source file: mills2.txt Date: Wed, 25 Oct 1995 10:38:45 -0700 From: "John H. Chalmers" From: mclaren Subject: Live xenharmonic recording --- As more and more of you get together and produce live microtonal music (whether with MIDI instruments, acoustic instruments, or a combination of the two), sooner or later you'll want to record your xenharmonic performances on DAT. Here are the fruits of my experience in recording live to DAT: [1] The first thing to remember is that whenever a "pro" is involved, the recording will be crap. In my experience "professional sound engineer" is a code phrase that actually means "incompetent toad." Most of these people are ex-stereo salesmen who wouldn't know a balanced line from their own bum. Among the examples of expertise I've seen demonstrated by "pro recording engineers:" using ordinary drugstore rubbing alcohol to clean the heads of a 16-track tape deck; using a Calrec soundfield microphone to zero in on someone snapping a rubber band while 5 other xenharmonic instruments were playing live; accidentally forgetting to record one of the tracks of a critical stereo master during a live take; and clipping most of a digital recording. This last is worth a mention or two. One of the hallmarks of a "professional" recording is that the digital audio clips a lot. A WHOLE lot. Fortunately, all these problems can be eliminated very simply: if there's a "pro" involved in the recording process, get rid of him. (It's always a him. Women don't seem to be able to plumb such Stygian depths of ineptitude.) Do the recording yourself. The results will be *infinitely* superior. [2] The first problem you'll encounter is the ground loop. This is a pesky 60 Hz hum that can drive you out of your mind. It's caused by two grounds at slightly different potentials; 60 Hz wall current flows between the 2 grounds, producing an audible buzz. Ground loops can arise at many points in the audio chain. The 1st and most obvious place is the wall socket. In this case, 2 different pieces of audio equipment are hooked together electrically (say, a synth plugged into a mixer) but plugged into 2 different wall sockets. A ground loop can occur if the ground on one wall socket is faulty; in that case current will flow from one wall socket to the other through your equipment (an electrical signal always seeks the lowest level of electrical potential). Another particularly interesting cause of ground loop is a connection between your computer's MIDI port and your synth. This is NOT supposed to cause a ground loop because the MIDI's supposed to be optoisolated; my best guess is this sometimes occurs because the metal shell of the MIDI cable is connected both to the digital ground of the computer and the analog and digital grounds of the synth, causing current to flow twixt the analog ground of the synth and the computer's backplane. In this case you'll also get a distinctive whining noise--clock noise from the computer. The way to solve all ground problems is with an isolation transformer. The Ebtech Hum Eliminator handles both balanced and unbalanced lines and does a good job. It's also dirt cheap: < $60 for 2 stereo channels. I keep 3 of these on hand at all times whenever I do live recordings, since (as usual) if you ask any "pro" for an isolation transformer, he'll fumble around like a lobotomized trilobite & come up with a single mono balanced line version. (As always, the "pro" is too inept to realize that in the real world most synths use stereo unbalanced line outs.) If all else fails, you *can* fix ground loop hum in the mix. Digitize the audio to your hard disk, feed .25 seconds of the hum into the Dolson DNOISE program, then filter it out of the recording on your hard disk by running DNOISEwith the Fourier filter generated by the hum. Warning: this usually adds considerable reverb, since the notch filters generated by DNOISE are usually set at harmonics of 60 Hz, and as we all know running a bunch of different FFT bins at once through a soundfile always & unavoidably adds reverb due to the nature of the short-time FFT. [3] When live instruments are combined with synths, it's important that everyone be able to hear hi/rself. Often the live instrumentalist will use outboard effects and a small speaker to enhance the sound; thus as a recording engineer you'll often be faced with the problem of recording a live instrument with a speaker nearby. If you feed the output of your recording to reference speakers so that everyone can hear what the mix sounds like, you'll inadvertently recreate Robert Ashley's classic electronic music composition "The Werewolf." The only reliable way to deal with feedback in my experience is with a feedback eliminator. Turning down the monitor speakers doesn't work because no one can hear the total mix; moving the small speaker back from the mike doesn't work because the live performer can't hear what hi/r output sounds like. The Sabine feedback eliminator is in my experience the best way to deal with the problem. Because it's a digital widget, it's especially effective in killing screech & howl before it starts. There are other feedback eliminators on the market, but this one is cheaper and does a better job than anything I've come across. The cost is about $250. Naturally a "pro's" solution to the problem of feedback is: "Uh, uh, hm, uh, play lower." Typically useless; typically incompetent. [3] When recording to DAT or ADAT, you'll discover that Robert Fripp's rules of recording apply: *The SPL level of the group during live performance will always be at least twice the level during the sound check. *The percussionist will always hit the microphones at some point during the performance. *If the monitor speakers are placed close enough to the performers that they can hear themselves, there will be feedback; if the monitor speakers are