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  • CaioCosta
    Contributor - Level 2
    2018-07-15
    I'm able to hear my voice through Direct Monitor, however my microphone does not work on Discord, Google Hangouts or Skype, even though it works on Reaper.
    I'm using Windows 10 64bit, a BM-800 mic connected to a U-Phoria UM2 audio interface via XLR.
    I have the latest drivers installed and have already reinstalled them as well as rebooted my PC many times.
    I've already checked the audio configurations in the mentioned programs and made sure the correct input was selected.
    I really don't know what else should I do.
    read more...
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    • CaioCosta
      Michael_Lapke

      Caio Costa;149112 wrote:

      I'm able to hear my voice through Direct Monitor, however my microphone does not work on Discord, Google Hangouts or Skype, even though it works on Reaper.

      I'm using Windows 10 64bit, a BM-800 mic connected to a U-Phoria UM2 audio interface via XLR.

      I have the latest drivers installed and have already reinstalled them as well as rebooted my PC many times.

      I've already checked the audio configurations in the mentioned programs and made sure the correct input was selected.

      I really don't know what else should I do.




      Hi Caio,



      Have you tried changing the sampling rate of your UM2 to 44.1kHz in the ASIO4ALL driver control panel? A number of the programs you list (Hangouts, Skype etc) may only work with audio provided in that format, so maybe that's why it's not working the way you expect when using those programs?
      • July 16, 2018
    • CaioCosta
      CaioCosta

      Michael Lapke;149189 wrote:

      Hi Caio,



      Have you tried changing the sampling rate of your UM2 to 44.1kHz in the ASIO4ALL driver control panel? A number of the programs you list (Hangouts, Skype etc) may only work with audio provided in that format, so maybe that's why it's not working the way you expect when using those programs?




      Hey, Michael!



      There's a "LINE IN" in the Recording tab and it's already set to 44.1kHz
      • July 19, 2018
    • CaioCosta
      Michael_Lapke

      Caio Costa;149365 wrote:

      Hey, Michael!



      There's a "LINE IN" in the Recording tab and it's already set to 44.1kHz




      Not sure that has anything to do with your UM2 interface, are you using the ASIO4ALL driver which is provided on our website?
      • July 19, 2018
    • CaioCosta
      CaioCosta

      Michael Lapke;149374 wrote:

      Not sure that has anything to do with your UM2 interface, are you using the ASIO4ALL driver which is provided on our website?




      Yes, I'm using those drivers.

      Have tried different versions of it already, like 2.13 and 2.14. The problem remains...



      People from Discord once "fixed" my problem by giving to me access to a beta version of their problem which worked for some time, but I don't think the problem was really in Discord because I also have this problem and other programs, as mentioned.



      Is there any test I could perform to check my audio interface? I bought it used, by the way...
      • July 21, 2018
    • CaioCosta
      Michael_Lapke

      Caio Costa;149469 wrote:

      Yes, I'm using those drivers.

      Have tried different versions of it already, like 2.13 and 2.14. The problem remains...



      People from Discord once "fixed" my problem by giving to me access to a beta version of their problem which worked for some time, but I don't think the problem was really in Discord because I also have this problem and other programs, as mentioned.



      Is there any test I could perform to check my audio interface? I bought it used, by the way...




      You could try to connect your interface to another PC and see if the behavior continues.



      Can you please confirm (with a screenshot) which device on your computer is set as the microphone input? Can you also show me what you have selected as your Discord audio input source?
      • July 23, 2018
  • EddieFranklin
    Contributor - Level 2
    2017-02-03
    Why is that? According to the system data specifications, the umc404hd has a dynamic range of 100dB, A-weighted, while the umc204hd has 110dB. That is a pretty big difference. Why is that so? What have you done to the umc404hd to make it's quality so much worse? Not that it's bad, 100dB of dynamic range is ok, but 110dB is better.

    Thank you.
    read more...
    0 7,997
    • EddieFranklin
      EddieFranklin Nobody?



      I've been trying to guess why, but I don't know. My only guess is that the analog circuitry is in fact different. It is of inferior quality of the umc204hd.



      Am I correct? Any Behringer engineers around to backup my theory or prove I am wrong?



      I'm just being curious. I am the sort of guy that opens up his hardware to do some mods here and there (even if I void the guarantee). I have a couple of mods I want to perform on my umc404hd. The first being the substitution of two of the inserts for headphones out-puts, which I find much more useful.



      Can anybody explain the difference in sound quality between the umc404hd and the others? I do not have a umc204hd to compare.
      • February 7, 2017
    • EddieFranklin
      Michael_Lapke Hi Eddie, welcome to the forums!



      My apologies for not getting back to you sooner as I noticed your posts last week but I had to do some research to find the answer to your inquiry.



      Here's what our U-PHORIA UMC product developers had to say in this regard:



      "If comparing the smaller (UMC202HD/UMC204HD) to the bigger model (UMC404HD), Digital-to-Analog Out (signal from a DAW), the dynamic range is quite the same.



      If comparing from Analog In to Analog Out (via Direct Monitoring) the noise floor on UMC404HD is slightly higher due to the more preamps/channels, so the dynamic range in turn is a little bit lower. Our department measured a difference of approx. 6 dB, not 10 dB as mentioned in the QSG, but this might be caused by a different measurement method used to determine the specs listed in our QSG.



      However, please know that the different code chips specifically have no influence on the dynamic range.



      Long story short:

      UMC404HD has more channels > higher noise floor > resulting lower dynamic range"



      Hope this helps clarify things for you. Please let me know if there is anything else I can do to assist you here in the forum.
      • February 13, 2017
    • EddieFranklin
      EddieFranklin Thank you for your answer. It's more or less what I suspected, more channels imply more components, and more components may mean more noise.

      I am very happy with the umc404hd. I have been using quite intensely as a portable sound card for broadcast applications, and the only problems I see are it's weight (it's a wee bit heavy) and the non-existance of the xlr latch. Apart from that, it's a fine little thing.

