Feedback noise in mic signal I've been having an issue for a while now with this feedback noise coming from my mic. Makes it impossible to record anything. I have a pretty decent audio setup (Rode NT1 condenser mic, Mogami XLR cable, Focusrite Scarlett Solo interface). I have no idea what the problem is. I've been trying to troubleshoot this issue for months to no avail. If anyone can help that would be greatly appreciated. Here is an audio clip of the feedback noise recorded into the mic signal (increase the volume on your speakers or headphones if you can't hear it, but it's definitely there peaking at around -40 dB): [https://audio.com/pj-svndsn/audio/feedback-noise](https://audio.com/pj-svndsn/audio/feedback-noise)

Thats not feedback thats 60Hz breakthrough (from the power/power lines), possibly a ground loop - theres a troubleshooting guide above - its a whole thing.

So then how do I figure out the source of the noise? I’m flummoxed

[https://www.reddit.com/r/audioengineering/wiki/troubleshooting/#wiki\_computer\_noise.2Fground\_loop](https://www.reddit.com/r/audioengineering/wiki/troubleshooting/#wiki_computer_noise.2Fground_loop)

I tried reading through that but I’m struggling. I’m usually pretty good with tech stuff but this has me pretty stumped

I’ve got a Black Lion Audio Auteur (MkI version) that has a channel with an unresponsive gain knob. I have thoroughly performed routine testing and troubleshooting, but haven’t been able to diagnose or resolve it. I’ll be reaching out to Black Lion for support, but also wanted to see if anyone here had either suggestions on probable cause, or a recommendation for a sub with a pro audio repair/maintenance focus where I might be able to get help diving into the chassis and repairing myself. I’m handy with a soldering iron, have built some effects pedals and do all my own electric guitar and amp service, have just never gotten into preamps like this one. The issue: Gain knob makes no difference in the audio when manipulated. Sound passes through, 48v is working, gain knob makes absolutely no change in loudness, distortion, etc. No sound passes when the unit is powered off, as expected. Maybe this is an op amp issue? Appreciate any insights!

How do I set my RME interface to stop sending browser audio to Zoom? I'm trying to use Zoom to broadcast Facebook Live and share my laptop browser screen. I don't want to share the browser audio, just the browser video display. Instead, I want to share the audio of my piano --> https://basilwhite.com/zoommic.png I'm sending the piano into my RME Babyface Pro at Input 3. I tell Zoom to use Input 3 as its microphone, and it does. However, the mixer software for the RME takes input to Input 3 from both my piano **AND** the browser --> https://basilwhite.com/mixersettings.png I've set Zoom's "share sound" option to off. How do I get the RME mixer software to stop sending the audio from the browser to Input 3?

Hi all! I'm looking to purchase a very basic setup for live podcasting at a venue. From what I understand, the simplest set up is: Microphone into Powered Mixer into Passive Speakers. I'm just wondering does anyone know if the speakers and mixer I'm thinking of purchasing would be suitable. Mixer - Allen and Heath Zed 10 (not sure if this is powered?) Speakers - Behringer PK108A Thanks a million.

Hey folks I'm trying to connect a Focusrite Red 8pre and an X32 through an existing Ethernet network. Everything tests ok and works when using Dante Virtual Soundcard. The issue here is that we then can't hear playback in the studio, and it's not utilizing the Dante functionality on the Focusrite interface. Is it possible to connect the Focusrite and the X32 essentially by one Ethernet cable and have them communicate without using the Virtual Soundcard? I'm not having much luck. The ultimate goal is to be able to freely send channels back and forth between two rooms; and whilst this is possible through Virtual Sound Card, it's not utilizing the interface's Dante capabilities and leaving us a bit short. Thanks in advance.

just upgraded from focusrite 4i4 to apollo twin quad setup for engineering in pro tools & producing in fl studio. ive got it set up for fl studio finally and made my first beat with this setup. off the rip i noticed the way the apollo's master output doesnt redline for me. ive already double checked my fl studio settings. everything looks normal and ive been producing with this laptop for years which leads me to believe it has something to do with the apollo. it sounds like there's some kind soft clipping stage before the output or something else preventing me from clipping the apollo and hearing the true sound. whats crazy is the beats i exported sound and look overly distorted but when im making them they sound fine. this is an issue for me as part of my sound is me having nothing on my master letting fl studio clip & this translates to other speakers better for me. anybody know how i can fix this or should i just continue to use my 4i4 just for making beats?

24bit interfaces SHOULD have headroom above 0dB on the DAW, Have you considered putting a clipper and limiter on your master, so it does translate to all systems properly, not just bad ones?

Hey there! I have a hybrid recording setup consisting of various tape machines and an interface, leading into my beloved ACE recording console. I have this weird noise (comes and goes), even if all possible sorts of gear and devices on the same power phase are switched off. I even powered the power supply of my desk with an UPS, no luck there. Funny thing, the exact same behaviour also occured with my former Soundcraft console. Initially the noise was not present but since a few weeks it drives me nuts… Anybody got an idea? Thank you! Edit: [Link to YouTube video for demonstration purposes added](https://youtube.com/shorts/U7wpvlq25pQ?si=zkBoUNzm4W7K2yla)

What kind of noise? Got a welding shop nearby or similar?

Welding shop is a good guess but fortunately - no welding shop nearby, also no heavy machinery. I uploaded a youtube short. Notice: the noise is not coming from the delay unit which i switch off while the video runs. It also occurs when every input possible is switched off. [YouTube demo of the noise](https://youtube.com/shorts/U7wpvlq25pQ?si=zkBoUNzm4W7K2yla)

Hello, I'm having major audio lag in FL studio 21 while I mix and master. My CPU goes to 100% and the stuttering is unreal. It is nearly impossible to work with. I work with SSL, Brainworx, Ozone 5, Plugin Alliance, TR5, RX 10 for mixing and mastering. Anyone have ideas on what might be the issue? Is it FL studio? The plugins? or is it the CPU? Do i Need DSP? My CPU is AMD Ryzen 9 7900 12-core processor Audio setup: I use scarlett solo for audio interface. Audio settings in [https://drive.google.com/file/d/188fYhsW\_JqE6xqTmGCVgSR3nZmUI8TK4/view?usp=sharing](https://drive.google.com/file/d/188fYhsW_JqE6xqTmGCVgSR3nZmUI8TK4/view?usp=sharing)

Use the Focusrite driver. NEVER EVER use a generic universal driver where there is any other option, ALWAYS use hardware specific ASIO drivers or you are defeating the point of ASIO. [https://www.makeuseof.com/what-is-an-asio-driver/](https://www.makeuseof.com/what-is-an-asio-driver/) Using that you should be able to get your buffer to 256 or lower which should fix most of your issues Also make sure EXPO/XMP is enabled in your BIOS, running at default RAM speed cripples Ryzens, esp in audio work. Even better, tune your RAM properly.

Hey man, I appreciate your response. I talked to my brother who is in IT and he told me to update BIOS and all drivers... I had to update a bunch of things, I also did some performance tweaks in BIOS. I then opened up my latest project which has been mixed and mastered. I disabled all supersampling (I had it on for rendering, not while working of course) I bypassed everything RX related, disabled everything to do with supersampling and bypassed ozone 5. I reinstalled focusrite drivers and installed new ones, I changed buffer size to 192, sample rate to 48000hz in focusrite. I tried to use focusrite ASIO, but it ran poorly, I tried every buffer size possibly. Choppy audio, so I changed back to FL Studio ASIO. Buffer length for FL Studio ASIO is 384. I disabled some things in the audio settings. Trade off is a little underruns, but it plays back smooth. No stuttering at all. I will provide a screenshot of my settings in hope if anyone else stumbles upon this and found no other answers, I hope they can copy what I did. screenshot of my audio settings in FL Studio: [https://drive.google.com/drive/folders/1JvvWZ31sF5f77qBmye5uK7Nux1sCpbsl?usp=sharing](https://drive.google.com/drive/folders/1JvvWZ31sF5f77qBmye5uK7Nux1sCpbsl?usp=sharing)

Something is very wrong if the focusrite ASIO driver runs poorly - it should always be the best option, by far. As i said try 256 or higher buffer at 48k in that

Think the audio interface is defect? I thought too it was odd it was worse. Let me clarify: I tried 256 and higher, I tried every buffer size at 48k, no dice. Only FL Studio ASIO works smooth

If its working via Windows Sound API (which is what FLStudio ASIO connects to) and not via the factory native ASIO driver for it, it sounds like an issue with your machine config.

Honestly at this point I don't care as long as I can work

So I decided to start a home studio, I’m brand new to this. And I bought a mackie 24.4 and i don’t know how to connect it to my Mac mini to use it with protools. The mixer works. I’ve plugged mics and the channels works. And I can hear it on the monitors. But I can’t figure out how to connect it to the Mac mini so I can use it on protools.

Are you able to return it and get an Audio Interface for recording rather than a live desk for rehersal rooms?

I bought it off from someone on offer up. I thought that would work for recording.

Which specific Mackie 24 channel 4 Buss mixer is it? there have been a few named 24.4, but afaik theyre all intended as live mixers. Does it have USB?

