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Keep in mind there are both physical outputs and virtual (UsB) outputs as well as AUX outs so you have a lot of variety to find a routing configuration to meet your needs.Assuming you are sending your broadcast mix to a streaming device that only recognizes USB 1/2 (like Facebook, OBS or VMix) you might need to use the USER OUTS section to build some custom routing and then send that out the Routing > Card section. This video might provide more insight on the Routing screen. ruclips.net/video/PpeqRuPGi2Y/видео.htmlsi=ejIjrMQlO-72qsuk

You will have to utilize a Matrix to get the second plugin inserted. Here is a video that talks about how I do this on a weekly basis in my broadcast scene. ruclips.net/video/9abR8wv4a9c/видео.html

I'm glad the video was helpful and you got the job completed.

I'm glad that was helpful. Take it up a notch now and refine your broadcast audio. Check this video out for a walkthrough of my process. ruclips.net/video/pMnaiuC55RY/видео.html

MD mic needs to be removed from the LR mix so select the MD. Channel, and then push the Main LR Bus button that is just to the left of the screen. You want the button to NOT be orange.

Yeah (like most things) the more experience the better you can make it sound. I agree with your points about practice/recording, and I feel most environments are conducive to acoustic kits if you have a musician as a drummer...not just a drummer 😉.

Thanks for the input. My Roland TD-4 did not have that functionality which was the supporting info for my statement there. All of the other e-kits I have played did have lots of capability.

​@@AllamHousejust read the td4 manual quickly because I was surprised at this lack of functionality, and the manual says you can adjust each pad's volume.

Interesting. Now I wish I had that kit back so i could redo some of my heartache. 🤣

@@AllamHouse Hehe :) I now use a TD12 (which i love), but also used to have a cheaper TD model (TD5 ? TD7 ?) before. In hindsight, i would have way better results with it today, just because my knowledge of these modules has increased so much throughout the years :)

Thanks for the kind words. I e done this trick on djembe, Cajon and even the bad sounding kick drums.

Thanks for the encouragement.

Thanks for the follow-up. Proximity is really a challenge with podium mics.

That is correct. This short talks about the needed cable. ruclips.net/user/shortsEwWBYPgSXwE?si=WwwoDBL8_Ke_DHdr

Thanks for sharing with the community. I’ll have to keep it in mind the next time I get in that situation.

Sounds good. Let me know if you need individual assistance.

Correct. You only get gain going through a mic preamp. The AUX inputs will provide digital trim only, but the method explained in this video does open you up to the rest of the channel strip features.

I use an old 2012 MacBook Pro, i7, 16GB RAM, 512 GB SSD

@@AllamHouse okay thanks but i having a issue when i try to do virtual sound check with logic pro x the sound is glitching i have try all the buffer sizes still glitching my Macbook pro specs is a 2013 Macbook pro ram 16 gb hard drive 1TB Samsung ssd processor i7 any help will be greatly be helpful

Check the sample rate. Usually audio runs on 44.1khz, but many people are also handling broadcast with their consoles which run at 48.0khz. Just make sure you have the console and DAW set the same and it should take care of your issues.

This raises a few additional questions, but the most important is what mix are you listening to exactly? House, monitor, broadcast?

I’m glad the video was helpful. Let me know if you have other questions.

Hopefully someone will offer some insight. I don't have the network background to offer a good answer to you, but you might also try people on the Behringer X32 facebook groups as they might have more specific info to offer.

@@AllamHouse I've been testing.... but i heard something about ports 10024 and 10023 TCP/UDP

That’s awesome. Thanks for sharing your experience with the community.

Thanks for the feedback and I do think there is more support out here for the Behringer/Midas ecosystem. Let me know if you have other questions and I can point you to quality resources.

Based on your request it sounds like you are trying to do something different than what this video calls for specifically. Are you looking to build a broadcast mix to get out of your console to a streaming / recording device? Are you interested in multitrack recording out of the console? Help me better understand your end goal and I can point you to the right setup to use.

I;m happy to help where needed. Other folks have some nice videos on the Wing, but if you want to send me an email we can start a conversation outside of RUclips and help you out directly. Check out the contact card on my website at sound.AllamHouse.com

Thanks for reaching out. A Matrix does remove some of the flexibility of individual instruments into your livestream since you are using the main mix and/or buses to feed the matrix. That being said, you do have a few things I would want you to check. (1) Check the gain on the fader input to make sure you have enough signal coming into the board. Often times we have subs in our PA which require way less gain from kick drum and bass guitar and then we only turn up the gain according to what we hear in the house mix. That causes problems downstream and I tend to like having my input levels at -12dB on the input meter of the console. (2) Once you have the correct gain set, now move forward with making sure your house mix is still correct. You might need to turn down the input fader for the bass if it is too much now. (3) Finally, set the bass guitar in the broadcast mix if possible. You mentioned a guitar bus being used. If that is being used for just broadcast then that is the location you want to turn it up a little (if needed). If that bus is sending to your house mix and causing it to be too much then you might consider adjusting your house PA a little to handle the higher level of the bass guitar.

