Abstract: A howling detection and prevention circuit which receives an output of a microphone as its input signal and detects howling therein includes a computing section which divides frequency of the input signal into a plurality of frequency bands on the basis of a predetermined sampling period and computes power of each of the frequency bands, an identifying section which sequentially shifts the frequency band and identifies whether howling exists or not in accordance with a predetermined condition by employing a value of the computed power of each frequency band, and a gain adjusting section which, when howling has been detected as a result of the identifying, adjusts gain of the frequency band in which the howling has been detected to prevent the howling. A loudspeaker system employing this howling detection and prevention circuit is also provided.

Abstract: A method and apparatus for controlling acoustic gain during a non-speakerphone audio mode of computer system independent of acoustic gain during a speakerphone mode of the computer system is provided. The BIOS code of the computer system detects whether the computer system is in a speakerphone mode or a non-speakerphone audio mode. If the computer system is in a speakerphone mode, an upper gain limit is not applied to the microphone path of the computer system. If the computer system is in a non-speakerphone audio mode, a predetermined upper gain limit is applied to the microphone path of the computer system. The predetermined upper gain limit is preferably applied to the microphone path at either a microphone pre-amplifier or an audio mixer chip in the microphone path of the computer system.

Abstract: The pulse-shaping look-up table with transient suppression (530) avoids hard turn-on and turn-off transients by modifying word segments of initial and final digital words during transmission of a digital data sequence. A controller (570) sends a mode signal and a digital data sequence to the pulse-shaping look-up table with transient suppression (530). A mode buffer and command block (560) uses the mode signal to control the creation of initial and final digital words created by a data buffer and control block (550) from the digital data sequence. The digital words are used by a look-up table (540) to create a sampled digital output waveform sequence. The pulse-shaping look-up table with transient suppression (530) provides an accurate output waveform sequence with reduced spectral emissions even during start-up and shut-down of digital data transmissions.

Abstract: A problem in hearing aids or other audio/acoustic amplifier circuits is that external sources of EM energy may be coupled into the electronics of the hearing aid so as to contribute to the acoustic output. The invention provides a circuit for removing the effects of EM interference. A separate reference generator is used to detect the external EM energy. This is fed into an interference canceller which may be adaptive, which effectively removes the unwanted component in the hearing aid signal, leaving only a signal representative of the desired acoustic output.

Abstract: Disclosed howling eliminating apparatus comprises frequency analyzing means 32 for detecting a frequency which causes howling, band eliminating means 20 for eliminating a band including the howling frequency detected by the frequency analyzing means, and controlling means 30 for controlling both the frequency analyzing means and the band eliminating means in their characteristics. The frequency analyzing means is controlled to divide input frequencies into a plurality of wide bands, detect howling status in each of the divided wide bands to isolate one wide band suspected to include the howling frequency. When the one wide band causing the howling is isolated, the frequency analyzing means divides the suspected wide band into a plurality of narrow bands, and detects the howling frequency in one of the narrow bands. The band eliminating means is controlled to eliminate the narrow band including the howling frequency. The apparatus thus shortens the time it takes to detect howling with a minimum of hardware.

Abstract: The present invention is embodied in a method and apparatus for eliminating audio feedback which includes an active microphone coupled to an amplifier which transmits signals received at the active microphone to a parametric speaker for broadcasting. The apparatus comprises at least one transducer for detecting at least one sonic frequency and generating an electrical signal representative of the at least one sonic frequency, a processor for receiving the electrical signal and generating a first ultrasonic frequency which has modulated thereon the at least one sonic frequency, a parametric demodulator for recovering the at least one sonic frequency from the first ultrasonic frequency, and a speaker for directly emitting the at least one sonic frequency.

Abstract: Feedback cancellation apparatus uses a cascade of two filters along with a short bulk delay. The first filter is adapted when the hearing aid is turned on in the ear. This filter adapts quickly using a white noise probe signal, and then the filter coefficients are frozen. The first filter models parts of the hearing-aid feedback path that are essentially constant over the course of the day. The second filter adapts while the hearing aid is in use and does not use a separate probe signal. This filter provides a rapid correction to the feedback path model when the hearing aid goes unstable, and more slowly tracks perturbations in the feedback path that occur in daily use. The delay shifts the filter response to make the most effective use of the limited number of filter coefficients.

