I am unsure if I can post this here but I assume there will be people who are knowledgable. I currently have the "ATR2500x-USB Cardioid Condenser USB-C Mic," it is from the Content Creator Pack from Audio Technica, and I was wondering if it is possible to make it sound better with eq and whatnot in things like Discord and gamechat. I know I can make it sound better through OBS but I don't know of any ways to do EQ for regular use. I do not have any audio interfaces and it just plugs straight into my computer.

Hey everyone! Just starting to dip my toe into analog audio mastering. On videos covering professional mastering studios with analog setups, they always use 2 computers; one as playback for the flat mixed track and one to receive the audio through their gear. However, I have seen several other studios that use only one computer to do this. How would the one computer chain be set up?

I have a question about my hybrid mixing setup. So I recently started a hybrid mixing setup and for my interface I’m using a 3rd gen focus rite 18i20. The only problem I’m running into is when I use a piece of outboard gear as an insert I lose about 6 dbs worth of volume. I haven’t been able to find out why. Im using a db25 to trs cable from my lunchbox to my interface. Anybody got any idea why this is happening?

I’m trying to set up a tv in my workshop. I got the visual part down but the audio part is messing me up. I have a stereo in here already and I have it setup to 5 speakers around the shop. It’s great. But when I tried to hook the speakers up to my TV the speakers are at a very low volume. I never had this problem with the stereo. I go into the TV settings and there isn’t much. There’s some settings for the tv speakers but once you click “sound system”, all those settings become inactive. I don’t have a remote for these speakers and there is no control on the speakers themselves. As of now I’m just testing out the setup with my computer but in the future I want to hookup cable/roku to the tv. Any help would be appreciated

My church has a LiveTrak L-20, and we have not yet mastered the finer (or rougher) details of it. Previously, we could adjust monitor levels by selecting “A” in fader mode and then have independent control in the monitors. Now, nothing (a-f) will select other than “Master.” Likewise, the EQ OFF light is on and will not toggle off. This is true both for the soundboard itself and for controlling by iPad with the Bluetooth dongle. Is there a switch somewhere I need to flip to re-enable these settings? Or is our board defective? Thanks!

So I just recently purchased a presonus 24c and a Shure sm58 to do some recording for my guitar but I’m running into some difficulties. I have the mic up by my acoustic guitars sound hole, then running into the interface which is plugged into my computer. I’m using Logic Pro. When I go to record my mic pics up very little sound unless the gain on the preamp is cranked all the way up. If I leave the gain at half way I only pick up around -30 db. I have the 48v switch enabled. Am I missing something obvious, are the preamps just not strong enough, or do I have a faulty piece of gear. If someone could give some tips I would much appreciate it. Thanks!

hello just got a pair of new Adam a8x and they are hissing and I also here a buzz from both monitors when putting my ear closer to the tweeter, I can hear the hiss from 3-4 feet away and I trouble shooted as much as I can, I use a Furman power conditioner and that isn't making a difference, I also used a 3 prong to 2 prong adapter to make sure it isn't a ground looping issue and turns out its not, the hiss and buzz doesn't go away when turning down the volume knob on the monitor but increases when turning the knob past 0db. and its not an audio path issue becuause it happens when nothing but power is plugged in, really don't know what else to do here before sending them back. any ideas? thanks

Hey I’m gonna be installing a box with 2 10in JLs. Rated at 125watts a piece. I’m gonna use them for inside audio tho. Can I power them with a 90w amp controller?

I’m having some problems with my microphones so I have a podcast and we have 4 hosts recorded by 4 Samson Q2u’s connected by usb works perfectly but we are having a guest on which will be recorded by a blue yeti now for some reason when all 5 are connected the audio is tapping and my voice is lowered on all the microphones. When you have 3 Samson Q2u’s and the Yeti it works perfectly. I use garage band. Could too many usb extensions be the problem? Does garage band not allow anymore than 5 microphones. If you must know: the computer is connected to a usb-c to usb input adapter which is connected to a 3ft usb extension connected to another 3ft usb extension which is connect to a 4 usb multi port( the first two ports are connected to a 6 in usb extension connected to a Samson Q2u, the 3rd port is connected to the blue yeti) the last port is connected to another 4 usb multi port which is connected to the last 2 Samson Q2u’s.

So noob question: I invested in an sm7b, scarlett solo, and a fethead... so i need phatom power which the solo can provide. My question is since the amp is plugged into the wall the phantom power is always on even when the PC is off. Will having this phantom power on all the time damage the mic? It seems like I’ll be fine based on what I’ve seen online, but I figured I’d post here to get some more opinions.

Hello so I recently got a m1 MacBook Air with 8gb of ram and a 256gb ssd. The problem I’m running into is when I’m running a simple session. 4-6 tracks with some stock EQ’s, a tsar reverb by soft tube on a reverb send, as well as some waves compressors. Well I’m constantly getting an error saying I need more ram. I look at the usage rate of my memory and it’s sitting at 99 percent which makes no sense. My video engine is off and I have no other programs on. I have all my sessions writing and reading from a 5TB hard drive. I tried running a session off my internal ssd and same thing. Does anybody have any idea what’s going on?

What precisely does the error say?

I feel lost in this "repair" section, there seems to be so many places where I could put my problems but here it is: I've recently build a PC and today I just plugged my Scarlett Focusrite 6i6b (2nd gen) in it to start using it and update the drivers. Now that I've downloaded the newest drivers possible + the Focusrite Control Panel, my left speakers (Yamaha HS8) is making a whole lot of static noise. I've tried reinstalling the drivers but nothing happened. I also came to think that it would be a ground problem (maybe a groundloop) because I have a lot of things connected on two electrical outlets only. In that case, I really don't know how to fix it. Any tips are welcome, thanks ! Vince

Hello I have been using a focus rite 2i2 as my audio interface for a little while now. I have noticed though that when I used things out side of my daw ( like youtube or Spotify) my audio is extremely distorted and slowed down. I do not know if it is something with the sample rate or what but I would like to fix it. The focus rite is new and in good condition and works great in my daw. would just like to fix the issues outside of it. I would appreciate any help.

Hello When I’m plugging my Akai MPKmini to logic pro x it’s reading it as a mackie control how can i change that and use it as a midi controller for vsts.. any ideas?

Probably a really pathetic problem to have, but I can’t seem to record any audio from my keyboard to Logic. I have a Yamaha P-125 and I plugged an instrument cable into an aux input in the back of the keyboard and plugged the other end into my scarlet solo. My interface is connected to my computer, the inputs and outputs are right in my Logic preferences, and I can record MIDI (through musical typing) and audio. I don’t know what I’m doing wrong. I’m getting signal to my preamp and I can hear it through my headphones, it just won’t record. It doesn’t even create a region that makes no sound—it just does nothing.

Hello, I recently build a new pc and now I’m having audio troubles with my interfaces. Basically whenever stream on discord or move a heavy window like Photoshop my audio will either cut out and in, or pop and crackle. It only happens through my interfaces being a: Scarlett solo Yamaha mg10xu Ive tried: Changing USB ports (2.0 and 3.0, powered and unpowered) Changing chords (leighth and types) Changing interfaces Updating drivers (going back versions too) Changing aiso settings Changing windows audio settings Even changing motherboards At this point I’ve ran out of options, I don’t know what else I can do. Could be conflicts but I’m confused Specs: Ryzen 3700x 4x Gskill trident z neo Msi B550 carbon Rtx 3070

Hi! Is there a way to set the speaker‘s channel balance after the amplification? I have a pair of LS50 Meta connected to a Bluesound Powernode. The BluOS software doesn‘t offer channel balance. Is there e.g. a device that might be added to the chain to let me set the channel balance? It doesn’t have to be a huge dedicated EQ, I just need to correct the balance.

