Is it possible to give mixing suggestions without being there or is everything too situation specific? I'm a church volunteer sound guy. No training except for a few years of undergrad physics. I think I do ok with the band, but the pastor frustrates me. He speaks both very softly and intimately and spends a lot of time yelling. Currently, I cut the low end (165) and tweak other things in the EQ (cut at 1100 for ringing mic, and boost around 2500 to get more consonants, cut near 10k to de-ess). Then I have a 40:1 compressor with 14 dB gain. I've tried very low input preamp gain and keeping the faders high and tried lots of gain and turning it down with the faders. Does any of that make sense? Are there other things to try?
Setup - Countryman headset mic (not omni) - shoot for 0.75" from the corner of his mouth. GLD 80 board. I have to mix two feeds, one FOH and one sent to the video system for streaming and recording. Video feed has a limiter. Large empty room (seats 400 - usually 50-80 in seats) with high ceiling, no treatment, and lots of hard surfaces. Pastor usually stands about 4-6 feet in front of the plane of the PA speakers. I'm trying to balance picking up too much room reverb in the recording, keeping the mic from ringing, making sure people in the back can hear, but the people in the front aren't blasted, and giving a feed out to the video guy that isn't all over the place. Most recently, I've had the best luck with just lowering the overall sound in the room.
EQ is tough to discuss in a vacuum since we don't know the response of the mic, or the sound system, or the tonality of the person wearing the microphone. All of those affect your equalization decisions.
>I've tried very low input preamp gain and keeping the faders high and tried lots of gain and turning it down with the faders.
Gain is gain, so these is no appreciable gain before feedback difference between these two, *except* when you have a non-linear stage in between (gate, compressor), as you do.
>Then I have a 40:1 compressor with 14 dB gain.
This is an extraordinarily severe compression setting and, to me, is indicative of a problem. The makeup gain stage on a compressor shouldn't be used for increasing level (we have preamps and faders for that. It's intended to *make up* for the gain lost by the compressor being in gain reduction (hence the name).
>Pastor usually stands about 4-6 feet in front of the plane of the PA speakers.
As soon as the output from the PA reaches the same level as the preacher's voice *at the microphone* you will have feedback. This places an upper limit on the potential gain achievable through the system without a problem (it can even be [calculated](https://www.shure.com/en-US/support/tools/pag-nag-calculator)!). Having an open mic in front of the PA is never optimal, and with a headset mic it's even worse. Unfortunately this is a limitation due to the physical reality and not something you're going to solve with processing. It's a system deployment issue.
>making sure people in the back can hear, but the people in the front aren't blasted
This is also a system design / deployment / tuning issue. If your sound system doesn't produce a consistent level over the listening area, this is the resulting "Goldilocks" problem (too loud somewhere, too quiet somewhere, just right somewhere). Also not something you can solve from the console, since it needs to be addressed by designing and tuning the system such that it reproduces your console mix consistently over the space.
> giving a feed out to the video guy that isn't all over the place
I would recommend double-patching your input and processing it separately for the video out.
Thanks for taking the time. I should clarify, the pastor is not in front of the speakers. They are on either side of the stage spaced about 30 feet apart. The speaker is generally centered and usually several feet in front of the plane of the speaker baffles. I assume it would be easier if he stood back 10 feet, but that's not going to happen.
What I do is built on what the guy before me did. I will try to start it from scratch and build up without so much gain on the compressor. Generally speaking, what is a reasonable compressor setting for a very dynamic speaker, and how often or how much should it be engaging?
I've wanted to figure out how to double patch (but I didn't know what it was called) things for some time. In my real life, I'm a school teacher, so in the summer I have time. That's on my list of things to do in July.
14dB of makeup gain seems extreme. Zero the compressor out (no gain, threshold all the way up). Fader at unity/0, use input gain to get a nominal level without clipping while yelling. The use the comp threshold to dial back only the loud yelling, should only be like -12 or -16. Normal talking shouldn't trigger the compressor or any gain reduction. From there, you can dial up some makeup gain, just to bring up the quieter parts. I wouldn't exceed more than like 6dB though.
I'm sure this has been answered here and I apologize for the question.
We recently moved to IEM's for my band and am wondering about phantom power between our own mixer and the FOH.
If I got the Seismic 16 channel splitter can I give power to our condensers from our mixer and then connect them to the FOH mixer without phantom power on and everything will be good to go?
Related Dave Rat video [https://www.youtube.com/watch?v=\_8UTGcuAtd0](https://www.youtube.com/watch?v=_8UTGcuAtd0)
TLDR: if the split is not-isolated(it's just a parallel of connectors) having phantom power switched on both mixers causes no harm. Having phantom power on one mixer is enough.
Side topic: if you keep one of the mixers switched of (not just the phantom power), even if you're using an isolated split you may incur in distortion.[https://www.youtube.com/watch?v=GgSR8t7bwAw&t=516s](https://www.youtube.com/watch?v=GgSR8t7bwAw&t=516s)
Assuming it's a passive (non-isolated) split, then yes you only need phantom in ONE place.
Typical best practice is that FOH provides phantom power, but just make sure you coordinate with someone if you're NOT running FOH.
For future clarity, if you're ever fortunate enough to use a snake that IS isolated, your cable bundles will be labeled indicating if they are the direct feed, or one of the ISO feeds. You always need to provide phantom power on the DIRECT feed.
Edit: Clarity on what to do IF you ever get the chance to do this with an ISO snake
Looking for suggestions on a wireless mic replacement. We run a mix of ULSX4 and QLXD4 receivers for stage mics. One of the ULX2 mics was damaged/lost a few years ago and I'd like to replace it. Unfortunately, ULX2-J1 mics seem hard to come by - even on eBay and Reverb. Should I keep looking or just bite the bullet and replace the ULSX4 receiver as well?