Abstract: In the invented speaker system, a passive radiator 401 is mounted on a top board 403a of baffle 403 in a direction opposite to a speaker unit 402. The speaker unit 402 is mounted on the opening of a cylinder 403b protruding from top board 403a at a place inner from the end. The remaining part of the opening is closed with a sub-baffle 406, to form a front closed cavity 404 and a back closed cavity 405. In this way, both the passive radiator 401 and the speaker unit 402 are fixed to the top board 403a whose rigidity being the highest; which reduces the unwanted vibration.

Abstract: Two or more microphones are mounted in an environment that contains an equal or lesser number of distinct sound sources. Acoustic energy from each source, with its attendant echoes and reverberation, impinges on each microphone. Using direction-of-arrival information, a first module attempts to extract the original source signals as if the acoustic environment were anechoic. Any residual crosstalk between the channels, which may be caused by echoes and reverberation, is removed by a second module. The first and second modules may be implemented using existing technology.

Abstract: Techniques for reducing unwanted acoustic feedback in a space are carried out by an adaptive notch filter algorithm that adjusts a notch to a plurality of different notch values in order to locate feedback. The results obtained by performing the algorithm at various notch values are compared. Based on the comparison, the parameters for the algorithm are adjusted for processing of the input signals to reduce the feedback.

Abstract: Apparatus and method for echo cancellation that incorporate a psychoacoustic model in determining the necessity to update compensator coefficients for the generation of artificial echoes for echo cancellation. The method includes the steps of dividing an audio signal having at least one echo into a plurality of subbands. For each of the plurality of subbands, the echo is reduced based on a variable coefficient transfer function to produce a compensated subband. The signal level of the compensated subband is then compared to a threshold value on a psychoacoustic model curve at a corresponding subband frequency. If the signal level of the compensated subband is greater than the threshold value, the coefficients of the transfer function are updated. Conversely, if the signal level of the compensated subband is less than or equal to the threshold value, the coefficients of the transfer function are not updated.

Abstract: An acoustic transmission line speaker enclosure with concentric cylindrical structures establishes acoustic coupling between a rear-traveling sound wave and a surrounding air mass. Inherent rigidity or high bending resistance of the cylindrical structure allows use of very thin walled cylinders without a massive and large overall enclosure. An audio amplifier tunable to a listening room removes very low narrow frequency band components of an audio signal. Listening room cavity resonance is measured by injecting a frequency-varying sound wave into the listening room while detecting peak sound energy within the room. The filter the eliminates from audio signal frequencies associated with listening room cavity resonance.

Abstract: A method (FIGS. 6-8) for detecting and attenuating N feedback frequencies in a digitized signal uses a tree structure containing a plurality of staged filters. In a step (602), an array of digital filters (FIG. 8) having N branches (40) is constructed. The array is arranged in a tree structure with each branch (40) having several stages (42, 44, and 46). Many of the N filters are used simultaneously in multiple different branches of the tree structure thus reducing the total number of filters required to detect all N feedback frequencies. Within each branch, N-1 of the N filters are notch filters, and each of the N- 1 notch filters attenuates the digitized signal at one of the N feedback frequencies. The remaining one filter in each of the N branches is a bandpass filter that passes the remaining of the N feedback frequency. Therefore, each branch of the tree passes a unique feedback frequency absent of all other N-1 feedback frequencies.

Abstract: Echo suppression is accomplished by comparing the ratio of the microphone (101) and speaker (121) energies to a dynamic threshold that varies with the background noise (310) at a second communication device (130), and transmitting locally generated noise when the potential for echo exists. The use of the dynamic threshold provides for echo suppression with significantly better doubletalk performance.

Abstract: A signal amplification system wherein a signal coming from a microphone is applied to a loudspeaker via a signal processing system. The signal processing system includes an estimator for estimating the transfer function from the loudspeaker to the microphone. The equalizer is set to a transfer function that is reciprocal to the transfer function from the loudspeaker to the microphone to equalize the transfer function from the microphone to the listeners. The estimator may include an adaptive filter which has a transfer function that approaches the transfer function from the loudspeaker to the microphone.

