bluuebunny/crossref_metadata_2025 huggingface dataset:
Coding of Species-Specific Vocalizations by the Squirrel Monkey Auditory Cortex
Authors: Newman John D.
Published: October 2005

Single neurons in the auditory cortex of squirrel monkeys (Saimiri sciureus) were tested with tape-recorded vocalizations representing this species' six major call groups. A small minority of these cells were highly selective, responding only to one or two call types. For these cells, a role in categorizing discrete call types appears likely. Some selective cells behaved as “tuned filters,” by virtue of the correlation between the best frequency of steady tone bursts for a cell and the major acoustic energy in the effective call type. Most of the neurons responded to several vocalizations. A simple scheme of “feature extraction” does not appear to explain readily the responses of these cells: (a) calls effective for a given cell usually represented structurally different call groups and did not share the same simple acoustic features; (b) many cells responding across call group boundaries did not respond to all the calls of an effective call group; (c) while most of the neurons responded to steady tone bursts, their responses to tones were generally not satisfactory indicators of their responsiveness to vocalizations; (d) in some cases, the nature of a cell's response was shown to depend upon interactions of nonadjacent parts of a vocalization.
Physiological Control of Vocalizations in the Japanese Monkey: Inferences from a Field Study
Authors: Green Steven
Published: October 2005

A field study of vocal communication in the Japanese monkey reveals their repertoire to be composed of patterns that intergrade acoustic features. If the repertoire is divided into classes on the basis of a few distinctive features, a regular pattern of association is evident between these classes and the social situation in which they are uttered. The association is not arbitrary. It can be largely accounted for by the internal state of the animals. Uniformly tonal sounds, for example, are given only by unaroused monkeys and are used in situations of affinitive contact. On the other hand, uniformly atonal sounds are given by very agitated animals and accompany aggressive threats. Within the broad framework of covariation of classes of sound with internal state, there are opportunities for small differences between closely related vocal patterns to be utilized socially. Evidence is presented that these opportunities are exploited in a number of ways; for example, minor variations of vocalizations within a single class mirror fine nuances in social usage. The acoustic nature of these minor signal variations can also be interpreted as graded reflections of the physiological-emotional substrate of the vocalizing animal.
Auditory Processing by Dolphins of Signals Related to Signature Whistles
Authors: Hollien Harry , Caldwell Melba C.
Published: October 2005

It is now well established (Caldwell and Caldwell, 1970) that one of the major forms—if not the major form—of dolphin communication is the signature whistle. In an effort to obtain information concerning the types and levels of auditory discriminations dolphins must be able to make if they are to process these calls, ah extended series of related studies is being conducted. In the first investigation, the research focus was on auditory pattern recognition within three domains: those of frequency, time, and acoustic configuration. Specifically, a “master” dolphin whistle was patterned and synthesized—as were three other whistles varying only with respect to their frequency, temporal, or spatial characteristics, respectively. Additionally, sets of synthesized whistles were obtained for several equally spaced intermediate points along each of the three experimental continua; all signals were generated through the courtesy of the Speech Transmission Laboratory, Royal Institute of Technology, Stockholm, Sweden. For the three substudies, an experimental animal (an Atlantic Bottlenose dolphin) was administered 50 trials (25 stimuli each) of the “master” whistle and one of the experimental stimuli. Following completion of this procedure, a second set of 50 trials would compare the master whistle with another set of experimental signals; these sets were always ordered from the most different comparisons to the least different within a substudy. Once all data were obtained from one substudy (frequency, say), the whole procedure was replicated for the next. With one exception, the experimental animal was able to discriminate between all stimuli—even those that were almost identical. A discussion of these results and a review of the direction of subsequent research will be provided.
Temporal Modulation Transfer Functions for Audition
Authors: Viemeister Neal F.
Published: October 2005

Systems analysis has been extensively used to study the spatial and temporal processing of visual inputs and recently has been applied to auditory filtering. Two techniques are discussed in the present paper, which extend this approach to auditory temporal processing. In the “modulation-threshold” procedure, the threshold modulation amplitude of a sinusoidally modulated waveform (either sinusoids or noise) is determined. The function relating this threshold to modulation frequency is a temporal modulation transfer function (TMTF) and ideally is the attenuation characteristic of the temporal processing system. TMTFs obtained with this procedure show only low-pass filtering and are 3 dB down at approximately 10 Hz. In the “temporal probe technique,” the threshold amplitude of a brief probe (either a bandpass filtered or a wide-band pulse) is determined at several temporal locations in a noise masker whose amplitude is sinusoidally modulated. The function relating probe amplitude to its time of occurrence (the masking curve) can be well-fitted by a sinusoid with a dc component. The amplitude spectrum (TMTF) and phase spectrum are determined from masking curves obtained at several modulation frequencies. The empirical TMTF indicates a shorter time constant and steeper attenuation rates than those obtained with the modulation-threshold procedure. [This research was supported by the National Institutes of Health and by a Special Traineeship from NINDS.]
Reaction Time as a Measure of Suprathreshold Masking
Authors: Lanson R. N. , Garrett T. , Matsuo V.
Published: October 2005

