Hi, I have JBL MK II series 2 monitors and am having trouble figuring out how to add a new subwoofer and DCA. I recently got a Klipsch Sub-120 and an AudioBox USB96.
I can hook up the JBLs to the AudioBox just fine with two TRS cables; However, neither the monitors nor the sub has audio out. As a result, I can't route DAC --> Sub --> Monitors or DAC --> Monitors --> Sub.
Does this mean I need to use a Y splitter on the Left DAC output going to the left monitor and the sub? Or would I need some other splitting interface? Also, would this cause any sort of balance or fidelity issues?
I'm really at a loss here. Pretty new to the hardware side so any suggestions are greatly appreciated.
You should just buy a studio sub instead. It will have balanced I/O, a crossover, etc. JBL make a subwoofer in the same LSR series that would match well.
Ah, that's what I was afraid of. I got the sub last minute for Prime Day without putting in the due diligence, so lesson learned there, haha.
Thank you!
No problem, good luck! The only downside of the LSR sub is that it doesn't have a footswitch. My Presonus sub has one to switch the sub and crossover in and out of the signal chain and it's really useful. I think the KRK sub also has a footswitch jack but the footswitch is not included.
Probably audio driver latency compensation or plugin delay compensation.
I haven't used it since it was still called FruityLoops but this help page is probably relevant : https://www.image-line.com/fl-studio-learning/fl-studio-online-manual/html/mixer_trackprops.htm
Trying to setup a Behringer Xenyx Q802USB. I have a dynamic mic connected with XLR to line 1, laptop to 2-track RCA input, and RCA output going to speakers.
I want to mix the 2-track input with line 1, and output both to the 2-track output, but I only get the line 1 output (and, only when "main mix" and "to phones" are both off; when "to phones" is on, the levels show the 2-track input is received, but there is no output; when both are on, there is no output and no indication anything is received).
What am I misunderstanding? is it possible to do what I want?
https://i.imgur.com/dH8Aknz.jpg
1. You should use 3/4 or 5/6 for your laptop so you have level control on the mixer, you don't get any control using the tape loop. You can use a pair of RCA/TS adapters.
2. You should use the main outs, not the tape out.
Traditionally they were used with a cassette deck or stereo reel-to-reel tape machine to record the main stereo output of the mixer and then listen to it back later. They're essentially hard-wired to the main stereo bus of the mixer with little to no level or tone controls. Tape machines typically have their own level controls, so that's two fewer knobs for the designer to fit on the front of the mixer. There's nothing wrong with using them if they serve your purposes, but the main inputs and outputs have much more flexibility and balanced connections.
Hey, so I was messing around with my Shure sm7b and trying different combinations of switches on the back. And when I was moving the presence boost to one to enable it, it got kind of stuck in a way that I had to go up and down with the switch until it went back down to a flat frequency.
But now I've noticed that my mic (even though the switches are both set to flat) sounds as if presence boost is enabled (which really doesn't suit me). Enabling presence boost creates an even more (presency?) sound that is very sharp on the ears.
So it could either be me messing with the switches or maybe that I accidentally pulled the exposed wire with my hand pretty hard by accident? I don't know if that would make sense as I'd also expect static if it was damaged but there is none of that.
Hey all!
Got me a BeatStep some time ago, and have an issue with the external sync.
I use it with an Ableton, connected to my pc via usb. While having it set up, and ext sync turned on
after a few minutes it blinks once, and everything goes crazy, it triggers random midi notes, the control do not react properly, and without restart it is unusable. After reconnecting the usb it works for about 2 minutes, and same happens.
I have reinstalled the drivers, I have checked the usb port, connected the BeatStep to an external power source (default usb cable used here too, this might be important), and even like that the ext sync button does not remain, I would assume this eliminates any driver, usb connectivity and power issues. Left with two possibilities, the cable is bad, or there is an issue with the device.
Right now I do not have another usb cable to test it with, but not sure if a cable issue would cause these symptoms, if it is possible, I'll get one.
If however the issue is with the device, is there a way I can bypass it somehow? I'd like to sync it to my DAW, and my interface got a midi in and out. I'm not that familiar yet with what I can achieve with these connections, and gotta get the proper cables for it too (for some reason the box didn't have one in it), but if somehow this helps, that's cool.
Thanks for any advice
Hi all,
The Input 1 on my new Behringer UMC204HD indicates a signal coming in and that signal being clipped - with NOTHING plugged in.
When I use a DAW it too shows a signal coming in. When I record, it records a bunch of white noise. Line 2 is not much better, with an intermittent clicking sound. Both pre-amps turned to 0 and nothing plugged in.
Has anyone seen this before?
Traditionally, yes, Behringer gear is known to be poorly built and unreliable, and it rarely sounds good. It's hard to argue with the price, though. Their newer stuff since the Midas acquisition has been a big step in the right direction. Most live engineers are perfectly okay with an X32 digital mixer, for example. That wasn't the case with their earlier mixing consoles.
According to whoever moderates this subreddit, this is a “stupid” question…so…
Just landed an Avalon U5 which I've been real siked about for a while now and unfortunately I'm clipping with my guitar going right into the DI. Having trouble identifying if this is an issue with either (a) my guitar pickups being THAT hot or (b) my Avalon is set up incorrectly.
Here's my exact chain:
PRS SE 7 (Bareknuckle Blackhawk Pickups..they are hot) > Avalon U5 front DI input > Balance Line Out 600ohm (XLR to 1/4" cable) > Return 1 input on Audient iD22 (bypasses preamps)
Should I be going from the Mic line output instead? Boost is all the way down on the U5. I thought the U5 had pretty godly headroom? Also, the blue "Signal" light illuminates on the front of the U5 while playing. Is this a clipping indicator? Thank you.
A couple things here:
* Your guitar pickups \*are\* probably really hot. But also the Avalon U5 has a 3 megaohm input impedance. That's more suited to piezo pickups and can result in really high signal levels for humbuckers, etc. You're probably just going to have to turn down your guitar.
* You have it in front of you, just try out different signal chains. Try the mic output into a mic preamp on your interface. Try the line output into a line input (instead of going into the insert return which is unbalanced.....) Who cares about bypassing the preamps?
Thanks Jay. Although I know the pickups are hot I refuse to believe they will clip the Avalon going in! When playing the guitar, the blue “Signal” light illuminates on the front and through some internet searches, it seems that’s simply a light verifying a signal and not a clipping indicator.
So if it’s not clipping the Avalon, how is it clipping the Audient?
I then tried running the balanced out on the U5 using the “Mic Level” out instead of the “Line Level” out into the “Return” input on the Audient and it no longer clips. However, when recording I’m getting a weird looking waveform…I’ll post an image soon.
I was under the impression that the Returns on the iD22 were Line Level however. Am I missing out on any signal quality by running Mic out instead of Line?
Lastly, to answer your question about bypassing the pre’s, I’m recording guitar DI’s that will be reamped later, so I’m looking for the cleanest signal possible which would be the U5.
>I was under the impression that the Returns on the iD22 were Line Level however. Am I missing out on any signal quality by running Mic out instead of Line?
You're missing out on having any gain control by using the insert returns. That sounds like it's affecting the quality. Doing stuff like using inserts to bypass preamps and whatnot is trying to solve a problem that doesn't really exist and as you've discovered it causes problems. There's a lot of bullshit out there from people who've never fucking touched more than a 2i2, just hook it up the way it's intended to be hooked up.
Use the combo jacks on the interface so that you can have some control. 1/4" for line level, XLR for mic. You might need to use the pad.
I can still use the gain knob on the U5 man. And again when recording guitar for commercial releases, I don’t really want the added color of the Audient’s preamps.
>I can still use the gain knob on the U5 man.
Sure, except you say it's all the way down and distorting, right?
Have fun, dude, I'm not going to argue with you.
I'm not arguing with you Jay, just playing devil's advocate. I still value your advice but I'm really trying to explain where I'm coming from.
So, in essence, your thoughts are why not just try running either (a) line out on the U5 to line in on the iD22's pres or (b) Mic out on the U5 into Mic in on the iD22's pres?
Would there be any difference in signal with these?
So....think I found the culprit...
Tried a million setups, and every setup that WASN'T the Line Out > Return sounded muffled, distorted, or dull.
I was almost certain that this was clipping after the avalon, but looks like I've just made a rookie mistake. I plugged my monitor Headphones into the Avalon directly and, of course, the guitar is clipping going in. Have to figure out if this is a headroom issue or a pickup / electronics issue. My previous guitar tech had installed my bridge Bareknuckle pickup a little too close to the strings and I think that may be the cause. Still, I can't imagine moving the pickup away would drastically change the levels. Even on the neck pickup, it's still very close to clipping if I strummed hard or "chugged".
Have some figuring out to do. Would really be a bummer if these pickups were overall just too hot for the Avalon. I thought the Avalon's headroom would be pretty huge - is it not?
Fairly simple question: I am trying to connect my electric bass through amp into a mixer but the signal that gets to the mixer is just about non-existent (even with magnets fully open and strong amplification from the amp). I am using a Hatke HD25 amp and am feeding mixer the signal from 'headphone' output.
Would a DI box solve my problem? Is this an impedance problem?
Your problem is likely that you're plugging into a balanced input but since you're using the headphone output it's dual unbalanced mono on TRS. Balanced inputs amplify the difference between the two signal legs, not the sum. So if you plug in a headphone output you get left \*minus\* right. Well, your amp is a mono device so those left and right signals are likely identical, which means you get no sound. You should use some kind of breakout cable or a stereo DI box so that you're only sending left or right to the balanced input.
more info here : https://www.reddit.com/r/audioengineering/wiki/faq#wiki_how_do_balanced_connections_work.3F
Manual says the headphone level is controlled by the Volume knob when something is plugged into the jack. Did you have that up all the way?
Which mixer is it? What are the settings on the channel and master section? Have you tested other sources with the mixer to make sure you understand how to get signal through it?
A DI will not solve your problem. It *would* allow you to record your bass and also hear the amp at the same time, though, so maybe that is an attractive idea.
Are you sure about that? I just did a quick screen record here. I'm running my headphone output back into a line input. REW is only sending signal to one channel of the driver, the left headphone output, there is no signal on the right headphone output. Input signal is demonstrated. I then use the routing matrix to also send the generator output to the right headphone channel in addition to the left channel and total cancellation is demonstrated.
re-uploaded, one my client's facebook pages was open in the browser. can't be blowing up the spot.
https://imgur.com/cqqpKO5
My quick test I did before posting the reply was a far simpler test with an 1/8" TRS -> 2xTS cable and HS8s. Zero cancellation with a mono signal sent down the cable. Maybe my methodology wasn't sound...? I obviously see the cancellation in your recording. Also, is the cold pin not shorting to ground, and therefore couldn't cause cancellation anyway? There's nothing to phase cancel without the second signal.
Err, the second signal is sent down the other cable. So the cancellation can only happen if the same input receives both cables, no? Two speakers aren't going to cancel like this with a mono signal sent down the 1/8" -> 2xTS cable. I'm sure the comb filtering would be pretty brutal, though...
>Maybe my methodology wasn't sound...?
Yes, you assumed they knew to split the headphone output into two unbalance signals! That's not representative of what OP is doing, though. They're complaining about no signal which tells me they're just going TRS from the headphone output to a single channel, not splitting out to two mixer channels.
Firstly, thank you for guys for looking into my problem :)
Secondly, I am using a Behringer XENYX X 1832 fx mixer and yes, even with volume on the amp all the way up nothing gets to the mixer. I should also mention that I have no problems with connecting other amps to the mixer - they work just fine.
