Richard Dudas

Cort Lippe

The Phase Vocoder is a tool used to perform time-stretching and pitch-shifting on recorded sounds. Its name derives from the early "vocoders" (contraction from "voice encoders"), which used a set of bandpass filters in parallel over many frequency bands, to crudely analyze and reconstruct speech. The infamous hardware vocoders of the 1960s and 1970s (as used, for example, by Wendy Carlos in the soundtrack of Kubrick's film "A Clockwork Orange") were based on this technology, and allowed the spectrum of one sound (in the Carlos example a synthesizer) to be controlled by that of another (a voice). In the Max/MSP examples folder, there is an example by Marcel Wierckx called "classic_vocoder.pat" (located in the "effects" sub-folder) which shows how this traditional vocoder works. Unlike the classic vocoder, which is based on bandpass filters, the phase vocoder is based on a Short-Term Fourier Transform (STFT) – a Fourier Transform performed sequentially on short segments of a longer sound – and in practical use has little to do with the hardware vocoders of the 1960s and 1970s. The phase vocoder can, however, be considered a type of vocoder because the Fourier Transform returns a set of amplitude values for a set of frequency bands spaced evenly across the sound's spectrum, similar to the older vocoder's set of bandpass filters. Of course the phase vocoder, as its name suggests, not only takes into account the amplitude of these frequency bands, but also the phase of each band.

If you are not familiar with the Fast Fourier Transform (FFT) as it is used in MSP, we suggest you review MSP Tutorials 25 and 26 , which deal with the fft~/ifft~ objects and the pfft~ object set, respectively.

In MSP Tutorial 26  on the pfft~ object, we are shown a simple phase vocoder patch which analyzes an incoming sound and records the FFT frames into a buffer~ object. The data in the buffer~ object is then reconstructed so that a basic sort of time-stretching (and compression) may be performed on the recorded data. Although it works as advertised and introduces the basic concepts, it lacks some of the flexibility of a more standardly-designed phase vocoder, which lets us transpose our original sound, as well as start at any point in the original sound.

Let's review the STFT and basic phase vocoder design. A Short Term Fourier Transform (STFT) is a series of Fourier Transforms, usually spaced evenly in time:

Whereas the old fft~ and ifft~ objects just perform the FFT, the pfft~ object actually goes one step further and performs an STFT, as shown above. The input signal is sliced into frames, which start at a regular offset known as the hop size. The hop size determines the overlap, which can be defined as the number of frames superimposed on each other at any given time. If the hop size is half the FFT frame size, the overlap is 2, and if it is one-fourth the frame size, the overlap is 4. Theoretically, an STFT needs a hop size of at least 4 and although a hop size of 2 can be made to work for many musical purposes, the phase vocoder will sound better with an overlap of 4.

If our FFT size is 1024, our FFT will give us 1024 frequency bins (these would be called bands in filter terminology) from DC (0Hz) to the sampling frequency. Because the data above the Nyquist Frequency (half the sampling rate) is a mirrored copy of the data below the Nyquist Frequency, the pfft~ object by default eliminates this redundant data for efficiency's sake, and reconstructs it for us before performing the inverse FFT. (In our phase vocoder patch we will need to tell the pfft~ object to override this elimination of data for reasons that will become clear later in this article).

For each bin, the FFT provides us with coordinates which can be converted to amplitude and phase values. (This is covered in MSP Tutorials 25 and 26). The difference between the phase values of successive FFT frames for a given bin determines the exact frequency of the energy centered in that bin. This is often known as the phase difference (and sometimes also referred to as phase derivative or instantaneous frequency if it's been subjected to a few additional calculations).

If we record a succession of FFT frames, then play them out of order, the differences between the phase values in the bins will no longer produce the correct frequency content for each bin. Therefore, in order to "reconstruct" a plausible playback of re-ordered FFT frames, we need to calculate the phase difference between successive frames and use it to construct a "running phase" (by simply summing the successive differences) for the output FFT frames.

