Statistics

254,292 members
131,967 posts
  • New
    daveh13
    Newcomer - Level 1
    2021-10-09

    Hi,

    I do not know if this is the place for suggestions/requests and if Behringer is reading here, but :

    having a bunch of old synths/rythm box and other unbalanced gear directly plugged into the rear XLR in my studio, it would be great to have a "phantom power" param safe in the channel param safe list. Just to prevent any bad manipulation of the 48V button that is easy to press when you fly your hands above the console... I would not like to fry my nice instruments... ;-)

    read more...
    0 8
    • daveh13
      Paul_Vannatto Gain and phantom are part of the HA Config safe.
      • Oct 9
    • daveh13
      daveh13

      Yes, I know this safe "group" of params, but I think it would be nice to be able to lock only the 48V phantom, that is really an unsafe feature, but still be able to tweak gain/lowcut, etc...
      For example, when I unplug and plug a new synth at the rear, I have to tweak the gain again, and my hands are flying close to the 48V button... Scarry !

      • Oct 9
    • daveh13
      Paul_Vannatto Why not make a snippet to turn off the desired phantom. The OSC command is /ch/01/preamp/hpon OFF. Another option is to get a stagebox (SD8 or SD16) and plug the synths into the TRS jack of the combo jack, thus eliminating the risk of phantom.
      • Oct 9
    • daveh13
      daveh13

      I am not familiar with snippets, which seems to be a way to save some parameters. But I don't understand how a snippet could prevent me from accidentally togglin the 48V button ? Is it possible ?
      For the SD16, I know, it would be great, but alas, the budget is not here for the moment ;-) I wish Behringer also put some mixed XLR/TRS at the rear of the X32...

      • Oct 9
    • daveh13
      Paul_Vannatto Safes do not prevent you pressing the phantom button. All safes do is limit what settings are applied to the console when a scene or snippet is loaded.
      • Oct 9
  • New
    daveh13
    Newcomer - Level 1
    2021-10-09

    Hi,

    When tweaking the gain of an analog input, I usually aim the -18 to -12dB area to keep some headroom,
    and usually, no clipping ever occurs.

    But I have a strange thing for the digital inputs, coming from the X-USB card and from my DAW.

    The output of my DAW is by definition limited to 0dB, and I even put a limiter to -0.3dB, so that no sample ever touch 0dB, and that no digital clipping ever occurs.

    So when I connect the digital DAW output to a X32 channel, even with a 0dB gain, there should be no digital clipping.

    It's not the case !

    I even have to lower gain on the X-USB input to -16dB to avoid clipping. And in this case, as we can expect, I see the vumeter staying largely under -16dB (should be precisely -16.3dB).

    So I wonder why the X32 points out some clipping. Is this clipping LED really a digital clipping detection ?
    Or is it some kind of prediction algorithm that prevents intersample peaks ?

    I understand there could be a predictive algorithm for analog inputs,
    but, when everything stays in the digital word, clipping is very simple to detect.
    And -16dB to avoid intersample clipping seems huge...

    What do you think about it ?  An idea ? Should I forget this clipping light on true digital inputs ?

    Have a nice day.

    read more...
    0 13
    • daveh13
      Nigel67 Hi davehbsk. The clip led lights up from -0.3 dBFS at the digital input, as a cursor warning that the signal is close to clipping rather than it being too late and the signal is clipping. It gives just a small amount of headroom and to compensate for the fact that not all samples are used for clip detection.
      It happens also on the analogue inputs but the digital level (after ADC) is not always what you see on the input meter because the preamp gain is in 2.5dB steps, and the rest is digital trim. The console will not clip though because of floating point processing, the trim being in the dsp domain. Hopefully this answers your question. Many thanks
      • Oct 9
    • daveh13
      daveh13

      Hi Nigel, thanks for your answer. But if I understand your first sentence about the digital input clipping, the return from my DAW thru X-USB (which is obviously digital), should never clip as long as I introduce a little gain reduction of -1dB (to stay under your -0.3dB limit). BUT in my case, to avoid red clipping light, I have to set a -12dB on the gain of the DAW return... If I set only -1dB, the channel clips all the time (especially when the return of my DAW is a "mastered" final signal, compressed and itself limited to -0.3dB).

