Analog Synthesizer Research Articles

Review: Switched-On Bach, by Roshanak Kheshti, and Wendy Carlos: A Biography, by Amanda Sewel

Review: <i>Switched-On Bach</i>, by Roshanak Kheshti, and <i>Wendy Carlos: A Biography</i>, by Amanda Sewel

Composition-Interpretation-Improvisation Processes in XO pt. II by Dino Rešidbegović: Comparative Analysis of two Interpretations

The composition XO pt. II for flute with processors, amplified cello, amplified piano and analog sound synthesizers, by Dino Rešidbegović, was published in 2016 as an electroacoustic adaptation of its previous version, XO for undetermined instrumentation. The phenomenon of the notation of both versions, presented through the graphic score, comes from the very idea of/about the composition. The score is divided into nine squares which form a network with incorporated abstract shapes of different colors. In the middle of the score is a chart of a spectral analysis of the composition The Impact ofthe Analog Synthesizer for ensemble, also by Rešidbegović. The only difference between the scores of the two versions is the instrumental parameters table involved in XO pt. II for three performers. The main task of this paper is primarily the analysis and comparison of two interpretations of XO pt. II by the same three performers, with a difference in duration of almost 19 minutes. The goal is to reconstruct and explicate the improvisation as a potential basis for a compositional-technical analysis. A composition-interpretation-improvisation processes chain represents the main structure of the general process which forms the composition XO pt.II, involving composer and performers – interpreted as equal, interdependent creators of the musical work.

Explicit Fixed-Point Computation of Nonlinear Delay-Free Loop Filter Networks

An iterative method is proposed for the explicit computation of discrete-time nonlinear filter networks containing delay-free loops. The method relies on a fixed-point search of the signal values at every temporal step. The formal as well as numerical properties of fixed-point solvers delimit its applicability: On the one hand, the method allows for a reliable prediction of the frequency rates where the simulation is stable, while, on the other hand, its straightforward applicability is counterbalanced by low speed of convergence. Especially in presence of specific nonlinear characteristics, the use of a fixed-point search is limited if the real-time constraint holds. For this reason, the method becomes useful especially during the digital model prototyping stage, as exemplified while revisiting a previous discrete-time realization of the voltage-controlled filter aboard the EMS VCS3 analog synthesizer. Further tests conducted on a digital ring modulator model support the above considerations.

Design and Implementation of Feasible Direct Digital Synthesizer to Eliminate Manual Tweaking

This paper presents design and simulation of the programming model of optimal and feasible Direct Digital Synthesizer that eliminates the need for the manual tuning and tweaking related to component aging and temperature drift in analog synthesizer solutions. A Direct Digital Synthesizer is a measure part o Digital Down Converter where the DDC (Digital Down converter) has become a cornerstone technology in communication systems. Digital Down Converter (DDC) is key component of RF systems in communications, sensing, and imaging. This paper also evaluates the performance of DDS under various programming parameters and finally performs the realization of DDS using Virtex II Pro.

Hands-On Parameter Search for Neural Simulations by a MIDI-Controller

Computational neuroscientists frequently encounter the challenge of parameter fitting – exploring a usually high dimensional variable space to find a parameter set that reproduces an experimental data set. One common approach is using automated search algorithms such as gradient descent or genetic algorithms. However, these approaches suffer several shortcomings related to their lack of understanding the underlying question, such as defining a suitable error function or getting stuck in local minima. Another widespread approach is manual parameter fitting using a keyboard or a mouse, evaluating different parameter sets following the users intuition. However, this process is often cumbersome and time-intensive. Here, we present a new method for manual parameter fitting. A MIDI controller provides input to the simulation software, where model parameters are then tuned according to the knob and slider positions on the device. The model is immediately updated on every parameter change, continuously plotting the latest results. Given reasonably short simulation times of less than one second, we find this method to be highly efficient in quickly determining good parameter sets. Our approach bears a close resemblance to tuning the sound of an analog synthesizer, giving the user a very good intuition of the problem at hand, such as immediate feedback if and how results are affected by specific parameter changes. In addition to be used in research, our approach should be an ideal teaching tool, allowing students to interactively explore complex models such as Hodgkin-Huxley or dynamical systems.

레이더용 주파수합성기의 위상잡음 모델링 및 분석

In this paper, we proposed a phase noise model of a frequency synthesizer for a radar system. Especially, it was proposed a phase noise model in a DAS(Direct Analog Synthesizer) and a frequency up converter system using Leeson's model. The proposed phase noise model was derived from the measurement data of model 1 and evaluated by adapting to model 2 and model 3 frequency synthesizers. The prediction phase noise by modeling was totally matched to the measured data and the effective analysis of the phase noise was done in a frequency synthesizer and a frequency converter of radar system.