It’s the mackie sr24.4 VLZ it has no USB connection. In the back it only has 1/4 inch connections and XLR connection.

The problem is you need exactly the same thing to record from it to the computer as you would to plug the mics in and record them direct - an audio interface (ie an Audient Evo16 for $460) - and with a modern interface you will probably get better quality from its internal preamps than from the desk. Another $200 will let you add another 8 channels over ADAT with ie a Behringer ADA8200, up to 24 i/o

The reason I bought that was because I was hoping to use the knobs to adjust volumes and all that with physical knobs. Rather than the digital ones on the screen. So then does that mean I pretty much won’t be able to use the mackie mixer?

Theres certainly ways you could make it work, but it would be a LOT less complicated to use a mix controller (i have a Mackie MCU and a Maschine JAM, Novation Launch Control and Akai APC are very popular, but im in Ableton) to control ProTools, if the intent is to record and monitor what you recorded. Avid even make specific controllers for the purpose. And interface direct is gonna sound better - unless the point is chasing 'old mackie sound' for lo-fi (they have a distinct, almost iconic distortion if you overdrive them, that a lot of people love) I'm sorry - i really dont mean to seem brutal. Theyre cool, but they were never intended for PC or even multi-track recording. You would probably be sending all the channel inserts to the audio interface and then back out to the desk, Or i guess you could record the busses but that would b unusual these days, and modern interfaces have its functions built in.

I got a Motu M2 audio interface and was wondering which mic would be the best for keyboard sound recording and for my voice for discord maybe a little bit of content making? I was thinking about rote nt1 (will definitely pick that pink xlr cable up. that thing looks fire)

Looking to plug in my Big Muff Opamp pedal to my Digitech Valve FX rack unit or otherwise use them together. Both except instrument level. Will this damage either unit at reasonable levels? Checked the manuals for specs already. Thx

Do you own Cymatic LR-16 drivers? Hello! Does anyone own (or knows where to get) the widows driver for the LR-16? (the 16 inputs recorder). The website drivers are down I can't find any useful post on their legacy forum. Thanks in advance!

I need a living room advice. Will be moving next year to new apartment. It's in empty state so I'll be able to put as many cables in the walls as I want, but I have a tricky space situation. Currently I'm having yamaha rn-802d, B&W ASW610 and pair of B&W 683 s2. Currently it's placed in rectangle room 6x6 meters and feels on point. My plan is to place my audio-video in the 200cm space as on the photo, to avoid placing sitting area against the wall. I don't think my current setup will fit in such a small space. What would be your advice? I can move on to bookshlef speakers, or soundbar, but I don't want to resign from great audio experience. I'm also thinking if I have possibility to go dolby atmos (I can place cables anywhere), but is it really feesable? Waiting for your ideas on how to fill the space. Additional info: Usage: 50% music/radio 30% gaming 20% movies Location: EU Budget: up to \~3k EUR Living room plans: [https://imgur.com/a/NohuD0N](https://imgur.com/a/NohuD0N)

probably need more room treatment, and microphone cables, for a recording studio

I'm trying to wire an aircraft microphone into my home cockpit It's a Holmco mh-50, I have the wiring diagram but I'm a little confused what the best way to wire it is. It has a 4 pin xlr type connector. From my understanding, it has a push button which is both hard muting the microphone and also stopping the audio from transmitting to the connector. Then it has to other 2 other pins on the connector, I can see it's a + and - which i'm assuming means its unbalanced. It also has the screen which is wired into the - (white cable, at the pcb end) which I assume is grounding the screen. I can easily wire the switch into an IO board I use, but I'm not sure what the best option for connecting the audio to my PC would be? Any advice would be awesome Link to wiring diagram on 2nd page: [https://www.yumpu.com/en/document/read/29027199/data-sheet-pdf-holmco-holmberg-elektroakustik](https://www.yumpu.com/en/document/read/29027199/data-sheet-pdf-holmco-holmberg-elektroakustik)

Its an electret condenser, It needs 1.5-10V power. Very few interfaces will power it, But the mic input on your computer can though not optimally (the cheap presonus Revelator io44 is one of the few, but its a terrible interface otherwise). Beltpack transmitters will also power it. Anything designed to run a computer headset should power it fine. Making an electret driver is really simple (its an example cct in many op-amp datasheets) too. The switch is completely separate. (edit - forgot about these plug-in power adapters form Rode [https://www.amazon.com.au/R%C3%98DE-VXLR-3-5mm-TRS-Adaptor/dp/B071LNDKBJ/](https://www.amazon.com.au/R%C3%98DE-VXLR-3-5mm-TRS-Adaptor/dp/B071LNDKBJ/) that will power it form phantom)

What if I bought this, cut the mic from that out and wired my mic into it? https://www.amazon.com.au/BOYA-Electret-Condenser-Microphone-Smartphones/dp/B00MP566OM

might work i guess, seems like the hard way and i would suspect the cable is not the quality kind that likes to be patched easily

I guess the issue here would be finding a 4 pin xlr to 3.5. I might just gut it out and replace it with a gutted pc mic

Dude its REALLY simple to drive, way simpler than all that, you can just buy an electret driver, you seemed like you would be fine making one [https://www.amazon.com.au/WELWIK-Microphone-Amplifier-Adjustable-Converters/dp/B0CZ8MG5ZM/](https://www.amazon.com.au/WELWIK-Microphone-Amplifier-Adjustable-Converters/dp/B0CZ8MG5ZM/) just cut off the end or buy a plug, wire the mic + and - to the board, wire the tip of the computer mic plug to out, ring to VCC and ground and the shield wire of the mic to ground.

I’ll give this a go! Mics are not my strong point hahaha

Hi everyone, I'm facing a persistent issue with buzz/hum/white noise in my recording setup, and I'm hoping to get some advice on how to resolve it. Here’s a [link](https://youtu.be/j39XqbE33mM?feature=shared) to a sample recording. Here are the details of my setup and the troubleshooting steps I've taken so far: Setup: - Microphone: Shure SM7B (supposed to have low noise floor but I’m out of ideas) - Audio Interface: ssl2 and scarlet solo 4th gen - Cables: Mogami XLR cables - Headphones - Computers: Tested with 2 different computers - Recording Environment: Noise-treated studio with -75 dB noise floor, no echoes or reverbs - Power: All equipment connected to the same power outlet surge protector Some things I’ve tried 1. Recorded with room light on and off. 2. Recorded with keyboard mouse Unplugged and plugged. 3. Switched monitors. Recorded with no power to monitors 4. Upgraded to high-quality Mogami XLR cables. 5. Upgraded the audio interface. 6. Tested with different computers. 7. Plugged and unplugged headphones. 8. Ensured all power connections are from the same outlet. Despite all these efforts, the noise persists. Here’s what I’ve observed: - The noise remains constant regardless of whether the room light is on or off, or whether the keyboard and mouse are plugged in or not. - Switching monitors and computers does not affect the noise. - Upgrading the XLR cables and audio interface didn’t resolve the issue. - The noise is present even when the headphones are unplugged. - It is a constant buzz/hum/white noise that is faint but noticeable. I’m using audacity, no post processing in the sample audio besides normaling rms to -22 for acx standard. Sorry for some plosives in the recording didn’t use a pop filter just trying to get a quick sample Im pretty lost at this point and welcome to trying any suggestions or insights on what might be causing this persistent noise and how I can eliminate it. Any advice on additional troubleshooting steps or potential solutions would be greatly appreciated! Thanks in advance for your help

# Everything is 180° Out Of Phase in Studio Strange issue. In my studio, every time I go to record, I do a soundcheck on drums. Typical mic set ups... I check the waveform in Pro Tools and almost every track is out of phase, except the bottom snare which is in phase. It's typical for the bottom snare to be out, but for it to be the only one that's in, that's strange. It can't be the case that ALL of my cables are wired backwards. I am running a Scarlett OctoPre, OctoPre and Behringer Ultragain. All through ADAT into an RME RayDAT. It's not an end-of-the-world issue. but merely a tedious one that requires an extra step in post. Does anyone know why this might be?

# Clock issues in studio with ADATs I am not too educated on clock syncing etc. I am running an RME RayDAT which is connected to a Behringer Ultragain, Scarlett OctoPre and an OctoPre Dynamic. I believe the Ultragain is the master. I have no issues with this preamp (apart from some inputs not working, which I think is simply because it's a bit shit). The OctoPre Dynamic works fine too. But the standard OctoPre has clock sync clicks, which makes it basically unusable. I've messed with the settings in the RME Hammerfall DSP and managed to move the clicks to a different preamp, but not managed to get rid of them at all. Clicks also used to get sent out of the OctoPre Dynamic's outputs, but I managed to fix this in the settings. Can anyone suggest how to fix this please?