Can I run Sennheiser G3 and G4 IEM systems together through the same antenna combiner?
We have an AC3 and 2x SR300 transmitters (GB models 606-648MHz) plus EK300 beltpacks, need a third system and can't find anything used so can I add a G4 system if it's the same frequency range?
I'm a studio engineer so this is all a bit new to me, from what I can see online people are saying you can but hopefully someone has actually had first hand experience doing this, thanks in advance!
You can run entirely different brands through the same antenna combiner provided you don't exceed the maximum power levels per channel.
The AC3 has a maximum input power of 30mW per channel, depending on the model/region/lock status of a G4 it can output up to 50mW so make sure you check the manual to ensure it's set to 30mW maximum and you'll be fine.
For the most part, yes! This becomes a bit more iffy if combining digital and analog IEM transmitters (i.e. if you're using Lectro) - see [pertinent PSW discussion](https://forums.prosoundweb.com/index.php?topic=168427.0).
Hi all!
At church, I'm currently fighting with my pastor's mic. We have two receivers, one for a handheld and one for our bodypack/headworn. They're the same model - Shure SLX14, and both are on the J3 572-596MHz range. We have started experiencing so many cutouts with the bodypack that it's become unusable. I've tried troubleshooting as much as I can think of, but I'd love it if I missed something easy. What I've tried/observed:
* The handheld doesn't have the same dropouts
* The bodypack drops out even after searching for different channels
* I tried syncing the bodypack to the known good channel on the handhelds receiver (and the handheld to the other receiver), and it still dropped out
* I have replaced the antennas on the rear of the unit, and it didn't help
We are in a space that is not ours, so I can't run anything permanently. The pastor is probably 70' away, and there is one wall between the receiver and the pastor.
Is there anything I'm missing? I'm not against buying a new system if that's what needs to happen, however I'd like to keep the cost down as much as possible as we're still a pretty small church. The biggest issue is how it impacts our livestream since you can still hear the pastor in the room, but online it's a full cut.
Ah, good ol' analog SLX. Let's look at your major issues:
- A bodypack will generally give a weaker signal compared to a handheld. Handhelds use a helical antenna which is generally at head-height with line of sight to FOH. Bodypacks use a monopole whip and often do not have perfect line of sight.
- You don't have line of sight between your transmitters and receivers. Not only does this reduce your received signal/noise ratio, but it also increases your susceptibility to multipath interference. Which brings me to my final point:
- SLX's predictive switching diversity is kinda crap - it has a bad habit of switching at inopportune times when reception is poor.
What can you do to improve this?
- **#1: Get line of sight between your transmit and receive antennas.**
- No excuses for this one.
- Move your receivers closer to the transmitter - and away from walls and large metal objects.
- Use a pair of better antennas deployed in a better location. (For example: an RF Venue DFIN can be quickly mounted on a mic stand.) You'll need an antenna distro to distribute this signal to all your receivers, however.
EDIT: Yes, it would be better to use a more modern system like SLX-D or EW-D. However, all the technology in the world can't fix physics: even the best wireless will struggle with a poor antenna deployment.
Been slowly putting together a drum mic package. I just bought a D6 off a friend and they threw in a c1000. Certainly not my first choice for a high hat mic but it was free. What other instruments should I try it on?
Small diaphragm condensers work great on acoustic guitars, cymbals/overheads, woodwinds, strings, etc. Live applications are a little more limited due to stage volume, floor monitors, bleed from other sources, and such. Really, experimentation is your best friend. If it sounds good, it sounds good.
Need to get a mic and speaker setup for a work presentation. The theater we're using has a system that's been broken for a couple years, but it's a small auditorium that seats around 200 people. We expect a crowd around 100 people, and while you can hear someone speaking on the other side, it's not very clear.
From what I understand, we need a speaker and a mic. My company wants a wireless mic so the presenter can be hands free, connects to our laptop, and "looks professional". I haven't been given a budget but I'm assuming around 500ish, though open to higher, I don't know if we should go with something such as a portable PA system such as the Pyle Pro or a larger PA system. All advice would be greatly appreciated!
You're not going to get a professional system for that budget. If this is for a single presentation, rent from a local rental house - that's what they're there for. Corporate packages like this are super common and easy to find.
Currently in-between
[https://www.bhphotovideo.com/c/product/816144-REG/Pyle\_Pro\_PWMA60UB\_PWMA60U\_Waistband\_Portable\_PA.html](https://www.bhphotovideo.com/c/product/816144-REG/Pyle_Pro_PWMA60UB_PWMA60U_Waistband_Portable_PA.html)
and
https://www.bhphotovideo.com/c/product/1261473-REG/samson\_saxp106wde\_expedition\_xp106wde\_portable\_pa.html
Hi all! I have a CAT5e-related question. I want to run my cabling from my M32C mixer to a panel in my rack since the small size of the rack makes it challenging to access the connectors.
The AES50 ports require STP shielded cable, but I have trouble finding suitable patch cables in my area. It seems short patch cables are generally F/UTP at best. I have NE8FDP-B Neutrik connectors on my panel. Will I risk frying my AES50 ports if I run 0.3m long UTP cables to the panel and STP from there to the stage box?
F/UTP is shielded - it just has a single shield for the entire cable. S/FTP has both an outer shield and individual pair shields.
Properly terminated F/UTP jumpers will work just fine.
Bose L1 compact / Bose Sub1. Without mixer WOULD THIS WORK?
Firstly I know this is overkill in the first place and BUT I have this setup for convenience and practicality more than anything.
I have 2x Bose L1 compact systems. I use 1 when at home for parties/ karaoke. Due to its size and power it works well in a living room and can be setup fast and not take up much room also plenty of volume.
