Abstract: A method of producing a plurality of broadband response files derived from empirical testing on at least one human subject, for convolving with an audio input signal represented as digital samples to produce a stereo output signal (having a left field and a right field) such that the stereo signal emulates the production of said audio signal from a specified audio source location relative to a listening source location. A reference signal is derived for each of a plurality of test positions. An originating signal is derived from at least one reference signal. Each reference signal is deconvolved with the originating signal to produce a broadband response file for each test position. A data storage facility having a plurality of broadband response files stored therein.

Abstract: The microphones used in hearing devices normally possess different characteristic lines that are to be adapted to one another. For this purpose, the amplitude of an output signal of a first microphone and the amplitude of an output signal of a second microphone are measured. The output signal of the first microphone is subsequently filtered dependent on both measured amplitudes, such that the difference between the two output signals is reduced. One of the two microphones hereby serves as a reference, and an absolute normalization can be foregone.

Abstract: A communications system and method performs centralized signal processing on received audio signals. A plurality of terminals are coupled to a processing switch via links. The terminals can be, for example, dedicated speakerphones, desktop handsets, or personal computers with audio capabilities. The links can be wired or wireless, can carry analog or digital signals, and can be shared with other users in a network. The switch receives the audio data from the terminals, processes the data according to desired acoustical procedures, creates one or more output mixes, and provides the output mixes to the appropriate terminals. The operation of the processing switch is controlled by a communications support module (CSM) which can receive, process, and send data to/from multiple terminals simultaneously. The CSM receives audio signals from the terminals.

Abstract: A method of tuning an alert device in a portable communication apparatus having a microphone. The method involves the steps of recording, through the microphone, an acoustic signal which is emitted by the alert device in response to a drive signal; deriving a characteristic value of the recorded signal; comparing the characteristic value with a reference value and generating a comparison result; and controlling the drive signal of the alert device in response to the comparison result.

Abstract: A person receiving a Voice over IP call, such as a call center agent, remotely controls media settings on a caller's multimedia terminal device functioning as a telephone terminal device. In some embodiments, the call center agent may remotely control media settings by entering a DTMF (dual tone multi-frequency) sequence from the telephone keypad. The DTMF sequence may be recognized by a gateway through which an Internet-to-PSTN call is made, stripped out of the audio stream and sent to the caller's multimedia terminal device in a separate command message through the Internet, or the DTMF sequence may be encoded into packets along with the voice information for detection and translation into a command message by software at the caller's terminal device. In pure VoIP applications, command packets may be sent directly.

Abstract: A microphone preamplifier. An electrical signal from a microphone is received via a shielded cable by an input stage cascaded with an output stage. The input stage contains an impedance converter and a phase inverter and receives feedback that reduces input capacitance, actively shields the coaxial cable, and stabilizes the input stage. The output stage contains a buffer and output driver and provides amplified versions of the input stage output. The preamplifier contains means for transformerless phantom powering the microphone and the microphone preamplifier, for sharing current between the input and the output stages, and for limiting the frequency at the output of the microphone.

Abstract: An anti-oscillation loudspeaker device where a reproduced sound is compensated based on a acoustic signal detected by a microphone and a reproduced band is enlarged is provided. In this loudspeaker device, a microphone is placed near a position where sound pressure of resonance occurring in a closed space for at least one of height, width, and depth of the inside of a cabinet is minimum. An influence of the resonance is restrained, a feedback circuit becomes stable, and a feedback amount increases.

Abstract: A device for operating voice-controlled systems, such as communication and/or intercommunication systems in motor vehicles, includes a plurality of microphones and at least one loudspeaker. Voice signals received by the microphones are transmitted to the at least one loudspeaker. The voice signals are subjected to a low-value frequency shift before being transmitted to the loudspeaker(s) or to the input of a voice-controlled device to thereby suppress feedback.

Abstract: In the method according to the invention the communication device has a microphone and a signal path leading from the microphone to a speaker, where the signal path comprises a programmable signal processing unit. According to the method the user is given control in a training session over one or more signal processing parameters within the signal processing unit. In the training session the user listens to the sound of his or her own voice transmitted through the communication device, and adjusts the one or more signal processing parameters until he or she is satisfied with the sound quality of his/her own voice. The values of the signal processing parameters chosen by the user during the training session are stored in a storing means within the device, and the programmable signal processing automatically uses the stored parameter when detection means within the unit detects the users own voice.

