80 Years On: Getting it Right for Speech Reinforcement

April 27 marked the 80th anniversary of a historic milestone in the history of audio. On this date in 1933, the Philadelphia Orchestra under deputy conductor Alexander Smallens was picked up by three microphones at the Academy of Music in Philadelphia—left, center, and right of the orchestra stage—and the audio transmitted over wire lines to Constitution Hall in Washington, where it was replayed over three loudspeakers placed in similar positions to an audience of invited guests. Music director Leopold Stokowski manipulated the audio controls at the receiving end in Washington.

This historic event was reported and analyzed by audio pioneers Harvey Fletcher, J.C. Steinberg and W.B. Snow, E.C. Wente and A.L. Thuras, and others, in a collection of six papers published in January 1934 as the Symposium on Auditory Perspective by the IEEE, in Electrical Engineering. Paul Klipsch referred to the Symposium as "one of the most important papers in the field of audio."

April 27, 1933: Leopold Stokowski at the controls with Harvey Fletcher observing

 

Prior to 1933, Fletcher had been working on what has since been termed the wall of sound. “Theoretically, there should be an infinite number of such ideal sets of microphones and sound projectors [i.e., loudspeakers] and each one should be infinitesimally small,” he wrote.

Fletcher’s dual curtains of microphones and loudspeakers

 

Fletcher continued, “Practically, however, when the audience is at a considerable distance from the orchestra, as usually is the case, only a few of these sets are needed to give good auditory perspective; that is, to give depth and a sense of extensiveness to the source of the music.”

In this regard, Floyd Toole’s conclusions—following a career spent researching loudspeakers and listening rooms—are especially noteworthy. In his 2008 magnum opus, Sound Reproduction: Loudspeakers and Rooms, Toole noted that the “feeling of space”—apparent source width plus listener envelopment—which turns up in the research as the largest single factor in listener perceptions of “naturalness” and “pleasantness,” two general measures of quality, is increased by the use of surround loudspeakers in typical listening rooms and home theatres.

Given that these smaller spaces cannot be compared in either size or purpose to concert halls where sound is originally produced, Toole noted that in the 1933 experiment, “there was no need to capture ambient sounds, as the playback hall had its own reverberation."

Localization Errors

Recognizing that systems of as few as two and three channels were “far less ideal arrangements,” Steinberg and Snow observed that, nevertheless, “the 3-channel system was found to have an important advantage over the 2-channel system in that the shift of the virtual position for side observing positions was smaller."

In other words, for listeners away from the sweet spot along the hall’s center axis, localization errors due to shifts in the phantom images between loudspeakers were smaller in the case of a Left-Center-Right system compared with a Left-Right system.

Significantly, Fletcher did not include localization along with “depth and a sense of extensiveness” among the characteristics of "good auditory perspective.”

Regarding localization, Steinberg and Snow realized that “point-for-point correlation between pick-up stage and virtual stage positions is not obtained for 2-and 3-channel systems.” Further, they concluded that the listener “is not particularly critical of the exact apparent positions of the sounds so long as he receives a spatial impression. Consequently 2-channel reproduction of orchestral music gives good satisfaction, and the difference between it and 3-channel reproduction for music probably is less than for speech reproduction or the reproduction of sounds from moving sources.”

The 1933 experiment was intended to investigate “new possibilities for the reproduction and transmission of music,” in Fletcher’s words. Many, if not most, of the developments in multichannel sound have been motivated and financed by the film industry in the wake of Hollywood's massive financial investment in the "talkies" that single-handedly sounded the death knell of Vaudeville, and led to the conversion of a great many theatres into cinemas.

Given that the growth of the audio industry stemmed from research and development into the reproduction and transmission of sound for the burgeoning telephone, film, radio, television, and recorded music industries, it is curious that the term “theatre” continued (and still continues to this day) to be applied to the buildings and facilities of both cinemas and theatres. This reflects the confusion not only in their architecture, on which the noted theatre consultant Richard Pilbrow commented in his wonderful 2011 memoir A Theatre Project, but also in the development of their respective audio systems.

Theatre is Not Cinema: The Differing Requirements of Speech Reinforcement

Sound reinforcement was an early offshoot, eagerly adopted by demagogues and traveling salesmen alike to bend crowds to their way of thinking; yet, as Don Davis noted in 2013 in Sound System Engineering, “Even today, the most difficult systems to design, build, and operate are those used in the reinforcement of live speech. Systems that are notoriously poor at speech reinforcement often pass reinforcing music with flying colors. Mega churches find that the music reproduction and reinforcement systems are often best separated into two systems.”