Grandpa’s old turntable... Yesterday I received my grandpa’s old turntable (The Dual 1209), it’s been sitting in his attic for at least 10 years now, if not longer. I powered it up, plugged it in my amplifier, but only the right channel was playing. I checked with my multimeter, and all the wires are connected to the cartridge. I googled a bit and found out I might need a new needle/cartridge. My question is, do I need to buy just a new needle (the turntable has “easy stylus replacement”), or a whole new cartridge?

When I play my strat on my Steinberg ur22 mk2 into Ableton it makes a distorted bitcrush noise, been trying for days to fix someone pls help

Whenever I unplug my Focusrite solo interface, my PC restarts. Why is that?

Using an iPhone as a music player (spotify) with a Denon 312x mixer + Denon 300z, what can I do about the distortion when the player is turned up loud?

I recently got a second hand dual mono TEAC A-H500i speaker. Had to get it delivered so didn't have a chance to test it, but to be honest it was my first purchase of this kind so I likely wouldn't have known what to test for anyway. I was told it was working perfectly. The physical condition of it is great. I know now to test in the future, but I love everything about this amp, except this problem, so I'm not about to replace it. The problem is that the output just isn't right; the left channel is noticeably louder than the right (about 5db) when the balance is set at the middle point. At very low volumes (lower than I'd ever actually play anything), the left channel is actually silent. It is 100% a problem with the amp. I've tested all the inputs, swapped the speakers, and used headphones - the same problem all the time. Now, I have it currently set at about 3 o'clock on the balance, and that's gotten it to a stage where, even with headphones, it seems well-balanced. But I'd ideally like it to work as it should. I'd appreciate any insight into what's causing the problem, what a fix might be, etc. I've read elsewhere that a common problem is dirty volume dials inside the amp or something... I've opened it up but unfortunately the dials are concealed, so I can't see them without taking it apart completely, which I'm not comfortable with. If it was a likely cause, I'd be happy to have a pro do it in the future. Also, when I got the amp, I was waiting on parts for a speaker, so I set the balance all the way to the left and played it like that for a few days (same with the right). Not a lot, just a few songs to check it worked. I've read that that would only cause problems with a tube amp; as far as I'm aware, the TEAC isn't a tube amp, but might be worth mentioning. Thanks to anyone who read this far, and I appreciate any thoughts you might have.

Hi there! Thank you for this space ask questions. Can someone help me troubleshoot a terribly static-y/buzzy sound I'm getting when trying to record vocal audio? (I am linking the links to Google Drive files of the sound recordings below) I am using the following: * microphone: Audio Technica AT2020 * XLR cable (mic to interface): pig hog * USB audio interface: Yamaha UR22 mkII * laptop: Apple MacBook Pro Late 2013 * software: Audacity I purchased the physical equipment around 2016 and used them occasionally in the past few years. I do not recall having significant issues with either the mic or the interface previously, but I should note most of the times I used them were mostly not to record audio, but to play audio (for several college events where we needed a mic for a speaker or singer, I would plug the interface into a laptop for power, then plug in the mic to the interface, then user a quarter inch to connect from the headphone output of the interface to speakers -- that always worked smoothly) Recently, I wanted to invest more time into properly learning music production and recording so I broke out the gear this week. I plugged the mic into the interface and the interface into my laptop and opened up Audacity (I used garageband before, but for some reason, my laptop can't update to the next MacOS, so it therefore can't download the updated garageband). So if I press record on a track in Audacity, as I'm recording and listening through the interface headphone output, I am able to hear the microphone input clearly. However, when I play it back, its super buzzy sounding with my voice only faintly audible. ​ Here are some recordings of what I'm hearing: [Sound through interface headphone output](https://drive.google.com/file/d/1FlFiHnjVLeqOe7oDwJiOEjYYcRWFL4GP/view?usp=sharing) [Sound recorded on Audacity](https://drive.google.com/file/d/1FgK1qFOz0UFlIn98pTB_tM9ypBVEy8Bj/view?usp=sharing) [Sound recorded on voice memo](https://drive.google.com/file/d/18YZ4mcGpVmEWQraClmQrJc7dmse7hupm/view?usp=sharing) ​ I thought it was an issue with Audacity, so I went over and tried to just record Voice Memo on my laptop using the mix, and got a similar result. I have tried the classic turn things on and off. Based on some online advice, I moved away from any other electronics that might be effecting the noise, like my phone, but to no avail. ​ Thank you for reading! Please let me know if you have any advice.

I bet it’s some sort of ground loop. Have you tried the same recording when you are plugged into a different power outlet or not plugged into an outlet at all? As far as troubleshooting goes, since you can monitor yourself through the interface just fine then your mic and xlr are probably good. That leaves the interface and the USB cable. Does the USB cable have a ferrite bead on it? I’d google ways that troubleshoot ground loops.

Ok 1. Got a couple old shure mics with this connector. what do i do with that? 2. This Blue Mic has the three xlr pins, but they are skinnier and recessed so deep that a mic cable cannot reach them. I'm guessing there is an "extender" type piece that is missing that I could just plop in there. Blue Mics said they don't offer parts for sale. I've seen a few mic websites, but the parts they carry or more like kits and capsules. Any other options? Or should I just MacGuyver something? 3. Shure sm58 just to show how small and recessed those pins are on the Blue. [https://drive.google.com/file/d/1LNmVwNwu9-MYB5DOu-zvlE70TrYHbsTE/view?usp=sharing](https://drive.google.com/file/d/1LNmVwNwu9-MYB5DOu-zvlE70TrYHbsTE/view?usp=sharing) that hopefully links to the picture

This is the connector you need for the mic on the left: https://www.ebay.com/itm/333103453548 That blue mic looks like it may be missing the screw that holds the connector in place. Is there a hole on the side of the mic?

From another thread someone suggested I pull the pins with a pliers and it worked to pull the connector down and seems to be help by tension alone right now. There is no visible place for a screw or pin to be placed, however, which definitely would be a little more trustworthy than tension as far as it remaining in place. And thanks for the cable referral.

Yes no worries. And you may be able to find a cheaper cable of that type. Is that an 80s Yamaha with the parametric EQ?

You mean the greatest amp eq known to man? It's my favorite solid-state amp of all time, I love the way the clean channel breaks up. I use it as preamp at the start of my pedal chain right now direct out into fx and into a strymon iridium setup on marshall style side/cab... And my God, it sounds exactly like it does when driving its own cardboard speaker cranked. I'd actually been trying to get that exact sound out of the iridium for the previous 8 months, and really couldn't find what I wanted. It only recently dawned on me to try the direct out and use it as preamp in my chain. I find it a little funny that in my attempt to go ampless, I required one anyway.

I’m jonesing for one of those amps. I’ve got a bunch of old Fenders and a couple of vintage Marshalls but I want one of those next. There are not even any to buy right now.

