摘要:

This paper discusses the generation and detection of acoustic waves in a piezoelectric material by cavity excitation. By invoking electromagnetic field concepts whereby the acoustic waves appear as equivalent current sources, the reciprocal nature of both problems can be proved. Also, from a knowledge of the cavity electric field and the acoustic excitation vector, the detected cavity responses can be evaluated, showing that two distinct responses (even and odd) are obtained for differing electric fields on the piezoelectric surfaces.

ISSN:0001-4966    

DOI:10.1121/1.383655

出版商:Acoustical Society of America    

年代:1979

数据来源: AIP

摘要:

Theory and preliminary experiments are outlined relating to a nearfield method of evaluating electroacoustic transducers. The theoretical results are conveniently organized into two theorems: (1) The total complex force on all parallel infinite planes to one side of a transducer and perpendicular to an arbitrary direction, has a constant magnitude equal to the magnitude of the farfield pressure in that same direction multiplied by the wavelength. (2) The output voltage of a baffled, reciprocal, plane‐piston receiver is proportional to the total incident force normal to its face. These two theorems suggest a compact and relatively simple method for evaluating directive or moderately directive ultrasonic transducers. It is demonstrated experimentally that there is good agreement between the farfield pattern of a directive transducer in the main beam region, measured by a large, plane‐piston transducer in the nearfield, and the pattern measured directly with a small probe in the approximate farfield. Further, it is experimentally demonstrated that, as a piston receiver is moved axially, with respect to a transmitter, no discernible change in the output voltage of the receiver is detected, provided the nearfield beam of the transmitter is intercepted by the receiver (even though the nearfield of either transducer differs significantly from that of a plane wave).

ISSN:0001-4966    

DOI:10.1121/1.383656

出版商:Acoustical Society of America    

年代:1979

数据来源: AIP

摘要:

A finite element analysis method is used to obtain the electromechanical coupling coefficientkefor plank‐shaped piezoelectric transducers with a widthWto thicknessTratio of not more than two. Calculation results for typical piezoelectric ceramics show that there exist maximum values ofkeat certain values ofW/T. For PCM‐5Rmaterial the maximum value ofkeis 0.69 at aW/Tof 0.6. It is shown that only one vibrational mode is very strongly coupled around this value ofW/T. This vibrational mode is very useful for application to electronically scanned arrayed transducers.

ISSN:0001-4966    

DOI:10.1121/1.383657

出版商:Acoustical Society of America    

年代:1979

数据来源: AIP

摘要:

The investigation uses a flue organ pipe designed so that mouth heightY, position of the upper lip in the air streamX, and width of the air passage are adjustable. Other dimensions can be altered with interchangeable parts. Rectangular and circular resonators of several lengths, open and stopped, are used. Speaking frequencies are 125 to 500 Hz. Speech for a supply pressure of 55‐mm water is described for the usefulX,Yrange in qualitative terms, and also in terms of sound‐pressure level and harmonic content. Behavior is related to resonance frequencies. Stable oscillation of longer pipes occurs in a limitedX,Yrange unless the first resonance frequency is some 2% lower, relative to harmonic relationship with the next few modes of vibration. AsXis varied, for the speaking pipe, the second harmonic has an amplitude minimum, where the phase relative to the fundamental reverses. The acoustic pressure is proportional to mass flow rate over a range of at least 4:1 in tests involving reduction of air pressure or narrowing the passage. Threshold operating pressure for 7‐mm mouth height is 5‐mm water, in good agreement with Coltman’s flute data. Air supply energy is converted to radiation with about 0.5% efficiency for the stopped pipe.

