Recently Mix magazine published an article, “Analog Rules”, describing the audio production methods used by several top producers and engineers. Key to these methods is the combination of analog and digital techniques. This is nothing new, but what was interesting was the renewed interest in tape as key component. Tape imparts a unique sound to tracks as a result of many factors including saturation (a kind of pleasing distortion, often termed musical distortion), subtle variations in the speed and imperfect contact with the heads, as well as the influence of discrete analog electronics (resistors, capacitors, transformers, etc.) on the signal path.

Now in 2010 digital audio has improved so much that in conjunction with good analog outboard gear (preamps, compressors, eq) it sounds very good indeed. However I recently acquired a 2-track SONY MCI JH-110 1/4″ tape deck from a Nashville engineer, Chris Mara, who runs Welcome to 1979.com, a primarily analog recording studio. Using this deck as a final link in the mastering chain has given me one more tool. I can take a previously recorded digital mix, bounce it to tape, and bring it back into the DAW, and compare it to the original mix. Often (although not always), I like the added nuances that tape provides. I describe the tape sound as a kind of “rounding and smoothing” of the audio, especially for transients. Sometimes this dulls the mix, in which case if I like the original digital version, I will keep that. But more often than not I prefer the tape version.

Tape as a production medium was nearly pronounced dead 20 years ago as digital tape and later digital audio workstations (a.k.a. “souped up computers with professional audio interfaces and sequencing software”) gained acceptance. Digital was welcomed for its improved editing, low noise, and relatively lower cost compared to tape. However digital audio has had from the start its detractors, including such notable musicians as Neil Young, and top flight engineers such as Bob Clearmountain, according to the Mix article.

I also track directly to tape and dump the tracks into the DAW. In that case I usually do not have to bounce in the final master stage. However with the cost and extra work of using tape, and the fact that many of my clients are amateur musicians or part-timers who are not always fully prepared, I tend not to use this technique for tracking as often as I might.

Combining tape and digital is the same approach that many engineers, including those mentioned in the Mix article, are using. Nice to know that this hybrid approach is working for so many others. I certainly like it, and will continue to employ it in my own music production.

When mixing and mastering, there are many ways to approach the workflow, especially when working both inside and outside the “box”. In this case my “box” is an Intel Quad Core, 4 Gig, XP Pro system using SONAR 8.5 PE for tracking and mixing (and occasionally Reaper 3), and Wavelab 6 for mastering and CD production.

Here is my workflow:

Using SONAR, where I originally laid down the tracks, I mix out through my console and outboard and then back into SONAR, and export the final stereo mixdown to a project \Mix folder. The mix version retains the original bit depth and sample rate, usually 24/48 or 24/88.

In Wavelab I load each mix file separately for mastering, create the master section fx chain, and save that as a per-song preset. I will set the loudness using the WL meters, or the meters in UAD Precision Limiter, set to K-14. I also use UAD Precision Maximizer, or possibly the Fairchild 670 plug, to get the sound where I want it in terms of loudness and clarity, warmth, etc.

Then I render out to same bit depth, with final gain set for the target avg RMS (I use the Global Analysis tool all the time to check this), generally -14 DBFS plus or minus a db. No dither yet. Note that this loudness level is average perceived loudness, set to less than today’s pop music standard, i.e. it retains much of the original dynamic range and is still loud enough to sound good on CD compared to many other similar tunes. As I am mostly doing acoustic, folk, Celtic, electronica, country and bluegrass, i don’t feel the call to squashing the sound just to make it competitive for radio play, as has been the trend in recent years. Anyway…..

Each of these 24-bit mastered files is kept in a \Master folder in the project subdirectory, so I can keep track of when I last rendered out a final version. At any time I can go back to the mix versions restore the preset for the tune, and work it some more if I need to.

I create a CD project montage in Wavelab, set order of the tunes, do the fades (top and tail), spacing, and only then apply dithering to the final rendered stereo file and basic audio CD file. That way I can adjust final levels in the montage if I want to before dither.

It might sound like a lot of work but it gives me pretty good control over each song, without having to render the entire CD montage with all of the original per song plugs, which takes a while. The final render only applies dither.

After the final render of the project I can then burn reference CD’s, which i listen to in a variety of contexts (home stereo, car stereo, iPod). Once its all good, a final master is burned onto a high quality CD-ROM, such as Apogee or Tayo Yuden, at 8X speed (which my DAW seems to like better than the often quoted 1 or 2X speed).