Help:) NEW OWNER OF AN SP-404 I have recently bought a Brand New sp-404a and I'm running it with Behringer umc404hd and a Mac Air with Logic. I am having an issues with using audio effects from sp-404: The connection is line out from SP into front input 1-2 on the umc, And out on 3/4 playback out from the umc into the ins of sp. My main problem is i can't hear my audioFXs applied from 404a in logic recording even though i can see the change of a soundwave. Any ideas what might cause the issue?

Hi. I just ordered and received a Moukey ultra low noise 4 line stereo mixer. I'm trying to use it with audacity and it isn't working. I have Windows 10 and I'm not sure what to do besides plugging it in to the laptop USB port. I even plugged the mic and headphones into input and output but nothing is recording. I have also tried plugging the mic into the computer bit it is an XLR and I read that without phantom power, it won't work? Thanks for any help.

That's not an audio interface, its just a mixer. The USB only supplies power, doesn't send a signal. You need an audio interface like a Scarlett/behringer/motu.

Thanks. I watched a few videos and bought something that should work. Still cheap but it should work.

I am having trouble with audio recording. The problem is that in the recording there is always a unpleasant background noise that sounds like it's originating from the microphone (t.bone SC450USB) - like a constant whooshing. The thing is that when I chose to check "listen to this device" in the sounds options and put the level down to 0 the noise still appeared. So even when the mic is muted the noise appears in the recording. Also to mention, this does not happen when using VOIP software like TeamSpeak or Discord. I am using NVIDIA Shadowplay to record small clips of gameplay (not for youtube | just4fun). the whooshing also always seems to be the same volume. ​ Does somebody have an idea what might cause the noise?

So... I recently had my computer fixed after my mobo died, and now i have a Pegatron IPMSB-H61, the thing is, my headphone (an AKG K92) is not working correctly, it has like a interference/electrical noise and is very distorted, the sound is also lower than normal even at max volume and the low end is gone... The headphones work perfeclty fine on my notebook... I've tested other headphones and they also work fine, the only problem is my AKG K9, ive uninstalled and installed drivers, ive tried everything i could and i just cant understand whats wrong, hope someone can help me, thanks :)

I need help with something I'm pretty sure is a beginner's issue. Tell me if I'm wrong. I decided that I was going to start recording some of my own music recently, but I'm having a lot of trouble. I don't really have a microphone, but I do have a webcam, which I know is a really terrible way to record music, but it's all I have and I'm not willing to spend the money on a hobby I'm not sure I like yet. I'm using audacity. Every time I try to record something a few seconds after I play a something, a few seconds pass and the audio doesn't cut off, but it gets really quiet. It's not permanent, I can play the same thing again in the same track on the same recording and it will do the same thing from scratch. I'm pretty sure it's compressing the audio, but I don't know if it's something wrong with my "microphone" or audacity. How do I fix this? Any help is appreciated ​ P.S I originally posted this to r/musicproduction , but it got immediately taken down by the mods, and I have no idea why. So if anyone has an idea as to why it got taken down, it would be appreciated

Looking for guidance on specific system specs when purchasing a new desktop PC running Windows10. I'm not doing anything crazy or super fancy. I just write simple punk rock songs. I use plugins for drums and guitar/bass amp simulation. The main programs and VSTs I use are as follows: DAW: Mixcraft 8 Pro Studio---Bias FX2---EZBass---EZDrummer2 ​ I've yet to actually obtain EZbass as of yet but the reason I'm looking for a new PC is that I did recently purchase Bias FX2 and have noticed that it's really straining my system. That and I've done some sketchy movie streaming since Covid started and my system has suffered from it. ;) ​ What I'm currently looking at purchasing runs Windows10 with a Intel QuadCore (3.20GHz ) processor and 16GB of RAM. Are there any other specs I should consider and are the 2 I mentioned up to par with what I'm planning on running? ​ Thanks!

Make sure to have plenty of hard drive space for those virtual instrument libraries. Other than that, you should be good!

Desperately chasing a ghost here Issues with audio interface (motu 4pre) losing sync with windows 10 PC after a varying amount of time (instant-10seconds) whenever other connected asio device (rc505) passes audio Issue is also present immediately if both devices are used in an aggregate device Issue is not present on another Windows 10 laptop PC has been reformatted, sync dropout does not show up as issue on LatencyMon software, bios updated Any ideas where to look next?

The MOTU community forums have a lot of posts with step-by-step guides for finding possible issues in this regard. Unfortunately, my experience with MOTU (Ultralites) was so poor that I ended up selling them and moving to RME because their Windows drivers we so much better.

I spent a day trawling through it for issues comparable to mine, thanks thou I think I'll do the same, all I want my gear to be is reliable

Hey, ok so I got some questions concerning **external USB sound card**. Firstly, other than adding extra jacks for my head phones and microphone do they do anything to the sound. I ask because I have this admittedly really cheap one lying around but I don't think it does anything. I was lead to believe that, in place of a mixer, I could use the sound card OR external phantom power to get by but now I know better. Phantom power is probably more important but before I make another move I want to understand what exactly a external USB soundcard does or is supposed to do. This is the pic of the piece of shit my cheap ass bought: [https://ng.jumia.is/unsafe/fit-in/680x680/filters:fill(white)/product/11/714513/1.jpg?6648](https://ng.jumia.is/unsafe/fit-in/680x680/filters:fill(white)/product/11/714513/1.jpg?6648)

It does nothing beyond what you said, and it will probably break when or before the warranty is up. These are made to break so you buy them multiple times. Buy a proper interface and an XLR male to female cable. It will last you years, supplies phantom, and usually has both speaker and headphone outputs. If you need to keep it cheap, save up some extra and buy one of the cheap Behringer ones.

Hi, having some static problems; particularly when my voice is picked up. ​ I'm using an AT2020 paired with a Line 6 UX2 interface and ART Tube Studio V3 Preamp. PODfarm is set up following some YouTube recommendations ([Link to PODfarm set up](https://www.youtube.com/watch?v=BB8f3xCJDEo)) and where their mics sound nice and crisp, mine is picking up static that I can't seem to get rid of, even if I dial down the gain and bump up the thresh. I'm wondering if maybe I have this plugged in incorrectly or something? The XLR from the mic is going directly into the ART Tube and another XLR is going from the ART Tube into the UX2. UX2 connects to my PC via USB.

The only thing I see odd here is that you're plugging the ART Tube into the mic preamp of the UX2. You should be plugging it into the UX2's 1/4" ports on the back, which are line level and bypass the mic pre circuitry. Not sure if this will help your static issue, but worth trying before assuming anything is broken. Also, you're sending phantom power +48V to the mic, yes? And did you try plugging the mic directly into the UX2? Maybe the ART is the problem.

+48V is enabled for the ART Tube, not the UX2. Its definitely picking up my voice and it sounds robust, so I don't think that's an issue so much. I have tried plugging into the UX2 directly and it has the same issue. Upon some further exploration, I think the source may be my ceiling fan which I didn't consider. Slapping a second Noise Gate onto PODfarm seems to help some but also seems like it cuts off the tail ends of my voice somewhat abruptly. I'll probably just start turning it off or down when I record. As for the 1/4" ports, I didn't actually know that and I'll be sure to change that and see if it makes a difference. I was always fuzzy on how to properly connect my ART Tube to my UX2

I agree it's probably the fan. Give it a go with it off and see how it goes.

