SOUND POST FAQ: The "Bullets"
I have no idea what you don't know -- although I can make a guess based on questions students often ask. So this guide tries to convey the gist of what you need to know to get a decent mix on your LMU film project. If you've taken the intro Sound Design class, much of this will be familiar.
Since you're probably working under time pressure, I'll try to give you the basic facts in short little "bullets." Details are provided later (just click on the links.)
THESE ARE THE BASIC QUESTIONS WE TRY TO ANSWER:
HOW CAN I DIGITIZE PRODUCTION SOUND FROM NAGRA TAPES?
If you've taken RECA 367/353 you should be familiar with the procedure; what follows is just a guide to refresh your memory. While some students elect to transfer sound from Nagra to audio CD and then import from CD into Final Cut/Pro Tools, you can save a step by digitizing directly into Pro Tools to create 48kHz AIFF files (the native format for FCP.) That's the procedure we'll cover here. A transfer cart is available through RECA studio support on the second floor of the SFTV building.
Nagra transfer procedure checklist.
Trouble-shooting the transfer.
I'm playing back the Nagra tape, but I'm not getting any sound into the Mackie or Pro Tools.
First, confirm that you have the Nagra's "Line & Playback" knob turned up so that you're seeing appropriate levels on the Nagra itself. Look on the right hand side of the Nagra and confirm that the output jacks have not been disconnected. On the Mackie, check the Mute/Alt button above the Nagra slice level knob. (It's been given a "light" color treatment.) Make sure it is "up" and not down. If it's already up, check that the Source button is in the correct position (the uppermost of the possible buttons should be "down.") If all that is correct, re-check to see that the input slice level knob is at the detent "12:00" position and that the Main Mix or Master Output to Pro Tools knob is set likewise. Confirm that the separate Headphone/Speaker level is not turned all the way down. If you still don't see level on the Mackie Mixer or hear sound through the monitor, get some RECA tech support.
I see and hear good sound levels at the Mackie but nothing shows up on the Pro Tools audio meters.
Make sure that your audio track is RECORD ARMED (no levels will read on the meters unless the "R" button is red.) This is the most likely culprit. If the track is already armed and ready to record and you still see no audio on the meters, go to Setups >Hardware Setup and confirm that your input source is "analog" and not "S/PDIF." (S/PDIF is for a digital-to-digital transfer from DAT.)
I'm hearing a loud hum/buzz in the monitor.
Check the three banana plugs on the right hand side of the Nagra to confirm that all are connected correctly. Check the head shield on the Nagra; it should be up to shield the playback head from magnetic fields. Do not proceed with the transfer unless the problem is corrected; seek tech support.
I'm hearing distortion on my production recordings.
If you're listening to the output of the Mackie mixer, try switching the headphones and monitoring the output of the Nagra directly. (You may need a stereo-mono adapter for the headphones.) If you still hear distortion, it's undoubtedly a problem in the original recording. If the sound from the Nagra is fine but the Mackie output is not, check to see that the mixer input slice and master output are set to the usual detent position of 12:00 and that the loud parts of your dialog are not causing the "+22" clipping indicator on the Mackie's meter to light. If the clip light is coming on, or if most of your levels are in the "yellow" range on the LEDs, try adjusting the Nagra "Line & Playback" knob to a lower setting.
The pilot flag on the Nagra is not indicating sync.
Check the connector to the external 59.94 sync box. And make sure the Nagra playback selector switch is fully down.
I went to back up my session folder but I don't see my audiofiles in the Audio Files folder.
You may not have had your Disc Allocation set correctly. In that case, Pro Tools has created an identically named session folder on one of the other drives, and that should contain your audio.
About reference levels and how they line up (and when to fudge the settings.)
Why is a -10 dB tone on the Nagra supposed to line up to "0" on the Mackie, but read "-18" on Pro Tools?
While each device uses decibels on its meters, each device uses "0" dB on the scale to mean something different. On the Nagra, "0" represents a very strong peak -- but one that we can go slightly beyond. We can go 4 dB higher than that with a slight edge of distortion. We can actually go a bit higher still, but the tape itself will become saturated and introduce bad distortion.
The Mackie uses "0" to represent "good and strong" but with quite a bit of safety margin beyond that point before we begin to distort. (About +20 dB above and beyond.)
Pro Tools is a digital recorder and uses "0" to mean the absolute maximum. Digital clipping occurs beyond that point.
To deal with these different devices and scales, we have to choose a smart reference point, and "headroom" is the key factor. The Nagra uses a reference level of -10 dB to indicate "good and strong" -- with about 14 dB of headroom or safety margin above that point. We can line that tone up to "0" on the Mackie and our loudest sounds should only go up to about +14 on the Mackie, which is well within the headroom of the Mackie itself. So far, so good. The next step is to choose a point of reference on our digital recording. Logically, we might choose -14 dBFS for our Pro Tools setting. This would work out fine as long as our Nagra recordings stayed in an ideal range.
But if we've botched our Nagra recording slightly and over-recorded the tape, we could end up with peaks somewhat higher, maybe 18 dB higher than our Nagra reference level. This would mean that loud sounds that are already somewhat distorted on the original tape would go into digital clipping on Pro Tools and end up sounding still worse. So we allow that extra four dB of safety margin to help prevent digital clipping. (Since digital is a low-noise medium, it's better to play safe and let the levels be a little on the low side rather than risk digital clipping.)
Why not line the Nagra up to read -12 dBFS in Pro Tools -- after all, isn't that our mix reference level in FCP?
There is a certain logic to that idea but the fact is, the Nagra has more headroom than many of our final release formats, and it is the headroom of the original recording that determines how we "fit" the recording into the headroom of our target media. If we lined the Nagra tone up to read -12 dBFS in Pro Tools, a lot of our loudest recordings would go into digital clipping. (See above.)
When should we make an exception to the recommended line up procedure?
Despite our best efforts to build in enough safety margin, it's possible that you might have extremely loud recordings where your peaks might still go into digital clipping. In that case, back off the level on the Nagra's "Line & Playback" knob.
More likely is a case where your production sound is under-recorded at a consistently low level. In other words, your Nagra line up tone is fine but the production sound itself is on the wimpy side relative to the tone. How will you know? If you follow the recommended line up procedure and watch the Nagra modulometer, you should find that the peaks for well-recorded dialog tracks will end up bouncing around more or less straight up -- from about "-10" to "-6." A shout or door slam might deflect to "0" or slightly beyond. On the Pro Tools meters, this might translate to peaks in the -14 to -10 range. [If you're thinking that this seems a little "hot", especially since we used -18 dBFS as a reference, the difference is largely due to the faster response time of the Pro Tools meters compared to the Nagra.]