Abstract: In a signal amplifier system, a microphone (2) is connected to an echo canceller (16) via a decorrelator (6). The output signal of the echo canceller (16) is amplified by an amplifier (14) and fed to a loudspeaker (18). The echo canceller (16) is included to avoid instability caused by undesired feedback of the signal coming from the loudspeaker (18) through a feedback path (11). To improve the stabilizing effect of the echo canceller (16), the decorrelator (6) is included for decorrelating the signal coming from the microphone (2) and the signal transmitted by the loudspeaker (18).

Abstract: An echo removing apparatus for reducing the echo caused by sound generated turning from a speaker round to a microphone of a small-sized sound generated communication terminal, such as a portable telephone. A speaker output signal X(k), sent from a terminal 11 to a speaker 12, is sent to a smoothed power value calculation circuit 21 of a tap coefficient estimation circuit 21 where a smoothed input signal power value Px(k), which is a smoothed version of the square sum values, is found and sent to a tap coefficient updating circuit 22. Using the smoothed input signal power value Px(k), the tap coefficient updating circuit 22 updates tap coefficients of a filter 15. The filter 15 filters the speaker output signal X(k) to output a pseudo echo signal which is sent to a subtractor 14 where it is subtracted from a microphone input signal collected by a microphone 13.

Abstract: In an apparatus for amplifying a sound while suppressing a howling noise, a microphone collects a sound and converts the collected sound into a corresponding input signal. An amplifier amplifies the input signal to generate the sound through a loudspeaker. A plurality of equalizers are interposed between the microphone and the amplifier, each equalizer having a variable attenuation frequency for attenuatively filtering the input signal around the attenuation frequency. Detection is conducted for detecting a plurality of noise spectrum peaks of the howling noise contained in the collected sound. Then, an adjustment is conducted adjusting respective variable attenuation frequencies of the plurality of the equalizers correspondingly to the plurality of the detected noise spectrum peaks to thereby remove the howling noise from the generated sound.

Abstract: Acoustic feedback is removed from an audio signal (50) by digitizing the audio signal (50) to produce a digitized audio signal (54). The digitized audio signal (54) is then filtered with an adaptive bandpass filter (56) to detect the frequency of the acoustic feedback, where the adaptive bandpass falter (56) is aligned with the feedback based on a phase relationship between the input and the output of the adaptive bandpass filter (56). A notch filter (58) is then configured based on the frequency of the acoustic feedback, and the digitized audio signal (54) is filtered with the notch filter (58) to attenuate the feedback. The feedback-attenuated digitized signal (62) is converted to a feedback-attenuated analog signal (70).

Abstract: A howling prevention apparatus prevents a howling noise which is generated in a sound amplification system using a microphone and a loudspeaker. The howling prevention apparatus utilizes a plural number of first to n-th equalizers, which have a variable frequency response and which modulate an output of the microphone. A detector detects a frequency point at which a loop gain of the system reaches a peak while supplying a standard noise signal to the loudspeaker to produce a test sound and measuring the sound collected by the microphone through the series of the equalizers. A frequency response of the first equalizer is adjusted to suppress a noise peak at and around a frequency point detected by the detector. Then, a frequency response of the second equalizer is also adjusted to suppress a noise peak at and around another frequency point detected by the detector. Lastly, a frequency response of the n-th equalizer is adjusted by subsequently repeating the above setting procedure.

Abstract: An audio system for multimedia computer systems and method for use are provided including an audio module having speakers, a microphone and associated circuitry. The audio module is adapted to be interposed between a monitor and a monitor pedestal such that the audio module swivels and tilts in concert with the monitor. The circuitry is adapted to drive the audio module speakers and to drive only one speaker and the microphone in a speaker phone mode.

