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[ Synthesizers and Samplers]  


Electronic Musical Instruments

Classification by sub-system technology:

Completely analog signal processing with control voltages (CV), in polyphonic machines each channel is implemented as a separate monophonic synthesizer, VCO with square/sawtooth/triangle/sine/variable pulse waveform generation with CV for frequency and waveform shape, VCO optionally has a sync input (normally triggered via output of other VCO), analog multipliers (e.g. ring-modulator) for waveform mixing, analog noise generator with filter for spectral shaping, VCA for volume envelope shaping, VCF with resonance for "subtractive" spectral shaping
a) analog CV generation and processing by use of analog envelope generators, analog LFO (LF VCO) for modulation, analog random-CV generators based on noise generators with sample&hold (S&H) techniques. Sequencers mostly based on SSI TTL or CMOS counters+demultiplexers and analog CV gates.
Example: Moog Modular, MiniMoog, PPG Model 300, PPG 1002, Korg PS-3100, most early modular monophonic/polyphonic analog synthesizers and drum machines
b) "Hybrid analog": digital control subsystem, partially even digital envelope generators and LFOs, DACs for CV generation, typically "programmable" polyphonic with digitally storable sound patches and settings. Later models often use digitally controlled oscillators (DCO, e.g. Xtal master with frequency dividers) with analog waveform generation. Microprocessor-based sequencers.
Example: PPG 1003, Yamaha CS80, MultiMoog, MemoryMoog, Jupiter-8, Juno series, Prophet-5, TB303, CR78, TR808, TR909 (partially), most MIDI-era analog synthesizers and drum machines

Digital control, digital envelope generators, digital waveform generation, digital or analog (VCO) clock generators, microprocessor control
1) variable clock frequency for pitch variation per channel, each channel has a separate clock generator (or even VCO) as waveform data clock, separate waveform output DACs with analog post-processing VCA (or MDAC) and VCF for envelope and spectral shaping.
1a) wavetable synthesizer/sampler: clock advances sample address register by one.
Example: Fairlight CMI, PPG Wave 2, Emulator II, Emulator III, Drumulator, LinnDrum, TR909 (partially), early digital wavetable synthesizers/samplers and drum machines
1b) special FM synthesizer: clock triggers phase register increment, phase and phase increments are variable but fractional (i.e., 0<=phi<2*pi), digital phase/frequency modulation via sine lookup-tables and phase adders.
Example: Synclavier (digital FM synthesizer)
2) pitch variation by variable phase increment at fixed clock rate, phase accumulator has integer and fractional part (or even floating-point), decimation/interpolation algorithm to obtain sample value from phase accumulator, fixed waveform data output rate with parallel or time-multiplexed processing of multiple channels.
2a) Shared or separate DACs, envelope and spectral shaping by analog VCF/VCA post-processing, in modern designs waveform generation often implemented by use of ASICs.
Example: PPG Wave 2.2, Ensoniq Mirage, Ensoniq ESQ-1, Waldorf Microwave, early digital wavetable synthesizers/samplers and drum machines
2b) Completely digital signal processing, digital envelope-shapers and filters, typically digital effect machines included with multiple digital audio-busses (e.g. I2S), typically mixed multi-channel digital and analog outputs, typically implemented by ASICs and CPUs/DSPs.
Example: Yamaha GS1, DX7, Kawai K5, Emulator IIIX, most modern samplers, wavetable or algorithmic (e.g., additive, FM, virtual analog, acoustic modeling, granular synthesis) synthesizers and drum machines


  • The main classification of "Analog" and "Digital" is based on the waveform generation.
  • VCO/VCF/VCA with digitally generated CVs (via DAC) are sometimes called DCO/DCF/DCA. Quite often the term DCO is also used for a digital clock generator (e.g. based on bit-rate-multipliers ("BRM") and dividers).
  • Analog-a) could also be considered as a special-purpose analog computer for sound generation.
  • Analog-a) and Analog-b) differ in system/envelope control.
  • The exotic case Digital-1b) could also be considered as a combination of Digital-1) and Digital-2a).
  • Digital-2a) and Digital-2b) differ in post-processing.
  • Digital-2b) could also be considered as a special-purpose computer for sound generation, without the historically grown separation into logical units/sub-systems like oscillator, waveform generator, envelope generator and post-processing. Even the classical algorithms like additive, wave-table or FM/PM can be seen as compromises in older generations of digital synthesizers due to technical limitations.
  • The table above lists extreme categories, combinations are possible.
  • Early samplers and wavetable synthesizers with limited Bit-counts and/or sample-memory have used non-linear encoding/companding schemes of sample-values (e.g. 8-Bit mu-255 and/or DPCM) to improve the signal-to-noise ratio as compared to linear encoding. Examples: LinnDrum, Emulator.

Frequency- and phase-modulation synthesis:

Given are 2*pi-periodic functions a1(phi1) and a2(phi2) with time-dependent phases phi1=phi10+omega1*t and phi2=phi20+omega2*t, respectively. (Note that omega=2*pi*f). Frequency-modulation (FM) and phase-modulation (PM) are defined as:

  1. Phase-modulation (PM) is defined as:
    with the so-called modulation-index phi12.

  2. Frequency-modulation (FM) is defined as:
    with A2 such that a2=dA2/dphi and the frequency deviation omega12. The modulation-index thus is omega12/omega2, which is frequency-dependent. Note that the "momentary frequency of a1" thus reads dphi1/dt=omega1+omega12*a2 (hence "frequency modulation").

Note that the modulation-index determines the Fourier-spectrum of the resulting function a12. (In general, FM and PM are only equivalent if a1 and a2 are special functions such as sine or cosine). If the ratio of omega1 and omega2 is irrational (i.e. the frequencies are in-commensurable), the spectrum of a12 becomes non-harmonic (i.e. a12 is not periodic).