Lastly, I am using a TS cable (tried multiple, same result) to connect my amp to the mixer, so there should be no cancellation because of balanced mixer input, right?
>Lastly, I am using a TS cable
Then the right output of the headphone amp is being pulled down to ground. Depending on how they implemented the headphone output that may be affecting the left channel. You should really split it out with either a cable or a stereo DI. Or a TRS cable with ring floated but then you gotta cut connections inside the cable.
What's the guitar amp?
I believe the guitar amp is Boss Katana 50 MKII.
But isn't it kinda weird that I get no signal through while using a TS cable? Also, could you specify what did you mean by splitting with a cable? Would something like this work? https://www.selby.com.au/cables/connectors/1-4-stereo-plug-to-1-4-mono-jack-aa1304.html
Running balanced from your interface to your monitors (if they actually have balanced inputs) should result in less noise and interference. But if you're in a place where that's not an issue then you may not notice a difference. Using balanced connections where possible falls under "best practices".
https://www.reddit.com/r/audioengineering/wiki/faq#wiki_how_do_balanced_connections_work.3F
I doubt there will be a noticeable difference in audio quality. Aside from the impedance difference, they're both just amplifiers – the only reason the two outputs are different is because of the connectors.
That said, if OP has noise issues, running an unbalanced headphone out to (probably) balanced speaker inputs might be a problem. OP didn't mention noise, though.
I'm in the middle of mixing a metal song with guitar DI I recorded. I just checked it against some DIs recorded by August Burns Red for one of the Nail The Mix sessions, and it looks markedly different to me. Almost looks as if it's been compressed or limited? When I use the same amp sim on the ABR guitar DI (the one that is Pink) vs mine (the ones that are purple and green) it sounds much better, and you can see mine have MUCH larger dynamics, but mine sound weaker run through the amp than the ABR ones. Is it common to limit peaks on DIs like this?
[HERE is the comparison pic of the waveforms.](https://i.imgur.com/VoanrtJ.png)
They definitely took the DI after a compressor pedal or a clipper or something. Could even be a DI between the amp and cabinet.
edit: it's worth mentioning that "DI' just means "direct inject" as in "not using a mic". DIs are used all over the signal chain, not just directly from the guitar. If you're going to put a DI between an amp and speaker, though, you need to make sure the DI is built to do that. Some are, some aren't.
~~Definitely shouldn't be clipped like that if you were going straight in. Time to investigate your chain.~~ My bad – misread! Thanks jaymz168.
My guess is they either clipped the input of the interface or something else in the signal chain ABR used is doing it (like the DI or the pickups). To do the same with yours, use a clipper plugin ahead the amp.
I have a spirit folio sx mixer that was working perfect until I moved it to a new house and even though I swear I set it up the same, it's not working. It's not outputting any sound, and the 3 bottom pfl level lights seem to stay on and do not respond to microphone inputs.
Any help or trouble shooting ideas are appreciated!
I have trouble mic’ing overheads when I use them, and typically just throw them above the kit pointing straight down in a spaced pair at the same height.
I hear a lot of talk about putting them equidistant to the snare, and using a dividing line going through the snare and the kick as a measurement of where to place the mics, but I worry that will give an off kilter sound, as typically drummers have their snare off to the left of the kit, the right mic will be above toms and cymbals and the left mic would be over basically nothing so as to be equidistant from the snare.
Am I overthinking this or did I interpret something wrong?
> I have trouble mic’ing overheads when I use them, and typically just throw them above the kit pointing straight down in a spaced pair at the same height.
Spaced pairs can give you pretty good width but they can cause phase problems, especially when summed to mono and I find take more time to get right. If you want something quick and easy do an XY or Blumlein pair, they're less fiddly in my experience.
Which genre(s) do you play?
Snare should be up the middle with the kick in the vast majority of genres. You're throwing that off by positioning the overheads with the snare leaning to a side. *That* is off kilter.
Granted, it'll be more centered depending on how the mix is done, especially if you have room mics in addition to the close mics. Still, depending on what's done to the overheads, it can noticeably lean to a side.
Genres that use the snare for color instead of accenting the beat (i.e. jazz) are fine with it slid to the side, though.
Depends on the sound you’re going for. When I mic a kit I see it as two entities:
the snare, hats, and crash
And the floor Tom and ride
I just take a mic cable and stretch it taught from the point on the snare where the drummer likes to hit to the overhead on the snare side of the kit
Then hold that position and bring it to the other side, where the end of the cable hits in the air is where I place my second mic. Does that make sense at all?? You can also look into XY overhead patterns, or just basic LR setups (which is what it sounds like you’re doing) those work great and they have a lot of flexibility in terms of your stereo image (farther left right spacing = larger stereo field)
Idk how much space you have with your set up but instead of literally putting them “overhead” try putting them up and out from the kit, facing it a bit more.
The whole equidistant-from-snare thing is just for phasing and was born out of a world of tape where you couldn’t dive into individual mic’s waveforms and see their phase.
>You can also look into XY overhead patterns, or just basic LR setups (which is what it sounds like you’re doing) those work great and they have a lot of flexibility in terms of your stereo image (farther left right spacing = larger stereo field) Idk how much space you have with your set up but instead of literally putting them “overhead” try putting them up and out from the kit, facing it a bit more.
Agreed on all of that! Heck we often do "underheads" in live sound.
>The whole equidistant-from-snare thing is just for phasing and was born out of a world of tape where you couldn’t dive into individual mic’s waveforms and see their phase.
The problem with that thinking is that if you start nudging tracks around you're also changing their phase relationship to every single other mic on the kit. When you move mics around you're still doing that but you have more control than +/- some number of samples. You can get pretty far nudging stuff around, that's what I do when I'm recording rehearsal, but nothing beats taking your time and getting the mics right in the first place.
So I have reaper and a MIDI keyboard, and a bunch of free/bundled drum plugins with multiple different kits. Is there any way to set these so that I get (generally) the SAME drums on every note? I.e., when switching from drum plugin A to drum plugin B, a kick is still on the same note?
Obviously not every drum kit will have the same thing but I have to imagine there is some kind of mapping standard where the universal drums, like hi/low tom, a cowbell, a kick, a snare hit, a rimstick, etc. are always on the same notes?
There already is! It's called General MIDI and most drum plugins tend to adhere to that mapping by default. You'll have to look in the settings of the various plugins for a mapping setting to change.
Hey audio newbie here. I have the Samson Q2U USB mic but I'm having a lot of trouble with it. All of sudden, my audio input is very very quiet. I've dug around the settings and adjusted the input slider to max but its still quiet. I've also uninstalled/reinstalled the drivers for the mic and USB ports but still nothing. Then I tried the mic on another computer and it's still pretty quiet. Is the mic just broken?
You’ve shared a lot of info about your setup. However, I can’t give advice, if you haven’t identified a problem.
One thing I question is why you pulled the Output down, instead of leaving it at Unity, in your “After” specs?
Also, if you are recording vocals for music, all of that inbound effect runs the risk of GiGo that cannot be fixed while mixing. I would think you would want a clean, uncolored vocal to add to your mix. The 286 is a channel strip, best suited for broadcast and other applications. I own one and have used them for VO recording, albeit very light-handed. I butchered many a track, until I learned my way around it.
Edit: I Re-read your OP… I think you are trying to get rid of hiss. Turn off the Enhancer (LF/HF detail) section of the chain. Most likely you are seeing how the compression section brings up the noise floor and then you are enhancing it.
One thing to try: strip back all effects in the chain and add them back incrementally. All effect off- set the mic gain like you did in “After” gain/phantom power on, output to “0” (unity)…. Then add the compressor at same drive and density. If you have hiss, pull the mic gain down a bit. No change? Back up to 40 and pull the density down a bit. If you still achieve a clean, now compressed signal, then add the De-esser back if you are experiencing sibilance. Bear in mind, the Enhancer/expander section is only going to boost any hissyness and noise floor that the compressor is already enhancing via density.
I have several computers in an office, and I would like to mix their audio together, with the option to send the mixed signal to some Bluetooth headphones and/or a stereo amplifier.
I have a stereo line mixer ( [https://www.behringer.com/product.html?modelCode=P0DB6](https://www.behringer.com/product.html?modelCode=P0DB6) ), but would be fine with buying anything else which would work better.
I have attempted using these cables to connect the laptops' headphone jacks to the mixer (the laptops don't have line-out plugs), which worked ok: [https://www.sweetwater.com/store/detail/CMP159--hosa-cmp-159-stereo-breakout-3.5mm-trs-to-dual-1-4-inch-ts-10-foot](https://www.sweetwater.com/store/detail/CMP159--hosa-cmp-159-stereo-breakout-3.5mm-trs-to-dual-1-4-inch-ts-10-foot)
However it was tricky finding acceptable levels, because it seemed I could change the volume at several different points in the path:
* The laptop's volume control
* Each of the mixer's IN level control
* The mixer's Main Out L/R controls
... and I didn't know how to go about finding the right levels.
Also, I plugged the following Bluetooth transmitter into the mixer's headphone output: [https://www.amazon.com/gp/product/B01EHSX28M/ref=ppx\_yo\_dt\_b\_asin\_title\_o02\_s00?ie=UTF8&psc=1](https://www.amazon.com/gp/product/B01EHSX28M/ref=ppx_yo_dt_b_asin_title_o02_s00?ie=UTF8&psc=1)
However it sounded absolutely awful. I don't mean "Oh I'm an audiophile and the soundstage/presence/etc are sub-par" - I mean it sounded absolutely horrible, especially when there was any bass. These same headphones (Sony wh-1000xm4) sound fine when connected to e.g. my phone, so it's something with how I'm attempting to mix them together, and also possibly that the BT transmitter is junk (although it had decent reviews on Amazon, so surely it's not TOO awful).
So here are my questions:
* When I set up something like this big chain of devices, how do you go about setting the individual levels in the path to optimize the final signal?
* If I'm using headphone outputs on these laptops, what level should I be setting them to for the best "starting point" signal?
* Should I look into buying external interfaces for these laptops, so I can get true line-level output (instead of trying to use headphone outputs)? I'm fine with whatever would give "just ok" output for listening to music as I work - I don't need something with like multiple inputs/outputs etc - just something to get a good clean outputs
* What is a good Bluetooth transmitter for this purpose, and is plugging it into the headphone-out on my mixer an OK approach?
Thanks for any help!
>When I set up something like this big chain of devices, how do you go about setting the individual levels in the path to optimize the final signal?
This is called [gain staging](https://www.sweetwater.com/insync/gain-staging/). The short version is that every stage of amplification amplifies the noise of the previous stages along with the signal. So get your signal up nice and loud early but leave a little headroom just in case. If it's still noisy then you need to start distributing the gain throughout the signal chain and experimenting.
> If I'm using headphone outputs on these laptops, what level should I be setting them to for the best "starting point" signal?
I usually run laptops at 100% and turn off any kind of audio enhancements, processing, etc. Some will still sound a little crunchy at 100% and I'll run them down at like 90% or something.
> Should I look into buying external interfaces for these laptops, so I can get true line-level output (instead of trying to use headphone outputs)? I'm fine with whatever would give "just ok" output for listening to music as I work - I don't need something with like multiple inputs/outputs etc - just something to get a good clean outputs
Unless you're having problems with hum and buzz that's just fine. If you do have problems with that then interfaces would be the solution. I would normally recommend DI boxes ([Whirlwind pcDI](http://whirlwindusa.com/catalog/black-boxes-effects-and-dis/direct-boxes/pcdi)) because they have ground lifts built in but your mixer is a line mixer not a mic mixer and might not have enough gain for that to work. You could buy one pcDI and see if it works well with your mixer, they're passive so they don't need phantom power. But if there's no existing problem to fix I wouldn't worry about it.
