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Word clocking a Boom Recorder rig


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Here's the chain;

Yamaha 01V96 AES/EBU out to Mutec Format Changer on D-sub 25.

Mutec Format Changer to MOTU Traveler on ADAT Lightpipe.

MOTU Traveler to Boom Recorder on an Apple Mac Mini on Firewire 400.

Timecode from Denecke GR1 to Boom Recorder on an Apple Mac Mini through MOTU Traveler

All sampling rates and bit deapths will be 48/24 throughout.

Timecode already shows lock to the Mac at the Traveler.

Can I assume that work clock will be carried through the AES/EBU between the Yammy and the Format Changer and through the Lightpipe from the Format Changer o the Traveler with no need for BNC coax?

No need to speak about the setup.  This rig is fitting itself into an already working rig and I am using the only remaining digital outputs left on the Yammy.  This was the most stable cost-effective method of delivering eight channels from the Yammy to the Mac.

Of course I will test this, but I am not sure that it will go so far, at least before it is actually used, as to go to Post production and be sunk with picture, so I want to know that I have a passing chance of getting this right in the shop before it goes on the set.

Thanks.

D.

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This seems like it should work, although "received wisdom" is that WC isn't supposed to be chained in any way.  It sounds like the master clock is really the Yamaha, and your TC generator isn't taking any sort of clock feed from it (that you said) and that could be an issue for long-take sync.  If the Yammy is the clock master, it also seems like the MOTU, being clocked via the ADAT input, could be your TC generator and that that generator WOULD be clocked to the Yamaha--maybe you don't need the GR1 then?  This would work for a "closed-system" approach, where you were always the clock and TC master--otherwise you'd have to repatch things and take ext TC into the MOTU first, have it clock itself to the incoming TC, and then use its clock output to clock the Yamaha.

Philip Perkins

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Hi Doug and Philip,

I input my timecode from my Gr-1 as an analogue signal (split out to my Deva and Boom Recorder)

My fw interface to my Mac-Mini is the Fireface 400 - but the setup could be the same for the Motu.

I input time code to a line input of the Fireface and in the Boom Recorder "Patchbay" window I assign the tc analogue line input to the 9th track which I label as TIME CODE. I uncheck that 9th track from the "file" list.

On the Boom Recorder main screen I can now see the timecode signal level(9th track) and the TC readout will show up in the Timecode box.

Regards,

RL

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Hi Doug and Philip,

I input my timecode from my Gr-1 as an analogue signal (split out to my Deva and Boom Recorder)

My fw interface to my Mac-Mini is the Fireface 400 - but the setup could be the same for the Motu.

I input time code to a line input of the Fireface and in the Boom Recorder "Patchbay" window I assign the tc analogue line input to the 9th track which I label as TIME CODE. I uncheck that 9th track from the "file" list.

On the Boom Recorder main screen I can now see the timecode signal level(9th track) and the TC readout will show up in the Timecode box.

Regards,

RL

In this set up, is the RME or the console the clock master?  Do you have your board talking to the RME as analog or as digital?

Philip Perkins

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Philip,

The Yamaha is the master clock when I am at 48K/30fps or 29.97 etc. I am sending AES to my Deva via the MY16-AE card slot and obviously the return from the Deva. I use the ADAT "lightpipe" to send 8 channels of AES (and return)  to the Fireface 400.

I have successfully tested the 48.048 workflow. I explained this in a detailed post, almost two years ago, here:

http://www.jwsound.net/SMF/index.php?topic=1085.msg7070#msg7070

This setup has worked for me with out any glitches for over three years now. The only hang up I've had is I'm on my second Mac-Mini, now an Intel which has proved even more useful as I can now use Lectro's Lecnet!

RL

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Philip,

The Yamaha is the master clock when I am at 48K/30fps or 29.97 etc. I am sending AES to my Deva via the MY16-AE card slot and obviously the return from the Deva. I use the ADAT "lightpipe" to send 8 channels of AES (and return)  to the Fireface 400.