"FM synthesis" is very often implemented as a phase-modulation. PM has the advantage of a frequency-independent modulation index (which determines the spectrum and hence the "sound" or "timbre") and a DC-free output (which is especially important in higher order modulation chains to avoid detuning.) In general, multiple oscillators with variable amplitudes, phases and frequencies can be employed, especially complex modulation chains, where an oscillator output is employed as a modulation source for other oscillators. (Concerning the amplitude modulation, additive synthesizer concepts are closely related, see below.)
For technical reasons (stability + modulation-speed and -depth), almost all FM/PM syntesizers have a digital oscillator/modulator implementation (phase accumulator registers, multipliers, adders, waveform ROMs, see e.g. Yamaha GS1 or DX7). One of the strengths of FM/PM synthesis is the ability to easily produce complex (esp. non-harmonic) spectra (in contrast to subtractive synthesis and classical additive synthesis, see below) and the broad range for real-time timbre control/modulation. Using non-sine functions further enhances the possibilities of FM/PM synthesis (e.g. a combination of FM and wavetable synthesis as used in the FM-part of the Synclavier and later Yamaha synthesizers).
Historically, John Chowning of Stanford University has invented (and patended later on) the concept of using FM for audio signal synthesis.
(Note: Typically, classical FM/PM synthesis can produce "bell-like" and "metallic" sounds. FM/PM sound-programming, esp. trying to reproduce realistic sounds, is complicated and not straight-forward from scratch, although interesting artificial timbres can be obtained quite easily.)

Fourier, additive, and wavetable synthesis:

Let us first consider a given audio signal, which is nothing but a time-dependent real function a(t). In a digital synthesizer, the audio signal is sampled at equally spaced discrete points in time, and the sample-values are discretized, i.e. quantized (but not necessarily linearly, e.g. mu-law companded).
The so-called Fourier transformation provides a complex frequency-dependent function A(f) (the "spectrum") for such a given time-dependent audio sample. Firstly, we have to note that the values of a Fourier-transformed function are complex, i.e. they have a phase and amplitude. Furthermore, restricting the audio sample to discrete time-values with step width dt, the resulting spectrum becomes highly symmetric (A(f)=A(f+n*fmax) with fmax=1/dt and an integer n). (We assume that the original a(t) is bandwidth-limited to -fmax/2,...,fmax/2 in order to avoid aliasing. Then, the original a(t) can be fully restored via "sin(x)/x folding" according to the sampling theorem.). The assumption of a real sample function a(t) implies that A(-f)=A*(f) (i.e. complex conjugated). It is therefore sufficient to restrict ourselves to the interval f=0,...,fmax/2 (e.g. with dt=10microsec => f=0...50kHz). The next important property of the spectrum lies in the fact that the sample function a(t) has a finite length, i.e. t=0...tmax. Thinking of a periodic continuation (period tmax), the frequency spectrum also becomes discrete in the frequency-variable (providing a so-called Fourier-series) in steps df=1/tmax (e.g. tmax=10sec => df=0.1Hz). Hence we have Nt=tmax/dt time values and Nf=fmax/df frequency value, giving Nf=Nt/2. Note that the factor of 2 basically stems from the fact that the spectrum A(f) is complex-valued (real and imaginary part or: amplitude and phase), whereas a(t) is real, giving the same amount of information. The Fourier transformation of such periodic discrete-argument functions is called discrete Fourier transformation (DFT). As a special case, if Nt is a power of two (i.e. Nt=2^n with some integer n), the DFT can be computed very efficiently by use of the fast Fourier transform algorithm (FFT).
The periodic continuation of an audio sample (we have introduced above as a simplifying assumption) can be re-interpreted in the case of a very small tmax: If df=1/tmax lies in the audio range (e.g. tmax=1msec => df=1kHz), tmax becomes nothing but the so-called base-period of a stationary sound-wave. In this case, df is nothing but the base-frequency. Furthermore, since we have seen that f can only take discrete values, namely an integer times df, the spectrum of such a sound has an evenly-spaced comb-structure which is called harmonic.
So far, we have considered the purely time-dependent audio sample a(t) and its purely frequency-dependent spectrum A(f). Where a(t) is the object that samplers deal with, the A(f) for a large tmax (e.g. 10sec) is often not very useful in terms of synthesizers. One has to note that the human ear perceives something like a combined time- and frequency-dependent function: Roughly spoken, everthing above approx. 20msec is time-resolved. That is, the complete Fourier transformation as discussed above is not the mathematical method we are looking for. From this point we can take various paths for contructing an algorithm which is adequate for the ear's analyzing capabilities. Here, we want to discuss two synthesizer concepts:

  1. The so-called short-time discrete Fourier transformation (STDFT) as used in digital additive and wavetable synthesizers, as well as in effect engines. Here, the whole sample (the time-dependent function a(t)) is split into smaller, possibly overlapping "time-frames" of equal sizes (typically tmax(frame)<<20msec see above) giving b(t1,t2):=a(t1+t2) with two time-variables: time within frame t1 and frame start position t2. At this point, the STDFT analysis makes a partial DFT (or FFT see above) within each frame, providing a time- and frequency-dependent function B(f1,t2) (time-dependent "frequency frames" or "timbres"), in contrast to a complete Fourier analysis of the whole sample, which provides a purely frequency-dependent function. Non-harmonic spectra, broadened peak-structures, and sub-harmonics can be generated (to a certain degree) by choosing synthesis/analysis frames with a length given by an integer multiple of the audible base-period (e.g. an FFT frame of 2048 samples for a base-period of 128 samples). To avoid audible sub-harmonic artefacts (given by the time-frame length), overlapping frames are typically used.
    The STDFT is employed to resynthesize (from samples) or create sounds in a time-frequency domain, which is related to the human ear's sound-reception properties. After creating/editing this fuction B(f1,t2), the STDFT synthesis generates b(t1,t2) which in turn is played back by the wave-table sound generator. Here, the index t1 is looped continuously and t2 is slowly varied according to a given envelope generator ("small-loop sample-player with varying loop position"). In that sense, samplers (which play a(t)) can be considered as wavetable synthesizers which play each frame in b(t1,t2) only once (yielding the time-scaling "artefact" with varying pitch which is typical for samplers).
    As a generalization, the playback-speed/phase within the frame (t1) can be modulated, providing an additional frequency
    /phase modulation of the varying frame-waveform (see FM/PM synthesizers above, e.g. FM-part of Synclavier).
  2. Additive synthesis. Here, we start from a small tmax (<<20msec see above) for just one base-period of a stationary sound-wave a1(t), characterized by its spectrum A1(f). The concept of an additive synthesizer is to vary the individual amplitudes (and possibly phases) in A1(f) (normally slowly) in time during playback. (Please note that a stationary sound-wave is perceived by the human's ear as a sine-like tone after some time.) Mathematically, we hence introduce a function B1(f,t) (with B1(f,0):=A1(f)) which is is employed by the sound generator ("set of equi-distant sine-oscillators with individual envelopes") to compute an output signal in real-time.
    As a generalization, we can discard the condition of evenly-spaced oscillator frequencies and additionally vary each oscillator's frequency in time, providing frequencies f(i,t) (with i the oscillator index) and generalized envelopes E(i,t). Please note that we already produce a non-harmonic generator output (if long-time Fourier-analyzed) if B1(f,t) above varies in time: The phase variation provides a frequency shift of the individual oscillator (proportional to dphi/dt) and the amplitude variation broadens the peaks (proportional to d|B1|/dt, formerly delta-functions). Such a concept of a set of sine-oscillators with time-dependent amplitudes, phases and frequencies is nothing but an FM/PM synthesizer (especially if oscillator outputs serve as modulation sources for others, see FM/PM above). At this point, we speak of a classical additive synthesizer if we have oscillators with fixed evenly-spaced frequencies, and of a classical FM/PM design if sine-oscillators with variable frequencies/phases and amplitudes are employed.

One has to note a similarity between the B(f1,t2) of the wave-table synthesizer and the B1(f,t) of the real-time additive synthesizer. (Hence, wave-table techniques are sometimes also referred to as additive.) These two synthesizer concepts are mathematically equivalent under special conditions, the sound quality of such machines is closely related. Due to the large number of required oscillators in the concept of real-time additive synthesis (as defined above), historically, samplers and wavetable techniques in conjunction with non-realtime waveform generation software have been used (e.g. Fairlight CMI or PPG Wave). For technical reasons (oscillator stability), most additive and wave-table synthesizers employ digital implementations. Modern additive synthesizers are based on DSPs (e.g. Kawai K5000).
(Note: Typically, classical additive/wavetable synthesis can produce "technical", "organ-like" and "bright" sounds. Additive/wavetable sound-programming, esp. trying to reproduce realistic sounds, is tedious. Programming from scratch often results in stationary and primitive timbres. However, with the help of the resynthesis of recorded samples it becomes quite intuitive and provides interesting "surrealistic" timbres.)

Digital pitch generation overview:

Typically, two different approaches have been pursued in digital
synthesizers and samplers to vary the pitch (see above):
1) variable clock rate:
advance one original sample (in sampler/wavetable synthesizer, one phase step in Synclavier FM synthesizer) per clock pulse. This requires a programmable digital oscillator/bit-rate-multiplier/divider (or even a VCO) and a DAC for each output channel. Typically, a VCF/alias tracking filter and VCA is used for analog post-processing. Used in older (small series, custom design, expensive) wavetable synthesizers and samplers, also in older drum machines with a VCO as clock generator. With a perfect sample clock, the quantization and alias noise has always the same relative spectrum, independent of pitch because ratio "playback pitch : sample-output rate" is constant. Hence, such a pitch generation scheme is generally considered to provide a high quality natural sound from a musical point of view. However, typical digital oscillator implementations (based on bit-rate-multipliers and/or dividers) often have slight phase fluctuations (jitter) producing additional noise. Combined with low sample rates, insufficent alias filtering, and low Bit-count quantization (and even analog VCF/VCA post-processing) this provides typical digital synthetic sounds of the 1980s.
2) interpolation/decimation with fixed sample output clock rate:
variable fixed-point (or even floating-point) phase increment at fixed clock rate, combined with an interpolation/decimation algorithm (e.g. "drop/add sample"="take integer part as sample address" as the most primitive one or even "sin(x)/x folding") for sample output processing. In older systems with analog VCF/VCA post-processing. Modern instruments employ digital post-processing (mixing, routing, effects engines) and LSI/VLSI-IC (ASIC, DSP, or FPGA) based sound generators with parallel processing of multiple channels of waveform data. Typically used in modern algorithmic (e.g. FM or acoustic modeling) and wavetable synthesizers and samplers. This pitch-generation method can suffer from noise/artefacts (e.g. in early or low-cost machines) that depend on the ratio of fixed sample-output rate to playback pitch (i.e. the pressed key) and are determined by the chosen interpolation/decimation algorithm. In particular the "drop/add" scheme is generally considered to provide a "cheap" sound quality.

