[Stephen Muir]

Another option is to record the effected track to a new audio track. The easiest way to do this is through a post-fader send via internal bussing. Make sure that only the new record track is rec-enabled, highlight the audio in question (including any reverb decay time on the end) and just record it directly to the track.

You may want to leave the original track version in the session (disabled, of course) in case you have to go back to tweak the effect settings.

[Stephen Muir]

Hi, Tom.

Protools comes in four flavours: Protools Free, Protools M-Powered, Protools LE, and Protools TDM.

Protools Free, which is no longer supported with upgrades and isn’t rated for OS-X or WinXP, is a free-to-try software-only version of PT. It was limited to 8 tracks and uses the computer’s on-board sound card. Only the Mac version would allow you to import a quicktime movie. To be honest, I’m not sure if this is still available from the Digidesign website.

Protools M-powered is a basic entry-level Protools platform. This includes both a hardware interface and a version of the Protools LE software. I haven’t used this particular platform, so I’m unaware of it’s video playback capabilities. I do know that it will not support the extremely useful DV Toolkit software option.

Protools LE is a primarily software-based platform, and each version includes both the I/O hardware and the software. The popular Digi-002 and Mbox product lines are both Protools LE systems. LE allows you to import a video clip into your session, which can be displayed in a number of ways: as a video window on the monitor, as a full-screen video on a second monitor, or on a video screen using either an analogue video I/O card (eg. the old Pinnacle DC30) or via the firewire port through a DV transcoder (eg. a DV camera). If you can work within the 32-track limitation, this is the platform I would go for. I’ve owned an Mbox for some time now, and I’ve been very happy with its flexibility (not to mention the price). If you are going to be working with audio for film and video, you should look into the DV Toolkit 2 software option. Though expensive, the Toolkit adds a number of useful features (eg. OMF import/export, timecode and feet/frames functions).

The third platform, TDM, is probably going to be outside of your price range. TDM systems are primarily hardware-based, and provide a tremendous amount of horsepower at a steep price. Among the myriad hardware options available is the SyncIO, which allows a TDM system to chase a video machine (eg. Beta SP or V1M) via LTC or even RS422 machine control. Unfortunately, the TDM systems are the only ones that support surround mixing and TDM plugins.

The good news is that mono and stereo sessions are compatible across all three platforms, meaning that you can prepare your sound edit at home using an Mbox, and just hand it off to a TDM-equipped mix theatre for the final 5.1 mix. Also, foley and ADR recorded on a TDM system can be cut on an LE system, making that little Mbox pretty powerful in the cooperative world of audio post.

So, here’s what I’d reccomend as a minimum setup for sound editing:
– a fairly powerful PC or Mac desktop computer
– an Mbox
– the DV Toolkit 2 software option
– an iLok hardware key (needed for software registration)
– a Canopus ADVC110 (firewire in, composite and S-video out, just be sure to feed it compatible video files with the DV CODEC)
– a small television for your video monitoring
– a pair of Sony MDR-7506 headphones
– 2 additional 7200 rpm harddrives (one for audio and one for video, both rated for A/V work)

A small Mackie mixer and a pair of studio monitors would be very useful as well, but you can get those after the other gear has payed for itself.

[Stephen Muir]

The model of DAT deck you’re using probably doesn’t support TC read/generate. What model of deck are you using in post?

Also, you won’t find a standard TC track on the production DATs if they were recorded on a non-TC recorder (such as the Tascam DA-P1).

A non-TC DAT can be used to record (but not generate) TC by feeding an external TC generator into one of the audio channels. Just be careful that the TC doesn’t bleed onto the remaining audio track.

[Stephen Muir]

“Never boost when you can cut” as the old recording maxim goes.

Although a low-level recording made with good equipment may not accumulate an appreciable amount of hiss when boosted, the original recording will still have less detail than one made with propper levels. If you’re recording is 16-bit, then an under-mixed will have a very narrow dynamic range (perhaps only 12-bits or even 8-bits). The more you push it, the more digital artifacting will be apparent, because the subtle details just don’t exist in the recording. It’s analagous to under-exposing video footage, and then brightenning it until it reads propperly on a waveform monitor (in which case you can kiss the shadow detail goodbye). Although there’s no tape hiss in a digital recording, you’re still dealing with a finite amount of information.

I’m currently working on the dialogue edit for a film on which the production DATs were badly under-recorded (maximum peaks at around -30dB, averaging around -50 dB), and the dialogue sounds horrible no matter what we do short of ADR.

However, it is true that location sound is a bit more unpredictable, making it prudent to back off production mixing levels slightly (-20 dB rather than -12 dB). That said, between dual-mono techniques and a solid set of limiters (ie. a good ENG/EFP mixer) -12 dB can still be a good level to aim for.

As for the studio recording matter, unless the dialogue was being recorded to perspective (ie. it was going to be that low in the final mix), I don’t understand why the session mixer would want to intentionally provide less than optimum quality recordings.