The first thing we need to do is to be able to is to access our sound file directly in our pfft~ subpatch and perform an FFT directly on the sound, without sending it into any fftin~ inlets of the pfft~. Why do we need to do it this way? Since the whole point of a phase vocoder is to perform time stretching and compression on an existing sound, we need to be able to access that sound directly via a buffer~ object and perform the FFT at any given playback speed and overlap. In order to have independent transposition and playback speed, a phase vocoder needs independent playback of the sound to be transformed for each FFT overlap (4, in our case). Each playback frame needs to be synchronized with its respective FFT frame. Therefore we cannot send a single copy of the sound we wish to transform into the pfft~, but need to play a slice of the sound into each of the four overlap frames. Since we cannot send one slice of the sound into the pfft~ object (it keeps track of its input history and considers its input to be continuous sound, not individual slices of sound), we cannot use the fftin~ object inside the pfft~, but must do our FFT processing using the fft~ object. The fft~ object performs a full-spectrum FFT (i.e. mirrored), so we consequently need to make the pfft~ object behave the same way, so the fft~ can work in sync with the FFT frames processed inside the pfft~ object. We need to make a few changes to the default way that pfft~ deals with the STFT.

We first need to tell the pfft~ object to process full-spectrum FFT frames, instead of the default "spectral frame" which is half the FFT size (up to half the Nyquist). This is easily accomplished by adding a non-zero fifth argument to the pfft~ object. Because the full-spectrum argument is the fifth argument, we must supply all the other arguments before it, including the fourth argument, the start onset, which will be set to the default value of zero.

Next, because the fftin~ and fftout~ objects perform the FFT calculation at zero phase with respect to the FFT (the first sample in the windowed frame sent to the FFT is the middle of the window), and the traditional fft~ and ifft~ objects perform the FFT 180 degrees out of phase, we need to make sure any fftin~ and fftout~ objects in our patch have the same FFT phase offset used in the fft~ objects. We do this by specifying a phase offset to the fftin~ and fftout~ objects. A phase value of 0.5 means 180 degrees out of phase, so this is the value we want. While we do not need the fftin~ object in our pfft~, we can still make use of the convenience of the fftout~ object in order to get the properly windowed and overlap-added result out of our pfft~. The automatic windowing in the fftout~ object should behave like our manual windowing with the fft~ objects.

Now we are ready to start constructing our phase vocoder sub-patch.

We always need to access our buffer~ in two places – at the current FFT frame location, and at the location of the previous FFT frame of the buffer. We can use the index~ object to access the buffer~, just as we might use the index~ object in a simple playback patch. (Note that we are accumulating samples at the "normal" playback rate for now.) And because we're manually designing the input part of our STFT ourselves, using the fft~ object, we need to window the signal we read from the buffer~ before sending it to the fft~ objects. The patch uses a Hanning window (the default window used by pfft~).

Using two fft~ objects that are a quarter frame apart from each other, we calculate two FFT frames (the present frame and the previous). Using cartopol~, we convert from cartesian coordinates to polar coordinates to get amplitudes and phases, and then simply subtract the phases of the previous frame from the present frame to get phase differences for each bin. Just as in the simplified phase vocoder in MSP Tutorial 26 , we use the frameaccum~ object to accumulate the phase differences to construct the "running phase" for output.

Additionally, at the top of the patch we no longer have a fixed playback rate (it was set to 0.25 in the previous example image), but have added the capability to time-stretch (or compress) the sound by accepting a (variable) sample increment from the main patch.

Notice that now the "previous" frame that we read from the buffer~ might not actually be the frame that we previously read as the "current" frame! This is the whole point of the phase vocoder – we are able to read frames in any location in the buffer and at any speed, and by simultaneously reading the frame one hop-size before the "current" frame (regardless of the speed at which we're reading the buffer~) we can obtain the "correct" phase difference for the "current" FFT frame!

The FFT algorithm outputs transformed data in cartesian (x, y) coordinates. These coordinates are often referred to as the real and imaginary parts of a complex number. The amplitude and phase values that we normally associate with the FFT are the polar coordinates of the (x,y) values. The polar coordinates of cartopol~ conveniently give us amplitude and phase information. While working in polar coordinates is convenient, Cartesian to polar conversion of phase information uses the trigonometric math function arctan~, which is computationally very expensive. Avoiding the arctan calculation, which must be calculated for each of the eight FFTs used in a single phase vocoder, by using less intuitive Cartesian math means working with complex math (the complex multiply and divide instead of the simple addition, subtraction, multiplication, and division needed in the polar domain). In addition to the issue of computational efficiency, for reasons of accuracy, converting to and from polar coordinates can introduce small amounts of error which slowly accumulate over time and probably should be avoided. Finally, there are some additional features that improve the quality of the phase vocoder which we will see in Part II of this article (scheduled in two months' time), for which it is preferable to use complex math on cartesian coordinates instead of calculating on the polar coordinates derived from them. So we need to learn a little complex math and how it relates to the polar calculations we're performing on the amplitude and phase values.

