Abstract: In method the following is performed: receiving input audio content representing mixed audio sources; separating the mixed audio sources, thereby obtaining separated audio source signals and a residual signal; and generating output audio content by mixing the separated audio source signals and the residual signal.

Abstract: Apparatus for managing and/or reducing harmonic distortion arising with an audio signal including a phase generator for generating at least one phase-difference signal or a reference audio signal generated by the phase generator, wherein the or each constant phase difference is adapted to provide cancellation of the harmonic distortion components arising along the signal path. Respective amplifier channels for receiving and separately amplifying the audio signal acting as a reference audio signal are also provided as are respective loudspeaker channels for receiving and separately producing sound corresponding to the amplified audio wherein each loudspeaker channel has substantially equal performance parameters and is adapted to radiate the sound relative to other loudspeaker channels to produce a combined sound that corresponds to the audio signal with harmonic distortion components that are reduced compared to the harmonic distortion components arising along the signal path.

Abstract: A method and circuit for testing an acoustic sensor are disclosed. In a first aspect, the method comprises using electro-mechanical features of the acoustic sensor to measure characteristic of the acoustic sensor. In a second aspect, the method comprises utilizing an actuation signal to evaluate mechanical characteristics of the acoustic sensor. In a third aspect, the method comprises using a feedthrough cancellation system to measure a capacitance of the acoustic sensor. In the fourth aspect, the circuit comprises a mechanism for driving an electrical signal into a signal path of the acoustic sensor to cancel an electrical feedthrough signal provided to the signal path, wherein any of the electrical signal and the electrical feedthrough signal are within or above an audio range.

Abstract: An ear interface mode of headphones may be determined by measuring an acoustic response of the headphones. For example, the headphones may be determined to be in a leaky or sealed configuration. An adaptive noise cancellation (ANC) system may be controlled based on the determined ear interface mode of the headphones. For example, a set of configuration parameters may be loaded for the ANC system corresponding to the known ear interface mode. An anti-noise signal may be generated according to the selected configuration parameters, and that anti-noise signal added during playback of media, such as voice recordings, music, videos, or telephone call speech.

Abstract: The technology described in this document can be embodied in a method that includes receiving an input signal representing audio captured by a microphone of an active noise reduction (ANR) headphone, and processing, by a first compensator, a first frequency range of the input signal to generate a first signal for an acoustic transducer of the ANR headphone. The method also includes processing, by a second compensator disposed in parallel to the first compensator, a second frequency range of the input signal to generate a second signal for the acoustic transducer. The first frequency range includes frequencies higher than the frequencies in the second frequency range. The method also includes detecting, by one or more processing devices, that the second signal satisfies a threshold condition, and attenuating the second signal responsive to determining that the second signal satisfies the threshold condition.

Abstract: A headset accessory comprises circuitry and is configured to mechanically attach to an audio headset. The circuitry of the headset may be operable to establish a link to the audio headset that supports conveyance of bias voltage, bias current, and/or information between the circuitry of the accessory and circuitry of the audio headset. The headset accessory may be configured to attach to a housing of the headset on a surface of the housing opposite an ear cup. A state of the circuitry of the accessory may be controlled based on the information received from the audio headset via the link. The information may include characteristics of audio being processed by the audio headset. The circuitry of the headset accessory may comprise non-volatile memory, and the non-volatile memory may store parameter settings for configuring audio processing circuitry of the audio headset.

Abstract: A signal processing device includes an acoustic echo cancellation (AEC) circuit, a blocking matrix circuit, a controller, a subtractor, and a filter. The AEC circuit performs an AEC operation based on a far-end signal and a first input signal to generate a processed signal. The blocking matrix circuit suppresses a target signal component of the first input signal and a second input signal, to generate a reference signal. The controller generates a control coefficient based on the processed signal and the second input signal. The subtractor generates an output signal based on the filtered signal and the processed signal. The filter generates the filtered signal in response to the control coefficient, the reference signal, and the output signal.

Abstract: A headphone arrangement is configured to induce natural directional pinna cues. The arrangement comprises an ear cup comprising a frame configured to at least partly encircle the ear of a user, wherein the frame is at least partially hollow. The arrangement further comprises a loudspeaker arranged within a wall of a frontal part, a rear part, an upper part, and/or a lower part of the frame, the loudspeaker comprising a membrane, a first side of the membrane facing a cavity inside the frame, and a second side of the membrane facing the outside. At least one loudspeaker is arranged at a first angle with respect to a median plane crossing a user's head midway between the user's ears such that a main direction of sound propagation is directed away from the median plane, and the second side of the membrane is directed away from the median plane.

Abstract: Audio waveform data can be received from a plurality of client devices. The audio waveform data for each client device can be generated by sampling at least one portion of an audio stream received by the respective client device. The audio waveform data received from the plurality of client devices can be compared. Based on the comparison, which of the plurality of client devices are located in a same location can be determined. Further, based on the comparison, which of the plurality of client devices located in the same location are to mute their respective input audio transducers can be determined, and an indicator indicating to the client devices to mute their respective input audio transducers can be communicated to the client devices that are determined to mute their respective input audio transducers.