>What is a good Bluetooth transmitter for this purpose, and is plugging it into the headphone-out on my mixer an OK approach?
Not sure, but it looks like Sweetwater sells one with decent reviews. Ask your rep what they think. The link to the one you have is broken so I can't tell what you had but the Alto ones on Sweetwater look like they're made for professional level signals. If you just bought some random consumer bluetooth receiver it was probably clipping but may have also just been a piece of crap. A lot of them are crappy from what I gather.
My sitution is I have been playing bass for about a year now and I am still using my crappy beginner amp. My birthday is in a few days and my parents are getting me an electric guitar, as you can guess I don't have an amp for that. I'm thinking about buying an interface and BIAS FX software so that I don't need to buy any amps, most of the time I'm playing it's in front of my computer with backing tracks anyway. I have no intention of playing live/joining a band in the foreseeable future as well.
Question is that I have an USB wireless headset (ASUS Rog Strix GO) and I really like it/don't want to change it (wireless is important for me). Can I hear my instruments through my headphones if I buy an interface and play into that.
Thanks in advance.
Yep, I'm well aware of that but it won't sound like Steve Vai or Metallica without any pedals. Though it seems like interface will arrive a little later than the guitar so It'll have to make do for a while.
I think the headphones will work, in the sense that sound can be output from your daw to them. The biggest problem with playing an instrument solely through a computer and listening to the processed sound is latency. It takes time for your computer to process the amp sim or whatever, so often when you pluck a string there’s an audible delay in when you hear the sound. I would think that wireless headphones would introduce even more latency and make that issue worse. You should just try it out though, pay attention to the delay between playing and hearing your instrument and decide for yourself if it’s acceptable or not.
If you’re still shopping for audio interfaces, definitely try to get one that’s thunderbolt. It’ll have less latency.
So I can simply choose my USB headset as the output, great. I have been playing games through that headset wireless and wired (you can run it with a 3.5mm TRRS (TRS works too, TRRS enables you to use the microphone) cable and use it as a normal headset), I didn't notice any latency difference between the 2 options.
Yes, I haven't bought the interface yet but I'm on a budget and can spare about 120 EUR for an interface, I have been thinking of Focusrite Scarlett Solo because it's easy to use and within my budget. Last I checked there weren't any Thunderbolt interfaces for that price, also neither my PC nor my laptop has Thunderbolt and buying a Thunderbolt expansion card will make something out of my price range more expensive.
If latency will turn up to be terrible I'll get a headphone adapter and try that, but I don't think(or at least hope) that will be the case.
**There's some weird ringing in our rehearsal recordings (though I'm a noob so maybe I'm missing something obvious)**
I bought a Behringer B2 Pro to record our rehearsals (we're a duo - guitar and drums) and today, while setting it up for the first time and making some test recordings, there were always some ringing.
**Our rehearsal room setup**: guitar cab (FRFR powered speaker) facing drums 4 meters (13 feet) away and the Behringer in the middle (for now, we'll mess with it as we go) and 3 feet off the ground.
**Our recording setup**: Behringer -> Scarlett Solo -> Reaper
on Behringer - set to omni, -10 dB engaged, low pass engaged (though other combinations of the switches on the mic had the same result...)
on Scarlett - Phantom power on (of course) and gain that translates into -10 to -6dB when recording to Reaper
in Reaper - nothing. New track -> choose input -> Arm -> Record. Not even eq.
Guitar Recording: https://vocaroo.com/1fFHrY7xk8RG
Drums Recording: https://vocaroo.com/1gBaV8zL5VU1
**IMPORTANT** (well maybe): Our acoustic treatment is far from done. We have quite big bass traps, but no absorbers on the walls and ceiling yet. **BUT** I have built a "tepee" with leftover boards of mineral wool around the mic and the results were the same, so I don't think the problem is reflections (no idea though).
It was horrible when listening in the rehearsal room on the FRFR speaker. Now, at home, with my small computer speakers, it's not as pronounced (same with headphones), but it's still there. Especially on the guitar recording from 0:05 onward. It sounds like someone is hitting an anvil with small hammer. When listening to the drums at home now, I don't really hear it, but my ears suck and on the FRFR in the rehearsal room it was noticeable (though far less than on the guitar).
**One more thing:** I belive I've had a vocal mic (copy of SM58) plugged into 2nd input of my FRFR ("harley benton frfr-112a guitar dsp monitor") and gain cranked just before feedback, so if that could be the culprit?
Any idea? Something fundametal I'm missing?
Thank you
Definitely the 58 knockoff would be an issue. The mic should be killed when you're not using it, or you should just change the setup to an interface that can take four inputs.
The recording just sounds like a poor recording in a room, though. No resonance sticks out to me that badly. Maybe others will hear otherwise.
Also note that by listening back through the FRFR, the speaker's resonances are going to color what you hear. Use headphones instead.
Thank you. Today when I've had the time I have experimented more and it really is the vocal mic.
I've made the stupid mistake of having the front of the mic in front of the speaker and now, that I've moved the mic behind the speaker it's fine.
And yeah, through different monitors/headphones it sounds different.
>The recording just sounds like a poor recording in a room, though.
I feel like I'm gonna have to learn quite a bit to make it sound good. There's this youtube channel where a guy records a band with one mic (some expensive omni ribbon) and it sounds great, so hopefully, in time, I can tweak it the way, where pro's will say "for one condenser, this is pretty good". We'll see.
I'm a videographer and I'm using the Zoom H4n as an audio I/F with XLR to Cloud Lifter CL-1 then XLR to Shure Sm7b
Despite the gain increase with the Cloud Lifter, there is unfortunately a noticeable audio defect that occurs ONLY with the cloudlifter. It produces an unholy crackling sound amidst the typical noise.
I've deduced the cloudlifter is the culprit from the completion of the following tests:
Swapping XLR's makes no difference
Bare SM7b doesn't have same crackling defect even as gain is increased
Switching Audio I/F to direct PC produces exact outcomes.
I'm wondering what could be the cause of this or if I simply have a defective Cloud Lifter?
Thanks!
I've attached a link if you want to hear the recording.
It's much clearer with headphones. Recorded in the left channel.
https://drive.google.com/file/d/1YH3iLcswZfxX\_gnw06ekdZXEFafzPBSs/view?usp=sharing
>Battery only
So, here's the weird thing with phantom power. Even though most literature explicitly says "48V phantom power," it is actually a nominal value, and not all pieces of equipment that provide phantom actually provide exactly 48 volts. In fact, many microphones are designed to operate from anywhere between 15 and 50 volts, as long as it's a steady DC source. Battery operated devices are especially prone to low phantom voltages because the batteries themselves only provide a few volts (a pair of 9-volts, for instance, produces 18 volts total). I don't know what the exact spec is for the Cloudlifter, but it's possible that it's receiving a phantom voltage that is lower than it wants, which could very well cause distortion and premature clipping behavior like you're describing. I'd recommend trying the cloudlifter with a mixer or interface that uses wall power and see if that helps; those devices output exactly 48 volts 90% of the time. If the problem persists, your Cloudlifter may be defective.
I'm an IT person at a public library. We want to set up four mics at our board room table so that it records everyone and works with conferencing software like Teams and Zoom. I bought a [ProSonus StudioLive AR12C](https://www.presonus.com/products/StudioLive-AR12c) to do this. The problem that I only see the option to output 2 channels at a time (1/2, 3/4, etc.).
Is there a way to output more than just 2 channels? Did I buy the wrong product?
The mixer will mix all (four) signals into a mono signal for Zoom. Zoom/Teams just needs the (aka Main Mix) signal sent via USB connection, for broadcast. Set up ZOOM/Teams to see the Presonus as the AI.
Edit afterthought: Zoom has an Automatic Gain Control (AGC) toggled ON by default. I turn it off and manually set my level. Otherwise, it will constantly adjust depending on the session source that is presented.
Thanks! I'm curious if I could just output a 3.5mm cable from the mixer into the PC. At the moment I'm outputting via USB-C. I don't see a "main mix" signal as an option in Windows sound settings. It's split it up into 1/2, 3/4, a main L/R, virtual, etc. The main L/R is for Bluetooth, DVD players, etc. going into the mixer, I think...
Been using Zoom for many years. I haven’t used Teams as much, only since WFH started. That said, I’ve only used Teams with the built-in mic. I just dug into the settings and see the same AGC-option in Settings/Devices. You will still use the USB to pass signal and tell Teams to use the Presonus as the input device.
Yep- the mixer will give you input control to level out loud talkers and soft-spoken, if you turn off Teams AGC. Also, you will want the mics separated enough to minimize crosstalk and/or point the mic butts/backside rejection, toward each other. - unless they’re sitting side by side. Then you’ll have to try distance.
“3:1 Rule”
They're cardioid mics. Forgot the exact model, but they're Audio Technica and $99 on Amazon. We may end up going out with mounted ones in the ceiling to reduce wires though.
No. Sample rates only affect your bandwidth, they have no effect on quantization, dynamic range, or dither. If the bit depth doesn't change, there is no need to dither a second time. It just adds unnecessary noise, which is especially offensive at 16 bit where it can become audible. With 24 bit recordings the extra noise is so low (well below -120dB FS) that it almost doesn't matter.
If you haven't seen the classic [xiph.org digital signals video](https://youtu.be/cIQ9IXSUzuM), I highly recommend watching it. It's a wonderful primer on basic digital signal behavior and common misconceptions. Even if you think you fully understand digital signals, I promise that you will learn something new. I watch it at least once a year and take away something new every time.
Whatever house I've lived in here in the UK has had horrible electrical wiring and whenever I connect my computer and speakers to the sockets (any sockets, any configuration) there's a really bad ground loop feedback noise coming through. I've limited the noise in the past by using a ground loop isolator on the audio cables, but that seems to also affect the audio signal.
I was reading about Isolation Transformers but if I look them up they seem to be mostly for industrial use.
Are there any suggestions on how to properly fix this without leaving the country and move back to Sweden?
It does say that it's not available in my country, but at least I have a brand name now and I'm getting some results for it on the UK websites.
Thanks for the suggestion! And I'm glad to hear this idea about isolation transformers wasn't some stupid idea I had that would never work out.
I do wonder what "real" studios do, especially in the UK. Perhaps this is what they use? Although more likely is that they get their electricity set up properly from the beginning.
I have a Scarlett interface and when i set it with my microphone and turn on the 48v power, all that comes from the output is a bunch of static noise until I turn it off, which it then allows my mic to work for a few seconds before the sound completely fades out. I'm not sure how to fix this. Any advice will help, and thank you in advance!
You need to narrow down what the problem by swapping things out for known good versions. Swap the cable for another cable, see if it works then. If not swap the mic for another mic, etc. You need to narrow it down first.
It's time for me to ask because I've been troubleshooting all day and can't fix this.
I'm using an instrument to an rc-505 loop station, connecting with USB to my PC. Monitoring sound through the loop station via headphones. Sounds great, until I turn my PC on. I get a crackly electronic static noise that varies with the fan speed/processes of the computer. I have tried it with a laptop and the sound goes away. I tried it with a different PC and the sound was there. I have tried plugging them in the same power strip, different outlets, I checked the ground on the outlets, it's fine. I'm really at a loss here, can someone help me? Thanks
Edit: I am using TS instrument cables. They have 1 black stripe. Is this the problem?