      Thank you for your answer again.
      • February 23, 2017
    • EddieFranklin
      DimiPana

      Eddie Franklin;94217 wrote:

      Why is that? According to the system data specifications, the umc404hd has a dynamic range of 100dB, A-weighted, while the umc204hd has 110dB. That is a pretty big difference. Why is that so? What have you done to the umc404hd to make it's quality so much worse? Not that it's bad, 100dB of dynamic range is ok, but 110dB is better.



      Thank you.






      Reading this thread, I am so glad I did not go for the four input interface and convinced myself that the 2 IN 4 out UMC204HD is pretty much ALL I will ever need, plus the dynamic range is phenomenal and the best for its class among Behringer devices I mean. Somehow the 204 is the sweet spot and I am glad it's the one I got.
      • April 9, 2018
  • DavidStrausberg
    Contributor - Level 2
    2014-04-27
    I am finding the RTA function works well enough to pinpoint feedback quickly, but curious if anyone is using this combination and their thoughts.
    0 7,948
    • DavidStrausberg
      Paul_Vannatto Hi David,



      I'm contemplating the same thing. I'm using a couple of the older DSP1124 Feedback Destroyers on my monitors, simply because it was a carry-over from my analog setup and haven't yet made the transition to properly setting up the monitors. Now with the built-in RTA on both the Rack screen and the Mixing Station, I'm seriously considering it. Unfortunately, I have a couple of weeks in Winnipeg (visiting my sons). So I won't be able to try it out until the end of next month. If you do try it out before that, please post your results. I would think it best to leave them inline and just bypass them. So if you are unable to get the monitors under control, you can revert back to the destroyers.



      Paul
      • April 28, 2014
    • DavidStrausberg
      DavidStrausberg If one comes up on ebay on the cheap, I may try it out.
      • April 28, 2014
    • DavidStrausberg
      KeithBroughton

      haven't yet made the transition to properly setting up the monitors.


      I think this is the key point.

      A well set up monitor system doesn't need automatic feedback control.

      Just my opinion....
      • April 28, 2014
    • DavidStrausberg
      DavidStrausberg Sure Keith, I think both Paul and I agree with that statement 100%, but I think we are also talking about other situations as well. I will have presenters walk into the audience directly in front of FOH speakers after giving them an education on feedback. I have my systems go on rent to inexperienced users at churches. A band member will decide during the 1st song his guitar amp was too low and crank it. I have given control of a musician's iem mix to the musician with a remote surface. Or maybe I had something happen in my life that has got me to the gig with little time to spare. These are just a few reasons a feedback suppression unit might come in handy. Do I use one now, no. Do I tune my rooms, yes. If a suppression unit comes up on ebay cheap might I try it out, I think I will. Because as much as I like to think I am in control of the universe, sometimes at the most inopportune moment, it likes to prove me otherwise and I like to be prepared for those moments as much as possible.
      • April 28, 2014
    • DavidStrausberg
      Paul_Vannatto Yes I do agree David.



      For me, it is usually the case of not being given enough time (usually 2 hours or less) for load in, setup, ringing out FOH, sound check including monitor mixes, dealing with promoters and roadie wannabe's. And I work by myself most of the time. So I have relied on the Feedback Destroyers to take care of the monitor feedback issues. That said, I also agree with Keith that I need to learn to rely less and less on that "crutch". With the Mixing Station now having RTA and the ease of adjusting the parametrics, I really don't have any excuse not to at least try and do it right.



      Paul
      • April 29, 2014
  • BartMitchell
    Contributor - Level 2
    2013-03-19
    I have a question regarding getting the vocals to sound right coming through the P16 monitors. We just bought the X32, S16 and the P16 monitors. Installation (for someone that doesn't know sound) went very well, and all is working great! We are using SE425 Shure in-ear monitors, and none of the band has ever used them before. So we are VERY inexperienced! My question is how do we get the vocals to sound like it did before when we were using wedges. We have a good size sanctuary, so we get natural reverb in our vocals. If I just put one in-ear in, and leave the other out, the mix is great in my one ear from the monitor, and I can hear the rest of the band, the audience and my vocals as it normally sounded through the ear without the IEM, which is what I am looking for. But I've heard that if you do this, you can damage the ear with the IEM because of turning up the sound (which of course I did so I could hear the mix). I know that using the X32, you have different reverbs available, but I have no idea which one is best. I am currently trying to use the Hall Reverb, but it sounds terrible because I don't know how to set it to be like the natural reverb in the room. So here's the question:
    - Which reverb effect on the X32 is best for vocals and what should the settings be based on the size room we have so it sounds like it used to? Is there anyway to find out what each adjustment does on the different reverb effects so we can properly set it?

    Thanks everyone!
    Bart
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    • BartMitchell
      JohnDiNicola

      Bart Mitchell;3931 wrote:

      I have a question regarding getting the vocals to sound right coming through the P16 monitors. We just bought the X32, S16 and the P16 monitors. Installation (for someone that doesn't know sound) went very well, and all is working great! We are using SE425 Shure in-ear monitors, and none of the band has ever used them before. So we are VERY inexperienced! My question is how do we get the vocals to sound like it did before when we were using wedges. We have a good size sanctuary, so we get natural reverb in our vocals. If I just put one in-ear in, and leave the other out, the mix is great in my one ear from the monitor, and I can hear the rest of the band, the audience and my vocals as it normally sounded through the ear without the IEM, which is what I am looking for. But I've heard that if you do this, you can damage the ear with the IEM because of turning up the sound (which of course I did so I could hear the mix). I know that using the X32, you have different reverbs available, but I have no idea which one is best. I am currently trying to use the Hall Reverb, but it sounds terrible because I don't know how to set it to be like the natural reverb in the room. So here's the question:

      - Which reverb effect on the X32 is best for vocals and what should the settings be based on the size room we have so it sounds like it used to? Is there anyway to find out what each adjustment does on the different reverb effects so we can properly set it?



      Thanks everyone!