The difference lies partly in the relatively low channel count of audio reproduction systems that makes localization of talkers next to impossible. Since delayed loudspeakers were widely introduced into the live sound industry in the 1970’s, they have been used almost exclusively to reinforce the main house sound system, not the performers themselves. This undoubtedly arose from the sheer magnitude of the sound pressure levels involved in the stadium rock concerts and outdoor festivals of the era.

However, in the case of, say, an opera singer, the depth, sense of extensiveness, and spatial impression that lent appeal to the reproduced sound of the symphony orchestra back in 1933, likely won’t prove satisfying in the absence of the ability to localize the sound image of the singer’s voice accurately. Perhaps this is one reason why “amplification” has become such a dirty word among opera aficionados.

In the 1980s, however, the English theatre sound designer Rick Clarke and others began to explore techniques of making sound appear to emanate from the lips of performers rather than from loudspeaker boxes. They were among a handful of pioneers who used the psychoacoustics of delay and the Haas effect “to pull the sound image into the heart of the action,” as sound designer David Collison recounted in his 2008 volume, The Sound of Theatre.

Out Board Electronics in the UK has since taken up the cause of speech sound reinforcement, with a unique delay-based input-output matrix in its TiMax2 Soundhub that enables each performer’s radio mic to be fed to dozens of loudspeakers—if necessary—arrayed throughout the house, with unique levels and delays to each loudspeaker such that more than 90 per cent of the audience is able to localize the voice back to the performer via Haas effect-based perceptual precedence, no matter where they are seated. Out Board refers to this approach as source-oriented reinforcement (SOR).

The delay matrix approach to SOR originated in the former DDR (East Germany), where in the 1970s, Gerhard Steinke, Peter Fels and Wolfgang Ahnert introduced the concept of Delta-Stereophony in an attempt to increase loudness in large auditoriums without compromising directional cues emanating from the stage. In the 1980s, Delta-Stereophony was licensed to AKG and embodied in the DSP 610 processor. While it offered only six inputs and 10 outputs, it came at the price of a small house.

Out Board started working on the concept in the early 1990s and released TiMax (now known as TiMax Classic) around the middle of the decade, progressively developing and enlarging the system up to the 64 x 64 input-output matrix, with 4,096 cross points, that characterizes the current generation, TiMax2.

The TiMax Tracker, an ingenious radar-based location system, locates performers to within six inches in any direction, so that the system can interpolate softly between pre-established location image definitions in the Soundhub for up to 24 performers simultaneously. The audience is thereby enabled to localize performers’ voices accurately as they move around the stage, or up and down on risers, thus addressing the deficiency of conventional systems regarding the localization of both speech and moving sound sources.

Source-Oriented Reinforcement

Out Board director Dave Haydon put it this way: “First thing to know about source-oriented reinforcement is that it’s not panning. Audio localization created using SOR makes the amplified sound actually appear to come from where the performers are on stage. With panning, the sound usually appears to come from the speakers, but biased to relate roughly to a performer’s position on stage. Most of us are also aware that level panning only really works for people sitting near the center line of the audience. In general, anybody sitting much off this center line will mostly perceive the sound to come from whichever stereo speaker channel they’re nearest to.

“This happens because our ear-brain combo localizes to the sound we hear first, not necessarily the loudest. We are all programmed to do this as part of our primitive survival mechanisms, and we all do it within similar parameters. We will localize even to a 1 ms early arrival, all the way up to about 25 ms, then our brain stops integrating the two arrivals and separates them out into an echo. Between 1 ms and about 10 ms arrival time differences, there will be varying coloration caused by phasing artifacts.

“This localization effect, called precedence or Haas Effect after the scientist who discovered it, works within a 6-8 dB level window. This means the first arrival can be up to 6-8 dB quieter than the second arrival and we’ll still localize to it. This is handy as it means we can actively apply this localization effect and at the same time achieve useful amplification.