Reaction time (RT) to pure tones masked by white noise was determined with two observers. At least five noise intensities, covering a 0–80 dB SL range, were selected; the tone intensity range was similarly divided. Seven tone frequencies from 250 Hz to 8 kHz were investigated. Finger-withdrawal RTs to tone onset were taken on 30 successive light-cued trials at a given combination of frequency and tone and noise intensities, presented monaurally. Noise was present continuously; tone duration was 1 sec, and foreperiod was variable. Data were obtained for intensity pairs yielding less than 1 sec RTs. Noise intensities were run in increasing order; tone intensities varied randomly within a given noise level. At equal signal-to-noise ratios for a given frequency, functions relating RT to tone and noise intensity appeared similar. Equal RT contours showed threshold shifts as a function of noise intensity. Comparison of these results with threshold masking data illustrated the use of RT as a measure of suprathreshold masking.
Implications of Peripheral Frequency Selectivity for Non-simultaneous Masking
Authors: Duifhuis H.
Published: October 2005

Frequency analysis at the level of the VIIIth nerve is so sharp that onset and offset effects in responses to short signals produce a non-negligible extension of the signal duration. Therefore, excitation patterns produced by signals that were separated in time, can overlap. The overlap of excitation patterns can produce masking, in simultaneous as well as in nonsimultaneous conditions. It is suggested that this peripheral part of the masking process can be held responsible for the major part of backward masking and for a part of forward masking. We refer to the above masking as transient masking. The finding that backward masking decreases with increasing probe frequency favors this interpretation.
Temporal Summation for Tones at Threshold
Authors: Wright H. N.
Published: October 2005

The threshold for tones as a function of their duration (500, 200, 100, 50, 20, and 10 msec) was investigated with the tracking method in a population of 50 normal-hearing listeners (25 male, 25 female; 25 right ears, 25 left ears) at nine frequencies from 125 through 8000 Hz under test-retest conditions. The results indicate that temporal auditory summation, at least as measured with the tracking method, conforms with theoretical expectations. There was no variation in the function with frequency, sex, or ear. When the thresholds for individual listeners are normalized to the threshold for 20-msec tones, the variability both among and within subjects is about the same at all frequencies for all duration tones (standard deviation of 2 dB). These results permit further investigation on the effect of other variables (e.g., hearing impairment) on the threshold for tones as a function of their duration. [Supported by a PHS Research Grant from NINDS.]
A Timing Model for the Intensity and Frequency Discrimination Functions
Authors: Luce R. Duncan , Green David M.
Published: October 2005

For a pure-tone signal of intensity I and frequency f⩽1000 Hz, we attempt to arrive at the dependence of ΔI and Δf on I and f. Single fiber neural data for low frequency tones suggest that the pulse trains are approximately renewal process with interarrival times of the form IAT = I/f+X, where I is a geometrically distributed random variable with parameter p and X is a bounded random variable with E(X) = 0 and V(X) proportional to 1/f2. Because the distribution of IATs is almost perfectly multimodal, if the sample is reasonably large, it is usually possible to determine the value of I for each IAT. Since E(IAT) = 1/pf and E(IAT/I) = 1/f, p and f can be estimated from the sample in the obvious way. Using a d′ argument, we derive ΔP and Δf as functions of p and f. Assumptions are made relating p and the sample size to I and f, and from these assumptions are derived formulas relating ΔI and Δf to I and f. These predictions are compared with the existing data. Various theoretical problems are pointed out. [This research was supported by a grant from National Science Foundation.]
Can Amplitude Compression Secure a Distortionless Amplification in Hearing Aids
Authors: Nábělek Igor V.
Published: October 2005

It is generally believed that compression can limit the power output of a hearing aid without adding distortion to the signal. The occurrence of distortion is the main objective against limitation of output by clipping. However, similar distortion can occur also with amplifiers with compression. If the compression reacts to the increase of the input signal with a delay, during such a delay the gain is not reduced, and two possibilities arise: either even a strong signal is fully amplified and no protection occurs, or the amplifier is overloaded and the signal distorted. In the later case, compression reduces the time interval of distortion of strong signals as compared to distortion durations without compression. The duration of distortion can be reduced below perceptable level by the use of a very short attack time. Short release time also needed for short-term dynamic range reduction can have some negative consequencies. Requirements on different parameters of compression for various conditions of use will be discussed, and several examples of responses of hearing aids with compression shown.
A Modest Computer Psychoacoustic System
Authors: Campbell Richard A.
Published: October 2005