A complex multiply multiplies the amplitudes and adds the phases of two complex signals (i.e. signals which have a real and imaginary part – such as the two outputs of the fft~ or fftin~ objects).

A complex divide divides the amplitudes and subtracts the phases.

What is important about these complex multiplication and division sub-patches is what it does to the phases of our two complex signals. Complex multiplication adds phases, whereas complex division subtracts phases. Therefore we can calculate both the phase difference as well as accumulate the running phase of our output signal using these complex math operators. Because we only care about the phase in our phase vocoder patch (remember in the polar version shown previously we did not modify the amplitudes), we can make a further optimization to the complex division and remove the denominator by which both real and imaginary parts are scaled:

Now we're ready to use these to construct a phase vocoder which uses complex multiplication and division. The first part of the patch will remain the same – we are only changing what happens between the fft~ objects that read the buffer~, and the fftout~ at the bottom of the patch.

Notice how we have to use the send~ and receive~ objects to manually feed-back the previous output frame so we can use it to accumulate the running phase. Remember that a receive~ object "receives" signal data from its corresponding send~ object with a delay of one signal vector in order to avoid signal loops. The signal vector of a pfft~ is the size of the FFT frame, and since we are running a full-spectrum pfft~, the delay is 1024 samples, or one FFT frame.

Also notice how we invert the amplitude values so we can produce a frame which contains JUST the phases, with the amplitudes set to 1.0. (The inverted amplitude, or magnitude, multiplied by the real and imaginary values essentially cancels out the amplitude.) This is so we can use our complex divide and multiply and not affect the amplitude values of the "current" frame. In order to make sure we don't divide by zero, we add a very small offset to the complex number before calculating the inverse amplitude. This does not affect the sound perceptibly.

At this point you might want to compare both the polar and cartesian versions of the phase vocoder patch. The polar version which we first showed you is conceptually clearer at first glance. However, note the difference in CPU usage of the two patches. You may well decide that the extra effort in designing the phase vocoder using complex arithmetic is worth the payoff!

In listening to the phase vocoder patch, you may have noticed a somewhat metallic or slightly reverberated quality to the phase vocoder's sound as it runs. This is caused by phase inconsistencies from bin to bin. Since any given FFT bin might be affected by more than one component of a sound—for instance, a non-harmonic sound with two very close inharmonically related frequencies will struggle for precedence in a bin that they share—this can create a slightly reverberated quality to the sound. In Part II of the article we will learn some tricks to minimize this effect, as well as look at a buffer~ reader that allows for flexible and independent transposition and stretching at any location in the buffer~.

To sum up what we've just covered, the basic structure of the phase vocoder requires four overlapping pairs of buffer~ reads and four overlapping pairs of FFTs. The buffer/FFT pairs are exactly one quarter frame apart (one hop). The FFT pairs allow us to calculate the phase differences between where we are in a sound and where we were a quarter frame ago (one hop). Then, for each of the four pairs, we simply add their respective phase difference to the previous phase from a full frame before, accumulating our running phase. We do all this with Cartesian coordinates (real and imaginary values making use of complex arithmetic) using fft~ objects inside a pfft~ that is running in full-spectrum mode with a 180-degree phase offset for the fftout~ object so that it runs in phase with the fft~ objects.

The authors would like to thank Miller Puckette for developing what was more than likely the first real-time phase vocoder, and most certainly the first real-time phase vocoder using Cartesian coordinates!

The phase vocoder is a tool used to perform time-stretching and pitch-shifting on recorded sounds. Its name derives from the early "vocoders" (contraction from "voice encoders"), which used a set of bandpass filters in parallel over many frequency bands, to crudely analyze and reconstruct speech. In this article, Richard Dudas and Cort Lippe explain the workings of a phase-vocoder as well as how to construct and modify one in Max/MSP.

the-phase-vocoder-–-part-i

The Phase Vocoder – Part I

In our Part 1 of this phase vocoder tutorial, we saw how to create a basic phase vocoder for time-stretching. While it is by no means a simple MSP patch, it is a useful one. In addition to time-stretching, the phase vocoder has been used for transposition and "freeze" effects, which we will be discussing in this article. If you are unfamiliar with the phase vocoder principle, we suggest you review Part I of this series of articles. Additionally, if you are unfamiliar with Fast Fourier Transform (FFT) you may wish to familiarize yourself with MSP Tutorial 25 (about FFT and the fft~ object) and MSP Tutorial 26 (about the pfft~ object).

