Abstract: A method performed by a processing system. The method includes generating silence signatures of audio signals from a plurality of device based on energy levels of the audio signals, providing the silence signatures to an interaction service, and out-putting interaction information corresponding to the devices.

Abstract: In one embodiment, a pickup system includes a wind detector and a wind suppressor. The wind detector has a plurality of analyzers each configured to analyze first and second input signals, and a combiner configured to combine outputs of the plurality of analyzers and issue, based on the combined outputs, a wind level indication signal indicative of wind activity. The analyzers can be selected from a group of analyzers including a spectral slope analyzer, a ratio analyzer, a coherence analyzer, a phase variance analyzer and the like. The wind suppressor has a ratio calculator configured to generate a ratio of the first and second input signals, and a mixer configured to select one of the first or second input signals and to apply to the selected input signal one of first or second panning coefficients based on the wind level indication signal and on the ratio.

Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Abstract: A system for noise removal is coupled to a signal unit that provides a digital signal. The noise removal system includes a transformation module to transform the digital signal into an f-digital signal, a threshold filter to generate a noiseless signal from the f-digital signal based on a threshold profile, and a signal synthesizer to provide a gain to the noiseless signal and to transform the noiseless signal into an output signal.

Abstract: A crosstalk cancelation technique reduces feedback in a shared acoustic space by canceling out some or all parts of sound signals that would otherwise be produced by a loudspeaker to only be captured by a microphone that, recursively, would cause these sounds signals to be reproduced again on the loudspeaker as feedback. Crosstalk cancelation can be used in a multichannel acoustic system (MAS) comprising an arrangement of microphones, loudspeakers, and a processor to together enhance conversational speech between in a shared acoustic space. To achieve crosstalk cancelation, a processor analyzes the inputs of each microphone, compares it to the output of far loudspeaker(s) relative to each such microphone, and cancels out any portion of a sound signal received by the microphone that matches signals that were just produced by the far loudspeaker(s) and sending only the remaining sound signal (if any) to such far loudspeakers.

Abstract: An audio encoder has a window function controller, a windower, a time warper with a final quality check functionality, a time/frequency converter, a TNS stage or a quantizer encoder, the window function controller, the time warper, the TNS stage or an additional noise filling analyzer are controlled by signal analysis results obtained by a time warp analyzer or a signal classifier. Furthermore, a decoder applies a noise filling operation using a manipulated noise filling estimate depending on a harmonic or speech characteristic of the audio signal.

Abstract: Disclosed herein are system, method, and computer program product embodiments for maintaining a temperature of an audio device during playback of audio content without audio interference. An embodiment operates by receiving a data stream comprising audio content for playback. The audio device determines a volume level of the audio content using the data stream. The audio device then controls a cooling device based on the volume level compared to a threshold value. This reduces the amount of time the cooling device is turned on during playback of the audio content. This reduces audio interference, improves the user playback experience, and prevents overheating of the audio device which improves reliability and reduces audio device failure rate.

Abstract: A wake vortex avoidance system includes a microphone array configured to detect low frequency sounds. A signal processor determines a geometric mean coherence based on the detected low frequency sounds. A display displays wake vortices based on the determined geometric mean coherence.

Abstract: A method of wind noise reduction. Left side and right side microphone signals are obtained. In a first stage wind noise reduction is applied to a first sub-band of one of the signals, below a spectral threshold NA. In a second stage the wind noise reduced first side signal, and the second side signal, are both split into a sub-band below a threshold NB less than NA. The sub-band of the first side signal is mixed with the sub-band of the second side signal to produce an aggregate third sub-band signal having reduced wind noise, which is recombined with the respective sub-bands above the threshold NB to produce output first and second side signals.

Abstract: A method for beam-forming for a hearing aid system is disclosed. The hearing aid system has a left and a right hearing aid device for disposal on a head of a wearer. The hearing aid devices each have microphones for converting the sound into a left and a right input signal and also a signal processing device which receives the two input signals. In the method a number of different linear combinations of the left and the right input signal are provided, are assessed according to a predetermined signal criterion and, in dependence thereof, one of the linear combinations is selected as a beam signal.