      Bart




      Hello Bart,

      The problem you are having is common when using IEMs, since they tend to block the user from hearing most ambient sounds. This problem can be solved one of two ways.



      You could use a reverb as you suggest. If set properly, you can create a pleasing vocal sound in the IEMs using the many reverb choices on the X32. If doing so, I would assign the Direct Out of the FX return to a separate P16 channel from the dry vocals to allow the performer to create the blend they want.



      The problem with this approach is that it only solves part of the problem. While the reverb creates the impression of being in a larger space, it doesn't help with making a performer hear and "feel" the actual space they are in. For this reason, the ideal solution is to setup ambient microphones on your stage.



      This can be a single mic but is often a stereo pair of microphones placed on each side of the stage. Directional small-diaphragm condensers, like the BEHRINGER C-2s, C-4s, or B-5s, are often used for this purpose. For more placement tips and mic suggestions, google search "ambient mics for IEM" to find some great articles / forum threads. Because of feedback, be very careful NOT to feed these microphones to the Main LR mix, floor wedges, or any live speaker in the room. Where you would send them is directly to P16 channels, where they can be blended into the mix individually by each performer. These ambient signals are also useful for sending to recording mixes, video feeds, or as part of a multitrack recording.



      Hope it helps!



      Best,

      John DiNicola

      Senior Specialist, Product Support

      MUSIC Group

      BEHRINGER
      • March 19, 2013
    • BartMitchell
      BartMitchell

      John DiNicola;3933 wrote:

      Hello Bart,

      The problem you are having is common when using IEMs, since they tend to block the user from hearing most ambient sounds. This problem can be solved one of two ways.



      You could use a reverb as you suggest. If set properly, you can create a pleasing vocal sound in the IEMs using the many reverb choices on the X32. If doing so, I would assign the Direct Out of the FX return to a separate P16 channel from the dry vocals to allow the performer to create the blend they want.



      The problem with this approach is that it only solves part of the problem. While the reverb creates the impression of being in a larger space, it doesn't help with making a performer hear and "feel" the actual space they are in. For this reason, the ideal solution is to setup ambient microphones on your stage.



      This can be a single mic but is often a stereo pair of microphones placed on each side of the stage. Directional small-diaphragm condensers, like the BEHRINGER C-2s, C-4s, or B-5s, are often used for this purpose. For more placement tips and mic suggestions, google search "ambient mics for IEM" to find some great articles / forum threads. Because of feedback, be very careful NOT to feed these microphones to the Main LR mix, floor wedges, or any live speaker in the room. Where you would send them is directly to P16 channels, where they can be blended into the mix individually by each performer. These ambient signals are also useful for sending to recording mixes, video feeds, or as part of a multitrack recording.



      Hope it helps!



      Best,

      John DiNicola

      Senior Specialist, Product Support

      MUSIC Group

      BEHRINGER








      John:



      Thanks so much for the quick reply! I do have an ambient mic setup on stage. I believe it is an omni-directional mic though, not a directional one. Is that bad? It is routed only to the P16 monitors. The problem is that when I turn it up enough to hear it the sound is too muddy. Meaning the clarity of what I'm hearing from the band in the P16 channels goes away due to the blend with what is coming from the mic. I believe this is in part due to the natural reverb in the room, and the ambient mic picking up the sound of the band and it having a slight delay, but not sure about that. Could this also be because of the placement of the mic? We have a raised wooden stage, about 3 feet off the floor. We have steps in the center of our stage. The mic (if you are facing the stage) is mounted to the right of the steps about half way up before you reach the top of the stage, so about 1.5 feet from the floor. I drilled a hole into our stage so that the mic could be hidden somewhat. This was the only place I could find to place it. To the left of the stage I have my subwoofer, so that wouldn't work. I also have two speakers sitting to the left and right of the stage on the floor for the people that sit up front so they can hear, as we run a Mono set of 5 speakers dead center hanging down from the ceiling. Our ceiling is so tall that I would have to drop a mic 20' probably to get it close enough to the people so we could hear them so that wouldn't work either. Been wanting to move to a stereo sound but haven't done it yet. I'll do a search for placement as well as check out the mics you guys have.

      Right now I am using the "Hall Effect" for reverb, as well as using the FX1L in the P16 channel for my voice. But I am not also running my voice without the effect as a separate channel. And because I have no clue how to adjust the effect, it sounds terrible to me. Could you send me a "how to" on what each knob does for the reverb effects? I'd also like to try using reverb in the recording bus for the recording mix we have. Thanks again!

      Bart
      • March 19, 2013
    • BartMitchell
      InactiveUser Hello Bart,



      My name is Evan Hooton and I am the House of Worship Product Support Specialist for BEHRINGER. I have been reading your posts and I think it is great that you have decided to make the change to the P16 system. There are so many great things about having this system in a House of Worship and I believe that once you get more familiar with optimizing your setup that you will really be happy with the move.



      If I may, I have a few different scenarios for you and your ambient microphones situation.



      •I would really recommend purchasing a pair of the Directional condenser microphones such as the ones that John had mentioned. The “Muddy” sound that you are having can certainly be attributed to the omni-directional microphone which you are currently using. You want to block out as much of the rear stage volume as possible and focus on the congregation.

      •Placing the directional condenser microphones on the edges far edges of the stage is certainly a great place to mount them. I would recommend purchasing shock-mount clips for these microphones, which will help to further isolate these microphones to only pick up the noise that is in the direction that they are being pointed.

      •Although the goal of boundary microphones is to catch as much of the natural room ambience as possible for reference, I do recommend placing a high pass filter (also referenced as a Low Cut on certain mixing consoles) on these microphone channels to at least 120Hz if not up to 170Hz which will help to clear up the source that you are trying to pick up. If you have the ability to select your frequency point.







      Editing the Hall Reverb FX unit:



      You can find the description of the FX units within the X32 and what each slider does on page 21 of the X32 user manual, which can be downloaded from the X32 product page on behringer.com I have typed up the description below for you to take a look at.



