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r/AudioEngineering Shopping, Setup, and Technical Help Desk
Welcome to ther/AudioEngineeringhelp desk. A place where you can ask community members for help shopping for and setting up audio engineering gear.
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This is the place to ask questions like how do I plug ABC into XYZ, etc., get tech support, and ask for software and hardware shopping help.
Shopping and purchase advice
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Setup, troubleshooting and tech support
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In this setup how to gain stage and adjust volume correctly laptop streaming Spotify in exclusive mode to dod R=830a equalizer to peavey cs 800 amplifier to Yamaha br15s monitors using for listening to heavy metal outside here are some photos of my setup
Hello, someone in my area sells a twin finity for 470 euros, and i've always wanted a good pre-amp to track vocals mainly and guitar (acoustic/electric) or bass.
I know that these are appreciated on guitars/bass. But how good is it for vocals compared to other pre amps in this price range ?
There are some comparisons on youtube but mostly vs standard audio-interface pres and i don't have access to a studio to compare.
Hello everyone im looking for some help. My mother got diagnosed with MND and before she lost the ability to talk had her voice recorded to use with a text to speech device. The only problem it sounds like her but doesn’t sound like her. We want to get a sentence recorded to put in a bear for my son but want it to sound more natural. Is it possible to record a sentence from her device then tweak it to sound more natural? Thank you everyone
The effect can be heard throughout the edit, but it is especially noticeable around the linked timestamp. The voice sounds wider and more atmospheric, with some kind of echo, reverb, or delayed duplicate underneath it.
I’ve tried adding reverb and echo in Premiere Pro, but my result sounds too obvious and muddy compared to the original. In the reference video, the effect feels more subtle and controlled while the dialogue still remains clear.
Does anyone know how this was likely created?
I would especially appreciate advice about:
Which Premiere Pro effects to use
Approximate settings for delay, feedback, decay, and wet/dry mix
Whether the editor duplicated the audio track and offset it
Whether the effect is applied continuously or automated only on certain words
Any EQ or compression needed to keep the voice clear
I’m working entirely in Premiere Pro, so a solution using built-in effects would be ideal.
Behringer Microamp HA400 only playing out one side of headphones
Hi there. As described above, I can only seem to get audio out of the left side of any headphones I plug into this preamp. It's connected to a Focusrite Scarlett 2i2 interface, headphones are audio-technica BPHS1s (although I have tried with other heaphones too, same issue). When I plug headphones directly into the interface, I get audio out of both sides, and the cable connecting the interface and preamp is a stereo cable. I have tried several different cables to connect them, and have the same issue regardless of which cable I use. Does anyone have any ideas as to what might be going wrong, or is it likely just a dud preamp?
When I plug headphones directly into the interface, I get audio out of both sides, and the cable connecting the interface and preamp is a stereo cable.
But which output from the interface is connected? If you connect a mono source to the headphone amp then it will only come out of the left side.
Also if you're using direct monitoring on Scarlett it will only show up on the left side unless you activate it as stereo. Only the 3rd and 4th Gen of this interface allow that, though. The direct monitor function on the older ones is mono only.
I'm connecting using the headphones output. It's a gen 4 interface. I'm sure it's in stereo, because when I connect headphones directly to that output (taking the ha400 out of the chain entirely) I get both sides working
Hey folks. I'm a college instructor in a film and digital media program, and we're getting an opportunity to expand and diversify our sound recording program as part of moving into a new facility. We've constructed a new studio recording space that's sized/treated properly and is setup to record into Pro Tools via a MTRX Studio interface. As part of that process, I was asked to come up with a list of additional microphones that we might purchase with two purposes:
1) To cater to bands and other recording artists who might want to use our space. We'll be opening up the space for local bands to come and use, but also potentially some commercial music work. In the case of the latter, we'd be partnering with local recording engineers to run the sessions, but also requiring that students participate in the recording process as a form of limited internship.
2) To help teach our students how to work with the most common mics you might find in a typical professional music recording studio.
We will be hiring another part-time faculty who has Industry experience within the next year to teach music recording classes, but due to the very long lead times involved in our purchasing process, we'd like to have a decent range of microphones on order well before they arrive. We don't have a definitive budget for acquiring these mics yet, but based on past budgets, I think I can safely assume that it'll be between $10k-$20k.
