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February 14, 2007

Filed under: music»recording»production


At some point, I'm going to have to pick up a copy of Pro Tools for my home studio, for two reasons. First, I'll probably be consulting for WBI after my contract runs out, and I want to be able to open project files without any import/export issues. Second, I've honestly grown to like Pro Tools. Cubase is good software, but in my (admittedly limited) experience with it, it's the kind of software that believes every function needs a new window. Soft synths? New window! Mixer? New window! MIDI editing? New window! Channel strip? GUESS WHAT WE'RE DOING!

I kid because I love, of course. Cubase also has a lot of great features that Pro Tools doesn't have, like support for VST plugins and a wider range of hardware. And even though it opens up all those windows, it does get credit for making it easy to move between them--from almost anywhere in the application, you can get to a track's plugins, sends, and channel strip information. But for fast and flexible audio production, Pro Tools is still a monster.

But it's never been budget software, frankly. Some may disagree with the comparison, but Digidesign (the people behind Pro Tools) have always reminded me a bit of Apple: they like to use their own hardware (the merits of which are strongly debated), they're seen as more expensive than the competition, and they're pretty much standard issue at professional studios. Of course, in professional studios, a Pro Tools rig doesn't mean the same thing that we've got here at the Bank. A top-of-the-line Pro Tools HD setup offloads the effects and signal processing to outboard DSP chips contained in big, expensive rackmount units. Home users unwilling to spend more than $10,000 on a recording setup have two Pro Tools choices that are file-compatible but use host-based processing instead: Pro Tools LE and M-Powered.

My dilemma comes from picking between those two home versions. At the Bank, we're using an LE system (the very nice Digi 002), which requires Digidesign hardware (in this case, a big mixer-looking chunk of machinery that acts as both an interface and a control surface). That used to be the only budget choice. But then Avid (the parent company of Digidesign) bought M-Audio, makers of a ton of audio interfaces, and together they put out Pro Tools M-Powered, which is basically identical to LE but only runs using M-Audio hardware.

Now, I already own an M-Audio interface, the Firewire Solo. I like it. It seems solid, it's very low-latency, and other M-Audio hardware is relatively cheap so it would be easy to upgrade to a bigger system. Going to Pro Tools just means buying the software, which runs about $250. The downsides are that it's dongle-protected (so I'd have to carry around a little USB key in addition to everything else) and it doesn't come with as many plugins as the LE packs do.

Normally, I'd just have to bite the bullet on those deficiencies, because Digidesign's most affordable LE system was the Mbox 2, starting at $450. But they've just started shipping the Mbox 2 Mini, a small USB-based LE system. It doesn't offer very much in the way of input, but it's only $300, or $50 more than the M-Powered system. The Mbox also acts as a hardware dongle, meaning that I wouldn't need to carry the iLock dongle in order to use the software. On the other hand, it's expensive to upgrade an LE system (the cost of software is built into the price for new hardware) and I might still need a USB key for authorization if I bought any plugins.

So although I'm tempted, in the end I have to believe that for my small studio M-Powered will be the most logical choice. I like having a bigger choice of hardware, even if it does all have to come from M-Audio, and the overall costs are probably much lower (on par or cheaper with competitors, actually). If I had a couple thousand dollars, I'd probably want to shell out for a the Digi 002 system, because I've learned to appreciate having well-built physical faders for mixing, and then I'd add an Mbox 2 Mini for portable work. But I don't have that kind of money. I'm thinking I'll pick up the M-Powered, and then use the extra $50 toward either a Jamlab USB interface (very portable) or fxpansion's VST-to-Pro Tools plugin wrapper (thus negating the only feature I'll really miss from Cubase and Ableton Live).

February 4, 2007

Filed under: music»recording

Book Review: Capturing Sound: How Technology Has Changed Music, by Mark Katz

I've made no secret of my obsession with digital audio and its effect on listening habits. When David Byrne mentioned that someone had written a scholarly book exactly about that topic, I went looking for a copy as soon as I could, although I expected to be disappointed by it. But surprisingly, Capturing Sound is an interesting and accessible--if lightweight--look at the history of interaction between recording technology and music.

Those expecting an immediate leap into the intricacies of MP3 and DRM will have to be patient: Katz basically proceeds in chronological order, beginning with the first phonographs and how they were meant to make America "more musical." Subsequent chapters explore the ways in which jazz musicians altered their style and timing for recordings, how recordings began to emphasize violin vibrato, "grammophonmusik," and DJ turntable battles. In each of these, Katz's goal is not too show that technology irrevocably led to a specific cultural output (determinism), but to show that the musicians and the machines interacted.

