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  • striple66
    Contributor - Level 3
    2020-02-27

    Dear all,

    I am now somewhat familar with my new X32 Rack but I am missing physical XLR inputs. That's why I am considering buying a SD8 and a SD16 Digital Snake. The X32 Rack has only 16 physical XLR inputs and we need 20 Is it possible to connect a SD8 Digital Snake to AES50 A and a SD16 Digital Snake to AES50 B and have all 24 physical Digital Snake XLR inputs + the local 16 XLR inputs controlled by the X32 Rack? What I mean by "controlled" - mapped to the logical 40 inputs of the X32 Rack using the faders and channel strips on X-Touch and X32-Edi? Or am I missing here something?

    Thanks for you help

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    0 201
    • striple66
      DaveMorrison

      @striple66 


      You can use any of the 32 XLR inputs for the mixer's 32 channels + 6 other XLRs mapped to the auxiliary channels for a total of 38 channels. Aux 7 & 8 are "hardwired" to the USB recorder. Since the AES50 bus has 48 inputs and outputs, you could alternatively daisy-chain the SD16 and SD8 together and accomplish the same thing from just one of the mixers AES50 ports. 

      • February 27, 2020
    • striple66
      striple66

      Hi Dave, thanks a lot for your answer. In other similar (but not identical) questions and answers from other guys here in this community, there were often the recommendation to daisy-chain the two SD8 / SD16 Digital Snake boxes instead of connecting one to AES50 A and the other stage box to AES50 B. What are the implications on connecting both stage boxes in a daisy-chain manner and separating them on the two different AES50 ports on my X32 R? Second question: on the routing page on my X32 R, channels 17-24 and 25-32 are in [square] brackets, meaning that they are not physically existent on the console. This - I do understand But in the manual they say this:


      "Local inputs 17-24 and 25-32 are listed with brackets, indicating that these are not physically avaiable on this device. Assigning these will obviously carry no signal, but they still can be used when preparing shows for a full X32 console".


      What does this mean?


      Thanks and all the best

      • February 27, 2020
    • striple66
      GaryHiggins

      Hi @striple66 The implications are that ALL your aes50 inputs will be seen as Aes50 A, instead of 16 as A and 8 as B. (Or 8 as A and !6 as B). If your first connected stage box was the SD16, the first 16 ch's (sd16) would be aes50a 1-16 and the next 8 (sd8) would be aes50a 17-24. Using 2 different ports One would be aes50A 1-16 and the other aes50B 1-8. The outputs of either would be set by the switch at the end of each.


       


      It means that you could setup a scene at home with routing to those ch's included and when you get to a mixer that can load them (and has those physical inputs) they will work as intended.

      • February 27, 2020
    • striple66
      striple66

      Hi Gary, thanks a lot. But does this have any technical reasons in terms of "if you do it this way, you won't hear a thing in your console", or "it is not possible to do the routing then" if I choose one these connections alternatives? @GaryHiggins @DaveMorrison 

      • February 28, 2020
    • striple66
      DaveMorrison

      @striple66 


      I believe they put the brackets there to remind us we're using a full sized core which is missing half it's inputs and outputs. We could route signal there if we just had some way to add more inputs and outputs?. I used to think they were confusing, but now think they are totally unnecessary (and unnecessarily confusung).

      • February 28, 2020
  • DeclanSlater
    Contributor - Level 3
    2020-02-29

    A few things I'm wondering about on the WING... (Warning: long rant ahead!)

    I first must apologize for double posting, I mistakenly posted this first in the following link (portable live sound) - I created this post on my phone and posted it there by mistake.  I sure wish we were able to edit or delete the posts that we create.  Gary did respond to my post there.  I revised this post here to incorperate his thoughts.  Sorry for the double post. My bad.

    https://community.musictribe.com/t5/Live-Portable-Sound/WING-a-few-head-scratches/m-p/292852#M1641

     

    - TRIM: Why can we easily adjust the TRIM, right on the main page of any input channel, while we must push a button to be able to adjust an input gain? Why must we then have to hit the ROUTING button to gain access to phantom on the same channel?

    - Secret Mutes: Why is there a mute button located on the Preamp?  Why is that mute button seperate from the channel mute?  Why can we not control or monitor the status of that hidden mute button easily? I just triple checked this again, and yes - there is definitly a secret mute button there, that is NOT linked to the channel mute button in any way!  Switching between the MAIN and ALT channel input assignments has no effect on this "secret" mute button at all.  Under the CHANNEL INPUT section, in the MAIN input selection box, is one location of this "hidden" mute button.  Press the ROUTING button, and then pick the correct preamp that is currently assigned either MAIN or ALT for that same channel (hope you can remember it...) - there is the other location of that same hidden mute button again (this is how you can control it for the selected channel's ALT input source.