“If we don’t control these different arrivals they will control us. All the various natural delay offsets between the loudspeakers, performers and the different seat positions cause widely different panoramic perceptions across the audience. You only to have to move 13 inches to create a differential delay of 1 ms, causing significant image shift. Pan pots just controlling level can't fix this for more than a few audience members near the center. You need to manage delays, and ideally control them differentially between every mic and every speaker, which requires a delay-matrix and a little cunning, coupled with a fairly simple understanding of the relevant physics and biology,” Haydon said.

Into the Mainstream

More and more theatres are adopting this approach, including New York’s City Center and the UK’s Royal Shakespeare Company. A number of Raymond Gubbay productions of opera-in-the-round at the notoriously difficult Royal Albert Hall—including Aida, Tosca, The King and I, La Bohème and Madam Butterfly—as well as Carmen at the O2 Arena, have benefited from source oriented reinforcement, as have recent productions of Les Miserables, Jesus Christ Superstar, Into the Woods, Beggar’s Opera, Marie Antoinette, Andromache, Tanz de Vampire, Lord of the Flies, Fela!, and many others at venues around the world.

Veteran West End sound designer Gareth Fry employed the technique earlier this year at the Barbican Theatre for The Master and Margarita, to make it possible for all audience members to continuously localize to the actors’ voices as they moved around the Barbican’s very wide stage. He noted that, in the three-hour show with a number of parallel story threads, this helped greatly with intelligibility to ensure the audience’s total immersion in the show’s complex plot lines.

Based on the experience, Fry said, “I’m quite sure that in the coming years, SOR will be the most common way to do vocal reinforcement in drama.”

As we mark the 80th anniversary of that historic first live stereo transmission, it’s worth noting that, in spite of the proliferation of surround formats for sound reproduction that has to date culminated in the cinematic marvel of 64-channel Dolby Atmos, we are only now getting onto the right track with regard to speech reinforcement.

It’s about time.

(photo source: http://www.stokowski.org) 

The Passing of Online

The essential distinction between offline and online is that an offline process is one of construction; an online process, one of execution. In media production, online usually follows offline, as in the case of video editing, where a product that has been laboriously constructed in an offline edit suite—perhaps over the course of days or weeks—is executed by machinery following an edit decision list (EDL) in minutes or hours in an online suite.

Since the hourly rate of a well appointed online suite is typically several orders of magnitude higher than that of a small offline studio—often equipped with not much more than a desktop computer running editing software—the distinction between online and offline has long been etched into the steely heart of many a production manager.

Applying this distinction to the field of music, you might say that playing an instrument is generally an online process, and requires the talent to perform. Constructing a musical performance using MIDI step input, for example, is an offline process, and requires a different skill set.

Before Bing Crosby teamed up with Jack Mullin back in 1947 and seized on the potential for splicing tape offline to construct complete recorded performances, recording musicians had to execute a complete work flawlessly to the end while it was being recorded direct to phonograph disc—an online process. If they made a mistake, they had to go back to the beginning, scrap the disc, and start all over again.

Likewise, dialing a phone on a traditional land line is an online process. If you realize you’ve made a mistake, you have to abort—hang up—and begin again. Dialing a cell phone, on the other hand, is an offline process. You compose the number and, if you make a mistake, you go back a step and delete the wrong input—edit it out—and input the right number. When the entire telephone number has been constructed to your liking, you go online—literally, hit the green online button—and the call is executed by the service provider.

The ability to edit is what distinguishes offline from online processes.

Sound mixing for film used to be mostly an online activity. It was common practice in the early decades of film sound for an entire 10-minute reel to be mixed in a single pass, following one or more rehearsals. With the development of pick-up and record electronics for film dubbers making punching in possible, the two- or three-person re-recording team enjoyed the ability at last to go back and fix a flawed portion of a mix—usually refining their console settings listening to the sound backwards while the dubbers rewound in real time—without causing undue delay and excessive cost to the production.

Mix automation changed all that, from the introduction of console automation systems in the 1970s to today’s digital audio workstations featuring the ability to graph not just volume and mute, but just about every conceivable control parameter. Automation has allowed the offline construction of mixes to become standard operating procedure, with the mix being subsequently executed online in a single record pass or internal bounce-to-disk.

Now this has all changed again with the introduction of offline bounce in ProTools 11. This enables freezing a mix—that is, rendering the final mix up to 150 times faster than real time, according to Avid—and has made the notion of “online” something of a quaint curiosity.

Now a mix need never be onlined at all, since we are able to render into a single final file something that doesn’t ever need to be played through, prior to the playback for quality control checking and approval, after the fact.

