an hour and was able to use music I recorded in my own studio as well as commercially-available recordings and some of the bit-for-bit digital transfers that JBL engineers used), the M2s simply weren't there. I'd never before heard speakers that imposed so little of themselves on the resulting sound. Each recording sounded completely different — there were virtually no sonic characteristics common on multiple recordings that I could ascribe to the speakers. They were showing me more of what were on those recordings (including the ones I myself had made) than any other speaker I'd ever heard before (or since).

    Alan and Charles had indeed created a tool that would tell people involved in the creation of music recordings exactly what they were putting on their recordings. Being a musician and recording enthusiast I think this is no small achievement. Others here (and elsewhere) are welcome to second-guess the choices made by Alan and Charles (and a host of others like myself, who were peripherally involved in the M2's development), but my money is on the real thing.
    Thank you.

    I know that the M2 can be used with 3rd party DSP as well as amps. I know they can sound better than any passive speaker I’ve heard. I don’t know what you’ve experienced but I appreciate you sharing your background.

    I can only encourage others who don’t wish to use the Harman DSP that they don’t need to and can still produce state of the art sound reproduction with the M2. I know there are some folks who are interested in the M2 but don’t like how Harman deployed the DSP. I first heard the M2 at a Harman demo at CEDIA a few years ago. At the time I wanted to buy them but I balked due to the DSP-in-box I was told MUST be used with the M2. I’ve owned some very fine loudspeakers with state of the art off axis measurements. I can say the M2 I’m hearing is a much better speaker than those others. So I’m glad I took a chance and set them up the way I did. I’ll post some REW measurements and additional details concerning setup later this week.

  15. #270
    Senior Member Ian Mackenzie's Avatar
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    Quote Originally Posted by Fitero View Post
    And therein lies the key point for me; is the perceived difference due to the amplifiers or the speakers themselves?

    There is so much speculation on the net concerning these speakers that it is hard to find the truth.

    I think the only opinions acceptable are the ones generated from actual experience with trial and error.

    In my own experience, the 4367's clearly revealed every cable and component change. It would seem logical to assume that the M2's will do the same. Following that same logic, any difference in performance between Crown's amplifiers and other manufacturer's amplifiers will be similarly revealed.



    I'm software challenged, unlike member POS who has aided many M2 owners with his knowledge and expertise.
    Agree with the rational except you did not mention the M2 had the inclusion of dsp. The 4367 does not.

    What l do draw from your comments is the 4367 must have a level of transparency to reveal differences of associated equipment. With rational thinking why do you think this is?

    I have read reviews with similar feedback describing the 4367 delineation of music instruments.

    This actually brings the discussion back on topic to the 4367.

    Applying my own rational thinking sometimes less is more in terms of delivering the absolute purity of the original signal.

    This post by Mr Widget sums it up.
    http://www.audioheritage.org/vbullet...l=1#post381833

    http://www.audioheritage.org/vbullet...8&d=1450691243

    https://www.google.com.au/amp/s/home...er-review/amp/

    Elsewhere the designer describes his battle to obtain the musicality of analogue from dsp and where he ended up.

    Why think about f’ing with a very good loudspeaker before you even buy it?

    The next step is the S9900 or the D67000. Biamp of those systems is justified but done with care.

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