Several parametric representations of the acoustic signal were compared with regard to word recognition performance in a syllable-oriented continuous speech recognition system. The vocabulary included many phonetically similar monosyllabic words, therefore the emphasis was on the ability to retain phonetically significant acoustic information in the face of syntactic and duration variations. For each parameter set (based on a mel-frequency cepstrum, a linear frequency cepstrum, a linear prediction cepstrum, a linear prediction spectrum, or a set of reflection coefficients), word templates were generated using an efficient dynamic warping method, and test data were time registered with the templates. A set of ten mel-frequency cepstrum coefficients computed every 6.4 ms resulted in the best performance, namely 96.5 percent and 95.0 percent recognition with each of two speakers. The superior performance of the mel-frequency cepstrum coefficients may be attributed to the fact that they better represent the perceptually relevant aspects of the short-term speech spectrum.

Comparison of parametric representations for monosyllabic word recognition in continuously spoken sentences

An intelligent automated assistant system engages with the user in an integrated, conversational manner using natural language dialog, and invokes external services when appropriate to obtain information or perform various actions. The system can be implemented using any of a number of different platforms, such as the web, email, smartphone, and the like, or any combination thereof. In one embodiment, the system is based on sets of interrelated domains and tasks, and employs additional functionally powered by external services with which the system can interact.

Intelligent Automated Assistant

The temporal properties of speech appear to play a more important role in linguistic contrasts than has hitherto been appreciated. Therefore, a new framework for describing the acoustic structure of speech based purely on temporal aspects has been developed. From this point of view, speech can be said to be comprised of three main temporal features, based on dominant fluctuation rates: envelope, periodicity, and fine-structure. Each feature has distinct acoustic manifestations, auditory and perceptual correlates, and roles in linguistic contrasts. The applicability of this three-featured temporal system is discussed in relation to hearing-impaired and normal listeners.

Temporal Information in Speech: Acoustic, Auditory and Linguistic Aspects

We have argued that dynamically defined articulatory gestures are the appropriate units to serve as the atoms of phonological representation. Gestures are a natural unit, not only because they involve task-oriented movements of the articulators, but because they arguably emerge as prelinguistic discrete units of action in infants. The use of gestures, rather than constellations of gestures as in Root nodes, as basic units of description makes it possible to characterise a variety of language patterns in which gestural organisation varies. Such patterns range from the misorderings of disordered speech through phonological rules involving gestural overlap and deletion to historical changes in which the overlap of gestures provides a crucial explanatory element.Gestures can participate in language patterns involving overlap because they are spatiotemporal in nature and therefore have internal duration. In addition, gestures differ from current theories of feature geometry by including the constriction degree as an inherent part of the gesture. Since the gestural constrictions occur in the vocal tract, which can be charactensed in terms of tube geometry, all the levels of the vocal tract will be constricted, leading to a constriction degree hierarchy. The values of the constriction degree at each higher level node in the hierarchy can be predicted on the basis of the percolation principles and tube geometry. In this way, the use of gestures as atoms can be reconciled with the use of Constriction degree at various levels in the vocal tract (or feature geometry) hierarchy.The phonological notation developed for the gestural approach might usefully be incorporated, in whole or in part, into other phonologies. Five components of the notation were discussed, all derived from the basic premise that gestures are the primitive phonological unit, organised into gestural scores. These components include (1) constriction degree as a subordinate of the articulator node and (2) stiffness (duration) as a subordinate of the articulator node. That is, both CD and duration are inherent to the gesture. The gestures are arranged in gestural scores using (3) articulatory tiers, with (4) the relevant geometry (articulatory, tube or feature) indicated to the left of the score and (5) structural information above the score, if desired. Association lines can also be used to indicate how the gestures are combined into phonological units. Thus, gestures can serve both as characterisations of articulatory movement data and as the atoms of phonological representation.

Articulatory gestures as phonological units

Quantization, the process of approximating continuous-amplitude signals by digital (discrete-amplitude) signals, is an important aspect of data compression or coding, the field concerned with the reduction of the number of bits necessary to transmit or store analog data, subject to a distortion or fidelity criterion. The independent quantization of each signal value or parameter is termed scalar quantization, while the joint quantization of a block of parameters is termed block or vector quantization. This tutorial review presents the basic concepts employed in vector quantization and gives a realistic assessment of its benefits and costs when compared to scalar quantization. Vector quantization is presented as a process of redundancy removal that makes effective use of four interrelated properties of vector parameters: linear dependency (correlation), nonlinear dependency, shape of the probability density function (pdf), and vector dimensionality itself. In contrast, scalar quantization can utilize effectively only linear dependency and pdf shape. The basic concepts are illustrated by means of simple examples and the theoretical limits of vector quantizer performance are reviewed, based on results from rate-distortion theory. Practical issues relating to quantizer design, implementation, and performance in actual applications are explored. While many of the methods presented are quite general and can be used for the coding of arbitrary signals, this paper focuses primarily on the coding of speech signals and parameters.