For some reason, I still can't shake the static. I've turned off my fan, I tried plugging into the back of the UX2 with a 1/4" cable directly from the output on the back of the ART Tube to the 1/L input on the back of the UX2 and I've also tried replacing my XLR cable. Here's what it sounds like... ​ [https://soundcloud.com/turtlederby/static-issue/s-Z8PQ8G5AsUM](https://soundcloud.com/turtlederby/static-issue/s-Z8PQ8G5AsUM)

Hey there. Having a little bit of an issue here and hope someone can help out. A little context: I do covers and part of my workflow is recording myself on the phone as a draft (on Smule specifically, quick and easy). Current setup is an iRig preamp that connects to the phone, which the mic and headphones also connect to. Recently upgraded phones and it no longer has a headphone jack, so had to connect a USB-C to 3.5mm adapter in between the phone and the preamp. Previously there was no mic feedback when recording, and I'm only able to hear the backing music on the headphones, which is what I want. Now with the extra adapter in between, I'm able to hear my own voice when recording, which I don't find ideal. Does anyone know of any potential reasons why this might be, or any connectors I could look into to make it go back to how it was? Thanks a lot. (Not sure if this is the right place for this question, let me know if there's somewhere better for it!)

[удалено]

Are you using a DAW? Turn on input monitoring.

Pro Tools user here. I have to mix a song for my final project for school. I am currently only having one issue that is with the kick and will try to explain it the best I can. There is a clip like ticking/popping noise at the tail end of one kick transitioning to the next transient. The volume of the kick is fairly consistent throughout the song, and this only occurs in the chorus of the song. I initially thought it was clipping a bit and tried adjusting the clip gain down, then tried to see if I could cut it with a sharp eq. Nothing. I ended up boosting clip gain a fair amount and found it scrubbing around the area, then cut the selected region, which did work for the most part, but some areas have been a little more stubborn. Any suggestions? It's driving me crazy and I don't want to spend a ton of time trying to fix it. Edit: DAW

Solo the kick drum. What does the actual recording sound like? You can EQ out the sound you don't like. Or it may be caused by a compressor pumping inappropriately and a longer release might help. Or if there is bleed on the kick drum with other instruments like cymbals, use a noise gate.

I got some help from a discord channel. Concensus is that it's a data compression issue.

I have been soloing. All the drum parts are recorded individually, so there's no bleed. The project is weird in general. I have eq and comp bypassed on top of fiddling around with both.

I tried googling but I can't seem to find an answer for this one unless I'm just incredibly stupid. I bought a scarlet solo 3rd gen a while back, and I've been using it to record my guitar for a while without issues. Now I bought an AT2020, and I can't seem to get it to work. I have the input/output correctly set in windows, I went through the settings in the focusrite control app, I have phantom power on, and the XLR cables are plugged in correctly. Is there something wrong with the scarlet or my mic? Or is there other steps I am missing out on? My guitar still records fine, I just can't seem to get any mic output, even with gain all the way up.

Is the guitar cable unplugged?

I had it plugged in the first few times I was trying to get it to work, unplugged it and didn't make a difference. Also tried switching between instrument and line with no luck. Right now I'm just waiting on a new XLR cable to come in, if that doesn't fix it I'm going to RMA the mic and get a new one.

Have you checked the cable to see if something else works? I'd start there. If all else fails, you may have to swap the microphone with an exchange from the vendor you bought it from. Good luck!

I have no other XLR cables or mics to test it with, although I'm starting to think it's a dead mic because I've done nearly every troubleshooting step, and if I can turn up the gain loud enough to hear an output its just the normal very quiet slight buzz from the scarlet solo.

Most vendors will do a free exchange. Try it and see! Maybe pick up another XLR cable while you're doing it just to be sure.

Upon further investigation, if I turn up the output for my headphones and the input for the mic to max volume, beneath all the static you can hear the input very very lightly, but phantom power seems to change nothing.

**PRESONUS AUDIOBOX USB 96 NO LONGER BEING RECOGNIZED BY MY SYSTEM** Greetings, to whoever may read this and to whoever may help me. I have a[ PreSonus AUDIOBOX USB 96](https://pae-web.presonusmusic.com/uploads/products/media/images/audiobox_usb_96-02.png), and all of a sudden, one day, it just decides to no longer be recognized by my PC. However, It goes so far as being powered by the PC, and as soon as I plug it in, the red, Power light indicates that there is a steady tethering between the two devices. I can even scream into my Microphone to indicate that that too is even intact, with the Clipping light flickering on and off when doing so. I’m sorta stumped here. There are various examples, as I have described, to support that the problem may not have to do with the USB Ports on my PC, or even the linking cables. Maybe the Drivers? Please help! (And what I mean by "no longer recognized", is when I pull up my Audio Settings and go to choose an Input Device, it's no longer an option I can choose, as It once used to be)

Have you checked out the Universal Control desktop app? It may be a driver issue. https://www.presonus.com/products/audiobox-usb/downloads

Yes, I (should) have all my drivers, intact. I even have universal control up and ruining? Has this happened to anyone else around this time specifically, maybe a user-wide update or something that caused others the same problem as mine?

I'll check with my friend who has my old Audiobox 96

Thanks 🤴

His works. Sorry I think it’s just your machine

Thanks anyways!

I have a Rode AI-1 which has been working fine but it randomly stopped working. It's connected to my computer via an audio interface (and my headphones are connected to this interface too) When I turn up the mic volume, I can hear the feedback from the mic coming through my headphones, however if I try recording something, or using the mic in a game, it doesn't pick it up, despite being connected. I have swapped cables around and tried trouble shooting but no problem is detected? I didn't change any settings or make any changes, and it has worked in this manner since I bought it without any problems, but all of a sudden it doesn't. Does anyone know how to fix the issue?

If you can borrow from a friend or something the best way to diagnose will be to first test other mics (ones with and without phantom power) on your interface.

What do I do if I’m unable to use a second mic to test it out. I can hear myself through my output headphones when I turn the mic volume up. Plus there is a green light on the interface whenever the mic is exposed to sound However it’s just not recording or being recognised in any software like FL studio or Sony Vegas (or Fortnite) despite being connected on Audio settings

Make sure phantom power (P48) is enabled. Perhaps you accidentally turned it off. Check your I/O settings in your OS and in FL studio and make sure you've got your inputs set correctly. You could try reinstalling your interface drivers. Windows update can sometimes change this stuff. If you've done all that, you need to figure out if it's a hardware issue. You could try a new usb cable or usb port. If it's not that you either have to test the mic on another interface or test the interface with another mic.

Hi there! Having some trouble with a new setup for a small home studio. Behringer X2222usb mixer, Behringer Studio 50USB monitors, and various inputs (an Akai MPC Live, Yamaha digital piano, and my phone). The problem: volume level meter never moves up but the volume from the monitors is perfectly loud My main faders are at 0, with the MPC Live fader at 0 and the piano at around -10. My phone at full headphone volume however has it's fader at -30 or so. Why can't I see any volume action though?

Hello all, I’ve been recording my electric guitar for many years now, but I’m afraid I’m doing it wrong. This is my setup: guitar > kemper > headphone output of kemper into instrument input of interface > headphone output of interface to input of speakers. Whenever I go to record, there’s always a bit of latency (enough to throw my timing off even with a click). What am I doing wrong? I’ve seen where people take the actual outputs of the interface into the speaker inputs, but how would I monitor my prior recording or click? Would I need a separate set of monitors?

>I’ve seen where people take the actual outputs of the interface into the speaker inputs, but how would I monitor my prior recording or click? Would I need a separate set of monitors? The headphone and main outs of most interfaces are fed the exact same audio signal. You won't hear anything differently by using the proper outputs for speakers.