@@AllamHouse wow I’m Humbled sir for taking this much time out of your precious and busy schedule to answer my question. I appreciate you immensely and God bless you boss 🫡🙏🏿

All of the USB audio runs across one cable. It can run 32 channels from the console to the computer and 32 channels from the computer back to the console at the same time.

You can have separate mixes inside the same console using MixBuses. The main mix goes to the house speakers, then you can build a Broadcast MixBUS for everyone not in your room. You can even use other MixBuses to build monitor mixes… all in one console. Dante is something completely different. This video here shows you how to setup the broadcast mix on the same console as your house mix. ruclips.net/video/pMnaiuC55RY/видео.htmlsi=qwC_0bRUzJE4nm9v

@@AllamHouse thank you, it’s funny you sent me this link, I just finished watching this video. So I’m currently leading a team of young adults in sound, can I have a secondary person mix the livestream using the x32 software, mixing software, daw etc while I mix FOH?

If you want to have a second sound tech mixing your stream you will setup the MixBUS as Pre-Fader instead of Post Fader. This allows the inputs to move independently from the house mix. Then, connect your console to a network where you can install X32-edit on a computer (or use a tablet with the M32 app) and connect to the console. Here you can select the MixBUS for stream and then mix with faders and not interrupt the house mix. This video discusses the setup of X32-edit. ruclips.net/video/uijp9GD-gT0/видео.htmlsi=UP0QFZ_DKn1SyHHR

@@AllamHouse ok, I’m going to set all this up. I do however have a bit of a worry. Our previous sound techs have setup a livestream output from, do get remove all of that first? Also once everything is setup. Which output am I using to connect to the live stream service?

Step 1 is to save your existing scene and note how your livestream is connected to your streaming device from the console so you can always go back to your existing scene by loading the old scene and setting up and cables in teh same spot. Step 2 is to save a new scene that you can begin to alter for this new setup. I always recommend using recorded multitracks from your group because you can work on this setup by yourself. Step 3 is to setup a stereo BUS pair, similar to the video I sent which talks about having it set to Post Fader. I usually go with all of my sending fader set at unity which is effectively a direct copy of the house mix and then alter from there. Usually you need to add some drums and bass guitar and remove (or turn down) some vocals. FX are going to be huge for refining the mix, but that is all a refining process once you have the general setup and mix complete. Step 4 is about routing. You had a Matrix leaving the console going to your streaming device. In this case, I would just remove whatever was going to that Matrix and ONLY send your new BUS pair into it. Make sure to turn off any FX, EQ, Comp, etc. so it is really just a direct pass through (unless you feel it needs a little touch up). This saves you from having to move cables or reroute anything.

Thanks for the encouragement.

I could use them on every instrument, but I would avoid using them for vocalists. The latency isn't horrible, but this on top of the standard 4ms of a console gets to be too much. For budget vocal IEM I always run the Shure PSM-300.

We have the JVC GY-HM180. They are really great cameras and do well in most situations, but not as a back camera in our low-light environment. So we keep it up front and have the Canon CR-N500 as our rear camera due to the larger sensor and PTZ capability.

I’m glad you found it helpful. Let me know if you have other questions.

I'm not quite sure I understand your question and since there are so many possibilities, I'll ask you to ask it again but slightly rephrased. Are you asking about changing the actual sound coming out of the console? Are you referring to a computer program to send your live stream audio into and stream it (with video) out to the world?

Thanks for the encouragement. Let me know if you have other questions about the console or mixing techniques.

You are close for sure and XLR's are going to work just fine. First, go to Routing > Outs and make see what OUT your MixBUS is sourcing. This can be a confusing concept, but just go with it. Let's assume it is MixBUS 5 and you are sourcing OUT 5. Then go to Routing > XLR and make sure that OUT 5 is being sent to XLR 5. If this is the case you can then connect your XLR 5 to your IEM and it will have a mono MixBUS being sent to your head. Next thing to check is the setup of the MixBUS to see if it is PreFader or Post Fader. You do this by selecting the MixBUS and going to Home > Config and looking at the bottom left. See what the input tap point is set to and this will help you understand the setup. Usually IEMS will be set to Pre Fader. Finally, make sure the channels you are trying to send to your IEM are unmuted on the main layer that feed the Main mix. The Mute will impact all sends including your IEM. Last thing to check is that you are plugged into the left input on your IEM and then set your unit to Mono. I'm taking this as an assumption since that is typically how you would function with a mono setup going into an IEM transmitter.

The talkback mic is either routed to your broadcast mix or your house and you just can't hear it in the room (depending on how your setup is designed). 1) Are you using the talkback input on the back of the console or a different input and treating it as a talkback mic? 2) What signal are you sending to OBS? A copy of your house mix like this video explains or are you sending a MixBUS setup?

Yes, the P16 has a through-put cat5 connection so you can send your signal into one and then out of that one to the next one with no signal loss. Many people avoid this and use a switch when they run POE to power their units across the Cat 5 connection or to have flexibility from each unit and not have to rely on the others they are plugged in to.

