Hello again,

Let me start by describing my setup:

6 Radio Mics (Sony UWP-C3 Transmitters)
Each Mic assigned an input channel on an Alesis FireWire Mixer
Alesis FireWire Mixer attached to Fujistsu Siemes Notebook
6 Audio Channels created in Vegas to capture sound from each Mic/Channel

Recording went perfectly and all tracks identical in length and in perfect synch with each other.

Problem:

Using the Sony FX1 for video (DV) and the cameras audio track as a reference (to the above recorded audio tracks) the video starts out being perfectly in synch with the tracks recorded from the mixer (when placed correctly in the Vegas timeline) but gets progressively out of synch by the end of the tape.

In other words - I have 6 sound channels recorded using Vegas from a FireWire Mixer and I also have the audio track from the camera that is used as a reference to align or synch the 6 recorded sound channels to the video. Once correctly (manually) aligned on the Vegas timeline the video the video are initially in synch but the video and audio seem to go out of synch by the end of the tape / video footage i.e. after about an hour.

In order to correct this I have to stretch the video track by a fraction to get the whole video event synchronized with the audio recorded from the mixer.

I am confused about this.

I would have thought that because everything is being recorded digitally that this would not happen.

The project created to record the 6 audio tracks from the mixer had the identical settings (48,000 kHz, 16 Bit audio) to the final project where everything i.e. the 6 audio tracks and the video footage are being brought together again.

Why would the audio recorded from the mixer using Vegas be any different to the audio track of the camera?

Is this possible?

Any ideas or input?

Regards,

Dale.

Dale, the only thing that comes to mind is that the camera probably runs in drop-frame mode (i.e. 29.97 frames/sec.).
If the audio stream wasn't clocked to this (i.e. it was running at non-drop or 30 fps), the drift after one hour would be 3.6 sec. (108 frames).
Am I on the right track?

Mike

...the camera probably runs in drop-frame mode (i.e. 29.97 frames/sec.)...

I don't think this could be the issue. The camera is at 29.97 fps regardless of how the time code generator is counting the frames. The audio record chain has no frames, it just runs at its sample rate.

I'd be looking deeper into setups in the Alesis Firewire mixer. This is where you want the clock locked to 16/48, this is the clock that determines your audio sampling and sync. Is it possible that the Alesis is set up for 16/44, then you're recording in Vegas at 16/48 while resampling on the fly?

In dealing with your existing recordings, I think you'll have better results shrinking the duration of the audio than expanding the duration of the video. Definately shorter rendering times, and perhaps better picture quality.

BTW, multitrack recording and posting in Vegas has been very very good to me! Some with Sound Devices 744T, some with Alesis HD24. I've been recording at 24/48.

Hi, and thanks for the replies.

The Alesis was set to 16/48 but I don't ever remember clicking on the button that says 'set clock master' but I am not sure if this was ever necessary as 16/48 is the default for the installation.

BTW I am shooting in PAL so I can only assume that I am getting a straight and true 25fps :)

It is the strangest thing though and I cannot figure it out. And it is not out by that much either i.e. not even a whole second at the end of the video footage more like milliseconds.

Maybe I am supposed to click on the 'set clock master' in the Alesis software prior to recording in Vegas BUT if the mixer was actually recording at 16/44 then I am sure that the audio track would differ greatly from the cameras audio track but like I say it is only fractionally out.

Just had a thought though - if the audio tracks were recorded at 16/44 then Vegas would identify them as such would it not? Vegas does recognize the individual audio tracks as 16/48.

Dale.

Regarding "set clock master", I'm unfamiliar with Alesis' terminology here. If there is anything that looks like "lock sample rate" you want it.

Milliseconds out over an hour may be the best you can do. I'm not enough of a digital circuits person to say whether you "should" be able to do better. We're talking extremely small fractions of a percentage differences if, say, you're 200 milliseconds out after an hour... that's about 0.0055 percent error.

If I've got my math right, that would be about 5 frames at 25fps.

Not bad for clocks that aren't locked to each other?

Hi,

I suppose you are right - it is a small margin - but I was always under the impression that because all of this is digital none of this would matter.

I tried your suggestion about stretching the audio track but it gets far more complicated because although initially your video is in synch with the audio tracks in the beginning - by adjusting the audio tracks to synch with the end of the video you are changing the properties of the audio tracks in their entirety and the beginning then goes out again so you would have to repeat this process over and over until you got it just right.

The strange thing is that normally when you stretch or compress a video track the frames are recompressed when previewing on an external monitor via FireWire but this is not happening when stretching or compressing the video by such a small amount.

I have another thought though - do you think it makes any difference that I captured the video on another workstation (my desktop editing workstation) i.e. the audio was captured in the field on a notebook but then copied to my desktop and the video was then captured from the FX1 straight onto the desktop for synch and editing. I have not tried capturing the video to the notebook and then trying to align then stuff there (another three hours just to test)?

Still don't understand this clock thing!

By the way - is there any better way or technique of checking the synchronization i.e. I am previewing all of the audio tracks (including the video's original audio track) and then moving the video / video audio track back and forth on the timeline until there is no echo or delay and then I am assuming that at this point the audio of all tracks is in synch i.e. when the audio tracks are out of synch an echo or delay can be heard but when they are spot on with each other there is no echo or delay. Any other techniques?

Dale.

...do you think it makes any difference that I captured the video on another workstation...


No, don't think so. A capture in DV/Firewire is really more of a file transfer, no bits get changed from what's on tape.

...By the way - is there any better way or technique of checking the synchronization i.e. I am previewing all of the audio tracks (including the video's original audio track) and then moving the video / video audio track back and forth on the timeline until there is no echo or delay...

With a good ear and some devoted listening this is the best! Due to the way we perceive sound, our hearing is really very accurate for small timing differences (echo). Short of field equipment upgrades to cameras and recorders capable of jamming timecode, or better yet running synced, you can't do better, nor do you need to.

If you've reduced echo to nil, and lip-sync seems right, you're done.

If you're slipping the audio it can be handy to turn off "quantize to frames" so you can get sub-frame sync (better than 1/25th of a second for PAL). Don't forget to turn quantize to frames back on before you touch anything with video.

I'm not a Vegas user, but I had the same problem once before. Audio started out fine but later on was out of sync. Turned out that during the capture there was a frame drop in a couple of spots. Once I found out where the frame drops were (a tedious procedure) I realigned the audio clips at those points and things were fine.

In Premiere I've never had problems with drift importing 16/44k audio into 16/48k projects.

Barry

Thanks eveyone for the replies and info.

Seth - I followed your advice and stretched the mixers audio tracks and not the video and after only a slight adjustment the audio the individual video clips just fell into place which tells me that it is not the video capture but something to do with the audio capture.

By the way the difference per clip was 0.120 seconds - amazing how we (our ears) are able to perceive such small differences. I'm just glad I was able to align it and was just surprised that this kind of thing could happen in our digital world. Like you said - small differences like this (the audio tracks from the mixer were actually about 3 hours long and I only had to stretch them by a fraction) are actually not bad for unsnyched clocks (although I did not know this was a factor).

Funny enough - only yesterday - after doing the shoot - did it occur to me to turn off things like quantize to frames, snapping, etc. etc. in Vegas when capturing the audio from the mixer on the notebook. Maybe all of these settings played a part but as you say - I did get it right and there you go.