Abstract: A system for infrasonic pressure fluctuation (infrasound) control of a structure defining an interior. The system includes a reversible compressor including a housing defining an interior opening interface open to the interior of the structure and an exterior opening interface open to an exterior of the structure, and including at least one controllable element to generate positive pressure flows and negative pressure flows between the interior opening interface and the exterior opening interface. The system includes a first pressure sensor located to sense pressure indicative of the interior of the structure, a second pressure sensor located to sense pressure indicative of the exterior of the structure, and at least one controller configured to control the at least one controllable element of the compressor to cancel pressure oscillations within the interior of the structure based on the pressures detected by the first pressure sensor and the second pressure sensor.

Abstract: An improved scheme for identifying and removing musical noise in an audio processing device can be achieved in an audio processing device that includes an analysis path with a perceptive model of the human auditory system and provides an audibility measure. The device may also identify an artifact introduced into the processed signal by the processing algorithm and provide an artifact identification measure, and control a gain applied to a signal of the forward path by the processing algorithm based on the perceptive model. An advantage is to dynamically optimize noise reduction with a view to audibility of artifacts and is applicable to hearing aids, headsets, ear phones, active ear protection systems, hands-free telephone systems, mobile telephones, teleconferencing systems, public address systems, karaoke systems, and classroom amplification systems.

Abstract: Systems and methods for determining the operating condition of multiple microphones of an electronic device are disclosed. A system can include a plurality of microphones operative to receive signals, a microphone condition detector, and a plurality of microphone condition determination sources. The microphone condition detector can determine a condition for each of the plurality of microphones by using the received signals and accessing at least one microphone condition determination source.

Abstract: An apparatus is provided with a device storing machine readable code and a processor executing the machine readable code. The machine readable code includes sound setting code and audio processing code. The sound setting code detects use of a microphone and sets sound characteristics that are suitable for conversation in response to detecting the use of the microphone. The audio processing code processes sound on the basis of the sound characteristics set by the sound setting code.

Abstract: A device and method relates to reducing distortion of signal. The device includes a first filter unit for generating a correction band signal, a signal level detection unit for detecting a signal level of the correction band signal, a first lookup table unit for determining a control signal, a second lookup table unit for determining a correction amount, a correction band extraction signal generation unit for generating a correction band extraction signal, a correction signal generation unit for generating a correction signal, a first edge detection unit for generating an overtone signal from the correction band extraction signal, a filter unit for suppressing high-frequency range and low-frequency range signal levels of the overtone signal, a first amplification unit for amplifying the overtone signal, a second filter unit for generating a complement signal from the overtone signal, and an output signal generation unit for generating an output signal.

Abstract: According to an embodiment, an interface circuit includes a current replicator and a receiver. The current replicator includes a power terminal coupled to a first reference node, an output terminal configured to output a signal proportional to a signal received from a transducer, and an interface terminal coupled to the transducer. Using a single interface terminal, the current replicator may be configured to provide power to the transducer and receive output signals from the transducer. The receiver may include a first input terminal coupled to the output terminal, a second input terminal coupled to a second reference node, and a current converter circuit coupled to the first input terminal.

Abstract: Wind noise is detected in and removed from an acoustic signal. Features may be extracted from the acoustic signal. The extracted features may be processed to classify the signal as including wind noise or not. The wind noise may be removed before or during processing of the acoustic signal. The wind noise may be suppressed by estimating a wind noise model, deriving a modification, and applying the modification to the acoustic signal. In audio devices with multiple microphones, the channel exhibiting wind noise (i.e., acoustic signal frame associated with the wind noise) may be discarded for the frame in which wind noise is detected.

Abstract: There is provided a sound signal processing device, in which an observation signal analysis unit receives multi-channels of sound-signals acquired by a sound-signal input unit and estimates a sound direction and a sound segment of a target sound to be extracted and a sound source extraction unit receives the sound direction and the sound segment of the target sound and extracts a sound-signal of the target sound. By applying short-time Fourier transform to the incoming multi-channel sound-signals this device generates an observation signal in the time-frequency domain and detects the sound direction and the sound segment of the target sound. Further, based on the sound direction and the sound segment of the target sound, this device generates a reference signal corresponding to a time envelope indicating changes of the target's sound volume in the time direction, and extracts the signal of the target sound, utilizing the reference signal.

