Abstract: An integrated circuit for providing programmable microphone interface includes an input terminal for receiving an input signal and an output terminal for providing an output audio signal. The integrated circuit includes a bias circuit, an amplifier circuit, and two feedback circuits. The bias circuit provides a microphone bias signal to the microphone and provides a sensed microphone signal. The amplifier circuit includes a first input, a second input, and an output. The first input is configured to receive the sensed microphone signal, a first feedback signal, and a second feedback signal. The second input is configured to receive a first reference signal. The feedback circuits are in communication with the output and the first input of the amplifier circuit. In a specific embodiment, the first feedback circuit includes an RC circuit and the second feedback circuit includes an integrator.

Abstract: Various embodiments of the present invention are directed to methods and systems that reduce acoustic echoes in audio signals in accordance with changing conditions at first and second locations that are linked together in a communication system. In particular, one embodiment of the present invention relates to a method for determining an approximate impulse-response vector for canceling an acoustic echo resulting from an audio signal transmitted from the first location to the second location. This method includes forming a trust region within a search space based on computing a recursive specification vector defining the trust region. The method also includes computing a recursive shadow-impulse-response vector that lies substantially within the trust region, and computing the approximate impulse-response vector based on the recursive shadow-impulse-response vector and the recursive specification vector.

Abstract: A howling suppression apparatus suppresses a howling caused in an acoustic system including a sound collection device and a sound emission device. An estimation part generates an estimated signal by estimating a feedback sound reaching the sound collection device from the sound emission device. An adjustment part generates an estimated signal by adjusting the estimated signal. A spectrum subtraction part generates an acoustic signal using a result of subtracting a frequency spectrum of the estimated signal from a frequency spectrum of an acoustic signal. A filter part generates an acoustic signal by suppressing a component of a frequency band including a howling frequency F among the acoustic signal. An acoustic signal in which the acoustic signal is amplified by an amplifier is supplied to the sound emission device.

Abstract: An integrated circuit configured to provide a microphone output signal, comprising: a preamplifier coupled to receive an input signal, generated by either a first microphone member or a second microphone member, where one of the members is movable relative to the other microphone member; a voltage pump to output a pumped voltage; and a low-pass filter coupled to filter the pumped voltage from the voltage pump and to provide a bias voltage to either microphone member.

Abstract: The invention relates to a hearing instrument for processing an input sound to an output sound according to a user's needs. The invention further relates to a method of operating a hearing instrument and to use of a hearing instrument. The object of the present invention is to provide an alternative scheme for handling acoustic feedback in a hearing instrument.

Abstract: A portable telephony device has a mobile phone housing. Integrated in the housing are memory to store a telephony module, an earphone type detect module, a headphone port, and a processor. The headphone port may be a wired headset jack to receive a mating wired headset plug, or it may be a communications subsystem that makes a wireless connection with a wireless headset. The earphone type detect module, when executed by the processor, is to determine what type of earphone or headset is connected to the headphone port. The telephony module when executed by the processor is to adjust a sidetone function of the device as a function of the determined type of earphone, and apply sidetone in accordance with said adjusted sidetone function. Other embodiments are also described and claimed.

Abstract: A hands-free loudspeaker system which is capable of achieving high-quality voice amplification without requiring a human speaker to move to a microphone or a microphone to be moved to a human speaker. A microphone whose input level has continued to be above a threshold value for not shorter than a predetermined time period is detected, based on input signals from dispersedly arranged microphones. An input signal from the microphone is selected and outputted to a loudspeaker at an output level or with a delay time, according to a location of the loudspeaker. A preset lowest threshold level is initially set to the threshold value, and an input level of the microphone higher than the threshold value is newly set to the same, while when the input level is lower than the threshold value, a lower value is set to the same in a step-by-step manner.

Abstract: Provided is a digital microphone having a power supply. In this case the power supply is configured so as to provide a P48 V phantom power.

Abstract: An integrated circuit for providing programmable microphone interface includes an input terminal for receiving an input signal and an output terminal for providing an output audio signal. In an embodiment, the integrated circuit includes a bias circuit, an amplifier circuit and two feedback circuits. The amplifier circuit includes a first input, a second input, and an output. The first input receives either the input signal or a feedback signal, depending upon mode control signals. The second input receives either the feedback signal or the input signal depending upon the mode control signals. The first feedback circuit is in communication with the output and the first input of the amplifier and includes a first resistor and a first capacitor connected in parallel. The second feedback circuit includes an integrator circuit and provides the feedback signal. The mode control signals can be set in a programmable mode control register.