On the other hand, if your loudest production sound is peaking well below -10 dB on the Nagra and you're seeing similarly low levels in Pro Tools, you might wish to do a slight boost during this transfer stage. Since the Mackie Mixer and Pro Tools are aligned so that "0" on the Mackie produces "-18" on Pro Tools, it's best to do any fudging with the Nagra "Line & Playback" knob. Try to do this in an intelligent, non-arbitrary way. For instance, you may decide that a 4 dB boost overall is desirable. Go back to your Nagra line up tone and reset the Nagra playback so that the tone reads "-6" on the Nagra instead of the usual "-10." This should produce a reading of "-14" in Pro Tools. Make a note of this correction factor on the sound roll's sound report. This could prove handy if you need to go back and retransfer. And keep an eye on the loudest sounds during the transfer to make sure that they do not digitally clip.
About pull down and resolving the Nagra to 59.94 Hz.
If you're shooting on digital video none of this probably applies, because you have single-system sound with picture and sound already in sync and both are already compatible with video frame rates. But if you're shooting film, digitizing the film via telecine, and completing your post on a non-linear system, the film you shot on the set -- the "real time" that you recorded -- is going to get slowed down slightly during the telecine process. If you filmed for exactly ten minutes on the set, that footage, after being telecined, will play back slightly slower and last 10 minutes and 6/10th of a second. The difference is enough to make dialog obviously out of sync -- if you sync up the clapper at the start of the take, the sound will drift noticeably out of sync after just a minute or two. To avoid this problem, you need to "stretch" the Nagra recording, slowing it down by a corresponding amount, which happens to be .1%. This is the difference between a theoretical video frame rate of 30 fps and the actual NTSC frame rate of 29.97 fps. Likewise, it's the difference between the Nagra's original sync pulse reference of 60 Hz and 59.94 Hz.
During playback, the Nagra "expects" to see an external reference of 60 Hz to compare to its pre-recorded reference of 60 Hz. It will then make any tiny adjustments in playback speed that are needed to stay "in perfect pitch" with this external reference. When we use the 59.94 reference box instead, the Nagra will seem to be playing back a sync pulse that is subtly "sharper" in pitch than is desired so the correction circuitry will slow the Nagra playback down very slightly so that it is once again playing "in perfect pitch."
This is best done during the digitizing itself so that the resulting AIFF files will already be "stretched" to the correct length.
Remember that none of this resolving takes place unless the Nagra is playing back with the Selector Switch lever all the way down in the "speaker icon" mode. (The Nagra will play back audio in the "first click" position but will not resolve in that mode.) Also note that resolving and pulldown are not the same thing. If you failed to resolve a Nagra transfer, doing a pulldown through file conversion tricks could still result in sound that drifts in sync because the Nagra was essentially "freewheeling" during playback and not doing any subtle ongoing speed corrections. The pulldown is an overall speed adjustment that affects an entire take; resolving involves internal incremental adjustments during a take.
About achieving pulldown through file conversion.
Maybe you goofed and digitized a number of Nagra tapes while resolving to 60 Hz instead of 59.94. Or maybe you recorded sync sound on a standard DAT machine and you loaded it into Pro Tools through a digital-to-digital S/PDIF connection. In either case, you need to stretch those files so that they correspond to the "video speed" of the telecined picture.
If you were strictly working within Pro Tools, this would be relatively simple. Pro Tools defaults to using the Mac's clock reference to determine the playback speed of all audio files. That is, the session itself "assumes" a rigid speed based on the sample rate you choose for your session -- in this case, 48kHz. For instance, if you bring a 44.1 kHz file into the session without doing a file conversion, it will play back at the wrong speed -- it will occupy less space on the Pro Tools timeline, and play back faster than its original speed. So one way to achieve a pulldown would be to select a file and go to the Audio Menu of the Regions List, choose Export Selected as Files >Sample Rate>48kHz (Pull Up/Down)>48.048. You would then save these "sample rate tweaked" versions of the files into a different folder.
Now if these files were played back at their correct sample rate of 48.048 they would be exactly the same length as before. (And in fact Quicktime Player would do this.) But if you import these files directly into Pro Tools without converting them to "48kHz exactly" during the import, then they will end up sounding subtly slower and taking up more space on the timeline -- in other words, they'll be stretched to the proper pulled down length.
However, Final Cut takes more of a Quicktime Player approach to sound playback, essentially looking at the file and saying, "So -- you're a 48.048 file and if played at that speed should last exactly ten minutes. Okay -- we'll adjust accordingly so that you end up lasting exactly ten minutes." So FCP expects to deal with files of different sample rates in the same session and adjusts on the fly. Impressive, but in this case it complicates matters since we probably want files that will be stretched to work in FCP.
There are two solutions:
1. We can trick FCP into thinking our 48.048 file is actually 48kHz exactly. We can do this by using a sound utility to change the header information of each audio file to read "48 kHz." The header information is what tells Quicktime/FCP how to approach the sample rate. Disadvantage: While this method works, it is tedious since I haven't found a program that will change this header info in batches rather than one file at a time.
2. We can create true 48 kHz files that have already been stretched to be longer. Let's go back to our Pro Tools session where we created 48.048 kHz versions of our files and brought them into the timeline. They are now stretched in time. We simply need to create new copies that are converted to 48 kHz. One method is to highlight individual files and select Edit>Consolidate. But it's easier to drag a succession of files into the timeline, select them all, and then go to Audiosuite>Duplicate, making sure that we choose the option to "create individual files" rather than "create continuous file." This will result in a series of files with the same file names but with "DUPL-01, etc" appended. These will default to going into our Session Folder's Audio Files folder and we can later copy or move them for import into FCP.
About redoing a bad transfer and getting FCP to automatically replace the files.
There have been instances when someone has inadvertently introduced noise or distortion during the transfer process. For instance, you might forget to close the mike pots on the Nagra or you might have left some of the other inputs turned up on the Mackie -- this could introduce system noise/hiss. Or the EQ settings could have been altered from their neutral (12:00) position, giving your audio an unnecessary "tweak." In any case, suppose you've been editing away in FCP and only later discover your mistake. The original Nagra recordings are fine, but the sound in your workstation is awful. Can it be fixed? And can you avoid having to re-sync every piece of audio in your completed cut?
Yes, but you have to follow some steps. First, re-transfer the audio making sure this time to exercise proper quality control. (When in doubt, listen to the original Nagra tapes straight from the Nagra and compare.)
Next you have to fool FCP into accepting these new files as the original transfer. But first you've got to ensure that the new files are in exact sync with the originals. And this may not be as simple as simply aligning the beginning of the first slate marker. If you transferred an entire Nagra roll into a single file, remember that between each production take there will be a subtle interruption of the Nagra sync pulse. This can result in a slight "bobble" in speed as the Nagra adjusts and resolves to the new burst of sync pulse. So, you may have to readjust each production take slightly to accomodate. This may sound like a huge chore, but in fact it can go pretty quickly.
This is all best done in a Pro Tools session where you can easily make edits of 1/4 frame accuracy or less.