Abstract: A new apparatus, system, and technique for performing Matrix Spectral Factorization has been developed, which can factorize in real time high-dimensional matrices with high-order polynomial or non-rational entries. The method can be used for efficient implementation of engineering solutions to problems in Data Compression, Filtering, Wireless Communications, and Radar Systems, etc., which are computationally reducible to matrix spectral factorization.

Abstract: The present technique relates to a sound processing apparatus, a method, and a program capable of alleviating degradation of the quality of sound in a case where the gain of a sound signal is amplified. When equalizer processing for adjusting the gain of each frequency band of an input signal on the basis of a gain setting value is performed, an input signal is attenuated by an input attenuation amount derived from the gain setting value, and the equalizer processing is performed on the input signal attenuated. The amount of amplification of the gain of the input signal in the equalizer processing is estimated on the basis of the gain setting value and a weight coefficient of each frequency band derived from a generally-available music signal prepared in advance, and a difference of the estimation value and the input attenuation amount is calculated as a gain correction amount.

Abstract: A processor combines a first predicted signal generated based on an audio signal in an interval preceding a noise detection interval, and a second predicted signal generated based on an audio signal in an interval succeeding the noise detection interval, based on the ratio between the periodicity of the audio signal in the interval preceding the noise detection interval, and the periodicity of the audio signal in the interval succeeding the noise detection interval. The processor replaces the audio signal in the noise detection interval with the combined signal.

Abstract: An audio processing apparatus including: an obtaining unit configured to obtain audio signals; and a processor configured to process the audio signals, wherein if it is not determined that the audio signals contain an abnormal value, the processor calculates a next sample value based on a predetermined number of sample values of the audio signals obtained by the obtaining unit, and wherein if it is determined that the audio signals contain an abnormal value, the processor uses the calculated next sample value so as to calculate a further next sample value.

Abstract: FIR filters for compensating for fixed pattern jitter, and methods of constructing the same, are disclosed. In one embodiment, a FIR filter filters a signal having a desired frequency component, with the coefficients of the FIR filter selected so that the filter is the equivalent of two combined FIR filters, one having the desired frequency at the filter's peak output frequency, and a second in which the signal is delayed by a time equal to half of a period of a different frequency which is desired to be removed from the output signal. In another embodiment, a FIR filter includes a delay line with a total delay longer than the period of the jitter. A signal is passed down the delay line, the number of signal edges that have occurred as the signal passes each delay element in the counted. Drivers corresponding to the delay elements in which a number of signal edges occur at the desired frequency during the period of fixed pattern jitter activate impedance elements attached to those delay elements.

Abstract: A system for reducing airway obstructions of a patient may include a ventilator, a control unit, a gas delivery circuit with a proximal end in fluid communication with the ventilator and a distal end in fluid communication with a nasal interface, and a nasal interface. The nasal interface may include at least one jet nozzle, and at least one spontaneous respiration sensor in communication with the control unit for detecting a respiration effort pattern and a need for supporting airway patency. The system may be open to ambient. The control unit may determine more than one gas output velocities. The more than one gas output velocities may be synchronized with different parts of a spontaneous breath effort cycle, and a gas output velocity may be determined by a need for supporting airway patency.

Abstract: A method for separating signal sources by use of physically unique dictionary elements. The method is particularly advantageous for separating cerebral from artifactual sources in electroencephalographic (EEG) recording, making use of dictionary element pattern recognition methods that are tuned to the unique physical properties of each source domain.

Abstract: A method and apparatus for providing privacy for telephone conversations over communication networks is disclosed. For example, a telephony endpoint device comprises a microphone for receiving an audio signal and a gain boost circuit, coupled to the microphone, for applying a gain boost to the audio signal. The telephony endpoint device also comprises a switch, coupled to the gain boost circuit, wherein operating the switch will control whether the gain boost is to be applied to the audio signal.