The notion of online vs. offline, once so central to the production process and necessitating the development of the all-important EDL, is in the process of being relegated to the status of a quaint curiosity, a byway in the development of modern studio practices and procedures. It will soon be forgotten, along with such other bygone realities as the daily tape recorder alignment ritual, analog noise reduction devices, and uniformed gas station attendants.

It brings to mind the day that I finally sold my once invincible Synclavier and 16-track Direct-to-Disk recorder—to a couple of vintage synth collectors, no less. The only things I hung onto were two blank rack panels and an AC power bar. Some things, at least, are irreplaceable. 

Listening in on the Words and Music Demo Panel at Canadian Music Week

Canadian Music Week wrapped up last Saturday with a special Words and Music Demo Listening Session at the Toronto Marriott Downtown Eaton Centre. I have attended a few Date With A Demo sessions before under the auspices of the Songwriters Association of Canada, but this was by far and away the best yet, for a couple of reasons.

First, the 23 songs auditioned by the panel during the two-hour session were, as a group, of much higher quality than I had seen at any SAC session before; and second, because the panelists themselves, drawn from different sectors of the industry that are all relevant to aspiring songwriters, gave such precise prescriptions for making good songs great.

Moderated by SAC’s Ania Ziemirska, the panel included Juno Award-winning singer-songwriter Melanie Doane; radio promotion and music director Andrea Morris; Juno Award-winning producer Gavin Brown; and internationally acclaimed producer-songwriter Brett Rosenberg. As Brown said, their job was to provide analysis, not criticism. For those who were unable to attend, here’s a distillation of their advice, in no particular order.

General production advice

1. Keep intros short. This was hammered home many times during the session. Listeners will give an unknown song about 40 seconds, at most a minute, before moving on to something else. This is particularly true for radio programmers, who need to be grabbed immediately. No one will get to hear a great bridge if they’re not hooked by the first verse and chorus.
2. A demo produced for other people to sing should sound like a finished hit. Try not to allow the production to sound dated. However, a very simple demo, such as piano and voice, may allow a creative producer to imagine the song as it might be produced for different genres
3. Leave room at the beginning to build up excitement as the song progresses. A song that doesn’t change much from beginning to end will tend to sound boring. Make it quieter and louder, not just loud the whole time.
4. Ensure that the low end isn’t muddy. Roll off the low frequencies in the mix and see if that improves the song.
5. If singing from a first person singular point-of-view, maybe it’s best not to have multiple voices harmonizing on the word “I” when it comes around.
6. Make sure the lyrics are always clear. Don’t bury the vocal in the mix.

Lyric-writing

7. Ensure that the singer’s point-of-view is clear and unambiguous. Be careful not to slip from a first person (“I”) to a second person (“you”) or a third person (“she”) point of view as the lyrics unfold, unless the story demands it.
8. Above all, make sure the message is clear. A song is a vehicle for communicating. If a line isn’t communicating anything or isn’t amazing, it shouldn’t be in the song. The re-writing process is tremendously important. As the panel pointed out, prose writers rewrite constantly and have editors who help them revisit the text many times.
9. Look at the building blocks of the song and ask what emotion is in each part. Make sure the different blocks don’t contradict each other.
10. Avoid clichés. Don’t sing what you wouldn’t say. Extend the lyric to its logical conclusion and make sure you haven’t left anything important unsaid.
11. Always avoid awkward lyrics. If a line sounds weird or stilted when spoken out loud, then consider recasting it for the song.
12. Lyrically, something has to happen more than once, or else you’re writing a poem. If working with an extended metaphor, try to milk every association out of it, and make the whole song relate to that one thing.

Verse and chorus

13. Work on the melody. Then work some more. Don’t just sing over the chords. Try singing different notes of the triads or scale. Make the melody memorable.
14. Work on different melodic elements in the music track and the vocal so that they are different but complementary, rather than parallel and similar. For example, the guitar or piano should not be playing the melody in unison with the voice.
15. The title should be the hook. Make sure the song title is clearly stated, perhaps as the last line of the chorus. If you can’t fit it in naturally, then add a beat or two to let it fit. Or if that doesn’t work and it doesn’t fit in the lead vocal more than once, then try to have it sung in the backing vocals.
16. Don’t take too long to get to the chorus. The lift or pre-chorus should be followed immediately by the chorus without being repeated.
17. The chorus should be set up convincingly—most often it is set up on the fifth or dominant chord. A chorus should be awesome. Make it soaring, triumphant. If a chorus doesn’t sound triumphant, then keep trying. Experiment with big interval jumps. Big intervals are exciting.
18. Differentiate between the chords in the verse and the chorus.
19. Take care not to go to half-time or drop beats in the chorus.
20. Don’t let the drummer play over the payoff or the song’s title line in the chorus.
21. Scream it and mean it.

