Frank Nolan
Forum Replies Created
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When something that has been working well for some time suddenly doesn’t anymore it can sometimes be tracked down to something on the system that has been changed. Have you upgraded ANY software lately? Changed hard drives? Plugged in extra drives on the FW bus? Or added a FW hub? Anything at all changed?
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If you mean the speakers have inputs for 1/4″ or XLR, well as long as the 1/4″ inputs are balanced i.e. they take the TRS (tip ring sleeve) jacks then there is no difference. It should state this near the inputs or in the docuentation specs of your speakers.
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>However if that was the issue shouldn’t I see the horizontal shift during the shorter ouputs as well?
Not necessarily. As you said, in longer transfers it is random and sometimes happens further apart than others, so in therory shorter pieces may be fine. Blackburst is a composite signal with a horizontal sync pulse. So from your description of the horizontal shift, it would appear that the lack of a refernce between your equipment could be the cause of the problem. Then again it may not be, but I would at least hook up a blackburst generator to see if it solves it.
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Are you feeding blackburst to the AJA and the BVW-75?
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YOu will not get any better quality capturing from a DSR-11 through the AJA Io in fact you will lose quality and timecode. The dsr-11 only has s-video or composite analog outputs and no RS422 control. Also to answer your #3. Using that setting would still give you a dv capture not 8 bit uncompressed. Your best bet is to work with the footage in dv, captured over firewire. Then if you need to deliver on beta sp you could then drop your timeline into a 8 bit uncompressed sequence and render and print to tape or just print to dv tape and make a beta sp dub off that which would probably be quicker.
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>Its just a short piece 3 min final edit and I’ve got loads of room. I’m not going to redigitize it later. So I want to bring it in at a high quality.
In that case use the betacam US 8 bit component uncompressed setting or if you are adding grapics you could use the 10 bit setting. Then make sure your sequence settings match the capture settings and you wont get the render bar.
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Uncheck that box and see if that helps.
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Peter,
I think from a previous post you said you have a mackie 1202 vlz. If that is the case on the back next to the main outs there is a button that says +4/mic level. Try switching that and see if that gets you closer to the correct level. To set your gain structure correctly you first want to run a tape in your beta deck with a 1k tone, if you went in ch 1-4 on the mackie adjust the trim knob to “u” ch’s 5-12 have no trim. then set your channels knob to “u” and your master fader to “u”. If your tone on the beta is at “0db” on the machines meters then you should be at 0 on the mackies meters. If not then adjust the channels gain control on the mackie until the meters read “0db”. Now depending on what bit depth you are using for your audio capture settins in FCP, you will want the meters in the clip settings tab to register -12dBFS for 16 bit or -20dBFS for 24 bit. So adjust your sliders in the clip settings capture window accordingly. As a side note it would probably pay you to read chapter 3 of volume 3 of the FCP manuals to give you a better understanding of analog Vs digital audio. Also are you aware the capture setting you have chosen in the AJA set up is only giving you DV quality not beta? -
It will depend on your sequence settings and the clip settings your are trying to put on the second track. Also you may want to try unlimited RT and playback set to dynamic.
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I dont think that is possible yet.