  Digital synthesizers and samplers:
  • Fairlight
    CMI ("Computer Musical Instrument"):
    sampler + additive synthesizer
    -> Lots of hardware/software infos + pictures + links

  • NED Synclavier

    1976-1992: NED ("New England Digital")
    1992: The Synclavier Owners Consortium
    1993-1995: The Synclavier Company
    1996: hardware intellectual properties sold to DEMAS Inc. ("Digital Equipment Maintenance And Support", since 1992, info), software intellectual properties sold to AirWorks Media Services Ltd., DEMAS obtains software license from AirWorks
    1998: Cameron Jones aquires software intellectual properties and Synclavier trademark from AirWorks Media Incorporated (successor of Airworks Media Services Ltd.) w/o the Synclavier Sound Library and the S/Link software package
    1999: formation of Synclavier Digital Corporation ( info, press release) by Cameron Jones et al.
    (The content of this web site reflects the personal opinion of the author, the information given is not guaranteed to have anything in common with the products of these companies. This web site is in no way affiliated with them. Please visit their web sites for information about their products. See also disclaimer.)

    Synclavier European Services (pictures, info, manuals)
    Yaking Cat Music Studios (lots of infos + pictures!)
    Synrise: history+infos, Synthony: info

    digital 8-Bit FM/additive synthesizer, later 16-Bit sampling and harddisk recording, subsystems (ABLE, FM, poly, ...) later in separate "bins" in system rack
    (see also Yamaha FM synthesizer section for comparison of FM technologies)

    Sound generator :
    FM/additive voice 8-Bit FM/additive voice is called "FM voice", 8 voices = one FM generator card plus 4 controller cards (SS1-SS5 card-set), later optional Stereo upgrade: 8 Stereo voices = two new FM generator cards plus 4 controller cards (and new software), max. 32 FM voices per system, 1 carrier + 1 modulator oscillator per voice, 8-Bit phase and waveform, pitch by variable clock and variable phase increment per oscillator, digital phase adder (the "FM" is actually implemented as a phase modulation, diagram), variable carrier waveform (32 "timbre frames") for additive synthesis, analog multiplying DACs for carrier output envelope, sound can consist of up to 4 FM voices ("layers") in parallel (one FM voice then called "partial timbre"), FM-voice also used for resynthesis of samples
    Sample-To-Disk ("STD", "ADX/DAX") 16-Bit monophonic sampling/sample-playback option directly to/from IMI harddisk ("International Memories Incorporated", ST506 MFM type), 1kHz-50kHz (0.1kHz steps) mono
    Sample-To-Memory ("STM", "Analogics") 16-Bit monophonic sampling option, max. 50kHz stereo/100kHz mono
    poly (sampling) voice 16-Bit polyphonic sample-playback voice is called "poly (sampling) voice", "PSV" (PS versions) and "DDV" (DTD and later) cards, max. 100kHz sample rate, phase-locked sample-playback, optionally Stereo and panning, separate multiplying DACs per voice (2 if Stereo) with analog envelope, ABLE Model C or D required, max. 32 poly voices per poly-bin, shared "poly RAM" sample memory (missing in disk-recorder versions), max. 32MB poly RAM (128MB with Model D) per poly-bin, max. 3 poly-bins, sample input via STM
    Multichannel Distributor ("MD") output router

    : 16-Bit "single-instruction" system control processor/minicomputer (proprietary non-microprocessor CPU), XPL programming language (later called "Scientific XPL"), Model A/B/C/D versions (poly voices since Model C) ( info, cards)

    ABLE software:
    RTP ("Real-Time Program", Synclavier real-time system)
    SFM (sound editing environment)
    Monitor (command line interpreter)
    Screen Editor
    ("SED", text editor)
    SCRIPT (music/score editing/printing language)
    "Memory Recorder" (sequencer, 32/200 tracks)

    (more infos on models, cards)

    prototype/research model (1975):

    * Darmouth Digital Synthesizer (digital FM/additive synthesizer, developed in 1975 at the Dartmouth College/New Hampshire by Sydney Alonso/Jon Appleton/Cameron Jones, ABLE processor)

    early NED models (>=1976)
    8-Bit FM voices, Model B or C ABLE

    * Synclavier Digital Synth (1977)

    * Synclavier II (1980)
    (links: to info, control architecture, synthesizer architecture, synthesizer schematics, main unit, "ORK" ("Original Release Keyboard") keyboard, terminal)
    Stereo FM option
    MIDI option
    Sampling option ("Sample-To-Disk", 1982)
    IMI ("International Memories Incorporated", ST506 MFM type) harddisk controller (later SCSI)

    PS models ("Synclavier", 1984?):
    "VPK" velocity sensitive keyboard, Model C or D ABLE, PSV 16-Bit poly voices, 8-Bit FM voices, Digital VT640 terminal
    (links: to picture)

    * PSST (16-Bit sampler, small tower)

    * PSMT (8-Bit FM/additive synthesizer + 16-Bit sampler, medium tower)

    * PST (8-Bit FM/additive synthesizer + 16-Bit sampler, large tower with 3 poly-bins)

    Harddisk recording (1986?):
    Model C or D ABLE, DDV 16-Bit poly voices, no poly RAM (sample playback directly from disk)

    * DTD ("Direct-to-Disk", digital recording/editing, option to Synclavier or stand-alone system, 16 tracks)
    (links: to info, specs, picture, picture)

    3200/6400/9600 models (>=1988?)
    successors of PS series, Apple Macintosh as control console (instead of the Digital VT terminal), Model D ABLE, DDV 16-Bit poly voices (like in DTD), optional 8-Bit FM voices, Multichannel Distributor ("MD", output router)

    * Synclavier 3200 (16-Bit sampler, 32 poly voices (Mono), 32MB poly RAM, small tower)
    (links: to info+picture)

    * Synclavier 6400 (like 3200 but Stereo, small tower)

    * Synclavier 9600 (8-Bit FM/additive synthesizer + 16-Bit sampler, 32-96 poly voices, up to 768MB poly RAM, large tower)
    (links: to info, specs, info+picture)

