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  1. #11
    Nutcracker, sweet's Avatar
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    So, I thought this was cool - a little while ago, I was on a Skype call over a VPN. It was choppy as hell, so I joined the call from a second PC, dropped the call from the VPN'd PC, and didn't miss a second of audio. Everyone should have a mixing board.

  2. #12
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    Quote Originally Posted by submessenger View Post
    LOL. Sometimes I sing. I home-office, these days. Spend much of the day on Skype and other teleconf technologies.




  3. #13

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    Nowadays, USB-plugged mikes are a thing and works well for skype call/podcast and things like that. But since you seems to have a lot of fun with your mixer, I would recommand you to get a SM58 which is probably the best voice mike you could get for under 100.

    And because you seem interested in getting good voices tone for your skype call, be aware that having many mikes is not a ideal solution, because you will get the bad tone and reflections of an untreated room as ermghoti said. Plus, you have to get the phase relationship right or the mikes are going to sound worst when put together. Close miking is definitely the way to go.

    If you want to take your voice up to a whole new level, you might consider EQ'ing, compressing and de-essing your voice but we are already digging into more technical stuff there that only audio nerds care about

  4. #14
    Nutcracker, sweet's Avatar
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    Quote Originally Posted by Karish View Post
    Nowadays, USB-plugged mikes are a thing and works well for skype call/podcast and things like that. But since you seems to have a lot of fun with your mixer, I would recommand you to get a SM58 which is probably the best voice mike you could get for under 100.

    And because you seem interested in getting good voices tone for your skype call, be aware that having many mikes is not a ideal solution, because you will get the bad tone and reflections of an untreated room as ermghoti said. Plus, you have to get the phase relationship right or the mikes are going to sound worst when put together. Close miking is definitely the way to go.

    If you want to take your voice up to a whole new level, you might consider EQ'ing, compressing and de-essing your voice but we are already digging into more technical stuff there that only audio nerds care about
    It's not just about the mixer, it's about the multiple devices that I want audio to and from. So, USB mic is out of the question. I considered a VHF solution, but that got very expensive very quickly. IIRC, this is how they do things in environments like NFL, NASCAR, etc.

    I *am* interested in the audio nerd stuff, though, so please feel free to talk more about it.

  5. #15
    Nutcracker, sweet's Avatar
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    Following up, on this... I got the phantom in (I didn't previously notice the mixer had switched phantom built in, RTFM FTW). As predicted, it did nothing for me. What made a HUGE difference was using the XLR->XLR cable from the mic directly to the board. I was previously using XLR->1/4" mono, then XLR->XLR(phantom)->1/4" mono. Line noise has dropped to acceptable levels, and there was apparently a lot of mic attenuation, as well. Crappy cable, I guess. The phantom cost $19 and came with the XLR->XLR, about the same as a "good" XLR->XLR cable would be by itself, so I don't feel like I completely wasted the money.

    Now, I am going to get a better mic, at some point, soon. This one is omni-directional; I'd like to get a cardiod to help mask out keyboard clicking, and preferably also with shock absorber because the mic does pick up vibration from the desk.

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    Quote Originally Posted by submessenger View Post
    What made a HUGE difference was using the XLR->XLR cable from the mic directly to the board. I was previously using XLR->1/4" mono, then XLR->XLR(phantom)->1/4" mono.
    AUGH! God. No. Never do that. If you have to go XRL->1/4" without a direct box, you need the little plug adaptor with a transformer in it. They're like $15-20 or so.
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  8. #18
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  10. #20

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    Quote Originally Posted by submessenger View Post
    I *am* interested in the audio nerd stuff, though, so please feel free to talk more about it.
    Ok, are you by any means familiar with the usuals audio tools? (I mean EQ, compression) Because once you get your mic plugged in and placed right (neither too close nor too far) with a decent level (loud enough to be heard, but not loud enough to clip), you are not going to be able to do much without using these things.