1) go to Settings > Card and make sure you are set as 32/32 2) go to Routing > Card and make sure those settings match Routing > Input 3) If you are on a newer machine running USB-C and converting with a dongle. Make sure it is either Apple or Anker branded. Other less expensive brands will only convert channels 1/2 4) go into the settings of Logic an make sure the inputs are mapped properly so the DAW has a place to send all 32 USB inputs from the console.

I’m not sure I understand your question. Inputs will jump directly to the Main BUS as Post Fader. That being said, if you have a Bus setup for mixing that has inputs sent to it (instead of directly to the Main BUS) Pre-Fader, this means that any changes you do with the faders for the Main BUS will not impact the Bus mix. This is used for IEMs and some live stream applications. That is the bus configuration being Pre-Fader, but also have other tap point options. Once you have a mix in this Bus you need to send that Bus out somewhere (Monitor, IEM, etc). Of this Bus is leaving the console you can set the tap point for the output as Pre-Fader, Post Fader, etc. where Post Fader will allow you to use the Bus fader to adjust volume control of the output.

@@AllamHouse haaaa I got you now. When you say tap point could you please clarify what you mean?

The tap point is describing where the signal is “tapped” when it is sent to the BUS. You can choose to grab Post Fader, Pre Fader, Pre-EQ and others.

@@AllamHouse Thank you

You could do that and have stereo subs. Typically people only run subs in mono in an effort to minimize cancelation and muddiness within the venue.

@@AllamHouse Thanks! Run mono and stereo subs side by side (swap their signal) if you get a chance on modern bassy music like dub. Once you set the gains evenly the stereo's quality is very noticeably better. Even with only one sub. Tested in variety of spaces. Unanimous blind agreement on the quality. I'm not an expert but one of the manufacturer's reps said it has to do with phase cancellations in the mono signal processing. Try it out :D (I suggest using a sub that this would clearly make a difference if you can: KS28, PK G28, JBL VTX G28, Bassboss ZV28, etc.)

Sounds like a fun adventure for sure. I wish I had the gear available to run this experiment, but I'll have to take your word for it in terms of the difference.

Yep, you nailed it. Thanks for the comment and encouragement.

Yes, if you use this Matrix approach or a Post-Fader MixBUS, they will both be impacted by you muting a channel. In either of these setups when you mute a channel it is muted in the house and the live stream mix.

I’ll have to hope someone else can offer insight. This board I worked on in this video was one of two boards I’ve worked on. You could gently test some of the outward most keys to see how they act.

If you are using the 1/4” AUX outputs, yes. The XLR outputs are set to be sourced by the mixbuses by default so if you are using those there is no change required.

If you have the standard Behringer X32, there is a card on the back that has two, small metal poles coming out of it and it is labeled X-USB. There will be a red light next to the USB connector. Keep in mind you are using an older printer cable style so it is USB A on one side and USB B on the other. There will be a similar looking plug labeled "network", but that is NOT the connection you are looking to use. If you happen to have the X-Live card (also stock on the Midas M32) you will see the same USB connector. If you have a different card installed like a MADI or DANTE card then you will not find a USB option there.

I'm not exactly sure what to make of this comment, but I do connect and record across USB 3.0 ports all the time on my various laptops (Mac and Windows). What are you referring to with the part about "Drop outs, crackling sound, lost connection."? These symptoms are common when an install is completed without using the proper network cable between a console and a stage box.

@@AllamHouse Thank you, yes, my Lenovo mini desktop pc with usb 3.0 makes random clicks and pops during playback. I read somewhere that the issue is the usb port, so I tried connecting a usb 2.0 hub to the lenovo 3.0 port and conneting m32r Live usb console to the hub and all the problems are now gone.

Thanks for the encouragement.

Check out this video where I explain the three ways to get audio out of the console and into a live stream. Personally, I use the XLR method and send into an ATEM Mini using a stereo XLR to 1/8” cable. ruclips.net/video/xGt1z5I7blQ/видео.htmlsi=bnqExohsutapgJei

The first thing to keep in mind is that you only have 32 channels you can run to/from USB. So you can run 32 channels from your DAW into the console and 32 channels from the console into the DAW. Once you add your record player into the mix coming through the AUX In on the back of the console you will now need to determine which two channels (assuming your record player is stereo) you will give up to be able to push these AUX In channels to your DAW for sampling. If we choose Ch 31/32 to substitute with AUX In 1/2 to move forward with the example then we would do the following setup. Go to Routing > User Out and setup your Out25-32 to have your first channels matching your existing setup (25-30) and then 31 would have a source of AUX In 1 and 32 would source AUX In 2. Once this is complete you will need to go to Routing > Card and set your channels 25-32 to the USER OUT 25-32 and the right window (when you scroll wheel down) will show that your settings look similar, but now channel 31/32 should have AUX In 1/2 as their source. I hope this is helpful, but let me know if you have other questions.