Computer Sound Transformation

A personal perspective from the U.K.

Trevor Wishart

Introduction

For the past thirty years I have been involved in developing and using sound
transformation procedures in the studio, initially working on analogue tape, and
then through various types of computer platforms as computer music came of
age. Over these years I've developed a very large number of procedures for
manipulating sounds. Being a composer, I refer to these processes as musical
instruments and they are developed as part of my musical work. However, I have
not been prominent in publishing this work in academic journals as I'm primarily
a working artist. Nevertheless, the processes (and source code) have all been
available to others with the facilities to use (or develop) them through a
composers' cooperative organisation based in the UK, the Composers' Desktop
Project. As there has been a recent surge of interest in the Phase Vocoder F1"
name=FF1>1 as a musical resource, I've been advised by friends in the academic
community to put my contribution to these developments on record.

Origins

The earliest successful transformations I developed can be heard in the piece Red
Bird (1973-77) 2. The musical structure of the piece was conceived in terms of
such transformations between sound types, but techniques for achieving this had
to be developed on an ad hoc basis - through discovering what was practicable
with the facilities available in the local analogue studio. The transformations, all
from the voice to other sounds, include 'lis' (from the word 'listen') to birdsong,
'rea' (from the word 'reason') to animal-sounds, 'reasonabl-' to water, and
various machine-like events constructed from vocal subunits. They were achieved
by combining the elementary studio facilities available (tape editing, mixing,
mixer eq) with extended vocal techniques (developed while working as a free
improvising vocal performer F3" name=FF3>3 ). A discussion of the approaches
used in Red Bird, and the concept of Sound Landscape, can be found in On Sonic
Art F4" name=FF4>4. A more detailed description of the composition of this piece
can be found in Red Bird, A Document F5" name=FF5>5.

Realising that these notions of spectral transformation could in principle be

generalised in a computing environment, when major computer music facilities
became available in Europe (at IRCAM in Paris) I submitted a proposal for a work
based on vocal transformation and was invited on the induction course in 1981.
There I discovered a potential transformation tool (Linear Predictive Coding F6"
name=FF6>6), and was invited to compose a work. Unfortunately the mainframe
system at IRCAM, and much of the indigenous software, was changed
immediately following this visit, and the project could not proceed until 1986,
when the research and composition for Vox 5 was finally commissioned. It was
suggested to me that the CARL Phase Vocoder (Moore, Dolson) might be a better
tool to use, but no-one at IRCAM at that time had inside knowledge of the
workings of this program, so I took apart the data files it produced to work out
for myself what was going on.

I eventually developed a number of software instruments for the spectral

transformation of sounds which were then used to compose Vox 5. These
instruments massaged the data in the analysis files produced by the Phase
Vocoder. The most significant of these were stretching the spectrum (see
below) and spectral morphing – creating a seamless transition between two
different sounds which are themselves in spectral motion. These are described in
a Computer Music Journal article F7" name=FF7>7.

Establishing a personal computer based development environment in the

U.K.

Returning from IRCAM to the UK musicians were faced with an entirely different
development environment. There was no independent national research centre
for music – music research was confined to University music departments. Most
of these were small and very poorly funded – they were seen as primarily sites of
humanities research and hence could not attract the money required for
advanced computing equipment, which at that time was very expensive. A
number of departments had PDP-11 computers accessible to a few research
students and staff – updating this equipment was a constant financial worry.

During 1986-7, a group of composers (initially Andrew Bentley, Archer Endrich,

Richard Orton, and myself) and developers (Martin Atkins and David Malham)
based in York, (and all ex-graduates or current staff of the University of York),
working in financially astringent circumstances F8" name=FF8>8, ported Richard
Moore's Cmusic F9" name=FF9>9 and the Mark Dolson Phase Vocoder to a
desktop platform. I then implemented the instruments developed at IRCAM. The
platform chosen at the time was the Atari ST as this machine was just fast
enough to be able to play stereo soundfiles running at a sample rate of 48,000 –
at that time the Macintosh was not fast enough F10" name=FF10>10 . This was
the start of a larger project to make computer music tools available to composers
and institutions without significant financial resources. This project was called the
Composers' Desktop Project (CDP). In fact the idea of a user-group development-
environment based on personal computers originated out of necessity in this
environment, predating subsequent developments at IRCAM and elsewhere by
several years.