    Post-production packages:

    * Synclavier TS ("Tapeless Studio", PSMT + DTD)

    * PostPro (DTD standalone system + special console)
    (links: to info+picture)

    * PostPro SD ("Sound Design", integrated 6400 and DTD)

    DEMAS & Synclavier Digital Corporation (>=1999):

    ABLE subsystem (Model C or D) replaced by Apple PowerMac (PowerPC CPU) as control computer with ported Synclavier software
    (links: DEMAS info, www.earthlink/~yaking: info, brochure)

  • PODX on PDP-11/23 + DMX-1000
    DMX-1000: digital signal processor developed by Dean Wallraff at Digital Music Systems (see: Computer Music Journal 3(4), 44-49 (1979))
    POD/PODX: music software system by B. Truax (home, POD+PODX infos), first PODX version 1982 for PDP-11/23 + DMX-1000, since 1986 with real-time granular synthesis (B.Truax: GSX info)

  • Yamaha

    digital FM synthesizers:
    Note: In contrast to the NED Synclavier (1 modulator + 1 carrier per voice and variable carrier waveform), older Yamaha FM synthesizers (GS/CE/DX series) use various stack-like configurations (called "algorithms") with higher order stacks/trees of FM-modulating/modulated sine-wave oscillators (called "operators").
    (The content of this web site reflects the personal opinion of the author, the information given is not guaranteed to have anything in common with the products of Yamaha. This web site is in no way affiliated with Yamaha. Please visit their web site for information about their products. See also disclaimer.)

    links: info1,
    Markus Fiedler: FM (old link) Yamaha model history, timeline, DX1
    Dave Benson: DX7 Yamaha

    digital FM prototype
    see Markus Fiedler's infos and pictures

    GS1/GS2 (1981)
    preset digital FM synthesizer, 16 voices, memory for 16 sounds, later optional MIDI interface (GSMI-1/GSMI-2)
    links: info1, info2, info

    GS1: 8 operators per voice: 4 carriers with one modulator each (crossmodulation between 2 modulators), 88-key velocity and poly-pressure sensitive weighted keyboard ( picture)
    GS2: 4 operators per voice: 2 carriers with one modulator each (crossmodulation between 2 modulators), 73-key velocity sensitive weighted keyboard ( picture)

    The GS1 sound-generator is basically a "double-GS2" system with enhanced control/modulation possibilities (random detune modes and poly-pressure sensitive keyboard). Sounds were supplied by Yamaha on magnetic cards (the GS1/GS2 are not user-programmable). Each side of such a card contains sound data for 4 operators (thus, the GS1 needs two "sides" of sound data for its 8 operators, which can also be combined arbitrarily from different cards). Compared with later DX-series models, the GS1/GS2 provide only one "algorithm" (with crossmodulation).
    (Author's note: As for the offered features, the relationship between the GS1 and GS2 models is comparable to that of the DX1 and DX7 synthesizers. The GS1/GS2 design can be considered as the first LSI implementation of digital FM synthesis, building the basis for the following VLSI generation of DX series synthesizers and other "mass-market" FM-based keyboards by Yamaha.)


    The digital FM sound generator for 16 (mono-timbral) voices with 4 operators each (2 carriers with one modulator each, sine-waves) is implememented as the "FM" module PCB. The "FM" algorithm is actually implemented as a phase-modulation. Each modulator provides its waveform output as a modulation input to its carrier and also to the other modulator (with selectable cross-modulation mode). The FM module contains various proprietary LSI digital integrated circuits (5 per operator). Each operator is shared in a cyclic 16-phase round-robin scheme between its 16 mono-timbral channels with fixed sample output rate (370kHz/16=23kHz). The FM module also contains the 10-Bit(+sign Bit) audio output DAC (NEC uPD610D). The GS2 contains one FM module (4 operators), whereas the GS1 contains two FM modules in parallel (8 operators).
    The "KC" keyboard-scanner/controller module controls the FM module(s). It contains the master oscillator (at 3x370kHz) which is modulated by a vibrato LFO (on A module, see below). The KC modules also contains the key velocity controller ("inital touch"). In the GS1, additional key-pressure sensors and an additional "MPX" module decodes and generates poly-pressure ("after touch") control data, which is multiplexed with the velocity control data to the FM module. Detune effects are generated by use of dedicated detune/random/pitch control inputs of each individual operator (note: only the GS1 makes use of the random tune control inputs of the FM module). Optionally, key-code data can be redirected through a separate key-event generator module (e.g. a MIDI interface board).
    The "RW" sound memory module contains a 8035 (ROM-less 8048 variant) microcontroller with 2KB firmware ROM and 2KB/4KB sound SRAM for the GS2/GS1. It is connected to the voice registers on the FM board (via drivers on the KC board) and various control elements/switches. The sound SRAM (with battery-backup) contains 16 sounds (4x256Bits per sound with 4 operators). The card-reader (Canon K-15913-93), which can load/save sounds to/from the SRAM, is connected to this RW module. A programmer interface on the RW board provides the possibility to load/save sound data to the GS1/GS2 from an external programming device.
    Audio signals from the FM module(s) are post-processed on the "A" analog module, which contains an "ensemble" effect engine based on analog BBD chips (3x MN3009) and a "tremolo" effect circuit. Furthermore, the A module contains the LFOs for the tremolo and vibrato modulation effects, which are controlled by foot pedals and front panel elements. With the GS1, each FM module has its own audio output, connected to two audio inputs on the A module. The GS2 has the same A module, but here, both audio inputs are connected to the same FM module output.
    The GS1/GS2 has also a built-in three-band equalizer (on a PCB located on the front-panel) which is connected to the A module (between ensemble engine output and tremolo modulator input).

