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May 1990

Volume 87, Issue S1, pp. S1-S164

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back to top Session A. Engineering Acoustics I: Application of Signal Processing to Electroacoustics
Invited Papers
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Acoustic echo cancellation with a transform domain adaptive filter (A)

Sen M. Kuo and Huan Zhao

J. Acoust. Soc. Am. Volume 87, Issue S1, pp. S1-S2 (1990); (2 pages)

Online Publication Date: 13 Aug 2005

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The adaptive filtering technique is one of the major approaches to control the acoustic echoes in teleconferencing systems or hands‐free telephony systems. A brief overview of the performance of different adaptive algorithms for acoustic echo cancellation [cf, S. M. Kuo and H. Zhao, J. Acoust. Soc. Am. Suppl. 1 86, S56 (1989)] will be presented. A transform domain adaptive filter (TDAF) algorithm is proposed to construct an acoustic echo canceler that has a convergence rate faster than that of the conventional LMS adaptive transversal filter with colored signals. The proposed approach does not introduce extra delay during the process so that it can be used in the application where delay is not allowed. Different orthogonal transforms such as the discrete Fourier transform (DFT), discrete cosine transform (DCT), and discrete Hartley transform (DHT) are used for signals with different eigenvalue spread. Real‐time implementations of this class of canceler with digital signal processors (DSP) will also be discussed.
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A microphone array for echo cancellation in audioconference rooms (A)

Y. Grenier and M. Xu

J. Acoust. Soc. Am. Volume 87, Issue S1, pp. S2-S2 (1990); (1 page)

Online Publication Date: 13 Aug 2005

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This paper describes an adaptive acoustic antenna (or microphone array) designed for sound recording in audioconferencing rooms. This microphone array achieves three goals simultaneously: the sound from the useful sources (speakers) is recorded without distortion, several jammers emitting known signals can be cancelled (this is the case for the sound received from the other room and emitted by the loudspeakers), and finally, the ambient noise and the room reverberation are reduced. This last effect enforces the “presence effect” in the recordings. In the present approach, echo cancellation is seen as the problem of rejecting a jammer, and it is done by signal combination rather than by subtraction. To achieve these goals, the microphone array uses adaptive beamforming techniques. The beamformer is a compromise between data‐independent or fixed beamformers and data‐dependent or adaptive ones. This makes it robust with respect to localization errors: the beamformer uses an identified localization of the jammers but weighs this information by some knowledge of the signal‐to‐noise ratio. The localization of the main jammer (the echo coming from the loudspeaker) requres the identification of a long impulse response (the acoustic channel between the loudspeaker and each microphone). A model to identify this response in the time‐frequency domain is developed. This model involves a “double convolution” and its adaptive identification has some analogies with subband adaptive methods. The estimated parameters of this model are used to compute the optimal weights for the beamformer every 8 ms. The paper will present simulations of these techniques based upon a first prototype that consists of an array of 15 microphones on a semicircle with diameter 1 m. The performances of this adaptive microphone array are characterized in terms of echo rejection, convergence rate, and level of dereverberation.
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Acoustic echo cancellation in subbands (A)

Walter Kellermann

J. Acoust. Soc. Am. Volume 87, Issue S1, pp. S2-S2 (1990); (1 page)

Online Publication Date: 13 Aug 2005

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Acoustic echo cancellation for teleconferencing or hands‐free telephony constitutes a challenging task for today's digital signal processing techniques. Compared to the well‐established line echo cancellation in telephone networks, the problem takes on a considerably larger size: The impulse response of the echo path to be compensated is longer and may vary more rapidly in time. Moreover, larger bandwidths are often desirable. Therefore, acoustic echo cancellation requires algorithms that adapt far more coeffecients and converge faster than those commonly used in line echo cancelers. On the other hand, direct implementation of sufficiently advanced adaptive filtering algorithms is usually prohibited because of their computational complexity and inherent numerical difficulties. As a way out, the subband approach realizes a “divide and conquer” strategy and promises a reduction of computational complexity and favorable circumstances for fast convergence at the cost of some extra delay. After comparing several possible subband structures, the frequency subband structure is discussed in some detail. For adaptation using LMS‐type algorithms, reasons for the improved convergence behavior compared to a fullband implementation are explained. Design examples are presented to illustrate the effectiveness of the method.
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DSP beamforming and talker tracking with a linear microphone array (A)