      •The PRE DELAY slider controls the amount of time before the reverberation is heard following the source signal.

      •The DECAY controls the amount of time it takes for the reverb effect to dissipate.

      •The SIZE controls the “perceived” size of the space being created by the reverb effect. This is one of the key controls to set up a reverb, which is as close to the natural reverberation of your room.

      •The DAMP slider adjusts the decay of the high frequencies within the reverb hall. This can also help with creating the “open air” effect of a natural room reverb.

      •The DIFF control or DIFFUSION controls the initial reflection density.

      •The SHAPE adjusts the contour of the reverberation envelope.



      The last two controls help to create the tone of the reverb itself.



      I hope that my advise have helped you get closer to achieving what you are looking to do. Please feel free to message me through the forum or on this post if you ever have any questions about setting anything up in your facility. If possible and you have time, I would really like to get a few pictures of your sanctuary and your X32 and P16 system.



      Looking forward to talking more with you, Bart.



      Evan Hooton

      Specialist, Product Support

      MUSIC Group

      BEHRINGER
      • March 20, 2013
    • BartMitchell
      BartMitchell Hi Evan thanks for the advice. I will double check my ambient microphone that I am using to be sure it is the right kind. I am the one that had contacted you over 3 months ago about coming out to our church for the installation & setup of our system. We bought the entire system from Behringer. You had said that you would be able to come and help me set up the X32, the S16s, and the P16s, but it never happened due to your schedule. I have since tried to contact you multiple times with no results. I would still really love to have you come out if that was possible. I sent you another email 3 days ago just to see if you were available, and Chase McKnight has sent you a couple of emails as well but never heard from you. I see from your post that you are in Colorado Springs. I am only 4 hours away from you. Please email me back if possible by checking your email or let me know if there is a better way to communicate with you. Thanks again for the info. By the way, Chase on your support team has been INVALUABLE in helping me over the phone with getting the system setup. I know nothing about sound and he's been great. He is always available and if not calls me back very quickly. We still have a long way to go with properly adjusting the X32 for the best sound. The sound is very flat, and doesn't seem to have the same dynamics as our old analog Allen & Heath board. Everything is working great, but the issue with the IEMs and the overall sound quality we are disappointed with. I know this is probably due to our inexperience in using the system, hence why I've been hoping to set up a time for a visit from you. I am to the point now to where I am going back to wedges, because we simply cannot worship with the sound as is from the P16s. Thanks again for the help, and hope we can connect soon!

      Bart Mitchell

      Centerpoint Church
      • March 21, 2013
  • RyanHammond
    Contributor - Level 2
    2014-10-04
    I wonder why there's not a whole lot of talk about the Turbosound Siena series? These look like very serious speakers! The 15s are capable of 136dB, are wood, light, and have Ultranet functionality. I would love to hear these boxes!

    http://www.turbosound.com/docs2/range_intros/Siena.shtml
    read more...
    0 7,839
    • RyanHammond
      Paul_Vannatto Hi Ryan,



      I think these are the resurrected line of speakers from the Behringer iQ series which never launched. My understanding is that they won't be available til at least the end of the year. But I've heard nothing more.



      Paul
      • October 4, 2014
    • RyanHammond
      PatrickGMaillot ...136db :CALCULATED - Let's wait for actual measurements.
      • October 4, 2014
    • RyanHammond
      RyanHammond

      Patrick-Gilles Maillot;31894 wrote:

      ...136db :CALCULATED - Let's wait for actual measurements.




      Pretty much everything's calculated except for a select few who actually measure using pink noise at 1 meter dBA slow lol. Mackie does measurements with their HDA series, but the calculated number gives you something to compare with other speakers to see how well they stack up.
      • October 5, 2014
    • RyanHammond
      PatrickGMaillot Agree with you Ryan, but what surprises me is the Siena is the only ones from Turbosound with such a high db. All other speakers, including the ones with the same class D amp and 15" driver are rated at around 128db, which is already *very* loud.



      The Madrid series, using the same Carbon fiber loaded 15" driver is rated at 800W continuous (3200 peak) which is in line with the announced 2500W class D amp for the Siena, and is rated 126 dB continuous, 132 dB peak. The driver itself seems to be at 97db/1w/1m.



      Don't get me wrong, The Siena sure looks like great speakers. I hope I can get my hands on them at some point.



      -Patrick
      • October 5, 2014
    • RyanHammond
      RyanHammond

      Patrick-Gilles Maillot;31911 wrote:

      Agree with you Ryan, but what surprises me is the Siena is the only ones from Turbosound with such a high db. All other speakers, including the ones with the same class D amp and 15" driver are rated at around 128db, which is already *very* loud.



      The Madrid series, using the same Carbon fiber loaded 15" driver is rated at 800W continuous (3200 peak) which is in line with the announced 2500W class D amp for the Siena, and is rated 126 dB continuous, 132 dB peak. The driver itself seems to be at 97db/1w/1m.



      Don't get me wrong, The Siena sure looks like great speakers. I hope I can get my hands on them at some point.



      -Patrick




      Yeah, I saw that too which struck me as very interesting. They also spec the Milan 15 with a peak of 130, but the continuous SPL is 125—as measured by Sound on Sound (March 2012). Even thought the M15 uses a smaller amp than the Siena or IQ 15, both the Siena and IQ use the same amplifier—yet the IQ is 128. I assume that the driver sensitivities are similar.



      I wonder if they meant to say the IQ continuous SPL is 128—not the peak? That would make a lot more sense. I think we might be seeing some typos here. Still, even 125dB continuous is VERY loud... and 128 gives you a bit more headroom



      *edit*



      I did see some frequency response differences:



      Siena 15: 58 Hz - 17 kHz ±3dB, 43 Hz - 20 kHz ±10 dB

      iQ 15: 50 Hz - 18 kHz ±3 dB, 45 Hz - 20 kHz ±10 dB



      Perhaps the power is distributed "better" with the Siena's frequency response? Also, it doesn't say whether or not these speakers are used with subs—which as we all know, greatly enhance SPL efficiency.
      • October 6, 2014
  • JamesBurns
    Contributor - Level 2
    2012-07-12
    For technical and warranty support for your BEHRINGER product, the best way to contact us is through our CARE department.