Regards,

Dale.

Just to update this thread with some further thoughts on the subject:

It has dawned on me that using the audio captured by the camera as a reference to synch the video to the audio captured via the mixer is not actually an ideal or useable method. The reason I say this is as follows:

The camera was picking up the sound from the public address system (which was connected to the mixers main out). The notebook was recording the sound directly from the wireless mics. In other words - the camera was picking up the audio after it had come from the mics - then to the mixer - then to the public address system - and only then to the cameras mic. Also the camera was always a fair way away from the public addresss system. Is it not possible that this sort of 'round trip' compounded by the physical distance between the camera and the public address system could have caused my video/audio synch problems (although it still would not explain why either the video or audio track would have to be stretched or compressed given the same sampling rate/bit depth for a given period)?

Regards,

Dale.

It is easy to get a one frame difference between the optical record of the event and the audio. This can happen over a shorter distance than you might think. Just for grins, think 600 mph is the speed of sound for this calculation.

We take 30*60*60 frames per hour, that's 108,000 (NTSC)
Let's figure nautical miles cause that's 6000 feet.
600 x 6000 = 3,600,000 feet per hour. But we get to divide it by 3600 to get down to feet per second = 1,000 feet per second.
30 frames per second = 33.3 feet per frame

So you can see that you don't have to be too far away before the difference between a camera-local microphone and a radio microphone or a action-local recording system to have a significant difference in sound recording timing.

Even a small wedding can cause problems.

Some digital A/D converters run at 48.048kHz instead of exactly 48kHz but produce files that are timestamped as 48kHz. The purpose is to produce a 0.1% slowdown to match the speed of 30FPS viewed at 29.97FPS for editing film on video in Avid etc. I wonder if the Alesis Firewire mixer could be one of those critters? The slowdown would mean the audio would run just a hair longer than the video. Just guessing here, you'd need to find out the exact sample rates in the Sony camera and the Alesis mixer to know for sure.

Steve and Mike - thank you both for those replies.

Mike:

I'm not sure that I understand your calculation proper - particularly the 'nautical miles part'. Also I shoot in PAL and use the metric system so if you could explain a little more in detail (or adapt it for my specifics) I would appreciate it. Does the speed of sound not come into this equation? (I can't believe how ridiculous that question sounds)!

On the other hand - accepting you figures for an NTSC environment - it does make a lot of sense. At my last event the main camera was at least twenty metres away from the subject at any given point (and twenty five metres away from a subject at the far end of the venue) and that would equate (if I'm not mistaken) to about a two or three frame difference or delay (still not quite sure what we're talking about here) which seems to be exactly the difference between the audio and video after rendering (using the camera's original sound track to synch to the mixers sound track). The lip synch is definitely out by a fraction on the final product and I think that this is why.

I am going to use another method that I hope will work - I'll assign an extra channel on the mixer to a spare mic and at the beginning of each tape I will somehow generate a series of 'beeps' or 'test tones' using a small dictaphone or message pager or something like that (in close proximity to the camera and the spare mic). The idea being that I would then be able to 'visually align' the two audio tracks (at least at the beginning of each tape). Any thoughts on this idea?

None of the above of course explains away the difference between the length of the camera's audio track and the mixer's recorded audio track. Steve, however, may be on to something here.

Steve:

I did not know about the possibility of having a 0.1% slowdown for matching 30fps to 29.97fps and I am shooting in PAL so I would imagine that this could cause an even bigger difference if this is indeed the case. Am I right? I will investigate this further as I can just see this causing many problems for me in the future. It took me a whole day just to synch two hours of video to the mixer's audio tracks to the point where I was happy and even then, after rendering, the lip synch is slightly out (because of Mike's theory above I think).

How did I arrive at this point? What happened to the good old days of shooting and editing home video of the kids? What happened ...? Actually - not true! I think my fascination with video and film actually lies, not in what I already know but, in what I have still yet to learn! (How's that for a quotable quote)!

Thanks again.

Regards,

Dale.

Dale,

Here's a couple of thoughts as well - for what they are worth. Try using a line out of the mixer into your camera. Bypass that on camera mic altogether. Since everything is coming off the board, use the line in as your reference channel and sync up your 6 discrete channels to it. What I do is expand the height of the audio tracks and minimize the video track. I group the 6 other tracks so they move in unison. I then match the waveforms by expanding out the timeline with my wheel mouse. I'll drop my curser somewhere on a peak on the reference track and then slide the group so the same peak is lined up. It really helps to have a click or other loud, short duration sound to use for a mark. I do this at the front of each song and once it appears as close as I can get, I go to maybe halfway through and find another reference peak and confirm sync. If you are planning to edit in individual songs, you can cut your tracks (split) and resync all over again as you go down the timeline. This way you minimize any sync issues over short segments.

Hope this helps.

Jeff

Thanks Jeff for that.

Funny - I have been thinking about doing something similar (well actually the same - just a different way).

The Sony UWP-C3 Kit is made up of a plug-on transmitter and receiver. The nice thing about the transmitter is that you can switch the input between mic and line level so it would not be a big deal to take a line out from the mixer to the transmitter and then straight to the camera - mabe this is the BEST way of doing things. It would also most certainly identify which of the clocks is causing the synch problems (see details in the thread above) or at least which of the clocks is 'out'.

I really think that I need to give this a try.

Thanks,

Dale.

I was just reading this very LONG online article and it seems like this is a possible explanation for what you're facing: http://www.kenstone.net/fcp_homepage/location_sound.html - If you scroll down about 3/4 of the page to the "Dual System Sound" heading, you will find these paragraphs after that heading:

For the past few years, there has been much discussion about using Mini Disc, DAT or hard disk recorders for dual system sound recording when shooting DV. In theory, it's really easy. Sure, your VX-2000 may record horrible, unusable audio with a lot of hiss. Just bring along an MD recorder, pop in a disk and life is good, right? Well, it's too bad that in reality, it's just not that simple.

First of all, from a mechanical standpoint, even though both of the devices (camcorder and MD, DAT or hard disc recorder) are both digital, they are also both completely independently synchronized to their own internal master timing device. You would think that if both devices are digital and both are using the same sample rate that it would be a simple matter of lining up the picture and the MD, DAT or hard disc recorded audio on the same timeline and let them go. The problem is that neither device is synced to each other's master "clock" so eventually, the two sources will lose sync with each other. The two may lose sync at 5 minutes; they may lose sync at 45 minutes. There are no hard and fast rules about how long each set of devices can hold sync. There is no way to determine exactly how long it will take unless you test the devices and notate how long it takes for the two to lose sync. Still sounding easy and straightforward? Let's say that your particular combination of DV camcorder and sound recording device will stay in sync for 19 minutes at a time before they lose more than one or two frames sync. You did your homework and determined the number. The next challenges become ones of routing, organization and media management. Let's go over each of these:

Shawn - HOW DID YOU FIND THAT???

Thanks - that article describes EXACTLY ALL of the problems that I am having - and it would appear that the quick fix is to take a line out of the mixer (via a transmitter of course) to the camera (although as I read it it still does not guarantee that the different clock sources remain in synch with each other).

