Electric circuits containing vacuum tubes form an integral part of various audio equipment, such as guitar amplifiers, certain equalizers, microphone preamplifiers, and dynamic range compressors. Although most audio signal processing operations are straightforward to implement with modern computers, real-time digital simulation of vacuum tubes poses a significant challenge due to the dynamic nonlinearities of the tube circuits. Most of the current vacuum-tube emulators model the unidirectional signal path of the circuit using linear filters and nonlinear waveshapers, possibly with signal-dependent parameters. This paper introduces a physical model of a tube stage circuit using wave digital filters. In contrast to previous unidirectional signal models, the wave digital model implements bidirectional signal propagation. This allows realistic simulation of interesting dynamical nonlinearities, such as the bias variation under reactive load. The new model is an extension of the wave digital tube presented in ¿Wave digital simulation of a vacuum-tube amplifier¿ (M. Karjalainen and J. Pakarinen Proc. Int. Conf. Acoustics, Speech, Signal Processing, 2006, vol. V, pp. 153-156). In particular, the enhanced model for the tube grid-to-cathode connection enables the simulation of interstage coupling and blocking distortion.

Wave digital filters (WDFs) were originally developed for robust discrete-time simulation of analog filters, but recently they have been applied successfully to modeling of physical systems such as musical instruments and to model-based sound synthesis. While basic WDF elements are sufficient to implement arbitrary passive lumped-element models, the computational efficiency of such models is not optimal. In this paper, we explore methods to realize consolidated impedance or admittance wave ports that are compatible to WDFs and digital waveguides. In addition to efficiency, attention is paid to simplicity of parametric control. A modeling and sound synthesis case study of the bell is presented to demonstrate the performance obtained by the consolidated approach.

Virtual analog modeling is needed when simulating classic analog circuitry by DSP, for example when emulating analog sound synthesis and sound reproduction systems. In this paper we show how wave digital filters (WDFs) can be applied to efficient real-time simulation of vacuum-tube amplifier stages, typical in professional guitar amplifiers, which pose nonlinear behavior for desired distortion effects

Time-varying fractional delays are applied, for example, in physics based modeling of musical instruments, particularly for string and wind instruments. While Lagrange interpolation and allpass filters are used routinely in such sound synthesis models, they are found somewhat problematic, for example, in plucked string simulation when the length of the string is varied due to glissando or vibrato. There can be problems with signal energy levels and aliasing. We study two variable delay filter designs that have a physically realistic energetic behavior and keep undesirable side effects, such as aliasing, in control. The first one is sliding termination point simulation with energy correction and the second one is based on controllable wave digital filter delay lines.

A prerequisite in many systems for virtual and augmented reality audio is the tracking of a subject's head position and direction. When the subject is wearing binaural headset microphones, the signals from them can be cross-correlated with known sound sources, called here anchor sources, to obtain an accurate estimate of the subject's position and orientation. In this paper we propose a method where the anchor sources radiate in separate frequency bands but with common modulation signal. After demodulation, distance estimates between anchor sources and binaural microphones are obtained from positions of cross-correlation maxima, which further yield the desired head coordinates. A particularly attractive case is to use high carrier frequencies where disturbing environment noise is lower and sensitivity of hearing to detect annoying anchor sources is also lower.

In this paper, we describe how different the physics-based modeling paradigms can be in making a sound synthesis of musical instruments. We also discuss different ways to construct user interfaces for interactive virtual reality applications. For multi-paradigm modeling, we have investigated how digital waveguides (DWGs), finite difference time domain schemes (DFTDs), wave digital filters (WDFs), modal decomposition models, and source-filter models interrelate and mix together. We have investigated both the user interface control of these models in a cave-like virtual room using special hand-held controllers. Modeling examples of a 'virtual air guitar" and a virtual bell-xylophone are described and demonstrated.