Abstract: The invention relates to a method for processing customized data representative of the directivity of an customized audio system, the method comprising the following steps: —obtaining (101), for each individual of an initial set of individuals, at least one customized suite of filters; —determining (103) N independent components common to the suites of filters obtained; —decomposing (104) each of the suites obtained into a first base constructed from the N independent components with a view to obtaining, for each suite of filters, a first suite of weighting coefficients; —decomposing (105) each first suite of weighting coefficients into a second base of P independent components, so as to obtain a second suite of weighting coefficients; —storing (106) each second suite of weighting coefficients obtained in association with an identifier of an individual from among the initial set of individuals.

Abstract: Sound waves cause pressure changes in the air, and the pressure changes cause changes in the dielectric constant of air. Capacitive sensor measurements indicative of the changes in the dielectric constant of air can be processed to extract features associated with sound waves in the air. The features can include sound pressure levels represented and recordable as audio samples. Furthermore, the features can help identify types of sounds, determine direction of travel of the sound waves, and/or determine the source location of the audio. Instead of relying on movement of a mechanical member to transduce sound waves through a port into an electrical signal, an improved microphone uses capacitive sensing to directly sample and sense static pressure as well as dynamic pressure or pressure changes in the air to derive audio samples. The resulting microphone avoids disadvantages of the conventional microphone having the moving mechanical member and port.

Abstract: A sound collection equipment is disclosed. The sound collection equipment includes a speaker, a microphone and a processing unit. The speaker is used for generating a test sound having a first frequency. The microphone is used for receiving an external sound. The processing unit is electronically connected to the microphone and the processing unit. The processing unit is used for determining whether a frequency range of the external sound includes the first frequency and whether the energy of the external sound exceeds a predetermined energy value, and for judging that the sound collection equipment is in a usage state when the frequency range of the external sound includes the first frequency and the energy of the external sound does not exceed the predetermined energy value.

Abstract: Provided are a three-dimensional (3D) sound reproducing method and apparatus. The method includes transmitting sound signals through a head related transfer function (HRTF) corresponding to a first elevation, generating a plurality of sound signals by replicating the filtered sound signals, amplifying or attenuating each of the replicated sound signals based on a gain value corresponding to each of speakers, through which the replicated sound signals will be output, and outputting the amplified or attenuated sound signals through the corresponding speakers.

Abstract: An electronic device, method, and computer program product provide acoustic limiting of an audio output using a feed-forward, filter-based, acoustic control system. An audio signal is received that has more than one acoustic frequency spectra intended for conversion to an acoustic output by transducers of an electronic device. A determination is made of an expected acoustic magnitude respectively for each acoustic frequency spectrum of the more than one acoustic frequency spectra. A determination is made whether the expected acoustic magnitude of any acoustic frequency spectra will exceed an acoustic output threshold. The acoustic control system attenuates the first acoustic frequency spectrum to generate a filtered audio signal with an attenuated acoustic frequency spectrum that is less than or equal to the acoustic output threshold. The filtered audio signal is transmitted to the at least one transducer to the produce the audio output that does not exceed.

Abstract: The present invention relates to recipient customization of a bone conducting hearing device. Customization of the bone conducting hearing device may include attaching the bone conducting hearing device to a recipient, establishing communication between the hearing device and an external device, generating at least one control setting with the external device, and storing the at least one control setting in a memory device in the hearing device.

Abstract: A method in a computing device configured to perform volume-levelling processing on input audio data by at least applying one or more filters to the input audio data, the computing device being configured to obtain an estimated difference between a target output loudness level and a loudness level associated with the input audio data, and to adapt the filter coefficients of the one or more filters based on the estimated difference. The method involves starting, or stopping, the volume-levelling processing. The method comprises gradually increasing, or decreasing, a weighting applied to the estimated difference, in response to obtaining an indication to start, or stop, the volume-levelling processing.

Abstract: A method and a device are provided for optimizing a sound signal in the field of voice signal processing. The device includes at least two sound collecting units. In the method, the device locates one or more sound sources via the at least two sound collecting units. The device selects a designated sound source among the one or more sound sources. The device determines a sound signal emitted from the designated sound source among sound signals collected by the sound collecting units according to a spatial position of the designated sound source. The device optimizes the sound signal emitted from the designated sound source.

Abstract: A method for the acoustic analysis of a machine (M) including the acquisition of at least one acoustic signal supplied by at least one microphone (7) positioned in the machine, characterized in that it comprises the following steps: separation of at least one acoustic signal into a plurality of sound sources, the signal being modelled as a mixture of components, each one corresponding to a sound source; for at least one separated sound source, determination of a characteristic acoustic signature; comparison of at least one characteristic acoustic signature with at least one reference acoustic signature recorded in a reference database (5).

Abstract: In a directionality control system, a camera device captures a video of image capture area. A microphone array device collects a sound in image capture area. A signal processing section detects a sound source of the sound in image capture area, which is collected by the microphone array device. In a case where the detected sound source is within a range of privacy area, an output control section controls the sound in image capture area, which is collected by the microphone array device and is output from speaker device.

Abstract: A wireless headphone having a built-in flexible battery embedded therein. The wireless headphone having a built-in flexible battery includes a band part; a pair of headset parts including speaker units configured to receive a wirelessly transmitted audio signal and output the audio signal to the outside and connected to the band part; and a flexible battery embedded in the band part to supply power to the headset parts.