Mixing FOH mics with sbd feed for recording, delay?

Discussion in 'Tracking / Mixing / Editing' started by Terrapin, Oct 21, 2007.

  1. Terrapin

    Terrapin Guest

    Hi all,

    A question derived from the results of my last post. I had last set off to mix a show in a theater and do a multi track recording simultaneously. So I got a DI source recording track from the instrumental acoustic guitar performer as well as an XY SM81 stereo source from the FOH position. Ok so I also got the sbd outs (post Drawmer comp./mackie/onyx). Its a total of 6 tracks, 1vocal mic, 1 di, 2 main mix, 2 stereo mic at FOH.

    My question is regarding the delay time between the DI feed and the FOH mics as how it relates to mixing for a CD. I had first thought If I account for the speed of sound and then compare to distance of my board from the stage and set that same time in my DAW delay it would equal out. Hmm, Perhaps I'm doing it wrong. Anyohne else ever dealt with this?

    Although mixed straight together and the XY mics set much lower than the main source the recording seems to have a nice natural delay and reverb. How do I know if this mix is deep in phase cancellation?
     
  2. Boswell

    Boswell Moderator Distinguished Member

    Yes, we deal with this all the time, not only in live work but also in the studio. Others may have their own techniques, but two examples of when I use delay are:

    (1) Close miking an acoustic guitar when also taking a pickup signal. The pickup signal has no acoustic distance to travel, so will get to the board ahead of the microphone sound. For 12"-18" microphone distance, you need to delay the pickup by about a millisecond to reduce cancellations in the upper-mid frequency range.

    (2) Using flanker mics on a large coherent ensemble (e.g. choir), where the main pickup is from a stereo pair centered and set back from the ensemble. The amount of delay needed is this case may be slightly more than just the acoustic travel time difference so that the flankers give the required width and coverage without giving peripheral prominence or creating a central void.

    When recording in an acoustically live venue with lots of natural reverb, you may be hard pressed to hear the effect of adding the theoretically correct amount of delays. The big thing is to mix it so it sounds right, but remember that delay is just one of your mixing tools like EQ or compression, so don't overdo it, except for special effect.

    At the performance, was all the guitar sound acoustic or did the player use an amplifier?

    BTW, I don't see what the distance of the board from the stage has to do with acoustic delays.
     
  3. JoeH

    JoeH Well-Known Member

    You can't really know, until you listen to it. Sum it to mono, toggle one or the other off and on; listen for what you lose, listen for what you may gain. Any phase cancellation like you're concerned with may not be very linear anyway..some freqs will behave better than others. You may have to drag the delayed stuff forward a bit on the timeline if there are any serious problems. You may want to roll of the low end (if you haven't already) on the delayed stuff, so that there's less problems in the bass regions, where phase shift would be more pronounced.

    If it sounds good to you, then it is.

    I just did a three day festival (mostly acoustic music) with something similar. In our case, we had isolated recording splits from anything that went to the house or the monitors.

    I also put up my own pair of L&R audience/ambient mics, looking OUT into the house (not at the stage) to capture applause, sing-alongs, etc. These were about five or six feet forward of the front of stage line (due to the auditorium's layout, etc.) and well underneath the PA Mains, which were flown overhead.

    The onstage stuff was SM58's & 57's' for vocals and instruments and DIs or individual pickup mics for the various players that had their own. (Usually AT instrument mics, etc.) I'm always glad to have more than just a DI for any acoustic instrument / guitar. (A plug up its a** always sounds compromised to me....)

    Back here at mixdown in the studio, all of the onstage stuff is understandably dry and sterile - it could have almost been a studio recording. The Audience mics are great for MOST things, but not all, so it occasionally has to be dialed down for some tunes, etc. I also add some "live" and "Hall" reverb, depending on the source - vocal mic, instrument mic, percussion, etc.

    The point being that there's no one fix for this, and your audience expectations (hell, even your own) may not be happy with what you get from most of the X-Y mic pickup from the FOH position. (Visuals account for a LOT of the enjoyment at live shows, the FOH sound can actually be worse than you think, esp when you hear it later, without the visuals, back in the studio.)

    I used to go that route back when I had nothing else execpt the stereo board out and two mics at the mix positition, but it was always a compromise, and never enough to get things sounding as clean/live as they should with multitrack combined with ambient mics.

    As always, YMMV.
     
  4. sheet

    sheet Well-Known Member

    Yeah, there is no rule aside from some summing tests on-site. I wouldn't mess with the math, because there is no way to estimate the latency induced by your electronics before it gets to the speakers.

    Send click track, etc through the console (and PA) to one channel of your recorder and then record the room mic on the other channel. Listen to the difference. Insert a delay on the console send to the recorder. Adjust the delay until you get a match. You can try nulling the two by flipping the polarity on one channel, but I don't know that you will be get it exactly nulled, due to the room.
     
  5. bent

    bent No Bad Vibes! Well-Known Member

    Which DAW are we talking about?

    If you have it in digital land then you should be able to see the phase difference, a peak up and a similar peak down = cancellation. Zoom into a peak on the main track and compare it to the l/r mics' peaks at the same relative time. Move it in phase / in time and see what happens.
    If you want it to have some depth I suggest moving the waveforms a few milliseconds at a time until you come across your desired sound.


    -Hey Joe, you could take those audience mic tracks and slam them up and down and get that "Cheap Trick audience as a percussion track" sound!
    :lol:
     

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