Harvey F. Silverman

J. Acoust. Soc. Am. Volume 87, Issue S1, pp. S2-S2 (1990); (1 page)

Online Publication Date: 13 Aug 2005

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One of the problems for all speech input is the necessity for the talker to be encumbered by a head‐mounted, hand‐held, or fixed position microphone. An intelligent, electronically aimed undirectional microphone would overcome this problem. Array techniques hold the best promise to bring such a system to practicality, although the acoustic problems are manifold. High‐speed digital signal processing (DSP) chips have made it possible to attack the essential problems of forming the beam, aiming the microphone, and tracking a unique talker. A useful beamforming algorithm and a two‐step talker‐tracking algorithm are introduced. In the latter, step 1 is a rather conventional filtered cross‐correlation method; the delay between some pair of microphones is determined to high accuracy using interpolation on the sampled data. Then, using the fact that the delays for a point source should fit a hyperbola, a best hyperbolic fit is obtained using nonlinear optimization. Results indicate that this method works reliably for signal‐to‐noise ratios of less than 10 dB. [This work principally supported by NSF Grant No. MIP‐8809742 and DARPA/NSF Grant No. IRI‐8901882.]
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Active techniques for the suppression and reproduction of sound fields (A)

P. A. Nelson and S. J. Elliott

J. Acoust. Soc. Am. Volume 87, Issue S1, pp. S3-S3 (1990); (1 page)

Online Publication Date: 13 Aug 2005

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The active suppression of unwanted sound and the reproduction of a desired sound field are very similar problems. Recent work on the active control of sound has successfully used multichannel adaptive filters in the feed‐forward control of periodic and quasiperiodic low‐frequency sound in aircraft [S. J. Elliott et al., J. Sound Vib. 128, 355–357] and automobiles [S. J. Elliott et al., Proc. Inter Noise '88 2, 987–990 (1988)]. The essence of the approach can be extended to deal with temporally random sound fields generated by multiple primary sources. The necessary extension of the existing theory will be described and some practical applications of the technique will be discussed. The same theory will also be shown to provide an appropriate framework for dealing with problems in multichannel sound reproduction. In particular, the formulation of the problem allows the specification of the optimal inverse filters that give the minimum mean‐square error in reproducing a sound field recorded at multiple points. Some experimental results will be presented from the implementation of this approach in the two‐channel case by Hamada et al. [Proc. IEEE Workshop on Applications of Signal Processing to Audio and Electroacoustics (1989)].
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An adaptive modal‐based active control system (A)

Dennis R. Morgan

J. Acoust. Soc. Am. Volume 87, Issue S1, pp. S3-S3 (1990); (1 page)

Online Publication Date: 13 Aug 2005

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Active control for silencing sound and vibration has recently become the subject of much investigation, and is now being enhanced by the advent of adaptive filtering techniques. On another front, modal decomposition for active control problems has been shown to offer a number of practical advantages, as independent modal control can minimize the number of actuators, sensors, and corresponding dimensionality of the controller, as well as minimize the control energy; provide robustness to system parameter uncertainty; and, minimize the adaptive convergence time by uncoupling the modal responses. In this paper, details are sketched for a narrow‐band adaptive modal‐based active control system that incorporates all of these features. The major topics addressed deal with modal filters, actuator prefilters, spatial sampling effects, and design methodology. Although this work pertains only to the narrow‐band disturbance problem, these techniques can be readily generalized to the wideband case by using temporal filters to separate out different spectral bands for narrow‐band modal‐based processing. The analytical results are illustrated by numerical example for a vibrating cantilever beam, using a graphics display that was developed for real‐time interactive design.
Contributed Papers
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Volumetric selectivity from two beamforming microphone arrays (A)

G. W. Elko

J. Acoust. Soc. Am. Volume 87, Issue S1, pp. S3-S3 (1990); (1 page)

Online Publication Date: 13 Aug 2005

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A major problem in audio communication between large and intermediate rooms is the degradation of speech by noise and reverberation. The use of beamforming microphone arrays as a possible solution has been reported on earlier. It will be shown that the time‐delay bias in the estimate of the short‐time coherence function between two crossed beams can be used to obtain volumetric selectivity. The approach uses a time‐varying filter that depends on the received data itself. It will be shown that by simply changing the processing block window, “volumetric focusing” can be obtained.
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Recent study of a microphone with range‐dependent sensitivity (A)