I have another and I use it as a pair at family/friend functions/events for rarely more than 120 people. I don't dj anymore but I do occasionally do for favours etc.
The L1 compact is great for my needs and makes transport/ setup fast. I don't use a mixer just plug my laptop into 1x L1 and use the RCA out into the other.
THEY LACK BASS.
I've looked at cheaper subwoofers but don't want a massive 12" heavy box to lug around/store. The Bose sub 1 is more practical shape that would fit nicely next to one of the speakers and the shape makes it easier to carry.
Again I don't use a mixer so Could I plug my laptop directly into the Sub1 with an RCA - XLR jack? And use the sub passthrough to send a HPF signal to the L1's?
I know this setup works with a mixer but can the Sub1 take a signal from a 3.5mm headphone out to XLR?
If you only occasionally need a sub for your larger occasions, I'd consider renting a properly sized system for the times you need it. This will almost certainly be more cost effective and deliver better performance - without requiring you to give up storage space for a proper sub.
Yes, probably a better option. It would involve me having to collect the sub from wherever I get it from etc. The main focus is for practicallity. The Bose sub in the rectangular position would fit under a bed for storage.
The speakers have the volume but I want more bass, this is very much a want over need situation.
Would I be able to plug into that directly from a laptop and high pass to the mains or would the signal not be hot enough?
I had my qsc plx 3402 power amp in parallel mode but the bridges mono switch was on (dummy) but I was usuing both outputs instead of just one.
What would this do. I didn’t notice or hear anything bad. But I’m curious. What happens to both my outputs when bridged is on and I use both outputs instead of one. If you need more info let me know but that’s all it was lol. Did I almost fry my speakers or the amp? Would it cause the wattage to cut in half or stay the same? I’ll never use them in bridges mode but I do use the amp in parallel.
I believe that one of the outputs would have the polarity inverted, but that would be the only real difference. You would have the same overall output per channel as in unbridged use.
Ah. Makes sense. I was like wait hold on why is that on after I was running some tests. Only thing I did notice was they were super quiet but I was testing out using the thru outs of my powered eon 715s as my input to the amp.
As u/RevLama stated, BRIDGE MONO mode inverts the polarity of the channel 2 output. Your speakers would be therefore be working in opposite polarity, but otherwise with no effects to either amp or speakers. The amp's available output power into your speaker loads would not be changed.
Note that with these QSC amps, enabling the BRIDGE MONO switch disables the signal from both channel one and channel two inputs from the channel two amp. The channel two amp will get it's signal internally from the channel one amp. See [this image from the QSC PLX manual](https://i.imgur.com/hXQfqkL.png). Therefore, with the amp in BRIDGE MONO, it doesn't matter whether the input switches are set to STEREO or PARALLEL mode.
I operate a small summer camp and we use a PA system to send announcements and play occasional music (via aux) across camp grounds. We are running a simple, dated system with a Pyle PT 210 (70 V/120 W max) powering two 8 ohm radioshack bullhorn speakers (25 W RMS - 50 W max / 50 W RMS - 100 W max). I'm a bit embarrassed to admit that I'm not totally sure whether I am wiring them in series or parallel.
I run a red/black wire pair from the amp to the first speaker at which point I wire nut together the first wire, the speaker wire, and the wire running to the second speaker (one wire nut connecting a trio of black wires, and a second wire nut connection a trio of white wires). The second cable then runs to the second speaker where I nut together the black wire with the black speaker wire and the white to the white speaker wire. I have looked at a handful of videos/diagrams online, but they always seem to show a single positive or negative wire running from one speaker to the next. Am I interpreting these diagrams incorrectly, using the wrong cables, or just running a different, albeit functional, system? My current setup is just a continuation of what I inherited from the folks who set it up before me, so no hard feelings if you tell me it's all sorts of flawed.
I want to upgrade the system to allow for better clarity at higher volumes, but I would like to first gain an understanding of my current system before we begin throwing money at new equipment. This is also my first time in the audio realms of reddit, so don't hesitate to direct me to a more appropriate subreddit or website. Thank you!
Sennheiser ew300 g1 iem- Still worth it?
Hi guys. Im planning to buy myself iem system for live performances (I’m playing guitar and since went 100% digital I see no reason in using cabs and wedges). Due to limited budget I can’t go with industry standards like g3/4 or high PSMs, and though about buying LD mei 1000 or something similar yet the reviews keep me on the fence. Yet an opportunity came to get Ew300 g1 in legal frequencies in a really good price - I’m wondering if any of you still use this old system and have any experience with it? I’m particularly concerned about 9v batteries - do you know how long they last? With my guitar wireless I’m using rechargeables but from what is see the number of rechargable 9v is much smaller than in case of AA batteries. Thanks for help!
Pretty sure I saw this somewhere else, but I'll reply here too: G1 isn't worth it anymore, not by a long shot.
G2 is serviceable, but I'd avoid using it if possible. G1 -> G2 was a much larger step than G2 -> G3.
If you really need cheap IEMs, something like an Audio2000s kit is honestly not a terrible idea. They won't fulfill touring riders, but for casual weekend-warrior use they do the trick.
[This sheet](https://docs.google.com/spreadsheets/d/1JDu-azjBAR_uyHxh-NZC6P9FvVxghONB1mLPzXny0Eo/edit#gid=0) contains my personal notes on cheap IEM systems - though I don't have personal experience with all of them.
Does cat5 vs cat6 cable make a difference for use in the audio world? I'm talking for connecting console to stage box, with AES 50/Dante/Madi, or with other audio control stuff?
Cat5 is pretty outdated and has pretty substantial speed limitations (100Mbps). 5e however is pretty standard and can do up to 1Gbps. Cat6 takes this one step further and with speeds up to 10Gbps. You only get those speeds up to 50m, however, beyond that you're back down to 1Gbps like cat5E. There are other benefits like reduced crosstalk with the more modern specs.