    First of all, I don't think Skype have a effect chain build-in, and your mix doesnt have a build-in compression on the channels. Plus the 3-bands Eq on the channels are not really going to do the trick for what we are trying to achieve, we need a parametrical EQ. So the solution I see to achieve that is using a Digital audio Worksation to get your audio from your mixer to your computer, then to your DAW, then to skype. If you have no DAW, you can have a look at Reaper which is cheap and widely available. The trial licence last forever, you just have to click on a message urging you to buy the thing

    https://www.reaper.fm/

    Once you got your audio signals all routed, you can mess up with the whole thing to a new level. With Reaper feature's, you can do more than what you could with your mixing desk.

    But if you want to improve your voice and get this "professional broadcast-sounding voice" you need to work a bit on that. You will have to pick the stock compressor first. The goal here is to reduce the dynamic range so it will be easy to hear very clearly everything you say. This is also a point where you will bring up the little sound that you might not want like the keyboard sound you've mentionned before. This is why you need, as you have noticed, a cardioid mike like the SM58 so you don't record too much of unwanted noises in the first place.

    You should try these settings on the compressor : a 2.5 to 1 ratio, an attack time of 2 millisecondes, a release time of 300 millisecondes. You will lower the threshold of the compressor low enough so you can get about 5 to 10 decibels of reductions. You will then add enough volume with the make up gain so you don't hear a big volume difference whether the compressor is bypassed or not. If you start hearing the little details of the very bottom of the throat, like when someone is crying or when a singer sing a very emotional song, you are going a bit too far with the compression and you should back the threshold up a bit. There could be some others parameters on the compressor like Knee, auto-release, just don't bother with these, they are not interesting for what we do.

    Now let's talk about EQ. First you will pick up the stock parametrical EQ, and use a high-pass filter to get rid of most bass stuff below 100HZ to make your voice a bit more clear (there is not much useful stuff going there when it comes to human voices) I assume you have a average-pitched male voice, not much higher or lower than most guys around you. If that is the case, you need to use an bell-type filter to cut some frequencies around 200-300HZ where the "muddy" aspect of the sound is to be found. You then can make a similar cut in the 500-600HZ area, where you can hear the very nose-sounding frequencies. All of this is very classic EQ moves when it comes to voices, and that is what most professionals mixers will do whether we are in a broadcast or music context.
    Once you are done with that, you might consider boosting a little bit around 1.5 Khz to make you voice more clear and focused. And if you feel like your voice sound dull through the mike, boost a bit around 10Khz. These two last eq moves could be completely unecessary, trust your ears.
    By the way, don't be afraid to boost a lot to find the frequencies you are looking for, so you can adjust the boost or cut afterwards. Also, when I say "cut", I don't mean get rid of everything in the aformentionned spot, I mean reduce the thing by 2-4 decibels, which is usually enough to make a noticeable difference. Same goes for the boost, 2-4 db should do the trick.

    And to finish this, a thing to consider is De-Essing. This little thing is here to control the "S" that can sound extremely harsh and unpleasant through a microphone. It could not be needed, but in case you do need it, just put on the fx chain last and lower the threshold until the esses are no longer harsh. 15db of reductions could be not enough to get a convincing result on this specific subject. The only thing that we should really pay attention to is when you feel like you have a lisp. This is the signal that you are deessing too much and should back up the threshold.

    With all of this done to your voice, it should sound more clear, more focused and pleasant to listen to.

    Audio Engineering hack n1: A/B testing is important ! If you bypassed one of these plugin and you feel like something great is suddenly missing, then you're doing it right. On the other hand, if bypassing the plugin make your voice sound better, then you have probably not dialed in the plugin well and should tweak it a bit.


    Audio Engineering hack n2: With all of that being said, having a signal that doesn't clip is also super important. If the meter of your voice's track is getting red, that mean your signal is too hot and you should back it up enough so it doesnt clip

    Feel free to ask any further questions on that if you want more informations

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