placed far enough away from the performers to avoid feedback, they won't be able to hear themselves. *The tape will always run out during the best part of the performance. To avoid these inevitable problems, try the following: *Set the DAT record level during a sound check, then back it off by half, then drop it by another 3 dB. Most DATs are actually at -10 dB when they read 0 dB, so this will save your bacon during really loud sections. If the sound levels are generally within recording limits but momentarily rise by extraordinary amounts (this can occur with metallophone- type percussion instruments), add a compressor between the mixer and the DAT. *Place a coincident XY pair of mikes directly above the percussionist. Very few percussionists will try to play empty air. *Instead of using monitor speakers, use headphones for everyone in the group whenever possible. [4] Placement of microphones is critical for live recording. It differs with each room, acoustic instrument, and type of mike. You might be surprised to learn that with acoustic instruments, different frequencies radiate in different directions. Thus placing a mike front center and 30 degrees below a cello will pick up one set of low frequencies, while moving the mike to the left or right will pick up another set of mid-frequencies. This is one of the reasons why you'll often need a whole passel of microphones in different locations around one acoustic xenharmonic instrument. Cardioid condenser mikes should be placed as close as possible unless the source is metal bars or tubulongs; in that case they should be placed at least 8 feet to 10 feet away. When miking a guitar, you'll find that you get more realistic sound if you aim the mikes slightly away from the guitar's center hole. The entire body of the guitar radiates sound, and for best results place one mike up by the fingerboard and another mike down below the center hole, aimed at the instrument's body. The same appears to be true of the cello, the viola and the violin. In those instruments the bridge is one of the "hottest" sources of sound, and a mike moderately distant from the bridge, aimed at the body of the instrument, often needs to be balanced by another close-in mike aimed at another part of the instrument to get a recording that sounds like the live instrument. Again, as noted, to get the "full" sound of the instrument you may have to combine the output from a bunch of different mikes in different positions. (This is one reason why so many sampling CDs sound different from one another; only 2 mikes were used, and on each CD the mikes were placed in different positions! Yes, Virginia, mike placement is CRITICAL!) Wind instruments can exhibit sharp transients, so *always* use a puff filter. This can be something as simple as a clean sock placed over the mike or as elaborate as a $50 screen from the local music store. The acoustic result tends to be the same. If you're using PZM mikes, you can increase dynamic range by hot-wiring two 9 volt batteries in parallel. This will increase the apparent noise of the mike, so you'll have to use an additional hiss elminator between the PZM mike and the mixer. (More on hiss elimination in a moment.) PZM mikes operate unlike other mikes-- they produce the best results if they're firmly attached to large rigid slabs of wood, metal or glass. If possible, bolt the PZM mike to the floor or the wall! (Warning: if your recording site is near a street, you'll pick up low frequencies from traffic through the wall. A notable problem on recordings made with PZMs in London, since in London every recording studio is near a street with traffic.) The PZM mike also has a peculiar pickup pattern: totally hemispherical. Whereas cardioid mikes will do an excellent job of rejecting off-axis sounds, the PZM mike picks up everything indiscriminately within 180 degrees of its soundfield. This means that the PZM mike is suited only for an extremely quiet recording environment. By contrast, many of my recordings with condenser mike make been made in houses over which jet planes have been flying--yet the jet plane rumble isn't audible in the final recording because of the cardioid condenser's superb rejection of off- axis sound. The PZM mike is apt to overload during high transients. It should be placed farther away than a condenser mike. While a condenser mike will suffer drastic bass rolloff if it's farther than about 1 foot from the sound source, a PZM mike (firmly attached to a rigid plane of wood or metal) will exhibit excellent flat frequency response at almost any distance. The Calrec soundfield mike is a special case. It's noisy, but 4 of 'em output to an ADAT will allow you to "zoom" in after the recording on any of the corners of an imaginary tetrahedron. This can actually let you fix some recording problems in the mix. [5] Different engineers prefer different philosophies of mike placement. The 3 most common are the coincident XY pair, the binaural pair, and the widely separated pair. Coincident XY produces excellent results with a group in which everyone has about the same SPL; the binaural pair produces remarkable stereo versimilitude but only if the listener uses headphones. The widely separated pair (or quad, or octet) is useful for instruments physically distant and with very different SPLs, but tends to produce a final recording without a convincing soundstage. In general, the smaller the number of mikes, the more realistic the soundstage of the recording. Some of my CDs are reissues of Mercury Living Presence recordings made with coincident XY pairs in 1957-1959 and they sound more true to life than almost any recordings made today. With instruments which exhibit significant sustain (like a psaltery) you may want to place a mike inside the body of the instrument. If so, leave the sides open. Otherwise the