Abstract: A system and method for actively changing the sound perceived by listeners in an audio environment. A single transducer is used as both a sensing microphone and as an output driver. In one embodiment, the invention is implemented as an active noise cancellation system. The sensed noise signals are phase shifted to provide a cancellation effect, combined with the desired audio program signals, and output to the transducer, thereby reducing the level of unwanted noise heard by they listener. In other embodiments, the system can be used to sense the frequency response of a listening room and make appropriate equalization adjustments to the output.

Abstract: In a voice intelligibility enhancement system that controls a gain of a voice signal based on noise power and voice power of the voice signal generated by a voice signal generation unit, it is detected whether the voice power is equal to or greater than a predetermined level, noise power output when the voice power is less than the predetermined level is measured and stored, noise power to be output when the voice power exceeds the predetermined level is estimated to be the stored noise power, and gain of a voice signal is controlled on the basis of the voice power and the estimated noise power.

Abstract: An autobias vehicular microphone system (300) includes a microphone (301) uses an amplifier (306) for amplifying an output of the microphone. A first feedback path (308) provides an amplifier output signal to the amplifier input for providing amplifier linearity and a second feedback path (305) is used for providing bias to an voltage reference (303). The voltage reference (303) operates to provide an autobias to the amplifier (306) based upon amplifier loading. Thus, a DC feedback loop works as an average voltage sensing circuit operating to center the amplifier (306) to an operating point near one half its supply voltage. By allowing the bias point to vary, a constant clip level can be maintained depending on varying load conditions of electronic devices (307, 309, 311) using the microphone (301).

Abstract: Various embodiments of the present invention are directed to adaptive real-time, acoustic echo cancellation methods and systems. One method embodiment of the present invention is directed to reducing acoustic echoes in microphone-digital signals transmitted from a first location to a second location. The first location includes a plurality of loudspeakers and microphones, each microphone produces one of the microphone-digital signals including sounds produced at the first location and acoustic echoes produced by the loudspeakers. The method includes determining approximate impulse responses, each of which corresponds to an echo path between the microphones and the loudspeakers. The method includes determining a plurality of approximate acoustic echoes, each approximate acoustic echo corresponds to convolving a digital signal played by one of the loudspeakers with a number of the approximate impulse responses.

Abstract: An audio device includes a first audio path with a loudspeaker for reproducing an audio signal, and a second audio path. The second audio path includes in series a band-pass filter for filtering an audio signal, a detector for detecting the amplitude of the band-pass filtered audio signal, a multiplier for multiplying a periodic signal by the amplitude of the band-pass filtered audio signal, and a vibration device for reproducing the multiplied periodic signal. The frequency of the periodic signal is substantially equal to the resonance frequency of the vibration device.

Abstract: Disclosed herein is a headphone device, including: a sound pickup section configured to pick up an external sound; a directivity setting section configured to generate a directional pickup audio signal, which is an audio signal obtained by picking up the external sound with a desired directional characteristic, based on an audio signal outputted from the sound pickup section; a loudspeaker; an audio signal generation section configured to generate a cancellation-use audio signal for attenuating the directional pickup audio signal based on the directional pickup audio signal; and a driving signal generation section configured to generate a driving signal, which is an audio signal for driving the loudspeaker and includes at least the cancellation-use audio signal.

Abstract: A feedback reduction system adaptively processes a microphone signal. An adaptive feedback suppression filter circuit processes the microphone signal, and an adaptive feedback compensation filter circuit further processes the microphone signal. The adaptive feedback suppression filter circuit processes the signal based upon output from the adaptive filter compensation filter circuit.

Abstract: A dereverberation and feedback compensation system reduces the echo received by a first audio device while reducing the speech feedback received from a second audio device. A decorrelation logic decorrelates audio signals from the first audio device. A first processor generates a noise compensation signal based on the decorrelated audio signals and system determined filter coefficients. The second processor generates an enhanced noise correlation signal based on speech signals of a second audio device and the filter coefficients used by the first processor.