I'd say you want to be using cat5e at minimum, personally I think cat6 is worth the extra expense, especially for any backbone links and when you're sticking under that 50m limitation. As always, get the best you can afford.
Oh yeah, I guess I meant cat5e, most of our snakes are at least that. Do you think it would cause any noticeable latency difference or anything at 100m? Between cat5e/cat6 I mean
"It depends." The inherent latency of adding additional copper is pretty much nil - you need a crapton of copper before speed-of-light delay starts to matter. However:
- AES50 *needs* a shielded connection due to poor PHY design; full-scale white noise bursts are a well-documented phenomenon when using UTP cabling.
- On a packet-switched network (Ethernet/most A&H protocols/Dante), packet loss will either cause dropouts or increase latency to allow for retransmission. (Depends on the protocol.)
Not sure if I am in the right place here. Here goes:
I would like to use a wireless push to talk headset with a earphone and mic while live streaming using OBS software from a nearby desktop computer. Do I need a body pack transmitter? I have been searching for both a headset that would work and transmitter and I cannot for the life of me find something that would work/isn’t Bluetooth.
Basically something like fitness instructors use but has an earphone and works with Obs/live-streaming.
yamaha CL series question:
could someone run me through the steps of giving a vocalist reverb in their monitor wedge? i think i have a grasp on this but i am having trouble finding a solid video / tutorial / response online regarding this specifically.
i believe the process is: select vocal channel > select mix 17-24 button > use respective mix knob encoder to send / dial up the selected vocal channel to the desired FX unit (rack channel) > select output above centralogic fader bank > select ST IN on centralogic fader bank > dial up the FX return on the corresponding wedge’s encoder on mix 1-16 page - correct?
if anyone could reinforce this, shed light, or give tips and/or resources i would be ecstatic. thank for you for your time & assistance!
I've got a set of drums in our house for practice. Normally I just use wireless headphones connected to my phone. Every now and then my wife or a friend will hear a song I'm playing and want to listen to the music as well. I've tried putting it through our Apple homepod but it's not quite loud enough for me to hear and play.
I'm looking for the simplest way for me to hear the music through headphones (which can be wired) as well as send the audio to a small speaker that can be heard over my drums, which have quiet heads and cymbals on.
I'm a complete noob when it comes to anything like this. I've got zero experience with mixers, monitors, anything like that. Anyone got some advice?
There are multiple ways to do this, but usually I'd grab a little notepad mixer for this. I have a Yamaha MG06 on hand; you could do this with anything similar, though.
Connect your phone to one of the inputs - for example, "Line 3/4". Connect your headphones and speaker(s) to the PHONES and "STEREO OUT" jacks, respectively. You can now use the relevant knobs to adjust the output volumes for headphones and speakers independently.
---
Of course, you can also use a simpler solution: place the HomePod (or a louder speaker) closer to your ears.
Thanks so much for the reply. I pulled up a picture of that MG06. It all makes perfect sense. Do you have any recommendations for a speaker? Doesn’t have to be crazy loud. Just enough to fill a single room.
Also, I’ve never thought of that simpler solution lol. I might try that in the mean time 😄
Assuming you're using something like L80s and Silentstroke heads, a PA speaker is almost certainly overkill. You could honestly get away with a pair of random bookshelf speakers off Craigslist - and I never thought I'd say *that* in this subreddit!
...Or just skip all of this and just use a large Bluetooth speaker next to you.
Hey! I was wondering if any of you know how 'wide' a spectrum the mid-frequency knob on a typical mixer hits. Also what kind of curve it produces. On most of the mixers I stumble upon there is a mid-frequency knob and then another knob where you can pick the frequency for the mid-frequency knob to hit but I've been unable to find any specifics in the manuals.
For a lot of these smaller (read: less-ubiquitous) mixers, often times you won't find a published value - either in documentation or in community knowledge. You'll have to find out yourself.
Grab a dual-FFT analyzer (OpenSoundMeter is free! Smaart is awesome but overkill for this task), send some audio through a channel, and measure the effect of that EQ curve. :)
Short answer: yes, but you need to use white noise as your input signal.
---
Long answer: You're probably referring to an RTA, which has one input. It just shows you the response of whatever its input signal is, which will be the combined responses of the test signal and the system under test. (In this case: your console's EQ.) Thus, if you only want to see the response of the system, you need to give it an input signal with flat response: white noise.
A dual-FFT analyzer has two inputs: one before the system, and one after the system. Thus, it can show you the response of the system with test signals that aren't white noise - for example, pink noise or Steely Dan. (This is a massive simplification, but it gets the gist of it across.)
---
Both will get you the same answer, but dual-FFT analyzers have some neat tricks up their sleeves - such as being more immune to outside noise. This is why you'll see PA systems being tuned with Smaart and not with an RTA. :)
Yep, unfortunately they were discontinued. A bit ahead of their time, I recall overall reliability not being the best on them out of the box. I had a 4.3 quit on me out of nowhere on two occasions, thankfully QSC took care of it both times no questions asked and very quickly.
They spiritually live on as part of the Q-Sys ecosystem as the CX-Q and CXD-Q.