In Part 1 of this tutorial, we designed two phase vocoder patches - one which works with polar coordinates (amplitude and phase values), and one which works with cartesian (x, y) coordinates. While the former is easier to understand (and simpler to patch together), the latter is more efficient, since it avoids using trigonometric math functions (specifically the arctan function), which are computationally expensive. We will take our existing phase vocoder patch as a starting point, and show our modifications to both the polar and cartesian versions.

For many years in the first few decades of digital synthesis, the most convincing method of transposing a sound without changing its duration was to use a phase vocoder. In fact, using a phase vocoder you can change both the transposition and speed independently - so, for example, you could transpose a sound and octave higher while playing it back twice as slow!! Performing a transposition with the phase vocoder involves only a few changes to the buffer~-reading part of the patch.

The first change is the addition of a 3rd inlet to our pfft~ subpatch so we can control the transposition.  As with the time stretch inlet, we also use a sample and hold (sah~) object to make sure the transposition value is held constant for all bins in our FFT. Since transposition involves reading a larger or smaller chunk of sound from our buffer~, we scale the output of the counter~ by the transposition factor before we add it to our sample offset into the buffer~.  If our transposition factor is greater than one, we will be reading a larger window from the buffer~ albeit at a faster speed. Conversely, if the transposition factor is smaller than one, we will be reading a smaller chunk of the buffer~ at a slower speed.

One other change we need to make is to replace index~ with play~.  Since the index~ object does not interpolate sample values, we will degrade the quality of the sound if we use it.  The play~ object uses 4-point interpolation to read "fractional samples" from the buffer~, so its output will sound better when we read the buffer~ at faster or slower speeds.  Since the play~ object takes millisecond values instead of sample values as input, we need to add a sampstoms~ object to convert the samples to milliseconds and read the proper size chunk of sound from our buffer~.

Putting it all together, we can use this transposition for both the polar and cartesian patches, since these changes do not affect the actual phase vocoder part of the patch.

One other important change we are making to the cartesian patch is to use a unique number for the send~ and receive~ name. We do this by beginning the name with a #0 which will be replaced with a different number in each instance (This is described here). It lets us have multiple phase vocoder patches open at once without the send/receive names interfering with one another.

Note that for both patches we make use of the "args" possibility for pfft~. Following the 5th argument to the pfft~, note the word "args" and the FFT size.  This is convenient for allowing you to change the FFT sizes for the fft~ and ifft~ objects in the pfft~ subpatch. (If you do change the FFT size, make sure to change the size of the windowing function in the message box below the loadbang, and double-click the loadbang to recalculate the window function at the new size.) Also, you might want to refer to the "Time vs. Frequency Resolution" technical detail in MSP Tutorial 26, since different sounds might work better with different FFT sizes.

Although we can set the playback speed to zero, and thus "freeze" the sound at a certain point in time, the effect is rather static and mechanical. We can enliven our freeze effect by adding two things to our patch -  some random variance in the playback location, and some additional small random variance in the phase. Together they produce a much better freeze effect than can be achieved without them.

First, we need to only activate our freeze parameters when the playback speed is set to zero.  Since our first inlet to the pfft~ subpatch is the user-defined playback speed, we can simply check this value in order to activate our additional freeze parameters.

Next,  we can use the rand~ object to randomly oscillate the buffer~ read location around the given playback location. We can control both the oscillation speed (with a frequency to rand~) and the oscillation depth - which is our random playback location variance (with a signal multiply). Using just this technique automatically enlivens the frozen sound.

Finally, we can add a very small amount of random phase deviation to the bins of our spectrum. Generally this is something we try to avoid in a phase vocoder, because it adds strange audio artifacts to our sound; however, in the case of a freeze, a very small amount of phase deviation from bin to bin actually breaks the mechanical sound of the freeze!