transients will blow the top out of your mix and force you to record at such a low level as to produce junk. Mixing the output with the sound picked up from 2 nearby mikes can help to capture the live reverberant sound better than a pure coincident XY or binaural pair. It's important to realize that microphones are acoustic cyclopses: they see *only and perfectly* that tiny bit of the soundfield at which they're aimed. To get a recording that sounds like what your ears hear at a live performance, you'll often have to jump through hoops and use bizarre and irrational mike placements and mixes. One of the greatest fallacies in dealing with microphones is the old canard: "If your ears can hear it, the mike will pick it up." This is NEVER true. Sounds which are clearly audible to your ears during a live performance will invariably be ignored by the mikes; sounds which your ears cannot hear during a live performance will be magnified unbearably by the mikes. Small changes in mike placement can produce extreme changes in the recording. Above all, trust your ears by listening to the test recordings on your DAT: if the test recording doesn't sound like the live performance, change things--no matter how conceptually perfect your mike placement or mix levels. [6] All live recordings have hiss. Digital microphones, digital mixers, digital synths, digital recorders-- doesn't matter. There will STILL be hiss. In my experience the biggest source of frying bacon is the effects unit. Even if it's a 24-bit reverb, it will hiss like crazy. The only solution to this is to put a hiss eliminator on the effects unit BEFORE the stereo returns come back into the mixer. There are a lot of models of hiss eliminator on the market. They all work on the same principle: a level-sensing circuit activates a voltage-controlled filter which closes down depending on the sensitivity setting. The Hush IIcx is fairly cheap and works well; a used KLH Burwen Dynamic Noise Reduction unit will also work well; dbx makes (or used to make) a single-ended noise reduction unit that worked well; and other companies make similar widgets. They're all in the range of $150-$200. You'll need at least 2 of these for any recording session. 1 to kill the hiss from the effects unit(s) coming into the mixer, another to kill the hiss after it leaves the mixes and goes into your DAT or ADAT. [7] Fripp's rule of thumb that the tape always runs out during the best part of the performance can be easily end-run. I always record with 2 digital recorders-- one recording live from mikes, another recording the straight mix output from the mixer. As Warren Burt can testify, this approach works--what one recording misses, the other gets. Starting one machine later than the other assures that you'll never be burned by lack of tape. [7] "Pro" engineers will place great stock in balanced vs. unbalanced lines. In my experience balanced lines are useful mainly in running audio cables from the mixer to the DAT. As long as your other cable runs are short & you're not recording next to a microwave relay tower or a commercial radio tower, there's no practical advantage in using balanced lines. (Unless the stage is a long ways away from the mixing console! A cable run of more than 15 or 20 feets demands a balanced line, no question.) "Pro" engineers will claim that balanced lines eliminate ground loops. Naturally, this isn't true. Balanced lines are just as prone to ground loops as unbalanced lines when the digital grounds of digital synths are involved. The sad fact is that you'll still need isolation transformers even if you use balanced lines on all of your equipment. One further point: if at any point in the audio chain you introduce an unbalanced line, the whole audio chain might as well be unbalanced. A single unbalanced line can (& usually will) introduce a ground loop. Bear in mind that the unbalanced part of the audio loop can be inside a wall socket with 3 holes but no true ground! So much for the myth that "balanced lines eliminate ground loops." [8] You should bring along your own speakers, your own preamp, your own surge suppressor, your own power strips and your own line conditioner/backup UPS to any public recording session. Last year I had the delightful opportunity to handle audio on a live performance in which the "pro" sound engineer in charge thought it would be a real smart idea to let 20 people plug coffee machines and hot plates and toaster ovens into the same outlet the performers were using to power their digital synths. The result was interesting (as in the Chinese curse). [9] Always bring 4 of everything--4 3-prong-to-2-prong plugs (because the place where you perform won't have 3-prong plugs), 4 1/4-inch phone jack-to-RCA cables, 4 XLR-to-phone-jack transformers, 4 MIDI mergers, 4 mini-plug- to-headphone adapters, 4 durable equipment bags, 4 everything. 2 are never enough and someone always needs one extra. [10] All these pieces of advice deal with digital recordings rather than analog. Alas, there's just no comparison twixt DAT and analog reel recordings. The reel recordings don't measure up. Despite the frenzied claims of the LP and reel-to-reel fanatics, DAT offers infinitely superior sound quality to any possible reel recording. --mclaren Received: from sun4nl.NL.net [193.78.240.1] by vbv40.ezh.nl with SMTP-OpenVMS via TCP/IP; Thu, 26 Oct 1995 09:25 +0100 Received: from eartha.mills.edu by sun4nl.NL.net with SMTP id AA06609 (5.65b/CWI-3.3); Thu, 26 Oct 1995 06:30:36 +0100 Received: from by eartha.mills.edu via SMTP (940816.SGI.8.6.9/930416.SGI) for id WAA09429; Wed, 25 Oct 1995 22:29:20 -0700 Date: Wed, 25 Oct 1995 22:29:20 -0700 Message-Id: <199510260528.WAA03929@hopf.dnai.com> Errors-To: madole@ella.mills.edu Reply-To: tuning@eartha.mills.edu Originator: tuning@eartha.mills.edu Sender: tuning@eartha.mills.edu