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  • Behringer DSP1124P Feedback Destroyer

    I picked one of these up cheap a while ago from someone who couldn't figure out how to make it do anything useful . Anyways the manual mentioned a software app that Behringer is supposed to have to control it via MIDI but Googling around says it was either never available or was "pulled". Anybody actually ever see a copy? Any other software that "knows" about it? It would be cool if it could be used just as a PEq controllable via a GUI from the PC - it's too light to use as a door stop :lol: .

    "We Have Met the Enemy and He is Us" - Walt Kelly​

  • #2
    Hmm... just searching the forums for info on this unit and looks like I'm the only one askin' about it. Just an update - supposedly the software for the older version (DSP1100P) works with this unit as well:
    http://behringerdownload.de/_software/dsp1100.zip
    I modeled a 24db/octave 100Hz HPF using 3 PEQ's on it and it seems to work well:
    20Hz 55/60octave -48db
    20Hz 55/60octave -48db
    100Hz 2octave +16db

    I was playing around with mine yesterday as I want to use it in a spare rig I'm putting together.

    After a fixed filter becomes locked there appears to be no easy way to unlock them - they even survive a power cycle . The DBX AFS224 and Sabine 2020+ units I have work MUCH better for live applications. Those two also keep their locked fixed filters through a power cycle but can be easily unlocked :love:.

    EDIT> Hmm.. just checked it and the fixed filters are now unlocked :freak: - perhaps I didn't power it off long enough yesterday? I'm not set up here to easily get the filters to lock so I'll have to do that and see how long the power needs to be off. I could also copy my setup into all ten presets and just switch to a different one when I need to unlock the fixed filters to "prime" then at a gig. You generally only need to do it once anyways but stuff happens :facepalm:. FWIW I'm running the DSP1124P's level switches on the back in the -10db position (as I do with the DBX AFS224's) and changed the sensitivity of detection to -9db (highest allowed) from the stock -6db.

    "We Have Met the Enemy and He is Us" - Walt Kelly​

    Comment


    • #3
      That's some silly way to model the filter. Wonder what happens with data over/under-run? I suspect the dynamic results will not be as you expect.

      Comment


      • #4
        That's some silly way to model the filter. Wonder what happens with data over/under-run? I suspect the dynamic results will not be as you expect.
        Sounded like it was working - if I get ambitious I'll run a frequency sweep on it . I used their software to model it and programmed it in by hand:



        Dunno if they use a fixed or floating point DSP for sure but I suspect the latter?

        "We Have Met the Enemy and He is Us" - Walt Kelly​

        Comment


        • #5
          Back when I use to use Behringer gear I had a couple of those to use on monitor mixes. I used then inline with a graphic EQ and they actually worked pretty good. I did make the mistake of inserting one in a vocal channel once and it killed way to many frequencies and made the singer sound like crap. Not sure why.

          Comment


          • #6
            Sounded like it was working - if I get ambitious I'll run a frequency sweep on it . I used their software to model it and programmed it in by hand:



            Dunno if they use a fixed or floating point DSP for sure but I suspect the latter?


            I suspect the GUI shows a pretty picture but the signal aid the computations will not do well... regardless of the architecture.

            Is there a shelving option on the PEQ???

            Comment


            • #7
              Hmm... just searching the forums for info on this unit and looks like I'm the only one askin' about it. Just an update - supposedly the software for the older version (DSP1100P) works with this unit as well:
              http://behringerdownload.de/_software/dsp1100.zip
              I modeled a 24db/octave 100Hz HPF using 3 PEQ's on it and it seems to work well:
              20Hz 55/60octave -48db
              20Hz 55/60octave -48db
              100Hz 2octave +16db

              I was playing around with mine yesterday as I want to use it in a spare rig I'm putting together.

              After a fixed filter becomes locked there appears to be no easy way to unlock them - they even survive a power cycle . The DBX AFS224 and Sabine 2020+ units I have work MUCH better for live applications. Those two also keep their locked fixed filters through a power cycle but can be easily unlocked :love:.

              EDIT> Hmm.. just checked it and the fixed filters are now unlocked :freak: - perhaps I didn't power it off long enough yesterday? I'm not set up here to easily get the filters to lock so I'll have to do that and see how long the power needs to be off. I could also copy my setup into all ten presets and just switch to a different one when I need to unlock the fixed filters to "prime" then at a gig. You generally only need to do it once anyways but stuff happens :facepalm:. FWIW I'm running the DSP1124P's level switches on the back in the -10db position (as I do with the DBX AFS224's) and changed the sensitivity of detection to -9db (highest allowed) from the stock -6db.