This paper describes a systematic approach to specification and compilation of different physical modeling schemes particularly for sound synthesis studies. First we formulate theoretically a unified way of constructing physical interaction models which include elements that use both wave variables and Kirchhoff variables. These elements can be applied to build 1D and multidimensional structures as well as lumped element models. In addition, typical signal processing algorithms are supported. A software environment (Block Compiler) has been developed, allowing for high-level object-based specification of physical models and their compilation to efficient code for execution.

TheKelly-Lochbaumtransmission-line model of the vocaltract started the discrete-time modeling of speech production. More recently similar techniques have been developed in computer music towards a more generalized methodology. In this paper we will study the application of mixed physical modeling to speech production and speech synthesis. These approachesare Digital Waveguides (DWG), Finite Difference Time-Domain schemes (FDTD), and Wave Digital Filters (WDF). The equivalenceandinterconnectivityof theseschemesisshownandflexible real-time synthesizers for articulatory type of speech production are demonstrated.

A discrete-time realization of first-order (shelving) and second-order equalization filters is developed, providing bandpass/bandstop magnitude-complementary transfer functions. The bandpass transfer function is turned into a complementary one, and vice versa, switching between two structures that share a common allpass section containing the state variables of the equalizer. The present realization is an alternative to existing solutions, particularly in applications where the equalization parameters are dynamically varied.

Digital waveguide modeling of one-dimensional vibrating structures in musical instruments is developed further in this paper based on finite difference time domain (FDTD) formulations. It is shown that such realizations can be developed, efficient enough to run in real time on modem processors and more flexible than the computationally less expensive commuted synthesis. FDTD waveguide structures are analyzed with lossless and lossy propagation, input and output ports, terminations and scattering junctions. The formation of static initial state of displacement wave from pointwise excitation in plucked string models is demonstrated, not possible with simple waveguides. Related sound synthesis examples can be found at www.acoustics.hut.fi/demos/fd1.

Frequency warping using all pass structures or Laguerre filters has increasingly found applications in audio signal processing due to having a good match with auditory frequency resolution. Kautz filters are an extension where the frequency warping and related resolution can have more freedom. We discuss the properties of Kautz filters and how they meet typical requirements found in modeling and equalization of audio systems. Case studies include transfer function modeling of the guitar body and loudspeaker response equalization.

Collections of annotated spoken language have formed an important basis for the development of speech technology. Their existence has promoted speech analysis research as well as enabled robust synthesis and recognition methods to be developed. However, many complex relationships remain unspecified within a corpus due to a lack of meta-data that describes the raw information in sufficient detail as well as the interrelationships between signals, recording conditions, talkers, etc. A deficit of standards and formats, needed t o express complex relationships, has also hindered the potential use and value of available corpora. This paper presents a novel three-dimensional model for exploring temporal as well as atemporal information existing i n speech corpora. Examined are the potential benefits that are gained through corpus visualisation during the phases of creation, editing, verification, use, and exploration. The paper suggests that by providing a three-dimensional model of speech data, more of the inherent and potential value of a corpus can be utilised.

A lumped physical model of the glottal source is presented. Vocal folds are described as single masses, but vertical phase differences between upper and lower margins of the folds are taken into account by appropriately describing the non-linear interaction of the mechanical model with aerodynamics. This results in a modified one-mass model, or a “one-delayed-mass model”. Analysis on numerical simulations shows that the system behaves qualitatively as higher-dimensional models (such as the two-mass model by Ishizaka and Flanagan); in particular, control over flow skewness is guaranteed, allowing for synthesis of realistic glottal flow waveforms. As only one degree of freedom (one mass) is needed in the model, structure and number of parameters are drastically reduced, thus making it suitable for real-time synthesis applications.

In this paper we discuss various digital filter principles as models for synthetic speech generation. Warped linear prediction (WLP) and frequency-warped filters have been introduced earlier as a method to reduce the filter order in high-quality wideband speech synthesis. In addition to analyzing WLP and frequency-warped filters we introduce new related structures and techniques for arbitrary frequency resolution allocation. Kautz filters can be considered as generalized structures for pole-zero modeling. This study focuses on residual-excited synthesis and diphone-oriented reconstruction of speech signals. Control strategies for text-to-speech synthesis are discussed briefly.