The instruments ran initially in a command-line environment (a graphic

environment was developed later by Rajmil Fischman and others working at the
University of Keele). They were later ported from the Atari ST first to the Atari TT
then to the PC where they ran (and, in 2000, still can be run) under MS-DOS.
Almost all later platforms were chosen partly for their low cost – many CDP users,
including myself, did not have departmental salaries or budgets to buy expensive
personal computers or to constantly update them. The software also ran on the
Silicon Graphics machines at the University of York and elsewhere. More recently
two different graphic interfaces have been developed to drive the software, my
own contribution being the Sound Loom, written in TK/TCL so that it is potentially
portable from one computer platform to another.

Immediately after the IRCAM project in 1986, working in the CDP environment, I
developed a large number of other spectral transformation tools using the Phase
Vocoder data as a starting point. Subsequently, I also created a number of
original time-domain instruments (e.g. waveset manipulation, grain
manipulation, sound shredding) and extensions of existing instruments (e.g.
brassage F11" name=FF11>11). In 1994, a complete description of all the
spectral, time-domain and textural transformation possibilities available on the
CDP system was published in the book Audible Design F12" name=FF12>12. The
book has been used subsequently as a source by other software developers (for
example by Mike Norris who implemented many of my waveset manipulation
procedures on the Macintosh, now available from Sound Magic) some of whom
may well have had access to the CDP code.

I would stress that the work of researchers and developers at IRCAM (notably
Steve McAdams who introduced me to contemporary psycho-acoustic research on
the 1981 induction course and, later, Miller Puckette), and at the G.R.M. F13"
name=FF13>13 – where I attended the composition course in 1993 – were an
important source of knowledge, ideas and inspiration for my work. However,
when the sound morphing and spectral stretching instruments for Vox 5 were
originally developed as part of the public domain system shared by IRCAM,
Stanford and other major sites, IRCAM's research priorities were focused
elsewhere. The instruments did make their way to the USA via the University of
Santa Barbara and Dan Timis (the resident computer wizard at IRCAM when I was
working there). Later IRCAM did decide to pursue Phase Vocoder based
transformation and the Super Phase Vocoder (SVP) group was established (the
basis of the later AudioSculpt). During the development phase of SVP, when the
CDP spectral transformation suite was already quite large, I was a visitor at
IRCAM, and discussed possible transformational approaches with some of the
team working on the program.

The pioneering development work of the CDP has remained largely unknown or
forgotten about as the vast majority of the Computer Music community eventually
opted for the Macintosh as the machine of choice. Furthermore, being developed
primarily by a group without official financial support from within the University
infrastructure, the project was always short of resources. An initial grant from the
Gulbenkian Foundation helped propel us forward in the first 18 months, but this
was exceptional. Nevertheless the CDP continued to make both its instruments
and its code available to interested users and developers. I am indebted to the
work of many others developers (including in particular Richard Orton and
Richard Dobson) and to the C.D.P. Administrator, Archer Endrich, for continuing
to promote, support and develop the system, and make it more accessible to
users, despite the lack of financial rewards. The system has tended to be adopted
by independent composers or small educational institutions with limited budgets.
However, the source code has been available at a number of UK (and other)
University sites at different times, even after these moved to a primarily Mac-
based studio system. And some institutions, notably the Institute of Electronic
Music in Vienna, developed sophisticated graphic interfaces of their own.

The Instruments – (1) Spectral Transformation using the Phase Vocoder

There is not enough space to describe all the C.D.P. procedures in this article, so
I will describe only the more interesting ones, or those not available elsewhere.
Full descriptions of all these processes can be found in Audible Design.

In the first phase of development (post 1986), many spectral transformations

were implemented in the Atari environment. These included spectral morphing
(see above), and various types of spectral shifting and spectral stretching,
from a linear shift (adding a fixed value to all frequency data, thus e.g. making a
harmonic spectrum become inharmonic), through a multiplication (preserving
harmonic relations between data, but transposing the pitch – but with the ability
to split the spectrum at a given frequency, and hence produce doubly-pitched
output sounds) to differential multiplication of the data (spectral stretching, a
more sophisticated way to convert harmonic into inharmonic spectra, used in Vox
5 to convert vocal sounds into bells). Time-variable time-stretching
procedures were also implemented, more general than those existing in the CARL
Phase Vocoder implementation itself. These are important if one wishes to
preserve the attack characteristics of a sound while time-stretching the sound (as
a whole) by a large factor.