    You can contact us by phone at (1) 702-800-8290 for the US and Canada and (44) 1562 732290 in Europe.

    You can also contact our CARE department via email at [email protected]. We try to respond to every email within 24-48 hours, but please be patient if it takes longer due to the volume of emails we receive.

    We hear you!37244.jpg
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    0 7,838
  • SteveLynch
    Contributor - Level 2
    2013-12-04
    I'm waiting in breathless anticipation for any news on the line array. I saw the new video where you used it up in washington, yet no one outside of Behringer, anywhere has heard this system. I also noticed that you guys pulled it off the site so I an't even visit that page to fantasize while I wait for it's release.

    I know the question is asked a lot, but Why won't anyone from Behringer even acknowledge any questions about the system?

    Is there any tentative update? Any additional info?

    Saying Coming Soon, then going silent is not cool!

    Zillions of folks want to know what's up!

    Got anything at all for us?

    At least say why you removed it from online?
    read more...
    0 7,678
    • SteveLynch
      SteveLynch Ugh! Really? No response at all?



      We can get Uli Himself to respond to a thread about replacement knobs, but no one at all will respond to multiple inquiries about a new highly hyped product??



      hmmmmmmmm.
      • December 5, 2013
    • SteveLynch
      BrianMonroe Alright peppy. Calm down... the employees that are assigned to answer questions on the forums have other full time responsibilities, and you didn't even give them an full day to answer. I've waited up to two weeks (longer than a month if you count the linux app) to get a response back. There's a good chance that the ones on the forum aren't involved in that project, and will have to track down those that are to get answers for you.



      Patience is the key to success.



      Give it a few days, then, if we still haven't heard anything, I'll complain about timeliness with you! =)
      • December 5, 2013
    • SteveLynch
      SteveLynch

      Brian Monroe;12338 wrote:

      Alright peppy. Calm down... the employees that are assigned to answer questions on the forums have other full time responsibilities, and you didn't even give them an full day to answer. I've waited up to two weeks (longer than a month if you count the linux app) to get a response back. There's a good chance that the ones on the forum aren't involved in that project, and will have to track down those that are to get answers for you.



      Patience is the key to success.



      Give it a few days, then, if we still haven't heard anything, I'll complain about timeliness with you! =)




      Sorry, I staff a couple of other high volume forums, and if I don't respond within an hour my customers scream. Guess I just have a different interpretation of "timely".



      A company this size should have floaters checking the forums hourly at minimum.



      But wait I will!
      • December 7, 2013
    • SteveLynch
      JohnDiNicola Dear All,

      My apologies for the delay. I am looking into this now and will follow up here. Thank you for your patience.
      • December 10, 2013
    • SteveLynch
      BillYekel --Speculation--- I bet they are waiting for some new design tweak they brought over from Turbosound, or there is a delay because Turbosound is going to release some new product and they don't want competing sales.. ---Speculation--- Please, discuss amongst yourselves while we wait for the official approved answer...LOL!
      • December 11, 2013
  • PeteHanratty
    Contributor - Level 2
    2013-06-24
    Hi,

    We are about to purchase our X32 for church and I was wondering how you are all using the assignable control section, if at all.

    I would guess at the least turn efx on/off for speaking between songs.

    Any other ideas?

    Pete Hanratty
    Columbus OH

    ps. I got to meet Evan Hooton at Sweetwaters GearFest last Friday!!!
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    0 7,359
    • PeteHanratty
      Paul_Vannatto Hi Pete,



      Glad to hear you are in the process of getting the X32 for your church. We got ours in December and are very happy with it. My wife and I visited your lovely city last May, touring the downtown on a segway and visiting the famous Columbus Zoo, then off to the Amish area for a few days.



      The only thing I use the assignable buttons for is check the L/R (FOH) EQ on FX slot 5. The reason for this is that there is a known bug in XControl (and still in X32-Edit) in that once in awhile the EQ will be reset to flat. Hopefully they will fix this bug soon. The other thing I have is to goto the Routing Home page in order to switch the Aux In 1-4 between Aux 1-4 and Card 1-4. Oh and one other thing I changed. I change the 3rd and 4th button on set A to something other than Fx Time and Speed. The blinking light drove me nuts.



      Paul
      • June 24, 2013
    • PeteHanratty
      InactiveUser

      Pete Hanratty;6636 wrote:





      ps. I got to meet Evan Hooton at Sweetwaters GearFest last Friday!!!




      Pete,



      Glad we got to cross paths at GearFest. It was great to talk with you. Just wanted to let you guys know about two of the new webinars that are happening this month. One is on implementing the ProZone units (X32, S16, P16) units into your House of Worship and how they all go together, and the other one is setting up MixBusses, Mute Groups, DCAs, Subgroups and all of that good stuff on the X32. Here is the link to get to the page to sign up for them if you are interested.



      Also, How did you like GearFest?



      http://forum.behringer.com/showthread.php?1378-House-of-Worship-Live!-Webinar-amp-User-Group-July-2013



      Best Regards,

      Evan Hooton

      Specialist, Product Support

      MUSIC Group

      BEHRINGER
      • July 6, 2013
    • PeteHanratty
      Paul_Vannatto Hi Evan,



      Your webinars look very interesting. Unfortunately, it is during my workday. So I may be able to listen in, depending on the interruptions. Hopefully you will be posting them here shortly after the events.



      Paul
      • July 6, 2013
    • PeteHanratty
      JosephSchuurman

      Pete Hanratty;6636 wrote:

      Hi,



      We are about to purchase our X32 for church and I was wondering how you are all using the assignable control section, if at all.



      I would guess at the least turn efx on/off for speaking between songs.