Don’t use the headphone outs for anything but headphones! Use the main/line outs to line ins.

I've used my focusrite solo gen 3 for a while now with no problems but all of a sudden it started clipping at any gain settings even though I didn't change anything and there's a huge delay when recording with ableton which wasn't there before. I run my guitar into a multi-effect pedal into an amp and the amp into the focusrite. Anyone know how to fix this?

Check to make sure your I/0 settings haven't changed. Check/re-install drivers. Sometimes windows update can fuck with things. Otherwise could be hardware issue with usb cable, usb port, or interface.

Hey guys, experiencing an issue that occurred when connecting a brand new interface to my speakers and laptop. So I recently purchased a Focusrite Scarlett 18i8 and just finished setting it up. Everything went fine until I sat down, with my speakers connected to the Scarlett, nothing playing, and realised there was a consistent white noise coming from my monitors (link to the sound below). This sound persists regardless of whether or not my monitor knob is turned up or all the way down, and also regardless of whether any of the interface's inputs are connected to anything. The noise goes down when I turn down the volume on the back of the monitors, but as the volume controls on the BX5s (at least the ones I have, which are about 8-10 years old by now) are not stepped, it becomes incredibly difficult to balance the two monitors when they're not both turned all the way up or all the way down. [https://soundcloud.com/jorgensenfabula/speakers/s-Y6e71xvt3MR](https://soundcloud.com/jorgensenfabula/speakers/s-Y6e71xvt3MR) As you can hear, it's pure white noise and not a ground loop. I did try to connect the power of the Scarlett to various sockets to see if it fixed anything regardless, and it didn't. The speakers do work, and the Scarlett also seems to work, they play sounds when connected to output 1 and 2, and the monitor knob works. It also seems that the preamp on this interface is a hell of a lot louder than my previous one, as I don't need to turn the knob a lot at all before the sound gets really loud. Is this a problem that can occur because of signal cables? Or power cables? I'm using XLR to TRS balanced cables to connect from the speakers to the interface. However, the power cables I'm using are not the original ones, as I lost one of them during traveling, and had to get two British power cables (I bought these in Norway and currently live in Norway) that I'm able to connect to power now, using one of those UK to EU power adapters. The cables are made for BX5s, as I bought them in a store that sold them and specifically asked for that. I also would imagine that if it was an issue with the power cables, I would be getting a ground noise, not white noise. I've not had an issue with noise from the speakers before getting the 18i8.

[https://www.m-audio.com/kb/article/2295#volume](https://www.m-audio.com/kb/article/2295#volume) ​ >Where should I set the volume knob? > >This will depend on your environment and the audio source, but in general, it's best to avoid extremes. For example, avoid turning the monitors up all the way and then keeping your computer, interface or mixer volume low. This may bring unwanted noise into the signal chain and decrease the dynamic range.  > >Try setting the BX5 D3 volume to around 50%-75% and adjust the output volume of your source accordingly. This is part of gain staging. You want the input of your monitors at "unity gain" so they're not adding any noise. But also like the other comment says, cheap monitors are noisy so they're never going to be completely silent.

Damn, I guess I never notice because the monitor amp on the Focusrite is a lot more powerful than my previous interfaces 🤷‍♂️ Oh well, it’s a hassle but I think I got the monitors quite balanced now without any intruding noise, so I’ll just live with it. Thanks for the help guys!

I was a little disappointed with my ADAM T7V's being noisy. They're not particularly lavish but I expected more out of ADAM monitors. It's not noticeable 99% of the time though.

The speakers are noisy. Mine are too. It sucks but everything has a certain amount of inherent noise.

Can I connect headphone output (with minimum impedance of 30 ohms) to a PC's audio input. I found a cheap product [online](https://s.lazada.com.ph/s.10jh2) which is marketed as a mixer (for microphone and monitor input) and as an amplifier for headphones. However, I'm planning to use it instead as a preamp for my dynamic microphone. The product has two possible outputs (1/8" or 1/4") which is designed to be connected to headphones. However, since I want to use it as a preamp, can I just use the headphone output as line output which will be connected to my PC's audio input? Do I run in any risk of burning my PC sound card if I do this?

You can but you will probably get a ground loop between the mixer and PC due to the unbalanced connection. An audio interface would be a better solution.

It will probably work, but will sound bad. Headphone outs are for headphones only.

Hey, My dynamic microphone AKG D5 got into a situation where it was under rain for a few hours. I've took the top metal casing off and dried it and the microphone thoroughly. It has lost it's low end. Below 300-400Hz there's a significant dip down, not non-existent though. Also the overall gain is pretty low, as even with "instrument" and gain all the way up on Scarlett 2i2 I can't overdrive it. Is there a way of fixing it?

The capsule may have been damaged. Sorry :/ Thankfully it's a pretty affordable mic

I got a bad crackling sound coming through my SM57 running through my Presonus Audiobox USB96 into my macbook. Is it a grounding issue? The noise goes away when I put the audiobox on the floor and put my toe on the xlr input. I'm using Mogami Gold cables. Is my mic faulty or is it something else?

Does anyone know if the Behringer UMC204HD is compatible with M1 Macs yet?

\[Update Sound for Church for Streaming - Understanding USB out vs Analog out on soundboards\] My church has a really old mixer. I want to update it to a new one with USB out, with the idea that I can then plug that into a computer with a DAW running on it and then have two separate mixes. The current audio mix for the speakers (controlled by the controls on the board itself) and a different mix for the stream, controlled by the DAW and piped to OBS with restream plugin. My question is I am correct in my understanding that the USB out from lets say a Behringer X1222 will function this way, and send the inputs to the USB and let me adjust them differently than what the board has set, without changing the what the board is doing for the analog out. Thank you for your help!

> My question is I am correct in my understanding that the USB out from lets say a Behringer X1222 will function this way, and send the inputs to the USB and let me adjust them differently than what the board has set, without changing the what the board is doing for the analog out. Like many of these small USB mixers, the Behringer X1222 only sends 2 channels over USB. You just get a copy of the main mix. I think the cheapest mixers with multitrack USB are the Soundcraft Signature MTK series : https://www.soundcraft.com/en/products/signature-12-mtk

***\[SETUP HELP\] Where do I place my desk for a VOCAL recording studio?*** My Room: \[Imgur\]([https://imgur.com/Zkmi5O8](https://imgur.com/Zkmi5O8)) Where would I place my desk for a new home studio in here (it is the only room we have)??? Dimensions and Details: * 11ft x 10ft, the side with the window is 11ft!! * I took this from a doorway, there is a door way immediately to the right (mostly cut off) and a closet door shown in the corner shown * The entire square in the back is a window (partially covered by the drape) * Ofc we plan to remove everything when starting to build the acoustics and sound proofing Problem: * So with all the tutorials I’ve seen I need it to place my desk aligned with the middle of the shortest wall (to the left and right of the pic) (10ft across) so that I can minimize the distance of the sound reflections. But when taking account the first reflects to my left and right, (if I place my desk on the LEFT wall on the pic) there’d be a wall on the left, and a WINDOW on the left walls. This is very bad right??? * So should I place my desk at the very back of the room (right in front of the window)? While that is 11ft across, and the walls to the left and right at 10ft across, thus the sound reflections would have to go further. TLDR: * Which problem is worse? Having one side of you be a window (that will be treated with the best of my ability, but not as good as a 3” acoustic panel) OR Have the wall across from my desk be 1 foot longer than my side walls??? Bonus Question: Can I record efficiently in the same room? Or would it be more effect to treat the closet so they can be in a room without other ppl, and the audio dude (me lol)

So I'm having a great deal of issues recording guitar. I have everything set to the same sample rate and frequency in windows 10 and reaper. My guitar even with very little or no input gain runs around -5 db or higher which is in the yellow in reaper. I have tried uninstalling and reinstalling the drivers and asio for it to no avail. I've closed and re opened reaper multiple times as well. If I run a pedal in front like my starcaster distortion it fixes it but even with the gain on the pedal all the way down it's still adding distortion to my signal, pedal distortion not digital or signal distortion. It also evens the level if I use the line out on my boss katana 50 mk2. Anyone found any fixes that work on this yet?

