Behringer

Behringer

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Behringer

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127,373 posts
  • askuse7
    Contributor - Level 2
    2019-08-28

    Hi everyone,

    I purchased a Model D a few months ago and went through the calibration process and it is working great.

    I let the Model D warm up for 30 mins before I use it, but I still like to check the tuning occassionally so I connected the Low output to a Boss guitar tuner, and when I need to tune I just power on the tuner, make sure only Osc 1 is on, then press an A note on my controller keyboard, and then use the Tuner knob on the Model D to correct the tuning.

    This works well, except you can hear the A note when I play it through my speakers. Is there a way to do this so the note will not be heard so I can tune in-between songs when I play live? If I press the Main Out switch to off, will the signal still go out Phones or the 1/8" Main Audio Outputs?

    Also the Model included 2 1/8" mono cables. Are they exactly the same? Can either cable be used for audio and CV connections?

    Thanks!

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    • askuse7
      DaveMorrison

      @askuse7 


      How about an A/B switch pedal? Switch out the tuner when playing; Switch out the PA when playing.

      • August 28, 2019
    • askuse7
      askuse7

      Hi Dave,


      That would certainly work, but I was looking for a solution that doesn't require any extra cables or footswitches, and I think I've found it.


      I did a bit of research tonight, and I first confirmed the Phones output is completely separate from the 2 Main 1/4" Audio outputs, which makes sense. Next, I inserted a stereo 1/8" to 1/4" headphone adapter in the 1/4" input of my guitar tuner, and then connected a stereo 1/8" cable between the tuner and the 1/8" Phones output of the Model D. I then switched Off the Main Audio output switch. I then played an A note and the tuner recognized it no problem, so I now have a good working solution for checking the tuning without the sound being heard through my speakers.


      I also checked the 1/8" Main Audio output as well, and it is not separate from the Main Audio volume control. If you press the Main Audio switch to off then the signal is shut off from the 1/8" Main Audio output as well. Unusable for silent tuning.


      But I have a good working solution using the Phones output for checking the tuning of my Model D when I use it for live performance without it making a sound through my speakers.


      Hope this helps anyone else wanting to do the same!


      Andy

      • August 28, 2019
    • askuse7
      RonaldFigura

      Here's what I used to do with my MiniMoog... Took the headphone output of the Moog and ran it into a digital tuner that recogzed ANY pitch coming into it and showed the status of the tuning. Don't remember the make or model of the tuner, but it was cutting edge at the time (early 1980s). I powered it with a walwart so it could be always on. No need to mute the MiniMoog to tune because it showed whatever note I was playing. But if I needed to I could mute the Mini output and the headphone output would still supply an output to the tuner. I just had to be carefull not to drive the headphone output too hard or the tuner would "freakout". You can do this same thing on the Behringer Model D. All the functionality is there.

      • September 21, 2019
    • askuse7
      RonaldFigura

      PS- find a "digital chromatic tuner" This should work as I described in my other reply to your post.

      • September 21, 2019
    • askuse7
      askuse7

      Hi Ronald,


      My Korg Slimptich chromatic tuner arrived last week from Japan, and it worked exactly as I had hoped!


      Added bonus, it shows B Flat instead of A# and G# instead of A flat, so I can't make the same mistake I made last time


      See attached pic for final solution.


      Andy

      Model D Slimpitch Tuner.JPG
      • September 22, 2019
  • celoranta
    Contributor - Level 3
    2019-08-28

    Hello folks.

    I've been trying to use the X-LIVE card to perform virtual soundchecks with my band.

    As we generally do not have a front-of-house guy, my desire is to record a sound check tune to the X-LIVE, then move to the FOH with my iPad and X-Edit, then play back the recorded sound check and make channel adjustments as needed.  

    From what I've read previously, the process above is one of the intended purposes of the X-LIVE.