First, re-transfer/digitize the sound roll. Next, bring in the original NG audiofile -- this will serve as your sync template. Put the new improved audio in a track below the original and examine the waveforms. Line up the first slate marker of the new audio to match the original as closely as possible. Trim the beginning of the new audiofile so that it begins exactly where the original does on the timeline. Proceed down to the next slate marker and see if any sync adjustment needs to be made to get the new audio to line up. If so, edit the new audio to realign the waveforms. Continue the process to the very end of the audiofile. Trim the end of the new audiofile so it exactly matches the length of the original. (Note: If you have less audio than you need, you can cut & paste a little roomtone from somewhere else.)
Now highlight all the regions of this new improved audio you've created. Go to the Edit Menu and select Consolidate Selection. This will create a continuous audiofile with no edits that should match your original exactly in terms of length and sync.
After this you just need to do a bit of file management. Give the new file(s) exactly the same name as the originals. Move the original files from the folder where FCP "expects" to find them and set them aside. Replace the originals with the new improved versions. When you re-open your FCP session FCP should reference these new files and your session will look and sound the same -- only better!
WHAT KIND OF BAD PRODUCTION SOUND CAN I FIX IN POST?
You can't "undo" distortion in a bad recording.
If your record levels were set too high, or if you overloaded a mic input or some other amplifier stage, it's too late to fix it. That's why it's important to closely monitor the sound during production and fix those problems that need fixing when they need fixing.
But before you give up, make sure the problem is actually a bad recording -- you want to be sure that any distortion you're hearing is on the original production recordings and not the result of some error that further down the audio road. For instance, if the distortion was created during the transfer from original recording to digital editing system, you might solve the problem by re-transferring the sound. When in doubt, identify a take that sounds bad on your editing system and go back to the original production recording and listen to the original tape on a good monitor system.
It's even possible that the distortion you're hearing exists only in the monitor chain that you're listening through. You might just have a dirty connection on the headphone jack, so try gently wiggling the connector.
Some tips for distortion trouble shooting:
Does everything sound bad, including music tracks and sound effects? In that case, it's likely to be a monitoring problem -- a bad pair of headphones, an overloaded amp, some dirty contacts in the audio connectors, etc.
Does the entire dialog take sound bad, or just the loudest portion of the take? (If just the loudest dialog is distorted, this makes it somewhat more likely that it was just a production sound error.)
Do things like voice slates and reference tone beeps also sound distorted? If you can eliminate monitoring problems as a culprit, this situation makes it more likely that the problem may be due to a bad transfer/digitizing process.
Do the digital sound levels frequently "hit zero" on the distorted takes? Does the waveform show signs of clipping? If your production sound was recorded and loaded digitally then the problem is likely to be in the original recording. If the production sound was recorded on an analog deck like a Nagra, it's likely that your sound transfer/digitizing was made at too high a level. Listen to the original sound roll; you may just need to re-digitize the problem takes. (You can avoid re-editing by aligning the waveforms and trimming any re-digitized sound files so that they exactly match the existing files. Then give them the same file names and FCP should open the new improved files to recreate your edits.)
You can't make a "distant, unclear" recording sound close up and clear -- but you might help it a bit.
Try this: use the EQ capabilities of your editing system to do a little "frequency enhancement." (Final Cut Pro calls all its EQ functions "filters"; not the best choice of terms, but close enough.) Using the 3 Band Equalizer, try reducing frequencies below about 80 Hz by around 5 dB. Now try boosting the midrange around 2000 Hz by about 5 dB. You'll have to experiment a bit, but often a combination like this will improve the overall clarity of the dialog.
It's best to be a bit conservative with EQ -- don't overdo it.
Something like the above works best when the sound is only slightly off mic. If the mike was more than seven feet away from the actor...well...
One of the most important things about production sound: you have to get the mike in close enough so that you get a crisp clear recording without a lot of background noise or reverberance that can "muddy" the sound. So please -- when working with video cameras, don't rely on the built-in mikes mounted on the camera. And when working with double-system recorders, try to plan your shots so that at least some of the coverage will allow for decent mike placement.
Remember that it's easy in the mix to make a good clear recording sound "gritty" or "dirty" -- you can add ambience tracks or reverb or you can EQ the track to reduce the high and midrange frequencies that convey clarity. But it's very tough to try to clean up a bad recording.
So that's enough for the sadder-but-wiser routine; if you're reading this section, you already regret some of your production problems.
You can reduce certain types of noise (but don't expect miracles.)
Here's the basic rule that will tell you if your particular noise problem is fixable: the noise you don't want has to be in a frequency range that is different from the sounds you do want. For instance, you could get rid of quite a bit of low pitched rumble in a recording of a parakeet chirping because the chirps you want are high pitched and the noise is low pitched. That makes it a good candidate for an EQ fix.Try setting the 3 Band EQ to drastically reduce the gain on the low frequencies you select. Start off with the low frequency slider set to around 100 Hz and experiment with raising it to higher frequencies. In the case of this example, you might be able to reduce everything below 800 Hz without affecting the sound of the chirps; so you'd get rid of a lot of noisy rumble without cutting into the higher pitched chirps.
Dialog recordings are trickier to fix because dialog contains a much broader, richer range of frequencies than a simple bird chirp. However, the basic EQ approach described above can help reduce certain types of rumble such as wind noise or very distant traffic. Even some types of air conditioner noise can be lessened. But bear in mind that you have to be much more conservative in your approach --chances are you should confine your filtering to frequencies below 100 Hz. (Anything much above that begins to cut into the voice range.)
Noise that isn't confined to a narrow frequency range is a lot trickier to reduce. Expanders and other types of noise-reduction software can be applied to some of these problem noises. At the time of this writing, Final Cut Pro 4 has expanded its sound processing functions and you may be able to do quite a bit of fix-it work within FCP. For more exotic tweaking you might have to export selected sound files to Pro Tools to take advantage of some more exotic plug-ins. (And I can't give you a quick lesson in using them in just a paragraph or two so you may have to get someone to help you with it.)
But to summarize and illustrate the basic rule about noise reduction, I'll list a few likely suspects:
Fairly easy to fix:
Somewhat tricky:
Difficult to impossible:
If the original recording is poor quality you either have to live with it, find an alternate take (sometimes you can use the sound from a different take and "cheat" it over a different shot), or record a new version of the line. If it's a fairly brief line -- especially if it occurs offscreen -- you might get together with the actor and record this "wild" (without seeing the picture.) For more extensive replacement you'll need to arrange some type of ADR session. In either case, try to get a decent perspective match with the other dialog in the scene -- in other words, if the original recording was done in a carpeted room with the window drapes closed and the mike fairly close, don't record the new lines in a reverberant room with a marble floor. Do not try to match background ambience in this new recording; just because your original sound was recorded next to a busy freeway is no reason to record the new lines that way. (You'll use dialog editing tricks to fill in some matching ambience for the final mix.)
HOW DO I MAKE EDITED SCENES SOUND SMOOTHER?