The most worrying thing of all mentioned in that article is the fact that the clocks of the different devices may / may not go out of synch and if they do go out of synch it may be at different points each time (if I read it and understand it correctly) i.e. they may be in synch for the first five minutes and then one of them goes out or they may stay in synch for two hours and then go out or even worse one 'slips' two or three times within an hour of video!!!

I unfortuanately (due to the nature of my work) no longer have a choice as to whether or not I use a seperate audio recording device (we provide the public address systems for the events as well and there are just too many audio sources to be recorded / sent to the PA without using a mixer) so I have to make sure that this seutp works for me.

Thanks again for finding that article - it contains some really great information in it (aside from the references to my problem).

Regards,

Dale.

A question along a totally different line.Although in your situation ,it may not be a problem,is are you listening to a preview and finding the sync out?
I have found that in some situations the preview may not sync and yet after a render or prerender it is infact in sync.Just 1 thing to eliminate prior to further diagnosing.

Shawn - HOW DID YOU FIND THAT???
Hey Dale - I'm glad that the article was what you needed. It was just good fortune that I could recall it - I was reading the article due to another post from DVInfo and then I saw you post - I'm just happy I could remember where the source was for the webpage (reading too many articles turns into one big mush for me when I try to relocate them :)
... and it would appear that the quick fix is to take a line out of the mixer (via a transmitter of course) to the camera (although as I read it it still does not guarantee that the different clock sources remain in synch with each other).
I'm not quite sure I'm following you here. If you have all your sources (mics, instruments, etc.) going into the mixer, then Brockett is suggesting that you use a dual output from the mixer (via XLR or wireless in necessary), in which case your camcorder receives one output and your digital audio recorder receives the other output. This way, your camcorder and digital audio recorder have an identical source to record from. If you did this, the audio recorded on your camcorder would be in-sync because it's all done on your camcorder using the same internal clock for video and audio. However, your digital audio recorder will still go out of sync at some point. Brockett is suggesting this so that the two can be compared in the future, and he suggests that you "post conform" the audio but doesn't give a lot of detail IMHO on how to do this. The only thing I don't know about is your mixer - it sounds like you are using a digital mixer and that may also have an internal clock as well - it seems like this would have to be the case - yes?
The most worrying thing of all mentioned in that article is the fact that the clocks of the different devices may / may not go out of synch and if they do go out of synch it may be at different points each time (if I read it and understand it correctly) i.e. they may be in synch for the first five minutes and then one of them goes out or they may stay in synch for two hours and then go out or even worse one 'slips' two or three times within an hour of video!!!
Okay - I didn't get the same impression on that. I thought Brockett was saying that you can test the two devices and determine when they will go out of sync AND this will take place consistently each time if you're using those two devices.

As I look back over this whole thread, it seems like the advice from everyone is pretty similar to this article, but it's just in pieces (Seth is talking about non-sync'd clocks; Jeff suggests using a direct line out of the mixer). The only thing I now question is whether or not it's a good idea to stretch audio at all, since this would re-digitize the samples. As an alternative, I'm wondering if it might not be better to delete a small portion of audio at specific intervals related to the out of sync issue (e.g. 10min or 20min), but of course you would have to find a spot in the audio near those marks that was just room noise and nothing else. To those in the know, would this improve the audio quality or not make any difference verses stretching the audio?

I unfortuanately (due to the nature of my work) no longer have a choice as to whether or not I use a seperate audio recording device (we provide the public address systems for the events as well and there are just too many audio sources to be recorded / sent to the PA without using a mixer) so I have to make sure that this seutp works for me.
Well I'm no audio expert, but if you camcorder's audio recording has acceptable quality, can't you just record the mixer output to your camcorder and not have to mess with the sync issues? If you really need the higher quality audio, then use your audio recorder -- but if not it seems like you can get what you need with your camcorder. Based on what you've said, it seems like you have to record long uncut segments (like an event videographer) and as such, you face problems that fiction-'film' folks do not face since they would generally have takes short enough to keep everything in sync. If you can make a cut every 30 minutes or hour, it seems like that would help.

Honestly, all of this is good for me to learn because I have had my eye on the Edirol R4 as a future audio recorder. However, now I'm wondering if I wouldn't be better off with just a mixer, except that means that everything has to be just right when recording since there would be no option to mix the audio levels in post like there would be with the 4-track Edirol or something similar. I wonder also if there are any audio recorders that will allow some sort of manual tweak/compensation on the internal clock to allow it and the camcorder to remain in sync for a long time.

Please keep us posted Dale on your solutions since those should be of interest to anyone following in your footsteps. Thanks, Shawn

Hi Jack and thanks for the comment.

I think I know what you are talking about.

The video preview (particularly on an external monitor) will always be slightly out of synch with the audio due to latency of the various hardware devices BUT I was trying to use the camera's audio track as a reference point to the 'quality' audio recorded via the mixer and THIS is what goes out of synch. In other words - no matter what the latency of the relevant audio device is I am comparing audio with audio.

Thanks for the input.

Regards,

Dale.

Dale - I just looked up your mixer and I'm now realizing that you're using this to capture 6 separate channels (or more) using Vegas and that your mixer is essentially your second clock that's going out of sync. I think that Brockett is primarily talking about an analog mixer with a separate audio recorder, but if you can send a combined signal to your camcorder that still should help with sync in Vegas. I'm guessing that with 6 channels of audio it would be impossible to snip the audio like I was thinking, so stretching the audio now sounds like the best option. The only thing I would suggest is that you find out how long it takes until the Alesis and your camcorder go out of sync and then determine if you can shoot in shorter takes of audio and video just to keep everything in sync. I would appreciate hearing how you solve this. I also would like to know how you like the Alesis mixer as this could be much cheaper than the Edirol. Thanks again, Shawn

Hi Shawn,

I just want you to know that I really do appreciate the interest shown.

Personally I still think that the article is spot on. I mean - if you used a Creative Labs Nomad, for example, to record a stereo main mix out of an analog mixer made up of multiple inputs the Nomad (according to the article) could go out of synch anyway. The only difference between that setup and mine is the fact that my recording device is a notebook and the Alesis mixer sends the audio as data via firewire.

One thing I did not do was to actually click on a button that says 'Set Clock Master' in the Alesis Control Panel prior to recording this last event. I can only assume that by not doing this the notebooks clock was being used as the master clock although I fail to see how this would make a difference.

Another thing that I did not try was to increase the audio buffers in the Alesis Control Panel but again - there were no audio dropouts in over two and a half hours of audio (believe me - I listened to each track in its entirety more than once to check) - and all that increasing the number of buffers would do would be to ENSURE that there were no audio dropouts.

I have, however, learned much from all of the input from everyone on the board and I therefore now are not entirely convinced that the clocks went out at all - this conclusion reached after much pondering on the subject. The reason that I say this is that because of the (potential) delay caused by the distance between the camera's mic, the public address system, the individual presenters, and my using the camera's audio track as a reference for synch purposes the possibility exists that although my ears may 'think' that I have aligned the audio tracks perfectly at the beginning of the first tape a slight error at this point on my part would be compounded by the end of the first tape i.e. after an hour of video. In addition to this, using the camera's audio track as a reference, once again given the above (potential) delay, the audio / video (lip) synch could still be out even if the clocks did not stray at all and the camera's audio track was in perfect synch with the mixers's audio track.