In our polar coordinate phase vocoder adding the phase is straightforward - we simply add low volume white noise to our phase component.


However in the cartesian coordinate version of the patch things become slightly more complicated. We have to create a complex signal whose phase component has the noise in it, and perform a complex multiplication to rotate the phases. (Phase rotation via complex multiplication and division is explained in Part 1 of this phase vocoder tutorial) To achieve this we use two cycle~ objects whose phase is 90 degrees apart to represent the sine and cosine components of the complex signal, and control their phase input directly with the white noise. We're using our complex multiply subpatch, used elsewhere in the phase vocoder, to add the phase deviation to our complex signal.

Our cartesian phase vocoder now looks like this:

It is quite a bit more complicated than the equivalent polar version shown in Figure 7, but you will notice that, as with the simple cartesian version shown in Part 1 of the phase vocoder tutorial, it is markedly more efficient. For complete views of these patches, we suggest opening up the patches themselves in Max/MSP and trying them out!

With these additions to the phase vocoder we can control a sound's playback speed and transposition independently of each other, as well as "freeze" the sound with a bit of added liveliness. We have also improved the patch so we can provide arguments to the pfft~ subpatcher in order to change the FFT size of the fft~ and ifft~ objects that we must use to read the sound from the buffer~. 

In our last article about the phase vocoder we saw how to create a basic phase vocoder for time-stretching. While it is by no means a simple MSP patch, it is a useful one. In addition to time-stretching, the phase vocoder has been used for transposition and "freeze" effects, which we will be discussing in this article.

the-phase-vocoder-part-ii

The Phase Vocoder - Part II

Ben Nevile

Amnon Wolman creates music with technology and for instruments with two distinctive features: first, a subtle and complex relationship with popular culture and second, a concern for the dramatic nature of sonic evolution. His first release on the c74 record label is "Dangerous Bend." In this conversation with Ben Nevile, Wolman discusses his romantic compositional techniques and reveals the unusual nature of his introduction to computer-based music.

Can we start with a bit of your history? Where do you get your ideas? What are your earliest musical memories?

I was born and raised in Israel. In 1968 I spent a year in Los Angeles which connected me to American popular culture in a way that was not present in my life before. I played classical instruments throughout my early years. I served in the Israeli army for three years - compulsory. Not a pleasant experience but not very traumatic either. I first acknowledged to myself that I was gay when I was 17 and to the world when I was 25. That was not a simple or easy route.

My first musical memories are of family singing. We had Friday dinners at home with the whole family and we would sing together. It was a highly physical activity. I felt the music in my body and mouth and ears more than hearing it as an outsider and an audience member. The first major classical piece that I was involved with as a youngster singing in a children's choir was Bach St. Matthew's Passion when I was 12. It made a huge impression on me and is probably the single moment when I realized that I needed to be a musician. It was like a drug, I felt elated.

So how did you become involved with computers?

In 1981 when I was completing my masters in composition at Tel Aviv University, the Lebanon war started in Israel. I wanted a way out, so I applied for a student exchange scholarship from the government of the Netherlands. I received the grant to do something in music that I could not do in Israel. I heard that there was a small group at the University of Utrecht - Institute for Sonology - that was working on computer music. At the time I was this strange combination of a politically minded, conceptual, somewhat minimal romantic composer... and I suspect I still am. I guess some conceptual music is highly romantic, and to be political implies some romanticism and naiveté too...

Anyway, I decided to go to Utrecht because I thought that using the computer to create music was a terrible idea, that it was cynical and mechanical. As often happened to me at that age I had very strong opinions about things I knew nothing about. I decided that I needed to know more about it in order to talk intelligently against it. But I was bitten by the bug, and the idea of creating sounds on my own has been very exciting and the highlight of my day ever since.

Tell me, was there something specific that caused you to change your mind?

Yes, it was working with sounds that changed my mind. I love imagination, and when working on a score for instruments imagining what it would sound like was a wonderful fantasy. But now, since 1982, I can put my hands in the dirt. I can design the exact sounds I want, and just listen to them evolve - not pitch, not rhythm.

After Utrecht I applied for graduate studies at Stanford where I was off and on until the end of 1989. From 1990 I've been teaching at Northwestern University, and from 1994 I also hold a position at Tel Aviv University.

How does your time at the two schools compare?