Writing for radio

22. If you’re going to write songs for radio, make sure the song fits the conventions of radio. Listen to the radio, and figure out what stations and formats you’re targeting—even if not every song you write is intended for radio. If you’re new, you can’t start out by doing your own thing—you need to have already established your identity as an artist to pull that off.
23. If sound effects are absolutely essential to the song, then keep them for the album version and provide a stripped down remix for radio play, especially if the effects are at the beginning of the song.
24. Beware of using sexually-tinged lyrics; even a word as innocuous as “virgin” may limit a song’s potential for radio play.
25. Jump into the lyric right out of the gate and make the intro short. A radio programmers’ music meeting is not likely to listen past the first minute of your song, if that.

The quality of the songs was truly impressive. Due to time constraints, only the first verse and chorus of each song was played, but on several occasions, the panelists expressed a desire to hear more of a song. A few songs even elicited spontaneous applause from the audience: Kat Leonard’s witty, off-the-wall I’m My Own Asshole; David Keeble’s liberating, stripped down demo Maybe Freedom; Steve Onotera’s The Field of White with melodic acoustic guitar accompaniment; and—illustrating a soaring, triumphant chorus—Catherine Bacque’s Stand.

Moderator Ania Ziemirska laboured valiantly through a lingering cold to keep panelists on track and play as many songs as possible, skipping songs if the writer was not present. Noting that some of the other CMW sessions were running late, she graciously returned to songs that were skipped, after the writers were able to join the session.

Speaking with the participants after the wrap-up, I can say that most were deeply appreciative of the depth and originality of the advice offered by the panelists, and the gentle candour with which they analyzed each song. There were no bruised egos in evidence, but more than once I heard a writer say, “That was great—now where do we go from here with our songs?”

From that, it’s clear that most found this version of Date With A Demo to be both motivating and inspiring. Given the other sessions that were on offer at the CMW Songwriters Summit, like How Artists Are Being Discovered and Publishing 101, SAC’s Demo Listening Session provided an excellent springboard for writers to move forward with their songs.

To Mix or Not to Mix? That is the Question

In the Monty Python film, The Meaning of Life, there is an unforgettable scene in an upscale French restaurant featuring this exchange between John Cleese’s fawning waiter and Terry Jones’ more-than-morbidly obese patron, Mr. Creosote:

“Today we have for appetizers moules mariniers, pâté de fois gras, beluga caviar, eggs benedict, tarte de poivre—that’s leek tart—frogs’ legs amandine, or oeufs de cailles—little quails’ eggs on a bed of pureéd mushrooms. It’s very delicate, very subtle.”

“I’ll have the lot,” replies Mr. Creosote.

“A wise choice, monsieur. And now, how would you like it served—all mixed up together in a bucket?”

“Yeah . . . with the eggs on top.”

While the humour in the scene is partly visual, the Pythons’ unique stamp of taking things to the brink of the ridiculous, and then vaulting over it, contrasts the list of the individual, highly refined dishes on the menu—representing the pinnacle of classic French cuisine—and the way these “very delicate, very subtle” elements are offered to the patron in a gross, vulgar, and repulsive manner, “all mixed up together in a bucket.”

Of course, no-one would willingly order a meal this way, much less be served in this fashion by a trained professional. Yet, that is much the way sound is often presented to theatre patrons: all mixed up together, with the eggs—or rather, the voices—on top. Occasionally delicate, not often subtle.

What is at issue here is the very notion of mixing, of combining disparate elements into a single channel (center cluster), two channels (L, R), a combination of these (L, C, R) or perhaps even on very rare occasions, a surround mix of four or five channels.

While mixing a large number of individual audio signals together into a few channels may be a very real requirement for the limited channel count of broadcast radio and television, as well as channel-restricted media such as consumer audio playback systems, this is certainly not the case for theatre and other staged entertainment. Until recently, however, theatrical and similar live events have largely been mixed in much the same way as broadcasts and recorded music.