I use windows media player but whenever I'm opening/closing through multiple files to choose the right one every now and then WMP freezes and I have to wait like 30s before I get an error message so I can open up files again, is there any other software recommended for this purpose? Sorry if my question sounds dumb

Not something relevant to this sub - try r/pcmasterrace - but i don't think anyone uses WMP, mostly VLC player. Also it sounds like youre having a disk error, you dont say what - but i would deal with that

Hi everybody! I'm currently a Macbook pro user and I need to update my laptop. I record orchestras and travel a lot, so I need a laptop. I also do a lot of podcast post production and I'm planning to edit video as well... I also want to start working with VST Instruments again (Kontakt, Ominsphere) so I need a lot of power. I don't want to use apple anymore, but I really don't know windows and other laptops in the market. I would like to expand RAM and storage memory manually.  My budget is around 1,000 dollars, but I can get a second hand and expand my budget a bit. What are your best recommendations??

Absolute novice here investing in my first home hifi setup. I'm planning to buy genelec 8331a for home music listening. I want to connect my laptop and mobile to the monitors via the digital xlr aes/ebu input and daisy chain them over digital xlr. It supports upto 24bit / 192khz inputs. What recommendations do you have for (I'm an absolute novice here): 1. 24bit+ depth digital volume control. Does it sit before or after the interface? Are there limitations on the volume control sitting in the usb / pcm realm vs aes/ebu xlr realm? 2. Single channel usb to aes/ebu interface, preferably with volume control. Something which does the conversions well but has no extra frills 3. Connecting mobile for say tidal over wifi. Would I require a separate component? 4. If possible, I'd rather change this setup to front the monitors with a network sound card with aes/ebu support. Any recommendations? And do these support wifi connectivity or require an rj45 input? Where would volume control fit in this setup? Would it require an aes/ebu xlr realm volume control?

Why? Studio monitors are not made for relaxing listening Also, AES3 is the pro version of SPDIF, get an adapter [https://support.genelec.com/hc/en-us/articles/360019582539-Can-I-connect-my-digital-SAM-monitors-to-S-PDIF-directly](https://support.genelec.com/hc/en-us/articles/360019582539-Can-I-connect-my-digital-SAM-monitors-to-S-PDIF-directly)

Because I liked it. Is there a non SPDIF option which goes laptop -> usb -> interface -> aes over xlr -> volume control -> aes over xlr -> monitor or laptop -> wifi / rj45 -> soundcard -> aes over xlr -> volume control -> aes over xlr -> monitor?

How to set them up digitally is in that article. Run SPDIF/AES3 (same thing) to one, then out to the other. Set one to A and one to B. Run network to both and control volume with the app at the speaker. Done. There is ZERO audio quality advantage to an audio interface in this config, just a latency advantage. The DAC is in the speakers. But you will not have a pleasant listening experience because they are made for the opposite. For the same money you could be buying a REALLY nice used home theatre system with full-range tower speakers, several times the power, seriously nice multichannel decoders, and at least 5.1.

Thanks for the advice!

So I just bought my first electric guitar, and with it I bought the behringer UM2 usb audio interface which I connected to my PC. I used the Reaper, and when playing a note I just get crackling/whitenoise. It is not the guitar, because I tried connecting a mic to the interface, and the same thing happens, just when I speak. I have installed ASIO drivers and I tried changing USB ports on my pc. Any help would be appreciated!

Do you have it plugged in inst2? is the clip light flashing?

Sorry if this is dumb, but I have zero idea about audio. I have 6 ceiling speakers, not JBL, I think they are Power Dynamics (I think it's [these](https://djcity.com.au/product/power-dynamics-ncsp8-low-profile-ceiling-speaker-100v-8-inch-white/), NCSP8) they aren't very high quality speakers. The speakers are mounted on the ceiling and wires have been run, but they aren't connected to anything. What do I need to get these speakers working via bluetooth (even chromecast/airplay) I think what I need is a mixer or A/V receiver, but I'm not sure if I need one or both to be able to get these speakers working. Also, I was thinking if there are receivers or mixers (still not sure which I need) that support JBL's TWS or JBL Pro Connect so I can have audio play both on the JBL Eon Mk2 and the ceiling speakers.

This is probably one for r/StereoAdvice or r/hometheater - thees almost too many options, and its a long way outside the scope of this sub. Personally i'd grab a used 9.2 home theatre system (mine cost about 70US) and add decent-ish front and center speakers, use the 6 roof speakers as surrounds - fr a few hundred you could have something kinda amazing.

Did I damage the input on my audio interface? I have an Audient ID14 MKII. I plugged my guitar into the hi-z input of the interface for the first time today, only to be disappointed that my DI clips (2 guitars, 1 passive 1 active), even with the input knob on the device at zero. I read online to try plugging into one of the line inputs. I did this without thinking of impedance....just plugged my ts instrument cable into a line input and played through it. Did I damage this input?

Its fine. The active guitar should be going into the lo-z/line in anyway, as would a lot of pedals. Worst thing that will happen is bad tone. Seems a bit odd for a passive pickup to be clipping the hi-Z tho, what is it?

Bare knuckle juggernaut.

That shouldnt be clipping the interface at min gain, it was hitting the top of the mixer in the Audient mixer??

Not the Audient mixer, but it goes past 0 in reaper and the peaks hit the wall. My pickup height is pretty moderate too.

If it isnt peaking in the audient mixer it isnt clipping on the way in.

You wouldn’t think, but it’s audible and there’s not much headroom to work with on the wav.

clipping means running out of headroom and clipping off the top of the waveform. any headroom = not clipping What is your complete signal chain, inc in the box? what are you perceiving as 'clipping'?

Just a guitar going into the interface, into the daw. One amp vst. Nothing else. I can hear some distortion on the DI when the peaks of the transients hit the wall. When the plug-in is activated, it goes into that “fart” territory when the peaks clip off.

What amp-sim? sounds like its the sim clipping rather than the interface; does it have an input gain before the sim? otherwise you probably need to put a gain device before it in your DAW. Interface gain on hi-z should always be at min for amp sims. FWIW not an axe player myself cos my fingers are too unstable, but ive always been round em an have tone nerd mates round a few times a week, theyre always blown away by the tone of Two-Notes Genome & my Evo16

So, I have a cheap electric keyboard. It doesn't have a feature I want and I've accepted it. It was a gift. But I'm seeing if I can create the feature. It has no midi output. Only female headphone audio output. My phone has sound software so I can loop some audio for layering while I play. (The feature) I'm not trying to be a professional here lol. So I figure I could buy a 3.5 male input with a usb-A2 output female, and then plug that into my usb-A2 male with my phone's micro-usb. Of course then it goes into my phone. A bit Frankensteinish but the logic is flawless. But I know manufacture standards for cheap cables don't always connect all the pins and that certain types of audio need processing that is only available in male or female counterparts and therefore not included in certain cables for certain uses. Using females for output and males for input is unusual. (I might have typed that backwards, too confusing). The real question is will this work? The below product will be used in reverse to how it's meant to be. Are it's manufacture standards capable of that? Some people have had issues with this product but they're trying to decode MP3 and interface with car input. What I'm doing might is just routing raw audio. Below is the product I intend to buy: https://www.amazon.co.uk/Multibao-3-5mm-Female-Converter-Adapter/dp/B0C8T6S64B/ref=asc_df_B0C8T6S64B/?tag=googshopuk-21&linkCode=df0&hvadid=696452133737&hvpos=&hvnetw=g&hvrand=15061380360807706055&hvpone=&hvptwo=&hvqmt=&hvdev=m&hvdvcmdl=&hvlocint=&hvlocphy=1007415&hvtargid=pla-2197222742078&psc=1&mcid=36af33f71aea3cdf9843cc437a717a1a&gad_source=1

That adapter won't work in reverse - that goes for pretty much all USB audio adapters. You probably need an audio interface for your phone that's intended for recording, with a line input, and you can connect the keyboard's headphone output in to that. Can't give you any specific recommendations, because I don't use my phone for audio.

Thank you for the response. What about this cable?: https://www.amazon.co.uk/Hollyland-Headphone-Adapter-Microphone-Compatible/dp/B0B4W8WB7L/ref=asc_df_B0B4W8WB7L/?tag=googshopuk-21&linkCode=df0&hvadid=648921372479&hvpos=&hvnetw=g&hvrand=17079400643176824766&hvpone=&hvptwo=&hvqmt=&hvdev=m&hvdvcmdl=&hvlocint=&hvlocphy=1007415&hvtargid=pla-1720778422018&psc=1&mcid=3bc3399a5e8c34d6af503f668daf6ba4 It shows it sending audio from a device to the smartphone on the pictures.

Yes, it looks like that would do it.

Cool. Would it cause problems if the source is stereo?

I think it'll be fine with stereo.

Thank you.

