Abstract: A time domain impulse response filter may be used to equalize signals in the time domain to avoid error and artifacts that are introduced by domain transforms such as the IFFT. The disclosed time domain impulse response filter is based on the magnitude responses of the individual signals. The magnitude responses for each signal may be calculated in the frequency domain or with other techniques such as auto-regressive analysis and mathematical signal approximations algorithms, such as Padé approximations. An adaptive filter may then equalize the input sensor signals in their original time domain form using a filter calculated based on the processed signals.

Abstract: An embodiment of a data-read path includes a defect detector and a data-recovery circuit. The defect detector is operable to identify a defective region of a data-storage medium, and the data-recovery circuit is operable to recover data from the data-storage medium in response to the defect detector. For example, such an embodiment may allow identifying a defective region of a data-storage disk caused, e.g., by a scratch or contamination, and may allow recovering data that was written to the defective region.

Abstract: The present document relates to multichannel audio coding and more precisely to techniques for discrete multichannel audio encoding and decoding. In particular, the present document relates to systems and method for coding soundfields. An audio encoder (200) configured to encode a frame of a soundfield signal (110) comprising a plurality of audio signals is described. The audio encoder (200) comprises a transform determination unit (203, 204) configured to determine an energy-compacting orthogonal transform (V) based on the frame of the soundfield signal (110). Furthermore, the encoder (200) comprises a transform unit (202) configured to apply the energy-compacting orthogonal transform (V) to the frame of the soundfield signal (110), and configured to provide a frame of a rotated soundfield signal (112) comprising a plurality of rotated audio signals (E1, E2, E3).

Abstract: The present invention relates to a sound effect control method and apparatus, where the method includes: when a hands-free call is performed for a mobile terminal, detecting whether a hands-free call channel of the mobile terminal is shielded; and adjusting a configuration of the hands-free call channel of the mobile terminal and/or outputting an alarm signal to inform a user that the hands-free call channel is shielded, when it is detected that the hands-free call channel of the mobile terminal is shielded. According to the method, when a hands-free call is performed for a mobile terminal, whether a hands-free call channel of the mobile terminal is shielded is detected; when it is shielded, a configuration of the hands-free call channel of the mobile terminal is adjusted and/or an alarm signal is output, which can effectively improve quality of the hands-free call of the mobile terminal.

Abstract: A first electronic device uses a first wireless link to connect wirelessly to an audio source device, and a second wireless link to connect to a second electronic device. In some cases, the first electronic device and the second electronic device can be worn by, or mounted to, a user. Upon receiving audio from the audio source device, the first electronic device extracts stereo channels from the received audio, and transmits at least one of the channels to the second electronic device. In some embodiments, an antenna of the first electronic device and an antenna of the second electronic device create a coaxial antenna pair. Alternately or additionally, each antenna, when worn by or mounted to a user, propagates an electric field in a direction generally normal to the user's head. In turn, this generates a creeping wave that travels along a surface, such as the surface of the user's head.

Abstract: A loudspeaker controller (1) for controlling a loudspeaker (2), configured to determine time-varying impedance information of the loudspeaker (2) based on a loudspeaker voltage and a measure of a loudspeaker current and provide for control of the loudspeaker (2) in accordance with said time-varying impedance information.

Abstract: An electronic device comprising: a microphone array including at least three microphones; and at least one processor configured to: identify a kind of an application that is executed; activate one or more of the microphones in the array based on each microphone's respective position within the electronic device and the type of the application; and capture audio using the activated microphones.

Abstract: A speaker system with acoustic event detection, the speaker system including a number of speakers, a head unit including an output circuit structured to output an output signal to the speakers to cause the speakers to output an audible sound, and an acoustic event detection module structured to receive signals from the speakers and to detect an acoustic event based on the signals received from the speakers.

Abstract: Embodiments are provided for customizing an audio system of a vehicle. An example computing device for customization of an audio system of a vehicle includes a display configured to present a user interface, a processor, and a storage device storing instructions executable by the processor to generate the user interface for presentation via the display, and receive user input to the user interface requesting one or more adjustments to an audio system, the one or more adjustments including instructions to tune an amplifier of the vehicle by adjusting digital signal processing (DSP) settings of the amplifier. The instructions are further executable to transmit the one or more requested adjustments to a communication interface of the amplifier of the vehicle, the adjusted DSP settings being stored in the amplifier and retrievable by the computing device.

Abstract: A mobile terminal apparatus portable by a user includes a recording unit, a memory, an estimating unit, and an output unit. The recording unit records a sound that has been produced by another apparatus. The memory stores a cause of an unusual noise that occurs in the other apparatus and intensities of individual frequency components of the unusual noise in association with each other. The estimating unit estimates a cause of the sound that has been produced by the other apparatus, on the basis of intensities of individual frequency components of the sound recorded by the recording unit and content stored in the memory, the intensities being calculated from the sound. The output unit outputs information about the cause of the sound estimated by the estimating unit.

