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Mic and Mixer dont get along. I need clue

Discussion in 'Recording' started by itsNobi, Mar 8, 2012.

  1. itsNobi

    itsNobi Active Member

    First, let me say: this is not the first time I'm setting up an in home recording studio, but it is the first time I'm doing it correctly.

    So, I have:
    • a pc with a CPU strong enough to handle processing music (I'll soon be switching to a Macbook Pro).
    • 2 mics: 1-MXL770 & 1-Audiotechnica AT2020
    • Alesis IO2 Express
    • All the cables needed to connect all my stuff
    Now, the problems I'm having are varied, and I’m suspecting some of the problems I'm having aren’t really problems at all, but things I'm not used to dealing with, because I'm doing it correctly for the first time, lol; hence, all the writing I'm doing here.

    So, I started off using the Audiotechnica, & found it too sensitive for my recording environment. I set the gain on the interface, and viewed the monitor levels in the DAW, and found that EVERYTHING was picked up; the meters always moved.

    So, I switched to he MXL770, and used the Hi-pass filter, and the -10db setting, and found out a few things that I suppose are actual problems:
    1. Now, when monitoring from the headphone jack of the interface, it doesn’t allow me to hear everything in the mix anymore. That is, before (while recording) if you spoke into the mic, you could hear it the phones. Switched mics; now you cant. And that’ even when you have the interface set to only pick that up (as opposed to it set to only pick up whats coming from the pc; or any combo of the two).
    2. Gain doesn’t do much anymore, as it applies to the mixer. There is a db meter on the interface, and it doesnt react to the MXL770, but it did on the other mic. No combination of settings on the MXL770 allows the meter to react. In the DAW software, you can see that the gain's adjustments do something minute to the meter.

    Welp, I figured that some audio interfaces don’t have a meter at all, so I decided that the meter no longer moving wasn’t an issue since it still converts the signal, and the DAW's meter wasn’t clipping. So, I loaded up Adobe Audition, and dropped an instrument in track one, told the program to use my interface, armed the second track, hit record, and started rapping. I looked at the wave form, and was shocked that at how small it was. The instrumental I was rapping against completely dwarfed what I recorded. I think it's worth noting that the instrument didn’t clip at all. It peaks at around -6db. My vocals peak at around -36db.


    When I amplified the vocals, it was a solid recording. There wasn't any noise, which was a blessing to my ear, but I quickly realized that the amplification caused clipping in two areas.
    I, then, remembered that I hadn't used a compressor at all, so I undid the normalization, I added a bus, applied a compressor, and fed the vocal track thru it. It got louder (duhh) but the wave form didnt change—I expected it to change. The tube compressor wouldnt give me more than 18db on the make-up gain, so it didnt get anywhere near as loud as the normalization did, but it also didnt clip. Then, I decided to move forward. I turned the instrumental track down, then I set the next track to the same compressor bus, and recorded my double track. It came out the same way (small wave form), and sounded the same. I then recorded a stack track, but I decided to add the compressor to the track itself (as opposed to routing thru the bus), and learned that the compression is done post record no matter how/where you set the compressor.

    Then I thought: what should my volume be set to, initially?

    What do I have wrong?

    I’m in a new world. I’m on the other side of the mic, this time. I'm trying to recreate what I’m used to: I can monitor myself thru my headphones as I’m recording. I hear myself loud and clear. Also, playback thru my headphones sounds just as loud and crisp to as when I was recording/monitoring myself thru the headphones. Seemingly, no adjustments made (for playback). just record, stop, playback, record, playback, then send to mix.
    Instead, what I get is, no active monitoring...and low vocals. So, to even playback just to see what you have, you have to go thru every setting under the sun, to no avail.

    I see that, thru all the reading I've done, there is still something major I’m ignorant to, and it is information between the "purchase/set up" phase & the "hitting the record button" phase.

    Any help would be greatly appreciated.
  2. gehauser

    gehauser Active Member

    Well, the MXL770 and the AT2020 have nearly the same sensivity (15mV/Pa and 14.1 mV/Pa, respectively), so if you used the same gain on your interface for each, then they should be recording about the same level in your DAW, assuming the mics are operating as they should.