What options should I check out for my first studio monitors and maybe sub? I have seinheiser hd6xx and beyerdynamic dt880. My budget is around 500-2500 preferably around 1-1.5k

Before you decide how much to spend on monitors, you need to decide how much to budget for room treatment. Theres no point to $1500 monitors without it, esp with a sub

Need help setting up my Alto TS112A loudspeaker so that I can stream music from my phone to it. **Overview** Here's the speaker spec sheet: [https://www.altoprofessional.com/downloads/ALTO\_TS112A\_SPECS\_B.pdf](https://www.altoprofessional.com/downloads/ALTO_TS112A_SPECS_B.pdf) I have an Alto TS112A loudspeaker that I'd like to equip with Bluetooth capability so that I can stream music from my phone directly to the speaker. The speaker has two 1/4" inputs on the back with volume control for each and an XLR output. Previously I connected to the speaker using a 3.5mm stereo to 1/4" mono cable that went from my phone to one of the input channels. This cable wore out over time and I've had a hard time finding a new one. [This cable](https://www.amazon.ca/COLICOLY-3-5mm-Stereo-Interconnect-Adapter/dp/B07YPW1YLY/ref=sr_1_5?crid=2T8WF1LHRIPZ1&dib=eyJ2IjoiMSJ9.uRINdPTGdATVlCPpgFwr6u765UU5WJIKscQ9DIQ2fB6m42-e_t1YSzBeP84kZnNDIYz6qktK2A7wJin7g76DKXnovby9_9k_LbPEv5F5eCqJSYTVqqqi9cyxQ1cZl-emPvF-7nc7LcLR8i_RMFs-7onuS6pqFojGk7spDrN21PxsFj4DSBRUYgTrxVGBmfHfkSmT6NMjG7DE9hm3-8rYQB5ixYUIXDn6lH1kTgkCTJrPMrWjNcMY_t9E53qManFoIwrcCTlVO-WdG0XWaOKHkR4dIn545NWyLkjg5TVg-_k.dXW6b-98E3czoKfq8w5iAt9lFpHveVZ0-BGpwFn2IVU&dib_tag=se&keywords=1%2F8%2Bstereo%2Bto%2B1%2F4%2Bmono&qid=1718387107&s=industrial&sprefix=1%2F8%2Bstereo%2Bto%2B1%2F4%2Bmono%2Cindustrial%2C107&sr=1-5&th=1) is similar to the one I use, however, it states in the functionality description that using the cable as I've described (stereo input to mono output) can burnout the stereo output stage. **Questions** 1. Can I use [this dual output cable](https://www.amazon.ca/TISINO-Stereo-Y-Splitter-Cable-Breakout/dp/B083R5BJ6P/ref=sr_1_3?crid=3L0FHN1ME800P&dib=eyJ2IjoiMSJ9.uRINdPTGdATVlCPpgFwr6rrfE6m-bajobjGTJPDmGSeua42pIFRHDOD1FwbzT-4aJimYR2J-6StfymzsgeFoGqL6aXZBc1_KhGxiJCY6QQtBLZKuFIhjBB2c3ZMGcyC93a2e-z7SePzWuHIP7-Ridpe-5VX5PjXDAN9vd-r0DdQVeZQAflU5ru-JhyAOznniFIApxwK3zcC6PjKdzd_b3Uv5O9NJvIEvAyJ5IiCSKgaTIBXvPG4D5gMkM_L_pP7x87XId6U_rjh9QWP8dKh6iRIGRwvK5sUoMKyLKoQcwc8.xkaOa0nRmVpxlH21We6yJ7Ueuojpnprho-TgjfEKjqM&dib_tag=se&keywords=1%2F8%2Bstereo%2Boutput%2Bto%2B1%2F4%2Bmono%2Binput&qid=1718387212&s=industrial&sprefix=1%2F8%2Bstereo%2Boutput%2Bto%2B1%2F4%2Bmono%2Binput%2Cindustrial%2C116&sr=1-3&th=1) to connect my phone to both input channels of the speaker? Will the speaker just mix the two signals together during playback? 2. Will this [Bluetooth receiver](https://www.amazon.ca/Bluetooth-Receiver-Streaming-Wireless-Adapter/dp/B0CD3SY2HJ/ref=sr_1_4?crid=16JM6GWFUU5JQ&dib=eyJ2IjoiMSJ9.MqnnvTE0zLqteYgqat97j3sx8dqBrWiOIWXJPZW5Z9vLA8hCAThF6W-QQS2NcBfgIx1JBI_gyVFx4gVPkZ4pozNQ4To_K9ZOWJXDh1GtroHpP2hq6OpWwr3DXRGo0y6tx4ORTYvBeq06Pp39mOaXmd_einxOnJ5fTDnmIbxAkI32QdO1EurewgAZUSz0mzPUmmvR__oedX1YMkElVi7f4fi6ZRdHL4YylmPm6SdObY0.AseYkts0BC64i2lCc7TdsvOqMhIPSQGPXXJ_EMkH-to&dib_tag=se&keywords=bluetooth%2Baudio%2Breceiver%2Bmono%2B1%2F4%2Boutput&qid=1718389194&sprefix=bluetooth%2Baudio%2Breceiver%2Bmono%2B1%2F4%2Boutput%2Caps%2C114&sr=8-4&th=1) work in conjunction with the dual output cable for streaming music to the speaker from my phone? 3. Is there a better or similar alternative setup to achieve the functionality that I've described for a similar price (<$100)?

Hi, I'm new to this subreddit. I'm trying to play around with regulating microphone (Fifine T669), I found this Bertom Denoiser app whereas you can configure and test your microphone with various settings? However, upon installing the app, I dont see the .exe file to open the app itself or anything? (I used „Everything“ app from „Voidtools“ to search for it). There are only two plugins in my computer about the app, but not something to open it with. Am I doing something wrong, or do you need to do something else to open it? I'm not familiar with audio stuff, so I'm experimenting around. About the app, I installed Bertom Denoiser Classic 3.0.6 from official website if anyone is wondering. I'm hoping anyone out here can help out with my problem with Bertom, thanks.

Bertom Denoiser is not an application, it doesn't have any .exe file. It's a plugin, intended to be used within another application. To get started, you might try installing a free audio editor application called Audacity. The first time you run it, it'll scan your computer to find out what plugins you have installed, and within the application, Bertom Denoiser should appear on the Effect menu.

Thank you alot, I appreciate it, now I understand!

Are there ear bud style headphones which would provide me a quality similar to the **Beyerdynamic DT 770s**? I'm not expecting a 100% mirrored quality but I just need to have something I can switch to time to time which isn't over my ears. I'm hoping someone might have an option.

Mic recommendations please. Need a desk mic for PC. I'm not the only one in the space, there are other people speaking in the background which my current mic always picks up (cheap condenser mic). I am aware that I need a dynamic mic but I want one under 100 CAD and one that you don't need to be SUPER close to. I would like for it to still pick up my voice from 1.5 feet away. Some options im considering: [https://www.amazon.ca/FDUCE-SL40-Microphone-podcasting-Streaming/dp/B0CRYM5NLS/ref=sr\_1\_6?s=musical-instruments&sr=1-6](https://www.amazon.ca/FDUCE-SL40-Microphone-podcasting-Streaming/dp/B0CRYM5NLS/ref=sr_1_6?s=musical-instruments&sr=1-6) [https://www.amazon.ca/Microphone-Streaming-Headphones-Mic-AmpliGame-AM8T/dp/B0CBLSJTYW/ref=cm\_cr\_arp\_d\_product\_top?ie=UTF8](https://www.amazon.ca/Microphone-Streaming-Headphones-Mic-AmpliGame-AM8T/dp/B0CBLSJTYW/ref=cm_cr_arp_d_product_top?ie=UTF8)

Audio technica ATR 2100x USB mic or rode pod usb mic if you need a usb microphone, else ATR 2040 dynamic :) or the SE v7 - basically any mic with a hyper cardoid mic would be great

The sE electronics V7 would be well worth looking at, it has an extremely tight pickup pattern. I dont know the exchange (im Aussie, its bd enough converting to US lol) but its around $90US at thomann In general electrical sensing technology (dynamic/condenser/ribbon etc) doesnt matter, what matters is the pickup pattern (the reason people say condensers are bad for background noise is studio LDCs tend to have wide pickup patterns and higher electrical sensitivity). A shotgun condesner will be WAY better than a typical dynamic or even a hypercardoid like the v7 at rejecting off-axis noise. Also the inverse square rule will always apply - 2ce as close is 4x the signal to noise, always. The ideal for your situation is a headworn cardoid, 2" off your mouth, but it sounds like that doesnt suit your ergonomics.

I got a Maono PD400X but I don't see an official shock mount for it. Any suggestions for one that's work?

