Jean-christophe Boulay
Forum Replies Created
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SSD’s make kicka$$ system drives, but are a very very bad choice for media drives, where many successive write-delete-rewrite cycles will occur. With each cycle, they get slower and slower until their mechanical counterparts trounce them. For a better explanation than what I could possibly fit in this post, you can have a read here: https://techreport.com/articles.x/17136/2. Or google “block rewrite penalty”.
Take note that Mac OS does not have TRIM support yet, which would attenuate the problem. Apple seemingly has no plan for it in coming updates either, in typical “if it doesn’t start with the letter i we don’t really care” fashion. So on top of being bloody expensive, SSDs as media drives are technically a bad idea. For once, techs and accountants can see eye-to-eye.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Linear PCM is the uncompressed option, which gives greatest quality and compatibility. I’m not sure how your 16 channels will fold down to mono, though. That could be pretty tricky. Usually, the channel allocation in the metadata is used to matrix the mixdown. As there is no true standard for 16-channel audio, there is probably no flag that could define what has been done with those 16 channels. If they are seen as 16 discrete channels and not a standard configuration, they will probably all be mixed down to stereo, in the center of the image, which is effectively mono. That could sound like a trainwreck.
I’d recommend you use Pass-Through for your audio, as the Uncompressed already contains Linear PCM in the correct channel configuration. It’s already what you want it to be.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
The post audio software is almost irrelevant. I know PT inside-out and mix with it a lot, but if you plump me in front of an unknown system running Logic, I’ll probably get equivalent results out of it. The constant between the two situations is the knowledge of audio. Equipement is secondary. With the Apple ProApps, you’ve got pretty much all the tools you need to produce quality audio. What is likely limiting you is your knowledge of audio production techniques and audio itself, as well as your listening environment and metering.
You could invest into training and completely rebuilding a part of the house as a control room, buying pro VU’s and PPM’s and selling a kidney to pay for speakers, but I think that would be silly. As a videographer, your audio efforts and investment should be focused on capturing the best location audio possible. Eliminating the problems before they occur is the best time investment you can make right now. For critical mix jobs, I suggest hiring a pro mixer. He’ll do the job much quicker than you could, offsetting part of his cost to you. The fact that you don’t have to invest into a dedicated mixing room is quite non-negligeable financially, as well. On top of that, the overall level of production being better than if it’d been done by one person alone, you can maybe up the price of your services a bit.
Audio post is a lot more involved than many video guys seem to think. By having a few people each practice their specialty, the results will always be a lot better, and surprisingly less costly. There is a reason you never hear a one-man-band playing Chopin.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Those must be SD2 audio files, right? Despite its popularity in the good ol’ days of OS9, support for SD2 was dropped completely, largely because of the issue you”re facing now. SD2 files are very reliant on their ressource forks, which can be stripped way too easily by transferring through a FAT32-formatted system or most internet and network transfers. You then end up with a bunch of files the system can’t identify.
It’s been quite a while since I’ve had to deal with this situations but I used to do it almost weekly at one point. One tool I remember using is SD2 Fixer (https://mac.softpedia.com/progDownload/SD2-Fixer-Download-40152.html). It used to work for me and I don’t remember any particular problems with it. Once you fix the files, I strongly suggest converting them to a more future-proof file format. Or present-proof, actually…
IHTH
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Have you tried enabling JamSync?
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
It depends on the studios, really. We did two yesterday and one today. Some places haven’t done any in a year. You seem to be in the second group. In that situation, I don’t think it makes much sense to keep it, since the alternatives are now up to par.
A good alternative to ISDN is SourceConnect from SourceElements (https://sourceelements.com/source-connect/). We’ve been using it for a few versions already and it’s now very stable and seamless. It behaves like a Skype-on-steroids, with your contact list and connection parameters in a single window. You even get a chat window to diss the talent with the other engineer. Fun!
Most studios we connect with get by fine with the Standard version. The Pro version adds higher bitrates and functions to automatically replace compressed audio with the uncompressed audio when you aren’t recording. Really, the compressed audio sounds fine. The Pro versions can be purchased with an option to use the APTX codec, but it isn’t worth it, as almost no one has it.