Abstract: An adaptive control system for reducing undesired signals comprises a processor (36) which provides secondary signals for sources (37) for interference with the undesired signals. Sensors (42) measure the residual Vibration which is indicative of the interference between the undesired and secondary signals. The processor (36) uses the residual signal to adjust the secondary signals to reduce the residual signals. Noise generation means (48) is provided to add a low level noise signal to the secondary signal and to provide a low level noise signal to the processor (36). The processor (36) is adapted to transform the low level noise signal and the residual signal from sensors (42) to provide the amplitude and phase of spectral components of the signals. The processor (36) modifies the secondary signals using these spectral components to obtain better reduction of the undesired signals.

Abstract: An A/D converter converts an analog signal to a digital signal. A plurality of cascade-connected notch filters include a first notch filter which is connected to the output of the A/D converter. A D/A converter is connected to the output of the last stage notch filter for converting a digital signal to an analog signal. The output of the last stage notch filter is connected to the input of a fast Fourier transform unit for analyzing the frequency. Analysis results of the fast Fourier transform unit are supplied to a detector. A coefficient having the same center frequency as that of a peak frequency outputted from the detector is selected from a coefficient memory and it is transferred to a second coefficient memory. Thus, the frequencies of the notch filters are set to eliminate howling.

Abstract: An audio interface garment includes systems for attenuating the influence of sound waves generated by audio output devices on output signals from a plurality of input devices an. In one embodiment, input signals which are applied to the audio output devices are combined by a mixer to form a mixed audio signal. A plurality of Widrow-Hoff least mean square adaptive filters each form a corresponding filtered signal based upon the mixed audio signal and the output signal from a corresponding one of the input devices. A plurality of processed signals are formed by differencing each filtered signal from the corresponding output signal. The weight values of the adaptive filters are modified according to the least mean square method. The processed signals provide signals in which the first sound waves are attenuated.

Abstract: The acoustical-mechanical disturbance feedback between the electrical-acoustical converter and the acoustical-electrical converter of a hearing aid apparatus is compensated by means of an adaptive compensator filter which feeds back a signal derived from the output of an amplification filter to its input. At the input side thereby the signal from the acoustical-to-electrical converter and the output signal of the adaptive compensator filter are substracted at a difference forming unit, the output of which being led to the input of the amplification filter. The difference is thereby formed in time domain, and time domain to frequency domain transform is performed at the output side of the difference forming unit, accordingly inverse frequency domain to time domain transform at the electric input side of the electrical-to-acoustical converter.

Abstract: Sound is converted into an electrical signal by a microphone and is converted into an inaudible, pulse width modulated signal that is combined with the electrical signal from the microphone, amplified, and converted into sound waves by a speaker. The pulse width modulator includes an A/D converter coupled to a shift register in a digital encoder. Any sound travelling from the speaker back to the microphone includes the inaudible component representing the original sound. The inaudible component is separated from the audible components, and the original sound is reconstructed in a pulse width demodulator including a shift register in a digital decoder coupled to a D/A converter. The reconstructed original sound is subtracted from the signal from the microphone, thereby reducing any echo and cancelling feedback. The apparatus includes amplitude correction circuitry for flattening the frequency response of the apparatus and includes phase correction circuitry for eliminating phase shifts in the apparatus.

Abstract: In a noise cancelling device including a first subtractor (5) for producing a first difference signal as a noise cancelled signal by subtracting a first pseudo signal from an input signal having a main signal and a first noise signal superposed on the main signal and a first adaptive filter (4) for processing a second noise signal correlated with the first noise signal into the first pseudo signal in accordance with filter coefficients thereof, a second subtractor (7) subtracts a second pseudo signal from the first pseudo signal to produce a second difference signal. A second adaptive filter (6) processes the second noise signal into the second pseudo signal in accordance with filter coefficients thereof. First and second power averaging circuits (8 and 9) produce first and second averages (P1 and P2) of power of the second difference signal and the first pseudo signal, respectively.