Got into the world of recording and mixing a bit over a year ago now and I started off with buying a Behringer UMC22 Audio Interface and an AT 2020 Condenser Mic to start my journey. Its been a good year I've learned a lot and now I want to take the next steps in upgrading my home studio. My main thing for an upgrade of my audio interface is that I want two headphone jacks so that my Artist and I can both listen when recording is going on. I have done a lot of searching and have set my eyes on the EVO 8. My budget would be around the £150 mark so any recommendations would be appreciated. Was looking at the Komplete Kontrol 6, but reviews said that the software was a mind fuck and very tedious and hard to use, whereas the EVO 8 looked a lot more user friendly and simple to set up. Secondly I wanted to upgrade my mic. It feels like regardless of when I record vocals they are always sitting above the mix and require a lot of work to be carved into my beats. I want a microphone that will lessen my job, and with a semi untreated room want something a bit more forgiving. I have again done some research and the choice was between the Rhode NT1 and the Sure SMB7. the SMB7 is looking at around £350 on Amazon with the NT1 around 150. Recording rap vocals and with the condenser AT 2020 definitely could hear absolutely everything going on in the room. With the SMB7 the need for a pre amp was heavily stated. As I am going to be purchasing an EVO 8 (or whatever you guys suggest) would the built in pre amp be strong enough or is it necessary to buy a Cloud Lift as well which is going on Amazon for around £150. Will be getting professional acoustic treatment in December before I get comments saying to treat your room first, currently will be recording standing up in the centre of the room with a sheet over the artists head. Any stands for mics you guys recommend would be grateful, I'm 6"4 so want to be able to speak properly into the mic when recording. Apart from that I don't have too many other questions, currently I use a pair of Bose QC Ultras and ordered a pair of Beyerdynamic DT990 Pro Open Back 250 Ohms to mix on, and that's what Ill be using to listen when the artist is recording where he will just have on the QC Ultras. Once again I appreciate all the help and hope that I can learn a few new things :)

>Beyerdynamic DT990 Pro Open Back 250 Ohms If you go with the Evo8, go the 80 ohm or move up to the Evo16 or ID range for their extra headphone power, 250Ohm on the bus powered evos is not gonna get much volume. (competing brands are worse, the ID series are the cheapest that can drive that high an impedance can properly) >would the built in pre amp be strong enough Yes >Komplete Kontrol 6 Bullet dodged, its just not in the league of other modern interfaces. >I wanted to upgrade my mic. For the sort of mic you are looking at its REALLY hard to look past the BeyerDynamic M90X Pro atm, great sounding mic with a nice tight pickup pattern (almost supercardioid), $150 down frm $380 - couldnt resist one myself [https://www.amazon.com/beyerdynamic-Addressed-Condenser-Microphone-Storage/dp/B096LC5SR6/](https://www.amazon.com/beyerdynamic-Addressed-Condenser-Microphone-Storage/dp/B096LC5SR6/)

Imma mainly be mixing using my old UMC22 Audio Interface if that makes any difference with what headphones I should get. I read that the lower Ohms they are the less accurate the headphones will be, and as I am going to be mixing/mastering with them I wanted the best quality that will translate the best to all different sound types. I didnt really see any reviews about the M90X Pro, do you think you could give me some of its benefits? Leaning mroe towards grabbing the Condenser NT1 if you see my other comment below.

It has a far tighter pattern (better immunity to room noise) than most competing mics, incredibly low self-noise, and sounds like a much more expensive mic. [https://podcastage.com/rev/m90prox](https://podcastage.com/rev/m90prox) Its a direct competitor to the NT1 (not NT1A or cheaper signature series) but half the money atm >I read that the lower Ohms they are the less accurate the headphones will be, and as I am going to be mixing/mastering with them I wanted the best quality that will translate the best to all different sound types. That would be true with a perfect headphone driver but the UMC22 doesnt have that. You will definitely want the lower impedance options I recommend this vid [**https://www.youtube.com/watch?v=O\_L86wNbzi0**](https://www.youtube.com/watch?v=O_L86wNbzi0)

Your interface choices sound solid but concerning mics: a dynamic such as an SM7B fed a lot of gain will pick up just as much as a condenser, so you’ll again hear everything happening in the room. The Rode nt1 is probably going to sound appreciably different since it’s a large diaphragm. I’d suggest getting it. But there’s a chance that you’ll still need to devote time to get them to sit well. Vocals just require work, I would say.

Imma be getting a K and M mic stand so that were able to record standing up in the corner of the rrom, would you ercommend an isolation shield or something like an Alctron PF8 to go with the Rhode NT1? Definitely seen a lot of good things about it going for £139 aswell so might have to get copped.

I’ve never heard of these before and did a bit of Googling. SOS did a detailed study that uncovered [underwhelming results ](https://www.soundonsound.com/reviews/how-effective-are-portable-vocal-booths) for practically all designs of shield or filter. The article is well worth reading. They suggest hanging a duvet behind the singer, and I’d say that’s a pretty good idea and far less gimmicky.

I have a boat audio system that consists of the following: * Milennia MIL-PRV19 Stereo ([https://www.crutchfield.com/S-HBlDR0CXgdW/p\_822PRV19/Milennia-MIL-PRV19.html](https://www.crutchfield.com/S-HBlDR0CXgdW/p_822PRV19/Milennia-MIL-PRV19.html)) * [Kicker 46KISLOC K-Series Interconnect, Speaker to RCA W/Line-Out Converter](https://www.amazon.com/gp/product/B00I4EF1BC/ref=ppx_yo_dt_b_search_asin_title?ie=UTF8&psc=1) * Kenwood KAC - M5014 ([https://www.crutchfield.com/S-B3WaQ8EoNC7/p\_113M5014/Kenwood-KAC-M5014.html?XVINQ=GZ0&XVVer=1D3Z&awcr=628108505932&awdv=c&awnw=g&awug=9009752&awkw=pla-1414953783511&awmt=&awat=pla&gad\_source=1&gclid=Cj0KCQjwsaqzBhDdARIsAK2gqnfB86lvoyjAb\_Rm9bTNRJKXjHpAHLSCYXeiq9FlZUHDLsDhWpZofoQaAqrsEALw\_wcB](https://www.crutchfield.com/S-B3WaQ8EoNC7/p_113M5014/Kenwood-KAC-M5014.html?XVINQ=GZ0&XVVer=1D3Z&awcr=628108505932&awdv=c&awnw=g&awug=9009752&awkw=pla-1414953783511&awmt=&awat=pla&gad_source=1&gclid=Cj0KCQjwsaqzBhDdARIsAK2gqnfB86lvoyjAb_Rm9bTNRJKXjHpAHLSCYXeiq9FlZUHDLsDhWpZofoQaAqrsEALw_wcB)) * 4x JL Audio M3-650X-C-GW ([https://www.crutchfield.com/p\_13693511/JL-Audio-M3-650X-C-GW-Gloss-white-with-Classic-grilles.html](https://www.crutchfield.com/p_13693511/JL-Audio-M3-650X-C-GW-Gloss-white-with-Classic-grilles.html)) The stereo only have speaker level outputs which connect to the kicker line out converter and are set to the minimum setting. RCA cables connect the line output converter to the Kenwood amp which I tuned using a voltmeter and test tones and following ([https://jlaudio.zendesk.com/hc/en-us/articles/204374120-Amplifier-Level-Setting-Guide](https://jlaudio.zendesk.com/hc/en-us/articles/204374120-Amplifier-Level-Setting-Guide)). This set up worked great until I turned the engine on and got some noise in the system from the engine and trim system. After that the speakers sounded muffled. I hooked the speakers directly to the speaker level outputs from the stereo and they sound fine so I don't think the speakers or tweeters are blown. When hooked back up to the amp they sound muffled again. Any suggestions? I will be installing a separate battery for the non essential onboard electronics on the boat such as the audio system which should eliminate the engine and trim noise but now the amp doesn't seem to be working correctly?

you will probably get more help in r/StereoAdvice , but did you try plugging a line cable form a heapdhone out into the amp? It sounds like an issue with the line converter Personally i would just get something that works properly out of the box, you seem to have spend a lot of money to do things almost the worst way, rather than just buying a used head unit with bluetooth and at least a pair of RCA out. Line converters will; always sound bad

I need help troubleshooting buzzing noise in microphone. Recently, I purchased a cheap lavalier microphone, the Boya BY-M1, and noticed that the recordings have a constant electric buzzing noise, similar to the sound produced by a Tesla coil. This issue only occurs when my laptop is plugged in, leading me to believe it’s a case of electrical interference. I tested the microphone on another laptop from a different brand, and it too produced the same electrical noise whenever it was plugged in. I also tried different wall outlets in various rooms and even used a power strip, but nothing seems to fix the issue. Additionally, one of the laptops has a weird issue where the capacitive touchpad becomes laggy, skippy, and unresponsive after being on for a few hours. From my online research, it seems like this could be due to static electricity build-up on the touchpad. Is it safe to assume that the wiring in my house might be faulty? Can someone advise me on what to do? For context, I live in an Asian country. Update: I forgot to post a recording in my original post. Here is what it sounds like, by the way. [http://sndup.net/5cvf7](http://sndup.net/5cvf7)

Its just the line noise form your laptop switching power supply. Not much you can do but buy better gear, inside a computer is pretty much the worst place for analog audio equipment.

So I have an Astro A50 headset that I got to have simultaneous and seamless audio transferring from my PS4 to PC. I ended up getting a PS5 which complicated matters as the headset was rendered useless as I had to consistently plug and unplug my USB dock thing from PS5 to PC and vise versa. I ended up working around this by getting a USB to LINE IN - LINE OUT converter and have two separate cables which connect into my PC. I then installed Voicemeeter Banana and the virtual audio cable. I used some online tutorial to set it all up and it seemingly worked fine. I'm not sure what I did but when I was playing Phasmophobia I noticed my microphone was barely picking up anything. I fiddled with stuff I do not recall but some include maxing out my hardware input 1 (headset microphone), maxing out my discord audio, maxing out my in game audio when possible, fiddled with inputs, etc. I feel like when I first had Voicemeeter Banana I was able to swap to Voicemeeter Input just fine but now when I swap to that input there is no sound coming out. I am only able to use my microphone when my input is under my headphones. Additionally, but when I change my Windows Output to Voicemeeter Output, sound does not come through unless I swap it to my headset audio. I am not the most tech savvy and I have tried looking up potential fixes to no prevail. I'm not 100% sure I'm even in the correct subreddit to be asking this, and if so, may someone kindly direct me to the correct one. If you guys need additional information to help solve my problem I am happy to provide. TL:DR - ASTRO A50 + Combined PS5 / PC audio with Audio Splitter + Voicemeeter Banana and Virtual Audio Cable = Great microphone levels on playback, Terrible microphone levels while talking in Discord / Games.