    Algorithm block diagram:

    Integrated circuits (LSI):
    KC (YM311) "Key Coder": 1x on KC board, key-switch input from keyboard, key-code output to CP
    CP (YM312) "Channel Processor": 1x on KC board, key-code input from KC, enevelope state input from EC, control outputs to IG/AG/MPX/VRG/EG/PG
    IG (YM320) "Initial Touch Generator": 1x on KC board, key-code data input from CP, initial touch (velocity) control output to EC
    MPX (YM318) "Multiplexer": 4x on MPX board, analog pressure control input from keyboard, key-code data input from CP, multiplexed pressure information to AG
    AG (YM334) "After Touch Generator": 1x on MPX board, multiplexed pressure information from MPXs, key-code data input from CP, after touch (pressure) control data output to EC
    VRG (YM347) "Voice Register": 4x on FM board, sound data register, serial sound data input from RW board (via KC board), key-code data input from CP (for scaling), current-state input from EG, rate and level information output to EG, frequency control output to PG
    EG (YM321) "Envelope Generator": 4x on FM board, 4-state envelope generator (attack, 1st decay, 2nd decay, release), rate and level information input from VRG, key-code data input from CP, current-state output to VRG, envelope output to EC
    EC (YM322) "Envelope Controller": 4x on FM board, envelope input from EG, modulation input from IG/AG/VRG, envelope output to OPC/OPM, envelope state output to CP
    PG (YM344) "Phase Generator": 4x on FM board, variable phase accumulator, frequency control input from VRG and modulation elements, phase output to OPC/OPM
    OPC (YM34501) "Operator-Carrier": 2x on FM board, phase adder + sine-ROM + envelope multiplier, phase input from PG/OPM, waveform output to ACC
    OPM (YM34502) "Operator-Modulator": 2x on FM board, phase adder + sine-ROM + envelope multiplier, phase input from PG/OPM, waveform output to OPC/OPM, selectable cross-modulation mode input from VRG
    ACC (YM316) "Accumulator": 2x on FM board, waveform data accumulator for 16 channels, waveform input from OPC, serial channel sum output to ADD
    ADD (YM327) "Adder": 1x on FM board, serial channel sum input from two ACCs, output to DAC
    SECII (YM633) "Symphonic Ensemble Controller": BBD controller, 1x on A board

    MIDI kits


    - synthesizer (fully mounted)
    - inside (A module to the left, FM+KC+RW modules in black metal housing)
    - FM module (1 per GS2 and 2 per GS1, KC module to the right)
    - KC module (FM module to the left, RW module to the right)
    - RW module (KC module to the left, programmer connector empty)
    - A module
    - magnetic sound card (with write-protection "edge")

    CE20/CE25 (1982)
    preset FM (home-)keyboard ( info, info)

    DX7 (1983)
    programmable digital FM synthesizer, 16 voices, 12-Bit, 28kHz, 6 operators/32 algorithms (Dave Benson: MIDI + schematics, info, info)

    The sound generator for 16 voices with 6 operators each is implemented as two intergrated circuits (VLSI):
    EGS (YM21290): digital envelope generators (pitch, amplitude), LFO
    OPS (YM21280): digital operator (phase accumulator, modulator, adders, sine-ROM)
    The operator output words (6 operators and 16 channels) are generated sequentially by use of a round-robin scheme, running at 16x6x the fixed sample output rate (with a 4-phase 9.42MHz/2 master oscillator). The routing of operator output data within the OPS chip in each round-robin time-slice is controlled by the selected algorithm. Frequency information (i.e. phase increment) has a resolution of 14 Bits, amplitude envelope data is 12 Bits. The "FM" is actually implemented as a phase modulation, i.e. a modulator operator output is added to the phase of the carrier operator (behind the carrier's pitch phase accumulator) and then fed into the sine-ROM. The sine-ROM output is scaled by the amplitude envelope data and the result (i.e. the operator output) is stored temporarily in a register. The OPS is followed by a 12-Bit DAC (BA9221) with a 4-Bit binary attenuator.
    The sound generator subsystem is controlled by a 63B03 CPU, whereas a 6805 CPU is used as a keyboard scanner.

    DX1 (1983)
    basically a "double-DX7" with enhanced control and programming possibilities ( techinfo)

    DX9 (1983)
    programmable digital FM synthesizer, 16 voices, 4 operators/8 algorithms

    TXn16 (n=1...8, 1984)
    typically sold as TX216 and TX816
    TF1: 16-voice module (DX7-like FM generator, 1-8 in frame)
    MIDI RACK: MIDI multiplexer + power supply module
    built into 19" rack-mountable frame for up to 8 TF1 modules
    (links: info, info+manuals)

    DX5 (1985)
    successor of DX1 ( info)

    TX7 (1985)
    keyboard-less DX7 MIDI sound generator ( info)

    DX21 (1985)
    successor of DX9, split- and layer-modes ( info

    DX27/DX100 (1985)
    low-price variants of DX21, DX100 is mini-keyboard version of DX27 ( info)

  • E-MU timeline
    See also the Emulator Archive for many useful infos and links.
    (The content of this web site reflects the personal opinion of the author, the information given is not guaranteed to have anything in common with the products of E-MU or Digidesign or Apple. This web site is in no way affiliated with them. Please visit their web sites for information about their products. See also disclaimer.)