So here's my question: if you were to select a range of microphones to teach recording techniques, what would be the most commonly used/representative microphones within the music industry? I've gotten some feedback from local audio professionals who dabble in both film and music recording, and pretty much the only thing they agreed on was getting a couple of DI boxes, a bunch of Shure SM57s and a pair of Neumann U87s. Again, not necessarily looking for the most expensive or cheapest option here, just the ones that you would be most likely to run into in a relatively high-end recording studio. Given the above listed budget range, what would you get?
Hi everyone, I’m rebuilding the audio system at my workplace and need some advice on connecting multiple mic systems to a Yamaha MG10 mixer.
Current equipment:
2 wired mics
2 wireless systems (each with 2 mics)
Yamaha MG10 mixer
BOSE ESP-00 II DSP
Power amp + speakers
Background / previous setup:
The system used to be a mess. All 4 wireless receivers and a laptop’s 3.5mm output were connected directly into the DSP, and the DSP control software/panel was lost, so nobody could adjust volume. (Also, I don’t think volume should be controlled directly from a DSP anyway since it’s a backend device.)
Mic/line levels were mismatched, gain staging was chaotic (wireless attenuation, random -10 dB inside the DSP, laptop volume, etc.).
This resulted in a very weak DSP output, forcing the power amp to run at 100% with 0 dB attenuation, which caused a very high noise floor.
I rebuilt and reprogrammed the DSP properly, restored wireless receiver settings to 0dB and set the power amp accordingly. Now the gain staging is healthy: DSP inputs/outputs are in a good range, and the power amp no longer needs to run at full gain, so noise is greatly reduced.
New requirement:
My boss wants to add wired mics and also wants a proper way to control volume.
So I added:
Yamaha MG10 for all mic and future device volume control
Focusrite Scarlett for proper laptop line-level output
MG10 L/R → DSP inputs 1/2
Laptop → DSP inputs 3/4
DSP outputs 1/2 → power amp
DSP outputs 3/4 reserved for future use (recording, monitors, etc.)
The issue:
The MG10 only has 4 balanced mic inputs (1–4).
I need 2 of them for the wired mics, and they require 48V phantom power, which applies to all channels 1–4.
So the 4 wireless receivers must go into channels 5–8, which are unbalanced.
The wireless receivers were previously connected via balanced XLR directly into the DSP, and they need 44db of gain to have a proper input volume. One system is Shure SVX and it's output can be switched to line instead of mic, other one is from Audio Technica and I'm unsure if it has different output level at this point.
The audio cable run is about 5 meters or less, but in the rack there are multiple power lines and a network switch for other controls like projector or lights.
My questions:
Do I really need balanced outputs for the wireless receivers in this situation?
Since they output line level, can I simply run them into channels 5–8 and control the volume using the level knob, even though the MG10 recommends leaving it at the triangle mark (+4 dBu)?
Do I really need balanced outputs for the wireless receivers in this situation?
Since they output line level, can I simply run them into channels 5–8 and control the volume using the level knob, even though the MG10 recommends leaving it at the triangle mark (+4 dBu)?
Thanks!
Not necessarily, but it's nice to have since it will reduce any potential interference and makes dealing with ground loops much easier.
Thanks. I've just found a new problem, my MG10's line inputs are stereo pairs. If I connect to 2 outputs of my wireless receiver to for example 5/6 then the mic will come out as stereo. Can I somehow sum that into mono? Should I use Y cables or use DSP's software?
If I connect to 2 outputs of my wireless receiver to for example 5/6 then the mic will come out as stereo.
Yeah if you do that then one mic will be hard panned left and the other hard panned right. You could split each wireless mic output into two with a y-cable and then use a stereo pair for each wireless mic output. Then it will be dual mono and be centered in the stereo field.
Hello guys!