ISSN:0001-4966    

DOI:10.1121/1.383658

出版商:Acoustical Society of America    

年代:1979

数据来源: AIP

ISSN:0001-4966    

DOI:10.1121/1.383659

出版商:Acoustical Society of America    

年代:1979

数据来源: AIP

摘要:

Current activities on speech waveform coding technique developments in Japan are reviewed. The following topics of the activities on ADPCM and ADM are discussed. (1) An ADPCM code sequence optimization method by Fushikida for an ADPCM with an adaptive quantization and fixed predictor. The method uses a technique reminiscent of dynamic programming and optimizes code sequence. It is shown that about 3 dB SNR improvement is obtainable for speech at 32 kbps (4 bits/sample). (2) Another ADPCM technique by Araseki and Ochiai. This technique is suitable for use at the bit rate less than 32 kbps. Two adaptive predictors connected in a cascaded form are used. One of the predictors has an ordinary small number of prediction taps. However, the other predictor possesses a considerably larger number of predictor taps and is used to predict the speech waveform which is one pitch period apart. (3) Performance improvement techniques and LSI implementation by Tomozawa and Niwa for a discretely variable slope 32 kbps ADM (DVSD). Yatsuzuka’s residual encoder and Hosokawa and Yamashita’s adaptive predictive DM are also discussed briefly.

ISSN:0001-4966    

DOI:10.1121/1.383660

出版商:Acoustical Society of America    

年代:1979

数据来源: AIP

摘要:

This paper surveys recent developments in adaptive predictive coding (APC) of speech. Prominent among these developments are the use of a three‐point pitch predictor, a pitch‐adaptive quantizer, entropy coding of the residual, and adaptive shaping of the quantization‐noise spectrum. APC systems produce high quality speech at around 16 kbit/s; their quality diminishes rapidly at 9.6 kbit/s or less. For those lower data rates, some form of baseband coding system becomes desirable. In such systems, a low‐frequency baseband is transmitted. The high‐frequency regeneration of the excitation spectrum from the baseband is of special importance. Traditional regeneration techniques have used some form of nonlinear distortion (usually rectification) of the baseband, followed by spectral flattening. We introduce a new set of regeneration based on duplication of the baseband spectrum at high frequencies. The audible signal distortions in rectification and spectral folding are compared.

ISSN:0001-4966    

DOI:10.1121/1.383661

出版商:Acoustical Society of America    

年代:1979

数据来源: AIP

摘要:

Frequency domain techniques for speech coding have recently received considerable attention. The basic concept of these methods is to divide the speech into frequency components by a filter bank (subband coding) or by a suitable transform (transform coding) and then encode them using adaptive PCM. Four basic operations are involved in the design of these coders: (1) the type of transform or filter bank (analysis/synthesis), (2) the adaptive quantizer design (quantization theory), (3) the choice of bit allocation used by the quantizers (noise shaping and auditory masking), and (4) the control of the step‐size of the quantizers (spectral estimation). This paper briefly reviews the basic aspects of the design of these four operations particularly as they apply to low bit‐rate adaptive transform coding.

ISSN:0001-4966    

DOI:10.1121/1.383557

出版商:Acoustical Society of America    

年代:1979

数据来源: AIP

摘要:

In any speech coding system that adds noise to the speech signal, the primary goal should not be to reduce the noise power as much as possible, but to make the noiseinaudibleor to minimize its subjective loudness. ’’Hiding’’ the noise under the signal spectrum is feasible because of human auditory masking: sounds whose spectrum falls near the masking threshold of another sound are either completely masked by the other sound or reduced in loudness. In speech coding applications, the ’’other sound’’ is, of course, the speech signal itself. In this paper we report new results of masking and loudness reduction of noise and describe the design principles of speech coding systems exploiting auditory masking.

ISSN:0001-4966    

DOI:10.1121/1.383662

出版商:Acoustical Society of America    

年代:1979

数据来源: AIP

摘要:

This paper discusses practical implementations of low bit‐rate speech waveform digitizers for the pre‐1980 and post‐1985 time periods. Based on requirements of expected mathematical complexity, accuracy, and redundance of these coders, the basic hardware speeds and amount of logic to implement them are given for today’s implementations. Issues important to the commercial and military users are examined to ascertain how they can radically alter the architectures and hardware chosen for implementation. Then assuming that the speech digitization algorithms remain at least as complex as today’s techniques, well‐known procedures for forecasting LSI improvements and cost reductions are used in estimating the hardware needed to implement these systems in the mid 1980s.

ISSN:0001-4966    

DOI:10.1121/1.383663

出版商:Acoustical Society of America    

年代:1979

数据来源: AIP