Abstract: A method for operating an active noise reduction system that is designed to reduce the harmonic or sinusoidal noise emanating from a rotating device, where there is an active noise reduction system input signal that is related to the frequency of the noise to be reduced, and where the active noise reduction system comprises one or more adaptive filters that output a generally sinusoidal noise reduction signal that is used to drive one or more transducers with their outputs directed to reduce the noise. Distortions of the noise reduction signal are detected. A distortion is based at least in part on differences between the frequency of the noise reduction signal and the frequency of the harmonic noise. The noise reduction signal is altered based on the detected distortion.

Abstract: A headphone cable (150) is connected to a headphone set (190) with one, two or more loudspeakers (191, 192). The headphone cable (150) includes at least a first signal wire (151) and at least one reference line (152, 154). An electronic filter unit (110) performs a filtering, e.g. a common mode filtering, which is effective on at least one of the signal wires and reference lines (151, 153). A differential audio signal between a signal wire and a reference line (152, 154) is passed. Common mode signals except signals within a VHF radio band may be attenuated. The electronic filter unit (110) may be arranged in a connector case (129) of a headphone connector (120) provided at a connector end of the headphone cable (150) or in a pluggable filter assembly (130) provided between the mobile electronic device (200) and the headphone cable (150).

Abstract: Provided is a method and apparatus for encoding/decoding an audio signal. Sections which are not used to output noise components near important spectral components and sub-bands which are not used to output noise components, are determined to be encoded or decoded, so that the efficiency of encoding and decoding an audio signal increases, and sound quality can be improved using less bits.

Abstract: A Method for processing an audio signal to use a beamforming technique in a three-dimensional (3D) space is disclosed. The method may include generating a beamforming signal on a horizontal plane related to a sound source in a three-dimensional (3D) space and modulating the beamforming signal to head for a listener in the 3D space from the sound source.

Abstract: The present disclosure describes systems and methods to transition a hearing prosthesis between different stimulation strategies. In accordance with one embodiment, a method is provided and includes operating a hearing prosthesis in accordance with a first parameter, receiving an instruction to operate the hearing prosthesis in accordance with a second parameter, and in response to the receiving, transitioning to operate the hearing prosthesis in accordance with the second parameter by operating the hearing prosthesis in accordance with at least one intermediate parameter.

Abstract: An apparatus comprises a selected volume detector that detects a selected output volume; an analog output signal amplifier; a digital volume amplifier; a boost gain control element coupled to the selected volume detector; the analog output signal amplifier; and the digital volume amplifier; wherein the boost gain control element is configured to: keep a gain of a path of the digital volume amplifier and the analog output signal amplifier substantially constant, wherein the boost gain control element can adjust both: a) a gain of the digital volume control; and b) a gain of the analog output signal amplifier; to keep the gain of the path of the digital volume amplifier and the analog output signal substantially constant and equal to the selected output volume.

Abstract: A speech signal transmission apparatus includes an extractor to extract speech signals from speech source signals collected by a plurality of microphones, a power calculator to calculate powers of speech signals of multiple channels and set any one of the speech signals of the multiple channels as a reference speech signal, a synchronization adjustor to adjust synchronization of the other speech signals based on the reference speech signal, a signal generator to generate extraction signals by offsetting the reference speech signal from the other synchronization-adjusted speech signals, an encryptor to compress and encrypt the reference speech signal and the extraction signals, and a transmitter to transmit the compressed and encrypted reference speech signal and extraction signals.

Abstract: Audio signals are processed for use in a communication event. A data store may be queried to obtain an indication of an echo direction, which relates to a direction from which audio signals output from the audio output are likely to be received at a microphone array (plurality of microphones) of a device. Beamformer coefficients of an adaptive beamformer of the device are determined in dependence upon the received indication of the echo direction. Audio signals are received at the microphone array. The adaptive beamformer applies the determined beamformer coefficients to the received audio signals, thereby generating a beamformer output for use in the communication event. The beamformer coefficients are determined such that echo suppression is applied to audio signals received at the microphone array from the indicated echo direction.