Abstract: A sound field control device includes an acoustic feedback signal path including a chamber and one or more loudspeaker or loudspeakers and one or more microphone or microphones provided in the chamber, the loudspeakers being disposed at a predetermined distance from the microphones, and at least either the loudspeakers or the microphones being provided in the plural, an electrical feedback signal path for feeding back a collected sound signal collected by the microphone or microphones to the loudspeaker or loudspeakers, and a collected sound signal supply control circuit provided in the electrical feedback signal path for changing, with lapse of time, a signal level of each collected sound signal fed back from the microphones to the loudspeakers. The transmission route of the collected sound signal is completely switched or the ratio of distribution of the collected sound signal is changed. Coloration in hearing is thereby reduced and a margin of howling is expanded without causing unnaturalness in hearing.

Abstract: For detecting pulsatory interference in a sampled audio signal, the value of a given signal sample is compared with the values of previous as well as subsequent signal samples within a selected time window. If the given sample, multiplied by a selected attenuation factor, exceeds a certain percentage of all such samples, that identifies it as an interference pulse. It can then be replaced by an interpolated sample value derived from adjoining samples over a selected interval of the signal.

Abstract: A method for active noise reduction based on destructive interference of sound waves in order to reduce the energy in a sound field employs two omnidirectional microphones (M1, M2) provided in connection with a loudspeaker. The acoustic feedback of the microphones is eliminated by a closed loop consisting of the microphones and the loudspeaker. The loudspeaker used is an open loudspeaker with a dipole characteristic, thus causing one of the microphones to be more sensitive to the far field and thereby to the noise which has to be suppressed. The method is implemented by a device which comprises a digital signal processor (DSP) for processing the microphone signals and which transmits an output signal to the loudspeaker where the feedback component from the loudspeaker is substantially eliminated, while the output signal's phase and amplitude are adjusted in such a manner that an effective cancellation of the noise is obtained in an area around the loudspeaker's near field.

Abstract: An automated method for modifying a speech signal in a telephone network by applying a gain factor which is a function of the level of background noise at a given destination, and transmitting the modified speech signal to the destination. The gain applied may be a function of both the background noise level and the original speech signal. Either a linear or a non-linear (e.g., compressed) amplification of the original speech signal may be performed, where a compressed amplification results in the higher level portions of the speech signal being amplified by a smaller gain factor than lower level portions. The speech signal may be separated into a plurality of subbands, each resultant subband signal being individually modified in accordance with the present invention. In this case, each subband speech signal is amplified by a gain factor based on a corresponding subband noise signal, generated by separating the background noise signal into a corresponding plurality of subbands.

Abstract: Disclosed is a mixer console which is capable of providing a mix-minus audio monitor signal to several locations from which an input audio signal is provided. The mix-minus audio signal consists of the sum of all active input channels except for the input channel to which the mix-minus audio signal is provided. Thus, the mix-minus signal permits monitoring without feedback or echo effects.

Abstract: A sound signal picked-up by a microphone is processed in a howl suppresser including a digital filter. A frequency analyzer performs frequency analysis of the picked-up sound signal. A howl detector detects a howl contained in the sound signal from a result of frequency analysis by the frequency analyzer. A coefficient calculator calculates coefficients to be input to the digital filter to suppress the howl according to a detection result by the howl detector, and a controller inputs the calculated coefficients to the digital filter. The howl detector judges that a maximum peak power level among power levels of the sound signal in a frequency region analyzed by the frequency analyzer is a howl component when a ratio of the maximum peak power level to a mean power level of the sound signal is larger than a predetermined threshold level, preferably for a predetermined threshold time.

Abstract: An electroacoustic amplifier arrangement comprises a plurality of microphone arrangements each of which includes a microphone coupled to a comparator and coupled via a coupling circuit to a variable gain amplifier. The amplifier arrangement further includes a threshold signal generator. The threshold signal generator has a first and a second input coupled to the input and output, respectively, of the amplifier and is arranged to derive a threshold signal as a function of the signals applied to its two inputs. The output of the threshold signal generator is coupled to a second input of the comparator. The threshold signal generator is arranged to derive a threshold signal as a function of the gain factor adjusted in the amplifier.

Abstract: A pair of amplifiers are coupled to respective loudspeakers. The output terminals of the amplifiers are coupled to each other through a filter network for reducing the effects of time delayed acoustic feedback from the ambient surroundings and from the other displaced loudspeaker on the amplifiers.