Spectral cleaning was developed using a comparative method – part of the

spectrum deemed to be (mainly) noise (and, in some options, part of the
spectrum deemed to be clear signal) being compared with the rest of the signal
and appropriate subtractions of data or other modifications made.

From a musical point of view, the most innovative early new developments were
spectral banding, a rather complicated 'filter', which enabled the spectrum to be
divided into bands, and various simple amplitude-varying (and in fact frequency-
shifting) processes to be applied to the bands, spectral tracing and spectral
blurring.

Spectral tracing simply retains the N channels with the loudest (highest
amplitude) data on a window-by-window basis. If N is set to c. 1/8th the number
of channels used in the PVOC analysis, this can sometimes function as an
effective noise reduction procedure (the value of N which works best depends on
the signal). When N is much smaller than this, and a complex signal is processed,
a different result transpires. The small number of PVOC channels selected by the
process will vary from window to window. Individual partials will drop out, or
suddenly appear, in this elect set. As a result, the output sound will present
complex weaving melodies produced by the preserved partials as they enter (or
leave) the elect set. This procedure is used in Tongues of Fire F14"
name=FF14>14.

Spectral blurring is an analogous process in the time dimension. The change in

frequency information over time is averaged – in fact, the frequency and
amplitude data in the channels is sampled at each Nth window, and the frequency
and amplitude data for intervening channels generated by simple interpolation.
This leads to a blurring or 'washing out' of the spectral clarity of the source.

Arpeggiation of the spectrum (a procedure inspired by vocal synthesis

examples used by Steve McAdams at IRCAM to demonstrate aural streaming) was
produced by 'drawing' a low frequency simple waveform onto the spectrum. This
oscillator rises and falls between two limit values – values of frequency in the
original spectrum – specified by the user. Where this waveform crosses the
spectral windows, the channel (or surrounding group of channels, or all the
channels above, or all those below) is amplified. Spectral plucking was
introduced to add further amplitude emphasis (and an element of time-decay of
the emphasized data) to the selected channels.

A number of other processes (such as spectral freezing and sustaining of the

spectrum at particular moments, and spectral interleaving, timewise, the
spectra from different sources) were implemented in this early phase.

Tuning the spectrum was introduced a little later. Tune spectrum works by
selecting channel data lying close to the partials of a specified set of pitches, and
moving the frequency of that data to (or towards) the desired partial frequency.
The spectrum can also be traced (see above) before doing this. Choose partials
selects channels which should contain frequencies close to those of a specified set
of partial frequencies (harmonics of, odd harmonics of, octaves above, linear
frequency steps away from, or a linear frequency displacement from harmonics of
a given fundamental). As analysis channels above the 21st are sufficiently narrow
to focus on a semitone band of frequency or less, the channel number itself is
sufficient to grab the desired partials.
After discussing possible algorithms with the SVP developers, I implemented
some of their ideas for spectral filters (defining filters in a more conventional way
than the banding procedure described above), and implemented various types of
low pass, high pass, band pass, notch and graphic e.q. spectral filters,
together with a chorusing procedure suggested by Steve McAdams' work
(introducing jitter into the partials data).

After discussions with Miller Puckette about his work on tracking the pitch
produced by instrumentalists performing in real time, procedures to extract the
pitch of PVOC data were finally developed into a useful form, and instruments to
correct the data, to transform the pitch data (quantise, shift, vibrato,
approximate, or randomise the pitch, and exaggerate, invert or smooth
the pitch contour), and apply the pitch to other sounds, were developed.

At the same time, the extraction of formants from the PVOC data was
implemented satisfactorily for the first time within the CDP environment. This
enabled the inner glissando procedure to be developed. Here, the process
retains the time-varying spectral envelope (the formant envelope) of the sound,
but replaces the signal itself by an endlessly glissandoing Shepard Tone signal
F15" name=FF15>15.