The electric guitar has been developed to withstand electric amplification and utilize (mostly analog) signal processing in order to create a multitude of timbres and sound types. Sometimes it would be desirable to play the same electric guitar, yet with a sound that resembles a good acoustic guitar or some other member of the plucked string instrument family. In this study we have investigated DSP techniques that can be used to shape the magnetic pickup output of the electric guitar to simulate acoustic instruments. This includes linear filtering for body simulation, time-varying modulation to generate beating of the harmonic components, and techniques to simulate the general temporal envelope of the plucked notes.

A computationally efficient model for multipitch and periodicity analysis of complex audio signals is presented. The model essentially divides the signal into two channels, below and above 1000 Hz, computes a "generalized" autocorrelation of the low-channel signal and of the envelope of the high-channel signal, and sums the autocorrelation functions. The summary autocorrelation function (SACF) is further processed to obtain an enhanced SACF (ESACF). The SACF and ESACP representations are used in observing the periodicities of the signal. The model performance is demonstrated to be comparable to those of recent time-domain models that apply a multichannel analysis. In contrast to the multichannel models, the proposed pitch analysis model can be run in real time using typical personal computers. The parameters of the model are experimentally tuned for best multipitch discrimination with typical mixtures of complex tones. The proposed pitch analysis model may be used in complex audio signal processing applications, such as sound source separation, computational auditory scene analysis, and structural representation of audio signals. The performance of the model is demonstrated by pitch analysis examples using sound mixtures which are available for download at http://www.acoustics.hut.fi/-ttolonen/pitchAnalysis/.

A nonlinear discrete-time model that simulates a vibrating string exhibiting tension modulation nonlinearity is developed. The tension modulation phenomenon is caused by string elongation during transversal vibration. Fundamental frequency variation and coupling of harmonic modes are among the perceptually most important effects of this nonlinearity. The proposed model extends the linear bidirectional digital waveguide model of a string. It is also formulated as a computationally more efficient single-delay-loop structure. A method of reducing the computational load of the string elongation approximation is described, and a technique of obtaining the tension modulation parameter from recorded plucked string instrument tones is presented. The performance of the model is demonstrated with analysis/synthesis experiments and with examples of synthetic tones.

In low-delay audio coding (coding delay <5 ms) there is no time for detailed spectral modeling in the case of brief percussive sounds, e.g., the castanets, and onsets of music or speech sounds. On the other hand, it is known from psychoacoustic experiments that the ear is not accurate near the onset of a wideband sound. We study the audibility of coding errors near the onsets of musical sounds in a simulated low-delay audio codec based on frequency-warped linear prediction. It is suggested that for many musical transients it is sufficient to reproduce a rough temporal and spectral envelope of the original signal during the first 5-10 ms. Preliminary listening tests support this idea. It is proposed that the overshoot effect of hearing could be utilized efficiently in enhancing the performance of a low-delay audio coding scheme.

A model for multi-pitch and periodicity analysis of complex audio signals is presented that is more efficient and practical than the Meddis and O'Mard (see J. Acoust. Soc. Am., vol.102, p.1811-20, 1997) unitary pitch perception model, yet exhibits very similar behavior. We also demonstrate how to apply this model to source separation of complex audio signals such as polyphonic and multi-instrumental music and mixtures of simultaneous speakers. Such analysis techniques are important for automatic transcription of music and structural representation of audio signals.

Digital waveguide modeling of a nonlinear vibrating string is investigated when the nonlinearity is essentially caused by tension modulation. We derive synthesis models where the nonlinearity is implemented with a time-varying fractional delay filter. Also, conversion from a dual-delay-line physical model into a single-delay-loop model is explained. Realistic synthetic tones with nonlinear effects are obtained by introducing minor amendments to a linear string synthesis algorithm. It is shown how synthetic plucked-string tones are modified as a consequence of tension modulation.