Hey man, maybe try r/pcmasterrace ? We just avoid using the windows audio system entirely in pro audio, Steinberg wrote an audio driver system called ASIO that completely bypasses it

Sounds good thanks i’ll comment over there :)

Outboard gear set-up question I have some outboard gear, but my audio interface only has 2 main outputs for speakers. It is a Focusrite 2i2. I’m pretty sure I would need to plug my outboard gear into those and just use headphones. My question is this, could I use the headphone jack as an output to my gear? I’m trying to use my gear as analog effects in my daw.

Hi everyone! I'm looking to set up a system that allows me to monitor my own voice during Skype conversations. Here's what I'm thinking: 1. Connect a **Rode Wireless Go II** receiver to my laptop and use it as a sound card. 2. Attach a Rode Wireless Go II transmitter to my collar. 3. Connect wired headphones to the receiver. My goal is to hear both my voice and the person on the other end of the call through the headphones. Is this setup feasible? If anyone has experience with this or similar setups, your insights would be greatly appreciated!

Why does my SM7B refuse to pickup my guitar amp, but will happily pick up my voice? In FL Studio the mic refuses to pick up guitar at all when I hit record. But once I start talking into it, it picks that up perfectly fine. Does it think that the guitar is noise and is cancelling it out? I think that's what's happening, and it's driving me crazy. Does anyone know what this issue is?

I have a motu m4 interface that I have been using for over a year, and every aspect of it was fine until about a month ago: I even recorded an entire single and mixed/mastered it to my satisfaction in late april, but around a month ago, I noticed after I compressed a microphone recording I got an obnoxious hiss. I've tried changing brand new XLR cables, pressing the XLR cables in more aggressively, securing the USB-C cable, buying a new audio interface (the volt 276), different USB-C cables, and trying it out with my windows 11 mini pc (yes i installed the drivers). The problem was exactly the same every single time. Duct taping the USB-C cable to the interface only worked temporarily. I tried wrapping a string around the interface and the cable and it didn't help. After I did research online and just found that a lot of people were having the same problem with motu m4, I concluded the problem was maybe related to the fact the cable was both supplying the data and electricity I decided to try buying the volt interface: but after getting it in the mail and hooking it up, it still makes that same obnoxious hiss noise after applying the compression (it sounds pretty much exactly like old televisions). The weirdest part of this is that the computer output to my speakers works just fine with both interfaces. Absolutely no issues! Not even when I play back my project files in audacity... So I am now down to it being audacity, a general issue with USB class compliant devices, or Ubuntu studio. Have any of you all come up with more definitive insights into this problem? Thanks in advance. I'm definitely willing to pay some money and abandon linux distributions, but I would like more information.  EDIT: Also, i have tried disconnecting the mircophone entirely, and the same thing still happens, so we can eliminate that as an issue EDIT: now i have tried turning off the wifi and my phone in my house to see if radio interference had something to do with it, and the problem persists. EDIT: I have also tried removing the interface and the speakers from the equation, and the static noise now persists when i record and compress the signal! So, more than likely, it is the computer itself doing this or the audio drivers. Great.

Can you describe the noise more? Is it present without the mic plugged in? Is it like a buzzing? If so USB power noise is quite common, the simplest solution is generally to use a good quality powered hub, or AC powered interface.

Also, i have tried disconnecting the mircophone entirely, and the same thing still happens, so we can eliminate that as an issue

"it sounds pretty much exactly like old televisions" You know that static sound from an old television when the black and white screen is on? Here is a youtube video with the exact example of the type of sound i am talking about: [https://www.youtube.com/watch?v=cptiIDohFZc](https://www.youtube.com/watch?v=cptiIDohFZc) I wouldn't describe it as a buzzing, but we are all different. What do you mean by a hub, or an AC powered interface?

If you are getting white noise, thats either heat in the actual analog hardware of the interface. You will always have some sort of noise floor and you are somehow bringing it up far enough to hear it when you compress (which is effectively turning up the quiet parts). I dont get how you made a recording without your interface? If you used the onboard sound teh noise floor will be even higher. If the noise were a buzzing, that is often caused by the switchmode powersupply of the computer and the many switching parts inside, and is passed down the USB power. Supplying clean power to the interfaces often fixes that, but it doesnt seem to be the issue here.

Well in audacity, you unplug or turn off the interface and hit R. But i guess heat is another factor to consider, my computer is very small and runs hot.

>Well in audacity, you unplug or turn off the interface and hit R My guess is that leaves you recording the system input, which would be worse. >i guess heat is another factor to consider, my computer is very small and runs hot. Normal system temps are about 350 Kelvin, 5-10% either way wont make much difference.

I recorded with the interface disconnected and connected, and disconnecting does change the static noise sound, even though theres not much of a variation with that. But my issue is most likely caused by my computer itself: i get similar results with my other mini pc, but my laptop seems to record fine. And my system temps are around 50C, so i guess thats irrelevant. When im not idle, it raises to around 52C.

Hello everyone, can someone guide me on whats the most efficient way to setup my tv + pc + speaker "combo", please? So I have speakers which have this audio inputs: TRS Balanced, RCA, AUX. I want to connect to them a TV which have optical audio output, also I want to connect them at the same time (if possible) to my pc, which have "1 x optical S/PDIF Out connector" plus 5 audio jacks (product specification not very helpful), just for reference my motherboard is [B550M Aorus Pro-p ](https://static.gigabyte.com/StaticFile/Image/Global/a8da7099b8ac4a7004a6f1bad413ca19/Product/25847/png/2000). What's the best setup for this componets? hopefully with the fewer devices possible and not too expensive, thanks in advance!

This is probably more one for r/stereoadvice. How to do it very much depends on your budget and quality aims. One of the cheaper-better options would be an old home theatre recviever to use as a DAC/switcher between the optical sources - they tend to have excellent DACs and multiple optical inputs, and go for next to nothing. Would also give you the ability to use HDMI rather than optical (much higher audio bandwidth, tho you arent using multichannel anyway, you may later)

I just have a simple question. Looking to add equipment into my studio, I already have a pair of HS5s. Would adding a pair of HS8s and a sub be the best option? Not sure having two sets of Yamaha speakers would make sense. I know my Yamaha HS5s are pretty flat (which I love) not sure if the HS8s are similar. For a little more background info, I will be mainly mixing/mastering. My question is mainly to gauge whether I should get a different brand Or stick with Yamaha. Edit: I am very open to suggestions maybe the best answer is to get different 5 inch monitors and get the HS8s? Not sure just looking for some general guidance.

Hey everyone! I'm currently using a Hyper X Quadcast for general usage and some light streaming. I recently bought a Scarlett Solo and a pair of Beyerdynamic DT 770's and they've been a game changer. Since I have the functionality for an XLR mic now I was thinking about switching to that and then starting to do some music recording as well. Only problem is I have a semi noise free environment but not completely. I'm looking for something under $200. I was looking at the AT2020 basically because of price and good reviews the other day, but I don't know enough about XLR mics to even be sure of what I need. Any recommendations or advice would be greatly appreciated.

Incredibly hard to look past the BeyerDynamic M90X Pro for $150 down frm $380 atm [https://www.amazon.com/beyerdynamic-Addressed-Condenser-Microphone-Storage/dp/B096LC5SR6/](https://www.amazon.com/beyerdynamic-Addressed-Condenser-Microphone-Storage/dp/B096LC5SR6/) Please dont get an AT2020, theyre not a great mic for the money, not even a true condenser.

apologize if this is a dumb question. How can I control my computer in my bedroom, to a monitor and my USB devices(such as keyboard+ mouse, midi controller, audio interface and whatever else I might need) in my basement My studio is in my basement. Right now I only have one computer and I carry it up and down the stairs when we jam/ record. The basement is located directly under my bed room and I have full access to the crawl space under my room and on the outside of the wall in the basement my studio desk is on. Probably only about 20-30 feet between the two locations, if that. I see things about kvm switches that carry the signals over Ethernet but I'm just making sure that's the right solution before biting the bullet. I'm also kind of confused about them handling a powered USB hub (for more ports since I see the switches usually only have two to 4 ports on them) and all of my music equipment being able to be plugged in with no latency.