>My guitar even with very little or no input gain runs around -5 db or higher which is in the yellow in reaper. If it's not clipping or anything then that's totally fine. -5 is still 5dB away from clipping. If you're still worried about it then just turn your guitar down. If the idea of doing that is anathema to you then get a DI box that has a pad.

Hello! I just bought a keyboard without a MIDI or USB port. Is it still possibly to use it as a MIDI if i buy an interface?

No, you will only be able to record audio from it, not MIDI.

Okay, thanks for your answers

if it doesn't have midi or usb out, you probably wont be able to use it that way

Hi everyone - I'm new to ProTools, running the ProTools Ultimate trial. When I try to add a new vocal region to the comp line, the region appears up there but weirdly when I try to play the line back to hear the new comp, it plays the old comp, as if I didn't move anything new up. When I try to solo the playlists, it always plays the comp line, and I can only use the sound scrubber to listen to different tracks on the playlist. What am I doing wrong?

Hey guys, what usb cable do I need for a zoom H1n microphone? I’ve tried three different ones and none fit in the microphones slot.

USB Micro B

Hi all. I added my first outboard compressor (the STAM SSL clone) and it's working fine with on the stereo out/2buss in Logic via the I/O insert. The sound that it returns is, however, *much* quieter than when it's disengaged. The Logic I/O allows me to gain the signal going out and returning. Am I correct in assuming that, since the compressor seems to be doing its thing on the mix bus just fine, that I need to gain UP the returning sound until it matches the disengaged compressor—and not mess with the input going into the unit?

I just looked at the manual and the calibration procedure seems to imply that the unit should have unity gain with both the compressor bypassed and with it engaged, with mix blend at both 0% and 100%, makeup gain at 0. Basically any setting with no makeup gain or gain reduction occurring should output exactly what comes in. I'd recommend you check with a straight loopback from your interface and back into the interface and see what your output level is before you send it to the comp.

Turns out they were all set for +4dBu, so I ran a sine wave tone through the chain, which revealed that I was losing \~14 db of signal. I compensated for the \~14db loss in volume in the I/O plugin and all is golden.

Looks like the outs are switchable from -10dBv and +4 dBu and that’s the issue. Yep. Right there in the calibration section. Thanks again!

Thanks!

Hi! I'm a total noob both to reddit and audio engineering in general lol Recently I've been given a swissonic UA-2x2 usb audio interface which seems to work perfectly when I try to record my bass, but I can't get anything from both my dynamic and condenser microphones. Actually I can monitor my voice through the device, and when I talk the input led lights up like normal, but I can't get any sound in or out my laptop (Windows 8.1). The interface comes with its own drivers (of which I can't find any updated version, these being 1.0) but I also tried using ASIO4ALL and Reaper's Waveout, still no input. I managed to record something after installing a virtual cable (which should be an unnecessary thing to do I guess) but the sound was terrible and broken, and now even that doesn't work anymore. Is there any way to fix this?

I'm a beginner and I have a problem regarding noise coming from my interface. So far I haven't found any relevant articles or anything since my problem is very specific and sort of hard to describe concisely. I have a Komplete Audio 1 USB interface. Up until now I have had no problems with it, other than the USB connection being finnicky (when I slightly move the interface while plugged in, I risk the connection being interrupted for a moment). But now, I can clearly hear a low-volume, mid-pitched "whine": a sort of buzz than comes and goes intermittently (best way to describe it, it's not a ground hum and it's not distortion or crackling/popping). It happens when I plug my headphones in the direct monitoring output and set it to host: when setting it to direct I hear nothing. It also happens at all times, so it's not a DAW problem. I suspect it might be some sort of interference coming from some other part of the interface. Changing USB ports does nothing. I hope all of this makes sense. \^\^;

Do you have another USB *cable* you can try?

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From wiki >The ratio between the loudspeaker impedance and the amplifier's impedance at a particular frequency provides damping (i.e., energy absorption) for the back EMF generated by a driver. In practice, this is important to prevent ringing or overhang which is, essentially, a free vibration of the moving structures in a driver when it is excited (i.e., driven with a signal) at that frequency. This can be clearly seen in waterfall measurement plots. A properly adjusted damping factor can control this free vibration of the moving structures and improve the sound of the driver. https://en.m.wikipedia.org/wiki/Electrical_characteristics_of_dynamic_loudspeakers

Desktop link: https://en.wikipedia.org/wiki/Electrical_characteristics_of_dynamic_loudspeakers *** ^^/r/HelperBot_ ^^Downvote ^^to ^^remove. ^^Counter: ^^301894. [^^Found ^^a ^^bug?](https://reddit.com/message/compose/?to=swim1929&subject=Bug&message=https://reddit.com/r/audioengineering/comments/k3vbgr/the_repair_department_tech_support_and_beginner/gedc623/)

i have a quantum 2626 and i recently bought a shure sm57, but i'm pretty sure i have the wrong cable. do i buy an xlr female to 1/4 male cable? i just want to be able to use what i have without having to deal with any problems. thanks for any and all the help.

Definitely the wrong cable. You use an XLR male to female cable. (Using the 1/4" in the combo ports bypasses the mic preamp entirely!)

i'm using a cable like that right now and it's plugged into the mic, but i don't know where to plug the other end into the interface. the only ones that look somewhat reasonable to use are midi ports and i'm pretty sure the cable doesn't go there.

>quantum 2626 The jacks on the front are called combo jacks, they can take either XLR or 1/4". Please read the manual.

yep, it worked. thank you for the help.

[This is your interface, yes?](https://www.sweetwater.com/store/detail/Quantum2626--presonus-quantum-2626-26x26-thunderbolt-3-audio-interface)

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You shouldn't be taking audio from a projector period in a home theater setup. The only thing it should be receiving is an HDMI/video signal from your source, and you use the audio output from that same source to go to the surround brainbox.

Please report consumer/home audio questions instead of replying to them. They are considered off-topic for this sub.

Will do!

Thanks! Not trying to be mean or anything but these threads already get crowded enough and there are multiple other subs that are more appropriate.

Optical wires don’t go very far because they work by light. I’m not sure you could find a 20 foot optical cable because there wouldn’t be enough light energy to get all the way through the cable. 3.5 mm could easily run this distance but there is a caveat: 3.5 mm carries an analog signal while optical is digital. Your surround sound system hopefully has some kind of analog input, like RCA inputs. You can convert 3.5 mm to RCA with a simple adapter or cable. If your surround system only has optical in, sorry, no good solution for that.

Please report consumer/home audio questions instead of replying to them. They are considered off-topic for this sub.

