Time alignment -UM400a and Boom mic

[chris_bollard]

Hi all

Can't help but think this has been discussed before but  a search didn't turn up what I'm looking for.

I'm curious as to what the time delay required to align a UM400a radio and boom mic ( around 12 to 14" from subject). Was hoping to dial it in on my 788t.

Thanks

[ProSound]

No time alignment required just record as normal

[RPSharman]

Wow.  If you're capable of noticing a delay when your mics are merely a foot apart, then you have a very special extra sense.

There are the RARE times when I have a delay on some crazy wide shot, and I am trying to mix in a bit of lav (or a lot of lav, depending), but there is a delay.  In these cases I do the best I can and let the re-recording mixers sort it out in post.  It's very easy for them to slip the sources together and blend the mics to match their choice of perspective. 

There's just not time to figure it out in the field, and I don't like to mess around with those settings.

Robert

[JDirckze]

Could you calculate it based on the assumption that the speed of sound is roughly 344m/s, or would there be an additional delay introduced because of the processing within the UM400?

[Jeff Wexler]

I'm with Robert on this even though I have had the ability and function to deal with delay for the longest time (early adopter of Zaxcom gear). I have never, ever, used any delay anywhere to compensate for any sort of latency inherent in the devices (analog vs. digital) and I don't think the tracks have suffered at all.

I do remember some rather outrageous discussions/fears when the Zaxcom digital wireless was released (the first time most of us had even heard the word "latency" mentioned). There were those who said that it would be impossible to use these things with all that delay, they were worried that nothing would even be in sync! What will I tell the producer or the people in post that the lips don't match! Most of us realized that this was a non-issue and a total subversion of the issue of delay. It was pointed out, I think even by me, that in a movie theater the sound will be different for the person siting in the 4th row compared with what the person sitting 10 rows back (delayed) but on one complains that it's out of sync!

[Bob Marts]

A somewhat related anecdote: 

Back when I was mixing to one track only, I came up with an approach to mix some boom mic into an otherwise all lav mix without ending up with a lot of phasey comb sound. I would EQ the boom mic channel by removing all the lows and lower mids which would give some nice air to the voices but not create delay problems on the lower freqs. Had to really play it by ear but it worked well.

[afewmoreyears]

Never worried about time alignment... I always simply move the boom away a decent amount (20ft) from the dialog and point it way off access... then bleed in the boom for life to wake up a sterile Lav sound...  It's always worked nicely for me....  If it's too far from talent to get a boom in, I don't try, I try to make what I know will work, sound good... and live with it... But, that's just my take...

My boom is on an ISO anyhow in case they want a ambient track to mix later on their own... I just don't want the dialog on it... (So is the LAV...)  or do with what they please.... Throw it away ?

[Marc Wielage]

I believe the internal latency in a Lectro UCR-411a is about 3ms. My memory is that the 401's were the same in digital hybrid mode, about 1/10th of a frame (1/240th of a second).

I believe Larry F. has said this amounts to a difference of two microphones several feet apart -- not a big delay. I'd have to ask a dialog editor if they worry about timing differences this small, but my gut feeling is no -- they're more involved with fixing actual performance- and level-related issues.

I do the same thing as AFewMoreYears above does above: I'll typically mix a little bit of the boom in anyway, if we're primarily using a lav in a scene or an interview, just so it doesn't sound so close-miked. I've never heard any comb filtering or phase cancellations to the point where it's a problem.

--Marc W.

[Marc Wielage]

I believe the internal latency in a Lectro UCR-411a is about 3ms. My memory is that the 401's were the same in digital hybrid mode, about 1/10th of a frame (1/240th of a second).

I believe Larry F. has said this amounts to a difference of two microphones several feet apart -- not a big delay. I'd have to ask a dialog editor if they worry about timing differences this small, but my gut feeling is no -- they're more involved with fixing actual performance- and level-related issues.

--Marc W.

[LarryF]

Three milliseconds is total system delay. So it's 1.5 ms for UH400a and 1.5 ms for the corresponding receiver (UCR411a).

LarryF

[Wyatt Tuzo]

Or, look at it this way. The speed of sound is 1,126 ft/s, or 1.126 ft/ms. So if we approximate that sound is delayed 1ms for every foot from the source, the 3ms introduced by a 400/411 setup would be comparable to to a boom mic from 3ft overhead.

With such slight variations possible within the practical scope of what we do, comb-filtering due to phase misalignment would be minimal at most. I REALLY wouldn't worry about it. It's one of those theory vs. practice things.