If you have to connect with studio that are still running ISDN, most bridging services now offer SourceConnect-to-ISDN connections. If you have few remote sessions, the bridging costs will be a small fraction of the money you’re not spending on ISDN lines anymore.
On the cheaper side, I’ve run lots of sessions through Skype, with the client directing through that. We then send the session over for the engineer on the other end. That’s more of a replacement for a phonepatch than for an ISDN, but works well in a pinch. The client can be absolutely anywhere, with this method, not necessarily in a studio, which saves them loads of money. They tend to like that.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
There are two schools of dealing with the LFE channel on the mixing end, so there could be two good answers to this. It will depend on the material you have in hand.
Either the mixer removed the bass from the full-range channels and sent it all to the LFE himself or he mixed for bass-managed systems by leaving some bass in the LCR channels so the end-user’s amp routes that bass to the LFE through its bass management system. The first option has a lot more traction these days, but I still see some mixes with quite a lot of bass in the full-range channels from time to time.
If you listen to the individuals L, C and R channels and there is some material below 250Hz, you could probably drop the LFE channel. If those channels don’t contain bass material, you should include the LFE in the downmix. In that situation, -10dB should be good.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
I feel I need to rectify some info in here. A FAT32 partition will not cause any problems at all on a Windows machine. All versions of Windows, including 7, have full support for the format. The problem you’ll have with FAT32 is the 2GB filesize limit. If all files will be less than 2GB, you can use Disk Utility to format the drive to “MS-DOS”, no problems there.
In fact, if you format your drive to NTFS on a PC and bring it over to a Mac to transfer your files, you’ll discover the problem Mac OS has with NTFS: it is read-only. You will not be able to transfer your files to the NTFS drive. You can only transfer files from it.
If you need to use NTFS because of file sizes, you should use a third-party NTFS driver for Mac OS. One was listed earlier in the thread. I used Paragon’s NTFS for Mac (https://www.paragon-software.com/home/ntfs-mac/). There’s a free fully-functioning trial period IIRC, which is great if you only have this one project to settle. Otherwise, it’s 20$ for a full license.
If you are on Snow Leopard, you can natively enable NTFS read/write with this little maneuvre (https://hints.macworld.com/article.php?story=20090913140023382). As with anything using the “sudo” command in Terminal, only attempt if you understand what you’re doing.
EDIT: Even better, here’s a small app that will complete the above manoeuvre easily. https://ntfsmounter.com/
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
I’ve been involved in the mixing of a few albums and, usually, no decision is taken lightly. That reverb has probably been discussed, rediscussed and rerediscussed. If it’s there on the final, it’s that the decision was taken to pile it on there. The mixing engineer maybe didn’t like it either, but the producer / artist / label / whoever called the shots wanted it on the final. That is the song. You’re editing the video for that song.
With all due respect, it’s not your place as an editor of the music video to ask for a remix. You can talk about it subtly to the caller of the shots if you have a good relation with him, but a few music industry types I’ve worked for wouldn’t take kindly to that while others would welcome the input. It’s you who knows who you’re dealing with. Bringing up the fact that the sound and visuals don’t match may raise a flag no one thought of, so you should inform them, but you should be proposing things you can do on your end to make it work. If you ask for a remix, you will very likely be ignored. Remember, the song is the product, the video is the advertisement.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
FCP can’t edit picture on a grid smaller than 1 frame, of course, but it does for audio. 1 frame of video at NTSC framerates is still over 1000 samples, so there is ample material to fade in. A quick test with a 1-frame fade on a sine wave in an FCP7 timeline does give a clear ramp when importing to ProTools, not a square cut. It does not behave as a 1 frame video fade-in would.
Craig, if you opened a “teacher” session with short fades like this on every region, it’s surely as a safety against clicks. While FCP will fade the audio between frames, it won’t let you cut it between frames. This means that audio edited solely in FCP is more likely to have clicks, as you can’t cut at the zero-crossing point unless it’s smack dab on a frame boundary. Batch fades of very short duration just ensure the conformed audio is click-free. Those fades can’t possibly be heard and they safeguard against something that definitely can be heard. Almost all OMFs I get from FCP editors for mixing jobs have theses fades at the head and tail of all regions. I’ve never had to remove them that I can remember, either. It lessens the chances of us playing the “chase the click” game in audio post. It’s good practice, IMO.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com