    - "Midas" Layer Flipper: On the left / right layer shift buttons in the middle section of the console, when the BUS MASTER button is lit, Why are we unable to jump straight from bus 1-8 to bus 9-16 - with only one single push of that middle section layer shift button(s)?  Gary had some neat ideas that I am currently trying to digest, using the USER1 and USER2 custom layers, found in the link at the top of the post.

    - 2-Knob TRIM: Why did the designers go with the separate GAIN and TRIM controls, rather than the method used on the X32/M32 with those preamp controls (GAIN and DIGITAL TRIM, hidden away, behind the scenes) that were integrated into one single TRIM knob?  Why is there not an option to also be able to control the preamp as if it was with a single knob?  Gary also had some comments on this (above link) that I am also currently digesting.  Might it perhaps be related to a Gain Sharing strategy for a FOH/Monitor console setup?

    - Dual Mono Inputs: Why are all the many stereo input sources and stereo output busses that are available on the console, unable to be controlled as dual-mono? Why are we not able to patch an insert into only one side and not the other, but only able to patch it across both sides?  Is it just not realistically possible to be able to adjust even some little simple EQ differences between Left / Right sides of a particular input or output?

    - "Why won't this knob do anything?": In the top master section, INPUTS button pressed, ch1 selected and the GAIN 48v button pressed - why can I flip the phantom on/off with the corresponding button in that section, but not adjust the GAIN with the knob right next to it? This also applies to the GATE, COMP and FILTER buttons and related knobs in that section?

    - Dissapearing Parameter Controls: Why are the current parameter values not constantly visible in the USER CUSTOM CONTROLS section (located by the jog wheel)? Why must the current values only be displayed temporarily, and only after a knob or button is touched or pushed in that section?  It seems that after mapping out all the controls in that section, that there is some "space management" issues going on is this section, and if the parameters were to be displayed permanently, some of the current device assignment info would need to be removed (lack of space).  I did find that whatever is assigned to CUSTOM CONTROLS - 1 thru 3 (hit the topmost VIEW button in that section), looking at the topmost part of the MASTER section, those parameters are permanently displayed there under buttons F1, F2, and F3.

    Thanks for listening...

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    • DeclanSlater
      DeclanSlater

      Hey Paul (And Gary),


       


      you said: On the home screen here, both gain and trim are there side-by-each. Select the one you want to adjust and use the encoder on the right side of the screen to adjust.


      - your right, I saw what I did - I had a preamp assigned to MAIN and a USB input assigned to ALT.  If you have Preamps assigned to MAIN and ALT - then on the home screen, you'll see GAIN and TRIM.  If it is setup the way I had it, and the ALT source is selected on the channel's home page - then you just see TRIM there.  If you go to the CHANNEL INPUT then you'll see both GAIN and TRIM.  Not sure yet, but the GAIN there in that case, might adjust the MAIN input source, and the TRIM might adjust the ALT source because it is currently selected?  Huh?  That can't be right...

      • February 29, 2020
    • DeclanSlater
      GaryHiggins

       


      @DeclanSlater 


      Some of what I had tried to relate in my post is talked about here at around 1:53 and esp 3:00. I think it will clear up some of my garble...and explain better. This fellow Mick (should be named Mick the Quick) runs right along quickly and concisely....I think it clears up your questions some...maybe Maybe worth a few watches. I watched about a month ago and it all makes much more sense to me now than it did then.

      • February 29, 2020
    • DeclanSlater
      DeclanSlater

      Hey Gary,


      Thanks for the clip, good info there (and fast too!)


      The two big takeaways I got were, as you and Paul said;


      - Despite the ALT input being selected, the GAIN and TRIM controls on the selected channel, will only control the MAIN input source.


      - When the INPUT SELECT for any channel is set to AUTO, then those channels can be globally flipped between MAIN and ALT by pressing the SETUP button, then flipping the SRC SWITCH button.  That sure makes Vitrual Soundchecks much easier to pull off, rather than creating a special snapshot to flip those around.