      We haven't made full use of the assignable controls yet, but we've setup Set A to be effects with the following assignments:

      dail: MixBus Levels (allows us to dail up and down the overall effects sends for that bus)

      button 1: time (if applicable, lets us easly tap the tempo for a given song, to make the hits in delays to line up nicely)



      I'd love to hear how other people are using them as well.
      • July 17, 2013
    • PeteHanratty
      JamesMonroe Hi Pete, I use the "User Assignable" section of the X32 every service for both eFx and scenes. On bank "A", I use the rotary control for eFx 1~4 "Level" control so I can have instant hands on control of the eFx level depending on the type/speed/style/instrument/voice of the song. Button 1~4 I have assigned to eFx 1~4 mutes for quick on/off. Buttons 5~8 I have assigned scenes 1~4 for easy access since I primarily use only 3 scenes each service. Our order of service changes from service to service and I don't always have time to set the scene bank in order of how the service is ordered. I have bank "B" rotary controls set to efx 1~4 size or modulation. Buttons 1~4 are set to eFx 1~4 tap delay. Buttons 5~8 are still set to scene 1~4. Bank "C", I have yet to use, however I still programmed them to eFx 1~4 pre-delay on the rotary control. I have scenes 1~8 on buttons 1~8. This works great for me. Recommend just playing around to see what configuration works best for you and the audio team. Hope this helps!!! J. Monroe
      • July 18, 2013
  • LeslieFord
    Contributor - Level 2
    2014-08-25
    Good Morning X32 H.O.W. Family! Welcome to my 1st post!!

    My statement of usage and questions are as follows:
    I recently purchased an X32 and X32 Rack. These units will replace a wonderful Allen & Heath GL4800 in our environment. Here is where my issue begins. My GL4800 has 48 mic pre's, of which we use about 40 regularly. I understand that the only way to control more than 32 mic pre's w/the X32, is to sync two X32's together. My goal is to sync these devices together and use the Rack as an instrument in and Monitor board (as it will be on stage), and the X32 as FOH, effectively giving myself 48 "controllable" mic pre's. (right?)

    While I believe the information I received on the sync'ing of two X32's together to be correct (as it came from John DiNicola of Music-Group), I still wonder exactly how I can "see" and "control" the Rack from the FOH? Although I only have the X32 Edit app to go on right now (the X32 will be delivered today, and the Rack in two weeks), the standard "Routing" options only cover Inputs 1-32 in the "Channel Processing Block Patch"

    Now for the ?'s:
    1. To confirm, am I correct in the thinking that I will be able to simultaneously control (not just "sync") 48 mic pre's in this manor?
    2. If so, I see where the Inputs 1-32 section shows AES50 33-40, 41-48, etc. Is this where I assign the additional inputs from the X32 Rack? (i.e. the X32 will be local 1-32 and the X32 Rack will be AES50 33-48)?
    3. If my statement in ? #2 is incorrect, how then do I connect, see, and control the additional in's/out's of the Rack? The X32 Routing only states 1-32, and AES50 33-48 are simply places to route inputs 1-32... I can't figure it.
    4. Even if my assumptions made above about routing are correct, I still don't see where I will be able to access the additional rack simultaneously. Once the Rack arrives and I sync it, will it appearing in the "Connected Devices" area under "AES50 A" then give me additional "Routing" options?

    Please forgive me in advance as I feel all my questions kind of circle the same issue... So Maybe I should have changed question #1 to be... "Can I do this, and if so HOW?!?!?!?"

    I am a seasoned Audio Engineer, but seem to hit a wall in my understanding as it pertains to routing this small form factor unit.

    Thank you in advance for your time, assistance, and patience w/my long winded description of my issue! I simply want to provide as much info as possible that I may receive the most informed response.
    Please advise...
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    0 7,355
    • LeslieFord
      Paul_Vannatto Hi Leslie,



      Welcome to the forum. Unfortunately, I think you have misunderstood something here. The X32 can only process 4 banks of 8 inputs, plus the Aux In (bank of 6) at any one point in time. So the only way to handle more than the 32 preamp channels (plus the 6 Aux Ins) is to use the X32 Rack as a submix and send the results to the X32 FOH via one of the banks of 8.



      :edit: The other option is switching the banks of 8 around using scenes, snippets or routing presets, still controlling the 4 banks of 8 (plus Aux In).



      Paul
      • August 25, 2014
    • LeslieFord
      LeslieFord So I'd lose 8 in's/pre's on the X32?



      If I'm understanding you correctly, I need to submix the 16 Rack channels down to 8 (or fewer) from the Rack itself, then send those to the X32 via AES50. Then I'd only have control of their overall, not individual volume, correct?



      I wanted to get individual instrument channels in to the X32 off of the Rack (like an S16 would) so I'd have full control of them for the house, BUT, I didn't want to lose those 13-15 in's on the back of the X32 as I need them elsewhere. Also, I wanted the Rack to handle the sends for the in-ear mix for the musicians (6 mixes total) and possibly an additional 2-3 monitor sends.



      Please let me know if my original plan can work, and how. If it can't/won't, is there an alternative you can recommend? Otherwise, I have to re-evaluate this purchase! I need the option of 48 mic pre's as we regularly have concerts, plays, and conferences that require more than 32.
      • August 25, 2014
    • LeslieFord
      LeslieFord As a (not off topic) aside... How does sync'ing two X32's work for Monitor and FOH mixes?

      Does the Monitor console split the signal to the FOH via AES50 and the mixes are handled completely separate? (no control of the monitor console from FOH)

      The boards would also need two different routers as well for iPad control, right? As we normally use one engineer per service, said person would have to switch between wifi connections regularly to control both consoles.