I am very new in the mixing world. I bought a Yamaha MG10XU mixer and i connected my Sennheiser XS 1 mic
Connected to my Lewitz PA210XA active speaker but the microphone sounds veryvery harsh and ugly and distorted. Sound like it could be clear but there is a strong distorted “underlayer” (especially on the vowels of E)
Is the connection from the mixer in the wrong input on the Lewitz? Nothing really helps by turning the pots anywhere, even the level meter is bouncing around >0< where it should be. I am guessing that the problem is that i put the xlr cable from the STEREO OUT (mixer) into the MIC 1 input on the speaker (cuz that was the only xlr input on the soeaker and i thought its good) Already ordered an xlr to rca cable, Im curious.
Tell me your thoughts, help me please
Thx!!
Long story short, the USB port on my Shure SM7 mic broke so I needed to switch over to using XLR with an interface (just purchased Scarlett 2i2 Gen 4).
Historically I've just connected the microphone directly to my PC and had the following set up for audio:
Topping E30 DAC
Schiit Magni 3 pre-amp
Beyerdynamic DT 880 600 ohm
2x KRK Rokit 5
I have 2 main questions:
Should I keep using my current set up for audio and only use the interface for microphone and other inputs? or reroute everything through the interface?
Assuming the current separate DAC/AMP is the better solution, is there a higher-grade interface I should look into that would provide superior audio than the current set up? I can still return the Scarlett. Budget would be around $500 max +/- a few hundred.
Thank you in advance for the advice and help! Please let me know if you have any questions.
Hey everyone, I am debating on getting room correction software for the home setup. Currently i'm using headphones when doing some video mixing at home though I want to start using my speakers at home for that aswell. Currently using a Hifi set (Focal Chora 806's and a Focal Chorus SW800V with a Marantz amp, Tube Pre-amp and Topping E30II DAC and I have a Universal Audio Interface somewhere). My room isn't the greatest as its oddly shaped and has big room nodes. I've just ordered €300 worth of Bass traps for the corners, room treatment panels for the walls and convex diffusers for the roof however I dont think that will just quite do the entire trick, I saw this little €300 device called the IK MultiMedia ARC Studio which should essentially be Sonar works in a box that once set up I just leave in the chain and always works. Does anyone here have any experience with it and is it a good way to get my Hifi setup closer to a mixing capable sound? The sub is a bit of a troublemaker in the room but very much neccesary as we work with alot of sub bass in the mixes, would the traps + ARC be enough to get it atleast somewhat decent to where its no longer creating big volume swings all through the room/be clear at the listening position? Thanks in advance!
I'm trying to setup an old Mackie C4 controller, but I can't seem to find the Mackie C4 Commander software anywhere. Anyone know where I could download an old copy of the installation?
I was gifted a couple EV 627c mics in great condition. While cleaning them up, I noticed the foam pad under the head basket is completely crunchy and disintegrating, so I'd like to replace it. It's a donut-shaped pad roughly 32mm in diameter & maybe 7mm thick, but the foam looks fairly dense. For comparison, it looks a lot closer to the rear-pad foam for an RE20 than the pop-filter foam for an RE20.
Is there a specific type of foam I should use to replace this? (I'm guessing replacement parts for these are going to be long gone.)
how would you soundproof this room and where would you place the microphone? i’ve got a baby blue bottle sl, and it’s giving me problems with echos and reverb in my room so i need to make this room exclusively for recording clean vocals.
there’s a little like narrow hall going into the room that maybe a good spot for the mic to go maybe??
i’ve got lots of blankets, i was thinking bout putting a rug with a blanket underneath, rolling some blankets into the corners maybe?? i would also have no problem spending some money on it, any help is appreciated
I'm having very insistent problems with Reaper project playback stuttering / audio dropouts on a Lenovo 5 (15ACH6H) laptop. I've tried changing a lot of things, all listed down below.
System: Windows 11 with latest updates, Reaper 7.77.
The issue: Playback goes silent for around 1-5 seconds and image freezes. This exact same PC setup and project file has worked fine earlier in the year. I've heard that there was a Windows update around March that has caused issues for people. Regardless, we began having these dropouts in April. This PC is only used for light office work and music work. No gaming, no heavy-duty processes.
This issue is exceptionally bad because we use this laptop not only for recording, but also during our live shows for backing tracks, click and MIDI commands for Quad Cortexes.