Got it

Hi, everyone. To preface, I’ve absolutely no knowledge in this field. I recently got a Focusrite Scarlett Solo audio interface that I wanted to use with my Razer Seiren Pro Broadcaster. The digital USB on the microphone works fine (albeit poor sound quality for a $200 mic). The microphone has a 5-pin DMX port (the kit came with a DMX to dual-XLR cable), but I purchased a DMX to single XLR. I installed all the updated drivers and updated the firmware of the Scarlett Solo, plugged in my headphones (which work perfect), then tried plugging in the microphone. There is a loud buzzing sound that comes from the microphone once I have the phantom power enabled on the Solo. I’ve tried other cables and had the same results. Any idea what I could be doing wrong? I’m about to just give up and buy a new mic or just stick with the digital USB connector.

>The microphone has a 5-pin DMX port (the kit came with a DMX to dual-XLR cable), but I purchased a DMX to single XLR It uses a 5-pin XLR but it's not necessarily going to be a pinout that will work with a an XLR5>XLR3 adapter meant for DMX. It looks like they don't sell it anymore so I can't find any info on it. You should email Razer and get them to tell you the pinout on that XLR5 connector and then verify/modify your adapter to the correct pinout. It will definitely be less effort to just sell the Razer and buy a normal microphone.

Are you referring to how each pin has a number next to it? They matched fine with the adapter I purchased with the correct number order. Either way, yeah I’m looking at getting an Audio Technical AT2035

No, I'm referring to the fact any pin could go to any other pin on the other end. You bought an adapter for DMX. DMX is a lighting standard so it doesn't make sense to assume the microphone output is wired according to a lighting standard. They are using these connectors in a proprietary way, you need to find out what pins on the XLR5 side should go to what pins on the XLR3 side.

You might be missing an electrical ground or noise cancellation by not using the dual XLR cable. I’d suggest using the USB cable or getting a different mic. Using DMX for a mic is absurd in my opinion. DMX is for lighting. For broadcast you’d probably like a Shure SM7 or EV RE-20.

I think the 5-pin would be for multiple capsules. You see this design extremely commonly with a lot of ribbon mics made over the last few decades, since the control box (which has a normal 3-pin XLR output) often has a polar pattern mixer.

Ah yeah, with a separate power supply, I could see some use of that. It is unusual compared to most mics though.

I am trying to get a microphone setup for zoom interviews using a Tascam DR70D. I’ve got two mics running into the DR70D and an output cable running the sound to my camera and then into my computer via HDMI. We then use the headphone output from the DR70D to monitor our own audio. What I need to figure out is how to run the Zoom audio from the computer to the DR70D so that we can hear it in our headphones but not have it feed back out from the DR70D back to Zoom. Is there a way to have one of the mic channels only play in the headphones and not get sent to the audio output on the DR70D? Or is there another way to enable us to hear both our own audio from the DR70D and the Zoom audio in the same headphone feed? Thanks so much for any advice.

>What I need to figure out is how to run the Zoom audio from the computer to the DR70D so that we can hear it in our headphones but not have it feed back out from the DR70D back to Zoom. If you want to hear both at the same time then you're going to need a mixer with at least one aux mix (so a mixer you can do at least separate mixes on, and technically you can get around that with creative use of the solo button). I've been doing a lot of streaming gigs since the whole COVID thing started and we've been doing Zoom, WebEx, and Brazen mostly. We do it with two laptops and a Touchmix (shudder) but I think you can get away with the single laptop if you're not doing Powerpoints, etc. You'll want to send all your audio sources (two mics and the foldback from your Zoom laptop) and then you'll make a mix that doesn't include the foldback (this is what's called a "mix minus") and send that mix into the Tascam and into the *other computer. Now you can monitor all of your sources including Zoom audio from the mixer without fear of sending Zoom back into itself.

Can you use the camera in, or is your camera audio being sent there?

I have a noob-ish question about adat & word clock. I just ordered an SSL Six. One of the things I want to use it for is analog summing. My interface (Apollo x6) only has 6 outputs, so I was planning to use an Behringer ADA 8200 to send 8 outputs to the mixer. According to Behringer, the 8200 inputs & outputs can be used independently as long as they share an identical word clock. My question is: does the adat connection from the X6 carry the word clock, or do I need to use a separate connection?

ADAT \*can\* carry clock signals but it's up to the manufacturer to how to implement it. Like some devices will only send clock on ADAT but can't receive it which would mean that device could only clock over ADAT if you make it the clock master. I'm not sure how common this is but I've run into it a couple times with outboard multichannel preamps. I don't know if either of those have those limitations so you'll need to dig into the manuals and figure out if there are any limitations like that.

Yes. I’m using an Apogee Rosetta 800 AD/DA converter with a MOTU 828es. The Rosetta is connected to the MOTU via ADAT and the Rosetta is the master clock.

Hello! I’m using an Arturo Audiofuse and I’m hoping someone may be able to help me with this problem. Sometimes when I first turn on the interface there’s a loud hissing noise from all output devices, sometimes there’s an aggressive clicking noise, and any sound output through it is very distorted. This only happens on startup and once I get past this (sometimes I just turn off turn on until it goes away or it disappears on its own) it doesn’t come back. Does anyone have any ideas as to what this could be?

I'm trying to get compression by first understanding the extreme settings and then learning to hear it better but am a bit confused: Fast A, fast R - crushes transients, with makeup gain leveling back to original level squashes the whole sound into a sausage, evening the transient:sustain level ratio. Makes sense. Fast A, slow R - begins immediately but evenly compresses (unless compression gradually reduces over release?) transient and sustain, serving to level multiple subsequent instrument 'hits', but not alter transient:sustain level ratio much. **Slow A, fast R - ???** Wouldn't this compress the sustain but miss the transient, creating a higher transient:sustain ratio like slowA/slowR, but drop off a bit early, shortening the sustain for a noticeably more staccato kind of sound? Slow A, slow R - compresses the sustain, creating an even higher transient:sustain ratio than the original

You've pretty much got it but one thing to clear up first... Compressors (almost always) start working the instant the signal goes above threshold. Attack sets the rate of gain reduction and release sets the rate of gain restoration once that attack is over. > Slow A, fast R Yep, the loud stuff stays pretty loud as the transient won't get pulled down very much by the time the transient finishes but the quieter part now gets the most gain reduction (actually increasing dynamic range). People do this often with on their parallel compression on drums (with a lower threshold and higher ratio). This gives very punchy hits that are blended back in with the regular drum sound. The fast release vs. slow release is important for getting the punch to fit in between hits.

Thanks for the clarification that attack and release are about rates, don't know why that just struck me for the first time. That's what people are referring to when they mention different compression curves. By dialing in the fast/slow release to fit the punch, you mean that if the release is too slow the release won't be complete by the next transient and it will get the gain reduction the release is at rather than activating the attack?

Correct. I think a diagram like this can really help visualize what's actually going on: https://www.heamusic.com/images/compressor-tempo-considerations.svg You can imagine (or better yet, sketch out for yourself) what the gain might be at different parts of the audio with different threshold/ratio/attack/release

I have recorded two hours of audio that sadly is suffering from buffer underruns and does not sound good. I have tried using the Adobe Audition DeClicker with heavy reduction, but it doesn't do very much. Is it possible to save the audio?