A technique for rapid speaker adaptation, called eigenvoices, was introduced recently. The key idea is to confine models in a very low-dimensional linear vector space. This space summarizes a priori knowledge that we have about speaker models. In many practical systems, however, there is a mismatch between the conditions in which the training data were collected and test conditions: prior knowledge becomes improper. Furthermore, prior statistics or models of this mismatch may not be available. We expose two key results: first, we use a maximum-likelihood estimator of prior information in matched conditions, called MLES, leading to an improvement of adaptation by a relative 14%, and second, we show how we can apply a blind scheme for learning noise, MLLR, achieving an additional 7:7% relative improvement in noisy conditions.

We describe how our Finnish talking head was improved by using a new auditory speech synthesis method based on neural networks and optimal synchronization of the facial speech animation and the audio signal. In our first version of the talking head, the user typed in text and synthesized auditory speech and synchronized facial animation were created automatically. We combine a 3D facial model with a commercial auditory text-to-speech synthetizer (TTS). The auditory speech is produced by concatenating pre-recorded samples of natural speech according to a set of rules. The quality of the current speech synthesis is not yet adequate. A new strategy has been developed to improve the TTS and to integrate auditory synthesizer synchronization, especially when hardware capabilities are limited. We are developing a new method to achieve an optimal synchronization, independent of the platform used. This method is based on predictive visual synthesis. The new synchronization method gives us better control over audio-visual speech synthesis in the time domain. Using the diphone duration, we can use a more realistic interpolation function between the visemes. Thus, we can also take into account coarticulation effects.

A text-to-speech synthesis technique, based on warped linear prediction (WLP) and neural networks, is presented for high-quality individual sounding synthetic speech. Warped linear prediction is used as a speech production model with wide audio bandwidth yet with highly compressed control parameter data. An excitation codebook, inverse filtered from a target speaker's voice, is applied to obtain individual tone quality. A set of neural networks, specialized to yield synthesis control parameters from phonemic input in specific contexts, generate the detailed parametric controls of WLP. Neural nets are also used successfully to compute the prosodic parameters. We have applied this approach in prototyping highly improved text-to-speech synthesis for the Finnish language.

This paper describes an experimental architecture for interactive education of DSP in the World Wide Web environment. The architecture is based on a client-server model providing means of distributing resources. The major design goal has been to combine interactivity and computational resources. The architecture itself is open in a sense that it does not specify implementation and it can be used for a variety of applications. Computer based education (CBE) of DSP is one area where it can be beneficially applied. An example of implementation for that purpose is presented. With our implementation, special attention has been paid to minimize the requirements of additional software for the user because of educational reasons. Thus only a Java-capable browser and audio support are needed at the user end.

An automated speech signal labeling tool, developed for the QuickSig speech database environment, is described. It is based primarily on the use of neural networks as diphone event detectors. For robustness, only coarse categories of diphones, such as stop–vowel and vowel–nasal, are used. 64 such detectors are implemented to cover all of the Finnish diphones. The preprocessing of speech signals is carried out using warped linear prediction and the diphone events from neural network outputs are matched to the given text transcription using a simple rule-based parser. In the case of isolated word labeling of single speaker signals a well trained system makes about 1-2 % of coarse labeling errors and the deviation of boundary positions, compared to careful manual labeling, is on average about 10 ms. Generalization ability to label other speakers shows promising.

Bark-scale warped linear prediction (WLP) is a very potential core for a monophonic perceptual audio codec. In the current paper the WLP scheme is extended for processing complex valued signals (CWLP). Three different methods of converting a stereo signal to one complex valued signal are introduced. The philosophy behind the coding scheme is to integrate some aspects of modern wideband audio coding (e.g. perceptuality and stereo signal processing) into one computational element in order to find a more holistic and economic way of processing.