Can i suggest its probably gonna be cheaper and easier just to run a network cable an buy a miniPC? $450 will get you an 8 core ryzen with 32G and a 1T NVMe [https://www.amazon.com/MINISFORUM-UM690-6900HX-Threads-Bluetooth5-2/dp/B0BRN8ND1S/](https://www.amazon.com/MINISFORUM-UM690-6900HX-Threads-Bluetooth5-2/dp/B0BRN8ND1S/) What you are suggesting can work, but the audio interface will have to be within 15 feet of the machine and you will need to run optical/analog from there most likely - audio interfaces dont tend to like USB extenders, theyre really timing sensitive. If latency doesnt matter you could run the audio over HDMI.

Hello! First and foremost, I am sorry if this is a dumb question and I am aware that there is a plethora of knowledge and advice to find on the internet regarding my problems, though me completely not being an audio guy, some more personalised help would be very helpful! Here is the situation: I have a Eurorack Modular system which I am aiming to use as an FX processor for my outboard gear, as well as samples from my computer. Importantly, this is the mixer I am using in my Euro system: [https://pittsburghmodular.com/lifeforms-system-interface](https://pittsburghmodular.com/lifeforms-system-interface) My computer has a PreSonus 1810c soundcard connected to it, which receives input from my outboard gear (Vinyl turntables, Polyend Tracker, Elektron Machinedrum, Modular Output) and outputs those to both headphones and studio monitors. Now, what I am trying to achieve is me being able to route the output audio from my outboard gear into the effects rack and also into the Machinedrum for sampling purposes. Another thing I would like to achieve would be the ability to route already processed audio from my Eurorack to the Machinedrum as well. I realise I can not monitor the dry signal with my Soundcard while listening to the processed signal from my Eurorack, but all I have achieved thus far is a wide variety of feedback loops... How can my goals be realised? I.e what has to be hooked up to what and how does it need to be set up so that I do not have to rewire anything at any time and most importantly don't end up with feedback.

You seem like you are willing to invest in doing stuff right, is this something you wanna make work with what you have or somethign you wanna get the right hardware for? The Presonus is really the wrong tool for the job - you want multi-bus DSP mixing and it only has a single analog mix bus. You can get round this by using your DAW to create the busses/routing in your DAW but this will introduce audible 10ms+ roundtrip latency (where an interface with DSP will be under 1ms through). Also ideally you want some kind of IO for the modular - either balanced +4dB or optical to modular level (balanced + 5x pro level). Personally I use an Audient Evo16 (cheapest option with 8 channels and 16 optical channels) and an Expert Sleepers ES8 (modular interface) over optical - that gives me 5 stereo mix busses in the DSP

Ideally I would like to make things work with my current gear, though if I need to invest in some additional hardware in order to use all my current machines work the way they should within my setup, I will obviously do that. I assume the combination out of my chosen Eurorack-mixer and audio interface won't let me do what I am trying to do at the moment?

Hey you fellow audio enthusiasts! A friend in the same theatre I work at wants to build an analog intercom with party line feature. There once was a ClearCom system that had just one 3-Pin XLR going from the main station to the bodypacks and you could just patch one bodypack to another bodypack. You could talk in both ways and receive audio in both ways just over this one wire. Also, it had some kind of power delivery like phantom for the headphone amps in the bodypacks and some LED indicators. Has anyone any idea how they did that? Special circuit for both way audio? I mean if you just build it the easiest way it will work, but what if both are speaking the same time? Thanks in advance for your ideas!

I record videos on my iPhone with HD 30 frames per second and my phone is about 10 feet away because I need to film content from that distance. I also record this same content using a high-quality Shure SM7B microphone through an Apollo twin audio interface which is 1 foot in front of my face, into Logic Pro X on my MAC which records in WAV 24-bit. Then, I export that audio file as an MP3 with 320 Kbps. When I import the video file and Logic audio into iMovie to replace the iPhone audio with the Shure / Logic audio, the audio from the iphone and Logic files does not perfectly line up, so I have to spend a lot of time cropping the audio and dragging under the correct spot in the video. I'm not sure if the iPhone and Logic are recording at different rates or what. Do I need to film my iPhone at 60 frames per second instead of 30? Any ideas on why this is happening and how to fix? Thanks!

Dude, save yourself a bunch of pain - just use OBS with the iphone camera extension [https://obs.camera/docs/getting-started/ios-camera-plugin-usb/](https://obs.camera/docs/getting-started/ios-camera-plugin-usb/)

Are there any tonal detriments when using an unbalanced TS line output connected to a balanced TRS input via a 7 foot TS/TS instrument cable on a modern interface such as the Presonus Quantum, RME fireface or UA Apollo?

I am setting up a small studio and would like to know if you have some recommendations on monitors (not big). Budget around 1k.

If my charger of the laptop is connected, my guitar connected to my audio interface keeps humming/buzzing, and it all stops when I remove it, but I can't produce too much time without my charger. How can I solve this? OBS: I don't have money for *Hum Eliminators* and I can't find *ground lift adapters* where I live, because here we use rounded pins plugs, so a normal adapter like [this](https://www.amazon.com.br/Adaptadores-Tomada-Esponja-Benjamin-Grosso/dp/B0BXVWXHFK/ref=sr_1_6?dib=eyJ2IjoiMSJ9.0gDfSxKiF2i4XV7hKYYPhp7YFbt4cM7a2AjILTDQFmKDkoKH9KuT-ASmSl5dChp7g2NqSMtfhzQfI98VYh1nNmdOF5Q_EQWuTx7mHmTgBXIay4CIcT0op1i58O2l7AqEXWePYcXUQqwUVpa4M9Hf91ICmKDg7bEPuh3jV67PbK_WVNRJgFl-m7s7UB_r6uMnTvC6zj3EKhN0m6GSbV_HY-Aex5DC_Xi1OaKOjCFtI9nkxC-Zfu2ayyhzu1UJw48rL5giQ6OaxSNFftfefFLQ8YIEKDV6IDZr83JdwruQ1aw.HvomW8eR_ILc36rGBUBn85AYaXw6KoDJjGTmSdrQF9Y&dib_tag=se&keywords=adaptador+3+pinos&sr=8-6) will solve the problem?

Guitar pedal on stand by causes distortion: When I put a guitar pedal after my moog sub 25 it distorts the sound, even if the pedals are in standby. I can hear it, and on an analyser I see the fundamental lower in db + extra overtones that weren’t there before. It also happens with pedals that have true bypass and every pedal seems to impact the sound slightly differently. The synth and pedals are connected to the same power supply. Any clues on what might help with this issue? I cant seem to find people with similar issues on google and am kind of lost :)

Dunno why its happening with pedals claiming 'true bypass' but any other pedal, analog synths will clip the amp circuits whether the effect is engaged or not - turn down your synth.

# Are Genelecs still a go to monitor compared to modern monitors like IK Multimedia MTM with room correction? What set of monitors would you get for under 1K if you were starting out? Looking for a pair for monitors for my (barely treated bedroom) I know room correction isn't a cure for treating rooms, the bass will still swell up, early reflections, echos, and all that, but for someone who only mixes on headwith (with Slate VSX and AKG k702) I wanted to finally get some monitors I found a good deal on Genelec 8030's from 2015, they're kinda beat up but the guy only wants 600$ for the pair locally Looking at my other options I saw IK Multimedia MTMs with built in room correction and they're powered with their own DSP for 500$ or I could go for a pair of these weird FOSTEX NF01A from year 2001 for only 400$ for a pair (The tweeters are made of banana skin!) For some reason I heard these were really true to the sound but have no bass Then there's ADAM A7X, Yamaha HS8, which I know nothing about My room is standard bedroom size, and I honestly was thinking of buying the cheaper acoustic panels off amazon or ebay. I read before to truly get your room acoustically treated you need to spend like 100$ per 20" panel or so and make sure its fiberglass etcs. SEEMS very complicated and expensive lol

Unfortunately no such thing as go-to monitors as everyone has different tastes and preferences. Genelecs in particular tend to get very "love them or hate them" responses, you really need choose with your own ears. Note that if you want to use room correction with monitors that don't have it built in there's several software based options like Sonarworks, Dirac and ARC. There's also hardware options, the new ARC box is reasonably priced but you can do hardware Dirac (more expensive) or Trinnov (really expensive). For panels, DIY is by far the cheapest way. Plenty of info around the internet showing how to make them and what material should be used depending on availability of where you live.

I just bought a Shure SM7db. I need to connect it to two computers for separate use without fiddling with wires each time. Is there an interface that has dual USB outputs?

Lewitt Connect 6

Use a regular interface with a USB switch.

This got kicked out of the main sub so I'm posting it here. New mixer coming into the studio Just ordered a Presonus 32S. I've used one running sound at clubs but not with the software (Studio One). Anyone have any thoughts about that software? Anything I should know about it? There's many alternatives to that brand of course, but I don't want to get off topic. I live in New Orleans and if I have to get it fixed it's a short drive to the Presonus factory in Baton Rouge. Being able to strangle someone in person is a big selling point. Plus a lot of venues here use Presonus, so I should be able to bring my settings in on a USB and load em up. I also wanted something I could record with independently of a DAW. You can record straight to SD card, USB, and through firewire to the computer, multitrack. I can of course compose on the computer and have it run midi to the synths via the DAW, play them all at the same time, 32 physical inputs plus a handful of sends! I am just tired of trying to manage the composite ASIO sound driver on the computer. Look at it wrong and it refuses to connect to something. Thanks for any suggestions about the software side (I use PC) and if there's any little trickery in the hardware I should be aware of, please also advise. Thanks!