Where one COULD get burned, is when using a digital hybrid along side a non-digital hybrid model (200 series or earlier) to capture something where phase coherence is more critical (say, mid-side stereo recording, for example). But then I assume that most people doing stereo ambient recording aren't doing so wirelessly (with mismatched units).

Best,

Wyatt

[JDirckze]

So the answer to the OP's question is 3ms (UCR411/UM400 processing delay) + 1ms (distance of boom from lav 12") = 4ms delay on boom to match lav (approx.)

[Marc Wielage]

Three milliseconds is total system delay. So it's 1.5 ms for UH400a and 1.5 ms for the corresponding receiver (UCR411a).

I humbly bow to Larry! 

--Marc W.

[soundbeing]

The digital latency always bugged me when I added Lectro 400 series to my 205s.  There are 2 approaches that I have used that improved time alignment coherency. 

When I ran the boom on a 400 and actors on 205s I would route the mix as split channels separating 205s from 400s and dial in delay on the analogue 205 routing then re-combine the 2 channels to the mix track.  Usually 5ms worked best which corrected both latency and boom distance of 2 to 3 feet overhead.

Later I reversed my approach and now run boom on an analogue 205 and primary actors on digital 400s.  No fancy channel routing.  This way the latency works in favor of the mix effectively bringing the alignment of the  boom 3' closer to the wires.

M.Weber

[Jonathan Reyes]

Will there still be a 3ms delay if you switch your ucr411 and um400a to 200 mode so that the wireless are running non-digital not digital? 

[greg sextro]

yes there is still a delay - but it's 1.5 ms - not 3.

I personally prefer a digital mixer for this reason - to time align digital wireless with analog wireless and analog booms. 

I have always been unhappy with the sound when mixing without correcting these delay times.

-greg-

Will there still be a 3ms delay if you switch your ucr411 and um400a to 200 mode so that the wireless are running non-digital not digital?

[Jonathan Reyes]

yes there is still a delay - but it's 1.5 ms - not 3.

I personally prefer a digital mixer for this reason - to time align digital wireless with analog wireless and analog booms. 

I have always been unhappy with the sound when mixing without correcting these delay times.

-greg-

So switching modes would reduce the 3ms delay to 1.5ms?

[greg sextro]

yes - but the sound changes.  the 200 mode is designed to replicate the sound of the 200 series.

[Marc Wielage]

Will there still be a 3ms delay if you switch your ucr411 and um400a to 200 mode so that the wireless are running non-digital not digital?

But then you have the sonic problems of companding. I don't think 3ms is worth worrying about.

--Marc W.

[LarryF]

Their are some mis-understandings here: The big delay is in the digitization (A to D, D to A), not the DSP processing. The delay is precisely the same, whether 100, 200, IFB, S, S, or 400 mode. The 400 series hardware has a 3 ms system delay in all modes. By the way, the Venue receiver has a mode to match delays of analog and DSP transmitters. Since some users are getting improved results running 400 systems in 200 mode, this tells me the delays are more in perception than in substance. Just sayin'.

Cheers,

LarryF

[RPSharman]

Resurrection here for a second...

I have 3x 411s, and was considering adding another receiver. Found a 211. Is the delay between the 411 and 211 really an issue?

I hardly ever use 4 portable receivers, but I have a rack distro which holds 4, and the empty hole bothers my sense of order. But no point of filling it if there are issues.

I suppose I could dial in 3ms on the 211 input to the 788T. Did I just answer my own question?

Robert

[jason porter]

Yes, dial it in

[RPSharman]

I'd be using 400 or SM transmitters in 200 mode. They still have the delay, even in 200 mode, if I read Larry's post correctly. So it's really just 1.5ms

[Mark]

Dial it in.

In some situations it may be fine but get 2 side by side and you will notice it for sure.

[minister]

I work in post. I have never ever looked at whether or not the boom and the LAV were time aligned.  Not just because I don't care, it doesn't matter.  First of all, it is rare that the 2 channels would ever play together in a mix.  In the rare instance when it would -- when ADR was not an option, and I needed 2 actors lines heard -- I would just adjust it so they worked.

 

I have to also ask why you feel the need to do this.  The sound emanating from a source is a composite wave.  All you are doing with time-aligning is, at best, lining up the fundamental and/or transient. Doesn't make them sound "better". The harmonics and the rest of the frequency content of the source don't stay in synchronous phase, especially when there is room interaction. Look at the waveform from source X in Mic A and chose two points on a sine wave. When you try to look at those same points of source X on Mic B, they will be in a different point of their cycle. This shows that their phase relationship has changed.  And will change again.

 

Get the sound right at each mics and call it a shoot.  And no, nudging it in your DAW is not phase aligning.