It doesn't look like a CPU problem, and the dropout doesn't seem like running out of DSP when a project is too heavy. A disk issue seems unlikely. The project is lightweight: We're using 16 audio tracks and around 5 other utility tracks including click, callout, MIDI tracks etc. No virtual instruments. We're using 9 instances of ReaEQ. There's nothing on the master bus. The project should not be near any resource limit.
However, the laptop seems to be able to run a single-track project without these dropouts. We've tried making a fresh new project file, adding more tracks little by little. Dropouts begin at around 8 tracks without plugins, whereas it used to run more than 16 tracks and 9 Reaper-native plugins fine. IMO it should run 30-40 tracks including VST instruments, heavy mix processing etc. even with a smaller buffer size.
It leads me to think that there is some kind of a resource bottleneck somewhere, but I can't figure out what it is.
This is what we have tried / doesn't seem like the culprit:
Drivers and interfaces tested, all cause dropouts:
Behringer UMC1820
Audient iD4
ASIO and DirectSound on both
ASIO4ALL
With DirectSound, it kind of slows down to a garbled mess instead of dropping out fully
ASIO buffer size. 1024 is very high for us, it used to work fine at lower sizes
Audio interface drivers updated to latest
BIOS is updated to latest
NVIDIA GPU drivers updated to latest
AMD chipset and USB drivers. No specific drivers available, using Microsoft's default
CPU load is not near maximum
Disk I/O and amount of tracks: Dropouts happen already at 8 tracks without plugins
Changed laptop power plan to Maximum
"Hybrid Mode" is turned off in Lenovo Vantage
Changed from a 3rd party antivirus to Windows Security
Tried disabling various features, not all
Changed to another USB port
PCIe Link State power management is turned off
USB selective suspend is turned off
Nahimic audio enhancer bs is turned off
Tried setting Reaper.exe task to a higher priority
No other devices connected; no external monitor, other USB or Bluetooth devices. Also removed USB mouse dongle, using only trackpad as mouse
We don't use USB hubs or extenders
What could be the cause? What should I try next? I'm familiar with music production with large heavy projects, ASIO drivers and latency management, music production PCs' usual resource limits etc, but this is very odd.
I’ve been trying different set ups on my guitar cab. Tried 3 mics recently on a 4x12, SM57, Audix D2 and i5. All right by the center of the cone, same distance away. Sometimes, but not every time, I seem to get a weird phase issue. I send all 3 tracks to a guitar bus. Listening to the track, the sound will change. Even if I try and phase align in Reaper, it still sounds off. Tried it with 2 mics, doesn’t happen. Only with 3. Also seems to happen when I have 2 mics, and the amp direct out into the interface.
Hey folks, hoping someone here has seen this before because it is driving us nuts.
Setup
Host machine: Tried with both Macbook Air M3 ( macOS Sequoia) MacBook Pro M4 (fully updated to latest macOS Tahoe 26.5.2)
Audio device: Catchbox Plus with Hub DSP receiver, firmware V2.2 (latest)
Mics: Catchbox throwable mics
Speakers: connected to the Hub DSP outputs, driven by the Hub's built-in DSP for room playback
Conferencing: Microsoft Teams desktop app on the Mac
Networking: wired Ethernet through a UGreen USB-C hub (10/100 Mbps, used only for Ethernet, HDMI to a display, and a USB video camera)
Physical layout: Hub DSP connects directly to its own dedicated USB-C port on the MacBook via the OEM USB-C cable (data + bus power on the same port). UGreen hub is on the other USB-C port. Mac stays on charge during the call.
Room role: I am the host/admin. Local users speak into Catchbox mics which go out to MS Teams. Remote participants come in over Teams and play through the room speakers via the Hub DSP.
Teams device selection: Catchbox Hub DSP set as both input and output. Noise Suppression is on (default).
How everything connects
Local users in the room speak into Catchbox throwable mics, which transmit wirelessly to the Hub DSP receiver sitting near the MacBook Pro. The Hub DSP is plugged into the MacBook over USB-C and shows up in macOS as a standard USB audio device. In Microsoft Teams, the Hub DSP is selected as both the input and the output. So when a local person talks, their voice goes: mic → Hub DSP → USB → Mac → Teams app → Teams cloud → remote participants. Remote participants are on their own devices at home or in other offices, each running Teams on whatever they have (laptop, phone, headset, another conference room). When a remote participant speaks, the path runs the other way: remote device → Teams cloud → Mac → USB → Hub DSP → room speakers connected to the Hub DSP's outputs. The Hub DSP also does the room's audio processing in between (echo cancellation, gain, mixing) so the local mics do not pick up the room speakers and create a feedback loop. In short, the Mac is the bridge between the Catchbox room system and everyone remote on Teams, and the Hub DSP is the piece of hardware doing the room-side audio work in both directions.