      I used an X32 in this manor a couple weeks ago. It was the front end for the stage instruments & monitors, and split the signal to my GL4800. It worked GREAT for the conference!! But since the GL4800 is analog, I couldn't network them together (dante, AES50, etc.). It required the use of a 150 foot splitter snake, which was an eye-sore and not suitable for longterm use. I was hoping to get the same overall functionality, while upgrading to digital, w/o breaking the bank... I figured I'd get the two X32 units and use them in the manor mentioned above. Not so?
      • August 25, 2014
    • LeslieFord
      Paul_Vannatto Hi Leslie,



      To answer your questions fully will take a bit of explaining. Hopefully other members will jump in and help here. Even though there are limitations to the X32, there are also creative ways to get around those limits. When I get a moment (have some pressing home issues do deal with at the moment), I'll try and help explain these as best as possible.



      Regarding your question of syncing the two X32 as FOH and monitor, have a look at what I posted on the X32-Wiki regarding this. It may help, even though this explanation involves two Racks. Here is the link. http://x32wiki.music-group.com/index.php?title=X32_Rack_Specific_FAQ



      Paul
      • August 25, 2014
    • LeslieFord
      Paul_Vannatto Hi Leslie,



      Sorry for the delay. I had a chance to re-read your initial post as well as the other subsequent posts. To answer your overall question in simplified terms, the answer is yes, you can "simultaneously control (not just "sync") 48 mic pre's" - but not in the manner you are thinking. Let me start with some background of how the X32 is designed.



      Your observation is correct that on an X32 unit (full console, or any other model), only 32 mic pres can be controlled at any point in time. By default, these mic pre's are assigned to the channel strips 1-32. The sources of these mic pre's can be from a combination of various physical inputs, including the back of the console, S16's or other X32 units via AES50, ADA8200 via X-ADAT card, etc and are assigned in banks of 8 (Routing, home screen).



      You are also correct in observing that the AES50 assignment screens (Routing, AES50-a and AES50-b) shows 6 banks of 8, including the last 2 (Outputs 33-40, 41-48). Typically these are used to send signals to the Ultranet (P16). If the Ultranet bus is not used, these banks of 8 can be used creatively for other routing purposes.



      Here is where you need to "think outside the box". Even though each X32 unit can only control 32 mic pre's, each X32 unit has the ability to control its assigned inputs independently (almost - I'll explain later) using its own surface controls (eg X32 full console, Compact, Producer) or via one of the apps (X32-Edit, X32-Mix or Mixing Station). So for your application, the FOH engineer could use the X32 full console, controlling its assigned mic pres via its surface and/or one of the apps - AND control the mic pres (as well as the full mixing control) of the X32 Rack via one of the apps located beside the X32 full console. Using a wireless router for each X32 unit, as you suggested, is a very good idea.



      In order to get all signals where they are supposed to be going is going to involve some creative and complicated routing. But I do believe it is doable. We'd love to help you with that, whenever you get to that point.



      Do we assume that the X32 full console (FOH) will be located somewhere in the audience and the Rack located somewhere near the stage (typical setup)? If so, there may be an issue with the required outputs from the Rack for the IEM's and floor wedges, particularly if the IEM's are stereo. The Rack only has 8 XLR and 6 Aux Outputs (total of 14). If the IEM's are mono, there are enough outputs. If not, the Rack would need to be supplemented by another output unit (S16, ADA8200 and X-ADAT card, etc.).



      Hope that helps



      Paul
      • August 25, 2014
  • Paul_Vannatto
    Volunteer Moderator
    2013-03-11
    I have had my UltraCurve DEQ2496 for a few years now, and have never been able to use the automatic feedback control. So I have always had to use the old way of finding the feedback frequency with the RTA and notching that frequency on the graphics equalizer.

    This last weekend I realized that I only had 1.3 version of the firmware and was successful in updating that to 1.4, hoping that would fix the automatic feedback problem. But it didn't.

    I have all the PEQ set to automatic. During yesterday's jamboree, I monitored the graphics screen to see if any notching would occur, but to no avail. Can anyone help me figure out what I'm doing wrong, or what I've missed in setting this up?

    Thanks

    Paul
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    0 7,286
    • Paul_Vannatto
      ZacRoss Dear Paul,



      Use the steps below for programming the AUTOEQ function of the DEQ2496!



      1. Use an XLR mic cable to connect the ECM8000 reference mic into the RTA/MIC input on the back of the unit.



      2. Press the UTIL key to reach page 1 (GENERAL SETUP), use the upper small jog wheel to select ‘CHANNEL MODE', then use the large jog wheel to set the MODE to ‘DUAL MONO'. Press the B key to ‘ACCEPT MODE'.



      3. Press the I/O key to access page 1 of the I/O configuration.



      4. Use the big jog wheel to highlight the ‘MAIN IN' box and press the B key to accept the source.



      5. Press the I/O or PAGE key to enter the I/O page 2 and highlight the ‘WIDTH' box.



      6. Press the I/O or PAGE key to enter the I/O page 3, use the large jog wheel to highlight ‘RTA/MIC', and press the B key to highlight the +15v phantom power box (if it isn't already).



      7. Press the RTA key to go to RTA page 1.



      8. Press the B key (AUTO EQ), and you will enter AEQ page 1.



      9. The DEQ2496 will now request that you edit the target curve. Use the A key to select the ‘LEFT' channel. With the large and small jog wheels you can adjust the EQ bands to reflect the curve you want to end up with (it can be ‘flat' – without any adjustment). Press and hold the B key to reset if you feel it's necessary at some point to restart the curve adjustment.



      10. Press the PAGE key until page 2 (AEQ) is accessed. Highlight ROOM CORRECTION if desired.



      11. Adjust NOISE GAIN with the large jog wheel as desired.



      12. Adjust AUTO EQ to ‘slow', ‘mid', or ‘fast' with the lower small jog wheel for the Auto EQ calculation speed.



      13. Press the PAGE key until page 3 of the AEQ menu is displayed. The GEQ is shown with outlined virtual faders.



      14. Adjust ‘MAX' with the upper small jog wheel for the desired maximum difference (in gain) between adjacent frequency bands. A suggested starting point would be a value of ‘3'.



