How to Fix Latency with Dual Boom Mics

Joshua Milligan

Joshua Milligan

Well-known member
Hey friends, feel free to correct my terminology here as I’m a camera op and not a sound engineer, but I’m looking for help fixing a latency (if that’s the correct term) issue I’m having when running dual boom mics for interviews.

I’ve been contracted to film a lot of interview videos for numerous types of doctors lately and cannot use my Sanken lav mics for the audio because of the scratchy clothing they wear. The lavs just pick up the sound of the horrible doctor clothes rubbing together (not rubbing the mics, just the loud clothes themselves), so overhead booms are a must. That being said, I typically like the audio from a boom better anyway, so that’s not a huge problem for me.

Where the problem does come in is when filming dual interviews of two doctors at one time which is something a plastic surgeon has been asking me to do a lot lately. The mics I’m using for this scenario are a stereo matched set of Sennheiser MKH 8040 cardioid mics. Because I don’t have a sound op (not in the budget), I am running these mics stationary on boom poles with stands.

The audio sounds great except for the (what I call) latency I’m getting when mic 1 picks up what mic 2 is recording when the person under mic 2 is speaking (and vice versa). In post, if I shut off one mic, the issue goes away and the audio sounds great, but when I combine the two together at the same time, that’s where I pick up the hollow sound that’s created from one mic picking up the other’s audio.

As of now my solution has been to go through and key frame the two audio tracks so that when one person isn’t speaking, their mic is keyed out. This works, but it takes forever and considering the amount of videos I’m doing, there has to be another way. And I understand that I could just use a “Reverb” effect and message it to make it sound better, or just use one cardioid mic and split it between the two speakers, but because I have these really nice mics and because I have the time during the shoot to set them up, I want to find a way to do this right.

Do any of you have suggestions on how I can work trough this more easily in post? If so, it would be much appreciated! Again, I’m not a sound guy, but I do appreciate good audio and am willing to do this to the best of my abilities if someone can help me find a better way. Thanks!
 
It's most likely phase cancellation. Try pointing the mics away from each other a bit and/or changing the distance between them.

I've had trouble with it in the past until I did the above (but I never really knew what to do exactly with each new set up). Eventually I started recording into a F6 (with the AutoMix on) and haven't had any issues since then. (Maybe just lucky.)
 
The hollow sound is phasing because of the difference in distance between the two mics. Using only one at a time is the best way to solve the problem. You could also keep them in stereo that should minimise the phasing. The one thing you absolutely do not want to do is mix them to mono, that will guarantee phasing issues.
Flipping the phase won't fix the issue, flipping the phase of both gets you back to where you are now.
You might be able to do assuming the people are not moving around is to phase the tracks so they are in sync. That "may" work. It would put one track slightly out of sync with the picture but I doubt it would be enough to be noticable.
Depending on what NLE you are using you may not be able to manually shift the track by a small enough increment. In that case you either need to do the shift in a DAW or use something like PluralEyes to sync them. Resolve and Premiere have a syncing feature that will work also.
 
Lots of ways to mitigate the issue. Volume envelopes can be used in lieu of a gate (or expander), which can be triggered unintentionally. Or you can cut up the timeline and eliminate the mic when it is not the primary.
Of coarse it all depends on what NLE you are using and if audio plug-ins are an option.
 
I definitely figured eliminating the mic not in use would be the best way, but I wasn’t sure if there was a software or plug-in out there that could do that for you automatically in post. Something that could detect the primary mic and turn off the other automatically so that you could run the entire interview through the software and have it cut that up for you. I’m editing in Premiere and use Audition for all of my audio.
 
I'd try in order:

1. Flip/invert phase of one channel: use Audio Effect Invert.
2. Try to perfectly align the two audio tracks: select them both and right-click Synchronize (audio).
3. Shift one track a few samples until issue resolved (right click on timeline cursor, select Show Audio Time Units, then Win/Command-Arrow to shift by samples). Will start to sound flangey if too much.
4. Automatic ducking: https://helpx.adobe.com/premiere-pro/how-to/automatic-audio-ducking.html . You'll need to do this twice (2 pairs of audio), where you duck the opposite in each case.
 
A noise gate is automatic. Many of them available for purchase and freeware. AA problem has one already. They ain't perfect though.
 
Joshua Milligan said:
I definitely figured eliminating the mic not in use would be the best way, but I wasn’t sure if there was a software or plug-in out there that could do that for you automatically in post. Something that could detect the primary mic and turn off the other automatically so that you could run the entire interview through the software and have it cut that up for you. I’m editing in Premiere and use Audition for all of my audio.
Click to expand...