Here are the clips I referred to in my previous post about the effects of partial phase cancellation when recording vocals with two mics and two artists. The first clip shows the results of two tracks, each with the vocal mic but the bleed from the second mic is retained. The second clip shows what happens when I edited out the bleed on the second mic track.

PhaseExample_In

PhaseExample_Out

Just finished a voice over recording session for a radio play. The play will be on an upcoming CD release for the heavy metal group, Butcher. The voice over was performed by two actors spaced 3 feet apart, facing each other. The large diaphram condensor mics were set to a cardioid pattern, but of course there was considerable bleed between the two.

I used a U87a on the female actor’s voice, and a Michael Joly modded MXL V67 on the male’s voice, each recorded to separate mono tracks in SONAR. The U87 was clear and powerful, full of presence, as I expected, but the sound from the V67 was thin and hollow sounding, which I noticed only when playing both tracks back together. This was not what I was expecting from the Joly modified V67.

Then it hit me! The bleed into the U87, being 3 feet away, was introducing a 3 ms partial phase cancellation for the male actors voice, when listening to the U87 track.

Its about 1 millisecond delay per foot of separation, that’s recording 101 for you. And yes it was considerably noticeable. However when I muted the U87 track and listened to the male actor voice soloed, it sound very good. Not U87 good, mind you, but much better. Considering the V67 is a $300 mic (after the Joly mods), it holds up well as a 2nd stringer to the U87.

So rather than try a different mic I just dealt with the phase cancellation in the DAW, when mixing. Its a simple matter to edit the tracks to eliminate the bleed, as long as the two voices are reading distinct parts and not talking over each other. Fortunately that is the case for this script, just a few places where you will hear both voices together.

Being out of phase is something that recording engineers really need to pay attention two, especially for vocals, when recording with multiple mics. Having better isolation also helps, but I did not have the room nor the available “gobos”. I might have used one of Ethan Winer’s Real Traps Portable Vocal Booths in this situation, but I wanted the talent to face each other as they were carrying on a “conversation”.

I will post some examples of this partial phase cancellation for anyone to hear, so you know what I mean.

So on to calibrating the console. This was more of an attempt to see if there were any level differences between any of the 16 channels. I have an older Soundtracs PC Midi 16, which was originally manufactured in Great Britain in the 80’s. This is a large, heavy beast for just having 16 channels, but I like the sound I get. However being an older console its important to check to see if any of the capacitors (caps) are wearing out, if there are wiring problems, etc. I have not opened it up to inspect it (yet), so for now I am just going to run signal into it and see if I can hear any major problems with any of the channels. Since I record both ITB (In the Box) and OTB, using the console, I have spent some time rigging a series of patch bays which will allow me to easily switch between recording to tape and to the DAW, as well to configure how the output from the DAW is routed. Normally I route a stereo pair of outputs from the Lynx Aurora convertors to my Central Stations monitor controller. When I want to use the console for analog summing, as well as to take advantage of mixing using real faders, EQ, etc., I patch all 16 channels out of the Lynx into each of the 16 Tape Inputs of the console, using the patch bay. I have rigged a pair of 8-channel snakes, one set for the outputs, and one for the inputs, into the patch bay, so switching my routing patches is pretty easy and fast. So I set up the console to take in all 16 channels from the Lynx, and played the Pink Noise signal through each. The levels on all but channels 15 and 16 matched perfectly, showing about +3 dB on the LED meters. I then adjusted the master faders until the stereo output was set to show 0 dB on the master output meter LED. This is not exact, using these LED’s but its close enough so that now I have a good reference level. The setting of the master faders is about -6 dB to attain 0 on the output. I have not actually measured any electrical signal levels, yet, which is what I would need to do to get a better calibration. I did find something interesting in channels 15 and 16. They were considerably hotter than 1-14, by about 10 dB. This led me at first to conclude that something bad had happened to the channel electronics. However, the difference was the same for both channels, not a likely occurrence. So I patched the 16 input into 14, and, yep the same level difference was noted on the meters. So this was coming from the DAW! So fired up the Lynx Aurora mixer software, and noticed nothing amiss in the soft fader settings, they were all set at 0 (max). Now this software mixer is not the most intuitive piece of soft kit, so rather than try to figure out why the last two channels were hot, I hit the software factory reset, and that fixed it! The levels are now the same across all of the 16 channels of the console. So the lesson is, don’t blame the old guy. Sometimes you have to spank the baby. Ok, bad analogy. But don’t blame the hardware, which could lead you down a path of big expense and waste of time, before you check to make sure your output from the computer is set correctly.