I feel safer as a teacher in Israel. I never question my language and my cultural references. My English is good, yet still I am not sure which words I may use with which person. When is it appropriate to use a slang word and which one, and what does it say about you when you use it? The same is true for cultural references. I know what it says about me if I say that I like the Beatles, that is safe, but when I say that I like Madonna, Tom Waits and Xenakis I know what it would mean for my students, and maybe my colleagues, but my other friends? People I meet in political activism groups? In Israel I feel that I know exactly what they mean, even though I may be wrong. I have less definition as a person here. For a long while it made my life easier because I could define myself, but now I miss being clear.

I really like the music on your C74 CD.  I find it incredibly soothing.

I can't decide if I like the description "soothing!"

Well, it's soothing to me. Most of the music I listen to is jagged... it's nice to hear something smooth. Is that not how you think of your music?

Interesting. It is not how I think of my music, mostly because of my politics. I guess for too long I've stood on barricades saying that music is an art and as such it presents many things, not just beauty and relaxation. But then I go and create music that is soothing... it feels weird in the context of my self-perception. But there is not a whole lot one can do about this. There is always a distance - huge I suspect - between the ideology and the art one creates. In the end I sit at the computer and listen to what I create and decide if it is what I want to create and ideology is never in the room during those times.

Is there anything that you've composed that comes more from an ideological standpoint, or has your creative process always been quite separate?

I think that the creative process is always an amalgamation of everything, the intellectual and the emotional. But I guess what I am talking about is the rigors of taste. That is, if I have an ideological or conceptual vision for a piece, by the time I work on it the final choices are about "what does this sound like" and "I like this" or "this is too much" or "this is too beautiful, what will the neighbors say." I don't have many voices of this last kind, the "what will my colleagues and friends say," but every once in a while I realize and chuckle when I finish a section and think to myself how so-and-so would like this...

But to make a long answer even longer, I think the final cut is irrational, and I've never finished a piece where I thought, "oh it does exactly what I decided in advance - so it's okay, even though it does not sound so great." I've always gone and fixed it, started all over, or scrapped it. I hate to admit my romantic streak, but the final decision is always left to my senses.

What compositional techniques did you use on this CD, both inside and outside of Max?

Mostly I used processed sounds. I have worked in the past with synthesis, but the last several years have been almost entirely based on processed recordings. I make recordings everywhere, with people and in nature, and then I process them in different ways. From filtering to different versions of granular synthesis using the recordings as the basis for the brain. I use a lot of time distortion, and different algorithms of phase vocoding.

I think that the idea of processing a recording of, say, a family conversation is very powerful. Somehow I feel that the essence of the sound stays. Even if I am the only one who knows where the sounds came from, some of the magic of the personality and the situation are folded into the sounds. Obviously the actual physical properties are inherent to the sound too. I think perhaps the most personal statement I can make about the pieces you called "soothing" is that I don't hear them as such because I listen to the sounds. A slight change, a disappearance of a harmonic, or an emphasis on a group of harmonics makes huge dramas in my listening experience and does not sooth me... but I know I hear sounds differently than other people.

Sure - you've made the sounds, so you understand them in a deeper way than anyone else, and your listening is therefore completely different. What has had the most direct effect on the romantic inclinations of your musical direction?

The most important musical influences were my teacher in Israel Abel Ehrlich, and John Cage- I met him in Stanford in 1985 and we remained friends until his death. Other influences have been friends and colleagues, usually young composers who were compatriots as students or faculty members and then a few of my own students. There are a few prominent names: Chaya Czernowin, Osnat Arbel, Yuval Shaked, Bill Schottstaedt, Joanathan Berger, Richard Karpen, Michael Pisaro, Jay Alan Yim, Kunsu Shim, Gerhard Staebler, Jeff Kowalkowski and Jenny Walshe. Most are not computer music people, and represent either an influence on my conceptual/intellectual side or the romantic side.

Listening to the greats, some of whom I had the luck to study with, has also obviously refined my process but listing them seems a little more removed from my day to day activity. With everyone on that list I can recall numerous conversations that altered some part of my thinking and helped me refine my own sense of music. I have no doubt that my personal relationships changed my sound world drastically. Most importantly there's my partner Eyal Levinson who helped me get back into listening after several years of trying to avoid music...

In the last little while several composers have told me they don't enjoy listening to music any more.  What pushed you away?

