Jean-christophe Boulay
Forum Replies Created
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From experience, using the pen tool to edit levels will be quickest once you get the hang. In the audio tool you’ll only be albe to record one fader move at any time and what you require is two simultaneous fader moves per edit. Once you read your waveforms right and know what levels you want, you can edit much faster than real-time, with a few touchups left. Don’t lose time grabbing faders with a mouse.
IHTH
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
This sounds like the classic old FCP audio bug. What framerate is your timeline? Is it the same frame rate as the currently-selected Easy Setup? First intuition is that they may not be. If that’s the case, remove the faulty audio from your project, change to an Easy Setup using your working framerate for the project and reimport the audio, preferably from a fresh copy of the file. FCP assigns a framerate to audio and takes it from the Easy Setup. When you take it to a timeline with a different framerate, it tries to pullup/down the audio to match, taking out of sync. As stupid implementations go… The fact that PPro works fine with the material tends to point to an FCP particularity. This has been covered many times on here.
If that’s not the issue, an audio sample rate mismatch could be the culprit but I’d have trouble believing someone well-versed in audio wouldn’t have tried to solve that one in the various mix versions used to troubleshoot. You are working with 24-bit/48kHz AIFF or WAV files, correct? If not, do.
I’d bet a drink your problem lies in one of those two causes.
IHTH
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Hi Chris,
They’re probably using a hexadecimal editor, like HexEdit, to close the file that has been unterminated by Quicktime, which should recover the data recorded up to the crash. You really have to know what you’re doing to make sense of the operation. This would be a good time to have a coder friend. I’ve had one of mine do something similar for me and all he asked for was a beer. Maybe some digging on Google can get you by.
AtomBox gives you access to the QT file’s structure, with built-in hex editing and could be used to repair the file if you know about atom structure. It runs on Windows. https://www.jongbel.com/?page_id=114
It’s up to you if your time twiddling about with this is worth 300$ or not. I’d look into a more rugged recording software for future gigs.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Like the others said, there is no such thing as Final Cut for Windows (or Linux, Unix…). If it is really FCP and is really running on a computer that is not made by Apple, it’s running on a Hackintosh, a PC tweeked to run Mac OS.
Be very wary if this is indeed the situation you’ll be facing. I’ve worked on one rock-solid Hackintosh system that was very impressive, even compared to a real MacPro. The other ones I’ve seen were quite unstable and in no shape to run FCP reliably.
The software would be exactly the same and operate in the same manner exactly. Before it bugs, that is. The possible operational difference would be the keyboard being mislabelled in the Cntrl-Alt-Option area. On another note, a Hackintosh is legally dubious, especially in a teaching environment. At least one EULA is being violated.
IHTH
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Just to specify, the Item Properties in FCP can absolutely be trusted for audio. I’ve never seen it wrong nor have I heard any stories about that.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Hi Joe,
The audio settings being set by FCP when conforming a timeline to the imported clip have a very tenuous relation the actual audio settings of the clip being imported. You can import 16-bit/44.1kHz all day and FCP will set that timeline to 24/48 every time for most codecs. Hence the recommended practice of having everything in the correct format before importing to FCP to edit, which is not your error, lucky you.
So you can’t trust FCP on this. QT must be right. Trusting FCP for audio is rarely rewarded with success, in my experience. Once you have your final product imported to FCP, on a timeline set to its video settings and with the audio (incorrectly) set at 24/48, you just have to check your Audio Outputs tab is set allright, render that timeline to be sure, which should take next to no time at all, and export that to a self-contained movie. That should give you a movie with the same video parameters and with 24-bit audio.
As an aside, I very much doubt that the file “needs to be 24-bit to play back correctly”. Any contemporary system that plays out digital files would have to have been set up in the most ham-fisted manner possible to not be bitrate-agile. But if I start telling people to not follow specs, night will turn to day and it’ll rain frogs, so you follow those specs!
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Exporting a 5.1 mix as discrete mono files does not give a “Dolby file”. That would be an AC-3 file.
If you want a “Dolby file”, you must take your six-discrete-channel mix to Compressor. In Compressor, there’s the “Add Surround Sound” button that you click on. You’ll get a dialog to select the mono audio file from your discrete-channel mix to assign to each channel. After hitting OK on that dialog, your surround audio is entered as the job source. You slap a “Dolby Digital Professional 5.1” preset on that and the resulting file will be the .ac3 that is often referred to as a “Dolby file”. Many parameters have a significant impact on the output result so making a copy of the preset for editing might well be necessary depending on input material and delivery specs.
IHTH
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Shane,
That tweet a while back about channelling your inner-Zelin? Getting there. Keep it up.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Hi Don,
You just need to go set your audio bitrate, subtract that number from 1000kbps and you get the video bitrate you can run while keeping the combined throughput under 1000kbps.
You’ll probably want to use AAC compression on the audio if that fits into your specs. It’s impossible to hit that bitrate without compressing the audio anyways, as 24-bit/48kHz stereo PCM is more than 2000kbps.
IHTH
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Hi Elliot,
The problem you were having with your files speeding up when changed to 48kHz is that you were not converting the sample rate, only changing it. I haven’t used Audacity enough to point to where you made the mistake, but the result you describe is pretty clear cut.
In some software, it is a bit too easy to export material at a different sample rate without conversion. This affects pitch and duration, as the same number of samples is being played back on a different time base. When going from one sample rate to another, it’s good to look for the word “conversion”. That’s an assurance the operation will create the missing samples or drop the superfluous ones.It’s the audio equivalent of the difference between adding pulldown or conforming in Cinema Tools to change video frame rates.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com