In nature, continuous sounds don't change instantaneously; rivers don't suddenly quit flowing, rain doesn't stop on a dime, wind doesn't cut off as if someone has thrown a switch. But all these things can happen when you edit various takes together into a supposedly continuous scene -- and our ears are trained to notice these weird discontinuities and wonder about them. And wouldn't you rather have the audience wondering what's going to happen next in your story and not distracted by these technical "bumps" in your soundtrack?
So how do you smooth the bumps? Sound editors rely on variations of just a few basic tricks:
1) "Hide" the change in ambience by making it happen during another strong sound that distracts the listener; this drowns out or "masks" the change.
2) Make the change gradual rather than instantaneous by creating a long fade-in or fade-out instead of a simple cut.
3) Run some continuous effects ambience underneath the entire scene; this can help mask subtle changes when you cut between different angles, especially if the continuous effects ambience is more "complicated" than the production ambience.
Example: You shot your "restaurant scene" in a converted warehouse and the room tone varies from cut to cut. But when you add some sound effects of background voices, dish clatter, and P.A. music those minor bumps aren't noticeable.
A bit more detail about "Trick #1." For subtle presence bumps, often it's enough to re-think the position of the edit point. Picture editors tend to cut the sound at the same point as a picture cut. When no sound except ambience is occurring at that point, the bump is noticeable. But if the ambience from shot A is continued past the picture cut to shot B, and the edit point occurs just prior to a strong modulation such as the first line of dialog in shot B, the strong modulation will tend to mask the slight change in ambience. (The sound for the two angles are placed on different units, but the same masking principle applies if they were cut together that way on a single worktrack.)
Here's what the original worktrack might look like if it was simply split onto two different tracks at the original edit point:
Since they're from two different takes (12/1 and 12B/2) there's probably a presence mis-match at the cut.
Instead, if we extend the presence from the first angle up to the point where the strong sound of the incoming dialog would mask the shift in presence, we might end up with something like this:
You can see that we had to "steal" some presence from elsewhere in take 12/1 in order to extend it far enough. In the above case that it took two different regions to "stretch" the presence that far.
Here's a variation on "Trick #2" in which a crossfade was created to ease a transition between two different ambiences.
Those crossfades might end up looking something like this:
Sometimes you need to mask the incoming or outgoing fade by having it occur while there's a strong modulation of dialog to "hide" the fact that the ambience is changing.
About "Trick #3" -- sometimes you run into cases where you have a particular "problem angle" with a distinctive ambient noise that cuts in and out in a jarring way. Then you may need to have the ambience extended from the beginning of the scene to the end so that there is at least a consistent ambient tone throughout. But remember that ambience is additive and that the dialog from that angle already has the noisy ambience tied to it. So don't just create a separate continuous piece of that background noise on a spare track, because when that noisy dialog occurs the ambience will still bump in and out. Instead, extend the ambience on the same track in which the dialog occurs, essentially "filling in the gaps."
In the first example, the background noise will still bump when Bob's angles cut in and out:
In the second case, the dialog should sound smooth and consistent (as long as Mary's lines are reasonably clean.)
STATEGIES FOR WORKING WITH FINAL CUT PRO & PRO TOOLS
This section assumes that you have a good working knowledge of Pro Tools. So it isn't a step-by-step "how to" guide; it's just an outline of general strategy for attacking various problems.
PICTURE
There are four basic choices for getting picture from FCP and viewing it on the Pro Tools workstations:
a. From Final Cut Pro export the FCP project as a "self-contained Final Cut Pro Movie." This is a high res 740x480 Quicktime Movie that will take up most of a Mac monitor screen, so in a room equipped with two monitors you can just drag the movie to the second screen. (If the second monitor is mirrored by a video projector, so much the better.) You really need two Mac monitors for this to work as the FCP movie hogs most of the screen. Note: This hi res approach is recommended only if you have some other reason for generating a Final Cut Pro Movie. See Choice B below.
b. Export a compressed Quicktime Movie. Recommended settings: Frame size--320x240; Frame rate -- 29.97; Audio -- None or Mono 48kHz*.; Quality -- Medium to High. (Options for key framing and internet streaming should be unchecked.)
If you view this in its "normal" size, it's a pretty small picture, suitable for viewing on a single Mac screen that must also display the Pro Tools window. This image size would not be very good for cutting detailed sync effects, but will work for cutting dialog when the worktrack is clearly in sync and no alternate take "cheating" or ADR editing is needed.
When there is a second Mac monitor available you can view this same movie in a "double size" window. First you must open the movie in Quicktime and select "double size" under the "Movie" menu. Then "save" the movie. The next time you import the movie into Pro Tools it will default to the double size window rather than the normal 320x240 window. Note that a Quicktime movie of medium quality will look a little blurry at this resolution, but for most purposes (even foley & ADR) it should do.
c. Generate a VHS or DV tape from FCP, then re-digitize it using the Miro card. This lets you view high-res picture on a regular TV monitor. (Disadvantage: It's time-consuming and you have to utilize a Miro-equipped workstation, which is a feature only found on a limited number of LMU's Pro Tools systems.)
d. From Final Cut Pro software you can export a Quicktime movie that will play back in an acceptable way through the Miro card to a standard NTSC TV monitor. The movie image will not fill the entire TV screen but it will be a workable image. The specs for "Option 1" will give you a picture that fills about 40% of the screen. "Option 2" will fill most of the screen, about 80 %. Both images will be fairly sharp & detailed.
Even if you've already created a FCP movie, you can still take advantage of the option above by opening the FCP movie in Quicktime Pro and exporting a second Quicktime movie with the above specs. One disadvantage: generating these movies is pretty slow even on a fast Mac; be prepared to take a coffee break or go have lunch. Also there's a very subtle distortion of the frame edge -- it can have a slight curve to it which can vary a little from shot to shot. But this is barely noticeable.)
*When should you embed audio into your Quicktime movie? Any time that the slightly larger file size it requires is not an issue. It's very handy to have a readily available worktrack whose sync relationship to the picture is established. This is especially important when the FCP session does not have a countdown leader and head pop (which it should.) However, if you're exporting a separate AIFF file or OMF session of the worktrack with a head pop on the track and a "2" frame visible on the movie countdown leader, then you should be able to easily establish sync between sound and picture.
DIALOG EDITING - FCP TO PRO TOOLS AND BACK
OPTION 1 - EXPORTING AIFF FILES
Ideally, all the finicky editing of the production dialog tracks will be done in FCP.The advantage of doing your dialog editing in FCP is that you have easy access to the all the complete audio takes -- very handy when you're trying to find matching ambience to plug "holes" in the track, or extending incomplete words, etc. The disadvantage is that, as of this writing, FCP is still just a little clunky when it comes to making fussy edits. To proceed you should:
Next you will export each individual track as a single audio file. For instance, mute all tracks except for A1. Go to File>Export>Using Quicktime Conversion. From "Format", choose AIFF. Under "Options" make sure to export the track as a 16 bit 48 kHz mono file. Repeat the process for each track you wish to export. Make sure there are head & tail pops on each track and that they align correctly with your picture.