      • Oct 10
    • daveh13
      GeorgeDougherty It may be the same with other DAW's but I use Reaper and recently resolved a similar issue. Reaper treats everything as stereo and I found if I send a channel to a mono hardware out it sums the L/R internally for a 6db gain as you get with any combination of identical signals. On the Reaper hardware out routing dialog there is a 1/2 dropdown. Clicking that gets an option to select a mono source and either channel 1 or 2. Pick one, and the output into the board will drop by 6db and should match what's in the DAW unless you have the Trim control adjusted on the board.
      • Oct 17
    • daveh13
      daveh13

      thanks Georges, infact, I must have made a mistake, because the problem is not there anymore and I cannot reproduce it. A "-0.3dB maximed" wav file now never clips when I send it to a channel with 0dB gain. But as soon as I raise the gain, it clips. So everything is normal again...

      • Oct 17
  • New
    Tjbasch
    Newcomer - Level 1
    2021-10-09

    I have a used x2222. I have signal showing on the LEDs but no headphones nor speaker output. I'm sure it's a button or know but can't seem to figure outwhere.

    0 16
    • Tjbasch
      Paul_Vannatto Are the mute buttons pressed, or the Main buttons not pressed, or the Main Mix faders down?
      • Oct 8
    • Tjbasch
      Tjbasch All of the above are correct.
      • Oct 8
      • Xenyx x2222 output
        Nigel67 Can you post a picture looking at the mixer surface showing the audio on the meters. maybe we can then identify where the issue is. Check that the two red 2tr / USB red buttons are in the up position. For the headphones, have you assigned the Main Mix button in the Source area? Is this a brand new mixer that you are just starting out with or one that has worked previously? You could also send us a Technical Support ticket and we will be able to help you further via that. This can be done by clicking on the support tab at the top of the page. A new window will open. Scroll down and click on the Technical Support tab and submit a ticket.
        • Oct 8
    • Tjbasch
      Paul_Vannatto If all of the above are correct, then that is why there is no output. For each channel you want to go out the Main outputs, unmute, press the Main button and bring the Main Mix faders up.
      • Oct 9
      • Xenyx x2222 output
        Tjbasch What I meant was the channels I'm using are unmuted, main buttons are pressed and the main mix faders are up.
        • Oct 9
    • Tjbasch
      Tjbasch This was purchased new a few years ago. It worked fine. I spent a couple years in the hospital and am now trying to re-learn. The LEDs were working, now aren't. I've tried so many combinations I have no idea what I pushed wrong. Hope you can see the picture ok.
      • Oct 9
    • Tjbasch
      Paul_Vannatto No picture available.
      • Oct 9
      • Xenyx x2222 output
        Tjbasch It's taking forever to upload
        • Oct 9
  • New
    illuminator777
    Newcomer - Level 1
    2021-10-08

    Hello I just recently purchased a shure MV7X and my GOXLR.
    When I launch the GOXLR app, the mic set up screen doesn’t receive signal from my microphone. I have tested my usb, mic, XLR cable all on a different interface and confirm they’re all working. Does anyone have a similar issue? I’m aware the MV7X is extremely new, but I can’t figure out why my GOXLR won’t even pick up any noise from my mic. Any help would be gladly appreciated. Thanks

    read more...
    0 3
  • New
    daveh13
    Newcomer - Level 1
    2021-10-08

    Hi,

    I'm quite a newbie in the X32 world, and I do not master the routing subject... So I need your help.

    I don't know if there is an answer to my question, which may be dumb... :-)

    I am in a studio configuration in a single room and I would like to send the main mix at will on 3 different pairs of speakers
    (one pair is power speakers on XLR OUT1-2 when we rehearse in the room, one pair is mixing monitors on XLR OUT3-4,
    and one pair is a basic HIFI on XLR OUT5-6 for reviewing the final result on a average consumer equipement)

    I would like to send the main mix (commanded by the master fader, and all the input faders) at will to any of these pairs of speakers, with an additional "send level", and easily switch between the pairs.