A psychoacoustic system, based on a minicomputer (Data General, Nova 1200) is described in which the computer not only controls attenuators and gates but also generates the sinusoidal stimuli by waveform reconstruction. Representative harmonic distortion values for the octave frequencies 125– 8000 Hz (with a digital output rate of 32 000/sec, and eight-bit conversion, and a 16 000-Hz low-pass filter) range from 0.08% to 0.4%, respectively. The psychophysical procedure used involves the random selection of stimulus frequency, undefined random observation intervals, a yes-no response, which is classified into one of eight categories (Hit, Miss, six kinds of False Alarms) with six possible results of each on the up-down adaptive method, on each trial. Despite the rather complex procedure being utilized, use of the direct memory access and interrupt capabilities of the computer result in the processor being idle about 95% of the time. The approach to auditory stimulus generation taken in this system, which minimizes the interfacing and programming attention required for basic stimulus generation, is seen as a powerful aid to many kinds of psychoacoustic research.
Effects of Partial Hamming Function Modulation of Tone Bursts
Authors: Mahaffey Robert B.
Published: October 2005

Some perceptual effects of Hamming function modulation of pure-tone bursts of short duration were presented at an earlier ASA meeting [Spring 1967]. In the present studies, subjects' pitch percepts were measured and thresholds were obtained for tone bursts having contours of the following descriptions: (1) full Hamming function modulation; (2) full rectangular function modulation; (3) first half—Hamming function modulation, last half—rectangular function modulation; and (4) first half—rectangular function modulation, last half—Hamming function modulation. The full-cycle Hamming function bursts provided the least variance in pitch perception measurements and the highest thresholds. The rectangularly modulated tone bursts afforded the greatest variances in pitch measurements and the lowest thresholds. The effect to tone bursts having contours beginning as Hamming functions and ending as rectangular functions approached those of the full Hamming function modulations, whereas the effects to the tone bursts having contours beginning as rectangular functions and ending as Hamming functions approached those of the rectangular bursts. On the basis of preliminary data from hearing-impaired patients, clinical implications are cited relative to the threshold shifts obtained with Hamming function bursts as a function of burst duration and of tone frequency.
The Pattern-Transformation Model of Pitch
Authors: Wightman Frederic L.
Published: October 2005

This report describes a new approach to pitch perception, based on a kind of auditory pattern recognition. The general approach is formalized in a mathematical model, the “pattern-transformation model.” In the first stage of this model, an acoustic stimulus is assumed to be “transformed” by the sense organ into what is called a neural activity pattern. This pattern is thought to represent roughly the power spectrum of the waveform. Thus, the model is phase insensitive; the temporal fine structure of the stimulus waveform is ignored. The first-stage pattern (power spectrum) is then assumed to be Fourier transformed (neurally) into another pattern of neural activity. This second pattern roughly represents the autocorrelation function of the stimulus. Pitch is derived from the positions of maximal activity in this pattern. From preliminary tests, it appears that the model can accurately predict the pitches of many types of acoustic stimuli. In addition, the model provides numerical estimates of pitch “strength” and “ambiguity.” While experimental confirmation of these predictions is lacking, the strength estimates agree at least qualitatively with available data. [Research supported by a National Science Foundation Postdoctoral Fellowship awarded to the author and the Institute for Perception—TNO, Netherlands.]
Prediction of Cubic Difference Tones from a Model of Pitch Perception
Authors: Ravindran A. , Thomas Ian B.
Published: October 2005

Using a model of pitch perception already shown to give accurate predictions of residue pitches and difference tones [83rd Meeting, J. Acoust. Soc. Amer. 52, 141(A) 0972)], we have been able to predict cubic difference tones for frequency ratios of the primary tones from 1.167 to 1.500. The model involves the determination of axis crossing intervals of the stimulus at an appropriate basilar membrane (BM) location through Flanagan's BM transfer function, taking into account both amplitude and phase changes. The model notes and marks the time of significant differences in adjacent axis crossing intervals of the signal at the appropriate BM location. Time intervals corresponding to cubic difference tones are found between these time marks. In almost all cases investigated, the axis crossing intervals must be determined at a BM location corresponding to the lower primary frequency. Contrary to our earlier statement, the prediction of cubic difference tones from this model is achieved without any assumption of nonlinear distortion mechanisms. [Work supported by NIH.]
High Noise Reduction Ear Enclosures
Authors: Allen C. H. , Malme C. I.
Published: October 2005

Personnel associated with operation and maintenance of combat vehicles and military aircraft are subject to noise environments so severe that the available ear protective devices do not provide adequate noise reduction. A study and development program has been conducted to improve available noise reduction by (1) design of an ear enclosure cushion to provide a better seal at low frequencies; (2) provision of a secondary seal inside the earcup between the communications earphone and the ear canal, and (3) utilization of electronic noise cancellation to improve further the noise reduction at low frequencies. The design evolved from this study features a low-profile ear enclosure in which the earcup and cushion form a smooth continuous shell that is curved inward to rest behind the pinna. The cushion is fabricated from material which has frequency-dependent viscoelastic properties. It is compliant at low frequencies to provide a good seal, but stiff at autio frequencies to act as an extension of the earcup wall. The design requirements that have been met provide for better than 20 dB noise reduction at low frequencies and better than 45 dB at high frequencies. [Supported by the U. S. Army, under contract.]
Noise and Pure-Tone Attenuation of Hearing Protectors
Authors: Bolka D. F. , Michael P. L.
Published: October 2005