This may be attributed in part to the large overlap in the designs of traditional recording, broadcast and live consoles; schools teaching audio (i.e., “recording schools”) continue to focus on the art and techniques of the mixdown; even one of the audio industry’s leading magazines proudly heralds the practice in its name, Mix. Originating in broadcast and recording sessions involving multiple microphones, and refined in multitrack recording studios producing mono or stereo masters, mixing has become entrenched in the industry and in the minds of many who dream of working in it, to the point where it’s almost as if no other way of working with sound is even remotely conceivable.

A great many shows are presented as if the audience were listening to a gargantuan stereo system, with massive line arrays hung to the left and right of the stage. Now this might not be inappropriate for a touring band well known from its recordings or for a big, dynamic rock musical where the design calls for a larger-than-life aspect.

Even so, many of the blockbuster musicals from the past quarter century benefited greatly from the creativity of such esteemed sound designers as Olivier Award winner Mick Potter, who, in the quest for more natural sound, have opted for separate vocal and orchestra mixes, striving simultaneously for clarity in the voices and power in the orchestra. Moreover, two voice mixes are sometimes derived, with one going to a duplicate set of loudspeakers in an A-B configuration pioneered in 1988 by Martin Levan for Aspects of Love, to eliminate electrical summing of mic signals and the ensuing phase problems that arise when performers are in close proximity to each other’s microphones.

For other production styles, however, an approach based on mixing may not be the most appropriate technique for conveying the nuances of theatre—including musical theatre, where sound systems have become ubiquitous—if the purpose of sound reinforcement is to allow every performer’s voice to be heard as it would unamplified in an optimum seat.

Leap year and drop-frame time code are conceptually the same

For those in the media production industries, February 29th is a good day to revisit drop-frame SMPTE time code, because both leap year and drop-frame time code came into being for the sole purpose of reconciling two different time bases on which we do things with mundane regularity.

Take the calendar first: our calendar simply charts the sequence of the individual days that comprise a single year. The day is based, of course, on a single rotation of the earth on its axis, whereas the year is based on a single revolution of the earth around the sun. Rotation and revolution are the two different time bases on which our calendar is constructed.

Since it takes about 365.25 days for the earth to revolve around the sun, we collect four of those quarter days and add them together into a single day—February 29—that appears on the calendar once every four years.

We do this because there's no such thing as a quarter-day: you couldn't start a New Year at 6:00 a.m. After all, a day is a day and cannot be partitioned like that. It's an integer.

It's important to see that the concept of the yearly calendar comprises 366 days—February 29 is not imaginary. But rather than adding it every four years, what we are really doing is dropping it from the calendar in every year that is not a multiple of four. If the year is not divisible by 4, then we drop February 29 from our count of days in that year.

It's exactly the same with drop-frame time code, where frames are analogous to days, and hours to years. A video frame is a whole thing, an integer, and we count 30 of them in one second. But the rate at which they proceed is a bit less than 30 per second, more like 29.97 frames per second.

This is the same sort of fractional discrepancy that exists in the annual rate of 365.25 days per year.

We deal with it the same way, by dropping 2 frames from the count at the very beginning of every minute that is not a multiple of 10. In that first second, there are only 28 frames.

So frames 00 and 01 simply do not exist at the beginning of every minute of time code that doesn't have a zero at the end of it (10, 20, 30, 40, 50, and 00 minutes being the exceptions), just as February 29 does not exist in any year that can't be divided by 4. It's as simple as that.

Why go to the bother of doing this? For the calendar, it's long been considered important that the seasons start at roughly the same time every year: if we didn't have February 29 as a corrective, then the beginning of Spring, for example, would progress steadily back through February, January, December, and so on as the years rolled by.

For producers, it's important that the time displayed by your time code reader agrees with the real-time clock on the control room wall. Without drop-frame time code, a one-hour program as measured by your time code would actually run 3 seconds and 18 frames too long, and that would wreak havoc with broadcast schedules.

Note that what we are NOT doing is cutting out frames from our program and leaving them on the cutting room floor, as some of my former students at the Toronto Film School used to believe. Those "dropped" frames are simply never there in the first place, just as February 29 will not "be there" in 2013, 2014, and 2015. The calendar works as "drop-day" code.