That's what that AutoMix feature mentioned above attempts to do during production, but it would be nice to have it in post.
 
The reason I would not even try flipping the phase is that what you have is a latency issue not a phase issue. Because it's a timing thing the phase difference between the two channels is frequency dependant. Lower frequencies are less out of phase than higher frequencies so flipping the phase doesn't help and since the mics are so close it might make it a lot worse.
It's basically no work to try it but I think that you will find it's either worse or make no difference. The phase flip is there to deal with electrical and wiring problems like a miswired XLR cable more than anything else.
 
Phase- try putting two mics very close together, e.g. a figure 8 and a cardioid, perpendicular. Then process in post as mid-side. Try flipping the phase on either mic and see what happens.

OP mentioned matched pair- close together into same recorder = possible phase issues = flip and/or shift as first tries.

Sounds like auto-ducking (mic A turns off mic B, and vice versa, 4 tracks) will provide the sound they are looking for (vs. manually key framing).
 
OP never said they were close together. He said they were a matched stereo pair. It doesn't mean he used them that way. He also mentions boom poles and stands (plural). If he did use them as a pair, there there is no reason to be using the two microphones. He can use one and get rid of the problem altogether. If the mics were close together then he didn't mic the talent properly. If he is using two mics, they should have been as close as possible to the talent to get the sig/noise ratio down as low as possible to reduce the problem, which is most likely what he did.
 
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Auto-ducking looks promising. Nice. I'm wondering if it will work if you apply it twice to the same two channels but in reverse. It seems like it would.
 
Yeah I think double auto-ducking would work nicely. A noise gate and/or expander could also work as mentioned earlier. The question for each method is effort and transition sound as the effect goes on and off. Given the large jump in levels (high SNR), it would seem double auto-ducking would produce the cleanest solution.
 
This is where the Dugan Automix feature on the Sound Devices recorders is worth the cost of the recorder itself. Waves seems to sell a Dugan Automix plugin for post work, but it is a bit expensive and requires you to process the audio.

If you are doing much of this type of work, I would highly suggest buying a recorder with either the Dugan Automix or a general type of noise gate automix built in.
 
Good point cpreston- the Zoom F6 supports Auto Mix (Menu/Input/Auto Mix). I've been using the F4 and now the F6 in the studio with the Schoeps CMC641 and Audix SCX1-HC typically 6 or so inches apart, and so far haven't had any phase issues (typically as stereo; listen in mono to check for phase issues). With the mics much farther apart (feet to many feet) no issues in mono though I typically manually duck the non-active mic (narrative, not interviews).

For interviews, if more than one person is talking at the same time, automatic double ducking for the OP use case would need to be turned off and manually ducked/gated for those sections.

Dugan Auto Mix algorithm is free for Reaper (which is free to evaluate and very reasonably priced (I bought it years ago as a Pro tools replacement)): https://forum.cockos.com/showthread.php?t=173289

See also: https://www.waves.com/plugins/dugan-automixer
Hardware: https://www.dandugan.com/products/
 
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Joshua Milligan said:
Do any of you have suggestions on how I can work trough this more easily in post?
Click to expand...

You don't say what software you are using? If it's FCPX because of it's no mixer no busses approach and the way it works with audio clips you may find Audified's "Auto Duck" plugin may do the job with a bit of working around. It's designed to duck music under VO but you may be able to use two dialogue tracks. Have a look.

https://www.youtube.com/watch?v=lQLtEKAdgaQ

In those sorts of situations for speed and pure simplicity without having to work with side chain compressing/ducking I do the auto ducking in Vegas Pro where I can just select the two audio tracks and run a script called "AdjustVolume" that will auto duck a track named "Music" under a track named "Narrative." These tracks could be exports from another NLE not tracks necessarily edited in Vegas Pro. This is a one click function after you have put volume envelopes on both audio tracks. It works perfectly well even if the "Music" track isn't music but another voice audio track. By default the script will duck the Music track by -12dB but if the script is altered this amount of "duck" in the level can be specified to whatever dB you want to duck by, 0dB if you wish. Also the the attack and release times of the duck can be adjusted to suit. For that matter the "Music" named track could be re-named "Narrative 2" or "VO2" or whatever as long as the track name you want to use is entered into the "AdjustVolume" script as the script looks for the track names so as to know which track you want 'ducking.' I often use auto ducking where I have a Boom and a Lav on a single interview and want to use a mix of both audio tracks or automatically kill one track down to 0db or whatever.