      • February 29, 2020
  • GeorgeDougherty
    Contributor - Level 2
    2020-03-09

    Is the EQ on a mixbus truly a post-fader processing point?  I suppose it may not really make a difference as EQ is still pre post-fade insert point, but I'd expect it to be pre-fade and listed above the 4 main bus sends.  Just seemed odd to me 

    Would also love some flexibility in the Gate/Comp key points.  Since you can tap a channel from a mixbus, should be able to key from one as well.  I'd thought maybe the aux/mixbus could have a gate added to allow creating a proper ducking delay, but that would also require the ability to re-order the processing to put the gate after the post-fade insert point on an aux. 

    Might be easier for the effects selection to include a proper ducking delay.

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  • rclanger
    Contributor - Level 3
    2020-05-16

    I purchased from B&H in NYC the Audio-Technica Two Person Podcasting Package which has the Behringer Xenyx q1002 USB mixer. All I want at this point is to be able to record a podcast with my wife.

    Every piece of equipment is connected with the exception of the headphones! I attempted to record using Audacity software but the line on the screen is flat.

    I really have no idea how to proceed. I am simply out of my element. I don't know the terminology at all. I took a while to figure out there was no on/off switch!

    I would be thankful for any and all help just to be able to record a podcast with the equipment I have.

    Thanks

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    • rclanger
      RexBeckett

      @rclanger 


      Hello Robert, welcome to the community.


       


      See if these notes help you to get it working. If not, take a photo of the mixer showing the connections and controls and include it in your post (click on Photos at the top).

      • May 15, 2020
    • rclanger
      rclanger

      Thank you for your suggestions. However, I am not doing so well. Going through the checklist you sent I came to a problem with the diagram on the first page, Sound! Your example has Line In, USB Audio CODEC, ready.


      Mine has, Microphone, USB Audio CODEC, default.


      This seems to be the only major difference between your guide and my laptop.


      I uninstalled Audacity and reinstalled with the mixer powered on just in case the laptop and mixer would plug and play. No difference. Is it possible that the numerous dials on the mixer are incorrect? Again I know nothing!


      I do believe everything is connected correctly. At least I don't have any parts not connected20200516_124739.png

      • May 16, 2020
    • rclanger
      RexBeckett


      @rclanger wrote:


      Thank you for your suggestions. However, I am not doing so well. Going through the checklist you sent I came to a problem with the diagram on the first page, Sound! Your example has Line In, USB Audio CODEC, ready.


      Mine has, Microphone, USB Audio CODEC, default.


      This seems to be the only major difference between your guide and my laptop.


      I uninstalled Audacity and reinstalled with the mixer powered on just in case the laptop and mixer would plug and play. No difference. Is it possible that the numerous dials on the mixer are incorrect? Again I know nothing!


      I do believe everything is connected correctly. At least I don't have any parts not connected20200516_124739.png





      @rclanger 


      Yes it could be a problem with the mixer settings but your photograph does not show the controls.

      • May 16, 2020
    • rclanger
      rclanger

      15896538545187590648476481451330.jpg


      Had to use my phone, other pic was to big. Hope that this is okay.

      • May 16, 2020
    • rclanger
      RexBeckett

      @rclanger 


      I would set the level controls on channels 1 & 2 to 0 (12 o'clock) and turn the channel preamp Gain controls down until the Clip LEDs just stop flashing with loud inputs. You may also want to turn the Comp controls down a bit. Raise the Main Mix fader to 0. I cannot see if the USB/2-Track To Main Mix button is pressed but it should not be when recording.


       


      What make/model of microphones are you using?

      • May 16, 2020
  • hoover
    Contributor - Level 1
    2019-06-21

    Hi,

    I'm new to this console 

    I'm looking at a mic on channel 32 which I've sent to reverbs and delays on bus 13 - 16 

    All the effects are routed back to the fx returns and everything is lighting up like it should. I am sending the mic on 32 to 4 effects units at the same time and all are returning to FX returns 1 -4 

    The stereo bus button is lit on all the effects returns but I can not hear them through the headphones that I have plugged into the console.

    If i solo the effects returns I can hear the effects through the headphones but if I take solo off I can only hear the dry mic on channel 32. No other solo lights are lit.

    Many Thanks

     

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    • hoover
      Paul_Vannatto

      Hi Scott @hoover,


      Welcome to the forum. Do you have any DCA's assigned to the Fx Returns? If so, they are probably muted. If not, I would suggest that you export the scene (using the Utility button) to a file, zip it and attach it to your response (Choose File below).

      • June 21, 2019
  • Mentor
    Contributor - Level 1
    2019-11-18

     

    I am having a issue when I try to rout audio between Midas M32R and Tesira Server. All the inputs look to be direct outputs, and I dont have any fader control on the input.

