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electronic MUSICIAN

Cool Tip of the Month

February 1, 2003

Setting Up a Talkback Mic in Pro Tools

The EM Cool Tip of the Month is presented courtesy of Cool Breeze Systems.

If you're recording tracks in a studio, a good line of communication between the control room and tracking room can be crucial to maintaining a creative production environment. To make communication as easy as possible, I set up a mic in the control room (out of the feedback zone) and control it with my DAW software. Personally, I hate to play with a talkback button, so I keep the talkback mic open to the headphones whenever we are not rolling, and then close it when the tune starts.

Most professional DAW programs let you set up talkback and automate the process. In this example (see example at left), I'm using Pro Tools with a multiport audio interface.

  1. Once you've configured the mixer and headphone mix, select New Track from the File menu and create an aux input (Command + Shift + N on the Mac, or Control + Shift + N in Windows). Name the aux input “Talkback.” (To review setting up a headphone mix, check out “Cool Tip of the Month” in the July 2002 issue.)

  2. Plug the control-room microphone in to the input you've assigned as the talkback. Don't forget that the microphone's signal must be routed through either an external preamp or a preamp built in to your interface.

  3. In this session, you're using sends 3 and 4 for your headphone mix, so assign the output of the talkback aux to sends 3 and 4.

  4. To control talkback, insert a compressor (Compressor II); set the threshold, attack, release, and knee to the lowest settings; and set the ratio to maximum (100.00:1).

  5. Click the External Key button and select a key source from the drop-down menu. The key source should be a continuous signal, such as a bus feed by another track or a SMPTE LTC signal if you're triggering one.

The talkback mic will now be open when no music is playing, and it will close when music begins playing.
Steve Albanese

New! Improved!

It should be obvious that when you're recording digital audio — especially on critical projects — you need the best digital converters you can get your hands on. Generally, that means using recently manufactured A/D and D/A converters. Like computers, converters tend to get noticeably better every year; even newer, affordable converters (such as the A.R.T. DI/O) tend to offer conspicuous improvements over older, more-expensive converters.

I recently mixed an album in a studio in which the mixdown deck was a nine-year-old DAT machine. I mentioned to the producer that a more modern A/D converter would likely improve the sound considerably. Thankfully, he heeded my advice and rented a high-end converter from a local pro-audio shop. While still monitoring through the DAT deck's original converters, we printed our mix for comparison purposes. Then we patched in the newer A/D converter — set to the same bit-rate and sampling frequency as the DAT — and printed the mix again. The difference was palpable: the low end on the second mix was bigger, deeper, more solid, and more pleasing to listen to; the highs were smoother and more airy; and, overall, the sound seemed more natural and true. Even the stereo imaging was enhanced, making it easier to pinpoint instrument locations on the soundstage. In the end, the couple of hundred dollars it took to rent the new-generation, high-end A/D converter was money well spent.
Brian Knave

Lord of the Ring Modulator

Ring modulation (also known as balanced modulation) is a process that takes two input signals (a carrier and a modulator or program) and outputs only the sum and difference of the two tones. For example, if you feed the ring modulator a pair of sine waves, 300 Hz and 200 Hz respectively, you will get tones at 100 and 500 Hz outputs. At the same time, the original signals should be removed.

If you are using a Big Briar (now Moog Music) MF-102 Ring Modulator and you hear the original tones bleeding through, a simple procedure can eliminate them. You'll need a sound source that gates quickly (such as keyboard sound with a brief decay). Begin by removing the bottom panel of the MF-102. Note where the VR1 trimmer is located: it is the one closest to the rear-panel jacks and has the blue cap. You will be adjusting this trimmer with a small screwdriver.

Set the oscillator Frequency control to a high pitch (around 2 Khz will do). Plug your keyboard or sound source in to the input and send the output to your amplifier. Play a short note; you'll hear the carrier die out about two to three seconds later. Keep hitting short notes in order to keep the squelch open. As you play, adjust VR1 until you hear the absolute minimum carrier bleed-through. You may be unable to attain absolute zero bleed; just strive for the minimum point. In doing so, you are nulling the carrier signal that comes from the balanced modulator.
Gino Robair

Time Off

Time stretching and time compressing are now standard features in most audio-editing programs, and those functions can be lifesavers if a block of dialog or a musical cue is a tad long or short for a particular time segment. Keep in mind, however, that stretching and compressing time are highly complex DSP operations, and they always degrade the audio to some extent.

Most audio editors offer a range that extends from half speed (50 percent) to double speed (200 percent), but the outer limits of the range are practical only when fidelity is not important or when you're purposely creating special effects. If your intent is to shorten or lengthen an audio clip without introducing excessive distortion, clicking, or other artifacts, you'll have to abide by some practical limits. Here are some rough guidelines to follow when stretching or compressing sound files:

In order to achieve the best results with speech, keep your stretch factors within about ±30 percent. For musical arrangements with overlapping instruments, it's best not to exceed about ±10 percent. For solo piano, classical music, and other orchestral styles, you will probably need to stay within ±3 percent.
David Rubin

Multiplying Loops

You can achieve interesting rhythmic effects by overlapping drum loops or any sounds with a beat-synchronized component. Instead of simply laying them end-to-end at measure boundaries, try placing a loop so that it starts at a quarter, eighth, or 16th note after the downbeat of the previous loop. This technique works equally well for different loops or for multiple iterations of the same loop.
Marty Cutler

Fade to Black

A classic recording-studio technique can create the illusion that your music is fading into the distance. At the end of a song, as you bring down the main faders for an extended fadeout, gradually bring up the reverb level on the main mix. As the sound level diminishes, its perceived distance will increase.
Geary Yelton

Make sure to check out the streaming movie tutorial of this tip to view this procedure in action. Log on to to take part in this online adventure. Also, if you dare, take the quiz to review what you've learned!

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