I have successfully tested the 48.048 workflow. I explained this in a detailed post, almost two years ago, here:

http://www.jwsound.net/SMF/index.php?topic=1085.msg7070#msg7070

This setup has worked for me with out any glitches for over three years now. The only hang up I've had is I'm on my second Mac-Mini, now an Intel which has proved even more useful as I can now use Lectro's Lecnet!

RL

Are you always the TC master on your shoots?  Do you have occasion to run takes of 20min + ever?  I ask because I don't see how the TC from the generator and the sample clock in the console are aligned, which doesn't matter if the takes aren't ultra long (obviously, since it is working for you).  It sounds like the Deva and the RME are following the Yamaha clock just fine.

Philip Perkins

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Yes I usually try to be the "TC master". That's why I like the Denecke GR-1 because it is quite accurate. However on some HD shoots including a features the DIT has given me a jammed lockit at either a 29.97 or 23.976 - which I feed externally into the GR-1 and slave to that time code.

I have had many takes that have been 20 minutes or more (on HD shows). I have not heard of any drift issues from editorial - but that as you know is not by any means a stamp of time code accuracy.

"I don't see how the TC from the generator and the sample clock in the console are aligned."

The clock rate in this case 48K and the time code rate will never be aligned as they perform different functions. The time code signal is never fed into my Yamaha. If either the FS rate or the TC rate is not running at their correct speeds, then, yes, matched sync between the audio and the image will not be frame accurate.

I monitor my returns frequently and if the FS clock rates were off, I would hear jitter. So I've assumed that since I have not had calls from post about any sync issues then it's all working fine.

Maybe I have misunderstand your concern on this, if so please explain.

Remember the old days of Nagra Pilotone - a much simpler world.

Regards,

RL

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Yes I usually try to be the "TC master". That's why I like the Denecke GR-1 because it is quite accurate. However on some HD shoots including a features the DIT has given me a jammed lock-it at either a 29.97 or 23.976 - which I feed externally into the GR-1 and slave to that time code.

I have had many takes that have been 20 minutes or more (on HD shows). I have not heard of any drift issues from editorial - but that as you know is not by any means a stamp of time code accuracy.

"I don't see how the TC from the generator and the sample clock in the console are aligned."

The clock rate in this case 48K and the time code rate will never be aligned as they perform different functions. The time code signal is never fed into my Yamaha. If either the FS rate or the TC rate is not running at their correct speeds, then, yes, matched sync between the audio and the image will not be frame accurate.

I monitor my returns frequently and if the FS clock rates were off, I would hear jitter. So I've assumed that since I have not had calls from post about any sync issues then it's all working fine.

Maybe I have misunderstand your concern on this, if so please explain.

Remember the old days of Nagra Pilotone - a much simpler world.

Regards,

RL

My concern is all in the realm of video and situations where one has to roll for a very long time.  For those situations I've found that it is important to have a common clock for video audio and TC, and I don't see how the 01v96 is "conscious" of the video clock in this setup.  If you got a TC feed (or Lockit/SBT) and were feeding the GR1 from that then your TC would match video's, but since your sample clock and theirs are not connected then how your audio would sit on the TC vs how the pix would sit on the TC would change over time.  I understand that the sample clock and the TC perform different functions (rate vs. address), but it is important that their relationship stay fixed, aligned, if you will, in this kind of situation if hard sync is to be maintained.  You would not hear jitter or sample rate error artifacts in your returns because your computer audio system is correctly locked to the mixer (as is the Deva).  The problems, if any, would not manifest themselves until the project got into audio post (if the picture editors were using a scratch mix you'd sent to the cameras), and they would appear as sync differences between the multitrack audio and the audio of the cut when conforming.  Is this a problem?  I ask this all the time of my Serious Engineer friends, and the answer seems to be "it depends".  It depends on what the expectations are (of sync, or automateability of the conform), workflow, and how the show will be cut.  It is pretty easy to correct sync on NL editing systems anymore--some editors are fine with doing it and others complain and get flummoxxed by having to even think about sync.  Something very cutty seems like it provides a lot more places to pull sync up than something with long takes.  Many shows mix in prosumer video cameras that have wildly drifting clocks that cannot be synced to anything.  I figure that I should provide as stable and accurate time-based a recording as I can, and the camera people will do what they will. 