    Keep the mics set flat for now with no pad, and cable them up to your interface XLR inputs. Then on your interface, make sure you have mic/line switch set to mic, and the 48V phantom power on, then adjust the gain on your interface until you get the proper recording levels (peaks in your DAW should be around -18dbfs, not -36dbfs). The recorded waveforms should be medium-sized, not large not small. If you cannot reach this recording level without hiss, it means your interface's preamps are simply too weak or too cheap to reach normal recording levels. This is often the case with the pres on cheap interfaces.

    Make sure you are getting results as described above, before you worry about playback issues. Playback of a mixed down wav or mp3 typically sounds much lower in level than playback of recorded tracks in your DAW. Part of mastering is the process of raising levels of the stereo mix to modern listening levels - read up on the loudness wars for more on this.

    This guy seems to have your "mic too loud" problem until he switched to ASIO drivers:
  3. itsNobi

    itsNobi Active Member

    Well, I definitely started off with the ASIO drivers, as Adobe made me aware from the start-up that it prefers ASIO. So, it's good to know that I'm on the right track there. I'm guessing that I should not be alarmed that I lost functionality, as it were, of direct monitoring on the interface when using my MXL770?
    Also, I won't be able to get back to my studio set-up for a few days. That being said, assume I'm able to adjust the gain so my vocals peak at around -18db, I still have two concerns:
    1. Should I be turning down the volume of my instrumental track (which peaks at around -6db), or should I be compressing my vocals (even turning up the volume on my vocals)?
    2. How do I recreate what I'm used to? That is: I can monitor myself thru my headphones as I’m recording. I hear myself loud and clear. (more important bit --> )Also, playback thru my headphones sounds just as loud and crisp to as when I was recording/monitoring myself thru the headphones. Seemingly, no adjustments made (for playback). just record, stop, playback, record, playback, then send to mix. As opposed to: no active monitoring...and low vocals.
    Is it a piece of equipment that allows this, or just a function of the DAW via plug-in?
  4. gehauser

    gehauser Active Member

    If you lost direct monitoring, you would lose it whether you used the AT2020 or the MXL770. That is an interface issue, not a mic issue. The fact that you think it is a mic issue tells me that your direct monitoring probably still works, but you did something different when you recorded with each mic.

    Does the Alesis come with routing software? You would need the inputs routed directly to headphones.

    If not, check the Alesis manual. I don't have the Alesis IO2 so I cannot say what is going wrong with your direct monitoring. Somewhere the signal is not getting routed properly, or you have a gain setting too low.
  5. gehauser

    gehauser Active Member

    Well, it is nice to have all tracks peaking roughly at the same level during recording, but it is not absolutely necessary.

    Now, if I am playing back a track while recording another track (overdub), I often turn down the DAW fader on the previously recorded track so I can better hear the track I am recording.
  6. itsNobi

    itsNobi Active Member

    I have to route it thru ASIO or thru a bus/send. I also found int he Audition manual, that the monitoring options are set via the DAW's options.
  7. itsNobi

    itsNobi Active Member

    Thank you. Thanks for all your help. Now, I have just one last question, to help me clear my last issue: as, I've stated, I'm new to this side of the mixing board. So, as an atist, Im used to telling someone, "I wanna record this hip-hop track," and stepping in the booth, putting on the head phones, waiting...and waiting...(maybe being told to speak into the mic)...then hearing my track, then recording. What, exactly, is the engineer doing while I'm waiting (before loading track, of course)?
  8. gehauser

    gehauser Active Member

    Well, he must select a preamp, connect the mic to the preamp, connect the preamp to the console and audio interface, create a new track, route the preamp signal to a track, arm the track for recording, check for signal, figure out what's wrong if he gets no signal, fix it, and adjust preamp level as you mic test. Then he needs to load the backing track(s) and set relative levels so you can hear yourself sing over the backing track.

    He probably does his connections via patchbay and console, so there may be more gain staging beyond my simplified description.

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