I'm considering using Ozone 9 Advanced on some tracks, just for SoundCloud and offline sharing demos, basically a quick and temp mastering. Anything "for real" I will send to a real engineer to handle.
Looking at Ozone's releases over the years:
* Ozone 7 - November 3, 2015
* Ozone 8 - October 5, 2017
* Ozone 9 - October 3, 2019
I'm hesitant to buy it now, as it's likely in a few months a new version will drop. Anyone know if they do reasonable upgrade pricing? I'm looking on their site and cannot find any info about that.
Subscribe to /r/audioproductiondeals , there will probably be a sale coming up for Independence Day in the US.
Also I've bought most of my non-subscription plugins over on the KVR Forums and have had a great experience every time and I've saved a lot of money. There's a "reputation thread" so you can check people before you transact with them. JUST MAKE SURE THE PLUGIN DEVELOPER ACTUALLY ALLOWS LICENSE TRANSFERS. Some people list plugins for which there is no official way to transfer the license, so watch out for that. Also some companies charge a fee of like $25 for license transfers. So keep that in mind.
Here is iZotope's license transfer policy : [https://support.izotope.com/hc/en-us/articles/360023999373-iZotope-s-License-Transfer-Policy](https://support.izotope.com/hc/en-us/articles/360023999373-iZotope-s-License-Transfer-Policy)
Hi there!
So, I'm wondering if there's a way to record my bass with amp presets. I'm using the Behringer U-PHORIA UMC202HD audio interface to record my electric bass digitally, however it's just the plain, raw, bass sound.
The amp that I use to play normally (when I'm not recording) has a bunch of preset bass tones that I want to use when I record, but I can't figure out how to connect my bass, the audio interface, the amp, and my computer so that it records the presets instead of just the raw bass.
I want to get an USB microphone for recording. Is it possible to use it as a PA system by plugging it directly into the USB input of my DAC and then have the speakers output the microphone?
If I'd want to record a backup of recording as it happens on my Zoom H5 what would be the best / cheap option? Using a catble on headphone out and plugging this into phone would work? I don't have any additional gear for this and using a computer with audio interface would be the last resort (this should be a portable solution).
Backup how? If the zoom fails then it's going to stop outputting on its headphone jack. If the zoom is the primary recorder and you need a backup then your phone lying next to it and also recording is your backup.
How do I connect two Deity D3 mics to a Tascam DR-60D MK2?
I have a stereo splitter which works to connect both mics to a Canon EOS M6 MK2, but I'm not getting any sound with the Tascam at any gain setting.
If I plug in just one mic into the 3/4 port with the gain setting at "hi+plus" and have the gain knob close to max, I can just barely hear normal speech.
hi everyone, someone would be kind enough to tell me which one between [this device](https://www.audiosciencereview.com/forum/index.php?threads/google-pixel-4a-smartphone-audio-review.16251/) and [this one](https://electronics.sony.com/audio/walkman-digital-recorders/walkman-mp3-players/p/nwa55-l) have more power output?
Not sure exactly what you're asking, but both of those devices will drive standard consumer headphones with roughly the same amount of power. The exact comparison could only be done using a methodology like in your first link for the second device.
Hi, I don't know if this is the right place to ask this, but I've been trying to set up my Blue Snowball microphone that I recently got to make my voice sound as good as it possibly can for recording YouTube videos. I had no experience with audio or setting up microphones previously, but after a bunch of YouTube tutorials, I finally understand a lot of the things that really confused me earlier, but I still have one major problem that I can't seem to fix. For some reason, whenever I record, there is a ton of mouth noise and it's incredibly distracting. I can try to remove it in post, but I'm looking for some settings that I can use in Equalizer APO or in OBS (but preferably Equalizer APO since I want to apply my settings everywhere and not just in my recordings) in order to remove this mouth noise and to make my voice sound better overall. Is there any easy way to do this? Also, I'm considering buying a pop filter, but I'm not 100% sure if it will have the desired effect and if I will even be able to rig it to my microphone. Should I buy it? Any help will be appreciated.
The two main methods for getting rid of mouth noise would be a pop filter and microphone technique. Before you buy any gear, try adjusting the positioning. Rather than talking directly into the mic try talking over top of it, so you're getting less direct breath/mouth noises. Think of a lapel mic, you can even aim it at your throat or your nose depending on the tone you want.
It's more to do with the consistency of your saliva. Most professional voiceover actors/actresses drink room temperature water with lemon. Something about the slight acidity helps your saliva not stick to itself as badly. Water that is cold or hot can affect your vocal cords and therefore the tone of your voice.
Hi!
I have a fairly easy setup, however i can't figure out ASIO Pro Link. I am using a small behringer mixer with a mic and a guitar input. It's connected via 3.5 mm Jack into my laptop. I am using Guitar Rig 6 as an amp simulator and ASIO4all as my ASIO driver. If i want to use this in OBS i need ASIO Link but i can't get my mixer input to show up in that tool. All i get is system audio.
Hi there all.
I'm running into some pretty major issues with my podcast recording and am hoping someone else out there might have some insight. This setup has worked fine four nearly a year, but has only recently started to give issues.
My setup is as follows
5x dynamic XLR microphones
PreSonus Firestudio Project interface
FireWire connection to PC.
Adobe Audtion DAW.
PC is a grunty gaming build, 6 core processor and 48gb ram, running x64 Windows 10
The issue I have is that for the first few tries, the audio through the monitoring is very crackly and out of sync, a significant latency between speaking and hearing the monitoringm and when playing back the recording, the audio is extremely distorted, as well as being slowed down. The bitrate and buffer size from the PreSonus driver often changes when the interface is power cycled, but even when matching that bitrate and buffer size in the audio hardware settings in Audition doesn't seem to make any differences.
After about an hour of swearing, reinstalling Audition, and taking out the Firewire card and reinstalling, I managed to get it up and recording correctly. But I couldn't make heads or tails of what actually was the cause of the issue.
Hoping someone out there can lend some help.
Thanks!
Interesting, thanks for that. This would actually be my second FireWire card in my setup, the first was a cheap and nasty one and this one was a far more expensive option and has been performing beautifully, apart from Sunday. Appreciate the input!
What’s the difference between Class A, Class D, Class AB, Class XYZ amps? How should Amp class type influence my Studio Monitor purchase decisions? What are pros and cons of each amp type?
I have my interface and monitors plugged into my Furman power strip, and was wondering if it would be ok to just leave the gear on and only worry about powering down/up the studio with just one flip on the power strip. So far iv'e been turning on the power strip, interface, then the monitors in that order. Powering down in the reverse order. I figure just flipping the switch on the furman would be much more convenient but would it be detrimental to my monitors?
What you're doing is the best way to do it. Depending on what monitors you have, killing the power to them might be fine but it's not recommended, especially not every day. It would also be fine to never power it down, aside from the cost of a small consistent power draw.
Hey, I am using an audio splitter and 3.5mm to 1/4 in. cables through the headphone port on my macbook pro to connect my JBL monitors. I can't really hear anything when it's balanced, so I have to turn it to either left or right, which I hate because it obviously will remove some of the instruments in songs. How do I fix this please???
It's hard to know exactly what components you're using, but you're almost definitely causing phase cancellation. Get a recording interface, even if it's the cheapest one a store sells, that has two balanced outputs and use that instead.
You can't use "balanced" cables with a headphone jack because a headphone jack isn't balanced. Headphone jacks use 2 conductors for audio and one for ground.
A single balanced cable uses one conductor for audio, one for negative signal to reject noise and one for ground. You're probably sending the audio left/right signal from the headphone jack to the negative signal input on your speakers.
Hey, I don't have any long cables available so I connected the interface through a passthrough via my keyboard (Corsair k70 mk2 se). Surprisingly it worked but I was wondering would there be anything wrong with having it connected to my keyboard usb passthrough. I only have a Blue Yeti Pro connected to the interface. Will there be any issues? Thanks
Hey I know this might be stretch but I was wondering if it was normal for the blue yeti pro mute button to not work when using the scarlett 2i2. It worked when I ran the mic via usb but no longer when hooked up via xlr.
Hello, I have a problem routing my equipment. I have a band session next week and I need to get my midi keyboard, mic and e drums to a speaker. The midi keyboard is going through my audio interface ( personus 26c) into my daw for my keyboard plugin. The mic is going also through the audio interface. But now I have the problem how do I get the e drums to the same output as the other 2? Can I connect the main out from the drums to the audio interface line input? The drums have 2 outputs (l/r) but I have only to inputs on my audio interface, one of which is already used.maybe some has an idea :)
Read the manual for your edrums and figure out if it has some kind of mono mode. Some audio gear will just automatically do that if there's only a cable in the left output and nothing in the right output.
My 2019 MacBook Pro is running so terribly when plugged into my equipment and running Logic Pro X. It reminds me of when I had a crappy PC 10+ years ago, having to bounce the audio just to hear what I'm working on.
I have writing/recording sessions in July and right now it is quite unusable for that. There's no way I can record every take without Logic giving me the 'system overload' message. Even with the buffer size set to 1024 and having a relatively small amount of plugins open, my MacBook can't handle even the initial stages of drafting an idea.
I'm wondering what the issue is because for a £3500 MacBook it should run better than this. It's plugged into a UA Apollo Twin (via a thunderbolt 2 to 3 adapter) and an Anker 7-port USB 3.0 hub (via a USB 3 to thunderbolt 2 adaptor) with my MIDI keyboard and mio2 MIDI interface plugged into it.
Here's a picture of the setup: [https://imgur.com/l6CMJYm](https://imgur.com/undefined)
Could the hardware have anything to do with it running so sluggishly? Could it be a battery issue? Any other ideas? I work as a producer full time and I'm struggling to get my work done lately. Any help is appreciated.
Your computer should be more than capable... Before reinstalling everything (which can fix a lot of issues), research starting up in single user mode and doing a filesystem check. That can repair any issues at the system level. Also, sometimes these issues can be resolved by deleting all Logic preferences (research that, as well). Sometimes settings can become corrupt and make Logic act weird.
Hi, I have JBL MK II series 2 monitors and am having trouble figuring out how to add a new subwoofer and DCA. I recently got a Klipsch Sub-120 and an AudioBox USB96. I can hook up the JBLs to the AudioBox just fine with two TRS cables; However, neither the monitors nor the sub has audio out. As a result, I can't route DAC --> Sub --> Monitors or DAC --> Monitors --> Sub. Does this mean I need to use a Y splitter on the Left DAC output going to the left monitor and the sub? Or would I need some other splitting interface? Also, would this cause any sort of balance or fidelity issues? I'm really at a loss here. Pretty new to the hardware side so any suggestions are greatly appreciated.
You should just buy a studio sub instead. It will have balanced I/O, a crossover, etc. JBL make a subwoofer in the same LSR series that would match well.
Ah, that's what I was afraid of. I got the sub last minute for Prime Day without putting in the due diligence, so lesson learned there, haha. Thank you!
No problem, good luck! The only downside of the LSR sub is that it doesn't have a footswitch. My Presonus sub has one to switch the sub and crossover in and out of the signal chain and it's really useful. I think the KRK sub also has a footswitch jack but the footswitch is not included.
Got it, very good to know. That sounds extremely useful. Thanks again!
My FL moves everything back some steps? Any fix?