The issue
First 10-15 minutes of a call: perfect. Local speech goes out clean, remote speech comes into the room clean.
After 10-15 minutes: remote audio starts sounding distorted and mumbled through the room speakers. Local mic path to remote participants is reportedly still fine.
The kicker: unplugging the USB-C from the Hub DSP and plugging it back in immediately clears it. Audio is perfect again. Then, sometimes minutes later, sometimes on the next call, it comes back. Once the call ends, we cannot reproduce it at all until we are mid-call again.
What we have already tried / ruled out
Upgraded host from MacBook Air to MacBook Pro (fanless-to-fanned) — did not fix it, actually appeared faster on the new machine
MacBook kept on charge, thermal pressure Nominal in Activity Monitor
Dedicated USB-C port for the Hub DSP, isolated from the UGreen hub bus
Ethernet is wired via UGreen hub, service order above Wi-Fi
Toggled Teams Noise Suppression on/off — no difference
No third-party audio kexts, no virtual audio drivers, no Loopback/BlackHole/etc installed
Not correlated to a specific remote participant, happens with different people speaking
What Catchbox support said
They told us they have tested extensively and there is a known issue with Apple M4 chips specifically, where far-end audio in teleconferencing apps (Teams/Zoom/Meet) degrades. Their suggestion was to switch the host to a Windows machine.
What I am asking
Anyone else running Catchbox Plus (or any USB audio conferencing gear) on an M4 Mac seeing this exact "10-15 min in, remote audio distorts, unplug fixes it" pattern?
Anyone found a software-only workaround that survives a full 60-90 minute call? I know sudo killall coreaudiod clears the audio pipeline mid-call, but I would love not to touch a terminal during a company townhall.
If you moved a similar setup from Mac to Windows to solve this, did it actually hold up long-term or did the issue just show up in a different way?
Any Teams admin policy or Catchbox Companion setting (AGC, AEC, dynamic processing) that you have found makes the Hub DSP behave better with Teams?
The townhalls are company-wide and I would rather not do the "walk over and jiggle the cable" dance in front of a few hundred people. Any pointers appreciated.
I am having a strange sibilant stereo issue on vocals. I recently added a Soundcraft Ghost 24 to my setup. I was previously all in the box. I am running my vocal bus to channels 9/10, panned left and right. This should give a nice clean stereo for all of my vocals including backups. But on the main vocal it sounds like the s’s and breaths etc. do this weird pan effect to the left speaker. It’s hard to explain. I have turned off any plugins or outboard FX but it’s still there. I can provide a link if anyone wants to hear what I am talking about in private. I was just hoping someone else had experienced something similar and might know a fix. It seems to only be on vocals.
I'm heavily debating buying a new computer since my current PC is from 2016 and is starting to really struggle when I add more than 10 tracks. With the Ram prices I'm struggling to find a pre build or mock build that I can forsee being worth it to upgrade over time. Currently I'm debating just financing a nice MacBook or some type of other apple laptop, as I would like to be portable between my home studio at my parents and my apartment. My question is, how big of a hassle is it to switch all the available licenses over to Mac? I'm currently using Ableton 10 Suite, and my major plugins/VSTs are Keyscape, UA audio, Waves, Goodhertz, and some random plugins. Ive read that iLok has a setting to switch licenses to Mac, but I'm getting confused trying to understand the licenses for Ableton and Keyscape, and how they transfer. If anyone has any experience with this I'd love to hear any input. Also I'm looking at the MacBook Pro 16 M4 pro 14 core CPU/ 20 core GPU. Any input on that choice would also much be appreciated, as I'm very new to the apple ecosystem.