Now it's just a question of importing those tracks into a Pro Tools session, making sure they are in sync with the Quicktime movie for the session, and refining your dialog mix within Pro Tools.
OPTION 2 - OMF EXPORT
If you must do extensive dialog editing in Pro Tools you will probably want to try the OMF file exporting procedure. Make sure that in FCP you create good-sized "handles" for the audio regions when you choose your OMF options (I'd say a minimum of 2 seconds; 4-5 would be better.) Note: Since the native sample rate for FCP is 48 kHz, you will probably want to stick to that rate that for the dialog session.
Once you have an OMF file you can use DigiTranslator to try to create a Pro Tools session. Notice I said "try" -- there can be bugs in the process. The most common is that if you specify very long handles beyond the range of the original audio file, the conversion process may hang/crash. However, I've successfully converted sessions with two or three second handles and that's often enough.
If you use an LE system that has an option called DVToolkit, you access Digitranslator from File Menu > ImportSessionData. If you don't have access to such a system you may still the free-standing program Digitranslator 1.x, a free OS9 version which will create a Pro Tools session from the OMF file. Note that Digitranslator 1.x defaults to creating a session in the SDII file format, a disadvantage when you're working on a Pro Tools LE system which doesn't support mixed file formats within the same session. One workaround is to create a new Pro Tools session that is set up for 48 kHz AIFF files and then use the "Import Tracks" feature to create an AIFF version of the SDII session created by Digitranslator 1.x.
With the DVToolkit version of Digitranslator, you will have the option of "transparently" accessing the audio from the OMF file without necessarily creating new audio files. However, it may be a good idea to use the options which will create a batch of new AIFF files within a Pro Tools session which is independent of the OMF file. In this way, you'll have a Pro Tools session which can be run on an LE system which does not have the DVToolkit option. (Confusing? It's all part of motivating consumers to spring for those extra options...)
If you're working on an LE system and would like the ability to use the original AIFF files you could either import & convert them individually as needed, or you might elect to create a new Pro Tools session that is set up for 48 kHz AIFF files and then use the "Import Tracks" feature to create an AIFF version of the SDII session created by Digitranslator.
Note: The Pro Tools session created via OMF will not duplicate any volume adjustments created in FCP. You'll be starting from scratch in many ways. But if you have also created AIFF exported tracks from FCP, you may find that you can use them much of the time but still access the OMF versions of the sound when needed. This requires a little extra thought in terms of track layout but gives you great flexibility in working with your sound.
OPTION 3 (You won't like this one...)
1. Export AIFF files as in OPTION 1 -- except you won't be doing any finessing of the edits in FCP.
2. Using a printed EDL, carve the files into regions, noting the take I.D.s.
3. Batch export the needed audio files from FCP as AIFF files.
4. Mod match all the takes. This gives you the equivalent of an "Auto Assemble" -- only it's not automatic. The advantage is that you have access to the entire audio take, not just limited "handles." (This approach may seem like a major chore, but it's routinely done on many feature films.)
GETTING SOUND FROM PRO TOOLS BACK TO FCP
DIALOG
You can create a dialog pre-dub bounce at 48 kHz and import it back to FCP for further mixing with effects and music within FCP, or you may continue to do all the post sound in Pro Tools to create a final DME mix that can then be imported into FCP.
If you're going to do more effects and foley work and final mixing in Pro Tools, I'd strongly recommend that you do a dialog pre-dub rather than try to handle all the tracks in one large complex session. For one thing, you're liable to use a number of plug-ins for cleaning up your dialog and that could tie up some DSP power that could limit your choices later on for your effects and music work.
A dialog pre-dub should contain your principal sync dialog and any principal ADR with your main actors. It should not contain Group ADR or walla. (You may wish to do a separate pre-dub of those elements, or you could leave them as separate tracks to carry in the final mix.)
In creating the dialog pre-dub you should concentrate on getting a smooth and believable balance from angle to angle within a scene, and good relative levels on scene transitions. This is the step where you should concentrate on minimizing background noise, getting extra clarity on "problem tracks" that could benefit from EQ, fixing mis-matches of perspective, and so forth. The audience shouldn't have to strain to hear any key lines of dialog; they shouldn't have their ears pinned back by jarringly loud lines either.
You may want to use some gentle compression where needed during the pre-dubbing. Remember, though, that in the final mix you'll be combining the dialog with new elements of music and effects -- so the level adjustments you're making in the pre-dub aren't necessarily carved in stone. You'll undoubtedly have to do some level riding and tweaking on the pre-dub itself during the final mix, so you may want to be a bit conservative during the pre-dubbing phase. (Example: It's easy to add more reverb to the pre-dubbed voices in the final mix, but to "undo" excessive reverb you created in the pre-dub would involve going back to the original dialog tracks and starting from scratch.)
Note: If you're going to do the rest of your post in Pro Tools, you may wish to bring a few effects ambience background tracks into your dialog pre-dub session. DO NOT mix these into your dialog pre-dub -- keep them separate until your final mix. But by having them available to play against the dialog tracks, you can sometimes tell if some ambience bumps in your dialog tracks will be "fixed" by the addition of a bit of background effects. This might save you some headaches in dialog editing.
FX EDITING/MIXING
The method I would like to see more students experiment with is for the FCP editors to hand off the FX editing and pre-dubbing to Pro Tools users (presumably RECA majors), who would then create pre-dubs of props, FX backgrounds and hard FX for importing into FCP. (The easiest option being to create bounces as 48 kHz AIFF files -- either mono or interleaved stereo.) The FCP editors would then be responsible for combining that with their own production dialog tracks for the final mix.
After all, the person editing the picture is most intimately familiar with the production sound and has all the material in FCP to do the bulk of the dialog editing.
Alternatively, you could create a dialog pre-dub in FCP for import into a Pro Tools session. The Pro Tools session would then contain the FCP dialog pre-dub and all other effects and music. Then the final mixing could be done in Pro Tools to create a bounce of the final stereo DME for reimport into FCP.
In which we deal with such mysteries as:
How do I set mix levels by the meters?
If you want a mix that will play well on a TV as well as in a medium-sized theater, here are some basic guidelines about metering:
How do I set mix levels by ear?
The first step is to set your monitor levels correctly so that a level that looks "good and strong" on the meters will sound "good and strong." In some of the Pro Tools rooms the playback levels have been calibrated; use the monitor level markings on the Mackie mixers so that the loudness you hear at the workstation will correspond to the loudness in a properly set-up theater. A good way to double-check this is by playing a bit of your reference tone and seeing that the Mackie meters read "0". (The Mackie meters use an analog scale in which "0" stands for reference level and not "maximum level.")
