For a while, I've been following a recommendation for field recording; it suggests using an XLR Y-cable to record two tracks from the same source. In my case, this source is usually a shotgun microphone at the end of a boom pole. It may seem redundant (it is), but over time, I've found it to be a method worthy of mention.
I touched on this over on the Documents page. If you're interested, go check out the downloadable sound report templates. I find this to be particularly useful in recording location audio for film, but it also shines when close-micing foley sound, or just about anything else at the right place and time.
First, understand this: options are good. It is certainly better to have extra microphones and channels, should you need them. However, there are situations where this isn't an option. For example, you have a limited selection of microphones, channels to record to, or limited space to record in. Recording dialogue in the corner of a cramped room (next to lights, camera, other crew, etc) doesn't exactly cater to having two boom operators or setting up mic stands.
Why, though, would you need two tracks for a single source? Well, when your subject goes from a whisper to a shout, and back again, the ground begins to shrink beneath you. Barring asking the director or talent to change their delivery, your choices are few:
- Ride the gain, and compensate for the shifting noise floor in post-production
- Use a limiter on capture, while the signal pumps and shifts automatically
- Capture the scream on location, while the whispers risk remaining in the mud
- Bite the bullet- mix for the lower signal, and ADR the scream.
Of course, this "scream" could be a loud response on impact, unpredictable ambient noise, or anything else that clips too far to be reasonably captured on the same track as a low level signal. Also, ADR and foley may be out, perhaps if you're recording an interview or documentary.
So, on we go, in search of a solution.
And here, we arrive, at splitting one source to multiple channels.
From the beginning to the end, the signal travels from the microphone, through a single cable, split through the Y-cable, and into two independent channels, each with it's own preamp and gain level. I have found the most painless procedure to be recording an "A-track," mixed as if it were the only one, while a "safety" track with a lower gain awaits something louder. This means that editing is the same as always, but that no content is lost, and can be recalled at a slightly lower signal-to-noise ratio (SNR) when needed.
| | In fact, Propellerhead's own Balance interface has a feature called "clip safe" that performs this very function when used with Reason. |
"Clip Safe is a magical lifeline for distorted takes – a red-eye tool for recorded audio. Clip Safe records your take on two channels, one at a lower volume. If your recording clips, it lets you substitute it for the other one. Don't ask us how it works, it just does."
The main drawback to this technique is that if you need it, it can involve heavy editing and some significant processing. For sources that frequently jump from barely audible to barely tolerable, the time required in post can be bothersome, at best. This is a sacrifice, but for mission critical work, time in post is insignificant compared to losing a recording of a rare event.
Moving forward, the tracks are imported to a DAW or NLE of choice, and editing begins. I'll be using Pro Tools 10. I'll also scale the waveform view to show detail; in other words, the level of each track is relative to the other, not to the size of the waveform depicted. The "flat" part is still 0 dB.
Important: If you are first synchronizing sound to picture and editing visually, make sure not to delete any audio tracks, even if they clip! It will sound better, but it's best left to audio post. Of course, if you're aware of this process, you can edit at your own discretion.
Moving forward, the tracks are imported to a DAW or NLE of choice, and editing begins. I'll be using Pro Tools 10. I'll also scale the waveform view to show detail; in other words, the level of each track is relative to the other, not to the size of the waveform depicted. The "flat" part is still 0 dB.
Important: If you are first synchronizing sound to picture and editing visually, make sure not to delete any audio tracks, even if they clip! It will sound better, but it's best left to audio post. Of course, if you're aware of this process, you can edit at your own discretion.
In this first screenshot, Tracks 1 and 2 are shown, recorded with the method described above. It isn't anything special, and is simply a verbal slate, followed by a clap and a cough that clip Track 1. This will do for a demonstration. Click the images for a better view.
The flat lines at the top of the waveform signify overmodulation of the signal. This is bad! Track 2 does not clip, but the verbal slate before the clap is significantly softer than the same portion of Track 1.
Editing in this manner isn't too difficult; I've chosen to select only the distorted parts of the clip, rather than the entire impulse. This is possible because the content is exactly the same on both tracks. I find it easier to make small cuts and fades here instead of treating it as if the tracks were mutually exclusive; any noise brought up will hopefully be masked by the actual signal.
After the selection is made, it's separated, as is the safety track.
At this point, the distorted audio is removed; however, as long as the selection remains unchanged, it will audibly jump down, then up by the difference in gain initially set on the recorder. To rectify this, we can use clip gain to quickly even it out. Alternatively, it can be done using the Audiosuite Gain plugin.
This is where quality preamps are important. I've raised the gain on this clip by 12 dB, but the difference can easily be much higher. The higher the difference here, the more your noise floor will come up. Fortunately, if your B-track has a strong, clean reading, the noise shouldn't be a problem.
This is where quality preamps are important. I've raised the gain on this clip by 12 dB, but the difference can easily be much higher. The higher the difference here, the more your noise floor will come up. Fortunately, if your B-track has a strong, clean reading, the noise shouldn't be a problem.
At this point, we can trash the backup track, since this is the last cut being made for this example. Two small fades (the minimum in Pro Tools is four samples long), and we're ready to mix and process.
But beware! If we consolidate it into a new file now, we could lose the information over 0 dB that we worked so hard to save! A floating point bit depth would allow us to retain it for normalization later on, but instead, we'll group the clips to process them altogether.
But beware! If we consolidate it into a new file now, we could lose the information over 0 dB that we worked so hard to save! A floating point bit depth would allow us to retain it for normalization later on, but instead, we'll group the clips to process them altogether.
Above, the early consolidation leaves us with the same problem we started with. Below, the signal hits far higher than 0 dB, and will clip the D/A converters if left unchanged.
Now, it's time to process. We can treat this as any other sample. We can do this non-destructively and in real time with an insert or send (the group has its own clip gain, and can be ungrouped at any time), or use an Audiosuite plugin to process and consolidate, all at once.
With a little compression, the VU meter on the left no longer peaks.
So, what are we left with? The dynamic range we've been able to capture is considerably large, and the noise floor is nice and low. Using a low-budget solution (under $20), we've saved ourselves a lot of hassle (or being hassled!) rescuing or recreating damaged content.
Other possibilities exist, such as implementing a multi-band limiter in field recorders, though this kind of processing would likely require more than a simple firmware update for existing equipment. Maybe it's already around. I don't know, but it's definitely something I'd like to see. As far as front end processing goes, you can mix and process before the signal hits the recorder, but it's costly to obtain the necessary equipment, takes up precious space, and results are rarely perfect.
Zaxcom's NeverClip has my attention. It uses two Analog to Digital Converters (ADC), operating as a function of the recorder, rather than manually routing through two preamps. I'd like to try it for myself sometime. Check out this demo.
Other possibilities exist, such as implementing a multi-band limiter in field recorders, though this kind of processing would likely require more than a simple firmware update for existing equipment. Maybe it's already around. I don't know, but it's definitely something I'd like to see. As far as front end processing goes, you can mix and process before the signal hits the recorder, but it's costly to obtain the necessary equipment, takes up precious space, and results are rarely perfect.
Zaxcom's NeverClip has my attention. It uses two Analog to Digital Converters (ADC), operating as a function of the recorder, rather than manually routing through two preamps. I'd like to try it for myself sometime. Check out this demo.
Anyone with thoughts on this is welcome to comment. You can reach me via email at Kyle@kylehughesaudio.com for a discussion over this or a request for a post on anything in particular.
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