Digital filters where unit delays are replaced with frequency dependent delays, such as first order allpass sections, are often called warped filters since they implement filter specifications on a warped non-uniform frequency scale. Warped IIR (WIIR) filters cannot be realized directly due to delay free loops. Specific solutions have been known that make WIIR filters realizable but no general approach has been available so far. In this paper we will explore the generation of such filters, including new filter structures. The robustness and computational efficiency of WIIR filters are studied and most potential applications are discussed.

This paper describes an interactive, self-study DSP tutorial package built for the World Wide Web (WWW, Web). The package includes concept explanation, graphical presentations, listening-ready audio examples and interactive exercises and demonstrations of basic concepts of signal processing. The WWW is a convenient and easy-to-use hypermedia tool. From the point of view of DSP education, the main problem is how to implement interactivity and signal processing. Our solution is to apply CGI (common gateway interface) and Java in tasks that they are best suited for. A common lisp based WWW server CL-HTTP together with the QuickSig environment is used for providing signal processing routines in a CGI-like manner. Java applets are used for taking care of the user interface at the client end.

It has been hypothesized that the Finnish language is well suited to speech-to-text conversion for the communication aids of the hearing impaired. In a related study it was shown that, depending on context, 10 to 20 % of phoneme errors can be tolerated with good comprehension when reading text converted from raw phonemic recognition. Two sets of phoneme recognition experiments were carried out in this study in order to evaluate the performance of existing speech recognition systems in this application. For telephone bandwidth speech both systems showed speaker dependent error scores of about 10 % or below, thus supporting the feasibility of the application. For speaker independent cases the error rate was typically more than 20 % which is too high for effortless and fluent communication. 1.

A computer-based education (CBE) system is described that is built on top of the QuickSig DSP programming environment. As two CBE applications the authors discuss the implementation and use of "Introduction to Signal Processing" and "Fundamentals of Psychoacoustics" that exploit modern multi and hypermedia features on new computers. The signal processing that is applied, the sound I/O, the graphical user interface, and the CBE navigation system are presented.

This paper discusses a discrete-time modeling technique where the length of time delays can be arbitrarily adjusted. The new system is called a fractional delay waveguide model (FDWM). Formerly, FDWMs have only been implemented with FIR-type fractional delay filters. We show how an FDWM can be implemented using allpass filters. We use low-order allpass filters that are maximally-flat approximations of the ideal delay. The advantages of the allpass approach are computational efficiency and reduced approximation error. The proposed structure can be applied to discrete-time modeling of acoustic tubes, such as the human vocal tract or resonators of musical instruments.

A linear prediction process is applied to frequency warped signals. The warping is realized by using orthonormal FAM (frequency modulated complex exponentials) functions. The general formulation of WLP is given and effective realizations with allpass filters are studied. The application of auditory WLP to speech coding and speech recognition has given good results.>

An extension to the traditional Kelly-Lochbaum (1962) vocal tract model is introduced. In the new model not only the diameter but also the length of each tube section can be continuously adjusted. This is achieved by using fractional delay filter techniques such as interpolation and deinterpolation. The filter structure consisting of bidirectional delay lines (digital waveguides) and interpolated ports that connect two or more waveguide sections together is called a fractional delay waveguide filter (FDWF). The interpolated version of the two-port scattering junction is presented and a technique for analyzing the degradation due to approximation errors in interpolation and deinterpolation is described. It is shown that when an FDWF structure with Lagrange interpolation is used a vocal tract model needs to be implemented using oversampling. For example, a sampling rate of 22 kHz is adequate for producing high-quality synthetic sounds at a 5 kHz bandwidth.>

An articulatory model of speech production is usually constructed by approximating the profile of the vocal tract using cylindrical tube sections. This is implemented by a digital ladder filter that is called the Kelly--Lochbaum model. In this paper we propose an extended approach, where the tube sections approximating the profile of the tract are conical instead of cylindrical. Furthermore, the length of each tube section in our model can be accurately controlled using a novel fractional delay filtering scheme. These refinements result in an accurate and intuitively controllable vocal tract model that is well suited for articulatory speech synthesis. I. INTRODUCTION An articulatory model for the human vocal tract is traditionally constructed by approximating the profile of the vocal tract using cylindrical tube sections. The resulting tube system is then modeled by a digital ladder filter. This approach was first used in [1] and is called the Kelly-- Lochbaum (KL) model. There are tw...

