Jean-christophe Boulay
Forum Replies Created
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Hi Alex,
There should not be any particular issues with what you want to do. Audio sample rates and video frame rates really have no relation at all with one another. The sync issues people run into are usually due to manipulation mistakes and settings mismatches. You look pretty set.
The one thing you should check before importing any audio into FCP is that the video framerate in the selected easy setup is the one you’ll be editing at. No matter what your sequence settings may be, FCP checks the framerate of the easy setup and will pull-down if it doesn’t match the sequence. That one gets a lot of people into trouble. You may need to go check your audio output device settings in the Audio Midi Setup in Utilities if you get playback problems. Aside from that, all should work fine.
IHTH
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
The bridging services are mainly the two big ones Ty mentioned, Digifon and EDNet. I’m not familiar with their service offer in Europe but they have connected me to studios in France. It is an additional cost layer, and quite a substancial one on long sessions. In my situation, the calculation was clear that it was not financially wise to ditch the ISDN system and link up to ISDN-only shops via bridging, but we run quite a few remote sessions. It might be worth it if you only run a few sessions and stack up recurrent savings by not maintaining the ISDN lines. We mostly use the bridging services to connect our ancient beige AptX box to shops running Telos or some newer codec. We’re lucky in that our clients are ad agencies with comfy budgets who don’t mind picking up the extra tab if everything works smoothly.
In the group of studios we regularly hook up to, the critical mass to abandon ISDN really isn’t there yet and I don’t foresee that changing super fast. I’d be in a right pickle if I had to choose wether or not to buy a new codec now.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
We use it quite a bit and the experience is mostly positive. There are a few niggles here and there but in practice it’s pretty seamless. Penetration is becoming better as more people go for it.
Most people seem to go for the standard version instead of Pro and there isn’t that much point in being the only side of the connection with Pro as the bitrate will have to match on both ends, most people don’t activate the QManager (which enables Auto-Replace and Auto-Restore), nobody I’ve seen uses the apt-X codec inside SC and we rarely if ever connect to more than one client or use VPN. I’d really check with studios you’d be connecting to what version they run as the price difference is quite significant.
It is more stable in standalone mode than as an RTAS plugin. I haven’t used it or set it up on non-PT systems so can’t speak to the stability of the VST/AU versions. Most people seem to run standalone but we prefer the plugin. Once in a while it loses contact with the RTAS engine but a quick transport start/stop solves that.
Sound quality at good bitrates is very good and we rarely replace with the uncompressed versions. Some basic knowledge of networks helps for set up. You’ll have to do port-forwarding through your router, for example. QManager features require that you have an FTP set up as well. Certification is included with the Pro version, but I’m not convinced it would be worth the fee if you get Standard. It doesn’t seem to be a big factor in getting more gigs. SC plays nice with ISDN bridging services so you should theoretically be able to connect to pretty much anyone through it. I’m keeping the ISDN lines, though. The Remote Transport Sync feature is a bit niggly and requires some coordination between both ends before the session starts but once you get it to work properly it’s pretty nice. Bandwidth demands are not too scary and we often run two SC sessions on the same connection without issues.
All in all, it works very well but you should really examine your needs before choosing which version to go for. The standard version pays for itself within a few sessions, especially if it saves you from using a bridging service.
IHTH
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Mark, you’re much closer to being an audio guru than you give yourself credit for. It’s all there.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
I’ll just throw in the obvious one for measure: in the View menu, is External Video set to All Frames? Cmd-F12 toggles it off so you might have done that by accident in some finger-stretching exercices…
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Hi Mike,
As long as you don’t do anything to alter the audio information itself, there should be no problem at all. Adding the tracks and exporting shouldn’t do anything to affect the audio. It’s wise to ensure you maintain the exact same bit-depth and sampling rate, though it can be possible to get away with that. You could even just add the tracks to a stereo ProRes export in QTPro if the timings are exactly on, which you have to trust they are since you can’t really monitor. If ever there are issues with your export and it gets flagged, you may be able to get the post house that did the Dolby E for you to integrate it to the file themselves. A bit costlier, but the responsibility is on them and they can monitor. All in all, it should work fine.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
I may be making it sound more complicated than it is in practice with my usual focus on details. Once you grasp the concepts, the application is rather straightforward.
Good luck!
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
I think all the troubles are related to the Dialog Normalization (DialNorm) data point of the AC-3 mixes.
We’re in a bit of a transitional period with regards to broadcast digital audio right now with the imminent implementation of the CALM Act in the US. Broadcasters, set-top box manufacturers and TV manufacturers are all at a slightly different step of this process and the results can be quite unpredictable, as you’re learning. All the theory is laid out in this year’s most gripping thriller read, ATSC a/85 (https://www.atsc.org/cms/standards/a_85-2011a.pdf). It’s a boring but important read.
As Wikipedia will tell you, Dialnorm is the meta-data parameter that controls playback gain within an AC3 stream. Originally, you’re supposed to set it to the average dialog level of the encoded program, as read in dBFS. For commercials, that doesn’t work so the overall level in dB LKFS is the level used to set it. That is laid out in section 5 of ATSC a/85. At some point, you’ll need a metering solution to measure LKFS so might as well sort that now. The DialNorm value of the AC3 stream for a commercial should be the positive integer of its LKFS reading, as set out in section 4. If you do that, you should be good.
That doesn’t mean the whole chain will be good, not by a longshot. If the stream goes through broadcasters, they could have their servers and encoders set any number of ways depending on where they’re at in their infrastructure switch to digital. If the stream passes that step unharmed, it arrives at the set-top box. If that box will decode the AC3, its internal settings like Loudness, Late-Night Mode and Dynamics come into play. If the set-top box passes the AC3 to the TV, that could be decoding and applying its settings. Then again, maybe the TV passes the encoded stream to the main audio receiver or amplifier’s decoder, which would apply its settings.
So you never know what the broadcast chain is doing to your level, and there is any number of things the consumer’s system could be doing to the level, all based on that little metadata point that is DialNorm. You can’t make any relations between an AC3 stream that’s gone through broadcast and a consumer’s system and your original export in your room.
Right now, some broadcasters ask that you specify your DialNorm level during the ingest process, some require it to be a certain value, so you have to mix to that level from the get-go. The vast majority don’t even mention it and that’s where the problems start. If there is any way of providing straight PCM audio, that would probably be the most reliable now as the encoding and its effects are on the broadcaster or distribution service and you can wash your hands of it as long as you mixed to broadcast levels, which you are since older TVs that disregard metadata get the correct level.
So I’m quite sure DialNorm is what you need to look at, though the Channel Mode meta-data point could also affect level, so you should check that too. Read the boring technical papers ATSC A/85 and maybe ATSC A/53, measure your levels in LKFS, set the DialNorm accordingly and it’s out of your hands after that.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
MXF is pretty much the same. You can’t export a timeline using stereo outputs to MXF, though, so all the outputs will be set to Dual Mono and you won’t do any of the panning shown in the tutorial. Just ensure all eight of your tracks route to the correct output and export. Works like a charm.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Sure you can. Go over to Apple’s support site and type in “10.6.7 combo update” in the search box and you’ll find it. The combo updates contain all the previous ones as well so you don’t have to do them all incrementally.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com