The acoustic attenuation characteristics of five ear muffs and four ear plugs were measured. Ten experienced listeners performed threshold shift determinations using pulsed pure tones and 13-oct bands of noise at 0° and 90° incidence in anechoic conditions and the same bands of noise in semireverberant conditions. Sound field diffuseness was determined by a directional microphone technique and subjectively. Ear muff attenuation at 1 and 4 kHz was significantly lower (p<0.05) for the noise bands in diffuse field than for frontal incidence pure tones in free field. This difference was not observed for ear plugs. The subjects agreed that threshold determinations were no more difficult with noise bands in diffuse field than with pure tones in free field; variability was somewhat lower for the noise band stimuli. Physical measurements of ear-muff attenuation on an “artificial head” test fixture exhibited statistically significant rank correlation with diffuse field subjective measurements at test frequencies of 2 kHz and below, although absolute attenuation values showed only fair agreement. The diffuse field subjective method and the “artificial head” method are being recommended for ANSI standardization. [This research was sponsored by DHEW, Public Health Service.]
Temporal Order Discrimination: Identification and Recognition by Trained Listeners
Authors: Warren Richard M.
Published: October 2005

Four trained subjects listened to successive sequence bursts, each burst consisting of three component items (filtered 1000-Hz square wave; noise band, 2500-Hz tone) repeated several times without pauses between items, until they decided whether alternate bursts had these items in the same or different order. Prior training consisted of listening to known same or different sample sets as requested by the subject until he indicated readiness to receive these sets as unknowns. Same/different recognition was significantly above chance for at least half of the subjects at each of three interburst intervals (50 msec, 1.5 sec, 3 sec) for item durations from 5 through 400 msec. With items shorter than 100 msec, the order could not be identified, and judgments were based on pattern “quality.” In separate experiments with single stimulus bursts, four listeners were trained by receiving immediate feedback concerning their accuracy in identifying (naming) order. For two item sequences (square wave and noise), correct identification was obtained for item durations from 5 through 400 msec. When these stimulus pairs were preceded and followed by a tone of the same duration as the test items, accurate identification was obtained for item durations from 10 through 400 msec. [Research supported by NSF.]
Internal-Combustion Engine Induction and Exhaust System Noise
Authors: Bender Erich K. , Brammer Anthony J.
Published: October 2005

The importance of induction and exhaust systems on community and passenger noise exposure and engine performance are introduced. Discussion is given of the types of engines and induction/exhaust systems they use. They include naturally aspirated two- and four-stroke engines, supercharging, turbocharging, and a special discussion on the similarities and differences between the Wankel engine, rotary engine, and others. The effects of mufflers on engine performance and their noise reduction capabilities are discussed.
Overview of the Two Special Sessions on Ultrasonic Energy and Power Measurements
Authors: O'Brien W. D.
Published: October 2005

The applications and uses of ultrasonic energy in the clinical practice of medicine, in industry, and in consumer products are increasing at an accelerated rate. Thus, it is essential, in many cases, that both the manufacturer and user have the knowledge and capability to measure accurately, reproducibly, and easily, both spatially and temporally the intensity (W/cm2), power (W) and/or energy (J) of the ultrasonic field. The purpose of the two special sessions on energy and power measurements of ultrasound is to present and discuss the radiation force, optical, calorimetric, and piezoelectric methods for measuring ultrasonic field parameters in an aqueous medium. Each method will be introduced with a tutorial talk describing the particular physical phenomenon which is being used to examine the ultrasonic field. This will be followed by a panel discussion of specific problem areas, during which time audience interaction will be encouraged. The principal ultrasonic frequency range to be discussed is 100 kHz to 30 MHz (c≅1500 m/sec). Both continuous and pulsed wave modalities will be considered. As guide-lines, the typical continuous wave intensity range is several hundred μW/cm2 to tens of W/cm2. For pulsed ultrasonic waves, duty cycles as low as 0.05%, pulsewidths as narrow as a couple of cycles and with peak intensity range a few mW/cm2 to hundreds of W/cm2 are considered.
TM Spectra: A Measure of Diffuseness of Reverberant Sound
Authors: Lubman David
Published: October 2005

Diffuseness of reverberant sound is a sine qua non for accurate reverberation room measurement of sound absorption, power, and transmission loss. Most authorities regard it as important for the concert hall as well. And yet, technique for the measurement of diffuseness is not well developed. In this connection, the recently proposed technique of TM (traversing microphone) spectroscopy shows promise. The technique exploits information on diffuseness contained in the spatial correlation function of instantaneous sound pressure. Mathematically, the TM spectrum is the Fourier transform of the spatial correlation function. Physically, it can be interpreted as the power spectrum obtained at the terminals of a small microphone traversing the room at constant speed. For an ideally diffuse field of narrow bandwidth, the TM spectrum is, conveniently, a perfect rectangle. For a less diffuse field, deviations from the rectangular shape will occur. Supporting experimental and analytical results are shown. The limitations and promise of this new technique are discussed. [This work supported by the Office of Naval Research, Code 468.]
True rms—The Digital Approach
Authors: Lapointe J. A.
Published: October 2005