Abstract: To sufficiently suppress an impact sound in a noisy signal. The impact sound in the noisy signal is suppressed. For this, the impact sound is detected in the noisy signal. It is characterized in that phase information of the detected impact sound is processed by using the phase information of a signal other than the impact sound in the noisy signal so that an amount of change in the phase information is reduced.

Abstract: An automated noise reduction circuit is configured to find an audio output with the lowest voltage level associated with undesirable noise. The circuit may include a plurality of audio input ports, ground isolation devices, electronic switches and a microprocessor. The microprocessor may be configured to toggle the ground isolation devices on and off, sample voltage at an output of the ground loop isolation transformers when the ground isolation devices are on and off, and switch the ground isolation devices to a position that produces a lowest voltage output based on a comparison of sampled voltage that is analyzed by a set of algorithms programmed into the microprocessor and that has been sampled from the output of the ground isolation devices for each of the multiple input ports.

Abstract: Provided is a hearing aid system including: a first processor that fast Fourier transforms N input signal tone data output from an input buffer memory, and then executes nonlinear compression; a second processor that inverse fast Fourier transforms amplitude spectrum data; an output buffer memory that stores the voice signal tone data, until the number of the voice signal tone data is N; and a digital-to-analog (D/A) converter that converts the digital voice signal tone data into an analog signal, to then output the analog signal to a receiver. Thus, certain ambient noise due to an acoustic feedback signal and a narrow frequency band that occur in a hearing aid is removed, to thus reduce discomforts due to the acoustic feedback noise of the hearing aid for hearing aid users, and to thereby significantly improve speech discrimination.

Abstract: An information processing apparatus includes: an acquisition unit adapted to acquire activities corresponding to a plurality of bases from an audio signal; a determination unit adapted to determine whether the plurality of acquired activities have a specific characteristic; and an output unit adapted to output an audio signal obtained by separating, from the audio signal, a component corresponding to an activity which is determined to have the specific characteristic by the determination unit.

Abstract: Spatially selective augmentation of a multichannel audio signal is described.

Abstract: It is determined whether or not a sound picked up by at least either a first microphone or a second microphone is a speech segment. When it is determined that the sound picked up by the first or the second microphone is the speech segment, a voice incoming direction indicating from which direction a voice sound travels is detected based on a first sound pick-up signal obtained based on a sound picked up by the first microphone and a second sound pick-up signal obtained based on a sound picked up by the second microphone. A noise reduction process is performed using the first and second sound pick-up signals based on speech segment information indicating that the sound picked up by the first or the second microphone is the speech segment and voice incoming-direction information indicating the voice incoming direction.

Abstract: An active noise control system for exhaust systems of a combustion engine operated vehicle comprises an anti-sound control connectable to an engine control of the vehicle and a loudspeaker connected to the control for receiving control signals and designed for generating an anti-sound in a sound generator, fluidically connectable to the exhaust system. In the control, at least two curves are stored in order to cancel airborne sound conducted in the exhaust system through outputting the signal to the loudspeaker. The curves cover different temperature ranges of the exhaust gas, which temperature ranges overlap one another by pairs or directly adjoin one another. The control is furthermore designed to select a curve suitable for a respective temperature of the exhaust gas conducted in the exhaust system from the available curves by means of signals output by the engine control and output signals to the loudspeaker making use of this curve.

Abstract: Provided is a method and apparatus for encoding/decoding an audio signal. Sections which are not used to output noise components near important spectral components and sub-bands which are not used to output noise components, are determined to be encoded or decoded, so that the efficiency of encoding and decoding an audio signal increases, and sound quality can be improved using less bits.

Abstract: A speech enhancement method is disclosed. The method includes the steps of: receiving a plurality of frames of sound signals by a microphone array; calculating an inter-aural time difference for each frequency band of each frame of the sound signals corresponding to at least one two-microphone set of the microphone array; calculating a plurality of values of cumulative histograms according to the calculated inter-aural time differences, wherein each value of the cumulative histograms is associates with a sound signal intensity of a respective frame; determining a first inter-aural time difference threshold according to the calculated value of the cumulative histograms; and filtering the plurality of frames of sound signals according to the first inter-aural time difference threshold.