    Emulator I (1981, 1982 mark 2)
    2/4/8 channels, mu-255 companded 8-Bit, 128KB RAM, Z80 CPU, "mark 2" with VCA analog post-processing
    Links: brochure, manual, info

    Emulator II (1984, 1985 II+)
    8 channels, DPCM mu-255 companded 8-Bit (i.e., the analog difference between the new input sample and the accumulated DAC output of the previous sample is converted via mu-255 to an 8-bit value), 512KB RAM (per bank), 2x Z80 CPUs (main and scanner), proprietary 74-TTL based processor (Emulator II "Microcontroller") for shared 8-channel address calculation (with forward/backward looping capabilities), pitch generation based on variable sample output rate via separate frequency dividers (from a 10MHz master clock) within the range of 12-66kHz with a rather poor resolution of 2-10 cents, fixed sample input rate 27.7...kHz (=10MHz/360), separate DAC (AM6072) and VCF/VCA analog post-processing (SSM2045) per channel, sample input ADC based on channel0 DAC with SAR, RS422 computer interface (shared with MIDI, used for control by Apple Macintosh), proprietary floppy format (low-level as well as filesystem layout, DS/DD 80 tracks with 0xE00 bytes per track and side).
    The Emulator II has a very unique sound quality due to its analog post-processing, the DPCM mu-255 companding, and the divider-based variable sample-rate principle. Sample editing can be accomplished by use of the "SoundDesigner for the Emulator II" (legacy software, originally by Digidesign) running on an Apple Macintosh computer (which can be attached to the RS422 interface).
    * Emulator II: 1 RAM bank (512KB), 2 floppies
    * Emulator II+: 2 RAM banks (512KB each)
    * Emulator II+HD: 1 harddisk and 1 floppy
    Links: story, brochure, schematics info
    Jeff Bergman: info

    For later models (e.g. the Emulator III or Emax) please see the Emulator Archive.

  • PPG
    PPG ("Palm Products Germany", 1974-1987, Hamburg/Germany)
    Hermann Seib's PPG page: infos
    antarcticamedia info,
    history info,
    Till Kopper's infos

    wavetable synthesizers with analog post-processing:
    (Hermann Seib: documentation/manuals):

    Wave Computer 360 (1978)
    A/B models: 4/8-voice, no VCFs
    ( info, info)

    Wave 2 (1981)
    8-Bit, 8 voices, 1 DCO per voice (variable sample rate), VCFs/VCAs
    ( info)

    Wave 2.2 (1982)
    8-Bit, 8 voices, 2 DCOs per voice (phase accumulator with fixed sample rate), 2-way multi-timbral, for use with Waveterm A
    (Hermann Seib: MIDI-upgrade)

    Waveterm A
    waveform editor/sampling option, for Wave 2.2, 8-Bit, additive synthesis/resynthesis, based on ELTEC Eurocom II V7 computer with 6809 CPU and FLEX9 OS, see mainpage for more infos about FLEX 9
    (Hermann Seib: info, info)

    Wave 2.3 (1984)
    12-Bit, 8-way multi-timbral, MIDI, for use with Waveterm B

    EVU ("Expansion Voice Unit", rackmount Wave 2.3
    (Hermann Seib: info, , MIDI-upgrade, info)

    Waveterm B (improved Waveterm, for Wave 2.3, 16-Bit, 6809+68000 CPU)

    Wave 2.3 Principle:
    Wavetable synthesizer with "drop/add" fixed sample output rate and analog post-processing.
    There is a total of 8 voices, each consisting of 2 digital oscillators, which gives a total of 16 oscillators. Oscillator sample address calculation is done via round-robin scheduling at a fixed sample output rate. The total waveform RAM has 64K 12-Bit words, consisting of 8 wavetables (in order to be multi-timbral), where a wavetable consists of 64 partial waves of 128 samples each. (Note: Once a new sound is selected, the corresponding wavetable is downloaded into the waveform RAM from a large wavetable-EPROM or an external Waveterm.) A phase accumulator at a fixed sample output rate with pitch variation by variable phase increment per oscillator is employed. The phase accumulator is implemented as a fixed-point 29-Bit register/adder with a 13-Bit fractional part and 16-Bit integer part. In the integer part, the lower 7 Bits address the sample within the current partial wave (128 samples) and the upper 9 Bits provide the partial wave number (8x64 partial waves). However, the phase increment is only 15-Bits (13 fractional and 2 lower integer Bits). Each oscillator has its own phase, partial wave number and phase increment register (contained within multiple 74LS189 SRAMs) with a phase incrementer logic (implemented via multiple 74LS283 adders and 74LS379 registers), shared among the oscillators in a cyclic round-robin fashion. The current output sample value is simply obtained by an add/drop sample output algorithm (i.e., given by the waveform sample at the integer part of the phase accumulator as the waveform RAM address). Waveform RAM and processing is located on the "PROZ. Board", also called the "Sound Computer". Each voice (2 oscillators) has a separate DAC and a VCF/VCA analog post-processing section, located on two "Voice Boards" "OF4" with 4 voices each. The OF4 contains the four 12-Bit sample output DACs (4x AD7545KN) and four VCFs with variable resonance (4x SSM2044) and four VCAs (2x CEM3360). The "I/O Board" contains the main controlling components, such as the 6809 CPU, main RAM, OS- and wavetable-EPROMs, keyboard and panel interfaces, parts of the MIDI interface and the control-voltage ADC (ZN427E-8) and DACs (3x AD558). The three control-voltages (for the VCFs/VCAs on the OF4) are demultiplexed (and sampled by following S/H amplifiers) on the OF4 to four voices. The "PPG COM. BUS" is directly connected to the "PROZ. Board".