There's a lot of emphasis on phonographs and old recording styles in the book, material with the potential to be boring or overly nostalgiac. But Katz has a deft hand for anecdotes and revealing stories that liven up his subjects. For example, when discussing the spread of the home phonograph, he talks about the Graduola--basically a volume control switch attached to the playback mechanism. Although it sounds silly to modern sensibilities, the Graduola apparently gave phonograph owners the feeling that by controlling the volume, they were "conducting" the record, and by extension lent a feeling of "musicianship." Then again, anyone who has watched an exuberant game of Guitar Hero may not think it's so silly after all. Are music games the new Graduola?

Katz's chapter on DJ battles was, for me, one of the most fascinating in the book. He explores the background and social culture of turntablism, then spends several pages on a step-by-step description of a winning turntable battle track. It's an impressive glimpse into a maligned musical genre, although he warns that the track itself (included on the CD in the back of the book) will not sound as interesting to untrained ears, and indeed it's barely comprehensible. Capturing Sound also gets off to a strong start in the final chapter, on digital sampling and audio, where he examines Fatboy Slim's "Praise You" at the same critical level of detail. It falls slightly short when he begins discussing MP3 audio, but peprhaps this has more to do with the preponderance of discussion around MP3 lately (there's little new anyone could say about it at this point), and less to do with Katz's focus.

Capturing Sound is carefully nonpartisan when it comes to its subject matter. Katz is no technology apologist, and although he may be an enthusiast he seems to be a cautious one. It's thought-provoking and, at 190 pages for the main text, a short read. After putting the book down, I felt like I had new tools to compare and understand the differences between live and recorded music, and obviously I think that's a good debate to be having.

January 8, 2007

Filed under: music»recording»production

Quick Fixes for Better Sound

Craig Anderton has an article in EQ Magazine this month with lots of cheap and easy recording fixes. I'm interested in making the change to 88.2kHz, 24-bit audio myself, after a Sound On Sound editorial discussed why bass sounds better when there's more resolution available to describe low-frequency waveforms.

Not to come across like a broken record, but what you can't do is increase audio quality after discarding large chunks of digital information--i.e., compression to MP3 or AAC. Wired reviews a few devices that claim to restore audio quality to portable music. My favorite snake oil is the third review, the Creative X-Fi, which garners a good score even though Wired can't say for sure what it did, or even if it did much at all. Judging by the demo on Creative's site, it sounds like a smile EQ setting, and possible a little drive for warmth. You'd be better off buying a bigger hard drive, and just ripping your CDs lossless. It takes a bit more work, but there would be a real, valuable audio difference.

Filed under: music»recording»mp3


A final version (by which I mean that the arrangement is solid and the quality is good enough for the web) of "Mastermind" is now available at Four String Riot. Obviously the little gimmick samples are not technically kosher with my manifesto, but I couldn't resist considering the subject matter.

I did manage to resist using the Digitech Whammy pedal I picked up this weekend, but it is probably the coolest pedal I've ever owned. When the Gib Cima Experience played "Werewolves of London" on Saturday, I kicked in the octave up mode to do the guitar solo, and with a little distortion it sounded just about perfect, including double stops. I may have to record it as a sketchpad. On triple stops or more complicated chord voicings, it still throws up all over the sonic spectrum, but overall it's certainly the best pitch shifter available short of a $1,500 Lexicon MPX-G2 or an Eventide harmonizer. The Whammy's octave harmonize function also does a nice 8-string bass imitation, and I've been playing Pearl Jam's "Jeremy" every time I turn it on.

January 2, 2007

Filed under: music»recording»production

Cubase Tip: ASIO Selection

I just found this the other day when I reinstalled Cubase on my recording laptop. The copy protection for Cubase LE is luckily (and thankfully) nonexistent if you have a working copy somewhere, although I did have to register a couple of DLLs with Windows. But after loading the software, even with the new Firewire interface, sound latency was awful--something like a full second round trip. It's impossible to monitor yourself through software when the delay is that bad, and although there's a direct mode on the interface, it involves opening up a mixer window and fiddling with the inputs. I didn't really want to do that, especially since I know that Ableton Live and Phrazor are perfectly capable of fast software monitoring with this hardware.

It turns out that Cubase LE can manage just fine, but the option is hidden. By default, Steinberg includes a driver that wraps the lowest levels of Windows functionality (MME, or maybe DirectX) in an ASIO layer. It works, but it's really slow. I'd known this all along, but I thought it was a restriction built into Cubase LE to encourage upgrades. That'll teach me to be cynical--they've just give the menu a very strange name. To change ASIO drivers in Cubase, choose "Device Setup" from the Devices menu, and then open the "VST Multitrack" tab. There's a pulldown menu that will initially read "ASIO Multimedia Driver"--when I opened it up, there was my Firewire Solo (as well as ASIO4All). Cubase still has trouble operating at the very lowest latency, but I had a comfortable experience running VST plugins on a recording track with 44.1KHz, 16-bit audio and a sample size of 128. Much better.

