摘要:

It is generally accepted that even trained listeners have great difficulty perceiving static phase differences in two‐tone stimuli. However, if the waveform with the changed phase is “inserted” into a longer signal, most observers can reliably detect whether the change has been made. This study used this “insert” method to compare a series of waveforms which differed from a standard waveform by 15°, 30°, 45°, 60°, 75°, 90°, 102°, 150°, and 180°. The standard waveform wasA sin2π(250)t+B sin[2π(500)t+90]. By the time the comparison waveform differed from the standard by 60°, observers could perceive the phase change correctly on over 75% of the trials. An attempt is made to identify the cues which permit the system to recognize such nonenvelope phase changes, but the data do not permit positive identification of these cues.

ISSN:0001-4966    

DOI:10.1121/1.1912672

出版商:Acoustical Society of America    

年代:1971

数据来源: AIP

摘要:

The function relating interaural signal‐frequency difference (Δf) to masking‐level difference (MLD) was obtained using a two‐alternative forced‐choice procedure. The masking noise was presented diotically (N0) at a spectral level of 45 dB. The signals to the two ears were gated simultaneously and each was 1 sec in duration. For all Δfconditions, the signal to one ear was 400 Hz. In separate sessions, signal frequencies of 406, 420, 440, 460, 500, 550, 600, and 700 Hz were presented to the other ear. The MLD was found to decrease from about 10 dB at Δf= 6 Hz to about 7 dB at Δf= 150, 200, and 300 Hz. This limiting value of the MLD is also the MLD for the 400‐Hz signal presented monaurally (N0‐SM).

ISSN:0001-4966    

DOI:10.1121/1.1912673

出版商:Acoustical Society of America    

年代:1971

数据来源: AIP

摘要:

Glass micropipets (filled with 2MKCl) were inserted through the round window of anesthetized cats. As the micropipet was advanced through the cochlea, we recorded (1) depth of penetration of the micropipet, (2) resistance of the micropipet, (3) dc potential, (4) magnitude and phase of the fundamental component of the response to tones, and (5) magnitude of response from a stationary electrode. At the end of the experiment, the micropipet was cemented to the temporal bone and kept in place during histological preparation. From examination of 54 sectioned cochleas, we determined which structures in the cochlea had been penetrated by the micropipet. The inaccuracies involved in determining the location of the tip of the micropipet with this technique made it impossible, however, to associate recorded electric events with structures as small as individual cells. The average dc potentials (referred to scala tympani) were scala media, +103 mV; scala vestibuli, −2 mV. Occurrences of large negative dc potential (−50 to −100 mV) were frequently recorded between scala tympani and scala media; these potentials were generally stable for a short time only (<10 sec).

ISSN:0001-4966    

DOI:10.1121/1.1912674

出版商:Acoustical Society of America    

年代:1971

数据来源: AIP

摘要:

Cochlear potential response to tones was recorded with glass micropipets in all three scalae of the basal region of the cochlea of 32 anesthetized cats. The fundamental component of the potential (referred to an electrode on the chin) is represented by complex amplitudesEV, EM, ET, where the subscripts denote the scala in which the potential was recorded.EM/EVis relatively independent of stimulus level and frequency and has a magnitude of approximately 1.0–1.8 and an angle of approximately zero.EM/ETis dependent on stimulus level and frequency. Its magnitude is as large as 10 at frequencies below 300 Hz and decreases for higher frequencies. The angle of the ratio increases from approximately zero at 100 Hz to a value between +90° and +180° above 1000 Hz. Similar results were obtained from two cochleas with a severed auditory nerve, which implies that these results apply to the cochlear microphonic (CM) potential response to tones. An electric‐network model of the spatial distribution of CM is analyzed.

ISSN:0001-4966    

DOI:10.1121/1.1912675

出版商:Acoustical Society of America    

年代:1971

数据来源: AIP

摘要:

Cochlear potential response to tones of known sound pressure was recorded with micropipets from the three scalae of the basal region of the cat's cochlea. By using an average transfer function for the middle ear of the cat, we calculated the complex ratio of cochlear potential response to stapes velocity,E/Ẋs. |E/Ẋs| is approximately constant for a decade or two of frequency in the range 100 Hz–10 kHz. As frequency decreases below 100 Hz, |E/Ẋs| decreases and ∠ (E/Ẋs) becomes positive. As frequency increases above a few kilohertz, |E/Ẋs| decreases and ∠ (E/Ẋs) becomes negative. The dependence ofE/Ẋson frequency for potential in scala media is somewhat different from that in scala tympani. To interpret our measurements, we postulate a model that relates cochlear microphonic potential,V(x, f), to stapes displacement,Xs(f), in terms of three cascaded transformations: mechanical,H(x, f); transduction,T(f); and electric,W(x,f). According to the model, the transfer function can be expressed asV(x,f)/Xs(f) = H(x,f) ⊗[T(f)W(x,f)], where ⊗ denotes convolution onx(distance from stapes). Special cases of this relation are discussed and compared with experimental data.