Many times you'll be working with headphones or on systems that have not been expertly calibrated. But you can set workable "ballpark" levels using the sample files provided. Just go to the desktop folder called LEVEL CHECK. If there is more than one folder, use the file format and sample rate that's appropriate for your purpose. (For Final Cut Pro, this would be the "AIFF 48 kHz" version.) Bring the file called "-12 dBFS LevelCheckV2" into your session. It's a stereo file, so put it into two tracks which have been panned left and right. Play the file and adjust your headphone/speaker level accordingly. (There's a voice-over that explains how things should sound. Basically you start off by playing a reference tone that should sound "strong & clear" -- a little on the loud side. Then there are some other sound samples that help give you a feel for how typical dialog might play, some fairly loud music, etc.)
Now that you have a good listening level that corresponds to a good meter level, you should be able to apply common sense to your mix -- if something sounds too loud,you should probably lower it; if it sounds too soft, you should probably raise it.
For future level checks, there's "QuickLevelCheck" that contains just the first few sound samples without the explanatory voice-over. The first two bursts of tone and pink noise should sound strong and clear but not obnoxiously loud. The Voice Over should sound pleasant and understated. The last pink noise burst should sound relatively stronger, probably strong enough to be a bit annoying.
Remember to check your monitor levels at the beginning of every mixing session.
"Reference Level" or "Standard Operating Level" represents a signal strength that is strong, but not too strong. Your subtlest sounds will fall well below that level; your strongest sounds will be above. How much above? Well, it varies depending on what kind of analog or digital media you transfer your mix onto. The number of decibels that your loudest sounds can go above reference level without distorting will vary from format to format. (That "safety margin" for the loudest sounds is called "headroom.") Beta SP tape, for instance, only has about 8 to 12 dB of headroom. DVD has 20 dB of headroom.
Now you may be thinking to yourself, "What a minute -- how can DVD have 20 dB of headroom? My Final Cut Pro mix can only go 12 dB louder than my reference level before it digitally clips." That's true as far as it goes. The difference is that a DVD has a reference level of -20 dBFS. We're using a higher reference level of -12 dBFS on our workstation mixes, so there is less headroom.
Doesn't a higher or lower reference level mean my movie will sound louder or fainter overall?
Not necessarily, because playback levels are based around the reference level. In other words, we adjust the playback volume so that sounds recorded at reference level will sound "good and strong." You may have noticed that when you play a typical dialog scene from DVD instead of a VHS tape, you tend to turn your TV volume up a bit higher. You may also have noticed that once you do so, when the big sound effects and music scenes come along, they tend to sound pretty loud.
Here's one way to think of it: for DVD, we record most of our mix at a lower level, with our average "strong" levels hovering around -20 dBFS. As a result, we have more safety margin for our loudest sounds because with that lower "average" level the loudest sounds can be 20 dB louder rather than just 12 dB louder.
In fact, the reference level for U.S. both DVD and digital theatrical film formats is -20 dBFS. This was determined by the Society for Motion Picture & Television Engineers, or SMPTE. (In Europe they use a -18 dBFS standard.) Our suggested reference level of -12 dBFS represents a kind of compromise that will work nicely for TV broadcast, home video formats, and theatrical presentation. Most film festivals and many broadcast facilities still use Beta SP as a preferred format, and a -12 dBFS standard for DV tape gives good "interchangeability" between different formats and TV broadcast. This is the primary reason that we have suggested that you use a -12 dBFS reference.
How do I transfer my mix from digital to an analog format like Beta SP tape?
If you use a reference level of -12 dBFS in Final Cut Pro and you are transferring to VHS Hi-Fi or Beta SP, play your reference tone and set the input of the Beta deck so that its audio meters read "0" VU. Then locate one of the loudest portions of your mix and double-check the levels on the Beta deck's meters. On Beta SP you would expect the meters to tip "into the red" a bit. That's fine; that's what you'd expect. What you don't want is for the meters to slam into the max and stay there for extended periods.
Note: VHS linear audio tracks only have about 8 dB of headroom, so if your mix tends to be on the loud side overall you may want to adjust your reference level lower to that it reads "-4" on the VHS deck. Or, if your film is on the quiet side you may not have to adjust at all. You may have a peak or two that will distort a bit on the linear tracks, but these days most VHS decks play from the Hi-Fi tracks as their default anyway.
It's a good idea to spot check the tape by playing back a few of the louder scenes.
Do I need to remix for different formats like DVD or Digi Beta?
The short answer is: not necessarily. If you use a -12 dBFS reference level and mix as suggested, your mix will fit into the available headroom of DV tape, DVD, and VHS hi-fi. (In fact, you'll have some "unused headroom" left over on DVD and VHS hi-fi.) For transfer to an analog format, you'll be dealing with the kind of line-up tone issues dealt with above.
For delivery on Digi Beta or authoring a DVD, you should start by taking the mix you did at a -12 dBFS standard and simply lower the entire mix file by 8 dB. This effectively changes your reference level to -20 dBFS, which is the standard for those formats. The result should make your average levels more or less the same as most DVDs, although your loud scenes won't be as loud as most recent commercial DVDs. For many films this 8 dB adjustment will be sufficient.
But if you want to fully exploit the greater dynamic range of DVD you should reopen your original mix session and lower the master fader by 8 dB. This basically switches you to a -20 dBFS reference level. You will then need to raise your monitor listening level by 8 dB to compensate for this change.Then go in and selectively raise the biggest loudest elements to take advantage of the increased headroom. (This approach might be good for a suspense film, where you are exploiting the shock effect of going from a quiet scene to a loud one, or using big music "stings" for emphasis.)
Another possible approach: depending on your film, you might just "ride the level" on your completed mix file. Start by creating a FCP session which contains your picture and completed sound mix file. Lower the audio level of the mix by 8 dB. Raise your monitor listening level by about the same amount to compensate. Now go in and selectively raise portions of the mix where you want the whole mix to go louder. (The disadvantage of doing this to a completed stereo mix file rather than going into the original mix session is that you don't have separate control of the sound elements. For instance, if you want to raise a music cue you'll also be raising ambiences and sound effects as well. So this approach is not recommended where you have dialog content.)
All of the above is the "short answer."The long answer gets complicated. When you're mixing on a simple workstation, the resulting mix is a kind of "prosumer compromise" to begin with, because you're creating a simple two track stereo mix.
The slang term for this is "Lo Ro", meaning "Left Only, Right Only". This is the format for standard audio CDs, and most people watch stereo television shows or home video in the Lo Ro mode.
But film mixes are not created in Left Only, Right Only stereo. Even the earliest stereo films added a separate center channel, and from the 70's to the mid 90's most stereo films were mixed in a four channel format: Left, Center, Right, and a single mono Surround channel. This is called "Dolby Stereo." The trick to this format is that it's actually stored on just two tracks. It takes a special encoding and decoding process to derive four separate channels from the two tracks, so to distinguish it from regular stereo it's called "Lt Rt" for "Left Total, Right Total." Most VHS tapes of popular films have Lt Rt tracks, and stereo TV is broadcast in this format, so if you have a Dolby Pro Logic amp and the right speaker setup, you can listen in this mode at home.