Modeling of the EMS VCS3 Voltage-Controlled Filter as a Nonlinear Filter Network

A previously developed nonlinear differential equation system, representative of the diode-based voltage-controlled filter on board of the EMS VCS3 analog synthesizer, can be modeled in terms of a filter network. The properties of this network can be converted block-by-block into a discrete-time version of the same model, in which the relationships existing between the system variables take the form of delay-free loops. In the presence of such loops, the discrete-time filter network is solved by iterating over a fixed-point scheme. Simulations running at 176.4 kHz show reasonable fidelity of the model in the regimes of interest, as well as shorter computation time compared to previous implementations of the same system. While reducing aliasing, such a sampling rate optimizes also the computation time as it requires fewer fixed-point iterations. The new model represents an improved alternative to previously developed voltage-controlled filter designs in the field of virtual analog.

Analog Days: The Invention and Impact of the Moog Synthesizer (review)

In the history of technology and in science and technology studies, "sound studies" have recently enjoyed some popularity. Hence a study on Robert Moog and the analog synthesizer is particularly welcome. Trevor Pinch, who, as a physics student, built his own synthesizer in the early 1970s, brings some biographical interest to the story. He combines this with his well-known conceptual and analytical expertise as well as his ability to do sound empirical research and to present the outcome in a way that is pleasurable to read.

BitHeadz Unity DS-1 Software Sampler and Retro AS-1 Software Synthesizer

BitHeadz's Unity DS-1 and Retro AS1 put an analog hardware sampler and analog synthesizer, respectively, as software onto Windows or Macintosh platforms, and allow them to integrate with packages like Digital Performer, Digidesign ProTools, Emagic Logic, and Steinberg Cubase. Using both BitHeadz packages in the studios at Waikato University in various configurations and versions and under various models of Macintosh for some time has afforded a sense of their practical implementation. The motivation for initially using these packages was the cost when compared to buying further hardware modules. The packages assume a traditional way of thinking about sound generation, in contrast to more flexible contemporary software packages. It is then most fitting to judge the DS-1 and AS-i in comparison to hardware units.

Analog synthesizer of orthogonal signals

A new adaptive architecture, an analog synthesizer of orthogonal signals which is able to produce a plurality of mutually orthogonal signals as functions of time, such as Legendre polynomials, cosine basis of functions, smoothed-cosine basis, etc., is proposed and analyzed. A method for the creation of analog devices with Chebyshev polynomials and Legendre polynomial transfer functions is proposed. A proof-of-concept breadboard version of the analog synthesizer is described. The results of smoothed-cosine basis synthesis, as well as characteristics of output functions, are then presented. The device is characterized by a very fast and stable process of signals synthesis.

A computationally efficient articulatory synthesizer

In this study, a computationally efficient articulatory synthesizer that utilizes the popular analog circuit simulator SPICE is developed. The synthesizer uses a transmission‐line analog model of the vocal tract. An analog model has many advantages over digital representations: (1) Side branches (needed for modeling nasals and /l/) can be simulated easily by additional transmission lines in parallel; (2) drive‐dependent sources, at any location, could be added; and (3) the number of sections can be varied without changing the sampling rate, as is the case with a digital synthesizer. A computer interface, using MATLAB, is developed such that the input to the synthesizer can be specified in terms of the area function of the vocal tract and the type and location of dependent or independent sources (voltage or current.) By simulating the transfer function of the vocal tract, transient and steady‐state responses are generated. Using Fant’s vowel area functions (1960), vowels were synthesized with their first four formant frequencies almost identical to those given by Fant. The feasibility of implementing the analog synthesizer using modern ICs, such as the gyrator‐based inductance simulator and switched capacitor filter circuits, is assessed.

Control of Klatt speech synthesizer with high-level parameters.

This paper describes a system for synthesis of speech with a terminal analog synthesizer like the Klatt synthesizer. The control of the synthesizer is accomplished with a small number (about 10) of high-level (HL) parameters that capture directly certain attributes of the state of the vocal tract during speech production. The multiplicity of low-level parameters in the synthesizer are derived from the HL parameters through a set of mapping relations. The HL parameters consist of (1) the fundamental frequency; (2) a specification of the vocal tract in terms of the first three or four natural frequencies of the tract assuming no opening to the glottis or the nose; (3) the areas of three major orifices in the vocal tract: the glottal opening, the velopharyngeal opening, and (for consonants) the principal constriction in the vocal tract; (4) a measure of active expansion or contraction of the vocal-tract volume for obstruents; (5) a measure of the efficiency of turbulence noise generation for obstruents. The mapping relations for obstruents involve the calculation of airflows, pressures, wall displacements, glottal waveform characteristics, vocal-tract losses, and noise due to turbulence. Some examples of the mapping relations for both obstruents and sonorants are given. [Research supported by the National Institutes of Health.]