    I know that I could create 3 submix and send them to the 3 pairs, but then I'll have to switch to the bus page
    everytime I want to modify the global volume (the master fader won't have any effect...)

    The main interest for me is : I'd always want to use the master fader to tweak the master volume,
    whatever the pair of speakers I use. And I'd like to have an additive layer of volume to set the average level of each pair of speakers, so that I can easily switch from one pair to the other, especially between the monitors and the HIFI to make comparisons.

    I don't know if I'm clear...

    I see the matrixes, but I don't know if they can do this...

    Maybe we can resume my problem : can we route the main bus to several submixes ?

    Thanks for your help.

    read more...
    0 9
    • daveh13
      KevinMaxwell Mix all of your inputs to L/R and then send those to the Matrix outputs. This should do exactly what you want to do and then just use the L/R master to change the overall level to the Matrix outputs and on the Matrix masters you can set the relative levels you want for each set of speakers.
      • Oct 8
    • daveh13
      daveh13

      Thanks, that's exactly what I need ! I felt that the matrixes was the way to go... ;-)

      • Oct 8
  • New
    MaciejKamil
    Contributor - Level 2
    2021-10-08

    Any dedicated thread/discussion for the topic ?

    Or should i start a new one ?

    0 10
    • MaciejKamil
      Paul_Vannatto This thread will do. Because of the way this "new fangled" forum interface is designed, you won't be able to find this thread in about a week's time anyways. So go ahead and add your feature requests and they will be relayed to the developers.
      • Oct 8
    • MaciejKamil
      MaciejKamil

      You can definitely go nuts with this new engine :)

      • Oct 8
    • MaciejKamil
      MaciejKamil - direct input to MTX – access to all physical/digital inputs should be granted – local and aux I/O, AES50 ports, stage connect
      - global TAP POINT configuration
      -global processing order configuration
      - delay on busses
      - oscillator parameters on User Defined
      - selected channel page on User Defined (TAP eq page, but not for a specific chnl, but for selected chnl)
      - direct OUTs for input channels as a pool in routing, and a dedicated page for configuring it’s tap and patch in channel view (next to input patch icon in the home screen?)
      - GEQ to faders in an automated fashion
      - stereo Spring Reverb
      • Oct 8
    • MaciejKamil
      Paul_Vannatto Thanks MaciejKamil for those suggestions - my comments below
      - direct inputs to MTX - I doubt they will implement this (already been requested), since the X32 and Wing design is that channels feed buses and buses feed matrix. Granted they did add the feature that buses now can feed buses 1-8 (for Fx to monitors purposes).
      - global tap points and processing order config - great idea
      - delays on buses - I believe there is a hardware limitation as to why that is not implemented.
      - more User Defined options - great idea
      - direct OUT - this is usually done in the routing section. For channel processing direct OUT, use the User Signal routing screen.
      - GEQ to fader - already has been suggested - great idea, except many of us rarely use GEQ anymore, since the PEQ are so much better.
      - stereo Spring Reverb - already available as a premium FX (slots 1-8). By the way, everything is stereo on the Wing.
      • Oct 8
    • MaciejKamil
      MaciejKamil

      Hey Paul,

      thanks for your contribution, here is what i think:

      Direct input to matrix - this is already available, but one can choose only from input chnls or AES/EBU inputs, and that is a pity.
      We should have an option to choose from all physical inputs. Honestly speaking direct input should be present in all buses including mains. That would give a chance to cascade desks and reach higher chnl counts.
      Regarding delays on buses - if this was possible on mains could be also possible on buses.
      For chnl processing Direct out the user signl is not so convenient.
      As for the GEQ on faders, i would be far from stating that GEQ is obsolete - apart from that GEQ on faders is an industry standard workflow - i would not argue with that.
      As for the reverbs, not only the Spring one, they are not exactly stereo. I double checked, and it seems that they only take input R for processing - try it out.
      When You make the stereo reverb bus as a post fader send, then You tweak channel pan left and send pan right, there will be no reverb at all (Spring reverb).
      BTW, sending POSTFADER after the chnl pan pot gets me confused - this would be ok when the bus is a SUBGROUP. But with the bus being a POSTFADER send including the pan pot makes no sense for me personally.
      I was really excited with all this STEREO stuff and possibilities it gives, but at the moment it does not work as expected.
      I cannot pan the instrument left ond its reverb right. Not possible. I tought its a matter of a single algotithm - the spring - but i double checked and even VSS3 does not do the job.