Abstract: A karaoke device is included within an enclosure and having a voice pickup element integrated into the enclosure, the voice pickup element for converting sound waves into an electrical signal. An audio input signal passes into the enclosure and connects with an electronic circuit for amplifying the electrical signal, for controlling the amplitude of the electrical signal and for mixing the electrical signal and the audio input signal into a mixed audio signal which passes out of the enclosure. A video input signal passes into the enclosure and directly connects to a video output signal that passes out of the enclosure.

Abstract: A wireless audio amplification system for classrooms and the like includes a system that allows for advanced listening and learning audio tools. The invention combines radio frequency and infrared technologies into one integrated system. A microphone includes a radio frequency receiver and an infrared transceiver. The microphone transmits voice signals to an audio amplifier unit that includes a radio frequency transceiver and an infrared transceiver.

Abstract: A speaker array and microphone arrays positioned on both sides of the speaker array are provided. A plurality of focal points each serving as a position of a talker are set in front of the microphone arrays respectively symmetrically with respect to a centerline of the speaker array, and a bundle of sound collecting beams is output toward the focal points. Difference values between sound collecting beams directed toward the focal points that are symmetrical with respect to the centerline are calculated to cancel sound components that detour from the speaker array to microphones. Then, it is estimated based on totals of squares of peak values of the difference values for a particular time period that the position of the talker is close to which one of the focal points, and the position of the talker is decided by comparing the totals of the squares of the peak values of the sound collecting beams directed to the focal points that are symmetrical mutually.

Abstract: An audio source tracking arrangement, integrated in or connected to a video conference system, for determining a position of a source creating a sound, including: at least an audio signal processing module configured to determine the position of the source creating the sound based on a plurality of audio signals originating from the source respectively captured by a plurality of microphones; and one or more microphone housings, respectively encapsulating at least one of the plurality of microphones, the one or more microphone housings including a cavity in which at least one of the plurality of microphones is localized, an aperture on a surface of the microphone housing, and a channel extending from the cavity to the aperture, wherein the channel and the cavity are dimensioned to form an acoustical amplifier with a frequency response having one or more high frequency peaks in a frequency band of the sound.

Abstract: A plurality of sound receiving units is installed onto an equipment body. An initial information memory stores an initial direction of the equipment body in a terminal coordinate system based on the equipment body. An orientation detection unit detects an orientation of the equipment body in a world coordinate system based on a real space. A lock information output unit outputs lock information representing to rock the orientation. An orientation information memory stores the orientation detected when the lock information is output. A direction conversion unit converts the initial direction to a target sound direction in the world coordinate system by using the orientation stored in the orientation information memory. A directivity forming unit forms a directivity of the plurality of sound receiving units toward the target sound direction.

Abstract: A system and method that reduces the perceptual effect resulting from ear occlusion include an electro-acoustic feedback network that produces phase canceling sounds in the ear, where a receiver and a microphone are located. A control mechanism controls the response of the feedback network to minimize distortion in the ear while maintaining a desired frequency response for external signals. Devices producing the external signal include hearing aids, personal sound devices, in ear monitors, communications headsets and hearing protectors. The integration of the above with these devices improves a user's perception of their own voice.

Abstract: A system for reducing an echo and/or a residual echo in a microphone output signal includes an echo compensation filter configured to receive audio input signals and generate an estimated echo signal. A speech activity detector is configured to detect speech of a local speaker, and a combining circuit is configured to generate an echo compensated signal by subtracting the estimated echo signal from the microphone output signal. A residual echo reduction circuit is configured to suppress the residual echo in the echo compensated signal based on the detected speech activity, and output an echo suppressed output signal. The echo reduction system may reduce local background noise and residual echoes in a microphone output signal so that local speech may be transmitted free from undesirable signals.

Abstract: An audio apparatus has a function of correcting an audio signal in response to a noise level. The audio apparatus includes a correction unit that corrects an input audio signal on the basis of a weighting factor, an output unit that produces a played-back audio sound on the basis of the corrected audio signal, a microphone for receiving an external sound that includes the played-back audio sound and noise, a noise-extracting unit that extracts a noise signal from an external sound signal, the noise-extracting unit including a speech-removing unit that removes a speech signal from the noise signal on the basis of noise spectrum data, and a weighting factor calculation unit that calculates the weighting factor on the basis of the extracted noise signal and supplies the calculated weighting factor to the correction unit.