Digital film release formats, including DVD, support "5.1" stereo which improves on the Dolby Stereo format by giving us a stereo or "split" surround and also a dedicated channel for very low frequencies (an LFE or "subwoofer" channel.)
The problem is: there's no practical way to mix for these formats unless you are listening in a room that has the correct number of speakers, properly tweaked. As of this writing, your basic Final Cut Pro or Pro Tools LE workstation is a "Lo Ro" environment.
How big a compromise is this? Well, good film mixers try to double-check their 5.1 mixes against a "worst case scenario" and ideally create actual re-mixes for formats with less headroom, fewer channels, etc. (This is called "down-mixing".) For lower budget projects this is sometimes done handled just by doing "compromise transfers", taking the 5.1 mix and tweaking it only slightly.
With a student project you have the opposite problem: you're taking a Lo Ro downmix that will probably play mostly in Lo Ro settings (like an agent or producer watching it on TV) and hoping it will play well in potentially "upmix" settings like a festival theater.
For a theatrical showing from Beta SP or DV tape, your film will probably be run through a Dolby Stereo decoder. That means that the decoder will be trying to derive 4 channels from your two track mix. The basic rules for the decoders is:
So anything you panned dead center -- like dialog -- should end up playing in the center channel. Anything that "leans" left or right -- like stereo music -- should play in the left and right speakers only. Now in theory nothing should go into the surround channels, but in practice you sometimes end up with "magic surrounds" -- a bit of your stereo recordings may bleed into the surround channel because of complicated phase relationships due to the original stereo mike placements. (This happens a lot with professional mixes done in this format, too. Sometimes it's a pleasing effect, sometimes not. Just one of the trade-offs on the encoding/decoding process.)
Also be prepared to have your stereo mix sound "narrower" or somewhat more "mono-ized." (Another case of reality vs theory.)
Note: To avoid having Lo Ro mixes decoded as if they are Dolby Stereo, DVDs contain a little "flag" in the metadata that "tells" a Dolby Pro Logic decoder whether the mix is Lt Rt or not.
I want my movie to be as loud as possible -- how can I do that?
It's true that by using a -12 dBFS reference level the loudest sounds in your mix won't play quite as loud as the loudest scenes in the latest Hollywood action movie -- but judging from surveys done of audience complaints to theater owners, that's not necessarily a bad thing. (I'm not necessarily trying to talk you out of your goal of a "loud movie", but let's kick around the repercussions first.)
Before around 1993, movies were released to theaters in formats that only had around 9 dB of headroom in four channels. This meant that the very loudest scenes in a movie might reach around 103 dB SPL.
Now trust me: audiences in the 80's weren't complaining that movies like "Top Gun" would be better if only they were louder.
But digital sound formats came along, and movies released with digital sound have more headroom, so the loudest scenes can reach around 118 dB SPL. And a difference of about 10 dB will strike most people as being twice as loud, so this is a pretty dramatic difference.
There's nothing necessarily wrong with this picture so far: that extra headroom was meant to be used for truly cataclysmic, awesome events, like "Death Stars blowing up" and so forth. The problem is that Hollywood historically isn't known for self-restraint -- just look at old posters hailing every cheesy B movie as "colossal, stupendous, tremendous!" So you have half the producers in town urging their mixers to make every two-bit "car crash effect"or music cue louder, right up to the max.
It's a little "needy", isn't it?
But if that's your goal, the obvious solution is to use the SMPTE standard of -20 dBFS as your reference level, then "mix to the max." Be aware though, that a format like Beta SP won't "hold" those loud passages without serious distortion -- you can't pour 20 ounces of water into a 12 ounce glass. So if you're transferring your mix to Beta SP you should line up your reference tone to "-8 VU" instead of "0 VU" -- which tends to defeat your purpose, as your loudest sounds coming off the Beta deck will play just as loud as anyone else's. (In fact your average dialog scenes might end up sounding on the faint side.)
If you output to DV tape then you should carefully label it as having a reference level of -20 dBFS. But note that for LMU screenings, the playback level will not be customized to your reference level, so again, you won't really achieve your goal. (And please don't tempt our hard-working projectionists with bribes to "crank it up.")
The best solution would be to create a DVD as your release format, or if you have a big enough budget,have a Dolby Digital 35mm print made.(And if you lean toward that approach, you should look into doing a full 5.1 mix. That gets pretty involved, and is way beyond the scope of this little FAQ.)
Or you can content yourself with knowing that, even with Lo Ro playback, if the theater is set-up for a -12 dBFS reference your loudest peaks should hit around 100 dB SPL.
HOW CAN I MAKE THE MIX SOUND BETTER?
Listen over good headphones or speakers, at an appropriate volume.
Seems like a pretty obvious statement. But what does "good" mean? And what is "appropriate"?
Basically, you want headphones/speakers that are accurate or have a "flat" frequency response. Avoid headphones that brag about "Mega Bass" and such, because they are designed to artificially pump up low frequencies. If you base all your mixing judgements on what you hear in the "Mega Bass Mode", you'll probably be disappointed when you hear the results played on properly designed speakers.
(Just as a personal aside, for headphones I've found that the Sony MDR-7506 types tend to provide a fair approximation of what you will hear on a good dubbing stage. They aren't cheap at around $90, but might prove to be a good investment.)
As for appropriate listening level, you should ideally listen in a room that has playback levels calibrated to match those of a good theater. That's not always possible, but you can use a reference tone or a sample soundfile to get your monitor level in the ballpark. (More about this topic.)
Note: If you're confident that your mix will be heard primarily in television viewing or home theater situations rather than a large theater, you'll probably want to cheat your monitor level down so that it more closely matches the typical listening situation. Most professionals mixing for television set their monitor levels about 5 dB lower than the standard level for theatrical film. (That translates to about a 78 dB sound pressure level instead of an 83 dB sound pressure level.)
Set dialog levels first, then music and effects.
This rule may not seem obvious at first, but suppose you were to set your main title music level before listening to anything else -- and you tended to push it toward the maximum level. Now suppose that same music cue continues and overlaps into your first dialog scene. You would probably end up pushing the dialog hard just to make it heard over the music. Then when the music ends and the dialog continues, you'd find yourself with an artificially "pumped up" dialog scene. If you follow that with a scene at normal levels, it will seem somehow out of balance, and you might be tended to raise it as well, creating an entire mix that is slightly off-kilter.
Dialog in a sense serves as our point of reference for what is "normal loudness" in a mix. There's a certain range of loudness that we expect to hear given a certain image size on the screen: a big close-up sounds one way, a person standing in a wide shot sounds another. What I'm suggesting is that you get the dialog in the ballpark first and then adjust other elements relative to that.