    Realizer (prototype, sampler/wavetable/FM/analog model digital synthesizer, 8x TI TMS32010 DSPs, 1x Motorola 68020 CPU, 4MB RAM, 12x 16-Bit DACs, 2x 16-Bit ADCs, SCSI, AES/EBU, 44.1kHz, 1986)
    ( info, info)

    sucessors: see Waldorf: Microwave, Wave

  • Waldorf

    Microwave (1989)
    rack-mount wavetable synthesizer with digital sound generator ASIC, MP7545 DACs, and analog VCF/VCA post-processing (Rev A: CEM3389, Rev B: CEM3387), 68000 host CPU, based on the PPG Wave 2.2 synthesizer

    Till Kopper: Microwave Wave

  • Akai

    samplers: info, info, history

    S612 (12-Bit, 128KB, 6 voices, max. 32kHz sampling, 1985, info+manual)
    S700 (12-Bit, 6 voices)
    S900 (12-Bit, 750KB, 8 voices, 7.5-40kHz sampling, 1986, info)
    S950 (750KB-2.25MB, 7.5-48kHz sampling)
    S1000 (16-Bit, 2-32MB, 8 voices, 44.1/22.05 kHz sampling, 1988, info+manual)
    S1100 (16 voices, 1990)

  • Ensoniq
    ( infos, timeline)


    Mirage DSK-8 (1984, 8-Bit, 144KB, 10-33kHz, VCA+VCF, Chips: "Q-Chip", 6809, Electric Enver: infos)

    Mirage DSM-8 (1985, rack version)
    Mirage DSK-1 (1987)
    EPS (1988, 16-Bit (13-Bit ADC/DAC), 2MB, 6.25-52kHz, Chips: "OTTO", "ESP")
    EPS 16 Plus (1990, 16-Bit, 2MB, 11.2-44.6kHz, Chips: "OTTO", "ESP")
    ASR-10 (1992, 16MB, stereo sampling, Chips: "OTTOR2", "ESP", 68302)
    ASR-10R (1993, rack version)
    ASR-88 (1995)
    ASR-X (1997, Chips: "OTTOR2"?, "ESP2")


    ESQ-1 (1986, VCA + VCF, Chips: "Q-Chip")
    ESQ-M (1987, rack version)
    SQ-80 (1988, Chips: "Q-Chip")
    VFX, VFX-SD (1989, Chips: "OTTO", "ESP")
    SQ-1, SQ-R (1990)
    SQ-1 Plus (1991)
    SQ-2 (1991)
    SD-1 (1991, Chips: "OTTOR2")
    KS-32 (1992)
    TS-10, TS-12 (1993)
    KT-76, KT-88 (1994)
    MR (1995, rack version, Chips: "OTTOR2"?, "ESP2")
    MR-61, MR-76 (1996)
    ZR-76 (1997)

    "SID" MOS6581: "Sound Interface Device" soundchip (by Bob Yannes, not Ensoniq at that time) for Commodore C64, DCO/DCF/DCA design
    "Q-Chip" "DOC-I" ES-5503: "Digital Oscillator Chip", 32 oscillators, 8-Bit
    "DOC-II" ???
    "OTTO" "DOC-III" ES-5505: 32 oscillators, 16-Bit
    "OTTOR2" "DOC-IV" ES-5506: 32 oscillators, 16-Bit
    "ESP" ES-5510: "Ensoniq Signal Processor", DSP, 24-Bit
    "ESP-2" ES-55??: DSP, 24-Bit
    "OTTOFX" ES-5540: integrated OTTO+ESP
    "OPUS" ES-5530: multimedia soundchip
    "SGLU" ES-5701: system glue logic chip
    Note: DOC wavetable playback chip is based on phase-accumulator principle running at fixed sample output rate with round-robin timesharing among voices

    ASR-10/EPS-16Plus file system manager and WAV-to-INST converter: asr10 (Rev.1.0) source, binary

    Please see disclaimer for downloads !
    Also try this to find more software on the KMI site.

  • KMI: 32-Bit floating-point Spectral Morphing Synthesizer (SHARC/MIPS based, 12-voice DSP-board prototype: 1/2001 version 209.5.3)
    based on a fixed sample output rate design with various interpolation schemes
  Analog and Hybrid synthesizers:


  Drum machines:
  • Linn:

    ( LM-1)

    (links: picture, samples, Forat: service manual, Forat MIDI Mod)
    8-Bit mu-255 companded waveform data format, pitch by variable sample output rate (controlled by VCOs, VCFs/VCAs for post-processing)), Z80A CPU, 8KB ROM, 8KB RAM,
    sound generators based on VCO+counter+ROM+DAC(+VCF+VCA)

    3 separate sound-generator systems:
    a) TOM/CGA

    Z80 address-space (periodic in 4000, except Forat MIDI Mod blocks):
    0000-1F7F: ROM (8KB)
    1F80-1FBF: INPORT (read, input ports), STROBE (write, see below)
    1FC0-1FFF: KEYBD (read, keys)
    2000-3FFF: RAM (8KB, battery-buffered)
    4000-5F7F: (Forat MIDI Mod)
    8000-9F7F: (Forat MIDI Mod)
    (Forat MIDI Mod)
    STROBE addresses:
    1F80: 2-digit LED display (PATTERN#)
    1F81: 2-digit LED display (STEP#/%MEM)
    1F82: LEDs
    1F83: LEDs
    1F84: LEDs, BEEP, output ports (TAPE SYNC, TRIGGER, CASSETTE)
    1F85: BASS
    1F86: SNARE/SSTK
    1F87: HI-HAT
    1F88: TOM/CGA
    1F89: RIDE
    1F8A: CRASH
    1F8B: CABASA
    1F8C: TAMB
    1F8E: CLAP
    1F8F: CLICK

    Linn 9000 Linn 9000

    (based on Linn 9000) by Forat
    Forat: Forat9000

  • Akai MPC-60

  • Roland
    TR808 ( samples, schematics)
    TR909 ( samples)

  • E-MU (Emulator Archive)
    Drumulator (8-Bit, 64KB waveform memory, shared DAC, 1983, info)

    SP-12 SP-12 (12-Bit @26.667kHz)

    SP-1200 SP-1200

  • Oberheim

  • Simmons info
  Effect engines:
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