ISSN:0001-4966    

DOI:10.1121/1.1912676

出版商:Acoustical Society of America    

年代:1971

数据来源: AIP

摘要:

A new experimental technique is proposed to estimate the bandwidth of the auditory‐filter mechanism when detecting short‐duration sinusoidal signals masked by wide‐band noise. The experiment requires the use of only wide‐band noise and two signals of the same frequency, power, and duration, but having different energy density spectra. The special feature of this technique is that the subjective impression made by the masking noise remains the same throughout the experiment unlike experiments that employ progressively changing bandwidths of the masking noise. This avoids the possibility of any change in the parameters of the detection mechanism due to changes in the subjective impression made by the noise. The analysis makes use of the energy detection model consisting of a bandpass filter, square‐law rectifier, and an integrator to describe the auditory detection system. A rectangular filter and an LCR single tuned filter have been considered in the analysis. From the experimental results it is found that the rectangular filter provides a better description of the auditory‐filter mechanism than the single tuned filter. The bandwidth estimates show an increase with signal frequency. The results also support the idea of a minimum auditory‐filter bandwidth for a continuous or long‐duration signal and increased values for short‐duration signals depending on the signal duration.

ISSN:0001-4966    

DOI:10.1121/1.1912677

出版商:Acoustical Society of America    

年代:1971

数据来源: AIP

摘要:

Several experiments have shown that the difference in masked threshold between signals in the S0M0 and SπM0 conditions, the masking‐level difference (MLD), increases markedly with reduction in masker bandwidth. However, there are two inconsistencies this result. First, apparently similar experiments conducted by different laboratories have produced quite different results. Second, the MLD‐bandwidth relation appears to be nonmonotonic; with very narrow‐band maskers (e.g., a pure tone), the MLD is often small. The results of the experiments reported here suggest that these problems arise from anomalies in the S0M0 data alone. It appears that in some narrow‐band masking conditions the SπM0 threshold is spuriously low, resulting in deceptively small MLDs. The proposed explanation is that observers detect signal energy which falls outside the masker band, for when this energy is attenuated by filtering the signal, the S0M0 threshold rises dramatically. The SπM0 threshold is unaffected by this manipulation. With specially filtered signals, the nonmonotonicity of the MLD‐bandwidth relation is considerably reduced, and MLDs with continuous pure‐tone maskers are observed which are nearly as large as the largest narrow‐band noise MLDs ever reported (28 dB).

ISSN:0001-4966    

DOI:10.1121/1.1912678

出版商:Acoustical Society of America    

年代:1971

数据来源: AIP

摘要:

We describe a procedure for efficient encoding of the speech wave by representing it in terms of time‐varying parameters related to the transfer function of the vocal tract and the characteristics of the excitation. The speech wave, sampled at 10 kHz, is analyzed by predicting the present speech sample as a linear combination of the 12 previous samples. The 12 predictor coefficients are determined by minimizing the mean‐squared error between the actual and the predicted values of the speech samples. Fifteen parameters—namely, the 12 predictor coefficients, the pitch period, a binary parameter indicating whether the speech is voiced or unvoiced, and the rms value of the speech samples—are derived by analysis of the speech wave, encoded and transmitted to the synthesizer. The speech wave is synthesized as the output of a linear recursive filter excited by either a sequence of quasiperiodic pulses or a white‐noise source. Application of this method for efficient transmission and storage of speech signals as well as procedures for determining other speech characteristics, such as formant frequencies and bandwidths, the spectral envelope, and the autocorrelation function, are discussed.

ISSN:0001-4966    

DOI:10.1121/1.1912679

出版商:Acoustical Society of America    

年代:1971

数据来源: AIP

摘要:

A binaural listener has the ability to concentrate on speech from a particular location while suppressing speech from other locations (binaural “cocktail party” effect). In some communication situations where sounds are picked up by a microphone system for transmission on a single path to a remote listener, it would be desirable to preprocess the signals to achieve a similar effect. We describe a class of nonlinear processes for the outputs of an array of microphones which emphasize speech coming from a particular (on‐center) location in a background of other sounds. These processes completely eliminate any off‐center impulsive noise which is nonoverlapping at the four microphones. Results of processing outputs of real and computer‐simulated microphone arrays for speech and noise signals are described. Under anechoic conditions, the processing results in reproduction of the on‐center speech without change, and in distortion and attenuation of an off‐center speech source. The distortion produced by the processing appears to be an important factor in subjective suppression of the off‐center source.

ISSN:0001-4966    

DOI:10.1121/1.1912680

出版商:Acoustical Society of America    

年代:1971

数据来源: AIP

摘要:

A wide range of speech investigation requires natural‐sounding synthetic speech whose individual acoustic features—such as pitch, formants, and duration—can be varied independently. This study describes an algorithm for obtaining such synthetic speech by automatically analyzing natural speech to obtain data for the control of a formant synthesizer. The lowest three formants of natural speech were determined by finding simultaneous solutions of the least‐square‐fit equations by means of a Newton‐Raphson technique. The data obtained from this formant analysis, together with pitch data which were also obtained automatically, were used for controlling a computer‐simulated formant synthesizer. This algorithm was tested for various sentences and speakers, and the synthesized speech was found to be of good quality.

ISSN:0001-4966    

DOI:10.1121/1.1912681

出版商:Acoustical Society of America    

年代:1971

数据来源: AIP