I still need to try a few things:

1) Generate my test tones immediately at the beginning of each video tape - in close proximity to a) the camera's mic and b) a spare mic assigned an additional channel on the mixer so that when I get back to the studio I can 'visually' align the audio tracks and theoretically everything else should line up perfectly (unless there is of course this clock synch problem).

2) Hook up another transmitter to another line out of the mixer and send this audio to the camera. This would certainly prove / disprove the 'drifting' clock theory but you would, of course, still be relyant on your audio perception to align these tracks correctly. Come to think of it a combination of 'beeps' and this extra transmitter might well do the trick.

3) User Acid Pro 6 instead of Vegas (although this should not make any difference I'm sure).

4) Try Steinberg CuBase LE that was supplied with the Alesis mixer as a comparison test.

5) Not use the mixer at all and just plug a mic into the notebooks mic in and then try and synch with this track.

Left any test out???

As far as the Alesis FireWire 16 Mixer is concerned:

It is a great, high quality, unit and has many features that work perfectly for me for example: you can mute the individual tracks being sent out to the public address system without affecting the FireWire output (to the notebook). The only thing that I find to be a problem (and I think this comes more from my lack of experience with the mixer than with the mixer itself) is that the gain knobs for each channel are EXTREMELY sensitive i.e. you can turn them up almost full and there is not much gain change but that last little bit really makes a huge difference with just a slight adjustment but I am sure that with a little more use I will get the hang of it. Other than that it appears to be quite robust and has all of the controls / filters / eq's etc. as per the specs. I have also checked whether or not it induces extra noise into the audio tracks and it would appear that if there is noise being induced by the mixer I either cannot hear it or see it.

The only thing that really upset me was the fact that it is supplied / bundled with Steinberg CuBase LE which only allows you to record up to four tracks simultaneously. As far as I am concerned this really is a con - nowhere on the Alesis website / packaging / manual are you informed of this and I did try to take Alesis to task on this and they were not interested in the least bit (you won't believe the answers that I got back from them). As far as I am concerned - make the customer aware BEFORE purchase so that they know that they are in for an additional cost if the software does not meet their requirements OR charge more for the mixer and include the necessary software.

The above is, of course, not a problem if you have Vegas / Acid Pro but what about the individual that does not have any other software and only after purchase finds out that he now has to go and spend quite a large sum of $$$ for additional software to be able to use the mixer's full capacity.

That aside - it works really well (for me anyway).

Regards,

Dale.

Hello again Shawn,

Sorry - I did not even see your previous (longer) message - I think I must have been replying at the same time that you were sending it.

Your are right - my problem is that I shoot events which can sometimes be up to four hours long (with one break somewhere in the middle) so what I do is just start the mixer recording a couple of minutes before everyone arrives - after checking the levels of course - and just let it run until the event is over and then (theoretically) it should be (should have been) and easy job to just pull up the video segments (tapes) (I never stop the camera) and synch them together (the only obvious breaks which I have to cut is when I have to change camera tapes).

I can't use the camera's audio (even if I take it straight from the mixer) for the simple reason that I have to edit each individual audio track (by edit I mean doing things like normalizing a track and passages within a track) in order to compensate for different levels of different speakers or presenters. This (theoretically) should not be necessary but in practice it is impossible to adjust the level of each mic / input on the fly and get them all just right.

Due to the above problem I started another thread somewhere on this board looking for a way to either compress / limit a 'too strong' signal PRIOR to recording or enhance a 'too weak' signal PRIOR to recording. I have looked at the Acid Pro 6 Demo and this talks about 're-wiring' a device. As I understand this it means that I could take the inputs from Acid Pro 6 - send the audio to another software package - apply a compressor / limiter / expander and then send this audio back to Acid Pro 6 and record it BUT I cannot use Vegas as a re-wire device, I cannot re-wire Acid Pro 6 to itself, so I need MORE software (like Steinberg's CuBase BUT the full version) in order to accomplish this!!!

Regards,

Dale.

...I still need to try a few things:

1) Generate my test tones immediately at the beginning of each video tape - in close proximity to a) the camera's mic and b) a spare mic assigned an additional channel on the mixer so that when I get back to the studio I can 'visually' align the audio tracks and theoretically everything else should line up perfectly (unless there is of course this clock synch problem)...

Dale, you should generate a test tone at the end of the tape as well. This should reveal to your satisfaction if there is a clock synch issue, and how much it is.

If (as I suspect) you are dealing with clock synch issues (or timebase errors)... such timing variations should be consistent. Meaning, if your Alesis clock runs at 100.05% speed of your camcorder's clock, it should always do so. As I wade through the many posts on this thread, there seem to be so many possible explanations explored, now you seem to not be as sure as you were that there even *is* a timebase error?

Timing with test tones at the beginning and end of a tape will reveal this for sure.

And here is an *easy* workflow if you have a test tone at the beginning and end:
Put the camera video/audio up on the timeline.
Trim the clip to the beginning of the first tone, and the end of the last test-tone, that is you'll have Tone-Program-Tone on the timeline.
Put the 2nd system audio up on the timeline.
Do the same trim.

Now you have visual references for everything you've been dealing with, which should make synch easy. Ctrl-drag the end of the audio track to match the end of the video track. If you've dropped a marker at the end of the video track, the audio will snap to it when you Ctrl-drag.

In fact, you don't need to wait until your next panel discussion to do this, just run a test whenever you can leave the camera and laptop sitting for an hour. This exercise should cut through the haze.

Then, there is the other timing "error", the speed of sound in the air, or offset. Sound reinforcement engineers deal with this frequently. Here's a chart I use:
20 feet = 17.9 milliseconds of delay
30' = 26.8ms
40' = 35.7ms
50' = 44.6ms
60' = 53.6ms
70' = 62.5ms
A millisecond is a thousandth of a second (.001). So, to apply a correction you would determine the difference in distance to the presenter of your camera's mic and the 2nd system mic.

Or, do as suggested above and record the second system output on the camcorder. Which will fix any offset errors, but NOT fix any timebase errors.

But I'll repeat. If lip-sync looks good, it is good. Lessee, a 30' offset = abt. 27ms = about 2/3rds of a PAL frame.

BTW, Quantize to frames has no effect in capture, only in editing on the timeline.

... The only difference between that setup and mine is the fact that my recording device is a notebook and the Alesis mixer sends the audio as data via firewire.

One thing I did not do was to actually click on a button that says 'Set Clock Master' in the Alesis Control Panel prior to recording this last event. I can only assume that by not doing this the notebooks clock was being used as the master clock although I fail to see how this would make a difference.
Dale - I don't think that your PC's clock is being used. My impression of how all this works is that your PC is solely a storage device that is accepting already digitized data from your Alesis mixer. Your Alesis mixer must a have a clock since it is converting analog audio to digital and sending that out as data via firewire.

I looked up the Alesis manual online (http://alesis.com/downloads/manuals/MMFW12_16Manual.pdf) and the 'Set Clock Master' option in your Alesis mixer is for using an external clock. On p. 38 of the manual it states "Setting the master device: If multiple audio devices are connected, one must be designated as the clock master." Take a look at this post (http://www.dvinfo.net/conf/showpost.php?p=247006&postcount=5), because ultimately you might want to do something similar if your camcorder will sync with your Alesis.