Abstract: A programmable speaker uses characterization data stored within the memory of the speaker and digital signal processing (DSP) to digitally perform transform functions on input audio signals to compensate for speaker related distortion and listening environment distortion. In a manufacturing environment, a non-intrusive system and method for tuning the speaker is performed by applying a reference signal and a control signal to the input of the programmable speaker. A microphone detects an audible signal corresponding to the input reference signal at the output of the speaker and feeds it back to a tester which analyzes the frequency response of the speaker by comparing the input reference signal to the audible output signal from the speaker. Depending on the results of the comparison, the tester provides to the speaker an updated digital control signal with new characterization data which is then stored in the speaker memory and used to again perform transform functions on the input reference signal.

Abstract: A microphone comprises a housing; an acoustic-to-electrical signal transducer disposed within the housing, the transducer being operable to generate an analogue audio signal; and a one-bit analogue-to-digital converter disposed within the housing for converting the analogue audio signal into a one-bit digital audio signal for transmission to other, external, audio processing apparatus.

Abstract: A system and method for actively damping boom noise within an enclosure such as an automobile cabin. The system comprises an acoustic wave sensor, a motion sensor, an acoustic wave actuator, a first electronic feedback loop, and a second electronic feedback loop. The enclosure defines a plurality of low-frequency acoustic modes that can be induced/excited by the enclosure cavity, by the structural vibration of a panel of the enclosure, by idle engine firings, and a combination thereof. The acoustic wave actuator is substantially collocated with the acoustic wave sensor within the enclosure. The motion sensor can be secured to a panel of the enclosure.

Abstract: In a digital voice enhancement communication system, a plurality of individually customized equalization filters are provided, one for each electrical path between a respective microphone and loudspeaker and custom tailored to the respective electroacoustic transfer function therebetween. Increased gain is enabled in a digital voice enhancement communication system by an equalization filter filtering a respective electrical signal from a microphone to a loudspeaker in inverse relationship to the electroacoustic transfer function between the microphone and loudspeaker.

Abstract: The present invention is a microphone that applies the principle of negative feedback directly to the diaphragm, greatly reducing the non-linearity of the diaphragm. In a further embodiment, digital negative feedback is used, incorporating the diaphragm into the digitization loop of a sigma-delta converter, creating a direct sound pressure to digital electrical output converter. In one embodiment, positive feedback is used in an analog circuit, causing a negative feedback response on the diaphragm.

Abstract: An amusement and entertainment system where participants try to continuously speak and/or sing while simultaneously listening to their own voices on a ½ second sound delay. The confusing feedback does not act as a sound enhancement echo but instead causes confusion, disorientation and amusement to the participants and others watching the participants. The system can be used by one or more participants and a third person can switch signals so that one or more participants or others watching can hear the resultant effects of another participant. The system can be table mounted so that participants can be seated about a table. Alternatively, the system can be mounted in a mobile vehicle. Video cameras and monitors can be used to further add to the amusement of participants and others watching the participants. Each headphone can include microphones built into the headbands.

Abstract: An electroacoustic assembly comprising a microphone having a diaphragm and supported on a face plate susceptible to vibratory effects. Vibration sensitivity is reduced by opposing the pressure effects on the diaphragm caused, on the one hand, by vibration of the assembly in the ambient air mass and by vibration of the air mass leading from the ambient air mass to the diaphragm, and on the other hand, by vibration of the effective mass of the diaphragm, generally augmented with additional mass, and including the effect of the internal air mass adjacent the diaphragm. Applications include hearing aids in which the microphone and its support are mechanically coupled to receiver components that may impart significant motion thereto.

Abstract: An audio delay and feedback apparatus includes i) a microphone for receiving an input audio signal and for generating a first electrical signal corresponding to the input audio signal, ii) a delay circuit for receiving the first electrical signal and for generating a second electrical signal corresponding to but delayed from the first electrical signal by a predetermined amount of time, and iii) earphones or a speaker for receiving the second audio signal and for generating an audio output signal corresponding thereto. An audio signal input to the microphone is delayed and fed back to the earphones so as to make speech difficult, humorous, discombobulated, and often unintelligible. The microphone, earphones and delay circuit are mounted on a headset. In one embodiment, the first electrical signal is digital and the delay circuit includes a digital memory. An A/D converter digitizes the first electrical signal and a D/A converter converts the second electrical signal to an analog signal.

Abstract: The long term average broad band gains of a plurality of individual signal channels, associated with a corresponding plurality of microphone elements, are electronically and dynamically adapted to one another. This is realized by periodically and dynamically processing the signals from the individual microphone elements. More specifically, the processing is such that the long term average broad band gain of the signal channels of the individual microphone elements is dynamically adjusted, an energy estimate of each microphone signal channel is averaged over the long term and the difference in energy between the signal channels is used to readjust the long term average broad band gain of the microphone signal channels to minimize those differences.