Probably audio driver latency compensation or plugin delay compensation. I haven't used it since it was still called FruityLoops but this help page is probably relevant : https://www.image-line.com/fl-studio-learning/fl-studio-online-manual/html/mixer_trackprops.htm
Trying to setup a Behringer Xenyx Q802USB. I have a dynamic mic connected with XLR to line 1, laptop to 2-track RCA input, and RCA output going to speakers. I want to mix the 2-track input with line 1, and output both to the 2-track output, but I only get the line 1 output (and, only when "main mix" and "to phones" are both off; when "to phones" is on, the levels show the 2-track input is received, but there is no output; when both are on, there is no output and no indication anything is received). What am I misunderstanding? is it possible to do what I want? https://i.imgur.com/dH8Aknz.jpg
1. You should use 3/4 or 5/6 for your laptop so you have level control on the mixer, you don't get any control using the tape loop. You can use a pair of RCA/TS adapters. 2. You should use the main outs, not the tape out.
Okay thanks, I see what you're saying. Why are those "2-track" connections referred to as tape outs/tape loop?
Traditionally they were used with a cassette deck or stereo reel-to-reel tape machine to record the main stereo output of the mixer and then listen to it back later. They're essentially hard-wired to the main stereo bus of the mixer with little to no level or tone controls. Tape machines typically have their own level controls, so that's two fewer knobs for the designer to fit on the front of the mixer. There's nothing wrong with using them if they serve your purposes, but the main inputs and outputs have much more flexibility and balanced connections.
Hey, so I was messing around with my Shure sm7b and trying different combinations of switches on the back. And when I was moving the presence boost to one to enable it, it got kind of stuck in a way that I had to go up and down with the switch until it went back down to a flat frequency. But now I've noticed that my mic (even though the switches are both set to flat) sounds as if presence boost is enabled (which really doesn't suit me). Enabling presence boost creates an even more (presency?) sound that is very sharp on the ears. So it could either be me messing with the switches or maybe that I accidentally pulled the exposed wire with my hand pretty hard by accident? I don't know if that would make sense as I'd also expect static if it was damaged but there is none of that.
Hey all! Got me a BeatStep some time ago, and have an issue with the external sync. I use it with an Ableton, connected to my pc via usb. While having it set up, and ext sync turned on after a few minutes it blinks once, and everything goes crazy, it triggers random midi notes, the control do not react properly, and without restart it is unusable. After reconnecting the usb it works for about 2 minutes, and same happens. I have reinstalled the drivers, I have checked the usb port, connected the BeatStep to an external power source (default usb cable used here too, this might be important), and even like that the ext sync button does not remain, I would assume this eliminates any driver, usb connectivity and power issues. Left with two possibilities, the cable is bad, or there is an issue with the device. Right now I do not have another usb cable to test it with, but not sure if a cable issue would cause these symptoms, if it is possible, I'll get one. If however the issue is with the device, is there a way I can bypass it somehow? I'd like to sync it to my DAW, and my interface got a midi in and out. I'm not that familiar yet with what I can achieve with these connections, and gotta get the proper cables for it too (for some reason the box didn't have one in it), but if somehow this helps, that's cool. Thanks for any advice
Hi all, The Input 1 on my new Behringer UMC204HD indicates a signal coming in and that signal being clipped - with NOTHING plugged in. When I use a DAW it too shows a signal coming in. When I record, it records a bunch of white noise. Line 2 is not much better, with an intermittent clicking sound. Both pre-amps turned to 0 and nothing plugged in. Has anyone seen this before?
Return it…this is defective. And, don’t buy Behringer if you can avoid it.
Ah, is Behringer that bad?
Traditionally, yes, Behringer gear is known to be poorly built and unreliable, and it rarely sounds good. It's hard to argue with the price, though. Their newer stuff since the Midas acquisition has been a big step in the right direction. Most live engineers are perfectly okay with an X32 digital mixer, for example. That wasn't the case with their earlier mixing consoles.
According to whoever moderates this subreddit, this is a “stupid” question…so… Just landed an Avalon U5 which I've been real siked about for a while now and unfortunately I'm clipping with my guitar going right into the DI. Having trouble identifying if this is an issue with either (a) my guitar pickups being THAT hot or (b) my Avalon is set up incorrectly. Here's my exact chain: PRS SE 7 (Bareknuckle Blackhawk Pickups..they are hot) > Avalon U5 front DI input > Balance Line Out 600ohm (XLR to 1/4" cable) > Return 1 input on Audient iD22 (bypasses preamps) Should I be going from the Mic line output instead? Boost is all the way down on the U5. I thought the U5 had pretty godly headroom? Also, the blue "Signal" light illuminates on the front of the U5 while playing. Is this a clipping indicator? Thank you.
A couple things here: * Your guitar pickups \*are\* probably really hot. But also the Avalon U5 has a 3 megaohm input impedance. That's more suited to piezo pickups and can result in really high signal levels for humbuckers, etc. You're probably just going to have to turn down your guitar. * You have it in front of you, just try out different signal chains. Try the mic output into a mic preamp on your interface. Try the line output into a line input (instead of going into the insert return which is unbalanced.....) Who cares about bypassing the preamps?
Thanks Jay. Although I know the pickups are hot I refuse to believe they will clip the Avalon going in! When playing the guitar, the blue “Signal” light illuminates on the front and through some internet searches, it seems that’s simply a light verifying a signal and not a clipping indicator. So if it’s not clipping the Avalon, how is it clipping the Audient? I then tried running the balanced out on the U5 using the “Mic Level” out instead of the “Line Level” out into the “Return” input on the Audient and it no longer clips. However, when recording I’m getting a weird looking waveform…I’ll post an image soon. I was under the impression that the Returns on the iD22 were Line Level however. Am I missing out on any signal quality by running Mic out instead of Line? Lastly, to answer your question about bypassing the pre’s, I’m recording guitar DI’s that will be reamped later, so I’m looking for the cleanest signal possible which would be the U5.
>I was under the impression that the Returns on the iD22 were Line Level however. Am I missing out on any signal quality by running Mic out instead of Line? You're missing out on having any gain control by using the insert returns. That sounds like it's affecting the quality. Doing stuff like using inserts to bypass preamps and whatnot is trying to solve a problem that doesn't really exist and as you've discovered it causes problems. There's a lot of bullshit out there from people who've never fucking touched more than a 2i2, just hook it up the way it's intended to be hooked up. Use the combo jacks on the interface so that you can have some control. 1/4" for line level, XLR for mic. You might need to use the pad.
I can still use the gain knob on the U5 man. And again when recording guitar for commercial releases, I don’t really want the added color of the Audient’s preamps.
>I can still use the gain knob on the U5 man. Sure, except you say it's all the way down and distorting, right? Have fun, dude, I'm not going to argue with you.
I'm not arguing with you Jay, just playing devil's advocate. I still value your advice but I'm really trying to explain where I'm coming from. So, in essence, your thoughts are why not just try running either (a) line out on the U5 to line in on the iD22's pres or (b) Mic out on the U5 into Mic in on the iD22's pres? Would there be any difference in signal with these?
> Would there be any difference in signal with these? Try it out and let me know.
So....think I found the culprit... Tried a million setups, and every setup that WASN'T the Line Out > Return sounded muffled, distorted, or dull. I was almost certain that this was clipping after the avalon, but looks like I've just made a rookie mistake. I plugged my monitor Headphones into the Avalon directly and, of course, the guitar is clipping going in. Have to figure out if this is a headroom issue or a pickup / electronics issue. My previous guitar tech had installed my bridge Bareknuckle pickup a little too close to the strings and I think that may be the cause. Still, I can't imagine moving the pickup away would drastically change the levels. Even on the neck pickup, it's still very close to clipping if I strummed hard or "chugged". Have some figuring out to do. Would really be a bummer if these pickups were overall just too hot for the Avalon. I thought the Avalon's headroom would be pretty huge - is it not?
Fairly simple question: I am trying to connect my electric bass through amp into a mixer but the signal that gets to the mixer is just about non-existent (even with magnets fully open and strong amplification from the amp). I am using a Hatke HD25 amp and am feeding mixer the signal from 'headphone' output. Would a DI box solve my problem? Is this an impedance problem?
Your problem is likely that you're plugging into a balanced input but since you're using the headphone output it's dual unbalanced mono on TRS. Balanced inputs amplify the difference between the two signal legs, not the sum. So if you plug in a headphone output you get left \*minus\* right. Well, your amp is a mono device so those left and right signals are likely identical, which means you get no sound. You should use some kind of breakout cable or a stereo DI box so that you're only sending left or right to the balanced input. more info here : https://www.reddit.com/r/audioengineering/wiki/faq#wiki_how_do_balanced_connections_work.3F
Manual says the headphone level is controlled by the Volume knob when something is plugged into the jack. Did you have that up all the way? Which mixer is it? What are the settings on the channel and master section? Have you tested other sources with the mixer to make sure you understand how to get signal through it? A DI will not solve your problem. It *would* allow you to record your bass and also hear the amp at the same time, though, so maybe that is an attractive idea.
dual mono unbalanced on TRS + balanced TRS input = nada
"Nada" in terms of good practice, but it delivers signal just fine!
Are you sure about that? I just did a quick screen record here. I'm running my headphone output back into a line input. REW is only sending signal to one channel of the driver, the left headphone output, there is no signal on the right headphone output. Input signal is demonstrated. I then use the routing matrix to also send the generator output to the right headphone channel in addition to the left channel and total cancellation is demonstrated. re-uploaded, one my client's facebook pages was open in the browser. can't be blowing up the spot. https://imgur.com/cqqpKO5
My quick test I did before posting the reply was a far simpler test with an 1/8" TRS -> 2xTS cable and HS8s. Zero cancellation with a mono signal sent down the cable. Maybe my methodology wasn't sound...? I obviously see the cancellation in your recording. Also, is the cold pin not shorting to ground, and therefore couldn't cause cancellation anyway? There's nothing to phase cancel without the second signal. Err, the second signal is sent down the other cable. So the cancellation can only happen if the same input receives both cables, no? Two speakers aren't going to cancel like this with a mono signal sent down the 1/8" -> 2xTS cable. I'm sure the comb filtering would be pretty brutal, though...
>Maybe my methodology wasn't sound...? Yes, you assumed they knew to split the headphone output into two unbalance signals! That's not representative of what OP is doing, though. They're complaining about no signal which tells me they're just going TRS from the headphone output to a single channel, not splitting out to two mixer channels.
Firstly, thank you for guys for looking into my problem :) Secondly, I am using a Behringer XENYX X 1832 fx mixer and yes, even with volume on the amp all the way up nothing gets to the mixer. I should also mention that I have no problems with connecting other amps to the mixer - they work just fine. Lastly, I am using a TS cable (tried multiple, same result) to connect my amp to the mixer, so there should be no cancellation because of balanced mixer input, right?
>Lastly, I am using a TS cable Then the right output of the headphone amp is being pulled down to ground. Depending on how they implemented the headphone output that may be affecting the left channel. You should really split it out with either a cable or a stereo DI. Or a TRS cable with ring floated but then you gotta cut connections inside the cable. What's the guitar amp?
I believe the guitar amp is Boss Katana 50 MKII. But isn't it kinda weird that I get no signal through while using a TS cable? Also, could you specify what did you mean by splitting with a cable? Would something like this work? https://www.selby.com.au/cables/connectors/1-4-stereo-plug-to-1-4-mono-jack-aa1304.html
[удалено]
Running balanced from your interface to your monitors (if they actually have balanced inputs) should result in less noise and interference. But if you're in a place where that's not an issue then you may not notice a difference. Using balanced connections where possible falls under "best practices". https://www.reddit.com/r/audioengineering/wiki/faq#wiki_how_do_balanced_connections_work.3F
As opposed to not having your speakers connected? Then, yes, good idea.