      Thanks for Your input.

      I'll be around if You'll need me.

      • Oct 14
  • New
    chadf9942
    Newcomer - Level 1
    2021-10-08

    I am looking for a pro 6 power supply. 
    one of mine isn't working

     

    0 6
    • chadf9942
      Nigel67 HI chadf9942. Is your Midas PRO6 still under warranty? Its 10 years from 1st Sept 2014 if you are the first owner. If so, please send in a Technical Support ticket and we will be able to help you further. This can be done by clicking on the support tab at the top of the page. A new window will open. Scroll down and click on the Technical Support tab and submit a ticket. If the PRO6 is out of warranty, then you can purchase a new power supply through our Spares team.
      Please send in a spares request ticket. This can be done from the same page that the Technical Support ticket can be found, just select the Spare Parts tab instead. Many thanks
      • Oct 9
  • New
    davep54
    Newcomer - Level 1
    2021-10-07

    I just need to know the configuration on the aes50 cabeling and and how to send all my  outputs from the dl32. I so far have no problem getting all the inputs to both consoles but the outputs are a problem.I wanted to ust xlr out 7 and 8 as my m32C foh mix output and 15 and 16 for x32  for foh in another room. Aslo wanted to send buss mixes 1-6 out of dl32 for monitor sends from x32.

    Hope this make sense. This routing system makes no sense to me and I'm not new here.

    I thought the newer firmware would make it easier and for inputs it does but I cant make sense of the output section.

    read more...
    0 22
    • davep54
      Paul_Vannatto Unfortunately your description of your outputs is unclear. Also you didn't provide your interconnectivity mapping (how everything is connected together). For example:
      X32 [B] <-> [A] M32C [B] <-> [A] DL32
      • Oct 7
    • davep54
      davep54 I didn't provide mapping cuz that is what i'm looking for to start. then i need to figure how to get the routing where I need it. so I will explain again. I have an X32 at Foh that will get its inputs from DL32 at the stage. that x32 will provide house l+r and 6 monitor sends to stage where dl 32 is. Then I have an M32C at the stage that will get its inputs from the DL32 also. I need the M32C to send the its l + r outputs out of The DL 32 and the X32 Also to send l +r out plus the 6 monitor sends out of the dl32. The M32C is Basically mixing for a different room but all the outs from both boards ar coming from the dl32. Hope this is more clear.
      • Oct 7
    • davep54
      Paul_Vannatto Your second description is much clearer. The interconnectivity mapping I provided will work for your scenario. Give me a bit of time to work out the routing. Which console do you want to control the DL32 preamps?
      • Oct 7
    • davep54
      davep54 The X32
      • Oct 7
    • davep54
      Paul_Vannatto Here you go Dave.

      Interconnectivity mapping ([A] and [B] are AES50 ports)
      X32 [B] <-> [A] M32C [B] <-> [A] DL32

      X32 and M32C
      Routing, Inputs
      * Inputs 1-8 -> AES50 B1-8
      * Inputs 9-16 -> AES50 B9-16
      * Inputs 17-24 -> AES50 B17-24
      * Inputs 25-32 -> AES50 B25-32

      Routing, Out 1-16 (setting up a bank of 8)
      * Output 1 -> bus 1 (post)
      * Output 2 -> bus 2 (post)
      ... (etc)
      * Output 6 -> bus 6 (post)
      * Output 7 -> Main L (post)
      * Output 8 -> Main R (post)