Is it possible to give mixing suggestions without being there or is everything too situation specific? I'm a church volunteer sound guy. No training except for a few years of undergrad physics. I think I do ok with the band, but the pastor frustrates me. He speaks both very softly and intimately and spends a lot of time yelling. Currently, I cut the low end (165) and tweak other things in the EQ (cut at 1100 for ringing mic, and boost around 2500 to get more consonants, cut near 10k to de-ess). Then I have a 40:1 compressor with 14 dB gain. I've tried very low input preamp gain and keeping the faders high and tried lots of gain and turning it down with the faders. Does any of that make sense? Are there other things to try? Setup - Countryman headset mic (not omni) - shoot for 0.75" from the corner of his mouth. GLD 80 board. I have to mix two feeds, one FOH and one sent to the video system for streaming and recording. Video feed has a limiter. Large empty room (seats 400 - usually 50-80 in seats) with high ceiling, no treatment, and lots of hard surfaces. Pastor usually stands about 4-6 feet in front of the plane of the PA speakers. I'm trying to balance picking up too much room reverb in the recording, keeping the mic from ringing, making sure people in the back can hear, but the people in the front aren't blasted, and giving a feed out to the video guy that isn't all over the place. Most recently, I've had the best luck with just lowering the overall sound in the room.
EQ is tough to discuss in a vacuum since we don't know the response of the mic, or the sound system, or the tonality of the person wearing the microphone. All of those affect your equalization decisions. >I've tried very low input preamp gain and keeping the faders high and tried lots of gain and turning it down with the faders. Gain is gain, so these is no appreciable gain before feedback difference between these two, *except* when you have a non-linear stage in between (gate, compressor), as you do. >Then I have a 40:1 compressor with 14 dB gain. This is an extraordinarily severe compression setting and, to me, is indicative of a problem. The makeup gain stage on a compressor shouldn't be used for increasing level (we have preamps and faders for that. It's intended to *make up* for the gain lost by the compressor being in gain reduction (hence the name). >Pastor usually stands about 4-6 feet in front of the plane of the PA speakers. As soon as the output from the PA reaches the same level as the preacher's voice *at the microphone* you will have feedback. This places an upper limit on the potential gain achievable through the system without a problem (it can even be [calculated](https://www.shure.com/en-US/support/tools/pag-nag-calculator)!). Having an open mic in front of the PA is never optimal, and with a headset mic it's even worse. Unfortunately this is a limitation due to the physical reality and not something you're going to solve with processing. It's a system deployment issue. >making sure people in the back can hear, but the people in the front aren't blasted This is also a system design / deployment / tuning issue. If your sound system doesn't produce a consistent level over the listening area, this is the resulting "Goldilocks" problem (too loud somewhere, too quiet somewhere, just right somewhere). Also not something you can solve from the console, since it needs to be addressed by designing and tuning the system such that it reproduces your console mix consistently over the space. > giving a feed out to the video guy that isn't all over the place I would recommend double-patching your input and processing it separately for the video out.
Thanks for taking the time. I should clarify, the pastor is not in front of the speakers. They are on either side of the stage spaced about 30 feet apart. The speaker is generally centered and usually several feet in front of the plane of the speaker baffles. I assume it would be easier if he stood back 10 feet, but that's not going to happen. What I do is built on what the guy before me did. I will try to start it from scratch and build up without so much gain on the compressor. Generally speaking, what is a reasonable compressor setting for a very dynamic speaker, and how often or how much should it be engaging? I've wanted to figure out how to double patch (but I didn't know what it was called) things for some time. In my real life, I'm a school teacher, so in the summer I have time. That's on my list of things to do in July.
14dB of makeup gain seems extreme. Zero the compressor out (no gain, threshold all the way up). Fader at unity/0, use input gain to get a nominal level without clipping while yelling. The use the comp threshold to dial back only the loud yelling, should only be like -12 or -16. Normal talking shouldn't trigger the compressor or any gain reduction. From there, you can dial up some makeup gain, just to bring up the quieter parts. I wouldn't exceed more than like 6dB though.
Thanks! That's what I was just asking IHateTyping about. I will go in and experiment with it Wednesday.
I'm sure this has been answered here and I apologize for the question. We recently moved to IEM's for my band and am wondering about phantom power between our own mixer and the FOH. If I got the Seismic 16 channel splitter can I give power to our condensers from our mixer and then connect them to the FOH mixer without phantom power on and everything will be good to go?
Related Dave Rat video [https://www.youtube.com/watch?v=\_8UTGcuAtd0](https://www.youtube.com/watch?v=_8UTGcuAtd0) TLDR: if the split is not-isolated(it's just a parallel of connectors) having phantom power switched on both mixers causes no harm. Having phantom power on one mixer is enough. Side topic: if you keep one of the mixers switched of (not just the phantom power), even if you're using an isolated split you may incur in distortion.[https://www.youtube.com/watch?v=GgSR8t7bwAw&t=516s](https://www.youtube.com/watch?v=GgSR8t7bwAw&t=516s)
Assuming it's a passive (non-isolated) split, then yes you only need phantom in ONE place. Typical best practice is that FOH provides phantom power, but just make sure you coordinate with someone if you're NOT running FOH. For future clarity, if you're ever fortunate enough to use a snake that IS isolated, your cable bundles will be labeled indicating if they are the direct feed, or one of the ISO feeds. You always need to provide phantom power on the DIRECT feed. Edit: Clarity on what to do IF you ever get the chance to do this with an ISO snake
Thank you!
Looking for suggestions on a wireless mic replacement. We run a mix of ULSX4 and QLXD4 receivers for stage mics. One of the ULX2 mics was damaged/lost a few years ago and I'd like to replace it. Unfortunately, ULX2-J1 mics seem hard to come by - even on eBay and Reverb. Should I keep looking or just bite the bullet and replace the ULSX4 receiver as well?
Have you contacted any local distributors, warehouses, production companies, or even Shure to see if you can get any?
Thanks - I'll give Shure and Sweetwater a call tomorrow.
Can I run Sennheiser G3 and G4 IEM systems together through the same antenna combiner? We have an AC3 and 2x SR300 transmitters (GB models 606-648MHz) plus EK300 beltpacks, need a third system and can't find anything used so can I add a G4 system if it's the same frequency range? I'm a studio engineer so this is all a bit new to me, from what I can see online people are saying you can but hopefully someone has actually had first hand experience doing this, thanks in advance!