Shuffling the sequence of windows, and weaving a specified path (including

possible repetitions and omissions) through the windows were implemented at an
earlier stage. A 'drunken-walk' through the analysis windows was suggested by
Miller Puckette's work in MAX. Miller also suggested the procedure of octave
pitch-shifting through selective partial deletion, while Oyvind Hammer of
NOTAM F16" name=FF16>16 proposed scattering of the spectral data.

The Instruments – (2) Original Time-domain procedures

Alongside this spectral transformation work, a large number of time-domain

procedures have been developed for sonic composition.

Waveset distortion was developed for the CDP while composing Tongues of
Fire. I defined a waveset as the signal between any pair of zero-crossings. With a
simple sine-wave the waveset corresponds to the waveform. But even with a
harmonic tone with very strong partials, the waveform may cross the zero more
than twice in a complete cycle. In this case the wavesets are shorter than the
waveform. With complex signals (e.g. speech) containing noise elements, the
definition of the waveset produces many varieties of technically arbitrary, but
potentially musically interesting, artefacts. A whole suite of procedures was
developed to manipulate wavesets. I have used three at prominent moments in
compositions.

The first of these involves replacing each waveset with a standard-shape

waveform (e.g. a sinewave). This produces a very pronounced spectral
transformation of the source, but one where the zero-crossings of the result are
exactly aligned with those of the source. It is thus possible to use a simple mixing
procedure (another CDP process, Inbetweening, does this) to produce a
sequence of sounds intermediate between the source and the new sound. These
two procedures were developed and used to produce the 'Wood' to 'Drum'
transformations in Tongues of Fire.

The second, waveset averaging, involves extracting the shape of each waveset,
and then averaging this shape over a group of N adjacent wavesets. Again, this
produces an extreme modification of the source (usually a relatively harsh sound
and often a transformation so distant from the source that little audible
connection is apparent!) and is used in the 'fireworks' transformation immediately
after the rhythmic climax of Tongues of Fire. The article Sonic Composition in
'Tongues of Fire' F17" name=FF17>17 discusses this in more detail.

Finally, waveset repetition generates unusual pitch artefacts in complex

signals. In particular, any small fragment of a noise signal, if repeated a number
of times, generates a tiny pitch artefact. The second movement of Two Women
F18" name=FF18>(18), based around the voice of Princess Diana, uses this
instrument to ornament and fragment the vocal material, different repetition
rates being used in the left and right channels to produce an irregular panned
echo/delay, with iteration (see below) being used as a reverb-like process to
sustain various pitch elements which arise from the first procedure.

Familiarity with the G.R.M.'s work on the classification of sounds F19"

name=FF19>19 drew my attention to the difficulty of time-stretching iterative
sounds i.e. sounds like a rolled 'r' or a low contrabassoon pitch, where the sound
is perceived as a series of individual attacks. In (realistic) time-stretching, we
need to avoid time-stretching the event attack itself as stretching this can
dramatically alter our recognition (or mental classification) of the source. Hence
we would usually apply a time-stretching parameter which itself varies through
time, being 1.0 (no stretch) during the attack, and increasing rapidly to the
desired stretch ratio immediately after the attack. With iterative sounds,
however, we are faced with a whole stream of attacks, and this simple solution is
not available. To deal with these a number of Grain manipulation instruments
were developed. These instruments extract the (loudness) envelope of the sound
by gating it. Using this envelope the source can be fragmented into attacked
elements and these elements repositioned in time (or in pitch or both) in the
output sound. (The process can also track the overall amplitude of the source and
adjusts the gate level for the grains correspondingly).

This approach also allows one to reverse an iterative sound. Most sounds have
an asymmetric form with a (relatively) loud initiating event at the beginning, and
a tailing away to zero at the end (these features themselves can have a vast
number of forms). Playing a sound backwards therefore rarely results in a sound
that we recognise as being a close relative of the original. Only sounds of (on
average) steady amplitude which have attack and decay as inverses of one
another e.g. a slow fade in matching a slow fade out, will appear similar when we
reverse them. Iterative sounds are particular difficult in this respect as every
attack within them gets reversed. If we extract the grains and then sequence
them in the reverse order, without reversing the grains themselves, we achieve a
convincing sense of retrograding the sound without change of source recognition.