Abstract: Method and system for use with audible signals that analyzes the signals into time-frequency frames over first and second time periods. Estimates of the noise and/or reverberation are derived from the frames. Gains are derived from the estimates and raised to a power to create modified gains. The modified gains are applied to the frames in the appropriate time periods. Modified audible signals are output after being processed by the modified gains.

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: A technique for determining one or more equalization parameters includes acquiring, via one or more electrodes, auditory brainstem response (ABR) data associated with a first audio sample and determining, via a processor, one or more equalization parameters based on the ABR data. The technique further includes reproducing a second audio sample based on the one or more equalization parameters, acquiring, via the one or more electrodes, complex auditory brainstem response (cABR) data associated with the second audio sample, and comparing, via the processor, the cABR data to at least one representation of the second audio sample to determine at least one measure of similarity. The technique further includes modifying the one or more equalization parameters based on the at least one measure of similarity.

Abstract: The present invention relates to a desktop speakerphone array processor for microphones which produces high quality sound. The microphone array has a front direction defined by the line of sight from a microphone inlet of the second microphone towards a microphone inlet of the first microphone, the array processor being connected to receive a front microphone signal from the first microphone, a rear microphone signal from the second microphone and an audio output signal representing a speaker sound emitted from a sound driver arranged near the first and second microphones and in the rearwards hemisphere with respect to the front direction of the microphone array, the array processor being configured to provide a first array signal having a first directivity pattern with a main lobe oriented in the front direction of the microphone array in dependence on the front microphone signal, the rear microphone signal and the audio output signal.

Abstract: A microphone includes a microphone base that has a first surface and a second surface. The microphone also includes a microelectromechanical system (MEMS) device coupled to the first surface of the microphone base. The microphone also includes a cover coupled to the first surface of the microphone base, such that the cover divides the first surface into a covered portion where the cover encloses the MEMS device, and a non-covered portion extending away from the cover. The microphone also includes one or more pads on the uncovered portion of the first surface of the base. The microphone also includes a port extending through the base from the first surface to the second surface.

Abstract: A system and method for speech reinforcement may determine the spatial location of an audio source and the spatial location of a listener. An audio signal generated by the audio source may be captured. The spatial location, relative to the listener, of two or more audio transducers that emit a reinforcing audio signal to reinforce the audio signal may be determined. The captured audio signal, responsive to the spatial location of the audio source, the spatial location of the listener and the spatial location of the two or more audio transducers to generate the reinforcing audio signal, such that, when emitted by the two of more audio transducers, the listener perceives a source of the reinforcing audio signal to be spatially located in substantially the spatial location of the audio source thereby reinforcing the audio signal.

Abstract: A system for removing noise from an audio signal is described. For example, noise caused by content playing in the background during a voice command or phone call may be removed from the audio signal representing the voice command or phone call. By removing noise, the signal to noise ratio of the audio signal may be improved.

Abstract: A mobile communication device having an acoustic divider for minimizing acoustic coupling is disclosed. The mobile communication device includes a housing having an outer surface and internal sidewalls. The outer surface and internal sidewalls define a void disposed at and below the outer surface of the housing. The mobile communication device includes a receiver disposed within the housing and below a first portion of the void, and a microphone disposed within the housing and below a second portion of the void. An acoustic divider is disposed within the void and laterally disposed between the receiver and the microphone. The acoustic divider acoustically isolates the first and second portions of the void, thereby minimizing acoustic coupling between the receiver and the microphone.

Abstract: A duty-cycled acoustic sensor saves power, for example, by operating for relatively short periods of time in a repetitive manner. A sensor bias current provides operating power to the sensor. An output analog signal from the sensor carries the information induced by the sensor upon the bias signal. Capacitive coupling is employed to remove direct DC voltage from the output analog signal to generate an analog input signal for acoustic analysis. A capacitor for capacitive coupling is pre-charged to reduce the charging time of the capacitor as the sensor is being powered up. After the capacitor is sufficiently precharged, acoustic analysis is performed on the analog input signal. The sensor is powered down by substantially blocking current flow through the sensor, which saves power. Results of the acoustic analysis can be used, for example, to control parameters of the duty-cycling of the acoustic sensor.

Abstract: Embodiments of the present invention relate to adaptive audio content generation. Specifically, a method for generating adaptive audio content is provided. The method comprises extracting at least one audio object from channel-based source audio content, and generating the adaptive audio content at least partially based on the at least one audio object. Corresponding system and computer program product are also disclosed.