Acoustical data typically extend over an extremely wide dynamic amplitude range, because of either source amplitude variations or of power spectral variations. Furthermore, it is highly desirable to afford the luxury of an amplitude range surplus to allow for the unknown level of a source that is to be measured. Digital detection can now provide a 70-dB input dynamic range plus true rms performance with simultaneous multiple-channel operation. This capability is made possible by the integrated circuit and by breakthroughs in the design of computational algorithms that are needed to perform rms calculations. Many analog devices have been built in the past to approximate such performance, but none can come close to the results obtainable with a straightforward digital approach. By performing the nonlinear operations of squaring and decibel conversion digitally, accuracy and dynamic range problems are greatly alleviated. Many other advantages also accrue, such as multiple-channel operation and flexible control of the averaging or measurement interval, and will be discussed.
Techniques for Processing EMG Signals
Authors: Port D. K.
Published: October 2005

Analysis of electromygraphic activity is potentially a fruitful technique for investigating speech production. However, the EMG signal picked up by bipolar wire electrodes is weak and inherently variable and must be carefully handled. At Haskins Laboratories a computer processing system was developed to analyze simultaneously up to eight EMG channels of data. This system allows for identifying, sampling, and digitally averaging EMG signals for a set of utterances of the same type. Integration of the EMG signals, both in the hardware and software, has been recently revised and its importance to the interpretation of EMG signals will be discussed. The relationship between an averaged EMG signal and the individual utterances in the sample will be discussed in the context of editing and display capabilities. Further, an experiment involving multiple placement of electrodes in both muscles of a bilateral pair has been conducted. Computation of correlation coefficients for utterances across the multiple electrode placements shows that one electrode can provide EMG activity representative of activity in the entire muscle. This is true for both individual utterances and for the average of a set of utterances. [This research was partially supported by the National Institute of Dental Research.]
Intermodulation Distortion Produced by Center Clipping
Authors: Mitchell O. M. M. , Yates G. H.
Published: October 2005

Recently, center clipping has been used in a number of speech processing systems [O. M. M. Mitchell and D. A. Berkley, J. Acoust. Soc. Amer. 47, 84 (1970); O. M. M. Mitchell and D. A. Berkley, Bell System Tech. J. 50, 1619–1630 (1971)]. In this paper, we apply Bennett's double Fourier series technique [H. E. Rowe, Signals and Noise in Communication Systems (Van Nostrand, Princeton, N. J., 1965)] to obtain integral expressions for the distortion products of two center clipping transfer functions for the case of two simultaneous sinusoidal inputs at arbitrary frequencies. Results of numerical solutions are presented for six clipping levels and six values of relative amplitude of the two sinusoidal inputs.
Low Bit-Rate Coding from the Autocorrelation Method of Linear Prediction
Authors: Markel J. D. , Gray A. H.
Published: October 2005

The term “autocorrelation method” encompasses the inverse filter, maximum likelihood, and PARCOR method. This method has a significant computational advantage over Atal's covariance method since the analysis does not require polynominal root testing programs and associated modifications, and is thus much easier to implement. Informal listening tests indicate that for low bitrate coding, the resulting synthetic speech is very similar for both methods with fixed frame analysis. Directly from the autocorrelation equation solution, the filter parameters, k parameters (also referred to as reflection coefficients and PARCOR parameters) are obtained, along with the synthesizer gain parameter. Low bit-rate coding based upon the k parameters has been obtained by performing the following steps: (1) prefilter sharply at the half-sampling frequency; (2) prewhiten and Hamm before analysis; (3) transform the parameter values to [ −0.7, 0.7]; (4) quantize with rounding; and (5) interpolate synthesis parameters pitch synchronously. These operations are all very fast and easily implemented. Good quality synthesis over a 3200-Hz input bandwidth is obtainable with [ki BITS, i = 0,1, ⋯,7,] = [6,6,3, ⋯,3]. Examples coded at 2100 and 1400 bits/sec (50-and 33-Hz frame rates, respectively) are presented.
The Application of Dither to the Quantization of Speech Signals
Authors: Rabiner L. R. , Jayant N. S. , Johnson J. A.
Published: October 2005