    Unfortunately it seems that the recording, when played back, differs greatly from the sound of the live band.  I do not believe this should be the case.  Here are the specifics:

    • We have NO acoustic instruments aside from our vocals.  Guitar is through an amp head, cab simulator and DI, bass is direct to board via DI, electronic drums and keys are direct to board as well.  
    • There are no stage wedges.  Our mix is via IEMs and P-16s.  The stage is quiet.
    • The level of our overall mix is very loud.  More than enough to overwhelm the acoustic voices as they pertain to the FOH mix.
    • I had previously (accidentally) messed with the TRIM settings, which certainly screwed up the playback a lot.  This has since been corrected and all 'tape' trim settings for all channels are set to 0db.
    • 'Record' mode channels
      • 1-8 are sourced from digital snake inputs 1-8
      • 9-16 from x32 input 1-8
      • 17-32 are irrelevant for now
    • 'Play' mode channels are assigned as defaulted:  Card 1-8 to Channels 1-8, etc.

    Am I missing anything here?  Is there any other reason the played back audio might sound different than the live audio?  Please lend a hand as I am very excited to change over our process, but my patient band mates are losing their patience somewhat with my failed attempts.

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    • celoranta
      RexBeckett

      @celoranta 


      Hi Chris, the fastest way for us to help is to look at your scene. Save the scene and attach the file to your post (see Choose File button below).

      • August 28, 2019
    • celoranta
      garyh

      @celoranta i am not sure but suspect the issue is with the tap points for the card, if you use the “local” ch source for card out, you will get pre everything recorded to the xlive. This will be true for aes50 sources like digital snake mic pre’s also. When you then playback you get the ch processing. You get half your sound in a way. It’s not just the xlive btw, other usb cards do it the same. It’s a flaw imo of local to card routing. by not allowing local sourced tap points other than pre. The work around is to use another routing menu like p16, aux out or even out 1-16 to use as the source to your card. Doing this will allow you to select pre fader or post fader (or other) ch taps to the xlive card. In card out routing, change it from local to whichever source you decide to use. P16 might work well since you already are using p16m’s in your system.

      • August 28, 2019
    • celoranta
      celoranta

      Thanks for the assist, but 'pre' is exactly what I'm looking for.  I want to record my channels before any processing, and then play them back through the same channels on the board (with the same settings,) which should allow any processing to affect the recorded track in the exact way it was affecting the live channel.

      • August 29, 2019
    • celoranta
      garyh

      You had started this thread by stating your recording (using local and aes50 inputs) wasn’t working for you? When you record pre everything card tracks and then play them back you are dealing with ch trim on a card playback ch. if you then adjust everything based on trim, as soon as you go back to the live gain based mic ch level of the given current gig, there will be a difference, the difference CAN be great. Mic placement, room acoustics, bleed, etc are all factors that may vary from when the recorded version was done. I also assume you listen to post fader main l/r.


      I find post fader recorded card tracks to be much closer to a live ch level and sound, even if you use flat eq on the recorded side going in. Your complaint in the thread open more or less appears to bear this out: "Unfortunately it seems that the recording, when played back, differs greatly from the sound of the live band." You should surely do what you find works the best for you. It may not be the usual way, I was only offering another possible solution. 


       

      • August 29, 2019
    • celoranta
      JackDemeis Chris, I have the exact same problems; 1) no or inexperienced soundman, 2) electronic drums, and 3) IEM. I'm the drummer in the band and a long time FOH engineer and would love to be able to get a FOH mix 85%-90% done and have the inexperienced soundman just adjust levels. While Gary brings up good points about "...the difference CAN be great. Mic placement, room acoustics, bleed, etc are all factors that may vary from when the recorded version was done." I seems really odd that the resulting differences between playback live can be great. To me, I think your right on regarding the channel gain. I think in the end what we need is the source during playback (in my case it's Logic Pro X) to have at least similar input level as live.
      • October 7, 2020
  • RayDoetjes
    Contributor - Level 1
    2019-08-28

    My DeepMind12 started to randomly freeze up.

    I have updated to the latest firmware 1.1.2, recalibrated even increased the default fan speed to rule out overheating.

    Nothing has helped. The store reverted me to this site to make an RMA request. But I can’t find how to submit an RMA request to send in the instrument to be serviced. I have the receipt.