Object-oriented programming (OOP) is a new paradigm and methodology for software development and design. A case study of applying OOP to the programming of the TMS320C30 signal processors in time-critical applications is presented. Data types and structures are defined as classes (abstract data types) in a systematic way without need for compromising between flexibility and efficiency. Procedurally the environment supports method (function) definitions in a high-level language, low-level macro instructions, and middle-level method instruction definitions integrated by a common syntax so that even low-level instructions may refer to the high-level OOP data types. The system is implemented in Common Lisp and follows the syntax of Common LISP Object System (CLOS). Several examples of digital signal processor (DSP) programming are given.>

A computationally efficient semiphysical time-domain model of the flute is presented. The traditional method of modeling the one-dimensional wave propagation as a transmission-line, where all the losses as well as reflection and dispersion effects have been lumped to linear filters at the ends of the line, is used. The model is excited by white noise, and the nonlinear interaction between the excitation and the tube resonator has been modeled by a static memoryless nonlinearity. The model is simple enough to be computationally efficient, but has retained many important features of a real instrument. The real-time implementation has been done on a Texas Instruments TMS320C30 floating-point signal processor using a sampling rate of 44.1 kHz.>

Algorithms that can be used to synthesize guitar sounds on a floating-point signal processor are presented. A finite impulse response (FIR) Lagrange interpolator is introduced to implement the efficient and precise fractional delay approximation that is needed to achieve arbitrary and varying-length strings. This kind of interpolation is especially good in avoiding distortion and undesirable extra effects when the string length is changing continuously during the synthesis of a sound. The interpolator can also be used in other cases, e.g. in transmission-line modeling of acoustic tube resonators in wind instruments and for vocal tract models in speech synthesis. In addition to the interpolation principle, the implementation of the guitar string model on the TMS320C30 floating-point signal processor is described.>

A transducer material consisting of foamized plastic, called electrothermomechanical film (ETMF), is introduced. The film has conductive electrodes on at least one side, so it can be used with either an electrostatic or electrodynamic driving force. The film can be used both as a transmitter and a receiver at audio and ultrasonic frequencies, also in hydroacoustical applications. Lumped-component and one-dimensional distributed-parameter models are developed to examine the behavior of the film. Frequency responses and sensitivities predicted by the models are compared with measured data. Audio applications of the film as a loudspeaker (tweeter and full-range) are introduced with data collected from prototypes. Means of controlling radiation pattern and frequency response are briefly discussed. Applications of the film as a microphone, hydrophone, and hydroacoustic source for audio and ultrasonic frequencies are mentioned.>

In a search for more flexible and powerful tools for DSP (digital signal processing) programming the author has developed an integrated programming environment called QuickSig. He describes the subsystem for signal processor code generation. The target processor in this study is the TMS320C30. The entire software system is based on the use of the Common Lisp language to achieve an exceptional degree of system integration, resulting in a high degree of flexibility, interactivity, and programming productivity. All levels from algorithm development to object code generation, including the assembly language, use the same Lisp notation. The principles and implementations of QuickSig are discussed along with some programming examples.>

A methodology for multiple-resolution analysis and event-based representation of speech signals is presented. The computation of multiple-resolution filtering, event detection and parsing of event structures is described with examples and discussions of auditory modeling aspects. The approach and its implementation are entirely based on object-oriented programming, which provides a systematic framework for the hierarchical nature of the method. Implementation of the analysis system is based on an object-oriented signal processing system called QuickSig running on the Symbolics Lisp machine. The QuickSig system has object classes such as signals, windows and filter banks, each with its own set of method functions for signal processing operations.>

