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About anothertom

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  • Gender
  • Location
    London, UK.
  • Interests
    All things AV, Rock 'n' Roll, DnB, making things go bang.
  • Occupation
    Event Technician


  • CPU
    Intel Xeon E3 1231V3
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    Asrock H97m/itx ac
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    2x8GB HyperX Fury 1600MHz
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    Sapphire R9 280X tri-X OC
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    Coolermaster Elite 130
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    500GB Samsung 850 EVO, 1TB WD Blue
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    Corsair CXM 600W
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    Asus VS228HR
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    Corsair H55
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    Corsair K70 Brown
  • Mouse
    Logitech G602
  • Operating System
    Windows 10

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  1. Something like a scarlett solo or 2i2, a Presonus Audiobox or a Behringer UMC22/UMC202 would suffice, at varying price points. You can directly use these with instruments or with microphone inputs. You may want to add a inline with your guitar/bass to prevent stray interference. When you say: Do you mean that you want to record the tone the amp produces or record the actual output of your amp? If you want to just take the tone/eq from the amp, then you can use the send from the amp (right hand side on the front) which should be a post-EQ/FX output. If you want to record the physical output of the amp (and speaker you've connected it to) then you'll need a mic to record it with. A typical mic to use on guitar would be a sennheiser e609 or e906, for bass a shure beta 57A.
  2. There are a few things it could be. A ground loop between the computer and speakers, where there's a difference in the absolute potential of the ground at each speaker/computer. This would be a very consistent 50/60hz buzz. Alternatively, internal EMI, where load on components (cpu/gpu mainly) will cause interference directly on the output traces. Alternatively, external interference. where you've run the unbalanced cables too close to power cables, which will also introduce EMI on the signal. The best way to eliminate this would be to get a cheap USB/thunderbolt/firewire audio interface with balanced outputs and go directly into the speakers. This will avoid having any analogue audio internally to the computer, and use a balanced connection externally. (e.g. UMC22 / UMC202 / scarlett solo / scarlett 2i2)
  3. I will reiterate that a "DIY" solution to this isn't going to be as turnkey as a paid solution. So physically, what I'd do is have the presenter (as we'll call them) running the meting from a desktop with the screen going through a splitter which also feeds either a large screen (55'' or larger) or a projector. The other people in the room observe what the presenter is showing through the larger screen. Ideally everyone is around a single table so something along the lines of this where the larger screens are easily view-able by everybody. The presenter should be sat at the table also. The desktop is running screen capture software (OBS) which records to the suitable format. A single omni-directional mic is placed central to the meeting going into an audio interface plugged into the presenter's desktop (alternatively, see below). OBS records both computer audio and the room mic. Short test recordings should be made to ensure consistent levels between all people. Once a recording is finished you can re-encode the output file to a more compressed format (either adobe media encoder or a free solution like handbrake, MPEGStreamclip or ffmpeg). One reason why i suggest a single mic, is that once you start micing people individually, then it becomes necessary to mic everyone individually, to get a usable end product. That said, giving the presenter a lapel mic and then a single 'shout' mic for the other participants, which can simply be something like an sm58, will probably be cheaper than a decent omni mic. In which case i'd go for a scaralett 2i2, a shure MVL or CVL for the presenter, then either an sm58 or pg58 on a short stand for the others to use when needed. This will require a knowledgeable person for each session, and you'll want to create as much of a permanent setup as possible, otherwise kit will start disappearing and breaking.
  4. anothertom