Similar sound-structural considerations apply to extending sounds using looping

procedures. Recording a rolled 'r', isolating a single tongue-flap sound, then
looping it to generate a 'rolled-r' at the same rate as the original produces an
entirely mechanical artefact sounding completely unlike the original rolled-'r'
source. Natural 'repetition' is usually micro-inexact. Thus the Iteration
instrument allows a signal to be looped, but imposes (user-controlled) random
pitch, amplitude and timing fluctuations on the repeated elements. Using
Iteration the sound generated from the single flap can be extremely natural (but,
of course, more distant transformations are also possible). Grain manipulation
and Iteration were both developed and used while composing Tongues of Fire.

Various instruments allow scrambling of a sound through simple editing and

rejoining of the edited segments. In particular, in Sound Shredding, I cut up a
sound (at random time-points) into a number of separate segments, shuffles
these segments, and reassembles them to the exact duration of the original. The
resulting sound is then cut up again, differently, and the reassembly repeated.
This process can take place any (user-specified) number of times. Applying the
process about 400 times to rapid speech material produced a sound very similar
to that of water running around rocks in a small stream, and this transformation
can be heard spanning a 2 minute section of Tongues of Fire.

The Instruments – (3) New perspectives on existing procedures

What are now referred to as granular synthesis procedures, but applied to

input sources, were developed at any early stage of the CDP. (The CDP
instruments are almost exclusively concerned with the transformation of existing
sources, rather than with synthesis). These are described as texture generation
instruments. Initially these procedures generated scripts for a simple Csound
F20" name=FF20>20 instrument which read (any number of) input sounds and
then distributed them in the texture according to the instructions given by the
user. The dependence on Csound scripts was superseded by direct use of the
soundfiles themselves.

Texture Generation was (and is) able to use an arbitrarily large number of input
sounds, to generate a stream of events where all the following parameters can
themselves vary through time:

• the average time between event repetitions (the density of events) or the
specification of a sequence of event times
• the scatter (or randomisation) of event timings (which means the
instrument can generate anything from dance-music-like regularity to
complete arhythmicity)
• a quantisation grid for times (or none)
• a specification of which range of input sounds are to be used
• the range and range-limits of pitch-transposition of the events
• the range and range-limits of event amplitudes
• the range and range-limits of durations of the individual events in the
texture
• the spatial centre of the texture on the stereo stage, and its motion
• the spatial bandwidth of the texture on the stereo stage.

A neutral texture is generated from independent events over a transposition

range without regard to tuning, tempering etc. However, the texture can also be
generated...

• over a harmonic field (not necessarily tempered) which can itself change
through time
• clustered into groups of events of specified or random pitch-shape
• formed from a line with arbitrary or specified decorating patterns (which
themselves have properties with independent parameters of their own).

The texture generation instrument are used extensively in all my sonic art pieces
since Vox 5.

The brassage techniques extensively and powerfully developed by the G.R.M.

and implemented (in various guises) in G.R.M. tools I have independently
implemented in the CDP environment. The G.R.M. have divided brassage into a
series of sub-categories based on musical outcomes (based on many years of
musical experience), providing the user with control of parameters over a
musically meaningful range for each resulting tool. I admire this approach and
accept that it is much more accessible to the user who is a computer user rather
than a programmer. However, compositionally I often find it interesting to explore
the areas where a process pushes against its limits and falls over into another
area of perception. E.g., if the size of grains used in a time-stretching brassage
routine (as used in the harmonizer) exceeds a certain threshold we begin to hear
the resulting sound as a rapid collage of elements rather than as a simple
timestretch. Hence the CDP brassage routine offers timestretch, pitch-shift,
granulation and source-scrambling as independent modes, but also allows
access to all the parameters of the brassage process at the same time..

• timestretch or compression and its range

• segment density and its range
• segment size
• segment transposition and its range
• segment amplitude and its range
• segment splice-length
• segment spatial position
• segment spatial scatter and its range
• segment timing randomisation
• segment search-range in the source

where every parameter can also be varied in time.