Does anyone know if the 01v96 (or the Mackie Onyx or the Presonus Live) align their sample clocks with the AC line freq?  Or if they are TXCO?

Richard--since you have a Deva on your cart, why do you have the GR1?  I figure that the Deva's clock is at least as good (and it's recording is aligned with its TC generator).

Life WAS simpler w/ Nagras....until TC...and then life wasn't simple at all, all of a sudden.  No desire to relive those days (and nights).

Philip Perkins

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How about the even simpler world --- sprockets! I guess that's actually before our time. Anybody here actually work on a production where the main production recorder was a mag film recorder?

-  Jeff Wexler

YES!  At the Univ. of Oregon we had a "portable" 16mm magfilm recorder ("Halley"?  "Hallet"?), and that's what we recorded audio onto for our projects to avoid having to pay for magfilm transfers at a lab.  Miserable piece of junk.  Audio recorded on this thing, then converted to 16mm optical sound gave new meaning to the term "low-fi".

Philip Perkins

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Philip,

I'll answer the easiest question first as to why I use a GR1 instead of the Deva as a master TC source. I just got used to using it, especially when I would be given a lockit at 23.976 (because the DIT insisted on this) and I could cross resolve it through the GR1 to 29.97 Hey, I also like to have a visual on all the time code running on my cart.

My working situations have been in feature films - using film and HD (Viper) and television episodics, again film and HD (Sony 900, 950 and Panavision Genesis).

In all those situations I have never had Telecine, the Picture Editors, the Post Sound department or the Rerecording Mixers complain about jitter or sample rate error artifacts.

"You would not hear jitter or sample rate error artifacts in your returns because your computer audio system is correctly locked to the mixer (as is the Deva)."

But I have - when the clock rates on either the Yamaha (I accidentally changed it to 44.1 in between takes) or the Fireface or Deva have mismatched FS rates. When all three devices have the correct FS rate then the recordings are pristine. So to me in a very non technical manner I know that the three clocks are in sync.

If I'm sending audio to either the HD cameras or the DIT - that is an analogue signal. So I don't think there would be an issue with these multiple units all with separate clocks.

The same occurs with the time code - it is being recorded simultaneously on the Deva, Boom Recorder, and the HD cameras. Assuming that lockit boxes and slates are all jammed correctly?

I think the weakest link in all this is us humans. Lockit boxes forget to be jammed, time code slates, wrong choice of TC frame rate etc. etc.

I'm sure there are frame offsets that occur and are readjusted when the material is being conformed either in picture editorial or sound editorial. Again if there were really horrible frame offsets causing the sound to drift, I'd be the first to be called and blamed!

And yes, on some HD shows they have used the camera scratch audio and reconformed from my DVD-RAM masters with no nasty calls from post.

On the series, Dirty Sexy Money, I got called by post only once in 30 episodes. They could not sync a few takes in one scene. I asked them if everything before that scene was in sync and afterwards as well. This included two separate DVD-RAMs.The answer was yes. After I asked lots of questions the sync issue was related to only one mag of film from the Steadicam. It turned out that one film mag was not feeding correctly and the camera ran way off speed.

So is was not "Sound's Fault" at all. But believe me as I was trying to figure it out, I thought about every possible "human" error that could have made it "my" fault.

I think it's worth asking Yamaha how the 01v96 aligns its sample clock.

Phil, how long are the takes you are rolling on?

RL

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Philip,

I'll answer the easiest question first as to why I use a GR1 instead of the Deva as a master TC source. I just got used to using it, especially when I would be given a lockit at 23.976 (because the DIT insisted on this) and I could cross resolve it through the GR1 to 29.97 Hey, I also like to have a visual on all the time code running on my cart.

My working situations have been in feature films - using film and HD (Viper) and television episodics, again film and HD (Sony 900, 950 and Panavision Genesis).

In all those situations I have never had Telecine, the Picture Editors, the Post Sound department or the Rerecording Mixers complain about jitter or sample rate error artifacts.

"You would not hear jitter or sample rate error artifacts in your returns because your computer audio system is correctly locked to the mixer (as is the Deva)."