The steps that you have mentioned are ones that you should do, especially to get rid of the sound-travel-delay to your camera, but the best solution IMHO is to ensure that there is no drift. Since you tape for such a long time, it would be suprising if there isn't some sort of drift due to using two different clocks, so syncing the camera and the mixer is another level of control. Thanks for the info on the mixer, Shawn

Good Morning - and thanks for the input.

Seth:

now you seem to not be as sure as you were that there even *is* a timebase error?
I probably should not have made this statement as I did. No - I am sure that there is a drift. All I was saying (trying to say) is that if I 'audibly perceive' that I have correctly synched the audio track from the camera to the mixers recorded audio tracks then - if I was slightly out - over time this would make a difference. However - this would only hold true for the audio / visual synch and not for the audio / audio synch i.e. even if my 'perception' of synch was indeed slightly off for the audio / audio synch at the beginning of a video tape I would also 'perceive' the audio / audio synch to be the same at the end of a video tape / hour and I can tell you that by the end of an hour I am no longer 'perceiving' the audio to be in synch with the audio - if you know what I mean.

This thread is not only making my own head spin - it's keeping me awake at night!!!

But you are right - I need to test this out - and I will do it today - and post the results.

Shawn:

A lot of other things are now beginning to fall into place for me as well.

Just for example: I recently had a problem with lip synch / external FireWire monitor preview / Vegas / Surround Sound Projects i.e. when I first started 'messing around' with mixing surround - sometimes (a lot of the time) I would play the clip (previewing on the external FireWire monitor) and the audio / video synch would start out OK (there is always a slight difference between the external FireWire monitor and the sound card) but would eventually drift way, way, out even over very short clips. I played around for days, buying different sound cards, messing around with DirectX and hardware acceleration etc. etc. and eventually found that the solution was to enable the midi ports in Vegas (any of the ports) and then set them up under the 'Synch' tab for (in my case) 25fps etc. etc. The problem - as if by magic - dissapeared there and then and has not reared its head again. Now - sometimes - the external preview will 'freeze' for a split second (while the workstation is obviously trying to do something else) and then go again but it obviously now 'catches up' to the audio at the right place and continues on in synch where as before enabling / setting up the midi ports in Vegas - the monitor never 'froze' now and then but the audio / video would just go further and further out of synch. Why this only happened on Surround Projects I do not know but there it is!

I digress - but for a reason.

I had a look at that thread and it really does (also) seem to be a possible solution i.e. the MidiStream from Kenton. The only thing that I cannot fathom out is how you would connect the MidiStream to my new notebook i.e. there does not seem to be a game / midi port anywhere on the notebook (and I did not specifically look for one either as it is the last thing that I thought that I would need). Be that as it may - thanks to your explanation - using the MIDI clock as a 'master clock' sure seems to be a good way to do things and like I said in my 'digression' above using the MIDI clock did solve another problem of mine.

Anyway - I am going to run some of these tests today - and I will post the results - as soon as they come off the press!

Thanks to both of you for the input.

Regards,

Dale.

I had a look at that thread and it really does (also) seem to be a possible solution i.e. the MidiStream from Kenton. The only thing that I cannot fathom out is how you would connect the MidiStream to my new notebook i.e. there does not seem to be a game / midi port anywhere on the notebook (and I did not specifically look for one either as it is the last thing that I thought that I would need). Be that as it may - thanks to your explanation - using the MIDI clock as a 'master clock' sure seems to be a good way to do things and like I said in my 'digression' above using the MIDI clock did solve another problem of mine.
Dale, The MidiSream (or Midi Time Code) does NOT connect to your new notebook - it connects to your Alesis mixer (if the Alesis has that jack). Seth seems to know more about this than me, so please tell me if you think I'm on target Seth. The Alesis is a digital mixer (or at least has digital output), so it needs the sync. The firewire connection is just there to create a link to your computer for storage/download, but I can't imagine how your notebook's clock would be used unless the Alesis was polling the notebook's clock (which seems unlikely). However, even if that was the case the Midi cable would still go to the Alesis mixer because it would then sync to the camcorder's clock provided you choose the 'clock master' option for the Alesis (even though that is done using software on the notebook). Here's another thread describing five methods, one of which is related to that post: http://www.sonyhdvinfo.com/showthread.php?t=4621 - To recap: since the mixer is sampling the analog audio and outputing digital audio, it has to use some clock to do that (whether it's own or polled from the notebook, or from the camcorder). Therefore, I'm 90% sure that you need to connect the Midi cable to the Alesis Mixer. Does this make sense Dale?

P.S. I just looked through your manual, and I'm not seeing a midi port anywhere, so this may all be for nothing. If it does have MIDI then great. If not, you want to figure out how they connect mutiple mixers and then see if you can use the connection with your camcorder to sync them together. It's now sounding like this may not be possible, but there's always hope.

Hi Shawn - thanks for that information - really appreciated.

I have not had a look at the last link you posted yet - I am busy testing this stuff.

These are the results of my first test:

Workflow:

Start notebook recording
Start camera recording
Place handheld Sony wireless mic next to and in line with camera mic
Generate a couple of very short 'beeps' (you would not believe what I am using for this)
At about 17 minutes into the recording generate a couple of very short 'beeps'
At the end of the tape (about one minute to go) generate a couple of very short 'beeps'

Upon completion - align initial series of 'beeps'

Result:

Camera video / audio (end of last 'beep') ended at 91,911,993 (absolute frames)
Mixer audio (end of last 'beep') ended at 91,907,971 (absolute frames)

Stretched the mixer audio and the everything aligned. This is good because it at least means that the clock differences are at least constant over time.

I must just say that aligning the audio tracks with these 'beeps' is a cinch - really easy and accurate particularly if you magnify the waveforms - you can be accurate to individual peaks and troughs - not bad.

Next result coming up.

Edit:

Something has dawned on me while I idly sit here waiting for another test to complete - is this not going backward? I mean - this reminds me of the old days of capturing analog video - you had to specify a master stream (be it audio or video) and then hope that eveything stayed in snych with no dropped frames etc. etc. and that you did not run out of disk space!

Regards,

Dale.

Hey Dale,

A couple of comments. First, to mention your problem with your gain control. The reason it affects differently the higher you turn the knob is that increasing gain is not the same as increasing volume. Increasing volume is at a constant rate. Increasing gain acts like increasing a per centage - the higher the db increase, the higher the overall per centage increase. It works sort of like an exponential increase so as you twist the knob, the more sensitive it is to change. Knobs are the cheaper way to manufacture than sliders.

Second, I'm far from an expert but if you haven't tried sending a stereo line into your camera, bypassing your camera mic, I think you are creating more headache for yourself. I have an Alesis 24 track recorder and when I capture footage to my camera, I record audio in from the mixer to replace the camera mic and I NEVER have an issue with sync. I think if you try this, it will solve your problem. If you need the pa recorded but it doesn't go through the mixer, add another track to record by putting a crowd mic close to the mixer and you can mix it in post.

I have also gotten deep into Vegas surround sound and there are some issues of settings that are not really covered in the manual. Notably the center channel defaults to pan type instead of constant power. That means by default, the center channel gets sent to bothe the front right and left which was making my front right and left clip. Switching the pan type on the center channel to constant power keeps 100% of the channel to the center channel. You can tell this is happening when you solo the center channel and it plays through the front right, left AND center monitor mixer.

Good luck.

Jeff

Hi Jeff - thanks.