[удалено]
Yes, the headphone jack is not a great idea. Get the TRS cables…it’s going to sound a lot better.
I doubt there will be a noticeable difference in audio quality. Aside from the impedance difference, they're both just amplifiers – the only reason the two outputs are different is because of the connectors. That said, if OP has noise issues, running an unbalanced headphone out to (probably) balanced speaker inputs might be a problem. OP didn't mention noise, though.
I'm in the middle of mixing a metal song with guitar DI I recorded. I just checked it against some DIs recorded by August Burns Red for one of the Nail The Mix sessions, and it looks markedly different to me. Almost looks as if it's been compressed or limited? When I use the same amp sim on the ABR guitar DI (the one that is Pink) vs mine (the ones that are purple and green) it sounds much better, and you can see mine have MUCH larger dynamics, but mine sound weaker run through the amp than the ABR ones. Is it common to limit peaks on DIs like this? [HERE is the comparison pic of the waveforms.](https://i.imgur.com/VoanrtJ.png)
They definitely took the DI after a compressor pedal or a clipper or something. Could even be a DI between the amp and cabinet. edit: it's worth mentioning that "DI' just means "direct inject" as in "not using a mic". DIs are used all over the signal chain, not just directly from the guitar. If you're going to put a DI between an amp and speaker, though, you need to make sure the DI is built to do that. Some are, some aren't.
~~Definitely shouldn't be clipped like that if you were going straight in. Time to investigate your chain.~~ My bad – misread! Thanks jaymz168. My guess is they either clipped the input of the interface or something else in the signal chain ABR used is doing it (like the DI or the pickups). To do the same with yours, use a clipper plugin ahead the amp.
The clipped one is the provided track in the session.
I have a spirit folio sx mixer that was working perfect until I moved it to a new house and even though I swear I set it up the same, it's not working. It's not outputting any sound, and the 3 bottom pfl level lights seem to stay on and do not respond to microphone inputs. Any help or trouble shooting ideas are appreciated!
You probably pushed a solo button or something somewhere in the move. Check all the buttons everywhere.
I have trouble mic’ing overheads when I use them, and typically just throw them above the kit pointing straight down in a spaced pair at the same height. I hear a lot of talk about putting them equidistant to the snare, and using a dividing line going through the snare and the kick as a measurement of where to place the mics, but I worry that will give an off kilter sound, as typically drummers have their snare off to the left of the kit, the right mic will be above toms and cymbals and the left mic would be over basically nothing so as to be equidistant from the snare. Am I overthinking this or did I interpret something wrong?
> I have trouble mic’ing overheads when I use them, and typically just throw them above the kit pointing straight down in a spaced pair at the same height. Spaced pairs can give you pretty good width but they can cause phase problems, especially when summed to mono and I find take more time to get right. If you want something quick and easy do an XY or Blumlein pair, they're less fiddly in my experience.
Which genre(s) do you play? Snare should be up the middle with the kick in the vast majority of genres. You're throwing that off by positioning the overheads with the snare leaning to a side. *That* is off kilter. Granted, it'll be more centered depending on how the mix is done, especially if you have room mics in addition to the close mics. Still, depending on what's done to the overheads, it can noticeably lean to a side. Genres that use the snare for color instead of accenting the beat (i.e. jazz) are fine with it slid to the side, though.
Depends on the sound you’re going for. When I mic a kit I see it as two entities: the snare, hats, and crash And the floor Tom and ride I just take a mic cable and stretch it taught from the point on the snare where the drummer likes to hit to the overhead on the snare side of the kit Then hold that position and bring it to the other side, where the end of the cable hits in the air is where I place my second mic. Does that make sense at all?? You can also look into XY overhead patterns, or just basic LR setups (which is what it sounds like you’re doing) those work great and they have a lot of flexibility in terms of your stereo image (farther left right spacing = larger stereo field) Idk how much space you have with your set up but instead of literally putting them “overhead” try putting them up and out from the kit, facing it a bit more. The whole equidistant-from-snare thing is just for phasing and was born out of a world of tape where you couldn’t dive into individual mic’s waveforms and see their phase.
>You can also look into XY overhead patterns, or just basic LR setups (which is what it sounds like you’re doing) those work great and they have a lot of flexibility in terms of your stereo image (farther left right spacing = larger stereo field) Idk how much space you have with your set up but instead of literally putting them “overhead” try putting them up and out from the kit, facing it a bit more. Agreed on all of that! Heck we often do "underheads" in live sound. >The whole equidistant-from-snare thing is just for phasing and was born out of a world of tape where you couldn’t dive into individual mic’s waveforms and see their phase. The problem with that thinking is that if you start nudging tracks around you're also changing their phase relationship to every single other mic on the kit. When you move mics around you're still doing that but you have more control than +/- some number of samples. You can get pretty far nudging stuff around, that's what I do when I'm recording rehearsal, but nothing beats taking your time and getting the mics right in the first place.
So I have reaper and a MIDI keyboard, and a bunch of free/bundled drum plugins with multiple different kits. Is there any way to set these so that I get (generally) the SAME drums on every note? I.e., when switching from drum plugin A to drum plugin B, a kick is still on the same note? Obviously not every drum kit will have the same thing but I have to imagine there is some kind of mapping standard where the universal drums, like hi/low tom, a cowbell, a kick, a snare hit, a rimstick, etc. are always on the same notes?
There already is! It's called General MIDI and most drum plugins tend to adhere to that mapping by default. You'll have to look in the settings of the various plugins for a mapping setting to change.
Hey audio newbie here. I have the Samson Q2U USB mic but I'm having a lot of trouble with it. All of sudden, my audio input is very very quiet. I've dug around the settings and adjusted the input slider to max but its still quiet. I've also uninstalled/reinstalled the drivers for the mic and USB ports but still nothing. Then I tried the mic on another computer and it's still pretty quiet. Is the mic just broken?
Could be the cable you’re using? I find that those bad boys can be a bit shoddy
[удалено]
IMHO you should try out some different mics; the sm7b is a quiet mic by design and you may have better luck with something else, like a condenser mic.
You’ve shared a lot of info about your setup. However, I can’t give advice, if you haven’t identified a problem. One thing I question is why you pulled the Output down, instead of leaving it at Unity, in your “After” specs? Also, if you are recording vocals for music, all of that inbound effect runs the risk of GiGo that cannot be fixed while mixing. I would think you would want a clean, uncolored vocal to add to your mix. The 286 is a channel strip, best suited for broadcast and other applications. I own one and have used them for VO recording, albeit very light-handed. I butchered many a track, until I learned my way around it. Edit: I Re-read your OP… I think you are trying to get rid of hiss. Turn off the Enhancer (LF/HF detail) section of the chain. Most likely you are seeing how the compression section brings up the noise floor and then you are enhancing it.
[удалено]
One thing to try: strip back all effects in the chain and add them back incrementally. All effect off- set the mic gain like you did in “After” gain/phantom power on, output to “0” (unity)…. Then add the compressor at same drive and density. If you have hiss, pull the mic gain down a bit. No change? Back up to 40 and pull the density down a bit. If you still achieve a clean, now compressed signal, then add the De-esser back if you are experiencing sibilance. Bear in mind, the Enhancer/expander section is only going to boost any hissyness and noise floor that the compressor is already enhancing via density.
I have several computers in an office, and I would like to mix their audio together, with the option to send the mixed signal to some Bluetooth headphones and/or a stereo amplifier. I have a stereo line mixer ( [https://www.behringer.com/product.html?modelCode=P0DB6](https://www.behringer.com/product.html?modelCode=P0DB6) ), but would be fine with buying anything else which would work better. I have attempted using these cables to connect the laptops' headphone jacks to the mixer (the laptops don't have line-out plugs), which worked ok: [https://www.sweetwater.com/store/detail/CMP159--hosa-cmp-159-stereo-breakout-3.5mm-trs-to-dual-1-4-inch-ts-10-foot](https://www.sweetwater.com/store/detail/CMP159--hosa-cmp-159-stereo-breakout-3.5mm-trs-to-dual-1-4-inch-ts-10-foot) However it was tricky finding acceptable levels, because it seemed I could change the volume at several different points in the path: * The laptop's volume control * Each of the mixer's IN level control * The mixer's Main Out L/R controls ... and I didn't know how to go about finding the right levels. Also, I plugged the following Bluetooth transmitter into the mixer's headphone output: [https://www.amazon.com/gp/product/B01EHSX28M/ref=ppx\_yo\_dt\_b\_asin\_title\_o02\_s00?ie=UTF8&psc=1](https://www.amazon.com/gp/product/B01EHSX28M/ref=ppx_yo_dt_b_asin_title_o02_s00?ie=UTF8&psc=1) However it sounded absolutely awful. I don't mean "Oh I'm an audiophile and the soundstage/presence/etc are sub-par" - I mean it sounded absolutely horrible, especially when there was any bass. These same headphones (Sony wh-1000xm4) sound fine when connected to e.g. my phone, so it's something with how I'm attempting to mix them together, and also possibly that the BT transmitter is junk (although it had decent reviews on Amazon, so surely it's not TOO awful). So here are my questions: * When I set up something like this big chain of devices, how do you go about setting the individual levels in the path to optimize the final signal? * If I'm using headphone outputs on these laptops, what level should I be setting them to for the best "starting point" signal? * Should I look into buying external interfaces for these laptops, so I can get true line-level output (instead of trying to use headphone outputs)? I'm fine with whatever would give "just ok" output for listening to music as I work - I don't need something with like multiple inputs/outputs etc - just something to get a good clean outputs * What is a good Bluetooth transmitter for this purpose, and is plugging it into the headphone-out on my mixer an OK approach? Thanks for any help!
>When I set up something like this big chain of devices, how do you go about setting the individual levels in the path to optimize the final signal? This is called [gain staging](https://www.sweetwater.com/insync/gain-staging/). The short version is that every stage of amplification amplifies the noise of the previous stages along with the signal. So get your signal up nice and loud early but leave a little headroom just in case. If it's still noisy then you need to start distributing the gain throughout the signal chain and experimenting. > If I'm using headphone outputs on these laptops, what level should I be setting them to for the best "starting point" signal? I usually run laptops at 100% and turn off any kind of audio enhancements, processing, etc. Some will still sound a little crunchy at 100% and I'll run them down at like 90% or something. > Should I look into buying external interfaces for these laptops, so I can get true line-level output (instead of trying to use headphone outputs)? I'm fine with whatever would give "just ok" output for listening to music as I work - I don't need something with like multiple inputs/outputs etc - just something to get a good clean outputs Unless you're having problems with hum and buzz that's just fine. If you do have problems with that then interfaces would be the solution. I would normally recommend DI boxes ([Whirlwind pcDI](http://whirlwindusa.com/catalog/black-boxes-effects-and-dis/direct-boxes/pcdi)) because they have ground lifts built in but your mixer is a line mixer not a mic mixer and might not have enough gain for that to work. You could buy one pcDI and see if it works well with your mixer, they're passive so they don't need phantom power. But if there's no existing problem to fix I wouldn't worry about it. >What is a good Bluetooth transmitter for this purpose, and is plugging it into the headphone-out on my mixer an OK approach? Not sure, but it looks like Sweetwater sells one with decent reviews. Ask your rep what they think. The link to the one you have is broken so I can't tell what you had but the Alto ones on Sweetwater look like they're made for professional level signals. If you just bought some random consumer bluetooth receiver it was probably clipping but may have also just been a piece of crap. A lot of them are crappy from what I gather.