Abstract: A sound system with a howling-prevention function comprises an all-pass filter having group delay characteristics, which vary with the elapse of time, provided on a line used for the transmission of an audio signal from a microphone. Owing to the provision of such a sound system, any deterioration in the sound quality with respect to high frequencies and a chorus phenomenon is not produced, thereby making it possible to prevent howling without causing inferior sound quality.

Abstract: The invention relates to a device for preventing the oscillation when no signals are supplied to a VCR. A switching circuit is designed to prevent any oscillation from occurring even though a noise signal due to a signal not being supplied is generated i na feedback loop formed between the aural signal input and output terminals of the VCR and the TV.According to the invention, the transmission line of the buffer operated by the emitter follower is formed between the aural signal input and output terminals of the VCR separately from the transmission line passing through the tuner/line transfer switch and the amplifier. In the normal condition under which the vertical synchronizing signal is detected, a feedback loop is formed through the transmission line passing through the tuner/line transfer switch and the amplifier.

Abstract: Audio signals are digitized and an FFT is conducted on samples of the digitized signals to produce corresponding frequency spectrums. These spectrums are analyzed, such as by determining one or more peak frequency magnitudes which are 33 dB greater than harmonics or subharmonics of the frequency in a plurality of several successive spectrums, to detect resonating feedback frequencies. The offending frequency is then filtered in the time domain, either in the digitized form or analog form, to eliminate the feedback.

Abstract: Compensation for interaction between separate channels of a noise cancellation system in a vibrating or noisy structure is obtained by the addition of pre- and post-processing circuits for channel controllers to electronically separate the channels so that each channel operates on uncoupled modes of vibration. The processing circuits include sum and difference elements.

Abstract: A sound field is generated responsive to direct sound picked up by a direct-sound collecting microphone and indirect-sound picked up by an indirect-sound collecting microphone. Respective processing circuits are provided for signal processing the sound signals of the direct-sound collecting microphone and the indirect-sound collecting microphone. An adaptive filter is provided having a transfer function equal to a transfer function from a reproducing speaker to the indirect-sound collecting microphone. The output of the signal processing circuit associated with the indirect-sound collecting microphone is filtered by the adaptive filter, and the output of the adaptive filter is subtracted from the output of the indirect-sound collecting microphone. The output of the subtractor is then applied to the signal processing circuit associated with the indirect-sound collecting microphone for signal processing.

Abstract: An echo canceler comprises an adaptive digital filter circuit for producing an echo replica signal on the basis of a transmit digital signal and a first digital signal indicative of a far-end signal, and an adder for eliminating the echo signal from a received signal with reference to the echo replica signal, wherein the echo replica signal is modulated by a delta-sigma modulation circuit and, thereafter, converted into an analog signal for allowing the adder to operate on the echo replica signal and the received signal both in an analog form so that the adder with a narrow dynamic range can be available.

Abstract: A feed-back noise-eliminating microphone circuit is provided. The circuit includes a positive-phase pick-up circuit to pick up sound source signal for amplification through a positive-phase amplifier for further output through a balance adjusting circuit and a power amplifier. The circuit further includes a reverse-phase pick-up circuit to pick up feedback signals from the loudspeaker for amplification through a reverse-phase amplifier, wherein feedback noise from the reverse-phase amplifier is offset at said balance adjusting circuit by the output signal from the positive-phase amplifier.

Abstract: A preferred duplex communications device includes a speaker for transmitting a signal received from a remote source, first and second microphones for transducing received acoustical signals wherein the first microphone is positioned significantly closer to the speaker than the second microphone so that the first microphone receives significantly more acoustical signal strength from the speaker than the second microphone. A first amplifier is connected to the first microphone and a second amplifier is connected to the second microphone wherein the second amplifier has a gain relative to a gain of the first amplifier such that the output signals of the first and second amplifiers are generally equal with respect to signals received by way of the speaker. A filter is then connected to the outputs of the first and second amplifiers to filter the signals transduced from the speaker thereby preventing transmission to the remote source of essentially all of the signals from the speaker.