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  • Laast
    Contributor - Level 1
    2019-08-27

    Hello I am really wanting the blue MS 1 and in Canada it seems like only red is available.  Do we know if the blue will be available and if so when?  Thanks.

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  • duelinmarkers
    Contributor - Level 1
    2019-08-27

    I just got a Behringer Neutron yesterday, and it seems like the FREQ MOD IN (VCF cutoff control) has no effect. Before I go looking into repair, is there anything anyone can think of that I might be messing up to make me THINK it's broken (e.g., some normalled connection that needs to be overridden by patching)?

    I first tried OSC2 OUT -> ATT1 IN -> FREQ MOD IN and raised the ATT1 knob, which I expected to bring some noise, then tried some simpler LFO patching and again heard no effect.

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    • duelinmarkers
      duelinmarkers

      Doh! Answered my own question this morning: I had MOD DEPTH all the way down. FREQ MOD is working fine, and I now know I don't need to use up an attenuator on it, thanks to MOD DEPTH.

      • August 26, 2019
  • KenMitchell
    Rock Star - Level 3
    2019-08-25

    In the spirit of XAirRemote (inspired by @EdGelmetti ), I've started work on X32Remote (inspired by @droberts ). 

     

    X32Remote is a bridge that sits between a TouchOSC or Lemur client and an X32 console and allows the client to have two-way communication with the X32.  Why is this needed? Anyone who's tried to get TouchOSC or Lemur to work with an X32 knows that they don't support the single "listen" OSC port in the same way X32 Edit does, so they can't carry on a two-way conversation.  Also, neither client was designed to periodically send the required "/xremote" OSC command to the console to keep updates alive.  X32Remote solves both ot these issues. 

     

    So far I have channel and bus master fader, mute, pan, and solo working as well as dca on/off and fader. My next task is to implement scene loading so X32 scenes can be loaded with a single button press. 

     

    I'm currently developing on the Raspberry Pi so all of the Alpha binaries will be for that platform. I plan to suport Mac, Linux, and Windows in future builds. Please let me know what controls you'd like to see implemented on the bridge.  Bus sends for monitor mixes come to mind. 

     

    Ken

     

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    3 279
    • KenMitchell
      KenMitchell

      I'll have the first alpha build ready later today for anyone who wants to try it out.  Again, it will only be available as a Raspberry Pi binary for now. 


       


      Ken

      • August 28, 2019
  • JamesCarney
    Contributor - Level 2
    2019-08-25

    I’m using an X32, connected to two daisy changed S-16’s on AES50 A. All was well until the board was powered up for an evening service last night.  No changes were made, and a base-line scene was loaded.  There are 4 condenser mics, all of which have phantom power applied.  The phantom power LED is “on” at the board, as well as on the S-16’s.  However, the condenser mics don’t work. When I turn phantom power on, or off, either at the board, or on the S-16, the board and S-16 show the same state (either on or off) Tried all 4 on another system and all functioned fine.  Put a “Sound Tools XLR Sniffer” on the line at the microphone, and it shows no power. Checked the routing and it’s OK (otherwise the lights wouldn’t show on the S-16 when toggled on or off I assume).  Also checked “Setup” screen, “pre-amp” tab, and the HA setting are appropriate (not split).  Finally, checked the “Sniffer” on the other system and it too functioned appropriately (lit up on a channel when phantom power was supplied.  I’ve also checked the cable and runs under the stage by plugging a dynamic mic into the same input and it worked perfectly. Am I overlooking something obvious, or do I have a power supply problem?

    Thanks in advance for any help.

    Jim C

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    • JamesCarney
      RexBeckett

      @JamesCarney 


      From the results of your tests and the fact that the problem is on multiple inputs, it seems likely that the 48V power supply in the S16 has failed. 