Mods suck, can’t even ask a question. Tells us to post here to a 2 day old thread that no one will respond to.

Help with aggregate device recording: I've been trying to record audio through two Samson Q2U microphones on my Macbook Air. I've created an aggregate device which (when testing the mics in the sound settings) seems to work. I can speak into each separate mic and the audio from both is picked up. However, when I go to record using this aggregate device, the audio is only recorded through one microphone. Is there any step I might of missed along the way that would explain why I can't get this to work? The only other thing I can think of that may be the issue is that both mics are connected via USB to a multiport USB hub which connects to my MacBook through the two USB-C connectors. Any help would be greatly appreciated!

What DAW do you use? Do you have two tracks set to take inputs from each mic, and are the tracks both record armed?

I was planning on just using garage band (unless there is a free alternative you’d suggest) and I’m pretty sure i have two tracks set up. Not sure what you mean by “record armed” (very new at this)

If each mic is set to a different input in your aggregate device, in GarageBand, make sure that you select the respective input on different tracks. The inputs for each mic are most likely gonna be 1 and 2.

I think I’ve figured it out now! Thanks for the help

I just recorded a 2 hour podcast on the Zoom H8 recorder. When I pressed stop recording there was a long load time about 3min. Went to start a new recording after and got an error message. Turned device on and off and it worked fine. Get home and first recoding with long load time will not play and the file size is much smaller than normal. After converting into an mp3 the files will now play but are only 6min long. Any help would be really appreciated.

Does anyone here have any experience with the 500 series version of the Dramastic Obsidian bus compressor? I want to use the external sidechain but cannot for the life of me figure out how to route the external sidechain signal into the unit. I have it housed in a Cranborne 500ADAT enclosure if that helps.

I'm looking at upgrading my interface to a rackmounted unit that provides 4-8 solid mic preamps I can use now and 8+ line-level inputs I can connect outboard preamps/channel strips to in the future, eventually bypassing the built-in preamps with higher-end stuff. I have an 8-channel preamp I'll be connecting via ADAT and am happy with 12 mic preamps for now, but 16 would be a bonus. My primary desire is line level inputs that do/can bypass the interface's preamps so they aren't in the chain with future outboard preamps/strips. My understanding is some/most interfaces don't allow bypassing preamps via the line level inputs, especially if they're combo jacks. Am I mistaken? Does the same apply for *separate* line level inputs from the XLR inputs? From reading some manuals I believe the separate line level inputs can or natively do bypass the preamps, but I'd like some assurances/verification of my assumptions from more experienced users. I'm looking at RME Firefaces for loads of reasons, but especially because they include more/separate line level inputs, so I could start connecting outboard preamps/strips while still using the onboard preamps to get more mic pres sooner. Background: I'm used to desktop and rackmount interfaces with combo jacks where bypassing the preamps is not possible, with the exception of the insert returns on my Audient id44, which are awesome and have allowed me to connect outboard gear bypassing the preamps. Most interfaces don't have insert points for their preamps, though; only the Audient and Arturia desktop units I've seen do, but I want more channels and rackmount form factor. Preemptive for those who will indicate outboard gear is a waste of money: I agree plugins have gotten amazing, but the more I mix, the more I want to get things as good as possible when tracking to facilitate less laborious mixing. The stuff I've mixed that was recorded on nice outboard gear simply requires less work/fiddling to make sound great ITB. Thanks!

a dedicated line-in will have no mic pre amps in the circuit at all

\[Shopping\] Thoughts on [Behringer 1273 2-Channel Pre-Amp/EQ](https://www.sweetwater.com/store/detail/1273--behringer-1273-2-channel-microphone-preamplifier-and-equalizer) + [Behringer 369 2-Channel Stereo Compressor](https://www.sweetwater.com/store/detail/369--behringer-369-2-channel-rackmounted-stereo-compressor) for my first outboard hardware processors? Mixing. Audio editing. Post production. Sound design. Budget options, of course.

I cant find a board pic but Behringer stuff is generally SMD and harder to modify/repair than other recreations like Warm Audio or Golden Age Project that are more traditional through-hole like the originals

I am planning on purchasing a Shure M7B mic, I'd like to use XLR connection, but I am not certain what kind of interface I should use. Does anyone have a recommendation? Is the Behringer UMC22 good enough? I mainly use my mic for Youtube stuff, and I know that thing compresses my audio anyways

Why on earth would you spend $400 on a mediocre mic, buy the cheapest interface, and nothing on room treatment? Buy an Audient Evo4 (or ID4 if you need to drive high end headphones) and an sE v7 (extremely tight pattern for poor spaces) and spend the rest properly treating your space. Dont buy on hype.

Well I currently use a sE v7 with a u-phoria um2 and I really hate how the audio comes out of it. So are you suggesting if I just upgrade my interface that may fix the root issue?

“What is the actual problem with the audio?” is a very valid question, that would be wise to explore; especially because the sE V7 is an excellent mic and sounds great.  Despite the price, it’s one of the best sounding handheld dynamic mics to be released in recent years. Might consider how you’re performing with it.  Is it far away from you?  If so, it needs to be quite close to your mouth, like within a few inches ideally and not like a foot away.  If you can’t record excellent sonics with the V7, it’s unlikely the SM7B will help the situation.

What is the actual problem with the audio? If there is a noise floor issue a better interface will help, and a better interface will have better drivers (& ASIO support) but if you are looking for 'radio sound' you are looking for mic technique, EQ and compression.

Hmm I'll this routing then. I will buy the Evo4 Audient, it does look like quite the step up over my current interface. Perhaps that will remedy my sound issues. If it doesn't I'll report back and return said interface. Thank yah muchly

You would probably be better off learning to use EQ and compression in whatever software you are using, or the free VB-Audio VoiceMeeter, by the sound of things, unless noise is the issue. The Audient is definitely a step up, its full pro grade as far is I/O quality, but thats only gonna help if thats your issue

Ignore the other guy. SM7b is a fantastic, classic, industry standard mic. Nobody who knows anything about audio would consider them "mediocre". Since you didn't specify what you hate about the audio, I'm going to guess you didn't buy an inline amp like a cloudlifter or sE DM1. The SM7b has low output, like other broadcast mics, and I'd bet the interface you have has nowhere near the gain it needs.

Its a unidyne III capsule, same as an SM57 or 58, it may be 'classic' but it is just not in the league of something like an RE20, and is nowhere near worth what is charged for it

I have a 2.1 setup, sub has an inbuilt amp. It powers two 15w 8 ohm satelite speakers however these plug into the sub via RCA. Trying to hook up a different pair of 50w 8 ohm passive speakers which use generic speaker cable input (banana plugs). I have spliced an RCA cable to speaker cables, using the inner core of the RCA as +ve and the outer wire as -ve, however there is no sound at all through the speakers. Wires show continuity through multimeter, why is there no sound at all? All guides that I have viewed state that this is the correct way

Yah- what you've written does sound like it should work. If it's some sensitivity issue, worst case scenario is output being quieter (if you drive the amp to clipping it could damage the speaker, but besides that). If you checked continuity, there should at least be sound. Have you also confirmed that the 50w speakers are working with another amp?

They did work through another amp, I don't have one to personally test but they were given to me by my grandad who showed me them working before I took them

All right, well- amp is probably not powerful enough to drive the speakers.

this is probably more a question for r/StereoAdvice

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I have a 2.1 setup on windows, and when I plug in my heaphones, the whole system output switches over to only the headphones. Is there a way to set it up so that the subwoofer plays while the main sound is coming through the headphones? I'm currently on a basic windows back of motherboard I/O setup until my interface comes back from the shop (with the interface i can achieve this easily) Any help would be apprieciated.

Not possible; at least not simply. Subwoofers take the full frequency range input, then split that into bass for the sub and higher freq to the other speakers. I suppose you could get what you want if you're able to output to both the headphones and sub simultaneously, but this would requite an audio interface. Perhaps there is some software out there that could do it, though, but I don't know of any.

probably more a question for r/stereoadvice or r/pcmasterrace

Wondering what alternatives there are to the CM-15? Just used one and it's amazing, but far out of my price range. I need something that will plug straight into my Sony a7S III via 1/8" TRS. Heard of the Trula, but honestly I don't need all the bells and whistles. I'm also not 100% sure how well the 1/8" out would work on that. Just need something with high quality audio that I can plug into my camera so I don't need to spend time aligning and editing audio. The advantage of the CM-15 is that it was high enough quality that I actually didn't need to do any post-production on the audio, which has proven to be very time consuming. I'm aware that I could use a lavelier or output from the Zoom H6. However, plugging in directly from the mic is highly preferred after experiencing how simple that workflow is for myself. My normal setup is an SM-57 recorded through Ableton, then aligned later. I'd like to find a mic that's good enough to avoid that. **TLDR:** Used a CM-15 and loved it, wondering what alternatives there are that us mortals can afford that support 1/8" out. (Although I could use an adapter if necessary)

probably more one for r/LocationSound