[Stephen Muir]

Although the questions you ask are simple enough, the answers can get extremely complex.
Here’s an example of a typical chain of events from non-linear picture editorial to final mix:
– The programme’s picture edit is locked.
– The picture is output to a work tape (eg. BetaSP, DVcam) or video clip. This work tape or clip will include the 2-beep and timecode burn-in.
– The audio content of the picture timeline is exported as an audio OMF (a media data interchange format available in most professional editing software) with the appropriate amount of “handle” on the regions (eg. 5 seconds).
– The OMF files and work tape/clip are handed to the sound editor to import into (typically) a ProTools session for audio post-production
– Depending on the complexity of the project, a number of specialised sound editors will begin working on the dialogue, effects, and music.
– Spotting notes are prepared for any foley or ADR required.
– The foley and ADR are recorded.
– Once all of the disparate sound editorial elements are complete (and often premixed), the tracks are screened for client approval. Any go-backs should be completed before moving on to the next step.
– The tracks are transfered into a mix session, and the re-recording mixer begins mixing the tracks into the way they will finally sound. This is also where the different mix versions (eg. music and effects only, “clean” broadcast or in-flight movie versions) are prepared, as well as slightly different mixes tailored towards different release and distribution formats (eg. theatrical, home video, broadcast, surround, stereo).
– Once the mix is ready, the new tracks are synchronized with the final picture version (with all of the colour correction, effects, and titles), which is then used to create the marriage print or programme master tape.

This workflow will vary from project to project, but the basic elements remain fairly consistant. So, to actually answer your question: a dubbing mixer would probably ask for the original audio material in an OMF so that they would have more flexibility in remixing your tracks. If a mixed stereo file is the only option available, then they may go about it in any of a number of ways (including fader automation, normalization, and dynamics compression). The method really depends on the existing material, the desired result, and how far apart the two are. The principle that “what sounds good is good” is true if (and only if) the monitoring environment is exactly identical to the the audience’s listenning environment. Since audience listenning environments vary so much, great care is taken in designing dubbing stages. An experienced mixer will also be aware the technical standards and variables associated with specific delivery methods, adjusting the mix with those in mind.

[Stephen Muir]

When you apply the “normalize” filter, it creates a new copy of the file with the gain increase applied.

When you increase the level of a track using the software’s faders, you are increasing the gain in much the same way except that the original audio file is not replaced (ie. the level increase is being applied to the same audio during playback, but the audio file itself is unchanged). This is called a non-destructive opperation, and has the advantage of easily being tweaked or modified. Such fader manipulations are generally processed in real-time by the software, making them much more flexible than rendered effects. Just bear in mind that you will have to re-export the program in order to render in the fader adjustments.

Try not to rely too much on the thumbnail view of the audio waveform shown on the track in the timeline. When it comes to the final mixing pass, the VU meters are a much more accurate tool for judging the audio level. And even the VU meters are a poor substitute for a decent set of near-field monitors and a propperly tuned monitoring environment.

[Stephen Muir]

Unfortunately, ProTools opperates in a different headspace than FCP. Much of the workspace is designed to mimic the traditional mixing boards, and as such you’ll need a basic understanding of traditional sound recording and mixing methodology. To that end, even books about music production will give you useful information.

Rather than employing “click & add” filters to specific clips, effects are achieved either by applying “Audiosuite” effects to specific regions, or applying effects plugins (RTAS only in ProTools LE) to the channel inserts. Ironically, the cheesey effects can be a bit harder to come by for ProTools, since most of the plugins on the market are designed with an ear to quality.

As far as books, “ProTools 6 Power” by Colin MacQueen and Steve Albanese provides a basic overview of the software and hardware. It even has a brief chapter concerning post-production, although it only scratches the surface.

Are there particular effects (cheesey or otherwise) that you’re having trouble achieving?

[Stephen Muir]

For the cost of an entry-level DAT recorder, you could get a compact flash recorder such as the Fostex FR-2. The sound quality of the FR-2 (up to 24-bit, 192kHz, uncompressed) blows DAT away, and the CF cards are immune to most of the dropout-inducing conditions to which DATs have always been prone. Plus, if you’re looking to play back your recordings without having to rewind, you’ll probably want a non-linear recorder anyways.

If your budget is particularly tight, check out the Marantz PMD660 (16-bit, 48 kHz) and the M-Audio Microtrack (24-bit, 96 kHz).

[Stephen Muir]

You should be able to either opperate the recievers off of alkaline batteries or else tap into the 12V DC power output of the mixer (if it has one, and if the recievers will accept it). If your wireless reciever’s won’t opperate off of internal batteries, you may want to reconsider using that particular model.

Other items I would strongly reccomend bringing along:

– Portabrace with accesory pouch for the mixer and recievers
– High-quality headphones (eg. Sony MDR-7506)
– An emergency wired handheld dynamic mic (eg. Shure SM-58)

You’ll probably never need the emergency mic, but at around $100 its pretty cheap insurrance.

[Stephen Muir]

There’s a number of 4×2 ENG/EFP mixers on the market. Check out the Sound Devices 422 and PSC Alphamix. A portabrace mixer bag with an accessory pouch will hold all four of your wireless recievers. If your reciever packs will run off external 12V DC power, you could even run the entire audio kit from a single NP-1 battery with a battery distribution system.