The process can also be applied to more than one input sound.

Modifying the loudness contour (envelope) of a sound is a fairly standard

procedure. Envelope extraction and superimposition (written by Richard
Orton) and envelope manipulation (which I developed from Richard's
programs) were some of the earliest processes to be developed in the CDP. These
allow the envelope to be extracted at different resolutions (e.g. a tremolo sound
which crescendos has a small-scale, rapidly-varying loudness envelope defining
the tremolo, and a large scale overall envelope defining the crescendo. These can
be extracted separately using a different window-size for the envelope
extraction). The envelope can then be changed (envelope warping – normalise,
limit, compress, exaggerate, corrugate etc), and applied to the original, or a
different sound. Even applying the envelope comes in two varieties. Simple
envelope superimposition is found in most mixing packages (often implemented
by drawing the envelope), where an envelope contour is imposed on the existing
sound. However, we can also envelope replace, where the new envelope
replaces (rather being superimposed over) the envelope of the processed sound.
In this case we force the original sound to have a flat level throughout (treating in
a special way points in the sound where the envelope approaches zero), then
apply the new envelope to the flattened sound.

Enveloping can obviously be used to produce tremolo. More radically, in a

reversal of the Karplus-Strong synthesis procedure F21" name=FF21>21, we can
produce a plucked attack on an existing sound. (The procedure involves
finding the first steady-pitch wavecycle in the source – assuming there is one –
then preceding it by copies which become increasingly loud and noisy). This
process was developed and used in Tongues of Fire but is not 'automatic' in its
operation and is quite tricky to sculpt.

Mixing is now usually carried out in a graphic environment displaying pictorial

representations of waveforms (and envelope and panning contours) in tracks on
the screen. The CDP Submix instrument (which I developed from existing CDP
mixing facilities developed by Andrew Bentley and others) is based on a much
earlier paradigm, mixing from a (text) list of soundfiles. Despite being much less
friendly than screen based mixing, this does allow for some powerful global
procedures to be applied to mixes. The CDP submix should be thought of as a
way to generate a new event from several source sounds, rather than as a
conventional track-mixing environment (although I do all my mixing in this
environment).

First of all, there is no limit to the number of 'tracks' used (apart from the
memory space of the computer). Any number of sounds can be superimposed.
Secondly, global operations on the mix are available, from simple features like
doubling (or multiplying by any number) the distance between event onsets, or
randomising them (very slightly or radically), to randomly swapping around the
sound sources in the mix, automatically generating particular timing-sequences
for event entry (from regular pulses, to logarithmic sequences etc.), or
redistributing the mix output in the stereo space in a new, user-defined way.

More specialised procedures involve synchronising the mix events (e.g. at

their mid-point, or end, as well as at their start), or synchronising the event-
attacks (where the search-window for the attack peak can be delimited by the
user). These latter procedures are particularly useful for building complex
sonorities out of less rich materials e.g. by superimposing transposed copies of
the sound (over the original duration, or in a different duration) onto the original.
Similarly, Inbetweening allows the generation of sets of closely related
sonorities (see above), while Cross-fading using a balance function allows a
sound to gravitate between its original form and a transformed variant in a time-
varying way.

There are no original filter algorithms in the CDP, but some powerful filter design
frameworks are available. In particular filter varibank allows one to define a
filter over a set of pitches which itself varies in time, where each pitch element
has an associated amplitude (which can go to zero so that pitches, or moving-
pitch-lines, can be 'faded out' or cut). The number of harmonics of those pitches
(and their relative level) can be specified (these serve to define further individual
filter frequencies), and the filter Q can also vary through time. This filter-building
algorithm was developed and used during the composition of Fabulous Paris F22"
name=FF22>22.

Finally, at a time when synthetic bell-sounds seemed to dominate computer

music works, I decided a bit of grittiness would be welcome, and developed
instruments which lower the resolution of the sound (reducing the effective bit-
representation, or the sampling rate), ring modulate and inter-modulate
sources, and even attempt (rather unsuccessfully) to simulate manually
scrubbing a tape over the heads of an analogue tape-recorder.