Abstract: A conference system comprises speaker units (12) coupled to a central unit (14). The speaker unit (12) comprises an echo canceller (20) with an adaptive filter (38) having an impulse response of comparatively short duration. In a speech pause of the user of the speaker unit (12) the common listening signal (LS) of the conference system is applied to the loudspeaker (18). Of the sounds then picked up by the microphone (16) only those sounds are compensated which are produced by the loudspeaker (18) and which reach the microphone (16) directly and which reach the microphone (16) indirectly via reflections from nearby objects. These sounds have a comparatively short impulse response time and enable a comparatively simple adaptive filter (38) to be used. The other sounds picked up by the microphone (16) reach the microphone (16) via reflections from the walls of the conference space (10) in which the conference system is situated. The other sounds have a comparatively long impulse response time.

Abstract: A headphone apparatus includes a speaker unit accommodated in an accommodating member of an open-air type, a microphone disposed at a front side of the speaker unit, a circuit arrangement mounted on a circuit board in the speaker unit including a filter for lowering a level of a high frequency component of a signal picked up by the microphone, and a negative feedback loop for feeding back an output of the filter to the speaker unit to thereby lower the level of a low frequency component of circumference noise output from the speaker unit. This headphone apparatus further includes an equalizer having a frequency characteristic substantially opposite to that of the filter, wherein an input audio signal is added to the negative feedback loop through the equalizer and then output from the speaker unit.

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: In order to prevent howling in a hearing aid (10) having a microphone (1), an amplifier (3) and a receiver (4) the hearing aid is provided with a second microphone (2) which is only sensitive to sound near to it. The microphones (1, 2) are connected to the amplifier (3) in anti-phase relative to each other.

Abstract: Headphones have a small cavity between the diaphragm and the ear canal with a microphone in the cavity closely adjacent to the diaphragm slightly off the axis of the ear canal and headphone diaphragm with the microphone membrane perpendicular to the headphone diaphragm. The microphone provides a feedback signal that is combined with the input electrical signal to be reproduced by the headphones to provide a combined signal that is power amplified for driving the diaphragm. The headphone transducer has a small 23 mm diameter diaphragm with a maximum excursion of 0.6 mm peak-to-peak and a low frequency resonance of 200 Hz. A disk of intracavity damping material inside the cavity isolates the microphone from the ear canal.

Abstract: An electronic stethoscope for reproducing at a user's ear as exactly as possible the sound pressure signals originating from a conventional diaphragm or bell type of chestpiece. The subject electronic stethoscope utilizes a signal processing approach which relies upon acousto-electronic feedback to provide an error or adjustment signal to amplifier gain control circuits. A pickup microphone detects audible sounds from a patient and produces an output signal representative thereof, and a variable gain amplifier amplifies the output signal of the pickup microphone. The amplifier drives an acoustic transducer in a headset which converts the amplified signal to audible sounds for audible detection by the user. A feedback microphone is placed near the acoustic transducer to detect the audible sounds produced thereby, and provides a feedback output signal.

Abstract: A multipurpose electronic assembly for a headphone assembly used in connection with a stereophonic system, such system comprising detection and signal means to ascertain noises extraneous to the stereophonic output system. Such assembly comprises a pick-up microphone in one or more speaker assembly for such ambient sound pick-up. The pick-up microphone is connected to an automatically actuated switch which closes whenever the pick-up microphone detects certain ambient sounds extraneous to the system. The latter automated switch assembly is directly interconnected with a voice compressor assembly, and then directly to an amplifier to provide minimal amplification of such extraneous sounds.

Abstract: A microphone unit particularly adapted to be used as a built-in microphone for a device such as a tape recorder in which vibration-induced components in the output of the microphone are substantially eliminated. An electret-type microphone element is mounted in a cylindrical-shaped microphone case having a central wall. A ceramic piezoelectric element is secured against the wall with a weight pressing against the opposite side of the piezoelectric element. The outputs of the microphone element and the piezoelectric vibration pickup unit are combined in a circuit in which the characteristics of the output of the piezoelectric element are corrected so that substantially all of the vibration-induced component from the microphone element is cancelled.

Abstract: A method of reducing the adaption time needed to adapt a synthetically generated secondary waveform so that it nulls a primary repetitive waveform (e.g. noise) from a source (1) of vibration, the secondary waveform generation (5) being synchronized by the source (1). The clarity and/or the amplitude of at least a part of the signal resulting from interaction between the primary and secondary waveforms is sensed (6) and used (7, 8) to reshape the secondary waveform. The invention finds application in the cancelling of repetitive vibrations (e.g. in quietening the driving cab of a vehicle).