      • August 25, 2019
    • JamesCarney
      KevinMaxwell


      @JamesCarney wrote:


      I’m using an X32, connected to two daisy changed S-16’s on AES50 A. All was well until the board was powered up for an evening service last night.  No changes were made, and a base-line scene was loaded.  There are 4 condenser mics, all of which have phantom power applied.  The phantom power LED is “on” at the board, as well as on the S-16’s.  However, the condenser mics don’t work. When I turn phantom power on, or off, either at the board, or on the S-16, the board and S-16 show the same state (either on or off) Tried all 4 on another system and all functioned fine.  Put a “Sound Tools XLR Sniffer” on the line at the microphone, and it shows no power. Checked the routing and it’s OK (otherwise the lights wouldn’t show on the S-16 when toggled on or off I assume).  Also checked “Setup” screen, “pre-amp” tab, and the HA setting are appropriate (not split).  Finally, checked the “Sniffer” on the other system and it too functioned appropriately (lit up on a channel when phantom power was supplied.  I’ve also checked the cable and runs under the stage by plugging a dynamic mic into the same input and it worked perfectly. Am I overlooking something obvious, or do I have a power supply problem?


      Thanks in advance for any help.


      Jim C





      Are the 4 mics on one S16 or spread out over both of them? If they are all on one of them did you try swapping the S16s to see if the problem is on both or only one of them?

      • August 25, 2019
    • JamesCarney
      JamesCarney

      Thanks for the reply Kevin.  They are spread across the two S-16's.  Podium mic and suspended choir mics on one unit, and some drum pencil condensers and a condensoer for an acoustic guitar on the other. My next step will be to disconnect the S-16's re-route the inputs back to local and see if I get phantom power on any channel that way.  After that, I'm out of ideas, other than it being a power supply problem. Is there a another way to test the power supply before sending the board out for repair?

      • August 27, 2019
    • JamesCarney
      JamesCarney

      Thanks Rex.  From your response it appears you think the problem is in the S-16 unit rather than the board, which would be the "best of the worst" situations as opposed to it being in the board. If it were just one S-16 that failed, would it impact both units since they are daisy chained? The problem is with both units. Any other way to check the power supply that would would confirm the problem is isolated to the the problem before shipping the unit back for repair (it's still under warranty)?

      • August 27, 2019
    • JamesCarney
      RexBeckett


      @JamesCarney wrote:


      Thanks Rex.  From your response it appears you think the problem is in the S-16 unit rather than the board, which would be the "best of the worst" situations as opposed to it being in the board. If it were just one S-16 that failed, would it impact both units since they are daisy chained? The problem is with both units. Any other way to check the power supply that would would confirm the problem is isolated to the the problem before shipping the unit back for repair (it's still under warranty)?





      @JamesCarney 


      I had not understood that phantom power wasn't working on both S16s. It isn't impossible but not so likely that both units have failed 48V power supplies. 


       


      I would try disconnecting the S16s from the console and all input/output cables. Set one S16 into mode 11 (Daisychain Master). Plug your XLR tester into one of the inputs and use the control panel to switch phantom power on. If there is still no phantom, a power supply problem is most likely. If phantom power works, there may be some problem in the cables.

      • August 28, 2019
  • ctzeninsane
    Contributor - Level 2
    2019-08-25

    Hi there. Our church bought an x32 Compact a week ago. I am still trying to figure out how to set up a stereo mix. I have a MA400 which was used for monitoring on the previous analogue mixer. I was just wondering about what you need to do in order to get the stereo monitoring from the x32 for the MA400. Our current bus setup is:

    Bus 1~6: 6 monitoring bus (pre-fader) assigned to output 1~6

    6 subgroups

    4 fx bus 

    I know I need to link bus 1 and 2, but it gets a bit confusing after that. MA400 only has one TRS input for monitoring. You do have a choice between going mono or stereo with a dip switch. Do I need to use the physical output 1 and 2 using some sort of a Y-cable (left and right XLR to a stereo TRS) and feed it to the MA400, or is there other proper ways of connecting it?

     

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    • ctzeninsane
      RexBeckett

      @ctzeninsane 


      Hi Ray, welcome to the forum.