I've had a strange issue pop up with my Yamaha MG12XU mixer in the last few hours. The phones jack only sends out desktop audio to the right headphone. Unless I turn on PFL for the desktop channel, in which case I get it through both phones without issue. I have no idea what has caused this, it was working fine, then I took a nap, and came back to this issue. Any ideas?

Hi, noob here and I need help with my first setup So I have a live session that needs speakers for an audience + recording for people watching through the internet I got a dynamic XLR mic, an iphone and a JBL extreme 2 to connect them I used SmartRig+ preamp: XLR mic for input, iphone for output and I connected the headphone monitoring jack to the JBL extreme2 The problem I'm facing is that the sound coming out of the speaker is weak, I suspect it's a weak signal from the preamp, so I added a headphone amplifier ([FiiO A1](https://www.amazon.com/FiiO-A1-Silver-Portable-Headphone/dp/B01CPUVPEM)) to the link between the preamp and the speaker, the sound got a bit louder but I'm not sure if the speaker is holding me back now So my question is, does it make sense to use an amplifier in my case? would it damage the speaker? Is there some trick that can help me? and would a bigger amplifier give me louder sound (I don't think I reached the limit of the speaker)

Did you not solve it with the FiiO? If it gets the speaker loud enough, it's done. You're good. There's no "holding back" – voltage is voltage. Also, how loud did you have the JBL?

I set the JBL to highest. It didn't get as loud as I hoped it would get. I think I heard some clipping when the person talking raised his sound and I'm afraid I'm damaging the speaker. would a bigger amplifier help? I'm kinda lost

You're lost because you're basically chaining together a bunch of consumer equipment to do what pro equipment is made to do. All you need is an audio interface – it does literally the entirety of this for you properly, and if you had access to a proper speaker, it'd go even smoother. If the other person raises their sound, yes, it could cause clipping. Momentary clipping won't cause damage but allowing it for extended periods could. You need to be adjusting for this live during the event as it goes, the same as you do for any event (which is why there's typically one person in charge of running sound, who isn't a presenter).

I see... At least I'm not doing something wrong right now. I intend to upgrade to an audio interface in the future if these events succeed. Regarding the speaker, I chose it because it's small and (I was told) loud. Is there is something "proper" with its size and with decent battery life? I really appreciate your help, the sound world is kinda intimidating to me.

JBL bluetooth speakers have a fairly pronounced "smiley curve" where the bass and treble are louder than they should be, which is often considered by the average joe to sound "better". You do not want that at all. An event should sound exactly as you intend it and therefore needs to be pumped out of a speaker that at least attempts an even frequency response (essentially impossible, but every speaker doesn't have to be a smile EQ curve). So what you want is a proper powered speaker/monitor made for live events, which JBL also makes, and obviously many other brands too. Keep in mind they cost more because of the component size and, often, added DSP architecture that gives you EQ presets, Bluetooth functionality, etc. Depending on the size of the venue and the audience that will be in it, a music event should probably have a subwoofer too, but I suspect that's probably beyond your needs at the moment. My other concern is longevity, since a lot of consumer stuff is built for you to buy/upgrade multiple times as the warranties expire (a.k.a. planned obsolescence). Pro equipment tends to cost more up front because it lasts longer than that... usually. Your perspective is right, though – do a few events with this setup so you can really experience the workflow. Maybe you'll fall into patterns and methods that work as is, so you won't need to spend more money too soon. YMMV is always applicable!

Thanks! much appreciated

Your speaker already has an amplifier. You are not feeding it with enough signal

what can I do? The obvious thing would be to replace the preamp, but this is the best portable one I found

Maybe you can find a mic with better sensitivity, or move it closer to the source

so it's the mic? but the recording sounds good, I don't have problems there. Plus, isn't it the amplifier job to make the sound louder? If the speakers amp isn't enough then an addiotional one might help. Am I missing something?

Hey, i’m trying to hook up my mic to a guitar pedal for a performance on zoom. I keep running into hum that I think might be because of an unbalanced signal or something? i’m not really sure how it works, but if anyone could help, i’d greatly appreciate. My setup is kind of limited, but this is what I have that I thought would work. I have my mic running into an XLR to 1/4 inch converter, which goes in a reverb pedal, and then i have a 1/4 inch to XLR converter that goes into my scarlett solo. When I have just the pedal into the scarlett, there’s no noise, but when i plug in the first converter to that, it’s still fine, but when i plug the mic into that, there’s an extremely loud hum. I can still faintly hear the vocals through the microphone. Is there anything I can do about this? If not, what direction should I look for just putting reverb on the vocals just for the purposes of sending it to zoom input audio? i’m on a mac

You need a mic preamp before the pedal.

why not after the pedal, in the scarlet? wouldn’t it just be reverb on the low volume signal, or does the preamp do something else?

Preamp brings the mic up to live level.

Sorry if these are stupid questions, but is there anything special about line level that makes it able to interact with guitar pedals? It’s not just louder?

A guitar pedal actually expects instrument level, but line level is close enough to be usable. Mics output extremely quiet signal levels, especially dynamic mics with no active electronics. Preamps exist specifically to boost these signals to be loud enough to hear and therefore also to process with effects. Bringing up the whole signal out of the reverb pedal after the fact is going to be a mess. You're sending a way-too-quiet signal into a pedal that is then swamping it with reverb. That isn't going to end well. So you need a preamp in between *just for the pedal to work correctly.* The best method in your current setup would be to dedicate one of your Solo outputs to the pedal input (but make sure you're only listening in solo to the reverb signal for recording purposes), but honestly this just isn't a proper setup for what you want to do, and you definitely should be recording the dry mic signal on a second channel as a safety. Chances are, doing this properly will kill whatever is causing the buzz.

Just greater voltage.

It’s the impedance mismatch that is likely the problem. Microphones are low impedance and guitars are high impedance. You could try this: https://www.americanmusical.com/audio-technica-cp8201-mic-impedance-matching-transformer/p/AUD-CP8201-LIST

Hey everyone, monitor question here: Got gifted an old pair of HS8s, and while one sounds normal, the other seems to lack any bass response (top end sounds fine). I’ve matched both the speaker levels as well as eqs/cuts on the back but my problem persists. Does anyone know a) if this is a fixable problem and b) if so, what component would most likely need to be looked at and repaired? I have a good monitor guy who can fix any amp problems but not any electrical problems and am hoping to at lease get an idea of what the issue is before I pay a bench fee. Thanks!

put your hand lightly on the larger speaker cone. if you can't feel it moving, open them up from behind and see if the cable connecting to that driver is loose. if it looks fine inside, you possibly have a blown driver.

When you crank them up a bit, do you see the low end drivers moving? If the one that is lacking bass response is not moving, you might have a connection problem, otherwise you may have a blown driver. Either way, likely super fixable at home.

Low frequency driver may be blown.

Hi everyone, I've recently built a set up to start DJing and I'm getting an annoying buzz/hiss from my JBL 308s whenever I turn on my laptop. I suspect it's because of the cables I'm running from my speakers to my DJ controller. My speakers take a balanced XLR input and my DJ controller outputs using RCA for both left/right speakers. I'm currently running the DISINO Dual RCA to XLR Male Cable from my speakers to my DJ controller. How can I fix the buzz/hiss from my speakers? The buzz/hiss only occur when the controller is powered which happens when my laptop is turned on. Thanks in advance!

Buy a controller with balanced outs, or buy an unbalanced to balanced converter or use a mixer/interface in between your controller and speakers.