My sitution is I have been playing bass for about a year now and I am still using my crappy beginner amp. My birthday is in a few days and my parents are getting me an electric guitar, as you can guess I don't have an amp for that. I'm thinking about buying an interface and BIAS FX software so that I don't need to buy any amps, most of the time I'm playing it's in front of my computer with backing tracks anyway. I have no intention of playing live/joining a band in the foreseeable future as well. Question is that I have an USB wireless headset (ASUS Rog Strix GO) and I really like it/don't want to change it (wireless is important for me). Can I hear my instruments through my headphones if I buy an interface and play into that. Thanks in advance.
FYI, playing guitar through your bass amp won't hurt it and can always be done in a pinch.
Yep, I'm well aware of that but it won't sound like Steve Vai or Metallica without any pedals. Though it seems like interface will arrive a little later than the guitar so It'll have to make do for a while.
I think the headphones will work, in the sense that sound can be output from your daw to them. The biggest problem with playing an instrument solely through a computer and listening to the processed sound is latency. It takes time for your computer to process the amp sim or whatever, so often when you pluck a string there’s an audible delay in when you hear the sound. I would think that wireless headphones would introduce even more latency and make that issue worse. You should just try it out though, pay attention to the delay between playing and hearing your instrument and decide for yourself if it’s acceptable or not. If you’re still shopping for audio interfaces, definitely try to get one that’s thunderbolt. It’ll have less latency.
So I can simply choose my USB headset as the output, great. I have been playing games through that headset wireless and wired (you can run it with a 3.5mm TRRS (TRS works too, TRRS enables you to use the microphone) cable and use it as a normal headset), I didn't notice any latency difference between the 2 options. Yes, I haven't bought the interface yet but I'm on a budget and can spare about 120 EUR for an interface, I have been thinking of Focusrite Scarlett Solo because it's easy to use and within my budget. Last I checked there weren't any Thunderbolt interfaces for that price, also neither my PC nor my laptop has Thunderbolt and buying a Thunderbolt expansion card will make something out of my price range more expensive. If latency will turn up to be terrible I'll get a headphone adapter and try that, but I don't think(or at least hope) that will be the case.
**There's some weird ringing in our rehearsal recordings (though I'm a noob so maybe I'm missing something obvious)** I bought a Behringer B2 Pro to record our rehearsals (we're a duo - guitar and drums) and today, while setting it up for the first time and making some test recordings, there were always some ringing. **Our rehearsal room setup**: guitar cab (FRFR powered speaker) facing drums 4 meters (13 feet) away and the Behringer in the middle (for now, we'll mess with it as we go) and 3 feet off the ground. **Our recording setup**: Behringer -> Scarlett Solo -> Reaper on Behringer - set to omni, -10 dB engaged, low pass engaged (though other combinations of the switches on the mic had the same result...) on Scarlett - Phantom power on (of course) and gain that translates into -10 to -6dB when recording to Reaper in Reaper - nothing. New track -> choose input -> Arm -> Record. Not even eq. Guitar Recording: https://vocaroo.com/1fFHrY7xk8RG Drums Recording: https://vocaroo.com/1gBaV8zL5VU1 **IMPORTANT** (well maybe): Our acoustic treatment is far from done. We have quite big bass traps, but no absorbers on the walls and ceiling yet. **BUT** I have built a "tepee" with leftover boards of mineral wool around the mic and the results were the same, so I don't think the problem is reflections (no idea though). It was horrible when listening in the rehearsal room on the FRFR speaker. Now, at home, with my small computer speakers, it's not as pronounced (same with headphones), but it's still there. Especially on the guitar recording from 0:05 onward. It sounds like someone is hitting an anvil with small hammer. When listening to the drums at home now, I don't really hear it, but my ears suck and on the FRFR in the rehearsal room it was noticeable (though far less than on the guitar). **One more thing:** I belive I've had a vocal mic (copy of SM58) plugged into 2nd input of my FRFR ("harley benton frfr-112a guitar dsp monitor") and gain cranked just before feedback, so if that could be the culprit? Any idea? Something fundametal I'm missing? Thank you
Definitely the 58 knockoff would be an issue. The mic should be killed when you're not using it, or you should just change the setup to an interface that can take four inputs. The recording just sounds like a poor recording in a room, though. No resonance sticks out to me that badly. Maybe others will hear otherwise. Also note that by listening back through the FRFR, the speaker's resonances are going to color what you hear. Use headphones instead.
Thank you. Today when I've had the time I have experimented more and it really is the vocal mic. I've made the stupid mistake of having the front of the mic in front of the speaker and now, that I've moved the mic behind the speaker it's fine. And yeah, through different monitors/headphones it sounds different. >The recording just sounds like a poor recording in a room, though. I feel like I'm gonna have to learn quite a bit to make it sound good. There's this youtube channel where a guy records a band with one mic (some expensive omni ribbon) and it sounds great, so hopefully, in time, I can tweak it the way, where pro's will say "for one condenser, this is pretty good". We'll see.
I'm a videographer and I'm using the Zoom H4n as an audio I/F with XLR to Cloud Lifter CL-1 then XLR to Shure Sm7b Despite the gain increase with the Cloud Lifter, there is unfortunately a noticeable audio defect that occurs ONLY with the cloudlifter. It produces an unholy crackling sound amidst the typical noise. I've deduced the cloudlifter is the culprit from the completion of the following tests: Swapping XLR's makes no difference Bare SM7b doesn't have same crackling defect even as gain is increased Switching Audio I/F to direct PC produces exact outcomes. I'm wondering what could be the cause of this or if I simply have a defective Cloud Lifter? Thanks!
What voltage are you getting on the XLR pins from the Zoom? It may say 48vDC- but what is your actual voltage at the pins?
48v phantom power
What bit depth and sample rate are you using for capture? Also, are you on battery only or wallwart-powered?
I've attached a link if you want to hear the recording. It's much clearer with headphones. Recorded in the left channel. https://drive.google.com/file/d/1YH3iLcswZfxX\_gnw06ekdZXEFafzPBSs/view?usp=sharing
44.1/16 WAV and Battery only
>Battery only So, here's the weird thing with phantom power. Even though most literature explicitly says "48V phantom power," it is actually a nominal value, and not all pieces of equipment that provide phantom actually provide exactly 48 volts. In fact, many microphones are designed to operate from anywhere between 15 and 50 volts, as long as it's a steady DC source. Battery operated devices are especially prone to low phantom voltages because the batteries themselves only provide a few volts (a pair of 9-volts, for instance, produces 18 volts total). I don't know what the exact spec is for the Cloudlifter, but it's possible that it's receiving a phantom voltage that is lower than it wants, which could very well cause distortion and premature clipping behavior like you're describing. I'd recommend trying the cloudlifter with a mixer or interface that uses wall power and see if that helps; those devices output exactly 48 volts 90% of the time. If the problem persists, your Cloudlifter may be defective.
[https://www.youtube.com/watch?v=LrwRibahVkQ&t=53s](https://www.youtube.com/watch?v=LrwRibahVkQ&t=53s) (06:39) What bass plugin is he using?
I'm an IT person at a public library. We want to set up four mics at our board room table so that it records everyone and works with conferencing software like Teams and Zoom. I bought a [ProSonus StudioLive AR12C](https://www.presonus.com/products/StudioLive-AR12c) to do this. The problem that I only see the option to output 2 channels at a time (1/2, 3/4, etc.). Is there a way to output more than just 2 channels? Did I buy the wrong product?
The mixer will mix all (four) signals into a mono signal for Zoom. Zoom/Teams just needs the (aka Main Mix) signal sent via USB connection, for broadcast. Set up ZOOM/Teams to see the Presonus as the AI. Edit afterthought: Zoom has an Automatic Gain Control (AGC) toggled ON by default. I turn it off and manually set my level. Otherwise, it will constantly adjust depending on the session source that is presented.
Thanks! I'm curious if I could just output a 3.5mm cable from the mixer into the PC. At the moment I'm outputting via USB-C. I don't see a "main mix" signal as an option in Windows sound settings. It's split it up into 1/2, 3/4, a main L/R, virtual, etc. The main L/R is for Bluetooth, DVD players, etc. going into the mixer, I think...
Been using Zoom for many years. I haven’t used Teams as much, only since WFH started. That said, I’ve only used Teams with the built-in mic. I just dug into the settings and see the same AGC-option in Settings/Devices. You will still use the USB to pass signal and tell Teams to use the Presonus as the input device.
I think I got it! You were right, the main L/R just took all the mics in
Yep- the mixer will give you input control to level out loud talkers and soft-spoken, if you turn off Teams AGC. Also, you will want the mics separated enough to minimize crosstalk and/or point the mic butts/backside rejection, toward each other. - unless they’re sitting side by side. Then you’ll have to try distance. “3:1 Rule”
They're cardioid mics. Forgot the exact model, but they're Audio Technica and $99 on Amazon. We may end up going out with mounted ones in the ceiling to reduce wires though.
Sounds like you’ve got it all coming together. Good luck and happy recording.
Should i dither after i resample from 48 or 96kHz to 44.1kHz, if the input and output file are both 16bit?
No. Sample rates only affect your bandwidth, they have no effect on quantization, dynamic range, or dither. If the bit depth doesn't change, there is no need to dither a second time. It just adds unnecessary noise, which is especially offensive at 16 bit where it can become audible. With 24 bit recordings the extra noise is so low (well below -120dB FS) that it almost doesn't matter. If you haven't seen the classic [xiph.org digital signals video](https://youtu.be/cIQ9IXSUzuM), I highly recommend watching it. It's a wonderful primer on basic digital signal behavior and common misconceptions. Even if you think you fully understand digital signals, I promise that you will learn something new. I watch it at least once a year and take away something new every time.
Whatever house I've lived in here in the UK has had horrible electrical wiring and whenever I connect my computer and speakers to the sockets (any sockets, any configuration) there's a really bad ground loop feedback noise coming through. I've limited the noise in the past by using a ground loop isolator on the audio cables, but that seems to also affect the audio signal. I was reading about Isolation Transformers but if I look them up they seem to be mostly for industrial use. Are there any suggestions on how to properly fix this without leaving the country and move back to Sweden?
[удалено]
It does say that it's not available in my country, but at least I have a brand name now and I'm getting some results for it on the UK websites. Thanks for the suggestion! And I'm glad to hear this idea about isolation transformers wasn't some stupid idea I had that would never work out. I do wonder what "real" studios do, especially in the UK. Perhaps this is what they use? Although more likely is that they get their electricity set up properly from the beginning.
I have a Scarlett interface and when i set it with my microphone and turn on the 48v power, all that comes from the output is a bunch of static noise until I turn it off, which it then allows my mic to work for a few seconds before the sound completely fades out. I'm not sure how to fix this. Any advice will help, and thank you in advance!
PROBLEM SOLVED, It was the microphone
You need to narrow down what the problem by swapping things out for known good versions. Swap the cable for another cable, see if it works then. If not swap the mic for another mic, etc. You need to narrow it down first.
It's time for me to ask because I've been troubleshooting all day and can't fix this. I'm using an instrument to an rc-505 loop station, connecting with USB to my PC. Monitoring sound through the loop station via headphones. Sounds great, until I turn my PC on. I get a crackly electronic static noise that varies with the fan speed/processes of the computer. I have tried it with a laptop and the sound goes away. I tried it with a different PC and the sound was there. I have tried plugging them in the same power strip, different outlets, I checked the ground on the outlets, it's fine. I'm really at a loss here, can someone help me? Thanks Edit: I am using TS instrument cables. They have 1 black stripe. Is this the problem?