The Radio Baton and Conductor Program, or: Pitch, the Most Important and Least Expressive Part of Music

In the late 1950s, the first computer-synthesized music (Mathews 1961) required many minutes to compute a second of sound. Computation time was greater than Performance was not possible. In order to perform, a performer must be able to hear the sound he is making in real time as he is producing it. He must be able to modify the sound depending on what his ear tells him. The performer's interpretation and modifications of the notes specified in the score add much meaning and live quality to music. We call these changes performance nuance. The impossibility of performance and the absence of performance nuance in early computer music was much missed by some of us. Today, computers and digital synthesizers are fast enough to play powerful and complex music in real time. Performance is possible. One of the first performance systems was Groove (Mathews and Moore 1970). Groove was a hybrid system in which a digital computer controlled an analog synthesizer. Although Groove was limited by its synthesizer and by its large, unwieldy control computer, many of the ideas developed in the conductor program (Mathews 1989a) were first demonstrated there. Although expressive performance with synthesizers has progressed rapidly in the domain of popular music, it has not been nearly so successful in other musical areas. Part of the problem may simply be that a cadre of highly trained synthesizer performers does not yet exist, but in addition, the richness of control possibilities in the synthesizer may exceed the capacities of human performers. We now believe that programs to facilitate expressive performance can be written. One class of such programs, of which the conductor program is an example, allows the performer to concentrate all his attention on the expressive aspects of music and relieves him of data processing and motor control tasks which are very demanding with traditional instruments. The conductor program can make the many hours of practice most performers have to put in to learn technique unnecessary, but these new possibilities raise some fundamental and unanswered questions about performance training. Specifically, do performers learn other crucial musical skills and sensitivities while they practice to achieve adequate technique, and if so can we teach these skills in some other way? The conductor program and its associated device, the radio baton, have been described elsewhere (Lawson and Mathews 1977; Mathews 1989b), so in this article we will only briefly review their characteristics. Mainly we wish to consider the possibilities, problems, and questions raised by programs designed to aid performers in achieving expressive performances. Our most interesting insights concern the possibility of dividing traditional music into a predetermined part and an expressive part. This division is the basis for the conductor program. Since computer-aided performance is very new, and performers require years or decades to learn their art, most of the questions we will raise can only be answered by future developments. A great part of this article is speculation. Computer Music Journal, Vol. 15, No. 4, Winter 1991, ? 1991 Massachusetts Institute of Technology.

Speech synthesis of Japanese from written Esperanto

With a subsystem designed to extract meaning from written Esperanto being added to the speech synthesis system of Japanese generated in the semantic/syntactic base [S. Sato and H. Kasuya, J. Acoust. Soc. Am. Suppl. 1 73, S5 (1983)], we present an overview of this translation‐synthesis system including a knowledge base editor for implementing linguistic information needed. Allowing a wide range of data modification and file manipulation operations, the frame editor can edit on a frame/slot basis the rules and dictionaries used in parsing Esperanto and generating Japanese. The main portion of the knowledge base is occupied with descriptions about case structures, selectional restrictions, hierarchial relations among items, etc. for detecting anomaly and ambiguity of a given Esperanto sentence. The semantics gives to the meaning a case frame expression, from which the sentence generator produces a surface form with some trimming with phonological and accentuation rules. The synthesis program further converts the surface into acoustic parameters, rendering a time series of control parameters for the terminal analog synthesizer.

The evolution of real‐time music synthesizers

A broad history of real‐time synthesizers will be traced. One root goes back to electronic tape studios pioneered in the early 1950's by Ussachevsky and Luening in the United States and by Schaeffer and Stockhausen in Europe. The RCA synthesizer (Olson, Belar), modular synthesizers (Moog), keyboard‐controlled synthesizers, and sequencer‐controlled synthesizers were notable developments along this path. Another root goes back to pure digital synthesis in the late 1950's (Mathews, Chowning). Developments include the computer‐controlled analog synthesizer (Mathews, Moore) and the digital synthesizer (Alonzo, Appleton). Today realtime synthesizers can be purchased at prices ranging from less than $100 for toys to more than $100,000. In the future we expect close association of synthesizers and personal computers plus special VLSI music chips which will greatly reduce costs and strengthen the power of the synthesizers.