But I have - when the clock rates on either the Yamaha (I accidentally changed it to 44.1 in between takes) or the Fireface or Deva have mismatched FS rates. When all three devices have the correct FS rate then the recordings are pristine. So to me in a very non technical manner I know that the three clocks are in sync.

If I'm sending audio to either the HD cameras or the DIT - that is an analogue signal. So I don't think there would be an issue with these multiple units all with separate clocks.

The same occurs with the time code - it is being recorded simultaneously on the Deva, Boom Recorder, and the HD cameras. Assuming that lockit boxes and slates are all jammed correctly?

I think the weakest link in all this is us humans. Lockit boxes forget to be jammed, time code slates, wrong choice of TC frame rate etc. etc.

I'm sure there are frame offsets that occur and are readjusted when the material is being conformed either in picture editorial or sound editorial. Again if there were really horrible frame offsets causing the sound to drift, I'd be the first to be called and blamed!

And yes, on some HD shows they have used the camera scratch audio and reconformed from my DVD-RAM masters with no nasty calls from post.

On the series, Dirty Sexy Money, I got called by post only once in 30 episodes. They could not sync a few takes in one scene. I asked them if everything before that scene was in sync and afterwards as well. This included two separate DVD-RAMs.The answer was yes. After I asked lots of questions the sync issue was related to only one mag of film from the Steadicam. It turned out that one film mag was not feeding correctly and the camera ran way off speed.

So is was not "Sound's Fault" at all. But believe me as I was trying to figure it out, I thought about every possible "human" error that could have made it "my" fault.

I think it's worth asking Yamaha how the 01v96 aligns its sample clock.

Phil, how long are the takes you are rolling on?

RL

Yes, you heard the jitter or artifacts when you were "incorrectly" locked, not when you are/were set up correctly.

I work in post as much of the time as I do in production anymore.  Without doubting your methods, I can tell you that in post if we have a problem with any aspect of production sound most of the time we try to fix it ourselves w/o dropping a dime on the production sound people, partly out of understanding that it's hard out there and partly due to there not being time to do anything BUT fix it however we can--the field sound guy can't really help us, usually.  So you might not have heard about resync issues, and if your takes are normal movie length then there probably weren't any.  I'm talking about (in answer to your question) a different sort of video production style where the rolls are much longer--live shows, concerts etc..  You asked about my roll lengths--my standard now is 3 hrs. from roll to cut--the system has to be able to hold sync for that long.  And for that kind of thing I've found that clocking sample clock and TC together with video is vital, and I have had sync issues w/ post when we did not do this. 

It would be interesting to know how Yamaha/Mackie/Presonus derive the sample clocks in their digital mixers.  The Yamaha is way ahead of the other two in my book because it CAN take ext. WC.

Philip Perkins

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I did find a few interesting numbers, maybe some of you other folks can fill in the blanks.

Internal clock accuracy:

Yamaha 01v96v2:            1000 ppm

Mackie Onyx:                  ?

Denecke GR1/SBT/SB3:    +- 1 PPM

Ambient Lockit:                +-0.5 PPM

Sound Devices 744T:        0.2 PPM

Deva V:                          ?

Philip Perkins

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Excuse my ignorance, but does this make the 744T the best?

My setup is this... (Analog board) 744T TC to analog input of FF800 to Boom Recorder - 744T spdif out (mix) to spdif in on FF800 to Boom Recorder.  The FF800 interface "locks" to the 744T via the spdif input.  Is this okay?  I have used this setup for 29.97, 30, and 30/48.048 - but as you say, no news is not necessarily good news.

Robert

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Excuse my ignorance, but does this make the 744T the best?

My setup is this... (Analog board) 744T TC to analog input of FF800 to Boom Recorder - 744T spdif out (mix) to spdif in on FF800 to Boom Recorder.  The FF800 interface "locks" to the 744T via the spdif input.  Is this okay?  I have used this setup for 29.97, 30, and 30/48.048 - but as you say, no news is not necessarily good news.

Robert

Seems good to me for situations in which you are your own TC master--the 744 etc is a great clock (Ambient) and it makes the sample clock of the interface much more accurate than it would be on its own.  To sync to something else (like TC from video) you could just take that TC into the 744 which would then automatically lock itself (and everything you have following it) to the incoming TC.  This is essentially how I work w/ MOTU boxes etc and it was worked out very well for very long takes (short takes too).