Vector quantization in speech coding

A speech bandwidth extension method and apparatus analyzes narrowband speech sampled at 8 kHz using LPC analysis to determine its spectral shape and inverse filtering to extract its excitation signal. The excitation signal is interpolated to a sampling rate of 16 kHz and analyzed for pitch control and power level. A white noise generated wideband signal is then filtered to provide a synthesized wideband excitation signal. The narrowband shape is determined and compared to templates in respective vector quantizer codebooks, to select respective highband shape and gain. The synthesized wideband excitation signal is then filtered to provide a highband signal which is, in turn, added to the narrowband signal, interpolated to the 16 kHz sample rate, to produce an artificial wideband signal. The apparatus may be implemented on a digital signal processor chip.

Speech bandwidth extension method and apparatus

As a first step toward automatic phonetic analysis of speech, one desires to segment the signal into syllable‐sized units Experiments were conducted in automatic segmentation techniques for continuous, reading‐rate speech to derive such units A new segmentation algorithm is described that allows assessment of the significance of a loudness minimum to be a potential syllabic boundary from the difference between the convex hull of the loudness function and the loudness function itself Tested on roughly 400 syllables of continuous text, the algorithm results in 69% syllables missed and 26% extra syllables relative to a nominal, slow‐speech syllable count It is suggested that inclusion of alternative fluent‐form syllabifications for multisyllabic words and the use of phonological rules for predicting syllabic contractions can further improve agreement between predicted and experimental syllable countsSubject Classification: 7040, 7060

Automatic segmentation of speech into syllabic units

A software articulatory synthesizer, based upon a model developed by P. Mermelstein [J. Acoust. Soc. Am. 53, 1070–1082 (1973)], has been implemented on a laboratory computer. The synthesizer is designed as a tool for studying the linguistically and perceptually significant aspects of articulatory events. A prominent feature of this system is that it easily permits modification of a limited set of key parameters that control the positions of the major articulators: the lips, jaw, tongue body, tongue tip, velum, and hyoid bone. Time‐varying control over vocal‐tract shape and nasal coupling is possible by a straightforward procedure that is similar to key‐frame animation: critical vocal‐tract configurations are specified, along with excitation and timing information. Articulation then proceeds along a directed path between these key frames within the time script specified by the user. Such a procedure permits a sufficiently fine degree of control over articulator positions and movements. The organization of this system and its present and future applications are discussed.

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An articulatory synthesizer for perceptual research

A synchronous discontinuous transmission medium access control (SDTX-MAC) method and apparatus for more efficiently using existing uplink channels by sharing these uplink channels between multiple terminals engaged in bursty data transmission. This is accomplished by assigning each mobile terminal an individual time slot and by not requiring each mobile terminal to broadcast its identity. This results in a reduction in the number of receivers on each base station and a reduction in the length of the synchronization message.

Medium access control scheme for data transmission on code division multiple access (cdma) wireless systems

In recent years, the telecommunications industry has witnessed the proliferation of a variety of digital vocoders in order to meet bandwidth demands of different wireline and wireless communication systems. The rapid growth in the diversity of networks and the number of users of such networks is increasing the number of instances where two vocoders are placed in tandem to serve a single connection. Such arrangements of low bit-rate codecs can degrade the quality of the transmitted speech. To overcome this problem the invention provides a novel method and an apparatus for transmitting digitized voice signals in the wireless communications environment. The apparatus is capable of converting a compressed speech signal from one format to another format via an intermediate common format, thus avoiding the necessity to successively de-compress voice data to a PCM type digitization and then recompress the voice data.

Paul Mermelstein

Papers

Speech bandwidth extension method and apparatus

Automatic segmentation of speech into syllabic units

An articulatory synthesizer for perceptual research

Medium access control scheme for data transmission on code division multiple access (cdma) wireless systems

Method and apparatus for improving the voice quality of tandemed vocoders