If you are on Windows and are prepared to render out .WAV files of your audio tracks once you have completed your edit you could process these files in the free program "Audacity" that has a very elegant simple and useful auto ducking capability which could be used in a couple of ways to achieve your goal. Once your ducking is done in Audacity you could just export your new ducked tracks as .WAV files and drop them back into your edit master timeline and use them for the final program render/export after you muted your original 'un-ducked' files on the timeline.

https://www.youtube.com/watch?v=gdevgTG37BM
https://www.audacityteam.org/download/

Side chain compression/ducking in the way most DAW editors would normally "duck". A much more time consuming way of doing it over an automated script which as I say is literally a one click application to run the ducking script.

https://www.youtube.com/watch?v=ruYQr8j55TA

Chris Young
 
To clear up some assumptions When I said "close" I meant "close" not paired mic. He is filming an interview/ conversation they are not going to be very far apart, JCS just made the assumption. IF it were paired mics then the latency would be basically nonexistent. The "flip the phase" to fix (whatever) is something that gets said a lot by non sound people. The "button" is in the DAW because it is on most mixers. It is used on mixers because in live situations you get the occasional mis wired cable and there was a time when outboard boxes were reverse polarity depending on where and in what industry they were used. East Coast and broadcast had a bunch of pin 3 hot gear and west coast and film was almost all pin 2 hot. The "phase switch" is for reversing polarity not for fixing phase issues. But folks who don't know a lot about sound get told by others that don't know a lot about sound that it does all kinds of magical things like making noise disappear, and fixing pretty much everything. It's a placebo, except for flipping polarity.

The case of M/S recording is a great example. It requires that the capsules are as close as possible to eliminate phasing in the audible range and then it uses polarity flipping to remove the center from one split of the side and add the center to the other split of the side to create a phantom "stereo" signal. Polarity flipping is what makes it work so of course adding another flip will change the output, should swap "left" and "right".

Polarity is NOT his issue, latency is his issue. The non "on mic" speaker is traveling a very short distance farther to the mic than the "on mic" speaker. If they are not moving around then that latency should stay fixed. So you might be able to shift a track to be in sync with the other, though since the latency may be different in one direction than the other you may still need to cut back and forth, which is the better way anyway.

BTW for those reading that may not get that core difference I will do a little explanation. If you do get the difference you can skip this paragraph.
Frequency is how many waves are complete in a fixed amount of time. 1Hz is one cycle per second. Phase is the relation ship of two waves. So if I start a second wave 1/2 way through the first wave and both are 1Hz then the two waves are 180 degrees out of phase (essentially what the polarity (phase) invert switch does). If you play those two waves together as Mono they will cancel eachother out and you won't hear anything. If you play them as "stereo" you won't hear any difference, but you might perceive the sound as being off center. I'm using 1Hz as a example for simple math/ visualization, in actual fact you wouldn't hear anything because humans don't hear below about 20Hz.
When you have mics that are a different distance from what they are recording the signal is delayed slightly (by the distance divided by the speed of sound). Going to the simplified model lets say the delay was 1/2 a second. At that delay your 1Hz signal would be 180 degrees out of phase BUT a 2Hz signal would be back in phase (but one cycle delayed), keep changing the frequency and the phase relationship changes. So in any normal sound you will have a ton of frequencies, some in sync and most out of phase by varying amounts. Which is why you get a weird "comb filter" (hollow "tubey" sound when you play it back in mono. You can not fix this with a phase adjustment because each frequency is a different amount of out of phase.

Auto ducking or auto mixing if you are lucky and the algorithm is good is about the only "easy" fix. The catch is that generally they are looking for the hottest signal and with two boomed mics that might get problematic, depending on the setup. The F4 has an auto mix function which seemed OK when I tested it, but I have never used it in a real situation. The worry with doing it while recording is that any mistake it makes is permanent. For things where you are just archiving a meeting or something I would be all over it but for something like an interview I would be hesitant. Though it really depends on the situation. If you are going to do the editing and you can just cut out any "mistakes" then I would be more comfortable, if you are handing over the recordings or you need to use all of the recording then... Job security would make me take the extra time to do it in post.

Now I don't know if the easiest fix has been mentioned but that would be to use the two tracks as a stereo pair. As long as you don't mix them to mono you won't get that comb filter effect and instead the slight delay will give you stereo placement.

Just found an interesting article on this topic
https://www.prosoundtraining.com/2010/03/11/which-pin-is-hot-and-when-does-it-matter/
 
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