Jean-christophe Boulay
Forum Replies Created
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Hi Cindy,
There are different surround formats, not all of them involving Dolby, so it’s important to first determine the destination of your mix.
If this will go straight to DVD, you can encode your mix to the Dolby AC-3 format with many media applications, including Compressor. The AC-3 format has its own downmix facilities, so it will fold down the 5.1 mix to stereo if only stereo playback is available. Some people rely solely on that, while other also include a dedicated stereo mix on another audio track of the DVD. Both work, and the AC-3 folddown usually sounds pretty good.
If this will go to theatres for screening, things get a little more complicated. You have to know from the theatres which format they want. More and more of them are asking for DCP (Digital Cinema Pack), which are normally created by a specialist post-house. In that case, you have to send out the discrete audio channels for the creation of the DCP, so Dolby are not involved in the process at all on your end. The ingest department of theatre chains can usually refer you to a post house for the DCP.
If an analog print will be created, you’ll probably need a Dolby DMU to create the MO Disk used at the print master stage. Now you’re opening a whole new can of worms. Dolby do not sell the DMU, they lend it to approved facilities. That means you have to establish a contact with your Dolby rep, have them come in and approve your mix room with all their gear. That normally involves some room treatment and EQ’ing, which normally involves a sizeable investment on your part. Then, they will lend you a DMU and you’ll have to pay the royalties for every show you encode. The setup of the DMU and the learning of it is not trivial either. If you have to go down this road, I strongly suggest you find a local sound post house that already has the DMU so they can create your master for you from your final mix. If you can find your local Dolby rep, they’re usually very helpful with this kind of thing and can steer you in the right direction.
If you can send in HDCam SR, you just have to dump your discrete audio tracks on it, so Dolby are not involved. You would be responsible for creating the stereo downmix, in this situation.
The DMU has mixdown facilities as well and creates a stereo-compatible mix from the 5.1 when you encode.
As for aesthetic considerations, you shouldn’t strive to fill in spaces just to fill in spaces. It should be natural-sounding, while steering clear of gimmicky use of the surround channels. This is a much deeper subject that isn’t even settled amongst pro mixers, so it’s hard to give steadfast rules on it.
Good luck. If you have further questions, fire away! If you know your output format, we can streamline this mass of information down a bit.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
There used to be a few little Mac apps that would do this, but none of them seem to have survived the Intel revolution. I keep falling on dead links while searching for them. Those that did survive, like More Missing Tools (https://www.angelfire.com/mac/rnc/files/MoreTools.html) don’t split the AC3, they just convert it or fold it down to stereo. I’d go for the clumsy method and be done with it.
You understood the FCP manoeuvre well. The 2-pop is just a sound that comes two seconds before the start of your program, usually to sync sound and picture. If you can find any sound that is present on all tracks at the exact same time (gunshot, hit, explosion…) you can check that your exported audio files are OK by checking if that sound is still well aligned on all channels. It’s just a safety.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Hi Rhodri,
The ProApps are great for creating DVD’s, not so much for reverse-engineering them. You can import an AC3 file directly to FCP, so what I’d do in a pinch is to take the AC3 to FCP, create a sequence with a single mono output and drag the individual sound channels to it to export each as a mono AIFF or WAV. A 2-pop would ensure sync when recombining them, though there is no reason for sync to be affected at all.
Alternatively, you can open the AC3 in QTPro, go to its Properties (cmd-J), select the Sound Track and go to the Audio Settings tab. There, you can access the setting of the audio track, which include the output assignment. Any channel directed to “unused” will be muted, while any channel directed to “Center” will be mono. You can set all channels to “unused” and toggle them to “center” one by one to be exported as a mono audio file through QT’s Export dialog.
Not particularly elegant processes by any means, but they’ll get you there.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Do the two sources sound the same?
If you’re using any kind of distortion effect, the output of those will vary a lot depending on the level of the input signal, so if one source has a lower level than the other, that could explain it. You’d have to match levels before sending through the effects chain. If more than levels vary (different mic positioning, different room) then you have to match the complete tone before the same effects chain will yield the same result.
If the two sources have identical character and level to begin with, you have more of a mystery on your hands. What happens if you merge both clips into one and effect that clip? If the difference is still there, it’s down to the sources being different, if the difference disappears, it hints at a difference in the effects chains. That should help to at least pinpoint the issue.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
That sounds like you may be using the general Core class drivers rather than the device’s specific drivers. Have you installed the latest drivers off the Focusrite website? (https://www.focusrite.com/support/saffire/saffire_6_usb)
What names does the selected output have in STP’s preferences, or the system AMS? That could be a hint.