Abstract: A system reduces echoes in an audio system by de-correlating audio signals. Adaptive filters provide compensation based on the de-correlated signals. A controller controls the de-correlation based upon the adaptation state of the adaptive filters.

Abstract: A method and apparatus for increasing phase margin in a feedback circuit of an active noise reduction headphone. The method includes providing an acoustic block comprising an acoustic driver comprising a voice coil mechanically coupled along an attachment line to an acoustic energy radiating diaphragm, the acoustic block further comprising a microphone positioned along a line parallel to an intended direction of vibration of the acoustic diaphragm and intersecting the attachment line, the acoustic block characterized by a magnitude frequency response compensating the magnitude frequency response by a compensation pattern that has a positive slope over at least one spectral range above 10 kHz.

Abstract: An exemplary method for locating sound sources is disclosed. The method includes the steps of: loading a sound source location program into a handheld device; activating the sound source location program; calculating a total voltage representing sound waves received by a microphone array via a waveform computation algorithm; calculating energy intensities of the total voltage according to the total voltage; and selecting a maximum energy intensity from the calculated energy intensities, and determining the location of the maximum energy intensity, the location of the maximum energy intensity is the location of the sound source. A related system is also disclosed.

Abstract: A method and apparatus for increasing phase margin in a feedback circuit of an active noise reduction headphone. The method includes providing an acoustic block comprising an acoustic driver comprising a voice coil mechanically coupled along an attachment line to an acoustic energy radiating diaphragm, the acoustic block further comprising a microphone positioned along a line parallel to an intended direction of vibration of the acoustic diaphragm and intersecting the attachment line, the acoustic block characterized by a magnitude frequency response compensating the magnitude frequency response by a compensation pattern that has a positive slope over at least one spectral range above 10 kHz.

Abstract: A system and method for monitoring for a specified frequency band is disclosed. The technology initially utilizes a micro-electromechanical system (“MEMS”) based acquisition device to monitor an environment. The MEMS device receives a signal from the environment, and generates an input signal comprising an electronic representation of the received environmental signal. This input signal is then conditioned for at least one frequency band. Embodiments of the invention next allow the conditioned signal to be compared to various pre-defined events in order to determine the signal's origin.

Abstract: According to one embodiment, an acoustic correction apparatus includes: a signal obtaining module configured to obtain an acoustic signal from a target space including an object and an external space; a signal output module configured to output to the target space a measurement signal; a coefficient identifying module configured to identify, on the basis of a response acoustic signal, a correction coefficient of a correction filter that reduces a resonance frequency component of a resonance in the object; a filtering module configured to use the correction filter, and filter the signal provided to the object; a noise cancelling module configured to remove, on the basis of the acoustic signal, a noise component comprised in the acoustic signal from the filtered signal; and an output module configured to output the acoustic signal, from which the noise component is removed by the noise cancelling module, to the object.

Abstract: Various embodiments of the present invention are directed to methods and systems that reduce acoustic echoes in audio signals in accordance with changing conditions at first and second locations that are linked together in a communication system. In one embodiment of the present invention, a first digital signal encoding sounds produced at the first location is output from the first location, and a second digital signal encoding an acoustic echo and sounds produced at the second location is output from the second location. The method computes a control state that depends on the signals transmitted between the first and the second locations and computes an approximate acoustic echo based on the control state.

Abstract: A sound reinforcement system which enables handsfree and high-quality sound reinforcement without requiring a person who is speaking to move to a microphone or move a microphone. At least one microphone and a plurality of speakers are arranged in a room. A speaker output adjusting section outputs sound picked up by the microphone to the plurality of speakers at predetermined levels.

Abstract: The invention concerns an audio system comprising a microphone, audio signal processing means, an output transducer and mans for detecting a possible feedback tone and the corresponding frequency of the feedback tone in the audio system between the output transducer and the microphone. According to the invention means for counteracting feedback are provided. Further, mans are provided for changing the phase of the audio signal at a given frequency.