Additional Aids to Composing

Over the years I have also developed a large number of utilities which I find
indispensable as a composer, starting with an instrument which searches a tape
of source recordings and extracts significant segments from surrounding
silences or clicks, using gating, and selection parameters specified by the user.
Next there are facilities to compare sounds, or compare the channels of a
single sound, (are they the same, or almost the same to within specified
limits?), to balance the level of sources, or the channels of a source, to invert
(or narrow) spatial orientation, and to invert phase (which, apart from
anything else, can be used to gain more headroom in a mix).
In the various instruments described in this article, alomst all parameters can
vary through time. Data for this is provided in simple textfiles containing
time+value pairs. To aid in working with such data, hundreds of automatic data-
creation and data-modification processes have been implemented, and are made
available in the Table Editor, now also driven from the graphic interface. I have
used it to design and modify complex filter specifications, to generate 'random
funk' accentuation patterns as envelopes over an existing stream of events
(Birthrite A Fleeting Opera F23" name=FF23>23) – and even to do my tax
returns(!). As an additional aid, a Music Calculator allows easy conversion
between a great variety of musical and technical units.

The future...

Currently (October, 2000) all this software works in non-real-time in a PC

environment, mainly with 16-bit soundfiles. The next version (early 2001) will
handle all currently available soundfile formats. In addition the sound-buffering is
being modified so that those instruments which could, in principle, run in real-
time can be enabled to do so. There is also no reason (apart from lack of time or
resources) why this entire environment should not run on the Macintosh, or any
other platform, as it is written in 'C' and TK/TCL. These two latter tasks could be
accomplished without great difficulty by someone with the time and enthusiasm
to commit.

Footnotes

1. A process which divides the source sound, timewise, into tiny

(overlapping) 'windows', performs a fourier analysis on each window to
determine the spectrum of the sound in the window, then deduces the
frequency of the components in the window by considering the change of
phase from one window to the next.
2. This can be found on the CD Red Bird: Anticredos (EMF CD022)
3. The vocal techniques were documented in a catalogue of extended vocal
techniques, The Book of Lost Voices (1979), later incorporated as a
chapter in the book On Sonic Art (see footnote 4).
4. Originally published privately in 1984: republished (edited by Simon
Emmerson) by Harwood Academic Publishers, 1996. (ISBN:371865847X)
5. Originally published privately by Trevor Wishart, 1978. See publications.
6. A procedure developed for the analysis and resynthesis of speech. It first
differentiates noise and pitch based elements in the source. It then
generates a sequence of filter specifications for consecutive moments in
the source which, when applied to a buzz (rich in harmonics) tone or a
noise source, reproduces the original sound.
7. The Composition of Vox 5 at IRCAM : Computer Music Journal Vol. 12: no
4: Winter 1988.
8. Despite the involvement of staff members, no financial support was
forthcoming from the University authorities at that time. I donated 100 to
help pay for materials to build the first 'SoundSTreamer', the buffering
device, designed and built by Martin Atkins and David Malham, which
enabled us to get sound in and out of the ROM port of the Atari ST.
9. General purpose software sound-synthesis environment.
10. To all those chuckling into their anoraks I would add that the Atari ST was
100% reliable. It simply never crashed in all the years it was used.
11. A process that cuts the sound, timewise, into segments (possibly
overlapping, possibly separated in time), then reconstructs the sound by
splicing these back together in different ways.
12. Published by Orpheus the Pantomime, UK. (ISBN : 0951031317)
13. Groupe de Recherche Musicale, Paris.
14. Tongues of Fire is available, by itself, on CD, or on the album
Voiceprints.
15. A tone which appears to rise (or fall) in (chromatic) pitch forever while
remaining in the same tessitura.
16. The Norwegian Centre for Computer Music.
17. Computer Music Journal: Vol 24 No 2 Summer 2000
18. On the CD Voiceprints.. See publications.
19. Solfege de l'objet sonore by Pierre Scaheffer and Guy Reibel.
20. General purpose software sound-synthesis environment, by Barry Vercoe.
21. A compact algorithm to synthesize plucked-string sounds of many types.
22. On the CD Or Some Computer Music: 1 from Touch.
23. Birthrite A Fleeting Opera with Max Couper: River Thames, London, 2000.
A score of Birthrite is also available, for dryland performance.