The only reasons that I can think for Steinberg's weird choices here are two-fold. First, they've obviously included the ASIO-MME driver for compatibility, and they don't want to take a chance on auto-selecting the wrong driver. Second, they've hidden the option in the "VST Multitracker" tab (which I had seen, but always ignored) because Cubase started as a sequencer instead of a recording workstation. Latency isn't as important for predetermined MIDI sequences, and from that perspective they might have referred to audio I/O as a "multitracker" instead of more straighforward terminology.

December 4, 2006

Filed under: music»recording»production

Standard Time

In a throwaway line from the introduction to his piece on tempo maps in Cubase SX, Sound on Sound columnist Mark Wherry asks:

It might be an interesting study to see how much music has been written with a fixed tempo of 120 bpm in four/four time over the years, just because this is the starting point presented to the user in almost every sequencer of the last 20 years.

This is a very good question. Another good question, depending on the answer to the first, would be to ask if 120 bpm (or divisions thereof) has become a unconscious tempo for many musicians precisely because we've heard so much music at that speed since the rise of digital sequencers.

November 25, 2006

Filed under: music»recording»mp3

Four String Refresh, Part 2

The process of re-recording old songs continues, with overhauled versions of "Lazy Sunday Eyes" and "My Foundation," as well as yet another rendition of "Voodoo Funk." Here's a few thoughts on recording these:

  • I finally figured out why I have so much trouble getting good volume out of my recordings. It's because the bass has such a huge dynamic range the way I play it--at any give time, I could be plucking, strumming, slapping, or literally pounding on the strings. Because I always wanted to avoid the nasty popping that results from overdriving a digital input, I was setting the preamp level according to my highest peak level--in this case, percussive slaps. Since my amp overdrives gently, I hadn't realized how much louder those peaks actually are compared to the rest of my signal. By setting them as the highest, everything else was recorded far too soft. The ultimate solution would be to record the amp itself, but my cabinet isn't in a great acoustic location, and I don't really want to mic my practice amp. After that, preferably I'd add a compressor before the interface, but I don't own a hardware compressor. Instead, I added a carefully set software compressor to the track to limit the pops but boost the gain on everything else. This raises the noise floor a lot, but it's good enough for now.
  • I've also switched to an M-Audio Firewire Solo interface, for a couple of reasons. One is that my Tascam US-122 was obtrusively noisy when using phantom power, which I need for my condenser mikes, and I was having to run them through a mixer. The Solo is externally-powered, so I can eliminate the mixer from my signal chain. Also, using IEEE 1394 instead of USB lowered my signal latency from 45ms to 6ms, making it possible to effectively monitor myself and (hopefully) cleaning up my timing a little. Neither of these were dealbreaking problems, but it's nice to have them addressed, and the Solo isn't that expensive.
  • The software pedalboard experiment is pretty much over. Although it was interesting and I still think there's a lot of power there, I was never entirely happy with the results. I've gone back to my MXR M-80 preamp, DOD Envelope, and Line 6 looper, and I'm just running the MDA Combo plugin to tame the harsh treble and boomy low-end. It's nice to have knobs to play with again.

Hear the results, as always, at the Four String Riot. Next up is to finalize/record "Mastermind," and then start writing again.

November 20, 2006

Filed under: music»recording»production»post

The Amateur teaches Pro Tools

Apart from producing the podcasts, recording voiceovers, and editing a couple of radio shows, most of my production work at the Bank involves supporting the video editors with their soundtracks. We use Final Cut Pro in the Multimedia Center, and although I'm sure it's a fine tool for video editing it doesn't seem to be very effective for more than the most basic audio work. For one thing, the effects need to be rendered before you can hear them, and I can never actually get any audible changes out of them (I'm probably doing something wrong, but the video people avoid the audio side like vampires in an Italian kitchen, so they're not much help). Soundtrack, the tool that accompanies Final Cut, is all well and good--but it's not really my cup of tea, and none of the editors want to put in the work to learn it.

So with all that in mind, here is a quick set of two tutorials for the common tasks that I perform while working my magic on soundtracks. I need to write a tutorial for my co-workers anyway, it might as well be now. Also, note that although Pro Tools can work with multi-track audio through the OMF/QT import part of the DV toolkit, you can also work with a stereo .wav of the whole soundtrack--and I often do.

1. Better Vocal Ducking:

When you bring in a voiceover on top of background noise, you want that background to get out of the way so that the vocals can be understood. That means lowering the volume, usually. Now, you can go through and manually adjust the volume using the track automation, but that's a pain and you might miss something. I prefer to do it through plugins. Now this is a pretty standard part of audio production, and you can google plenty of advice on it--assign a compressor to your background track, sidechain it to the vocals with a medium-high ratio, and its volume will automatically lower whenever the vocal track plays.

