JBL Master Reference Monitor

[cooky1257]

It's a fair point. The way i see it we have two options.

1: Just assume that what is on the record is the "answer". That means that our system must bring that into the room as uncolored as possible. Then absolute transparency is the goal.

2: Realize that we can't know the exact conditions in which the mastering was done. We can also assume that the monitors used were not perfect. .

Imo both methods have their merits. Personally I am in camp number 1 for the time being.

To a point I agree but..I wasn't suggesting tuning to taste rather that a quality level in these totl JBL systems is reached beyond which 'absolute transparency' can only ever be a subjective judgement call when the DSP units are included in the equation.

[Valentin]

Extremely fast and dynamic. Attack and decay are stunning. There is virtually no part of the bandwidth that isn't there. Headroom is insane. These are the kind of loudspeakers you would listen to for a typical ten to twelve hour day. It really isn't about the loudspeaker anymore. It's about everything else. The weakest link in the system concept is definitely the DSP engine for some people, if only conceptually or theoretically. These are monitors in the truest sense, GIGO, they will reveal the myriad bullshit in your media sources so beware

nice description of performance

[timc]

To a point I agree but..I wasn't suggesting tuning to taste rather that a quality level in these totl JBL systems is reached beyond which 'absolute transparency' can only ever be a subjective judgement call when the DSP units are included in the equation.

Ok. Then i don't agree. Absolute transparency for me is when the measured impulse response of the input equals the output. In this case it would be the impulse response from the input of the electronics, to the output of the speaker in an anechoic environment. You can't really DSP your way out of your acoustic environment completely.

[cooky1257]

Ok. Then i don't agree. Absolute transparency for me is when the measured impulse response of the input equals the output. In this case it would be the impulse response from the input of the electronics, to the output of the speaker in an anechoic environment.

Good luck with that.

[timc]

Good luck with that.

I didn't say it was possible. But it is the benchmark.

[pos]

I might be misinterpreting what you mean Pos. According to Rainer, you uppload your measurement to the HD2 with the free software. Then define your target response, both frequency and phase, and then it automatically does the necessary adjustments.

So it would be an automated correction base on some form of impulse inversion...
This is a powerful technique, but quite dangerous and which require a lot of care (see DRC-FIR documentation for examples...) and also place a lot of pressure on the measurement itself: you dont' want to correct diffraction effects that only occur at one point in space for example...

I prefer manual corrections (hence rePhase) and be able to choose what should be corrected or not based on one or several measurement and the knowledge of the limit inherent to them (gating, etc.).

I would be interested to know if the user can upload its FIR coefficient using this software.

[timc]

So it would be an automated correction base on some form of impulse inversion...
This is a powerful technique, but quite dangerous and which require a lot of care (see DRC-FIR documentation for examples...) and also place a lot of pressure on the measurement itself: you dont' want to correct diffraction effects that only occur at one point in space for example...

I prefer manual corrections (hence rePhase) and be able to choose what should be corrected or not based on one or several measurement and the knowledge of the limit inherent to them (gating, etc.).

I would be interested to know if the user can upload its FIR coefficient using this software.

I'll ask him for you.

[pos]

thank you timc

more10 is also working on a nice DSP solution (including downsampling).
He will probably open a thread when his module is finished.

[timc]

About rePhase. You are able to put the files generated from that into a convolver plugin right? So you could use a high-end sound card like the Lynx Aurora, and do all the filtering and crossover with multichannel output?

[pos]

yes some users are doing this for example using foobar+convolver for stereo phase linearization and amplitude EQ, or JRiver for multichannel active filtering (with linear phase acoustic filters).
Personally I prefer hardware solutions, for ease of use, power consumption, and reliability issues.

For stereo corrections the openDRC is all what anyone would ever need, even for serious corrections down low (with IIR filters complementing FIR when resolution starts to loosen), and is very affordable.
For multichannel active filtering the market is still open: the miniSHARC module (which is the module used inside the openDRC) is able to handle 8 channels, but the number of taps is limited (max 2048 taps per channel), and there are no ready made product including multichannel DAC just yet...
I think moer10's module could fill this gap when it is ready.

In the meantime, and to go back on topic, 2 openDRC units could do the filtering of the M2 without problem (one per speaker, with a stereo amp on each side), following the parameters of the crown preset, and even do a phase-linear version.
That is $300 per unit with digital in/out, to which you will have to add a good pair of stereo DAC, and a pair of amps...

[4313B]

G.T. is meeting this morning with ALLDSP via Skype for use with his DD67000, K2-S9900 and 880 Array. It will be interesting to find out how that goes. ALLDSP might be of interest to folks as another option.

http://www.alldsp.com/frame.php?cont...etail&ID=1800i

Another potential option from Mr. Widget:

http://lake.labgruppen.com/

[pos]

The AllDSP PLP series looks like serious processors on paper: 64bits internal precision (good for IIR processing), 118/114dBA dynamic range converters, 96kHz, and AVB input!
I don't understand why they are limiting filter slopes to 24dB/oct though, and you can only use one HP/LP pair per output (which is a problem for 3 or 4-way systems, even if allpass can partially address this)

[4313B]

I don't understand why they are limiting filter slopes to 24dB/oct though

That wouldn't work with the M2 which prefers a 36dB/oct on the 2216Nd.

[Valentin]

G.T. is meeting this morning with ALLDSP via Skype for use with his DD67000, K2-S9900 and 880 Array. It will be interesting to find out how that goes. ALLDSP might be of interest to folks as another option.

http://www.alldsp.com/frame.php?cont...etail&ID=1800i

Another potential option from Mr. Widget:

http://lake.labgruppen.com/

it is nice to see thing going forward to active systems

[sebackman]

Hi,

How would these above mentioned DSP's compaire to the BSS BLU-series?

JBL is stating that BSS BLU160 could be used instead of the Crown's for the M2's and that the BSS file can downloaded. But it cannot...

Would sure be nice to see how the configured the DSP filters and EQ's in BSS.

BSS BLU can do FIR filters, anybody here trired? Does JBL use FIR filter's for any application?

//RoB