     

     

    0 102
    • Mentor
      RexBeckett

      @Mentor 


      Hello Mentor, welcome to the forum.


       


      When you are setting routing on the Routing -> Card Output page, the Local and AES signal will be the direct inputs before any channel processing. If you want to use signals from other signal taps, you should Stage them on the Output 1-16, Ultranet or Aux Out tabs.


       


      If you upgade to firmware V4.0, you can also use the new User Out routing to manage the individual signals instead of being tied to blocks of eight.


       


      If you attach your scene file (see Choose File button below), and tell us what channels/taps you want routed to the card outputs, we could give you more specific help. Do tell us which firmware version you are using.

      • November 18, 2019
  • Motto26
    Contributor - Level 1
    2020-01-07

    Please help. I'm currently using UbuntuStudio 19.10. I'm using X-AIR Edit v1.5  It was working previously but as of recently the app's buttons and settings are no longer working after launching. I downloaded the app from the official Behringer site https://downloads.music-group.com/software/behringer/XAIR/X-AIR-Edit_LINUX_X64_V1.5.tar.gz

    The application launches but the faders, channels, settings and all others are not responding whenever i clicked it.

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    • Motto26
      Paul_Vannatto

      Hi @Motto26 


      Welcome to the forum. X-Air Edit 1.5 (64 bit) is working fine on my Mint 19.3. Are you certain that your Ubuntu is the 64 bit version?


       

      • January 7, 2020
  • one2one
    Contributor - Level 1
    2020-02-04

    Hey fellow tech heads i need some help. Recently purchased a Midas DP48 and am trying to connect it to the Midas Pro 1. any ideas on how to do it. thanks in advance

    0 77
    • one2one
      Paul_Vannatto

      Hi James @one2one 


      Welcome to the forum. You should be able to connect the DP48 to one of the AES50 ports of the Pro 1. Since the Pro 1 has a 96Khz clock rate, that would mean that the DP48 would only be able to access 24 channels instead of the 48 channels (at 44/48Khz clock rate).


       


      There is no DP48 support on the Pro 1 for group names and assignments, like on the M32. You will need to do that on the DP48 itself. To create the group names, you will need to create a file called names.csv. In it should be one line of comma delimited names, for example


      Gtr1Vox,Gtr2Vox,DrumVox,None,Gtr1,Gtr2,Bass,None,KickIn,DrumMix,None,TB


      and save that file in the DP48 folder in the SD card. Finally insert the SD card into the DP48 and go to the Setup, Group Names to load them in.


       


       

      • February 3, 2020
    • one2one
      one2one

      hello again, followed your steps but still getting nothing on the DP48. currently though the "in" symbol on the DP48 is switching between red and green and still cant get any audio out of it. is that a DP48 problem or a Pro1 problem. 


      thanks in advance

      • February 6, 2020
  • Chodi7
    Contributor - Level 1
    2020-02-07

    Necesito ayuda con mi consola xenyx 502. La uso para monitoreo en vivo, junto con in ears KZ As10. El problema es que se escucha todo saturado, no importa el volumen al que lo use, la mezcla que me manda el monitorista me llega rota a los oídos y es muy incómodo. Me pasó en varias ocasiones en distintos lugares. Algun consejo?

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    • Chodi7
      Vasco_Romijn

      Can it in English please?

      • February 8, 2020
    • Chodi7
      jd8888

      Per google translate:


      I need help with my xenyx 502 console. I use it for live monitoring, along with in-ear KZ As10. The problem is that everything is heard saturated, no matter the volume at which I use it, the mix sent to me by the monitor is broken in my ears and is very uncomfortable. It happened to me several times in different places. Any advice?

      • February 14, 2020
    • Chodi7
      DaveMorrison

      @Chodi7 


      Are you using the Xenyx 502 to create you personal mix from local sources or is your monitor signal coming from a another mixer such as a front-of-house (mian mixer)?

      • February 14, 2020
    • Chodi7
      Chodi7
       

      The signal comes from the monitoring mixer. It's the same mix for all musicians 

      • February 17, 2020
  • PatrickGMaillot
    Valued Contributor - Level 2
    2020-02-19
    Preparing new features for X32Automate programs: The X32 "GO" button can be used to launch different automation files (audio alone, audio+automation, etc.) to manage Musicals, Theater plays, shows, etc. First demo at https://youtu.be/bldx4bSH8DU More information at http://x32ram.com
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