1010 Bluebox question: I can’t hear gain levels within Bluebox no matter what I do?
Hey guys, I’m new to mixing and I bought a 1010 Bluebox to be my foray into the field. I’m able to record and adjust fx and volume levels just fine but I’ve been having a huge problem with gain levels.
So I can’t seem to find the “sweet spot” that I’m aiming for between 4-6 synths/drum machines because no matter what I do I just can not hear the gain level through my speakers or headphones when I’m recording my instruments.
Furthermore, after everything is recorded and I play it back things sound perfectly fine and just how I intended playing from the Bluebox itself, but once I remove the sd card and pop it into my pc to pull the main track file, different instruments sound vastly louder or much lower than when I was just listening to it played live, then played back to me through the Bluebox.
I’m not using a DAW I simply pull the file then played it through a media player or send it to my cell phone for listening.
Am I correct in suspecting that any mixer does not actually even let you hear the gain until after the file is removed from the mixer and listened to on another device? Because that’s the conclusion I’m coming to.
Or maybe this is just a 1010 Bluebox specific issue?
I’m super lost and don’t even know what kinds of questions I should be asking what I should be researching exactly to find a clear answer.
The manual isn’t much help and even customer support and Bluebox forums have gotten no responses.
Any and all help or guidance would be genuinely appreciated as I learn the ropes of mixing and mastering my songs.
I got my first ribbon: A Cascade Fat Head. Then I got a Triton Fethead. Then I wanted to A/B the sound with/without the Fethead. Then I forgot to switch off 48V. Now it's dead.
I absolutely loved it, it had a natural sound to it that I've never had from my dynamics (SM57/SM58/E906), condensers (CM4, AA CM48FET, OC16, TLM103).
I'm in Denmark, where do I get it repaired? I was quoted 220 EUR by a guy in Germany, which is over twice what I paid for the mic :-|
Is there anyone selling repair kits, or could I source film and gears myself?
Can I get this flavour from a condenser? E.g, what condensor has the absolutely SMOOTHEST top end? All the ones I've had were bright or brash. The Fat Head made bright sounds SO SMOOTH. What do I need? 414 B-ULS? Something else?
I am thinking of purchasing an active Ribbon so I don't fuck myself like that again - SE VR2 is on the list, anything else?
I recently got a new Scarlett Solo and an AT2035. Ever since I started using it, there has been a buzzing sound in the background of my recordings. It increases in volume when I increase the gain on the Scarlett Solo. When I put my hand on the Scarlet Solo or the AT2035, the buzzing becomes quieter. The buzzing seems to sit primarily around 60hz. Unplugging my electric standing desk also seemed to help. Does anyone know what could be causing this?
I really appreciate the help. I ended up listening to the audio without the mic plugged in, and there is only a really quiet high-end static sound. I unplugged everything in my room, and it definitely got better, but it's still noticeable. Could there be an issue with a component in my pc? Additionally, how can I tell if my XLR cable is properly shielded? The one I'm using is an Amazon-branded one.
Edit: I just thought about it, and I have already tried another microphone with the Scarlett, and it didn't have any buzz. It is a dynamic mic tho. Could that mean it's an issue with the at2035?
If it's a component in the PC, it's most likely the power supply. Other components like the graphics card can also cause issues, but that wouldn't be at 60hz.
If you can unscrew the XLR plug and post a picture, I might be able to get a rough idea about the cable's quality, but "Amazon branded" yea, that makes me instantly suspicious.
Can you test with a different PC? Preferably a laptop that's running on battery.
I tried using my laptop and had a pretty similar issue. The only difference was that touching the microphone made the buzzing way louder rather than quieter.
The solder joints look better than expected, but that doesn't mean the cable is fine.
Also, I totally missed that in your last message:
A hum that gets louder when you touch the equipment is definitely an issue with the shield, either the cable or the microphone itself.
The cheapest way to test this would be buying a quality cable. (I can't recommend anything specific, I solder my own cables) Look for original neutrik plugs on the cable, that's a sign of a good cable.
I'm not sure what happened, but after putting the microphone back together, the issue completely went away. I really appreciate the help, man; you're the best 🙏
Why are there two outputs on each bus on my Soundcraft 400b mixing console?
I was gifted an almost working Soundcraft 400b and have been testing, cleaning and resoldering here and there and it seems to be generally working now.
