Michael Gissing
Forum Replies Created
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Is the LTC continuous? A broadcast wav file would have been better as it has timecode embedded within. I think from 5.1.x onwards, broadcast wav embedded timecode has been supported.
Ask the audio house if they can provide a broadcast wav. If not and the LTC code is continuous, ask them what the first frame of timecode is at the file start of the mix. Put the mix into your sequence at that code and start editing.
If you need to generate an EDL then you will need to set the timecode of that clip. I don’t know how as I haven’t needed to do that, but I know it can be done in FCP.
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Eli,
Don’t be lead astray by PPM speak. Broadcast levels are speced in dbfs which is “decibel full scale” and refers to digital peak meters, which are what FCP uses. Also digital record decks use this scale too.
Broadcast specs vary slightly from country to country but the golden rule is to make sure peak levels do not exceed -10dbfs. This will work for practically all PAL and NTSC countries. Reference tone on a broadcast tape should be -20dbfs for NTSC and -18dbfs for PAL.
For the web, you can use full scale if you wish, peaking up to but not going over 0dbfs. You don’t need to do this though. Always give some headroom in digital in case your meters are not accurately showing short peak excursions.
Follow the peak rules, not how loud it sounds on your speakers. Next make sure your speaker/ amplifier combination gives you plenty of volume to monitor with so that you are not flat out. Again you need headroom in your monitoring so that distortion you hear is not monitoring distortion but program distortion. Once you have a comfortable monitoring level mark it off with a texta or chinagraph if you want to be retro. Finally, don’t mix too loud. It is fatiguing and gives a false impression.
Remember the wise words of Stanley Kubrick – Films are 60% audio and 40% picture. That from an image obsessed old deaf bugger.
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Dialog norm is just metadata. You still need to have your actual levels compliant with maximum peak levels.
Broadcast specs are fairly universal but can vary by regions. In NTSC land maximum peaks should be -10dbfs (decibels full scale on a digital peak meter). In PAL it is -9dbfs unless it is for the BBC then it is -10dbfs. To be safe, make sure your audio limiter is capping all signal at -10dbfs, then set dialnorm to -23 as Jeremy has pointed out.
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No.
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Mixed together – nah. Girls scream from Soundogs or other online FX libraries – easily sourced.
Get another scream, or ask a friendly female to do one for you.
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I doubt that the white levels will cause buzzing as the signal is going in via composite or component. Maybe if the connection is RF?
If in doubt, chuck a broadcast safe over the whole show. It is likely that the mpeg compression will be better if whites are at 100 IRE as there is less range to deal with.
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Did you capture from the DVCPro deck and convert to DV for the offline or did you dub to DV tape and edit? Did you use the deck to downconvert to SD or a card like Kona or Decklink? Did you capture via firewire or SDI?
I have been told that the Panasonic decks in PAL are buggy and particularly using firewire. I have only ever captured from them once and via SDI. The audio was 2 frames out of sync but picture code was correct. Never found out why, just told that PAL is poorly implemented in Panasonic decks.
As mentioned did you check the deck offset value in 5.1.4 and is it the same in 6?
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I don’t thinks so. What happened when you tried it?
If you have particular clips that you want to steal attributes from, open a new sequnece and copy the shot into it. Then toggle between the two sequences. I make the tracks really big with thumb showing so it is easy to pick off the steal shot. Handy for a long form show with lots of returning interviews that have the same grade, or cropping/ distort values etc.
Alternative, for saving just filters, I make new bins named for the interview clip and drag the filters into that bin from the clip in the viewer. The trick is to name the filters so that they are sorted into the right order. Easiest is to rename the filter with 1, 2, 3 to make them sort in order. Then you can highlight multiple filters and drag/ drop them onto timeline files. Other wise they will apply in the alphabetical order that they sit in the bin.
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I have only had that problem once before when a client hired in a portable digi beta deck and dumped out 13 half hour programs and then trashed his media. We couldn’t insert the audio on our Sony 2000 digi beta cause the portable had locked in the audio as stereo only! This prevents most digi machines from being able to insert edit on the audio tracks, even the stereo tracks as the data blocks on tape are incorrect.
Did you or are they using the Sony portable deck (can’t remember the model number offhand)? I know it is typical to blame audio people and that they often blame pic post people but a more positive approach is to work out why and inform. So please tell us if you sort this out.
The fix btw was to transfer all programs from my J3 to the 2000, via SDI and insert the audio onto the new tape via AES at the same time.
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If you can hear the sound then it can be recorded. For example, the output of my Win Media Player comes out my sound card as an SPDIF signal. I can re-record that on an external recording device.
I also have a variety of free downloadable software like Goldwave that will demux audio from QT and other video wrappers.
In short, no you can’t stop someone with access to the internet from finding a way to separate the audio from the video.