1) Gain control - now I understand the difference.

2) Mixer - I (probably) have the same mixer as you. My problem is that I have to edit / mix / 'fine tune' as it were the individual tracks for my final product - I can't just take a stereo main mix out directly to the camera live - that is why all of this is happening.

3) Vegas Surround - I just had a problem with the audio / video synch when previewing on an external monitor and enabling the midi ports / synch seemed to solve the problem. I have noticed what you are talking about though.

On with the tests!!!

I have just recorded another hour of audio / video but this time not using the mixer - just the notebooks onboard sound card - with one wireless mic plugged straight into the notebooks sound card - and following the same workflow as described in my previous post.

The interesting thing about this is that the camera audio / notebook audio is out of synch almost EXACTLY (I would go so far as to say EXACTLY) the same amount as it was with the mixer after an hour. I do not believe in coincidence! Something is telling me to look elsewhere i.e. maybe not a clock issue at all???

Any thoughts???

I am now going to play around with some Vegas settings and I think that I should also try using my editing workstation to record the audio via the mixer just to eliminate the possibility of the notebook having a problem (although as a control I captured the exact same video tape used in my first test from the camera using the notebook and that was also out by the same amount i.e. if the notebook itself was at fault or inaccurate then the video capture on the notebook would have matched the audio originally captured using the notebook and mixer).

Regards,

Dale.

Just another thought (full of them today).

When I capture video with Sony Capture XPress 6.0d I always get a report at the end that says something like 'Average FPS'. I forget what the figure is now but I do know that I have NEVER got a figure of EXACTLY 25fps after the capture i.e. it is always something like 23.976 and the like no matter what workstation I use.

I have never actually worried about this figure before (only the dropped frames part which I have never had).

What is actually being reported on?

Is it possible that THIS could have something to do with this synch issue?

I mean - the last thing that I would ever question would be my cameras - but - maybe they are not ACTUALLY getting EXACTLY 25fps (for these tests I am using my VX2100E but the problem originally occured with the FX1E). Is that possible?

I suppose I could plug a mic into my editing workstation and another mic into the notebook and record just audio using Vegas and see if those two align exactly after an hour. This would certainly eliminate the mixer / notebook combination from the equation.

I hope that this is interesting for members of this group (and OK by the moderators). I mean to say - I hope it is OK for me to be posting my thoughts / queries / test results as they come to me. If not - please be polite and let me know - and I'll stop taking up the space.

Shawn - by the way - no - the Alesis does not have a midi port at all - so I don't think that that will work. But - if there is a way to connect the MidiStream to the notebook (maybe there is a PCMCIA card that will allow this connection) then I don't see why the Alesis control panel will not let you choose the MidiStream as the master clock and your camera will already be synched to the MidiStream - if I understand this all correctly that is!

Regards,

Dale.

Dale Dale Dale,

Please try sending a signal straight to the camera. A mixer should have outputs for each channel you are recording AND an output of mixed stereo. Make the camera record audio from the mixer AND record discrete tracks to be mixed down later. Then you will be using the stereo mix for the reference and adding the 5.1 channels for finished audio.

Jeff

Dale,

Don't EVER think you are wasting space. This forum is to help everyone and God knows I need it to. Hang in there.

Jeff

Hey Jeff, thanks for that.

Sorry - I did not ignore your suggestion about sending the audio to the camera from the mixer AS WELL. The problem is that it is not going to make any difference if one of the clocks is 'slipping' or has a timing difference. Think about it. The video and audio on the camera WILL be in synch whether the audio is coming from the cameras mic or from the mixers output - no doubt about it BUT the individual tracks as recorded on the mixer WILL STILL be out of synch with the video and audio recorded on the camera - all you will have is a different audio source. If you are meaning that you can then use the audio from the camera as a reference point to synch the mixers recorded audio to the cameras audio / video that is a different thing but I have found that my 'beeps' at the beginning and the end of the recordings are far more accurate.

My problem is not so much that the stuff is going out of synch - this we already know (and we also know that it can be fixed by 'stretching' the audio by four frames of video for every hour in my case).

My problem firstly is to know WHY (and the general consensus so far seems to be timing differences between the different device clocks) AND how can I ensure that it does not happen AND exactly WHICH device is to blame.

I have just done another hour test - this time disabling WDM in the Alesis Control Panel and turning off the automatic detection for recording device latency in Vegas - have to wait for the video capture to finish before I know the results.

Assuming that the above makes no difference I think that the best test that I could do next is to set up the notebook to record audio using Vegas via the mixer and set up my editing workstation to also record audio using Vegas. If these two tracks align perfectly that would really eliminate the mixer / notebook and at this point I have a strange feeling that they will in fact be perfectly aligned.

Unfortuanately - I just have that kind of mind - WHY is the question? This could keep me awake for days!!!

Edit:

Sorry Jeff - I just read your message again. At any given event I am recording a MINIMUM of six channels (which have to be edited afterward). Just to check that I am not misunderstanding you - how am I going to send six discrete channels plus a stereo mix to camera in real time? I can tell you that the mixer only sends each discrete channel to the FireWire port - otherwise they are all mixed to the various stereo outs.

Regards,

Dale.

I was thinking that the audio from the mixer into the camera would be identical in stereo to that of the 6 discrete tracks so it would have to sync just fine. I guess I'm out of help. Good luck!

Jeff

OK - well - here endeth the lesson!

For my last test - I plugged a wireless receiver straight into the sound card of my editing workstation and another into the mixer connected to the notebook (easiest test as both receivers were set to the same receive channel and the same mic transmitted to both of them). Opened Vegas on both with identical record settings and let rip for an hour (with my 'beeps' at the beginning, middle and end i.e. after one hour).

The result - exactly the same difference over an hour i.e. between three and four frames difference per one hour.

What this tells me is that it is NOT the mixer that has the problem or is causing the issue but the notebook. The audio streams are different by the same amount, using the same software, using the identical settings, and it does not matter if the sound is recorded via the mixer or straight into the notebooks sound card - the difference is always the same.

As I hoped - this also means that the cameras do not have and are not causing these issues.

Bascially - after all of this - I can't help but get the feeling that I have not really accomplished anything. All I have proved is that with my combination of hardware I know that my audio is going to be three or four frames behind the video after one hour. Maybe tomorrow I will do the same test with my editing workstation and a spare editing workstation to see if the difference between these two workstations is the same.

The real problem is that when doing these tests it was a simple matter to 'stretch' the mixers audio at the end using the 'beeps' but in practice I can just see this becoming a nightmare. What if you forget to put the 'beeps' at the beginning of the tape / recordings AND at the end of tape / recordings and worse still - what if you are stop / start recording (video)?

Anyway - any input overnight will be greatly appreciated.

Regards,

Dale.

Good Morning!

This is driving me nuts!

This morning I had a look around for some solutions and I came across a thing called 'Horita Wireless Time Code System' - Model No. WTS100M.

Anybody had a look at this?

From what I gather these units (or at least these units in combination with some or the other unit of theirs) can send the timecode from one camera to other multiple cameras so I got to thinking - maybe they could send a timecode from the camera to the notebook and this would somehow synch the notebook / mixers clock to the camera?

I cannot find any documentation on these things other than the usual sales pitches.

Anybody tried these things (or something similar)?

Am I way off here?