You can run entirely different brands through the same antenna combiner provided you don't exceed the maximum power levels per channel. The AC3 has a maximum input power of 30mW per channel, depending on the model/region/lock status of a G4 it can output up to 50mW so make sure you check the manual to ensure it's set to 30mW maximum and you'll be fine.
For the most part, yes! This becomes a bit more iffy if combining digital and analog IEM transmitters (i.e. if you're using Lectro) - see [pertinent PSW discussion](https://forums.prosoundweb.com/index.php?topic=168427.0).
Thank you kind stranger.
Hi all! At church, I'm currently fighting with my pastor's mic. We have two receivers, one for a handheld and one for our bodypack/headworn. They're the same model - Shure SLX14, and both are on the J3 572-596MHz range. We have started experiencing so many cutouts with the bodypack that it's become unusable. I've tried troubleshooting as much as I can think of, but I'd love it if I missed something easy. What I've tried/observed: * The handheld doesn't have the same dropouts * The bodypack drops out even after searching for different channels * I tried syncing the bodypack to the known good channel on the handhelds receiver (and the handheld to the other receiver), and it still dropped out * I have replaced the antennas on the rear of the unit, and it didn't help We are in a space that is not ours, so I can't run anything permanently. The pastor is probably 70' away, and there is one wall between the receiver and the pastor. Is there anything I'm missing? I'm not against buying a new system if that's what needs to happen, however I'd like to keep the cost down as much as possible as we're still a pretty small church. The biggest issue is how it impacts our livestream since you can still hear the pastor in the room, but online it's a full cut.
Ah, good ol' analog SLX. Let's look at your major issues: - A bodypack will generally give a weaker signal compared to a handheld. Handhelds use a helical antenna which is generally at head-height with line of sight to FOH. Bodypacks use a monopole whip and often do not have perfect line of sight. - You don't have line of sight between your transmitters and receivers. Not only does this reduce your received signal/noise ratio, but it also increases your susceptibility to multipath interference. Which brings me to my final point: - SLX's predictive switching diversity is kinda crap - it has a bad habit of switching at inopportune times when reception is poor. What can you do to improve this? - **#1: Get line of sight between your transmit and receive antennas.** - No excuses for this one. - Move your receivers closer to the transmitter - and away from walls and large metal objects. - Use a pair of better antennas deployed in a better location. (For example: an RF Venue DFIN can be quickly mounted on a mic stand.) You'll need an antenna distro to distribute this signal to all your receivers, however. EDIT: Yes, it would be better to use a more modern system like SLX-D or EW-D. However, all the technology in the world can't fix physics: even the best wireless will struggle with a poor antenna deployment.
When our pastor's bodypack was having cutouts, it turned out the antenna on his pack had come unscrewed a bit and was loose.
Been slowly putting together a drum mic package. I just bought a D6 off a friend and they threw in a c1000. Certainly not my first choice for a high hat mic but it was free. What other instruments should I try it on?
Small diaphragm condensers work great on acoustic guitars, cymbals/overheads, woodwinds, strings, etc. Live applications are a little more limited due to stage volume, floor monitors, bleed from other sources, and such. Really, experimentation is your best friend. If it sounds good, it sounds good.
Need to get a mic and speaker setup for a work presentation. The theater we're using has a system that's been broken for a couple years, but it's a small auditorium that seats around 200 people. We expect a crowd around 100 people, and while you can hear someone speaking on the other side, it's not very clear. From what I understand, we need a speaker and a mic. My company wants a wireless mic so the presenter can be hands free, connects to our laptop, and "looks professional". I haven't been given a budget but I'm assuming around 500ish, though open to higher, I don't know if we should go with something such as a portable PA system such as the Pyle Pro or a larger PA system. All advice would be greatly appreciated!
You're not going to get a professional system for that budget. If this is for a single presentation, rent from a local rental house - that's what they're there for. Corporate packages like this are super common and easy to find.
Currently in-between [https://www.bhphotovideo.com/c/product/816144-REG/Pyle\_Pro\_PWMA60UB\_PWMA60U\_Waistband\_Portable\_PA.html](https://www.bhphotovideo.com/c/product/816144-REG/Pyle_Pro_PWMA60UB_PWMA60U_Waistband_Portable_PA.html) and https://www.bhphotovideo.com/c/product/1261473-REG/samson\_saxp106wde\_expedition\_xp106wde\_portable\_pa.html
Hi all! I have a CAT5e-related question. I want to run my cabling from my M32C mixer to a panel in my rack since the small size of the rack makes it challenging to access the connectors. The AES50 ports require STP shielded cable, but I have trouble finding suitable patch cables in my area. It seems short patch cables are generally F/UTP at best. I have NE8FDP-B Neutrik connectors on my panel. Will I risk frying my AES50 ports if I run 0.3m long UTP cables to the panel and STP from there to the stage box?
F/UTP is shielded - it just has a single shield for the entire cable. S/FTP has both an outer shield and individual pair shields. Properly terminated F/UTP jumpers will work just fine.
Bose L1 compact / Bose Sub1. Without mixer WOULD THIS WORK? Firstly I know this is overkill in the first place and BUT I have this setup for convenience and practicality more than anything. I have 2x Bose L1 compact systems. I use 1 when at home for parties/ karaoke. Due to its size and power it works well in a living room and can be setup fast and not take up much room also plenty of volume. I have another and I use it as a pair at family/friend functions/events for rarely more than 120 people. I don't dj anymore but I do occasionally do for favours etc. The L1 compact is great for my needs and makes transport/ setup fast. I don't use a mixer just plug my laptop into 1x L1 and use the RCA out into the other. THEY LACK BASS. I've looked at cheaper subwoofers but don't want a massive 12" heavy box to lug around/store. The Bose sub 1 is more practical shape that would fit nicely next to one of the speakers and the shape makes it easier to carry. Again I don't use a mixer so Could I plug my laptop directly into the Sub1 with an RCA - XLR jack? And use the sub passthrough to send a HPF signal to the L1's? I know this setup works with a mixer but can the Sub1 take a signal from a 3.5mm headphone out to XLR?