      15. Press the A key to START AUTO EQ.



      16. The DEQ2496 will measure the ambient (or room) noise for about 15 seconds.



      17. The DEQ2496 will now produce pink noise at its outputs and will analyze system noise for about 15 seconds. If you get a message that the ‘Pink noise level isn't high enough', turn up the pink noise level using the large jog wheel and restart the Auto EQ process or wait until a message appears that it is ‘Starting Auto EQ Process'.



      18. Once the Auto EQ process has begun you can view the ‘progress' from page 2 (RTA response) or page 3 (Graphic EQ changes). If necessary, the ‘MAX SPAN' and ‘MAX' values mentioned above can be changed in real time, so that you can see the Graphic EQ changes.



      19. Let this process run until it appears obvious that the adjustments begin to ‘settle' (or after about 90 seconds), then press the PAGE key until page 3 appears, and press the B key (DONE) to end the Auto EQ process. The unit will default to RTA pages 1-3.



      20. The DEQ2496 will have created the corrective adjustments to achieve the target curve that you specified earlier in this process. To view the Auto EQ changes, press the GEQ key. The settings will be visible and available for further editing.



      21. Repeat the process for the right channel on the unit.



      Note: In the beginning of the Auto EQ process, the DEQ2496 can be set to ‘STEREO LINK' mode (in the UTILITY menu) if you wish, so that the Auto EQ settings from one channel will be copied to the other channel.



      Hope this helps!







      Kind Regards,
      • March 12, 2013
    • Paul_Vannatto
      Paul_Vannatto Hi Zac,



      Thanks for that excellent and detailed instructions. It is much better than the manual for setting up the AEQ. I've printed that out and will use it for future events that I have adequate time for ringing out the room (which is rare).



      Unfortunately, that is not what I'm trying to get to work. I would like the Feedback Destroyer function to work. Sorry for wording my question wrong. I use 2 of the Feedback Destroyers (DSP1124) on the monitor feeds and they work wonderfully. I never have to worry about the monitors. But I'd like that same feature on the FOH using the DEQ2496. I always have the ECM8000 plugged into the RTA input. As I mentioned, I've had this setup for a few years now, and have never been successful in making this feature work. Could you help me with this as well?



      Thanks again



      Paul
      • March 12, 2013
    • Paul_Vannatto
      ZacRoss Hey Paul!



      There a few different ways to attain your goal. Here are a few ideas I think will help with programming the Feedback Destroy functionality.



      Using the Graphic Equalizer function on the DEQ2496:

      Press the GEQ button to display the GEQ menu.

      Turn the small Upper Data Wheel to select the frequency to work with.

      Turn the small Lower Data Wheel to select the bandwidth you want to affect.

      Turn the Large Data Wheel to adjust the GAIN of the frequency.

      For each control you will see visual feedback from the curve in the middle of the display and the numbers on the right of the display.



      Using the Parametric Equalizer function on the DEQ2496:

      Press the PEQ button several times to display page 1 of the PEQ menu.

      Turn the mall Upper Data Wheel to select the frequency to work with.

      Turn the small Lower Data Wheel to select the bandwidth you want to affect.

      Turn the large Data Wheel to adjust the GAIN of the frequency.

      For each control you will see visual feedback from the curve in the middle of the display and the numbers on the right of the display.





      Here are some other important settings to consider:

      SENS (upper data wheel) allows you to determine the point of onset for feedback suppression (describes the difference between feedback signal and overall level). When a signal reaches this difference, it get reduced in level. The setting range here is from -3.0 to -9.0 dB. Use the THRESHOLD (large data wheel) to select the threshold from which a certain frequency is considered to be feedback. The MAX. DEPTH paramtere below determines the maximim attenuation of a filter (-18 to -60 dB) in 6 dB steps, and thus the GAIN setting range as displayed on the first and second page (lower data wheel).



      Sens = sensitivity.

      this sets the trigger level (in dB below the overall signal level).

      -9 dB = most sensitive / -3dB = least sensitive

      If the feedback does not reach this level then no filter will be set.



      Threshold = this is the global threshold. Below this level there will be no filter set. This is to prevent the DEQ from setting spurious filters when they are not needed (eg when a child or soprano is "singing" a long note, or when a guitarist / keyboarder / flautist plays a long sustained note).



      In other words - before a filter is set, the level of feedback has to reach (a) the threshold and then (b) the sensitivity level.



      The level has little to do with the actual SPL - this is signal level measured at the input to the unit (ref the v/u meter or signal LEDs). If the unit is connected (eg) on an aux send, then it may be that the SPL is very high because the amps are set to maximum, even though the input level to the DEQ may be relatively low. Solution? increase the input level to the DEQ2496 and use the "gain offset" to bring the signal back down again before it goes to the amps.



      Even if the DEQ is setting the "best filters in the world" - you can't do much for bad mic placement. If you hold the mic right up to the speaker then feedback will continue to occur, regardless of whatever "notching" is taking place in the signal. You will need to get the mic positioning sorted out first before thinking about any feedback elimination.



      Kind Regards,
      • March 14, 2013
    • Paul_Vannatto
      Paul_Vannatto Thanks Zac,



      Yes I think it maybe the sensitivity that I don't have setup correctly. I'm printing this out and will go through all the steps you suggest. The reason why I asked the question is that the feedback destroyer function would never work at all. Yet the 2 DSP1124 worked flawlessly every time.



      I am very aware of bad mic placement, and I can assure you that is taken into consideration. The only event that I have a feedback issue is during a monthly jamboree, which is held in a fairgrounds hall. Unfortunately they have the stage and dance floor situated in the middle of the room facing the short side of the hall. And the accoustics is not the best. Where the real problem begins is when they stop the music and make announcements, talk quietly and expect me to amplify them sufficiently to cover the entire room of 250+ people. That's when I have to squeeze as much gain before feedback as I can. I do shoot the room ahead of time (quickly) to reduce the hall accoustic as best as I can. But a crowd will change that accoustic and that is where I need the feedback destroyer to help me with.



      Thanks again for your excellent explanation. I'm confident this will help me accomplish what I want to do.



      Paul
      • March 14, 2013
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