The problem with this, as my manager immediately pointed out when I first started using Pro Tools, is that it only ducks the volume right as the vocal comes out, and it sounds more natural if the background can start to fade just a little bit before the vocals actually enter, as if someone had anticipated the vocals. This also preserves the first word of the voiceover--it doesn't get lost in the slight pause before the compressor really kicks in.

I solve this problem, like most of my audio magic, with creative use of sends. Basically, you want to split your vocal track into two directions. Change its output to a bus, say Bus 1, and add a Send to a different bus, Bus 2 perhaps. Use Bus 1 as the background track compressor's sidechain input (the key icon in RTAS plugins will activate sidechaining if it's supported, and let you pick an input), but instead of setting the compressor to the usual settings, you want to give it a moderate attack and a very long release. I usually use the following settings (although I'm quoting from memory, so it may be off a little):

  • Ratio: 4.3
  • Soft knee
  • Attack: 153ms
  • Release: 2.5 sec
  • Threshold: -45db
Now, create an Aux track that takes input from Bus 2 and outputs to your speakers (or wherever). This is the path for the voice that you'll actually hear. Add a delay to it (I use the stock medium delay plugin set to 100% wet and 333ms, which is coincidentally twice as long as the attack on the compressor).

See what we've done? You've used the vocals to trigger a slow compressor, hopefully creating a fade, while simultaneously adding a delay to the audible vocal so that it'll arrive after the compressor turns down the background. You do need to be careful with timing, obviously, because the timeline display is now 333ms offset from your ears, but if you're after precision you should probably just invest in a plugin that supports look-ahead. This is the cheap way to do it.

2. Remove a noisy camera/audience:

Here's something to remember about audio: it's easy to add, but not so easy to subtract. Unlike video or images, you can't just cut a noise out, because it's part of the soundwave. Some tools, like Adobe Audition, will let you paint out chunks of the spectrum, but it's still not a perfect solution. And remember, sound is a representation of a physical phenomenon--in a lot of cases, it's the mic capsule or coil moving that is reproduced through your speakers. So physical movement or different sound frequencies can actually mask other sounds, because they're physically moving the mic and changing its interaction with the sound.

That's all very fun and technical, but what it amounts to in real life is that a lot of footage is shot in bad locations, through crappy equipment and sub-optimal mike technique. Maybe a speech was shot using the camera mike at the back of the room instead of plugging into the PA system. Maybe it was done on the move, and there's a lot of wind and crowd noise. Editors want that gone, or at least reduced, so you can hear the subject.

This is relatively easy to do. Remember that most of a voice's content takes place between the frequences of 100-5000Hz. Outside of that, you might miss some of the sibilant consonants, or low vocal rumble, but you'll be able to understand a person. Also, most electrical and camera noise takes place at the upper and lower limits of the spectrum. So to remove physical handling noise, like bumps, and boomy acoustics, I open up an EQ plugin and set a high-pass filter with a very sharp cutoff at around 180Hz, fine-tuning a little through headphones. I put a low-pass filter at around 6KHz, which minimizes clicks and a lot of tape whirr. If you add these through the AudioSuite menu of Pro Tools, using the preview function to listen before you apply them, you'll end up with a new, processed region that you can export back out for Final Cut.

November 19, 2006

Filed under: music»recording»mp3

Four String Refresh

There are new versions of "Voodoo Funk" and "We Used To Be Friends" up on the pretentious solo project, as well as the the hateful Myspace. With that said, if you were going to listen to them right away (crickets chirp, tumbleweed, the sound of echoes in a large empty space) you might wait until tonight around 8pm EST--they still need some tuning, and makeup gain (the Tascam US122 has many virtues, but hot input is not one of them). The goal of redoing these songs was that the previous recordings were pretty distorted and sometimes had boxy vocal sound.

Next up are my really old recordings, and then I need to put together three more songs or so before I go out and humiliate myself trying to gig for real again.

October 7, 2006

Filed under: music»recording»production

Sing Me Spanish Techno

GDLN World Forum Soundtrack, Take 1

As I said, I had to compose some soundtrack music for the GDLN World Forum this weekend. The only guidance I got was that one of them needed to be a techno remix, and the other needed to be more calm. They also needed a transition between the two. I'm no Fatboy Slim, but I think the results sound pretty good, and it's surprisingly fun doing this kind of grid-based music in Pro Tools.

Everything was created using the XPand! softsynth, including the techno breakbeats--I can't take credit for those, unfortunately, but it's not like most house composers waste a lot of time on their beats, and I was actually trying to get as close to the Amen Break as possible. I'm thinking about adding some audio samples if I have time on Monday. I'd like to get my coworkers to come in, say a few phrases in different languages, and then chop those up on top of the second half.

Future - Present - Past