Focusrite products are missing from Apple’s compatibility info for ProApps (https://www.apple.com/finalcutstudio/resources/supportdevices.php) but so are other interfaces I’ve seen working. All Core-compatible interfaces should work, in theory.
Latency occurs during any kind of playback, but is doubled while in input-monitoring situations, which is why most people associate it with recording. Non-optimized drivers are often the culprit with unusually high latency values. The click in the audio hints at other possible issues like clocking, though. Are there no glitches at all after that click?
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Hi Lina,
From what you describe, your initial hunch of looking at USB mics seems to me to be the right one. Other microphones would require a mic preamplifier, to amplify the mic signal to usable levels, and a computer sound interface, to bring the output of the preamp into the computer. The Shure X2U regroups the two. A USB mic basically has all this built in. Since the internal components are tailored to the particular mic, you get better sound quality for an equal economic investment, at the expense of versatility. For an investment similar to an SM58 and an X2U, you can get a USB mic that might well sound better. Don’t get me wrong, the SM58 is an institution in audio, but not for its sound quality. It’s a mic you can drop, step on, scream into, spit into, mix drinks with, hurl down stairs, roll over with your car… you get the picture. It is a tank, and you’re looking for a family sedan.
To record voice, the vast majority of engineers prefer condenser microphones. They have greater sensibility and will pick up nuances that a dynamic mic of the same category might not. Your concern about picking up unwanted signals is entirely legitimate, though. This will be addressed with the polar pattern of the mic. The polar pattern basically describes how much sound the microphone will pick up from various directions. The word you’re looking for in features is “cardioid”, which is a pattern that picks up the signal from the front of the microphone while minimizing how much sound it picks up from the back of the mic. Sound bleeding into the mic is inevitable, but the voice will mask all of that while there is signing to be recorded. The silent parts can be edited out.
For positioning, I’m really not a fan of eating the mic for singing. It boosts low frequencies are gives people that “OMG! Listen to my voice!” feeling, but sounds too shut in, in my opinion. Sound, especially musical sound, needs a little space to develop before being captured. The mic should be at least 6 inches away. It should be less than about two feet away, though, as you’re really asking for trouble over that. When you’re eating the mic, your head has to stay perfectly still while recording, which is hard when playing the piano with any kind of expression happening. Placing the mic on the piano should work, but there is the possibility of the vibrations of the piano’s body being communicated to the mic through the stand. If that happens, a small folded blanket or pillow between the mic and piano would decouple them and get rid of the problem. You can also get a decent mic stand for around 40$, which will give you pretty much endless flexibility in mic position. A pop-filter could also be a good buy, to tame the “p”, “b” and “t” sounds. The new mic itself will be a big improvement on the current setup, so you can build up as you find the weaknesses in the new setup.
I’m not in a position to comfortably recommend particular models you should be looking at, as my mic investments usually entail a visit to my banker, but BLUE, AKG, Audio-Technica and Shure are all companies with good reputations that make USB mics. I hear good things about the Audio-Technica AT2020 USB.
IHTH. Good luck in your hunt. If any of my incomprehensible tech-jargon has filtered in here, just raise a flag.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
dB are maths. You can’t really understand what a decibel ever represents if you don’t understand the maths behind them. That explains why so many audio pros don’t really understand them either. A Bell is a logarithmic unit of measurement that expresses a unit’s value relative to a given baseline of the same unit. It’s not just for sound, and even for sound it applies differently wether you’re talking about sound propagation through air of electrical propagation through a conductor. It’s not meant to be easy.
You can work with dBs without understanding them, many do, but if you really want to understand the deciBell, it’s all about maths, formulas and numbers.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
I don’t think Reaper has its own built-in denoising engine. For that, as for most signal processing, it is dependant on plugins. I’m sitting on some very good denoising plugs and do all that kind of thing in ProTools anyways. As far as plugins go, Apple’s own Noise Print is not so bad as a static denoiser, but there are way more powerful (and expensive) dedicated tools out there. I doubt you can do much better on noise removal without some outlay. It’s a particularly pointed market.
IHTH
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Reaper is the one even the pros with ridiculously expensive PTHD systems use. I’m far from the only person running it behind PT to sidestep some limitations. It really is a full-fledged production tool, more so than Soundtrack could ever aspire to be in its present form. And completely free. As in beer.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com