The only obvious thing to me is that the audio captured via the mixer does not have a timecode but maybe there is some way .....

Regards,

Dale.

HEY - I JUST HAD A BRAINWAVE!!!

I am making this assumption:

That all (Sony) DV Camcorders are created equal (at least as far as their clocks go).

SO:

What if I connected a 'spare' Sony camera to the notebook via a LANC to MIDI Cable and then set up this camera as the master clock and the notebook (or mixer) as the slave. Would this not ensure synch of the mixers recorded tracks?

Please - this idea I REALLY need input on.

Regards,

Dale.

Likely not, Dale. No two anythings are exactly identical in their clocks without a synching device. Working with DV without genlock or other master clock and only being 2-3 frames different after an hour is pretty good, actually. But, if you're dedicated to your quest...sounds like it's keeping you out of trouble.

At any given event I am recording a MINIMUM of six channels (which have to be edited afterward). Just to check that I am not misunderstanding you - how am I going to send six discrete channels plus a stereo mix to camera in real time? I can tell you that the mixer only sends each discrete channel to the FireWire port - otherwise they are all mixed to the various stereo outs.

Regards,

Dale.[/QUOTE]

What I meant was that the stereo out of the mixer should be on the same time as the six discrete channels. Record the six channels and send the stereo mix to the camera. You record via firewire AND use a 1/4 to XLR to the camera. The audio should be timed the same so the audio on tape will be the reference track for when you load the other six wav files. Make sure you have your camera set to receive the mixed signal and not just plug the xlr in and think that's your source signal.

Jeff

Hello again,

I have not yet managed to find a solution for this issue.

However - I did contact the company called Kenton who make the MidiStream referred to earlier in this thread. I described the problem and according to them the MidiStream will not solve my problem and they advised me to contact a company called Black Box Video. I have had a look at their website but have not found anything that I think will help nor have I contacted them yet.

I have, however, ordered a LANC to Midi Time Code Generator and I am waiting for this to arrive.

I am doing (what I think to be) and interesting test at the moment.

I have just used three of my cameras to simultaneously record my 'beeps' for an hour (one tape each). I am interested to see if all the cameras will actually stay in synch with each other (without additional help). The real purpose of this test is to see whether or not I can rely on one of them to be a sort of 'master clock' i.e. using the LANC to Midi Time Code Generator and make the camera the master midi clock during the audio recording process on the mixer / notebook.

One statement that I made has been bugging me as it was misinformation:

When I capture video with Sony Capture XPress 6.0d I always get a report at the end that says something like 'Average FPS'. I forget what the figure is now but I do know that I have NEVER got a figure of EXACTLY 25fps after the capture i.e. it is always something like 23.976 and the like no matter what workstation I use.

I have never actually worried about this figure before (only the dropped frames part which I have never had).

What is actually being reported on?

Is it possible that THIS could have something to do with this synch issue?
I'm embarrased!

It was really late (early) that morning and I was not thinking! The figure that was changing was the data throughput not the frame rate i.e. I ALWAYS get an average frame rate of 25fps - this does not vary. If I was not getting 25fps I would have REAL problems!!!

Sorry for that.

Regards,

Dale.

Hello again,

As promised - here is my update.

I simultaneously recorded one hour of tape on three different cameras and - guess what - three different results i.e. taking the FX1E as the 'standard' - the VX2100E was slightly ahead and the TRV27E was slightly behind.

Sorry Douglas - I just had to prove it to myself!

Anyhow - there goes my idea about using a 'spare' camera to generate a 'master clock' - whether it be a MIDI clock or anything else - right again Douglas!

I must tell you that this is extremely disturbing to me. If anything I thought that the cameras would be 'spot on' as it were.

It means in reality that even with a three camera shoot (let alone seperate audio) not only do you have to worry about cutting to the desired camera shot but even that shot could be out of synch with your 'master' track. How about that!

The worst part about it is that even if I tossed all of the equipment that I have and bought Z1's I would STILL have a problem with the audio synch from the mixer!

Basically the only way to get this right is to ensure that the sound guy gets the mix JUST RIGHT and send this mix to the camera while shooting BUT this is by no means a reliable or failsafe method of doing things. I have just ordered another two Sony UWP-C3's to send the main (stereo) mix out to one of the cameras - more $$$.

Anyway - I think that I have now put this baby to rest (not really) - and sincerely hope that the next time someone tries this they at least have a very good idea as to what they are in for.

Thanks for the input from everyone.

Regards,

Dale.

Basically the only way to get this right is to ensure that the sound guy gets the mix JUST RIGHT and send this mix to the camera while shooting BUT this is by no means a reliable or failsafe method of doing things. I have just ordered another two Sony UWP-C3's to send the main (stereo) mix out to one of the cameras - more $$$.


Dale I've been intently watching for your posts. What you say in the above paragraph is what I was suggesting all along. By the way you wrote it, it appears to me (sorry if I'm wrong) that you don't understand the concept. Please, anyone else chime in.

Dale, if you are having the mixer record six discrete channel to hard drive, you can still output a stereo mix of that to one of the cameras. Because the mix and the six channels are real time and your camera records in real time, send the stereo mix to a camera. That sound HAS to be sync'd with the video. Then match up the three video tracks and then match up the grouped set of 6 tracks to the stereo mix sent to camera one. When those two are sync'd, delete all three camera audio's, even the stereo mix (unless you want to keep it for reference only - then mute it) and you project should play fine. If this won't work for you then I am the one that doesn't understand and I'll go back under my rock.

Jeff

Hello Jeff,

I am glad to see that someone has not dismissed all of my hard work out of hand! :)

Now - I do not mean to argue with you BUT:

I know that you have been correct all along by saying that if I take a stereo main mix out (that is what it is called on the Alesis) from the mixer straight to the camera then the audio will obviously be in perfect synch with the video but if I cannot rely on the quality of that stereo main mix out i.e. if I need to edit those individual tracks - then I have a problem.

Put it this way - I send a stereo main mix out to the camera and the audio is in perfect synch with the video ON THE TAPE - no question. Now I have the audio on the tape as well as on the notebook as recorded via the mixer. Again - no problem so far. But when I then capture the tape from the camera and pull the individual audio tracks from the notebook recorded via the mixer into the same Vegas Project I am back to square one i.e. the mixers audio track is out of synch with the camera's audio / video track. If I want to replace the camera's audio track with the mixers (edited and better quality) tracks then I still have to manually synch the tracks before discarding the camera's audio track.

If I did not have to edit or tweak the individual audio tracks there would be no problem i.e. I would not even have to record the audio track on the noteboook - I could just use the audio on the camera's tape but this will never be the case so even although I have audio perfectly in synch with the video ON THE TAPE when I capture this tape to an editing workstation it will be out of synch with the, let us call it, the 'master audio' recorded on the notebook from the mixer.

Make sense?

Actually - on second thought - I think that we are talking at cross purposes here. I think that you are saying that by doing it your way I will have the camera's audio track as a reference to synch the audio tracks on the notebook recorded via the mixer - quite correct. That is true except for the fact that it is not so easy to synch the audio to the audio without my 'beeps' - I am trying to eliminate this 'manual synch' step altogether.

Regards,

Dale.

Are you positive you are recording 24/48 and not 24/44.1?
Jeff

Hi Jeff,

Definitely 48/16 but have also tried 48/24.