Abstract: A device and method are disclosed for testing the intelligibility of audio announcement systems. The device may include a microphone, a translation engine, a processor, a memory associated with the processor, and a display. The microphone of the analyzer may be coupled to the translation engine, which in-turn may be coupled to the processor, which is in-turn may be coupled to the memory and the display. The translation engine can convert audio speech input from the microphone into data output. The processor can receive the data output and can apply a scoring algorithm thereto. The algorithm can compare the received data against data that is stored in the memory of the analyzer and calculates the accuracy of the received data. The algorithm may translate the calculated accuracy into a standardized STI intelligibility score that is then presented on the display of the analyzer.

Abstract: A buffer is coupled to an acoustic motor. The buffer has an input and an output. The input has an input voltage and the output has an output voltage. The buffer is coupled to a load. The buffer includes an input transistor and push-pull transistor circuitry. The input transistor has a gate, a source, and a drain, a gate-to-source capacitance, and an area. The push-pull transistor circuitry is coupled to the input transistor. Under a first set of operating conditions, the gate to source voltage of the input transistor remains constant and the output voltage is a buffered copy of the input voltage. Under a second set of operating conditions, the push-pull transistor circuitry selectively sinks or sources additional current to the load so that linearity of buffer operation is provided. A gate-to-drain capacitance of the input transistor is buffered allowing the area of the input transistor to be increased without reducing the gain of the motor.

Abstract: A method for audio coordination. The method includes connecting electronic devices to a communication session. Distinct signals are assigned to each of the electronic devices. Input streams are established from one or more of the electronic devices. Signals are detected within the input streams. One or more electronic devices are selected for eliminating input streams based on an audio fidelity threshold.

Abstract: Two ear cups are coupled by a flexible member. Each ear cup includes a speaker mounted on an internal portion thereof and is configured to form an air pressure seal around an ear of a user when the apparatus is placed on the user. The flexible member couples the two ear cups and ijs configured such that, when the apparatus is placed on the user, each ear cup completely covers an associated ear of the user and forms an air pressure seal around the associated ear. A microphone is electrically coupled to the speaker in each ear cup for transmitting ambient sound to the user when the apparatus is placed on the user. An indicator light may be provided which is configured to turn on when the microphone is active. The microphone and/or the indicator light may be mounted on the flexible member or on one of the ear cups.

Abstract: A communication system for communicating with at least one loudspeaker is described where the loudspeaker is connected to audio equipment over standard two-wire speaker wire operable to carry an audio signal. The communication system includes a master node in electrical communication with a signal path carrying the audio signal between the audio equipment and the loudspeaker, the master node also including an interface with the audio equipment, a data encoder operable to encode data signals, a data transceiver operable to place the data signals onto the audio signal at frequencies above audio frequencies. The communication system also includes at least one slave node in electrical communication with the audio signal and each loudspeaker, the slave node including a data transceiver operable to receive data signals from the master node, a data decoder, and an interface able to communicate with the loudspeaker.

Abstract: An audio system that reduces or eliminates click and pop noise during power up and power down operations. In particular, the audio system includes an amplifier with an input adapted to receive an input audio signal and an output adapted to produce an amplified output audio signal for an associated speaker. The audio system further includes a noise reduction circuit adapted to smoothly apply and remove a DC voltage to and from the output of the amplifier in a manner that reduces or eliminates click and pop noise from being generated by the associated speaker. The DC voltage at the output of the amplifier may be derived from a DC reference voltage source and/or from the input audio signal.

Abstract: A system is provided for enhancing a sound signal produced by an audio system in a listening environment by compensating for ambient noise in the listening environment. The system receives an electrical sound signal and generates a sound output therefrom. A total sound signal is sensed representative of the total sound level in the environment, where the total sound level includes the sound output and the ambient noise. The system extracts an ambient noise signal representative of the ambient noise from the total sound signal, using an adaptive filter with an adaptive step size, in response to the total sound signal and to a reference signal derived from the electrical sound signal. The system generates a control signal in response to the ambient noise signal and adjusts the sound output of the audio system to compensate for the ambient noise level in response to the control signal.