Abstract: In a hearing aid with feedback compensation by virtue of at least one secondary microphone feeding into a delay unit and an attenuator feeding into one input of a difference amplifier, the other input with opposite polarity of which is connected to the output of the primary microphone receiving the ambient sound to be amplified and fed into the hearing-aid receiver and output duct, at least two secondary signal paths are provided. Each path comprises a secondary microphone with its associated delay unit and attenuator. An operational control unit may select the strongest signal to be used for feedback compensation. This makes it possible to achieve feedback compensation in various situations, such as may arise e.g. with an "in-the-ear" hearing aid when the user is chewing or yawning, creating various possible paths from the output duct past the housing of hearing aid to the primary microphone.

Abstract: A noise suppressor is disclosed in which a noise signal is detected by a first detector, then processed by an adaptive filter, and an output signal from the adaptive filter and an audio signal from audio equipment are added by an adder and reproduced by a speaker. The sound signal thus reproduced by the speaker and the noise signal are detected by a second detector located at a listening point and sent to a filter control circuit. The filter control circuit possesses the signal sent from the second detector and the audio signal from the audio equipment having a transfer function from the adder to the filter control circuit convoluted thereby to remove the audio signal component therefrom. The adaptive filter adaptively controls the noise signal in response to the output signal from the filter control circuit.

Abstract: An amplifier using vacuum tubes is mounted in a housing with a speaker in such a way as to minimize microphonics by having the speaker axis and all vacuum tube axes parallel.

Abstract: Acoustic feedback in digital signal processing hearing aids is suppressed by using signal processing techniques in the digital processor. A first processing technique causes the data to the main signal processing path in the digital signal processor to be delayed by varying amounts over time, preferably in a periodic manner, to disrupt the buildup of feedback resonances. In a second technique, a digital filter receives the input data and has its coefficients adjusted so that the output of the filter is substantially an optimal estimate of the current input sample based on past input samples. The output of the filter is then subtracted from the input signal data to provide difference signal data which substantially cancels out the resonant frequencies. In a third technique, the acoustic feedback path from the output to the input of the hearing aid is modeled in the digital signal processor as a delay and a linear filter.

Abstract: A noise component is extracted either from FM-detected output or from the so-called signal meter in the intermediate frequency amplifier. The noise component is amplified and rectified to provide a d-c signal indicative of the noise level. The rectified noise is then shaped to produce a control signal having a predetermined level and a width. The control signal is supplied to the gate circuit to control the opening and closing of the gate through which the FM detected output passes. The gate is closed during a period when the pulse noise is superimposed on the FM detected output, and is opened during a period when the pulse noise is not superimposed to the FM detected output. In the mean time, the rectified d-c signal is smoothed out to be used as a control signal for controlling the stereo separation, the attenuation of high frequency component of the MPX-demodulated audio outputs, and the overall signal level of the audio outputs.

Abstract: In a hearing aid with a microphone, an amplifier and a receiver supplying amplified sound to the user, a second signal path is provided comprising a second, feedback-suppressing microphone, placed at a location to receive feedback-causing sound from the receiver. The output of second provided microphone is suitably attenuated and delayed by an amount .DELTA.t corresponding to the effective acoustical distance a+b between the second microphone and the main microphone and supplied to a difference amplifier in opposition to the signal from the main microphone. Thus, the component of the signal from the main microphone likely to cause positive feedback or "howling" is substantially cancelled out. In another embodiment, the feedback-suppressing signal path is an acoustical path comprising a tube leading to a rear cavity in a microphone of the directional or differential type.

Abstract: An electronic filter for an electroacoustic system. The system has a microphone for generating an electrical output from external sounds and an electrically driven transducer for emitting sound. Some of the sound emitted by the transducer returns to the microphone means to add a feedback contribution to its electical output. The electronic filter includes a first circuit for electronic processing of the electrical output of the microphone to produce a filtered signal. An adaptive filter, interconnected with the first circuit, performs electronic processing of the filtered signal to produce an adaptive output to the first circuit to substantially offset the feedback contribution in the electrical output of the microphone, and the adaptive filter includes means for adapting only in response to polarities of signals supplied to and from the first circuit.