Hikaru Date

J. Acoust. Soc. Am. Volume 87, Issue S1, pp. S3-S3 (1990); (1 page)

Online Publication Date: 13 Aug 2005

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The spatially selective multimicrophone system, whose principle was first introduced at the 115th ASA meeting, can be easily modified into a directional microphone with range‐dependent sensitivity by appropriately selected distribution of gains of microphone pairs located on a virtual spherical surface. An optimum choice for realizing a super close‐talking microphone with this range‐dependent microphone is investigated by systematic change of gain distribution through computer program, while keeping the spherical boundary shape. The effect of deviation on sensitivity and position of microphone pairs located on the boundary are also analyzed. Furthermore, the simulation study is also extended to the spheroidal shape of the boundary surface and it is made clear that the ratio of major and minor axes up to 2 is allowable without significantly disturbing the spatial characteristics of the multimicrophone system. The experimental results on the first model, occasionally equipped with an analog signal processing part, is also reported.
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Hands‐free telephone using compact echo cancelers and voice‐switched attenuators (A)

Juro Ohga, Hiroyuki Masuda, Kensaku Fujii, and Yoshihiro Sakai

J. Acoust. Soc. Am. Volume 87, Issue S1, pp. S3-S4 (1990); (2 pages)

Online Publication Date: 13 Aug 2005

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An all‐in‐one type hands‐free telephone set was constructed by using a single chip digital signal processor integrated circuit (DSP IC). It includes a compact acoustic echo canceler, a sidetone echo canceler, and voice‐swilched variable attenuators. An echo canceler whose length is only 16 ms (128 taps for an 8‐kHz sampling) can suppress the power of an impulse response of the loudspeaker‐microphone acoustic coupling path by more than 15 dB, because the major part of the path for an all‐in‐one telephone set is due to short direct paths. Most of the sidetone can also be suppressed by more than 20 dB by using an echo canceler of 8‐ms length. A speech circuit including these echo cancelers and conventional voice‐switched variable attenuators was composed of only one medium‐performance DSP IC (75‐ns machine cycle, 2K words of ROM and 512 words of RAM, and an additional 1000 ASIC gates}, CODEC IC's, and a few analog IC's. Complete simultaneous conversation is feasible with this hands‐free telephone circuit because the voice‐switched attenuation could be reduced to 10 dB or less.
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Tuning logarithmic spectrum analyzers for music analysis (A)

W. F. McGee

J. Acoust. Soc. Am. Volume 87, Issue S1, pp. S4-S4 (1990); (1 page)

Online Publication Date: 13 Aug 2005

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Logarithmic filter banks for the real‐time analysis of musical signals permit the examination of the partials of notes in real time for transcription or analysis using low‐cost digital signal processors. However, many recordings (even CDs) are mistuned, sometimes approaching a quarter‐tone. Therefore, a tuning technique has been developed to allow the alteration of the filter center frequency, or all the filter center frequencies. The basis of the technique is to use a filter bank that results in complex outputs (magnitude and phase). The algorithm detects the imaginary part of the product of the filter output and the conjugate of the product of the filter output at the previous sample time and the expected phase shift. This quantity is positive for a sine wave below the filter center frequency and negative above, thus permitting a correction to the center frequency, and it is readily computed. Applications will be presented using a Motorola 56000 DSP and covering 7 octaves at the highest 85 piano frequencies.
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Results of experiments to detect moving vehicles with a one‐bit correlator (A)

Robert H. Gonter

J. Acoust. Soc. Am. Volume 87, Issue S1, pp. S4-S4 (1990); (1 page)

Online Publication Date: 13 Aug 2005

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Moving vehicles such as automobiles, trucks, and aircraft can be tracked with a one‐bit correlator using a technique similar to pattern velocity computers [G. R. Ochs and O. F. Miller, Rev. Sci. Instrum. 43, 879–882 (1972)]. Using two microphones spaced 30 cm apart, vehicles passing at a distance of 30 m or farther give significant correlation when the signals are in phase. Beamforming is easily done with shift registers so that the direction of movement can be determined as vehicles cross the beams. Although the correlations are confounded when more than one vehicle is passing, approximate vehicle counts were made using an AIM 65 microcomputer. It is possible that aircraft could be tracked by their sound using a three‐microphone array and an oscilloscope with X‐Yinput capability.
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