Abstract: An audio-coordinated visual indicator device reducing vehicle operator distraction associated with using a communications device during vehicle operation. The visual indicator being placed within the operator's field of view and provides a synchronized visual display associated with the output from the communication device. Providing an apparent visual anchor for the incoming sound freeing up the operator's mental resources, reduces distraction and allows greater attention to be placed on operation. Comprising of a receiver; a visual indicator connected to an output of the receiver having an attribute that varies according to the received signal from the communication device.

Abstract: A method of processing a sound signal in an audio amplification device using frequency transposition, the method including the steps of: (a) receiving (11) an input sound signal, (b) determining (19) gains for amplifying the input sound signal at a plurality of input frequencies, (c) transposing (20) one or more of the input frequencies of the amplified sound signal to generate one or more output signals at transposed frequencies, (d) determining (31-33) the presence of an undesired feedback signal component resulting from the amplification and frequency transposition of the input sound signal at the input frequencies, and (e) correcting (34) the output signal at each of the transposed frequencies to compensate for the presence of the undesired feedback signal component.

Abstract: An audio-coordinated visual indicator reducing vehicle operator distraction associated with using a communications device during vehicle operation. The visual indicator being placed within the operator's field of view and providing a synchronized visual display associated with the output from the communication device. Providing an apparent visual reference for the incoming sound freeing up the operator's mental resources, reducing distraction and allowing greater attention to be placed on operation. Comprising of a receiver; a visual indicator connected to an output of the receiver having an attribute that varies according to the received signal from the communication device.

Abstract: A microphone preamplifier, comprising a differential input (102) stage with a first and a second input terminal and an output stage with an output terminal; where the microphone preamplifier is integrated on a semiconductor substrate. A feedback circuit, with a low-pass frequency transfer function (103), is coupled between the output terminal and the first input terminal and integrated on the semiconductor substrate. The second input terminal provides an input for a microphone signal (105). Thereby a very compact (with respect to consumed area of the semiconductor substrate), low noise preamplifier is provided.

Abstract: Systems and methods are described that perform audio source localization in a manner that provides increased robustness and responsiveness in the presence of acoustic echo. The systems and methods calculate a difference between a signal level associated with one or more of the audio signals generated by a microphone array and an estimated level of acoustic echo associated with one or more of the audio signals. This information is then used to determine whether and/or how to perform audio source localization. For example, a controller may use the difference to determine whether or not to freeze an audio source localization module that operates on the audio signals. As another example, the audio source localization module may incorporate the difference (or the estimated level of acoustic echo used to calculate the difference) into the logic that is used to determine the location of a desired audio source.

Abstract: A system such as a speakerphone may include a processor, memory, a speaker and a microphone. The processor may be configured (via program instructions stored in the memory) to calibrate the speaker by: outputting a stimulus signal; receiving an input signal corresponding to the stimulus signal; computing a midrange sensitivity and a lowpass sensitivity for a transfer function derived from a spectrum of the input signal and a spectrum of the output signal; subtracting the midrange sensitivity from the lowpass sensitivity to obtain a speaker-related sensitivity; performing an iterative search for current parameters of a speaker model using the input signal spectrum, the stimulus signal spectrum and the speaker-related sensitivity; and updating averages of the speaker model parameters using the current parameter values. The stimulus signal may be transmitted during periods of silence in the external environment. The parameter averages may be used to perform echo cancellation.

Abstract: A broadside small array microphone beamforming apparatus comprises first and second omni-directional microphones, a microphone calibration unit, and a directional microphone forming unit. The first and second omni-directional microphones respectively convert voice from a desired near-end talker into first and second signals. The second and first omni-directional microphones and the desired near-end talker are respectively arranged at three points of a triangle. The microphone calibration unit receives the first and second signals and correspondingly outputs first and second calibration signals. The directional microphone forming unit receives the first and second calibration signals to generate a first directional microphone signal with a bidirectional polar pattern. The adaptive channel decoupling unit receives the first calibration signal and the first directional microphone signal to generate a first main channel signal and a first reference channel signal for noise detection.

Abstract: Signals dependent on the electrical output signals of two acoustical to electrical converters are computed to result in a result signal. A transfer characteristic between an acoustical signal impinging on the converters and the result signal is dependent on the arrival direction of the acoustical signals at the converters. The converters are matched for acoustical signals within a range of impinging arrival direction. The range of arrival directions is determined before matching.