Musical electromagnetic analog synthesizer controlled rocket engine

A musical instrument combination including a rocket engine type combustion chamber and a system for firing it either by single drum beat control or by variable frequency control pulses. The effect achieved by the rocket engine output sound is similar to that of a pipe organ and resonator and further is of a pitch controllable by the barrel length and diameter of the rocket engine chamber exhaust structure.

Vocal tract analog synthesizer with lumped losses

The vocal tract analog synthesizer generally requires a lot of calculation when a real signal is to be obtained. Thus, when you want to undertake studies, a real time synthesizer is more convenient. One way to carry out sound wave propagation modeling in the vocal tract is to consider the Kelly-Lochbaum model. In this case, we have to introduce a damping coefficient at the junction of the two elementary tubes. In order to decrease the computation size, we propose to concentrate the losses at three places along the vocal tract. These places are chosen so that the end characteristics (glottis, radiation) will not be disturbed. If there are about 20 elementary cells, the losses are places between cells number 2 and 3, 10 and 11, 14 and 15 (number 1 is next to the glottis). The simulation of this model, including the wall vibration and lip radiation according to the principle proposed by D. Degryse (The 4th F.A.S.E. symposium, Venezia, 193–196 (1981)], gives formant characteristics close to those of the distributed model: the maximum error is less than 1% on the first three formant frequencies and 20% on the respective bandwith. Then, the synthesizer requires about 25% fewer computations than without the improvement.

Representation of whispered vowels in temporal patterns of auditory-nerve fiber discharges

We have previously shown that the temporal patterns of auditory-nerve fiber discharges provide a robust representation of the spectra of periodic steady-state vowels [Young and Sachs, J. Acoust. Soc. Am. 66, 1381–1403, (1979)]. The purpose of the study described here was to determine whether the spectra of a whispered (aperiodic) vowel could be represented by such a temporal code. A terminal analog synthesizer was excited by white noise. The first three formant frequencies of the synthesizer were set to 530, 1840, and 2480 Hz which correspond to the vowel /ε/. The resulting whispered vowel was presented in 400 ms bursts once per second. Responses to the vowel were recorded in as many as 350 auditory-nerve fibers in individual cats. Interval histograms of fiber responses were constructed from the data and discrete Fourier transforms with a frequency resolution of 64 HZ were computed for each interval histogram. For each fiber the magnitude of the Fourier transform component nearest its CF was used as a measure of the fiber's temporal response to the vowel. Plots of this temporal response versus CF across the population of fibers in individual cats showed peaks at the vowel formant frequencies. Indeed, such temporal profiles provide a strikingly accurate representation of the vowel spectrum. [Supported by NIH Grant No. NS-12112.]

Timbre of Kiais: Analysis and synthesis

Kiai is a name given to sudden and violent vocalizations practised by Martial Artists. An investigation to determine formant frequencies, bandwidths and source‐spectrum giope was performed on vocalizations made by five high ranking Karateka. Spectral classifications were made using adaptations of those used for infant cries [H. M. Truby and J. Lind Act, Paed. Scand. 163, 11 (1965)]. All cepstra show strong peaks regularly spaced at pitch‐period intervals. This phenomenon is usually linked with irregularities in the pitch‐pulse timing [A.M. Noll, J. Acoust. Soc. Am. 41, 293–309 (1967)]. However digital simulations using a Terminal Analog Synthesizer indicate that it is likely that these peaks are due to an opening quotient of the vocal cords which is less than 10%. This leads to a nearly flat source spectrum. The fundamental frequency range was found to be in the order of 450–500 Hz and high amplitude resonances were found up to 6000 Hz for all vocalizations studied. Physiological considerations concerning the production and perception of these screams will be discussed and shown to be consistent with claims that Kiais have a characteristic timbre, which this analysis supports. [Work supported by U. M. E. R. G.]

Hybrid sound synthesis

Most general purpose digital computers are not fast enough to directly compute musically sophisticated sound waves in real time. A hybrid system in which a computer controls an analog synthesizer can generate interesting sounds in real time. The computer supplies three main services. It senses the gestures of the performer. It provides a memory for storing these gestures as time functions. It computes the time functions which control the synthesizer as a complex combination of the real time gestures of the performer and the time functions in memory. The hybrid system is a good instrument with which to study the real time interactions between a performer, his instrument, and the sound he is producing. At present, the precision, quality, and range of timbres which can be obtained from a hybrid system are limited by the analog synthesizer. Shortly these limitations will be removed by replacing the analog device by a special purpose digital synthesizer.