Philip Perkins

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Interesting that the Lock-it and the 744T are not the same accuracy.  As far as I know, they both use the same Ambient clock.

Thanks for all the varied information.  I like what Richard says about the humans being the least accurate link in the chain; it's very true.  I try my darndest to figure it all out, but then, if I don't hear anything bad back I gotta move on to other issues and consider it "good."

I think that Post on a season long show would say something if your track was continually the source of sync errors.  I had a TS-2 that had an intermittent stall and I heard about it until we figured it out (took a couple of trips to Denecke but we all kept on it until it was fixed).  Cool thing was that Post re-sunk when needed and never complained to production, so I never had to field any stupid questions "why."  It was very professional all around and I am so grateful for support like that.

I think I will set all the sample rates the same on all the devices, and expect AES/EBU to carry the clock through the chain.  My only concern is what happens to the clock in the Mutec format changer (until MOTU puts and AES Dsub 25 on their Traveler<g>.)

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FWIW I've been using an Apogee Big Ben as a master sync reference.

When working with video, I stab it with a blackburst signal from video village and lock to them.

Each device with a sync input gets a dedicated line from the Big Ben and everything ticks along happily.

I have used this setup on rolls sometimes lasting an hour or more and have not had any issues to date.

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  • 1 month later...

with BoomRecorder is it possible to feed audio via FW into the Mac but TC from the 744 into the lineIn of the Mac?

Basically I need to tell BR to look at the FW interface for track 1-8 and Mac-lineIn for track 9. Possible?

I`ve got 8 analog Ins on the FW interface that are occupied with the DiOuts from the mixer.

Cheers, Matthias

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Matthias,

I run TC into channel 8 for BR.  In the rare event that I need to plug in 8 mics, I simply unplug the TC and plug in the mic.  BR holds TC very well on its own.

I know it doesn't answer your question, but it is my understanding that you need to assign either your external interface OR your line input, and you cannot do both.

Robert

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You can aggregate your interface with the Mac's input, using the Aggregate Device Editor found under the Audio pulldown menu in Audio Midi Setup. It's in the Utilities folder inside the Applications folder. It can be done, just not sure how advisable this is.

Best regards,

Jim

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I thought someone had written about this somewhere already (was it you Philip?). I just can`t find it. I know I could plug and unplug but I rather keep everything cabled ...

Matthias

With Metacorder you can tell it to look for the TC on the Mac's own audio input, thus saving an input on yr interface. If you don't need the 8th input of the interface then you can use that too.  I'm almost always strapped for inputs when using that rig, so saving that TC input helps.  I thought BR could do this as well--I don't have an interface hooked up to my Mac now, but BR sees the Mac's inputs in the Patchbay.  I'm pretty sure the deal w/ BR (diff from Metacorder) is that it senses the presence of TC on any input automatically and will stamp files with that TC.

Philip Perkins

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Just curious...

What happens within Boom Recorder when there are multiple timecode sources on the inputs?

For example, one input contains your production timecode but another is playback timecode that you just want to record on to it's own track.

With Metacorder you can tell it to look for the TC on the Mac's own audio input, thus saving an input on yr interface. If you don't need the 8th input of the interface then you can use that too.  I'm almost always strapped for inputs when using that rig, so saving that TC input helps.  I thought BR could do this as well--I don't have an interface hooked up to my Mac now, but BR sees the Mac's inputs in the Patchbay.  I'm pretty sure the deal w/ BR (diff from Metacorder) is that it senses the presence of TC on any input automatically and will stamp files with that TC.

Philip Perkins

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You can aggregate your interface with the Mac's input, using the Aggregate Device Editor found under the Audio pulldown menu in Audio Midi Setup. It's in the Utilities folder inside the Applications folder. It can be done, just not sure how advisable this is.

Best regards,

Jim

response from vosgames (BoomRec):

It is possible to aggregate the M-audio with the line interface using the

Audio MIDI Setup utility, but this doesn't always work well on some

computers.

http://www.apple.com/pro/techniques/aggregateaudio/

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