Abstract: A hands-free loudspeaker system which is capable of achieving high-quality voice amplification without requiring a human speaker to move to a microphone or a microphone to be moved to a human speaker. A microphone whose input level has continued to be above a threshold value for not shorter than a predetermined time period is detected, based on input signals from dispersedly arranged microphones. An input signal from the microphone is selected and outputted to each loudspeaker at an output level or with a delay time, according to a location of the loudspeaker. A preset lowest threshold level is initially set to the threshold value, and an input level of the microphone higher than the threshold value is newly set to the same, while when the input level is lower than the threshold value, a lower value is set to the same in a step-by-step manner.

Abstract: An apparatus for suppressing feedback in an environment where a microphone and a loudspeaker are located, comprises a means for embedding a test signal into a loudspeaker signal, a microphone signal or a modified microphone signal, preferably by using a psychoacoustic masking threshold by using a pseudo-noise test signal, a means for determining a characteristic of a transmission channel in the environment between the loudspeaker and the microphone by using the embedded test signal and the microphone signal, a filter for filtering the loudspeaker signal to obtain a filtered loudspeaker signal, wherein the filter is adaptable to be adapted with regard to its filter characteristic to the characteristic of the transmission channel by the means for determining, as well as a means for subtracting the filtered loudspeaker signal from the microphone signal to obtain the modified microphone signal, in which the feedback is reduced due to the loudspeaker signal.

Abstract: An integrated circuit for providing programmable microphone interface includes an input terminal for receiving an input signal and an output terminal for providing an output audio signal. The integrated circuit includes a bias circuit, an amplifier circuit, and two feedback circuits. The bias circuit provides a microphone bias signal to the microphone and provides a sensed microphone signal. The amplifier circuit includes a first input, a second input, and an output. The first input is configured to receive the sensed microphone signal, a first feedback signal, and a second feedback signal. The second input is configured to receive a first reference signal. The feedback circuits are in communication with the output and the first input of the amplifier circuit. In a specific embodiment, the first feedback circuit includes an RC circuit and the second feedback circuit includes an integrator.

Abstract: A system for normalizing and calibrating the microphones of a sound-intensity probe or a composite of such probes, with respect to a stable comparison microphone with known acoustical characteristics. Normalizing and calibrating are performed using an apparatus 57 consisting of a tube with a loudspeaker inserted in one end and a fixture for holding the microphones of the probe together with the comparison microphone in the other end. The comparison microphone has known acoustical characteristics supplied by the manufacturer. Two banks of quarter-wave resonators 83 and 84 are attached to the side of the tube to absorb standing waves. The sound-intensity probe can be either a two-microphone probe used for measuring a single component of the sound-intensity vector or a probe with four microphones in the regular tetrahedral arrangement used for measuring the full sound-intensity vector.

Abstract: An autobias vehicular microphone system (300) includes a microphone (301) uses an amplifier (306) for amplifying an output of the microphone. A first feedback path (308) provides an amplifier output signal to the amplifier input for providing amplifier linearity and a second feedback path (305) is used for providing bias to an voltage reference (303). The voltage reference (303) operates to provide an autobias to the amplifier (306) based upon amplifier loading. Thus, a DC feedback loop works as an average voltage sensing circuit operating to center the amplifier (306) to an operating point near one half its supply voltage. By allowing the bias point to vary, a constant clip level can be maintained depending on varying load conditions of electronic devices (307, 309, 311) using the microphone (301).

Abstract: A combination of a microphone and a sigma-delta A/D converter include at least one feedback loop, such as a DC feedback loop. The DC feedback loop provides the bias current for a junction FET of the microphone. In this way, the signal current from the FET can be injected directly into the input integrator of the sigma-delta A/D converter without the need of signal resistors in series with the FET.

Abstract: This invention relates to a sound system and method for processing acoustical sound. The sound system comprises a microphone for converting an acoustical sound to an electrical sound signal, a processor for processing the sound signal and for generating a processed sound signal, and a speaker for converting the processed sound signal to a processed acoustical sound.

Abstract: A method and apparatus for automatically adjusting the volume of a headset is described herein. The headset includes a speaker and a pressure transducer. The speaker projects audible signals into the ear canal, while the pressure transducer measures a sound pressure level in the ear canal. Based on the measured sound pressure level, a control system controls the volume of the audible sound projected from the speaker.