The takeaway from this blog entry is that if you can intuitively grasp the concept of leap year, then you've already got the essence of drop-frame time code. Conceptually, they are one and the same.

Seller Beware—When You're Being Shopped for a Price

I got an email last week asking how much I would charge to mix 5 songs for a band's EP. I wrote back asking whether I'd be recording the original tracks, or just mixing tracks that someone else has already recorded for the band. Both, came the reply, and how much would it cost?

I wrote back to ask about the band, number of players, what instruments, etc., so that I could price out the job using the appropriate recording facility, and asking for a couple of possible dates when the band wanted to start recording. This is an important question, because in sales, there's a strong relationship between price, availability and delivery. You can sometimes get a better deal on studio time that would otherwise remain unbooked.

At this point, the answers started to get vague. What was clear, however, was that I was being shopped for a price. In other words, the prospect (not yet a client) had no intention of coming to me for the job, but was only trying to get a handle on the price of a job that most likely he was bidding on himself.

This happens to everyone from time to time, and is one of the reasons why it's not a great idea just to shoot out a price in response to an inquiry. Every job is different in some way, and a big part of the sales process is asking questions to qualify the buyer.

Asking questions not only keeps valuable business intelligence—your pricing policies— out of the hands of your competition, it also saves you from wasting time with tire kickers who would otherwise take up a lot of your time, but never end up buying anything.

The 80-20 rule seems to apply here: 80% of your business comes from 20% of your prospects. This is further refined so that in turn, 80% of your income comes from 20% of them. In other words, 4% (20% of 20%) of your potential clients are responsible for about two-thirds (80% of 80%) of your business.

Asking questions is your best line of defense here, and a genuine prospect will appreciate that you're drilling down in order to provide the best possible service.

MIDI Orchestration

I've just finished a first read-through of the 4th edition of Paul Gilreath's The Guide to MIDI Orchestration, published by Focal Press. Coming in at 600 pages, it's a pretty thorough introduction to the subject of orchestrating a musical composition using MIDI-based equipment and instrument sample libraries.

Several topics of interest to non-MIDI orchestrators and project studio folks alike are covered here, including instrument ranges and playing techniques, notation, voice leading, distribution of melodies and accompaniments to different instruments in the various sections, combining instruments to create different sounds, and achieving specific moods with orchestrations.

For the MIDI producer, there's a wealth of information on equipment, software, choosing sample libraries, and sequencing strings, woodwinds, brass, percussion, piano, harp and voices.

The author illustrates portions of the text with screenshots from his favourite digital audio workstations: Cubase/Nuendo, Logic, Digital Performer and Sonar. Unfortunately, Gilreath dismisses ProTools, saying it's "still working to catch up" in this area, which is too bad, given its market penetration together with the strides that ProTools has made in the whole area of MIDI sequencing and sampling since version 8 was released at the end of 2008.

In any event, readers working with ProTools can easily adapt the material to their way of working, which, in most respects, is not too different from the others.

Supplementary material is available online at www.midi-orchestration.com, including reviews of several good instrument sample libraries, supplementary tutorials, and audio examples, but you have to sign in to access it.

Much of this material—including several full-length, uncut chapters—is still freely available for download as a single zip file from Focal Press at www.focalpress.com/midiorchestrationfiles.aspx.

The book is a terrific reference, and it's refreshing that the author combines description and prescription in almost equal amounts, which is a rare feat. Anyone looking for a grounding in MIDI orchestration would do well to own this book, and will check in with it on a regular basis, if not frequently. It's a beautiful volume, well designed and easy to read, with a crisp and clear layout, and it should be on every music producer's reference shelf.

My criticisms are few. There are numerous errors relating to missing or misplaced illustrations or examples, including a missing reference section with bibliographical apparatus that the author himself refers to twice yet, strangely, is nowhere to be found!

There are also too many proofreading errors, not many of them spelling mistakes, which leads me to believe that spell-check may have served as a convenient substitute for a thorough proofing. This is a tad disappointing in a 600-page book priced at US$82.50 ($86.50 Canadian).

Fortunately, these shortcomings are overshadowed by the author's monumental achievement in turning out what was surely the crowning achievement of his career as a composer for film and television. May his new life as a dentist in Atlanta be as fulfilling!

The Guide to MIDI Orchestration, 4th Edition, by Paul Gilreath, published by Focal Press, 2010. ISBN: 978-0-240-81413-1