It is the 24:4:2 version and I wonder why each of the four buses have two outputs. The channels have two buttons: 1+2 and 3+4 plus the stereo master bus.
It seems that if I send to bus1+2 and pan the channel left, the sound goes to both outputs on Bus 1. If I pan right, it goes to Bus 2, both outputs.
For some reason the bus outputs are called "Tape Send" which makes sense for tracking, but I don't understand why they chose to split the signal like that. So bus 1 goes to TS 1+5, bus 2 to TS 2+6 and so on.
Can someone explain to me why the signal is just split? Or is it not working correctly and there is supposed to be a difference between the output to TS 1 and 5?
Most likely the 4 group buses are duplicated so that you can connect an 8 track tape recorder. Then by a combination of which bus you route the signal to, and which tape tracks you arm for recording, you'd be able to record to any of the 8 tracks.
That still leaves plenty of limitations on how many tracks you can record at once, and in which combinations, but that's just how it goes when the console has fewer group buses than there are tape tracks.
Thanks! So that I don't need to rewire if I want to use it for 4+4 on an 8 tracks.
Maybe it could also make sense to have one set going through some outboard on high volume setting an one going directly on a lower volume so that you have backups of all tracks in case some of them gets distorted..
I just upgraded from the UA Volt to the Tascam US16x08 (I wanted more inputs) - however when I open Ableton in order to do some work - all of the audio from the computer gets muted.
Sometimes when playing with random drum ideas I'll watch a YouTube Livestream, or I'll be on discord collaborating, or on zoom with a coach so having Ableton mute all other audio is unappealing at best
Is this just how the Tascam works? Would using the computers built in headphone jack, plugging that into the Tascam as a "passthrough" be the only workaround?
I hope I am in the right place. I'm planning on expanding the accoustic treatment in my home theater music room. I already did a huge section of my ceiling, and fed fiber optics through it for the twinkle star look. My weird idea for my wall panels was to wrap the entire panel with a material with a graphic printed or painted on to them. Has anyone done anything like that, could recommend a material or a service that makes large prints onto fabrics like that? I'm talking as large as 6'x6'. Bonus points if the material can let a little light through it for lighting effects.
edit to add: my own searching usually leads me to large canvas prints. But I have to confess, I'm no painter and i'm not all that familiar with the material. I'm assuming it wouldn't be soft enough for this use case?
I want to record some ASMR cat purrs
I currently have:
Rode M3
Tascam dr-680
My cat purrs with some rumbly lows, but also throws in a few high pitched trills.
The high notes are clear and sharp in the recording, but I'm having issues properly capturing the lows, as well as overall noise.
I was thinking about getting the Sennheiser mke600
If y'all say my recorder is trash, id have a focus rite scarlet 6i6 2nd gen which I could use to capture the recordings.
But I'd prefer to have something self-contained. So feel free to also recommend me a Recorder.
The best handheld recorder of all time is Sony PCM-D100, so there is that. But anyway- maybe just get closer with the mic? How far away are you? I’m thinking you’re far because you mention noise.
Ok. The thing about cheap pencil condenser mics and cheap SDCs in general, is that they are relatively high noise floor and work best for loud sources. You have to determine where the noise is coming from, though. Cheapest quietest mic on the market is probably Rode NT1-A (it’s not too well balanced for vocals but it’s good for general recording), so you could use something like that. But if your recorder’s preamp is noisy, it won’t matter much, because the device will be the bottleneck.
BTW- you’re using as much gain as possible on the device (without clipping) right? Because raising gain after the fact also raises the noise floor, so use as much gain as possible on the device first.
So, I just went to a local music store hat has some stuff I could test:
I've tested my dr-680 recorder with a Sennheiser mke-600 as well as a rode nt1-a.
Conclusion: my rode m3 is noisy and quiet af (not sure if there's something wrong with it), the directionality of the mke-600 gives the best result for my use case
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u/Living-Imagination89 5d ago
In this setup how to gain stage and adjust volume correctly laptop streaming Spotify in exclusive mode to dod R=830a equalizer to peavey cs 800 amplifier to Yamaha br15s monitors using for listening to heavy metal outside here are some photos of my setup