      X32
      Routing, AES50 B (to M32C)
      * Outputs 1-8 -> Out 1-8 (X32 outputs)

      Routing, XLR Out (outputs on the back of the X32 console)
      * Outputs 1-4 -> AES50 B33-36 (M32C outputs)
      * Outputs 5-8 -> AES50 B37-40 (M32C outputs)

      M32C
      Routing, AES50 A (to X32)
      * Outputs 1-8 -> AES50 B1-8 (DL32 inputs)
      * Outputs 9-16 -> AES50 B9-16 (DL32 inputs)
      * Outputs 17-24 -> AES50 B17-24 (DL32 inputs)
      * Outputs 25-32 -> AES50 B25-32 (DL32 inputs)
      * Outputs 33-40 -> Out 1-8 (M32 outputs)

      Routing, AES50 B (to DL32 outputs)
      * Outputs 1-8 -> AES50 A1-8 (X32 outputs)
      * Outputs 9-16 -> Out 1-8 (M32 outputs)

      Typically the console closest to the stagebox has control of the preamps. To relinquish control of M32C to X32, select HA Remote AES50 A, on the M32C Setup, preamps screen.

      If you want digital trims on both of the consoles, also select the HA Gain Split. That will mean that the X32 console will have to control the actual DL32 gains on its Setup, preamps screen.
      • Oct 7
  • New
    daveh13
    Newcomer - Level 1
    2021-10-07

    Hi there,

    I just received a X32 compact, for my studio use, and I wonder how I should connect my local synths (unbalanced line level TS outputs) to the balanced XLR inputs of the X32. They are very close (2-3 meters).

    I know that the MIDAS preamps can adjust the level, but I wonder what type of cable I should use, or if a DI is mandatory (I'd like to avoid buying 10 DIs...)

    So I bought simple unbalanced TS 6.35 jack to XLR cable, with XLR pin2 connected to jack tip and XLR pin1 & 3 connected together to jack sleeve.

    Will it be ok ? (I'd be careful with the phantom power, I swear ;-) ).

    Thanks for your help.

    read more...
    0 11
    • daveh13
      Paul_Vannatto That cable should work.
      • Oct 7
    • daveh13
      DavidThomson1 I connect mines direct to the xlr inputs with no issues (balanced) but as you said just be careful of that phantom power. on occasion I also use the auxiliary inputs (if I only have the TRS cables with me or for laziness at home), and simply remap the aux input to my regular channels.
      • Oct 8
    • daveh13
      daveh13

      thanks for your help. I tried and it worked like a charm !

      • Oct 8
  • New
    robgwin
    Newcomer - Level 1
    2021-10-07

    Hi all,

    I am considering a stage setup where I would connect the balanced outs of an onstage Behringer XR18 to the house snake and ultimately to FOH mic inputs. I would be sure to attenuate the XR18 output to send a mic-level signal.

    However, I'm concerned about the possibility of damage to the XR18 if FOH were to (perhaps accidentally) send 48v phantom power on those channels. I'm finding opinions on both sides of this question, but it ultimately seems to come down to whether the specific mixer has blocking caps on the outputs to protect against unwanted voltage.

    Does anybody know if the XR18 outputs have such protection? Or how I might go about finding out? Haven't been able to find anything about this in Behringer documentation.

    I know I can avoid the question entirely with DI boxes, but I'm trying to figure out if that's really necessary.

    Thanks!

    read more...
    0 6
    • robgwin
      Nigel67 Hi robgwin. Yes they do have protection on their outputs, but eventually this protection will fail if 48V is continuously applied to the outputs. The transient as the 48V is switched will be the main culprit for damaging the output circuitry. I would advise that if you know that there is a good chance the outputs are going to have 48V applied to them, then you should think about using DI boxes. Even the Pro consoles, no matter how much protection is built in to the output circuitry, are still susceptible to damage from unwanted 48V applied to them. I have repaired plenty.
      • Oct 9
      • Could Behringer XR18 outputs be damaged by unwanted phantom power?
        robgwin Much appreciated, thanks!
        • 1
        • ·
        • Oct 11
Go to page