By adding a pseudorandom “dither” noise to a signal X that is to be quantized, and by subtracting an identical noise sequence from the quantizer output, it is possible to break up undesirable signal-dependent patterns in the quantization error sequence, without increasing the variance of the error E. The idea has been widely discussed in the context of picture coding, and it is the purpose of this talk to demonstrate application of the technique to the quantization of speech signals. Computer simulations have shown how the use of dither whitens the quantization error sequence in PCM encoding, and renders it more acceptable than signal-correlated errors of equal variance. We demonstrate, for conditions of dither and no dither, typical speech recordings, illustrative error waveforms, and data on signal-to-error correlation C; and indicate how the advantage of dithering increases monotonically with crudeness of signal quantization and becomes significant when the number of bits per sample is less than about six. An experimental evaluation of the perceptual effects of dither on speech has shown (a) strong preferences for dithered speech over straight PCM encoding at identical bit rates; (b) an increase in word intelligibility for dithered speech over PCM speech when 4–6 bits/sample were used; and (c) a decrease in word intelligibility for dithered speech over PCM speech when 2–3 bits/sample were used. The above results imply that dithering is most advantageous for speech quantized in the 4–6 bits/sample range.
Adaptive Quantization in Differential PCM Coders
Authors: Cummiskey P. , Flanagan J. L.
Published: October 2005

A differential PCM coder is described in which the quantizer is automatically adapted to the error variance. The transmitted bit stream is examined by digital logic which in turn adjusts the quantizer step size to reduce slope overload and granular distortion. For a given bit rate, increased quality over PCM and DPCM coding results. The receiver incorporates adaptation logic identical to that of the transmitter, and reconstructs a signal estimate similar to that made by the transmitter. Digital hardware implementation of the coder is described, and transmission performance at rates of 18, 24, and 32 kilobits/sec is demonstrated.
Some Baseband Speech Processing Experiments
Authors: Coulter D. C.
Published: October 2005

Work with a voice-excited formant vocoder has led to experiments on several other methods of processing speech. These methods make use of the baseband and some of the same types of information which are normally derived by a formant vocoder, such as front-back vowel detection and voice-voiceless discrimination, but applied in a manner different from a normal formant vocoder. The systems so far tested are characterized by transmission of the baseband in unprocessed form and the regeneration of higher frequency information by deriving it from the baseband or simple parameters. While these systems do not yield high quality speech, the intelligibility may be high enough for certain limited applications. Tape recordings of these systems will be presented.
Signal Processing Techniques Based upon a Wave Packet Representation
Authors: Retz D. L.
Published: October 2005

The method of performing a time-domain approximation of the speech wave in terms of a linear combination of Gaussian wave packets is evaluated as a signal processing tool. This set of functions is used because of their ease of extraction from the original signal and their versatility. The parameters defining the set of functions are determined from the times and magnitudes of extrema in the band-limited signal. Because the wave packets used to represent the signal are analytically transformable to the frequency domain, various characteristics, such as the Fourier spectrum and the autocorrelation function, may be evaluated directly from the parameters describing the wave packets. The economics and accuracy of this technique are discussed.
Evaluation of Speech Processing Innovations to Improve Voice Quality and Intelligibility in Narrow-Band Voice Communication
Authors: Smith Caldwell P.
Published: October 2005

Using an experimental hardware channel vocoder as a test bed, a systematic series of tests including formal and informal listener evaluations were conducted to evaluate the merits of several speech processing innovations towards maximizing speech quality and intelligibility. A superior combination was found in a technique that enhances spectral peaks prior to digitization [a refinement of an old concept: C. P. Smith, J. Acoust. Soc. Amer. 23, 446–451 (1951)] together with a synthesis excitation arrangement using oscillators synchronously phase reset with each pitch excitation pulse [C. P. Smith, J. Acoust. Soc. Amer. 37, 170–171 (1965)], with added refinements for enhancing phase coherence in onsets of pitch epochs in voiced sounds. Although pitch extraction and coding were relatively crude (300 bits/sec pitch data rate, and occasional pitch errors), this combination yielded superior results such that quality and intelligibility at 2400 bits/sec, depending on the speaker, either exceeded or was fully comparable to that of a good 9600 bits/sec voice-excited vocoder. These findings suggest a potential for refining 2400 bits/sec hardware channel vocoders to derive superior performance/cost tradeoffs relative to any other existing technology. [This work was performed in the now-terminated Digital Speech Processing Branch of Air Force Cambridge Research Laboratories.]
Zero Quefrency in Speech Analysis
Authors: Makhoul John
Published: October 2005

The zero component of the cepstrum (transform of the log spectrum) is also known as the zero quefrency c0. c0 contains information about R0 (the total energy in the signal), as well as the shape of the signal power spectrum. The energy information can be removed by subtracting R0 from c0. If that difference is exponentiated, one obtains a normalized parameter V = ec0/R0. V can be interpreted as the ratio of the geometric mean to the arithmetic mean of the spectrum. For any spectrum, it can be shown that 0⩽V⩽1. The value of V depends only on the spectral shape. By taking advantage of the fact that most voiced sounds have the energy concentrated at low frequencies, while most unvoiced sounds have the energy relatively spread out across the spectrum, the parameter V can be used as a voicing detector. A detailed analysis of the behavior of V as a function of different spectral shapes can be found in Makhoul and Wolf [BBN Rep. No. 2304 (1972)]. Conditions under which V can be used as a voicing detector will be discussed.
Speech Quality of Computer-Simulated Voice-Switched Amplifiers
Authors: Silbiger Herman R. , Cullingford Richard E. , Pierce Linda
Published: October 2005