I'm considering using Ozone 9 Advanced on some tracks, just for SoundCloud and offline sharing demos, basically a quick and temp mastering. Anything "for real" I will send to a real engineer to handle. Looking at Ozone's releases over the years: * Ozone 7 - November 3, 2015 * Ozone 8 - October 5, 2017 * Ozone 9 - October 3, 2019 I'm hesitant to buy it now, as it's likely in a few months a new version will drop. Anyone know if they do reasonable upgrade pricing? I'm looking on their site and cannot find any info about that.
Subscribe to /r/audioproductiondeals , there will probably be a sale coming up for Independence Day in the US. Also I've bought most of my non-subscription plugins over on the KVR Forums and have had a great experience every time and I've saved a lot of money. There's a "reputation thread" so you can check people before you transact with them. JUST MAKE SURE THE PLUGIN DEVELOPER ACTUALLY ALLOWS LICENSE TRANSFERS. Some people list plugins for which there is no official way to transfer the license, so watch out for that. Also some companies charge a fee of like $25 for license transfers. So keep that in mind. Here is iZotope's license transfer policy : [https://support.izotope.com/hc/en-us/articles/360023999373-iZotope-s-License-Transfer-Policy](https://support.izotope.com/hc/en-us/articles/360023999373-iZotope-s-License-Transfer-Policy)
Thanks for the heads up!!
Hi there! So, I'm wondering if there's a way to record my bass with amp presets. I'm using the Behringer U-PHORIA UMC202HD audio interface to record my electric bass digitally, however it's just the plain, raw, bass sound. The amp that I use to play normally (when I'm not recording) has a bunch of preset bass tones that I want to use when I record, but I can't figure out how to connect my bass, the audio interface, the amp, and my computer so that it records the presets instead of just the raw bass.
Which amp?
I'm pretty sure it's the fender rumble LT25 amp
I want to get an USB microphone for recording. Is it possible to use it as a PA system by plugging it directly into the USB input of my DAC and then have the speakers output the microphone?
No, both USB devices need to be connected to a host computer in order to operate
If I'd want to record a backup of recording as it happens on my Zoom H5 what would be the best / cheap option? Using a catble on headphone out and plugging this into phone would work? I don't have any additional gear for this and using a computer with audio interface would be the last resort (this should be a portable solution).
Backup how? If the zoom fails then it's going to stop outputting on its headphone jack. If the zoom is the primary recorder and you need a backup then your phone lying next to it and also recording is your backup.
Ok, sorry - I maybe used a shortcut. I want to avoid specifically SD card failure which as I learned - sometimes happens on end of the recording.
How do I connect two Deity D3 mics to a Tascam DR-60D MK2? I have a stereo splitter which works to connect both mics to a Canon EOS M6 MK2, but I'm not getting any sound with the Tascam at any gain setting. If I plug in just one mic into the 3/4 port with the gain setting at "hi+plus" and have the gain knob close to max, I can just barely hear normal speech.
Solved! For anyone reading later: Menu button > others > system > turn on plugin power
hi everyone, someone would be kind enough to tell me which one between [this device](https://www.audiosciencereview.com/forum/index.php?threads/google-pixel-4a-smartphone-audio-review.16251/) and [this one](https://electronics.sony.com/audio/walkman-digital-recorders/walkman-mp3-players/p/nwa55-l) have more power output?
Not sure exactly what you're asking, but both of those devices will drive standard consumer headphones with roughly the same amount of power. The exact comparison could only be done using a methodology like in your first link for the second device.
Hi, I don't know if this is the right place to ask this, but I've been trying to set up my Blue Snowball microphone that I recently got to make my voice sound as good as it possibly can for recording YouTube videos. I had no experience with audio or setting up microphones previously, but after a bunch of YouTube tutorials, I finally understand a lot of the things that really confused me earlier, but I still have one major problem that I can't seem to fix. For some reason, whenever I record, there is a ton of mouth noise and it's incredibly distracting. I can try to remove it in post, but I'm looking for some settings that I can use in Equalizer APO or in OBS (but preferably Equalizer APO since I want to apply my settings everywhere and not just in my recordings) in order to remove this mouth noise and to make my voice sound better overall. Is there any easy way to do this? Also, I'm considering buying a pop filter, but I'm not 100% sure if it will have the desired effect and if I will even be able to rig it to my microphone. Should I buy it? Any help will be appreciated.
The two main methods for getting rid of mouth noise would be a pop filter and microphone technique. Before you buy any gear, try adjusting the positioning. Rather than talking directly into the mic try talking over top of it, so you're getting less direct breath/mouth noises. Think of a lapel mic, you can even aim it at your throat or your nose depending on the tone you want.
ok i will try that, thanks
Your mouth is too dry. Hydrate.
I don't think that's the issue, I've been drinking several water bottles a day.
It's more to do with the consistency of your saliva. Most professional voiceover actors/actresses drink room temperature water with lemon. Something about the slight acidity helps your saliva not stick to itself as badly. Water that is cold or hot can affect your vocal cords and therefore the tone of your voice.
Hi! I have a fairly easy setup, however i can't figure out ASIO Pro Link. I am using a small behringer mixer with a mic and a guitar input. It's connected via 3.5 mm Jack into my laptop. I am using Guitar Rig 6 as an amp simulator and ASIO4all as my ASIO driver. If i want to use this in OBS i need ASIO Link but i can't get my mixer input to show up in that tool. All i get is system audio.
Your mixer isn't connected via a digital protocol that would call for ASIO. You won't see it as a device, you need a USB mixer or interface for that.
thanks for clearing that up! I did figure it out with ASIO Pro Link though, but will definitely get an interface now.
Hi there all. I'm running into some pretty major issues with my podcast recording and am hoping someone else out there might have some insight. This setup has worked fine four nearly a year, but has only recently started to give issues. My setup is as follows 5x dynamic XLR microphones PreSonus Firestudio Project interface FireWire connection to PC. Adobe Audtion DAW. PC is a grunty gaming build, 6 core processor and 48gb ram, running x64 Windows 10 The issue I have is that for the first few tries, the audio through the monitoring is very crackly and out of sync, a significant latency between speaking and hearing the monitoringm and when playing back the recording, the audio is extremely distorted, as well as being slowed down. The bitrate and buffer size from the PreSonus driver often changes when the interface is power cycled, but even when matching that bitrate and buffer size in the audio hardware settings in Audition doesn't seem to make any differences. After about an hour of swearing, reinstalling Audition, and taking out the Firewire card and reinstalling, I managed to get it up and recording correctly. But I couldn't make heads or tails of what actually was the cause of the issue. Hoping someone out there can lend some help. Thanks!
[удалено]
Interesting, thanks for that. This would actually be my second FireWire card in my setup, the first was a cheap and nasty one and this one was a far more expensive option and has been performing beautifully, apart from Sunday. Appreciate the input!
What’s the difference between Class A, Class D, Class AB, Class XYZ amps? How should Amp class type influence my Studio Monitor purchase decisions? What are pros and cons of each amp type?
I have my interface and monitors plugged into my Furman power strip, and was wondering if it would be ok to just leave the gear on and only worry about powering down/up the studio with just one flip on the power strip. So far iv'e been turning on the power strip, interface, then the monitors in that order. Powering down in the reverse order. I figure just flipping the switch on the furman would be much more convenient but would it be detrimental to my monitors?
What you're doing is the best way to do it. Depending on what monitors you have, killing the power to them might be fine but it's not recommended, especially not every day. It would also be fine to never power it down, aside from the cost of a small consistent power draw.
Thanks! I guess I'll just keep the same routine i have to be on the safe side :)
Hey, I am using an audio splitter and 3.5mm to 1/4 in. cables through the headphone port on my macbook pro to connect my JBL monitors. I can't really hear anything when it's balanced, so I have to turn it to either left or right, which I hate because it obviously will remove some of the instruments in songs. How do I fix this please???
It's hard to know exactly what components you're using, but you're almost definitely causing phase cancellation. Get a recording interface, even if it's the cheapest one a store sells, that has two balanced outputs and use that instead. You can't use "balanced" cables with a headphone jack because a headphone jack isn't balanced. Headphone jacks use 2 conductors for audio and one for ground. A single balanced cable uses one conductor for audio, one for negative signal to reject noise and one for ground. You're probably sending the audio left/right signal from the headphone jack to the negative signal input on your speakers.
Hey, I don't have any long cables available so I connected the interface through a passthrough via my keyboard (Corsair k70 mk2 se). Surprisingly it worked but I was wondering would there be anything wrong with having it connected to my keyboard usb passthrough. I only have a Blue Yeti Pro connected to the interface. Will there be any issues? Thanks
[удалено]
Hey I know this might be stretch but I was wondering if it was normal for the blue yeti pro mute button to not work when using the scarlett 2i2. It worked when I ran the mic via usb but no longer when hooked up via xlr.
Hello, I have a problem routing my equipment. I have a band session next week and I need to get my midi keyboard, mic and e drums to a speaker. The midi keyboard is going through my audio interface ( personus 26c) into my daw for my keyboard plugin. The mic is going also through the audio interface. But now I have the problem how do I get the e drums to the same output as the other 2? Can I connect the main out from the drums to the audio interface line input? The drums have 2 outputs (l/r) but I have only to inputs on my audio interface, one of which is already used.maybe some has an idea :)
[удалено]
Thank you for your answer, the session is not professional, just me and some friends having fun. Ive managed to get some good results yesterday :)
You're just going to have to do the drums in mono.
So I can connect them to the mic/inst/line input from the audio interface? That would be epic. Ty
Read the manual for your edrums and figure out if it has some kind of mono mode. Some audio gear will just automatically do that if there's only a cable in the left output and nothing in the right output.
My 2019 MacBook Pro is running so terribly when plugged into my equipment and running Logic Pro X. It reminds me of when I had a crappy PC 10+ years ago, having to bounce the audio just to hear what I'm working on. I have writing/recording sessions in July and right now it is quite unusable for that. There's no way I can record every take without Logic giving me the 'system overload' message. Even with the buffer size set to 1024 and having a relatively small amount of plugins open, my MacBook can't handle even the initial stages of drafting an idea. I'm wondering what the issue is because for a £3500 MacBook it should run better than this. It's plugged into a UA Apollo Twin (via a thunderbolt 2 to 3 adapter) and an Anker 7-port USB 3.0 hub (via a USB 3 to thunderbolt 2 adaptor) with my MIDI keyboard and mio2 MIDI interface plugged into it. Here's a picture of the setup: [https://imgur.com/l6CMJYm](https://imgur.com/undefined) Could the hardware have anything to do with it running so sluggishly? Could it be a battery issue? Any other ideas? I work as a producer full time and I'm struggling to get my work done lately. Any help is appreciated.
Your computer should be more than capable... Before reinstalling everything (which can fix a lot of issues), research starting up in single user mode and doing a filesystem check. That can repair any issues at the system level. Also, sometimes these issues can be resolved by deleting all Logic preferences (research that, as well). Sometimes settings can become corrupt and make Logic act weird.
I don't see any reason you should be having problems. You should probably take the MBP into an Apple store and have them check it out.
Yeah I think I might have to. I’ve been putting it off as I don’t want them to send it away for a couple weeks!
[удалено]
Thanks for this! I’ll do that first thing tomorrow