An object-oriented DSP (digital signal-processing) environment called QuickSig is described that is based on recent developments in object-oriented programming (New Flavors on Symbolics Lisp machines). The design philosophy of QuickSig has been to extend the Lisp language by a layer of general DSP constructs, abstracts, and structures like signals, filters, windows, graphical presentations, and related signal-processing operations. QuickSig is a fast prototyping system for algorithmic development. It is extendable to include new ways of modeling signals and signal processing, both numerical and symbolic. The main features of the present system and some features that are under development are reported.>

Waveform concatenation has been the method used in many low-quality speech synthesis experiments. The objective of this study was to find new ways to overcome the inherent difficulties in concatenating speech sample waveforms. Our experiments in Finnish and Polish show that it is feasible to develop simple and inexpensive synthesizers with natural and high-quality human-like characteristics. The implementation can be based on standard microprocessors and D/A-converters without expensive signal processing hardware. This paper describes the results of the experiments and conclusions to the design of speech synthesis that we call the microphonemic method.

A new method to measure the acoustical impedance of an artificial glottal orifice is presented. The plate with the orifice is mounted to one end of a tube resonator with the other end being open. The impedance of the orifice seen from the tract can be solved from the resonance frequencies and bandwidths. The frequency characteristic of the orifice is easily otained under different DC-flow conditions. The results show that under turbulent flow the effective glottal inductance is clearly only a fraction of its flowless value. The measured glottal resistance is close to the theoretical value given by Flanagan's two-mass model.

A new computational model of auditory spectrum analysis is presented and its validity in sound quality measurements is shown. The model is based on the most important features of peripheral hearing in time and frequency known from the psychoacoustic theory. The methodology of sound quality measurement is based on the use of real signals (e.g. speech) as test signals and auditory spectrum distance as a measure of sound quality degradation. This paper describes the new auditory model and the results of testing its validity. A good correlation between subjective and our objective measures of distortion is shown. Auditory spectrum distance far out-performs the traditional distortion measures in this sense. Some limitations and further ideas concerning the model are also discussed.

Traditional methods of analyzing the sound quality of audo equipment are based on simple test signals as well as on simple measurement procedures. Sine wave or sweep is applied when measuring nonlinear (harmonic) distortion or frequency response. Combinations of sine waves and/or some kind of noise signals are the most complex test signals used in practice. Analysis methods as well are based on oversimplified models of auditory perception. It is easy to demonstrate how measures like harmonic distortion may be misleading when used to describe the perceived quality of sound, A new methodology is introduced here, which is based on a computational model of auditory perception as the analyzer of sound quality. The model makes it possible to use any real signal as a test signal. Some measurements carried out in practice are described.

The primary objective of this study was to determine what benefit could be gained in speech coding by using a psychoacoustical frequency scale instead of a linear scale. To partially overcome the well known difficulties in objective speech quality measurements, a computational performance criterium based on psychoacoustical models was developed. Several Finnish phonemes were then coded using regular LPC and LPC computed on a psychoacoustically correct frequency scale (Bark scale)and the coding performance of these both methods was tested via computational performance tests. The results indicate a significant improvement in speech quality for the same bit-rate, when applying LPC on psycho-acoustical frequency scale. Preliminary listening tests support both the better coding capability of the Bark LPC compared to the regular LPC and the reliability of the developed speech quality criterium as an objective performance evaluation method.

SYNTE 2 is a low-cost, high-quality, text-to-speech synthesizer designed for Finnish but applicable also to other languages if "phoneme writing" is used. After its first presentation in 1977 it has been adapted to many communication aids for the handicapped. The first application was a portable speaking machine with unlimited vocabulary for the speech impared. This paper describes the present applications of SYNTE 2, including the speaking machine, a talking data terminal for blind computer programmers, a system for automatic production of spoken information for the blind, etc.