    MIDI Mixer/Matrix

    So, you want to take the midi output from any of 3 computers and route it to and of 6 synths. Then take the output from those 6 synths and send them to a single pair of speakers? There are midi matrixes which allow you to route a number of inputs to a number of output, however these rely on a host device (pc) to configure and manage the routing. For example this 8 channel interface, or this 10 channel interface. Each allows routing any channel to be output from any other channel. (but make sure the device will do exactly what you're trying to do before you buy it). But to change the routing you need it connected to a host device. For the output from the synths any mixer would be fine, so long as you've got enough channels to handle the number of stereo (or mono) sources you've got.
  5. For recording you want to be going to a lossless format. This allows the best opportunity for post-processing over a lossy format. Due to the real-time nature of recording it's also easiest to go to an uncompressed format (i.e. WAV) over a compressed lossless format (i.e. FLAC). As for very high sample rates and bit depths, not sure what supports them but at a guess you'll end up with needlessly large files.
  6. A big factor in this, which you didn't clarify, is that if you need to be sharing from multiple screens or is it one person leading the session who needs to share their screen? There are commercial solutions to this, for instance zoom, but it is possible to do what you require DIY, but obviously there won't be a company to call and complain at if something stops working. If this all takes place on a single computer (or laptop or whatever) then capturing that screen and using a decent omni-directional mic centrally should produce reasonably usable results fairly cheaply. Then again, it depends on what the end audience is expecting quality-wise from the final product.
  7. Try out voicemeeter / voicemeeter banana.
  8. In this case i'd agree with @AAJoe and @rice guru. You don't need to spend a massive amount on a microphone to get a decent sound, and an AT2020 is going to be fine for anything going through youtube's compression. If there's a specific aspect you want to change then we can suggest specific fixes. If the sound is the most important thing to your content then you need to address the whole signal chain as well as the physical environment you're using. This includes the location and environment you record in, is there a large amount of background/external noise? do you get a very reverb-y sound? do you get a load of mechanical vibration coming through the mic? Most of this can be solved cheaply through isolation, damping, acoustic treatment, a little bit of processing/signal flow and proper mic technique. Since you don't mention an interface or mixer, i'm going to assume you're using the usb version of the 2020? If so then it may be worth flipping it for an xlr version or even up to a AT2035 or if you really want to a SM7b, but you don't need to be spending hundreds on the mic. Then it's worth thinking about pre-amps, compression and the interface/recorder. This depends on the mixer or interface you're using, you want to be looking at a focusrite scarlett / presonus audiobox or better in terms of pre-amp quality for an interface, and i wouldn't really suggest using a small usb mixer for anything serious. If you want a dedicated compressor, then look at a pre-amp/processor like the dbx 286, which will give compression, gate/expander as well as a de-esser. If you want to record a raw vocal and process it later then there are software solutions available. There are standalone EQ's available, both graphic and parametric. You can look at used equipment sites for most of this without much risk, but the interface is probably worth getting new.
  9. From what I can see, it's only that the wire going to the driver has been cut? If they were made properly, they'll be different colours, so you can simply solder the cables back together, assuming they're not too brittle to bend. If you're not able to simply reconnect the existing cables, then you can just run in a new set. As you don't really want to be trying to remove the stubs from the driver (as they look to be glued/resined to the driver) you'll still be connecting to the existing stubs. In this case you'll need to run the new cable back to either the crossover or the input connection (as i'm assuming these aren't active). Ideally you'll get the polarity right, so it's acting like the other one, otherwise you'll end up with some interesting phase issues.
  10. Sort of, you can choose which extra 2 channels you have on top of a 5.1 system, but not reroute the LFE (sub channel) to a different output, from what I can see. lol...
  11. Were they, perchance, being sold by a slightly dodgy looking gentleman in a white van? Otherwise, make and model, and we'll see what we can do. What you're looking for is generally known as a decoder, and as you've found, they're much rarer than a traditional receiver and annoyingly more expensive.
  12. To be able to process each input separately, you're going to have to use a program which can access the individual channels on the interface. Most live processing programs won't be able to do this, as processing multiple channels in real-time can get quite heavy. The only one i would suggest trying is voicemeeter banana, but i wouldn't be very hopeful. What you're really looking for would be a physical mixer, preferably with a USB output/ built in interface. I would ask why you decided to go with an 18i8, as it's pretty overkill and the wrong tool for the job if you only want to use two mics. If you've just bought the 18i8 new, you could have bought something like an X-Air XR12, which would provide as much processing on multiple channels as you could want, then mix it down to a final stereo mix to take into the computer.
  13. Well that's the wrong way round to start with. You should use the channel gain to get the microphone to a usable level (i.e. to the point where you are just not clipping the input at your loudest.). Then use the channel and main level to get the final level you need. Make sure to position the mic correctly and that you're speaking into it properly (it's side address rather than end address). You probably also need to use the channel EQ if you haven't already: insert the low cut filter (just below the input connectors), then you'll want to further take out the low frequencies (so it points somewhere between 9 and 10 o'clock) and probably roll off the highs slightly (to about 10 - 11 o'clock). Ignore the compressor until you've got it sounding right.
  14. What do you need to get the audio into? i.e .what hardware is the audio going into? is it going into a PC running a piece of software or some other system-in-a-box... Also worth considering is budget, style of microphone, as well as any region based regulation over wireless equipment.
  15. That is not what that button is. It simply puts a high pass filter over the input, it doesn't switch it between mic and line level inputs. Because that is the USB input channel. The USB output is taken from the master bus, after the master fader (I think, been a while since looked at the schematic). @DOutram if you haven't given up on this yet, you probably just need to turn it all up. You can easily give the mic channel more gain before clipping, and the master should be set higher. (You'll want to turn the channel level down slightly though). if you want to use the USB input to playback, then you need to select it as the output device in macOS, then on the mixer, the USB channel wants to be set as the 'USB' input and not the 'LINE' input and the destination 'to MON' which will route it to the monitor bus (which is split to the headphone output) and not 'to ST' which would send it back out the main output (which if you're recording will create a nice feedback loop). Depending on what you're doing and how your outputs are connected, you may want to have that button in either position. feel free to go back and read my previous post in this thread to learn how to properly set up the channel and mixer levels.