It is definitely this clock issue.

Anyway - I'm sure that somewhere down the line there is a solution which I WILL find!

Regards,

Dale.

Jeff,

Dale is experiencing two different synchronization errors that are under discussion here:

1) Offset error, produced by the time it takes sound to travel through the air to Dale's camera mic. vs. on-stage mics through wires to mixer and pc. This would be perfectly addressed by the wireless link you're suggesting.

2) Timebase error, produced by the clocks of the digital mixer and the camcorder running at slightly different speeds. This is not addressed by taking mixer audio to camera. This can be fixed in two ways:
a) Shrink or expand the duration of the mixer audio track in post until it matches the duration of the camera audio track. Made easier by beeps or other references that are visible in the waveform display. Inexpensive, and not very difficult once you've done it a couple times.
b) Upgrade camcorder and mixer (or digital recorder) to equipment that accepts timecode input and generates timecode output. Then, slave the audio recorder to the camcorder. Costs well over $10,000 but less than $20,000 for camcorder and audio recorder, depending on audio recorder selected.

To repeat, timebase error is not fixed by taking mixer audio to the camera, wireless link to the camera fixes only one of the two problems Dale is experiencing.

Why? Imagine two film cameras. Film a scene (with motion!) with one camera at 12 frames per second, and the other at 24fps. Take the processed film reels, and put them up on two projectors side-by-side and show them both at 24fps. The original 12fps-reel will show in half the duration at twice the (apparent) speed. This is timebase error, the two clocks were far from synched, it affects speed of action and duration when the content gets synch locked in projection, or on the timeline.

In this example, we had a 50% speed/duration difference. Dale's errors are much smaller, on the order of less than .1%, but that's enough to throw lipsync visibly off over an hour of content when the two audio tracks are locked to the soundcard's clock.

I've done a lot of synching in my time, with no beeps. Not so hard to fix offset error or timebase error in post. At least that's my experience, your mileage may vary.

Good Morning.

Thanks Seth for that explanation - explained as only a master could.

There is one other thing that is bugging me and would just like clarification (just for interest sake):

Correct me if I am wrong but even with cameras that have 'free run' timecode like the Sony Z1 you will still experience the same problem. Would I be correct in making this statement?

The reason I ask is that during my search for a solution to my problem I saw a thread somewhere (it was referring to the Canon GL2 but I think that you can follow the same procedure with the Sony Z1) that detailed how to start the 'free run' timecode with the remote on multiple cameras and then when it came to post things were just a lot easier as each tape (scene) had the correct corresponding timecode tape for tape / scene for scene.

After demonstrating that not one of three of my cameras run in perfect synch to each other I make the assumption that even with three or four Sony Z1's there would STILL be a difference between each cameras timecode i.e. timebase errors (learning all the time) over a long period. Correct? The 'free run' timecode would just make it easier to synch multiple camera shots IF AND ONLY IF the multiple shots were not of a lengthy duration.

Also - just for the record - I think I understand what a 'free run' timecode is but what is a 'record run' timecode?

Something else that has just come to mind: Does the difference between my cameras (for example) mean that they are not shooting at EXACTLY 25fps (PAL)? During capture when Vegas reports that my average fps for the capture of an entire tape is 24.98fps or maybe 25.03fps is it this timebase error that is being reported and should resampling each cameras video / audio track not solve the problem (although I did try this with no change in synch between the video /audio tracks from each camera)?

I know that none of the above has anything to do with the problem that I am trying to resolve - just want to know for interest sake.

Regards,

Dale.

Dale,

Free-run is handy, but, as you suspected, doesn't help with the offset and timebase errors you're experiencing.

The default for most prosumer camcorders is record-run, which runs more like a tape counter. Typically it starts at 00:00:00;00, what's recorded at the end of a one-hour tape will be about 01:00:00;00 or a couple minutes more, as tapes usually are 1-3 minutes longer than their stated capacity. So, the timecode generator lays down nearly continuous timecode on tape. (***edit - regardless of whether you've started and stopped the camera***)

So, the timecode generator only rolls when the tape is recording.

Then there is preset, which interacts both with record-run and free-run. Back in ancient times (as few as 10 years ago!), we were doing lots of linear editing with tape machines and editing controllers - they did NOT like seeing the same timecode twice in a project! Usual strategy was to "preset" starting timecode at hour 1 for tape 1, hour 2 for tape 2, etc, and use record-run.

Free-run means that the timecode generator rolls continuously, regardless of whether the camcorder is recording or not.

Free-run has all sorts of applications, but the one we're mostly interested in is so-called time-of-day timecode. Let's say it's 8:30am on the shoot day, and we've had our coffee and doughnuts. We'll preset all devices to 8:35am, switch them to free-run, then, whoever has the good wristwatch or the loudest voice will count down the last 10 seconds to 8:35, at which point everyone starts their timecode generator rolling.

Voila, all the timecode recorded now shows the time-of-day. With prosumer gear, that provides only rough sync - if done carefully, we're better than 10 frames, but that will still require a tweak in post, and, code will drift through the day. We might sync a few times if we care, but usually if this approach gets us within a couple seconds the editor can find the syncing points.

With pro-gear and a master timecode generator that is continuously wired to all devices, this is always frame-accurate. Usually in a multi-camera studio it's cameras, not camcorders, and all the tape decks are in a control room where they are frame-locked always (with master TC or house black, but that's another subject...)

The 'free run' timecode would just make it easier to synch multiple camera shots IF AND ONLY IF the multiple shots were not of a lengthy duration.

I do this all the time with music recording, very similar to what you have done with your alesis digital mixer/laptop. Yes, I'll roll continuously over an hour tape, or even a 3-hour tape in a studio deck. I always have to correct for offset error, and sometimes have to correct for timebase error, depending on the accuracy of the individual clocks.

My attitude about this is that we now have $3-10,000 USD cameras doing most of what we used to spend $40-60,000 on, only with better pictures. I'm so amazed by the fact that I can personally own equipment and an NLE with these capabilities (instead of being tied to a multi-million dollar facility) that I don't worry too much about having to do these sorts of corrections.

You're concerned with how elegant the solution is - I'm very impressed that the beast is walking at all.

Does the difference between my cameras (for example) mean that they are not shooting at EXACTLY 25fps (PAL)? During capture when Vegas reports that my average fps for the capture of an entire tape is 24.98fps or maybe 25.03fps is it this timebase error that is being reported...?

A frame is a frame, that is, it has its own begin and end information. So when you put it up on the timeline it's duration becomes (PAL) 1/25th of a second. If you lay it back to the Z1 after editing maybe it will be plus or minus some fraction of a percent different.

When does this matter? When it goes into large distribution systems like broadcast, cable or satellite. When there is a video switcher in use, and you want it to cut between sources glitch-free, which means cutting during the vertical interval between frames. And so, there are timebase correctors and/or frame synchronizers that retime frames during playback.

But it doesn't matter much for DV in general, because we usually either distribute direct to the consumer on DVD or VHS, or, we hand it off to a broadcaster who routinely fixes timebase errors during playback, because all tape machines have these errors - they are mechanical devices. I suppose things will be somewhat different when there are no tape players in use.

Seth - Thanks for your great explanation of various time-codes. So is there anything on the prosumer level that has time-of-day time code?