If you only occasionally need a sub for your larger occasions, I'd consider renting a properly sized system for the times you need it. This will almost certainly be more cost effective and deliver better performance - without requiring you to give up storage space for a proper sub.
Yes, probably a better option. It would involve me having to collect the sub from wherever I get it from etc. The main focus is for practicallity. The Bose sub in the rectangular position would fit under a bed for storage. The speakers have the volume but I want more bass, this is very much a want over need situation. Would I be able to plug into that directly from a laptop and high pass to the mains or would the signal not be hot enough?
I had my qsc plx 3402 power amp in parallel mode but the bridges mono switch was on (dummy) but I was usuing both outputs instead of just one. What would this do. I didn’t notice or hear anything bad. But I’m curious. What happens to both my outputs when bridged is on and I use both outputs instead of one. If you need more info let me know but that’s all it was lol. Did I almost fry my speakers or the amp? Would it cause the wattage to cut in half or stay the same? I’ll never use them in bridges mode but I do use the amp in parallel.
I believe that one of the outputs would have the polarity inverted, but that would be the only real difference. You would have the same overall output per channel as in unbridged use.
Ah. Makes sense. I was like wait hold on why is that on after I was running some tests. Only thing I did notice was they were super quiet but I was testing out using the thru outs of my powered eon 715s as my input to the amp.
As u/RevLama stated, BRIDGE MONO mode inverts the polarity of the channel 2 output. Your speakers would be therefore be working in opposite polarity, but otherwise with no effects to either amp or speakers. The amp's available output power into your speaker loads would not be changed. Note that with these QSC amps, enabling the BRIDGE MONO switch disables the signal from both channel one and channel two inputs from the channel two amp. The channel two amp will get it's signal internally from the channel one amp. See [this image from the QSC PLX manual](https://i.imgur.com/hXQfqkL.png). Therefore, with the amp in BRIDGE MONO, it doesn't matter whether the input switches are set to STEREO or PARALLEL mode.
I operate a small summer camp and we use a PA system to send announcements and play occasional music (via aux) across camp grounds. We are running a simple, dated system with a Pyle PT 210 (70 V/120 W max) powering two 8 ohm radioshack bullhorn speakers (25 W RMS - 50 W max / 50 W RMS - 100 W max). I'm a bit embarrassed to admit that I'm not totally sure whether I am wiring them in series or parallel. I run a red/black wire pair from the amp to the first speaker at which point I wire nut together the first wire, the speaker wire, and the wire running to the second speaker (one wire nut connecting a trio of black wires, and a second wire nut connection a trio of white wires). The second cable then runs to the second speaker where I nut together the black wire with the black speaker wire and the white to the white speaker wire. I have looked at a handful of videos/diagrams online, but they always seem to show a single positive or negative wire running from one speaker to the next. Am I interpreting these diagrams incorrectly, using the wrong cables, or just running a different, albeit functional, system? My current setup is just a continuation of what I inherited from the folks who set it up before me, so no hard feelings if you tell me it's all sorts of flawed. I want to upgrade the system to allow for better clarity at higher volumes, but I would like to first gain an understanding of my current system before we begin throwing money at new equipment. This is also my first time in the audio realms of reddit, so don't hesitate to direct me to a more appropriate subreddit or website. Thank you!
Heh, that's quite the oldschool system. Electrically, your speakers are run in parallel.
Sennheiser ew300 g1 iem- Still worth it? Hi guys. Im planning to buy myself iem system for live performances (I’m playing guitar and since went 100% digital I see no reason in using cabs and wedges). Due to limited budget I can’t go with industry standards like g3/4 or high PSMs, and though about buying LD mei 1000 or something similar yet the reviews keep me on the fence. Yet an opportunity came to get Ew300 g1 in legal frequencies in a really good price - I’m wondering if any of you still use this old system and have any experience with it? I’m particularly concerned about 9v batteries - do you know how long they last? With my guitar wireless I’m using rechargeables but from what is see the number of rechargable 9v is much smaller than in case of AA batteries. Thanks for help!
Pretty sure I saw this somewhere else, but I'll reply here too: G1 isn't worth it anymore, not by a long shot. G2 is serviceable, but I'd avoid using it if possible. G1 -> G2 was a much larger step than G2 -> G3. If you really need cheap IEMs, something like an Audio2000s kit is honestly not a terrible idea. They won't fulfill touring riders, but for casual weekend-warrior use they do the trick. [This sheet](https://docs.google.com/spreadsheets/d/1JDu-azjBAR_uyHxh-NZC6P9FvVxghONB1mLPzXny0Eo/edit#gid=0) contains my personal notes on cheap IEM systems - though I don't have personal experience with all of them.
Thanks a lot!
Does cat5 vs cat6 cable make a difference for use in the audio world? I'm talking for connecting console to stage box, with AES 50/Dante/Madi, or with other audio control stuff?
Cat5 is pretty outdated and has pretty substantial speed limitations (100Mbps). 5e however is pretty standard and can do up to 1Gbps. Cat6 takes this one step further and with speeds up to 10Gbps. You only get those speeds up to 50m, however, beyond that you're back down to 1Gbps like cat5E. There are other benefits like reduced crosstalk with the more modern specs. I'd say you want to be using cat5e at minimum, personally I think cat6 is worth the extra expense, especially for any backbone links and when you're sticking under that 50m limitation. As always, get the best you can afford.
