Michael Gissing
Forum Replies Created
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Until you have compulsory voting on a weekend with lots of polling booths and preferential voting, the USA can never claim to be ‘democratic’. In what other country can voting Green get a Republican in?
Even then, like here in Australia, the government is only as good as its opposition. But please get out and vote. Don’t let the Apathetic party win again.
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If you have shot progressive there is nothing to deinterlace. I think you are seeing motion jerking that progressive frames have when move speed and shutter speed combinations lead to a strobing or stepping.
The camera op should have seen the same thing in the viewfinder. Shutter control on the camera can help reduce this. What happens when you just view the footage straight onto a monitor, not going via FCP? If it looks different, then it may be FCP sequence settings.
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>>The sample rate effects the speed of the audio?
Effectively yes. If you take an aif file @ 48khz and slow it down to 47.952, it runs .1% slower. I suspect your sound post people, working at 24 fps are speeding up the program by that .1% . Slowing the mix down by that ratio should correct sound. Calculate the frame drift over time and see if it comes to +.1% . It is my best guess for the sync drift and it hopefully is a simple fix.
Ask you audio guys if they can provide a mix file varisped down the .1%
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5 frames over what duration? Have you tried varispeeding the mix file .1% (ie from 48khz to 47.952khz)?
I suspect it is the difference between 24 and 23.98 which a simple varispeed will fix. If correct, it will then be pitch correct when it is back in sync.
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I am in Australia, but what is relevent is the region coding on the DVD. Make sure your DVDs are set for all region playback. Although many in Australia are used to playing region 1 disks with DVD players that are set to ignore region coding, there may be many instances where players are not ‘adjusted’.
If the disk is region free then it will play on all machines here in Australia, computers included. Most of us have monitors that autoswitch to NTSC.
The real question however, is NTSC safe anywhere? I read the daily horror stories here about 3:2 cadence, 24 or 23.98 versus 29.97. PAL is so much easier!
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My G5 has an SPDIF digital output (optical). Modern amps often have an SPDIF input. Use that instead of a 3.5m jack. Also as suggested have the firewire/ deck monitoring on another input.
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OMFIs don’t transfer dynamic gain, panning or FX like EQ anyway. When I import FCP or AVID OMFIs I choose to ignore clip gain. So I always tell editors not to bother removing mix automation or clip gain as it doesn’t translate anyway.
Check with your audio post people as they will tell you if they want clip gain or not, or if you actually need to remove it. Also they may really want everything, in which case you will need software like Automatic Duck to transfer all the stuff that OMFI ignores.
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Digital audio in video land should be recorded at 48khz, 16bit or 24 bit. The sample rate is critical. In your workpath the 44.1khz is being converted by FCP to 48khz. YOu then send that out as a QT @ 44.1khz (another conversion). Then when you bring it back into FCP it gets converted back to 48khz again.
Much better to stay in 48khz at all times to avoid the multiple conversions.
Regarding the Flash – if the audio remains as PCM 16bit then it will be the same as the VO (providing it stays @ 48khz). I know Flash can compress audio. It depends on the settings.
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FCP has clip gain. Open in viewer and the level adjust is at the top. Also the mixer tool can be used. For extra gain add the compressor filter and there is extra gain that can be applied. Set compression to 1:1 so it is just gaining.
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A much simpler trick is to add the compressor filter to quiet clips – don’t use any compression ( ie 1:1 ) but there is a heap of extra gain you can add.
To answer your question in an earlier post; as a sound facility, I don’t need normalised clips for mixing. I have so much gain in my digital mix path that normalisiing would be unneccessary. Also a normalise render in FCP may not be floating point processing. 32 bit floating point gain, as you can expect on most digital mixers, is much better than 16 or 24 bit normalising.
Did you do a test to make sure the rendered normailsised clips had handles? If not then you are making a nighmare problem for sound post. I suspect you are going to a lot of trouble for a rough mix for pic editing. Try the compressor filter gain trick instead as this doesn’t need to render or affect the OMF.