Voice-switched amplification can be used in telephone circuits to provide additional amplification in the direction of transmission only when speech is present and thus increase the apparent S/N ratio. The effect on speech quality is a function of operating characteristics such as attack time, release time, switching thresholds, and amount of switched amplification. To evaluate the detectability and speech quality with such amplifiers, subjective tests were conducted in which computer simulation was used to provide convenient and accurate control of the operating characteristics. In the detection tests, the subjects determined whether they could hear the operation of the voice-switched amplifier, using a modified double staircase method with variable step size. The results were used to map out a region in which the operation of the switched-gain device was not detectable. A category judgment procedure using a 5-point scale was used for the transmission quality listening tests in the previously determined detectability region. Decay times of 25, 100, and 300 msec were presented with switched gains of 0, 10, 15, and 20 dB, over circuits with S/N ratios of 10 and 30 dB. Results indicated that for a 10-dB S/N ratio, the effects of amount of gain switched and decay times were both significant and had a strong interaction. For a 30-dB S/N ratio, only the effect of decay times was significant. The 10-, 15-, or 20-dB switched gain conditions were always preferred over the 0-dB condition.
Calculating the Wave Field Diffracted at the Boundary of a Shadow Zone
Authors: Hudimac A. A.
Published: October 2005

The conventional ray theory often generates shadow zones. Previously the author has given the density of simple sources and dipoles whose radiation accounts for the diffraction at shadow boundary. For a stratified fluid, the boundary is a contiguum of parallel rings starting with one tangent to the shielding plane. Three classes of rings are treated. Those nearly the same distance from the obstructing plane as is the field point are treated by an approximate wave theory. Their contribution is insignificant except for points near the boundary. Those beyond contribute insignificantly. Those nearer are treated by an approximate ray theory. Most of this diffraction comes from the rings near the tangent ring. For a field point inside the boundary, ray rings all the way to the tangent ring contribute, and the diffraction drops off slowly with distance from the boundary. As the point moves into the shadow zone, more rings near the tangent plane fail to contribute and the field drops off rapidly. The degree of accuracy of other types of calculations (e.g., Rytov, phase integral) shall be discussed.
Ray and Wave Solutions for Bottom Reflection for Linear Sediment Layers
Authors: Morris Halcyon E.
Published: October 2005

Agreement between experimental and calculated values of bottom loss have been shown in earlier work [H. E. Morris, J. Acoust. Soc. Amer. 48, 1198–1202 (1970)]. The mathematical treatment involves solution of the wave equation for a plane wave incident on a set of absorbing sediment layers with velocity gradients. In order to clarify the physical mechanisms that produce the total reflected sound beam, a ray theory solution is presented in which the sound that reflects for the water-sediment interface can be separated from sound that is refracted in the sediment layers. Curves showing the contributions of reflected arrival and various orders of refracted arrivals are presented for several cases of interest.
Sound Propagation along a Sloping Bottom in Shallow Water
Authors: King Louis A. , Browning D. G.
Published: October 2005

One of a series of sound propagation experiments recently conducted by the Naval Underwater Systems Center in the Gulf of Aaen took place across a basin with a sloping bottom. Three receiving hydrophones were located near one edge in 60 m of water. Results of this experiment in which bomb shots were used were compared with both a propagation model based on ray theory by J. P. Jones and P. W. Smith, Jr. (Bolt Beranek and Newman TM 84 of 7 July 1971) and a model based on normal mode theory. Since the received signal appeared as an impulse followed by a reverberant portion, each with different spatial properties, the two segments were treated separately.
Displacement First Passage of the Transient Response of Linear Vibration System with Stationary Random Exictation
Authors: Jacquot Raymond G.
Published: October 2005

The first-passage problem for a linear oscillator starting from rest excited by stationary wide band noise is examined. A Monte Carlo simulation technique is employed to obtain first-passage time statistics. An error analysis ia accomplished, and it is found that probabilities on the order of 0.04 may be predicted with 1000 response sample functions with a probable error of less than 2%. Upper and lower bounds on the first passage probability of other investigators are applied, and these suggest a simple empirical equation form to fit the first-passage distribution function. This equation involves two coefficients and Rice's probability of an upward crossing of the level of interest in the elemental interval (t, t+dt). These coefficients are evaluated for a large number of combinations of barrier level and oscillator damping ratio. Extension to a class of multidegree of freedom systems is discussed.
A Nonlinear Model for Basilar Membrane Motion
Authors: Kim Duck On , Molnar Charles E. , Pfeiffer Russell R.
Published: October 2005