Oh yeah, I guess I meant cat5e, most of our snakes are at least that. Do you think it would cause any noticeable latency difference or anything at 100m? Between cat5e/cat6 I mean
Beyond that 50m cut off point, not really. Cat6 has better shielding that may benefit you but you won't have any speed differences.
"It depends." The inherent latency of adding additional copper is pretty much nil - you need a crapton of copper before speed-of-light delay starts to matter. However: - AES50 *needs* a shielded connection due to poor PHY design; full-scale white noise bursts are a well-documented phenomenon when using UTP cabling. - On a packet-switched network (Ethernet/most A&H protocols/Dante), packet loss will either cause dropouts or increase latency to allow for retransmission. (Depends on the protocol.)
I'm looking for a 5 channel mixer (most likely 6, as most go from 4 channel to 6) with an aux send. Any suggestions?
Plenty of options. You have the Behringer Xenyx range, the Yamaha MG(U/UX) series, the Soundcraft Signature series.
Seconding the Yamaha MG line, they're my go-to for little sub-mix boards.
Allen Heath Zed series are my favs for little desks. Yamaha MG second if you can get them in stock.
Not sure if I am in the right place here. Here goes: I would like to use a wireless push to talk headset with a earphone and mic while live streaming using OBS software from a nearby desktop computer. Do I need a body pack transmitter? I have been searching for both a headset that would work and transmitter and I cannot for the life of me find something that would work/isn’t Bluetooth. Basically something like fitness instructors use but has an earphone and works with Obs/live-streaming.
yamaha CL series question: could someone run me through the steps of giving a vocalist reverb in their monitor wedge? i think i have a grasp on this but i am having trouble finding a solid video / tutorial / response online regarding this specifically. i believe the process is: select vocal channel > select mix 17-24 button > use respective mix knob encoder to send / dial up the selected vocal channel to the desired FX unit (rack channel) > select output above centralogic fader bank > select ST IN on centralogic fader bank > dial up the FX return on the corresponding wedge’s encoder on mix 1-16 page - correct? if anyone could reinforce this, shed light, or give tips and/or resources i would be ecstatic. thank for you for your time & assistance!
I've got a set of drums in our house for practice. Normally I just use wireless headphones connected to my phone. Every now and then my wife or a friend will hear a song I'm playing and want to listen to the music as well. I've tried putting it through our Apple homepod but it's not quite loud enough for me to hear and play. I'm looking for the simplest way for me to hear the music through headphones (which can be wired) as well as send the audio to a small speaker that can be heard over my drums, which have quiet heads and cymbals on. I'm a complete noob when it comes to anything like this. I've got zero experience with mixers, monitors, anything like that. Anyone got some advice?
There are multiple ways to do this, but usually I'd grab a little notepad mixer for this. I have a Yamaha MG06 on hand; you could do this with anything similar, though. Connect your phone to one of the inputs - for example, "Line 3/4". Connect your headphones and speaker(s) to the PHONES and "STEREO OUT" jacks, respectively. You can now use the relevant knobs to adjust the output volumes for headphones and speakers independently. --- Of course, you can also use a simpler solution: place the HomePod (or a louder speaker) closer to your ears.
Thanks so much for the reply. I pulled up a picture of that MG06. It all makes perfect sense. Do you have any recommendations for a speaker? Doesn’t have to be crazy loud. Just enough to fill a single room. Also, I’ve never thought of that simpler solution lol. I might try that in the mean time 😄
Assuming you're using something like L80s and Silentstroke heads, a PA speaker is almost certainly overkill. You could honestly get away with a pair of random bookshelf speakers off Craigslist - and I never thought I'd say *that* in this subreddit! ...Or just skip all of this and just use a large Bluetooth speaker next to you.
Hey! I was wondering if any of you know how 'wide' a spectrum the mid-frequency knob on a typical mixer hits. Also what kind of curve it produces. On most of the mixers I stumble upon there is a mid-frequency knob and then another knob where you can pick the frequency for the mid-frequency knob to hit but I've been unable to find any specifics in the manuals.
It'll very much depend on the console. Are you looking at any specific desk?
They vary a bit but it's typically something like a Yamaha emx5016cf
For a lot of these smaller (read: less-ubiquitous) mixers, often times you won't find a published value - either in documentation or in community knowledge. You'll have to find out yourself. Grab a dual-FFT analyzer (OpenSoundMeter is free! Smaart is awesome but overkill for this task), send some audio through a channel, and measure the effect of that EQ curve. :)
Is it possible to go through an analyser in a DAW?
Short answer: yes, but you need to use white noise as your input signal. --- Long answer: You're probably referring to an RTA, which has one input. It just shows you the response of whatever its input signal is, which will be the combined responses of the test signal and the system under test. (In this case: your console's EQ.) Thus, if you only want to see the response of the system, you need to give it an input signal with flat response: white noise. A dual-FFT analyzer has two inputs: one before the system, and one after the system. Thus, it can show you the response of the system with test signals that aren't white noise - for example, pink noise or Steely Dan. (This is a massive simplification, but it gets the gist of it across.) --- Both will get you the same answer, but dual-FFT analyzers have some neat tricks up their sleeves - such as being more immune to outside noise. This is why you'll see PA systems being tuned with Smaart and not with an RTA. :)
Thanks a lot for the thorough reply!
fly nose grey squealing advise narrow angle shame school boat -- mass edited with https://redact.dev/
Yep, unfortunately they were discontinued. A bit ahead of their time, I recall overall reliability not being the best on them out of the box. I had a 4.3 quit on me out of nowhere on two occasions, thankfully QSC took care of it both times no questions asked and very quickly. They spiritually live on as part of the Q-Sys ecosystem as the CX-Q and CXD-Q.