              I had one when it first came out. Maybe 14 years ago or more. I believe there is an easy way to unlock the filters.

              http://www.behringer.com/EN/downloads/pdf/DSP1124P_P0124_M_EN.pdf

              When the filters are Locked they were locked because they were in Single Shot mode when set. Just select the fitler that is locked and put it in Single Shot mode again. When there move it to PA mode to use the filter as a parametric.

              See 7.2 on page 15 of the manual. Like you I ended up using it for a stereo 12 band par. EQ. It was not very noisy. I did end up sending it back. The problem I had with mine it would reset if the input voltage dropped below 93 volts or so. This was long before I carried a distro. Not a bad unit.

              Dookietwo

              Comment


              • #8
                That's some silly way to model the filter. Wonder what happens with data over/under-run? I suspect the dynamic results will not be as you expect.

                Do you mind elaborating a little?

                I've got a passing interest in signal processing and a degree in Comp Sci, so you shouldn't have to dumb it down too much. :lol:

                Usually when I see the term "over-run" it's in relation to a fixed size buffer in an input stage, like the keyboard ring buffer in DOS.

                Is this the same thing, or are you talking about some type of loss of precision or accuracy in the computations?
                Kickin' it in the sticks...

                Comment


                • #9
                  I am talking about computations that when cascaded (like in RR's example) the computations may not be an accurate representation of the function... ie. out of bounds. That's 2 cascaded 48dB/octave filters plus a large boost filter. I do not know as much as I used to about DSP, but I do know that if you will be doing steep operations, it generally pays to double-check that the ALU can handle it. I doubt those conditions would have been considered remotely normal or desireable by the engineers developing the product. All kinds of artifacts can result depending on the architecture (including software). Since we are not privy to the system, it pays to double-check things like dynamic range, actual center freq., noise and noise artifacts, distortions etc.

                  Comment


                  • #10
                    Ah, got it.

                    I think I've seen people refer to this as "overloading" the DSP.

                    My Alesis Micron can exhibit some strange sonic artifacts if I tax the system too much.
                    Kickin' it in the sticks...

                    Comment


                    • #11
                      Is there a shelving option on the PEQ???
                      If there was I'd not be mucking about with faking one .

                      "We Have Met the Enemy and He is Us" - Walt Kelly​

                      Comment


                      • #12
                        I think I've seen people refer to this as "overloading" the DSP.
                        That's running out of processor time to do all the calculations for all the filters or other signal processing algorithms in real time. Here I believe AH is concerned that the math for the three filters might exceed the maximum numeric value that the DSP can represent or the value could drop low enough to lack enough precision to work properly. Early DSP's were fixed point and required a lot of expertise in programming to avoid such. Most DSP's are floating point these days because it is so much easier to program properly as they represent both HUGH and minuscule numbers with equal precision .

                        OTOH the PC software many not be doing a proper job of calculating the response of the filters either so best to do some measurements if I care what it is really doing.

                        "We Have Met the Enemy and He is Us" - Walt Kelly​

                        Comment


                        • #13

                          Here I believe AH is concerned that the math for the three filters might exceed the maximum numeric value that the DSP can represent or the value could drop low enough to lack enough precision to work properly.

                          I think what you are talking about is loss of accuracy, due to the inability to accurately represent base 10 values in IEEE floating point format?
                          Kickin' it in the sticks...

                          Comment


                          • #14
                            I think it might be a little of both, actually. The "real time" aspect comes into play when you have ISR's that fire and don't get responded to, due to a long running calculation. I'm not sure what type of O/S is onboard your unit and how dependent it is on interrupts, but that could be a concern.
                            You're thinking of how stuff like that is done on a PC or Mac. A DSP normally has no OS and may or may not have any ISR's.
                            I think what you are talking about is loss of accuracy, due to the inability to accurately represent base 10 values in IEEE floating point format?
                            No.

                            "We Have Met the Enemy and He is Us" - Walt Kelly​

                            Comment


                            • #15
                              You're thing of how stuff like that is done on a PC or Mac. A DSP normally has no OS and may or may not have any ISR's.No.

                              I'm just wondering how the result could lose precision, since that is fixed by the architecture.

                              Loss of accuracy happens in floating point math....

                              My micron has an O/S: the DSP isn't used to drive the control panel lcd and accept input. I'm just kind of "assuming" that maybe your device has a similar architecture?

                              Oh well, interesting speculation anyway....:wave:

                              (BTW: technically speaking, you should say "Fixed-Point" and NOT fixed precision. IEEE floating point is fixed precision. Nitpicky, I know)
                              Kickin' it in the sticks...

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