       


      You are correct - for a stereo monitor feed you would link two buses and route them to XLR outputs. To connect the two XLR outputs to the MA400 input you need a cable like this.  The cable wiring is:


      Stereo TRS Cable.PNG


      You could also consider using individual monitor amps so that each user has a convenient volume control. See this thread for details on using a Powerplay P2 for this. The thread includes information about the required cables.

      • August 25, 2019
    • ctzeninsane
      ctzeninsane

      Thank you very much for your fast response. Really appreciate it. Just a quick another question - after I hook them up with the correct cables, would simply panning the channels in my monitor mixbus be all I will need to do?

      • August 25, 2019
    • ctzeninsane
      RexBeckett


      @ctzeninsane wrote:


      Thank you very much for your fast response. Really appreciate it. Just a quick another question - after I hook them up with the correct cables, would simply panning the channels in my monitor mixbus be all I will need to do?





      @ctzeninsane 


      By default, the send from a mono channel to a stereo Mixbus pair will feed equal signals to both sides. If you open the Sends tab for the channel, you will see a pan control that allows the channel to be moved in the stereo field. The channel sends for your monitor buses should be one of the pre-fader options. Post EQ is usually a good option as it includes most channel processing apart from compression.


       


      If you need any help with mixer configutration and/or settings, save your scene file and attach it to your post (See Choose File button below).

      • August 25, 2019
    • ctzeninsane
      ctzeninsane

      I appreciate your advice and sorry about the late reply. My last question would be, in that case, since I would only get 3 monitor outputs if I wanted to have stereo mixes for everyone, is it possible to use the 6 aux outputs as extra 3 stereo monitor sends by converting my subgroup to prefaders then route them to the aux outputs, so I would get a total of 6 stereo monitor outputs?

      • August 27, 2019
  • goinz
    Contributor - Level 1
    2019-08-25

    I am not sure if it is a bug in sequencer but if you edit a pattern it is not possible to overwrite a rest with a note. vice versa is possible or note with note too. I run OS 1.07. BTW Great work! Love the Synth.

    0 137
    • goinz
      goinz

      Ahh....  have to rmeove the rest and the accent parameter before adding the note . But Sequencer is a bit buggy in some cases. But I think it is a gr8 feature and very powerfull. But a very big feature will be to control all other parameters too, like VCF or OSCI tune or what ever. 

      • August 24, 2019
  • doubleJ
    Contributor - Level 2
    2019-08-23

    My intention was to feed my sub from the Mono bus.
    Since I have to move to a different layer, just for that handle, I thought it would be cool to assign Mono to a DCA.
    I couldn't figure out a way to do it, so I used a MixBus (instead of Mono) and assigned it to a DCA.
    Is there a way to assign Mono to a DCA, do I have to do it the way I ended up doing it, or is there an even better way?
    Thanks...
    JJ

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    0 279
    • doubleJ
      ChaseMcKnight

      Hi @doubleJ ,

      Matrices, Main L/R, and Mono busses cannot be assigned to a DCA Group. Depending on what you're trying to achieve, you can select 'M/C depends on Main L/R' under Link Preferences in SETUP. This will link the Mono buss level to the Main L/R fader (Turn down the Main, you turn down the subs as well). Otherwise, using a MixBus is your best bet. 

      • August 23, 2019
    • doubleJ
      doubleJ


      @ChaseMcKnight wrote:


      Hi @doubleJ ,

      Matrices, Main L/R, and Mono busses cannot be assigned to a DCA Group. Depending on what you're trying to achieve, you can select 'M/C depends on Main L/R' under Link Preferences in SETUP. This will link the Mono buss level to the Main L/R fader (Turn down the Main, you turn down the subs as well). Otherwise, using a MixBus is your best bet. 





      Thanks for the info.
      I did see the Depends option, but I still want to have manual control.
      MixBux it is.
      JJ

      • August 23, 2019
    • doubleJ
      RexBeckett

      @doubleJ 


      You could use a custom layer to add the Mono bus fader to your normal channels. On X32 Edit, click the User 1 button on the left and then click Edit to define the Channels, Returns, Matrix, LR, M/C etc that you want. If you are using a tablet, Mixing Station MX32 also supports custom layers.

      • August 23, 2019
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