IEEE -Trans on Audio & Electroacoustics

An audio response unit has been built that synthetizes messages composed of a fixed sentence and any number from 0-999 999. The method used is synthesis by concatenation of words, and automatic corrections on pitch and rhythm are used to improve naturalness and intelligibility. The synthesizer is a part of a channel vocoder. This audio response unit is to be used in a telephone exchange to answer cost inquiries from the subscribers. Larger applications using touch-tone telephone as data input device are under consideration.

J. Genin

IEEE Transactions on Audio and Electroacoustics

An audio response unit for telephone needs

电话需求的音频响应单元

This paper presents a model for machine recognition of connected speech and the details of a specific implementation of the model, the HEARSAY system. The model consists of a small set of cooperating independent parallel processes that are capable of helping in the decoding of a spoken utterance either individually or collectively. The processes use the "hypothesize-and-test" paradigm. The structure of HEARSAY is illustrated by considering its operation in a particular task situation: voice-chess. The task is to recognize a spoken move in a given board position. Procedures for determination of parameters, segmentation, and phonetic descriptions are outlined. The use of semantic, syntactic, lexical, and phonological sources of knowledge in the generation and verification of hypotheses is described. Preliminary results of recognition of some utterances are given.

D. Reddy

L. Erman

R. Neely

A model and a system for machine recognition of speech

机器语音识别的模型与系统

With the exception of relatively sophisticated methods such as cepstrum analysis, the problem of reliable pitch-period extraction has remained largely unsolved. This paper examines the feasibility of pitch-period extraction by means of the nonstationary error process resulting from adaptive-predictive quantization of speech. A real-time hard-ware system that may be realized at low cost is described.

J. Maksym

Real-time pitch extraction by adaptive prediction of the speech waveform

实时音高提取通过自适应预测语音波形

The problem of developing discrete-time models for continuous systems from integral approximations is considered. In particular, the model obtained by approximating the input by a piecewise linear function is developed. By an appropriate approximation of this model, an equivalence between it and the bilinear transform model is established.

B. Haberland

S. Rao

Discrete-time models:Bilinear transform and ramp approximation equivalence

离散时间模型：双线性变换与斜坡逼近等价性

An analysis of the three possible types of quantization effects in the direct form realization of finite impulse response (FIR) digital filters is presented. These quantization effects include roundoff noise, A-D noise, and filter frequency response errors due to coefficient quantization. Since the analysis of roundoff noise and A-D noise for the direct form is straightforward, this paper concentrates on an analysis of the effects of quantized coefficients on the resulting filter frequency response. Based on this analysis, statistical bounds on the error incurred in the frequency response of a filter due to coefficient quantization are developed and verified by extensive experimental data. Using these bounds, a procedure for applying known techniques for FIR filter design to the design of filters with finite word length coefficients is presented. On the whole, the direct form is shown to be a very attractive structure for realizing FIR filters.

D. Chan

L. Rabiner

Analysis of quantization errors in the direct form for finite impulse response digital filters

有限冲激响应数字滤波器直接型结构中的量化误差分析

The present study demonstrates that spectrograms of Swedish utterances can be read with great accuracy under nontrivial conditions. This result is to be attributed primarily to the development of a formalized strategy that was designed to make it possible for spectrogram readers to derive information on certain grammatical features of an utterance such as word class, word boundaries, endings, and function elements. The input to this strategy consists of segmental phonetic features that the subjects extract from the spectrographic display and of information on prosodic features such as stress and tonal accent. The latter information is specified on the spectrogram for each syllable. An experimental situation is thus created that differs from the informal recognition of unknown utterances from spectrograms. A subject can base his final identification of lexical items not only on segmental phonetic features but also on an error-free specification of prosodic features and, in so far as he has been able to use the strategy successfully, on grammatical information. Experimental results are reported indicating that subjects improve their performance markedly with the aid of the strategy. In conclusion, attention is drawn to the important role that grammar and prosody appear to play in the present experiments and to the implications of the findings for future work on automatic speech recognition and speech perception.

B. Lindblom

S. Svensson

Interaction between segmental and nonsegmental factors in speech recognition

语音识别中分段与非分段因素的相互作用

The loudspeaker system small-signal figure of merit is a useful performance criterion that must be evaluated and used with some care. In practice, this figure of merit is significantly reduced by enclosure dissipation, including that contributed by filling materials.

R. Small

Loudspeaker system figures of merit

扬声器系统性能指标

A new realization of nonrecursive digital filters that are used to operate on analog signals is proposed. This realization requires no multiplications, and exploits the relative simplicity of delta modulation as a means for analog to digital conversion. This realization also permits a mechanization as a "very fast" digital filter, using read-only memory (ROM). An evaluation of this realization in terms of computation time storage requirements and mean-squared error is presented. These characteristics are compared with their counterparts for existing realization methods of nonrecursive digital filters. Computer simulation results that tend to confirm the theoretical results of the error analysis are included.

A. Peled

None Bede Liu

A new approach to the realization of nonrecursive digital filters

非递归数字滤波器实现的新方法

It is proved that the z transform of a sequence of data values cannot be exactly computed using a binary number representation for values of z on the unit circle, except z =±1, z = ±j. It is also proved that the discrete Fourier transform (DFT) of a sequence of data values cannot be evaluated with rational numbers.

C. Rader

A note on exact discrete Fourier transforms

关于精确离散傅里叶变换的注记

This paper presents a general-purpose computer program which is capable of designing a large Class of optimum (in the minimax sense) FIR linear phase digital filters. The program has options for designing such standard filters as low-pass, high-pass, bandpass, and bandstop filters, as well as multipassband-stopband filters, differentiators, and Hilbert transformers. The program can also be used to design filters which approximate arbitrary frequency specifications which are provided by the user. The program is written in Fortran, and is carefully documented both by comments and by detailed flowcharts. The filter design algorithm is shown to be exceedingly efficient, e.g., it is capable of designing a filter with a 100-point impulse response in about 20 s.

Publication Information	Publisher: Institute of Electrical and Electronics Engineers Inc.，Publishing cycle: ，Journal Type: journal，Open Access Journals: No
Basic data	Year of publication: 1966，Proportion of original research papers: ，Self Citation Rate:， Gold OA Rate:

IEEE -Trans on Audio & Electroacoustics

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