For dialog scenes, avoid selecting music cues that clash with the dialog.
Sometimes music cues can create some unwelcome competition for your actors and the performances you are trying to impress on the audience. Just playing the music lower isn't always the answer; at times it's the nature of the music cue itself that is the problem. For one thing, some types of music are simply "designed" to play big -- Beethoven's 9th Symphony just doesn't sound right when it's pushed to the background.
Or sometimes the music has so much energy in the mid-range frequencies that it tends to drown out the dialog. This is due to a quirk of our hearing called "frequency masking." It's easier for our brains to sort out two equally loud sounds if they are in very different pitches. When the sounds are in the same range it's easier for one to "drown out" the other. Mid-range frequencies are especially important when it comes to understanding speech, so as a rule if you're selecting or composing music for a dialog scene it's best to avoid having lots of instruments working busily in the mid-range.
Sheer "busy-ness" is also a factor; if the music is in a fast tempo and highly melodic, part of our brain is trying to follow the complex melody line and this tends to compete with the dialog for our attention.
And for a stereo mix, since dialog typically plays from the center speaker, music that contains a lot of information assigned "dead center" tends to fight the dialog more than music which has a wider stereo spread.
The above are just general observations and you can obviously think of many exceptions to them. Just be aware that there are sometimes technical reasons why a particular music cue tends to muddy your dialog track.
When you have ambience mismatches in your dialog tracks, attack the problem with dialog editing tricks first.
I've noticed that students will often approach an ambience mismatch from one shot to another by making level or EQ adjustments. This may work to improve the offending "bump" on the cut -- but what is it doing to the dialog itself? Will it play at the proper level? Will it sound artificially "tweaked"?
It's usually best to address these problems through editing first. You can eliminate many problems just by good editing, and at least minimize others so you won't have to resort to major processing in the mix. Often this will result in more natural-sounding tracks than if you resorted strictly to mixing/processing tricks.
Remember that the audience is hearing the dialog for the first time; you know the words by heart, but they may need help understanding them.
So obvious it hardly needs saying, right? And yet you can probably think of more than one recent film where you couldn't quite make out a line and felt frustrated. (This is especially deadly in a comedy, when the audience really wants to get the joke.)
It's understandable that after spending months going back and forth over the editing filmmakers can recite the entire film in a sleep-deprived dream-state -- and often do. And it's understandable that it's easy to lose perspective in the mix.
But smart filmmakers develop a knack for asking themselves such dumb questions as: can I understand what the actor is saying? If the answer is no, they look to fix it.
Learn to listen critically for elements that are out of balance or "stick out."
This is really the tough one, and I have no stunning insight to offer as to how you develop this ability, other than to listen to what is considered a good professional mix and then compare it to some not-so-good student film mixes. (You can also learn a lot listening to good student mixes and not so good major studio films.)
Foley in particular often needs careful adjustment in order to blend in with the rest of the track. Footsteps, for instance -- how do they sound in one mix as compared to the other? If you know something isn't quite right, can you pinpoint the difference? Are they louder? Too soft? Brighter, more trebly? Too bassy? Or maybe the scene is set in a big empty room and the foley sounds too "dry" and lacks reverb.
I'll risk a big generalization here: it seems that in a lot of student mixes foley tends to be mixed a bit too loud and "upfront." It may be a natural inclination -- "I went to all this trouble to record this stuff, I want to hear it" -- but the result can sound very unnatural. So in addition to playing the foley tracks lower, you may want to experiment with rolling off some of the high frequencies and adding a touch of reverb (when the space of the scene is appropriate.) This can help create a more realistic perspective to the sound.
Ambiences, too, are often mixed a bit on the hot side. Again, for a dialog scene, the dialog itself should give you a pretty good point of reference for giving other elements the appropriate weight.
Other times it's as simple as imagining yourself standing at the same location, right next to the camera, and imagining how something would sound.
If it's not broken, don't fix it. Avoid running your tracks through compressors, equalizers, etc, unless it's called for.
You can really get yourself in trouble by doubling and tripling processors and running the sound through a lot of "roundabout routes." Sometimes you can even have one processor trying to "undo" the processing of another. (For instance, if you've run sound through an expander in order to reduce some background noise, then do heavy compression on it, you can end up bringing a lot of the background noise back up.) And if there's still a problem with the overall sound, it just makes it that much harder to pinpoint the cause because you've introduced so many new variables into the signal path.
So please don't apply a processor simply by rote; approach the tracks case by case and decide when some special treatment is actually needed.
When you have to do processing, periodically use the bypass feature to compare the "new improved" version to the original. (It's easy to overdo the processing.)
This is closely related to the above. The bypass switch is a great tool for judging whether your "fix" is actually an improvement or whether you're better off leaving the sound alone.
Audiences respond to change. Provide some contrasts to keep your mix dynamic.
Successful entertainers understand this need for dynamics; musicians may do a series of increasingly intense songs and build to a sort of crescendo -- then follow that with a slow ballad to take the energy down before building it up again for the finale. By contrast, a strategy of trying to sustain one long peak may backfire and result in a kind of plateau. Even though in theory you've kept the intensity level constant, the audience feels let down.
Some films lend themselves better than others to big contrasts in sound -- suspense or action films, for instance. But even a relatively quiet dialog-centered film can benefit from little sonic changes-of-pace. Hopefully you've scripted and designed your film to create those opportunities.
Sure it helps if your sound is technically polished -- meaning that the recordings aren't unintentionally distorted or noisy, that the dialog sounds clear and natural, etc. But to really have an effective soundtrack, it's important that the sound conveys information that the audience cares about. If the dialog is hackneyed and irrelevant, the audience won't care if they can't make out a few words -- why should they? If the music doesn't take us anywhere interesting, it's really more of an annoyance than anything else. And if sound effects are merely used to cover the basics of "seeing a car, hearing a car" then a filmmaker really isn't working with the full potential of the medium.
One trick to get beyond the basic "two dimensional" treatment of sound and to really engage an audience, to make the sound matter to them, is to create story situations where the sound matters to the protagonist. Your main character should have a "stake" in the sounds going on both in the story and the film. We need to see the character actually listening to something and reacting to what they hear. And we need to hear it as well (although not necessarily at the same time as the character hears it.)
Suspense films provide obvious examples: a muffled conversation heard through a wall, the squeak of a footstep on a staircase, a doorknob being turned. Or: the deep creaks of metal under stress, sonar beeps, and the concussions of approaching depth charges.
It needn't be so melodramatic. You could imagine a comedy about a petty office rivalry where the sound of a Xerox machine might have great significance to one of the characters. The point is this: the soundtrack is not just a given, some minimum-daily-requirement of technical competence, but must be integral to the story. Oddly enough, an audience will overlook some lack of technical polish if they're really engaged and moved by the film.
Your job as a filmmaker is to find new and interesting ways to do that.
Copyright © 2003, 2004 by Rodger Pardee