A phenomenological model for basilar membrane motion is presented. The model consists of 10 cascaded elements defined by the nonlinear differential equations ẍi(t) + 2Di[1 + ηẋi2(t)]ẋi(t) + ω0i2xi(t) = Cxi−1(t) for i = 1, 2, ⋯, 10, where x0(t) is the input (stapes displacement) and x10(t) is the output (displacement of a point on the basilar membrane). The model demonstrates a close relationship among seemingly disjoint nonlinear phenomena of the peripheral auditory system by showing the following response characteristics: (1) saturation of the output amplitude, decrease of Q, and decrease of the maximally responsive frequency with increase of input amplitude; (2) “amplitude” and “temporal” nonlinearities in response to combination clicks; (3) generation of combination tone 2f1 − f2 in response to inputs of two equal-amplitude tones (f1 < f2); (4) two-tone suppression for frequencies of the suppressing tone above the characteristic frequency; (5) change of the phase angle of the output with increasing amplitude of sinusoidal input; and (6) change of click response pattern with increasing click amplitude. The characteristic (1) has already been observed directly in basilar membrane motion. We suggest that the characteristics (2)–(6), which are similar to nonlinear phenomena observed in spike-discharge patterns of single cochlear nerve fibers, may actually be properties of basilar membrane motion.
Local Potentials Recorded from the Tunnel of Corti
Authors: Lawrence Merle , Nuttall Alfred L. , Clapper Maxine
Published: October 2005

Microelectrodes are inserted into the cells and fluid spaces of the organ of Corti under direct visual control monitored by closed-circuit television. Local dc resting potentials, tissue-electrode impedance, and ac potentials resulting from fluid vibrations applied by two methods are simultaneously recorded. Vibrations are applied either directly to the stapes or to the basilar membrane in the area of electrode insertion. The ac response recorded from the polarized cells and spaces of the organ of Corti are different depending upon whether the ac is locally or remotely generated. The magnitude and distribution of dc resting potentials are greatly dependent upon the local physiological condition of the exposed organ of Corti. Of most importance are an intact capillary system in the local area and good general physiological condition of the animal. Location of electrode tip is determined visually followed by microscopic examination of surface preparations.
Comparison of Effects of Biasing the Cochlear Partition with Electrical Current and Sound Stimulation on the DIF SP Responses
Authors: Durrant John D. , Dallos Peter
Published: October 2005

Summating potential responses were measured utilizing the differential recording technique. Two experimental designs were incorporated, one in which the cochlear partition was electrically biased with direct current and the other in which the movement of the partition was “biased” with a low-frequency sound stimulus. The DIF− and DIF+ components were of prime interest. It was found that dc polarization and the mechanical bias have strikingly similar effects on the SP under certain conditions. The SP can be reversibly augmented or depressed according to the direction of current flow or direction of the biasing displacement, and, furthermore, the two DIF components exhibit change in the same absolute direction. A decrease in the DIF− and an increase in the DIF+, and vice versa, is produced by a given biasing stimulus. It is surmised that the net biasing effect is such that when the scala media potential is caused to increase (directly by current or indirectly by deformation of the cochlear partition), the DIF SP responses are made more negative, and more positive when the scala media potential decreases. The similarity of the changes in these SP responses observed under the two forms of biasing suggests that they act on essentially the same distortion process, which appears to be primarily of electromechanical origin. [Supported by grants from the National Institute of Neurological Diseases and Stroke.]
Ethacrynic Acid Ototoxicity, Potentiation by Kanamycin
Authors: Prazma Jiri , Browder J. P. , Fischer N. D.
Published: October 2005

Endocohlear potentials (EP), cochlear microphonics (CM), and action potentials (AP) have been recorded from guinea pig cochleas in animals anesthetized by Nembutal. The experimental animals were divided into four groups for two acute experiments and two chronic experiments. In the acute experiments, groups 1 and 2 received 50 or 100 mg/kg of Kanamycin and 10 mg/kg Ethacrynic acid, both drugs twice at 1 h intervals. In the chronic experiments, groups 3 and 4 received 50 or 100 mg/kg/day of Kanamycin for 14 days, and on the last day, after evaluation of base line potentials, received 10 mg/kg of Ethacrynic acid only. Each drug administered alone was not ototoxic or was only mildly ototoxic; however, combination treatment by both drugs potentiate the ototoxic effect. Detailed findings on cochlear potentials will be described. [This research was supported by NIH grants.]
Nature of the Cochlear Changes Induced by Ethacrynic Acid
Authors: Bosher S. K. , Smith C. , Warren R. L.
Published: October 2005

A comprehensive study of the effects of ethacrynic acid upon the cochlea in a single species is being undertaken to determine more exactly the nature of its toxic action. In the rat, the drug produces rapid replacement of the endocochlear potential by a negative potential and, while recovery is initially rapid, its rate decreases considerably at 30 min. The endolymphatic chemical composition remains unchanged, however, until 35 min, when increase in the sodium and decrease in the potassium concentrations occurs. The drug is consequently considered to cause transient inhibition of the strial enzymes, followed by alterations in membrane permeability. The ultrastructural strial anomalies are associated with and thought to arise from the permeability changes. The microphonic responses and morphological examination provide evidence of independent hair cell involvement, but its nature and relationship to the endolymphatic changes is uncertain. In addition, the findings strongly support the concept that the endocochlear potential has two components, a positive secretion potential and a negative potassium diffusion potential.