For many years it was hard to listen to music "for fun" for me. It was too much work. I enjoyed silence. I still would rather get home to a quiet house. I know for a fact that it is hard for me to work on creating music while listening to music. If I wake up in the morning and listen to the radio it's hard for me to get into composing. This is why I've become a morning person, and compose before anything else. Also I think I became a musician because I like being active at this game, the playing of music. Listening to music is a passive activity, you don't get to do anything, and it's a different activity, a luxury. I know many musicians who are bored in concerts unless they are on stage.

Yes. So how does that relate to something like "the communal techno machine," where there was mass participation... or was that something completely different? 

Hmm... that started because I wanted to create a piece based on the Andy Warhol Diaries. These were not really "diaries" as such. He would call "the factory" every day and tell the person who was keeping the diaries, Patricia Hackett, what he did the day before and how much it cost. This was done because he was audited yearly by the IRS. But little details crept in. For example, and I am paraphrasing off the top of my head, "I had lunch with Jacqueline Onassis, she had a great hat, $10..." etc. It describes an important period of US history between 1975-1985, for me, and it describes it as a pointillistic picture full of insignificant or important details without a hierarchy. This period is when the liberal community's only ideology was "live and let live" and the right wing community developed a very clear ideology and agenda. It is also the period of the last big gay parties and the beginning of AIDS.

I decided to use the texts and realized it would have to be a really long piece - 11 hours. At the time Canton Becker was a student of mine, and he was a rave fanatic. I knew nothing about it and he introduced me to techno music and the rave scene. I suggested to him that in order to create music for the piece he should build the communal groove machine, which would, with the help of others on the internet, allow us to create the music needed for the piece. He built it under my direction. Indeed at the last moment I discovered that we forgot to add a transposing element to it, so most of the piece is in C...

Ha, that's funny.  How long did you say the piece lasted?

How were the people on the internet able to affect the music?  In what form was their input?

The techno machine had a library of rhythm sections, bass lines, chord progressions, and feels - tempo accents, etc. People would respond to questions by choosing or entering new bass lines, or new rhythms and then it would create the song that they imagined but could not hear. The resulting information was interpreted through a MAX patch written with Matt Moeller that allowed us to mix the songs. The final production of the songs was done with the help of other students, most prominently Andy Knight. The piece also involved three actors/singers for which I wrote parts and a series of videos created by H.D. Motyl.

It was a fantastic experience. The communal groove machine was interesting in many ways, but most profoundly, for me, because it gave me a glimpse on how many people compose and have a clear idea of what their music should sound like, even though they are not musicians. You must remember this is the internet before browsers, only text was moving around. As part of the machine we asked people to include a description of the songs as they thought they should come out, hoping to use it as an indication for the mixing and production. People were very forthcoming, very clear and precise. I was very excited: composition was no longer an artistic activity for specialists, it was open for anyone who had ideas.

You mentioned The Beatles, Madonna, Tom Waits... how has your relationship with popular culture affected your artistic output?

I have a relationship with popular culture. It is part of who I am, part of my thinking and musical world. How it falls into my work is hard to explain. The Andy Warhol Diaries is a clear example where an element of popular culture was the impetus for the piece. In other pieces it may be a sound, or a formal musical element using only four chords...

It's hard to explain. I don't write popular music, that is, I have not in 20 years. But I do listen, when I listen, and it's a part of my sound world. My friend the composer Michael Pisaro once suggested to me that in order to incorporate pop music successfully into classical music you have to let go of your respect and love for it. To my mind, most composers my age find that hard to do. So the pieces seem to be orchestrations of pop music with a large orchestra, or with computer music. It seems that I may have decided that perhaps I too can not do it, so I just don't try. I don't try to create popular music with my music, but I use ideas from popular music as metaphors for some of the pieces I create.

Learn more about Dangerous Bend, the Amnon Wolman CD on the C74 label, by clicking here. You can visit Amnon's personal web page by clicking here.

Amnon Wolman creates music with technology and for instruments with two distinctive features: first, a subtle and complex relationship with popular culture and second, a concern for the dramatic nature of sonic evolution. His first release on the <a href="http://www.cycling74.com/c74music">c74 record label</a> is "Dangerous Bend." In this conversation with Ben Nevile, Wolman discusses his romantic compositional techniques and reveals the unusual nature of his introduction to computer-based music.

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