Abstract: Mechanisms for controlling an audio level of an HDMI audio system are provided, the mechanisms comprising: causing audio data to be presented by an HDMI audio system at a current system volume level; receiving a requested volume level from a second screen device; and controlling a system volume level by: (a) determining the current system volume level; (b) determining a change in volume based on a difference between the requested volume level and the current system volume level; (c) determining a direction in which to cause the system volume level to change; (d) sending a volume control message to the system using a consumer electronic control bus connected to the system indicating whether to increase or decrease the system volume level based on the determined direction of system volume change; and (e) repeating (a)-(d) until the current system volume level reaches a predetermined value.

Abstract: An omnidirectional acoustic sensor is provided. The omnidirectional acoustic sensor includes first, inner resonators connected to an inner circumference of a supporting frame and second, outer resonators connected to an outer circumference of the supporting frame. The first, inner resonators vibrate in a height direction of the supporting frame and the second, outer resonators vibrate in a lateral direction of the supporting frame.

Abstract: A smart-home device may include a recording device configured to record sound during a first time interval and a memory device comprising a plurality of stored sound profiles. The smart-home device may also include a processing system configured to receive an environmental input, select a stored sound profile from the plurality of stored sound profiles based on the environmental input, and perform a noise-cancelation routine on the sound recorded during the first time interval. The stored sound profile may be used as an initial background noise profile for the noise-cancelation routine.

Abstract: Some embodiments of the invention are directed to an audio production system which is more portable, less expensive, faster to set up, and simpler and easier to use than conventional audio production tools. An audio production system implemented in accordance with some embodiments of the invention may therefore be more accessible to the typical user, and easier and more enjoyable to use, than conventional audio production tools.

Abstract: An audio control apparatus particularly suited for an active outdoor type of person integrates multiple electronic devices. Audio source devices can include a mobile phone, a music player, and a walkie talkie. Audio sink devices can include a video recorder. The audio signals from the audio source devices may be mixed and sent to the video recorder to capture live audio signals. The user controls the audio control apparatus using either integrated or external UI buttons, which are context specific, configurable and allow control of the external devices from a UI. External UI buttons may be mounted to a helmet or other accessible location. A USB port on the external UI allows charging the audio control apparatus without needing to access or otherwise disturb the apparatus while it is nicely positioned within the helmet. Incoming phone and walkie talkie calls may be selectively allowed or blocked depending on sensor feedback.

Abstract: A system is provided to analyze cross-modulation distortion in audio devices, which may include testing with audio frequencies. One or more distortion signals from the audio device may be measured for an amplitude, phase, and or frequency modulation effect. In another embodiment a musical signal may be used as a test signal. Providing additional test signals to the audio device can induce a time varying cross-modulation distortion signal from an output of the audio device. Also utilizing at least one additional filter, filter bank, demodulator and or frequency converter and or frequency multiplier provides extra examination of distortion. Also frequency and or phase response can be measured with the presence of a de-sensing signal and or another signal that induce near slew rate limiting or near overload condition of the device under test.

Abstract: Acoustic sensing systems having improved vibration cancelation, and methods of achieving improved vibration cancelation. In one example, an acoustic sensing system includes an acoustic sensor configured to produce a sensor output signal representative of a response of the acoustic sensor to acoustic excitation and vibration excitation, at least one accelerometer configured to provide an acceleration signal responsive to the vibration excitation, and a controller, including an adaptive digital filter, coupled to the acoustic sensor and to the at least one accelerometer, and configured to receive the acceleration signal and to adjust coefficients of the adaptive digital filter so as to minimize coherence between a residual signal and the acceleration signal, the residual signal being a difference between the sensor output signal and a filter output signal from the adaptive digital filter.

Abstract: A system is provided to analyze cross-modulation distortion in audio devices, which may include testing with audio frequencies. One or more distortion signals from the audio device may be measured for an amplitude, phase, and or frequency modulation effect. In another embodiment a musical signal may be used as a test signal. Providing additional test signals to the audio device can induce a time varying cross-modulation distortion signal from an output of the audio device. Also utilizing at least one additional filter, filter bank, demodulator and or frequency converter and or frequency multiplier provides extra examination of distortion.

Abstract: An acoustic output device and a control method thereof are provided.

Abstract: A microphone assembly includes a capacitive MEMS transducer and a bias circuit having a DC output coupled to the transducer. The microphone assembly also includes a differential amplifier having a first input coupled to a first output of the transducer, the differential amplifier having a second input coupled to a second output of the transducer. The microphone assembly further includes a first impedance matching network and a second impedance matching network configured to balance a first capacitive load at the first input of the amplifier and a second capacitive load at the second input of the amplifier. Electrical signals applied by the transducer to the first and second inputs of the amplifier are matched or approximately matched in magnitude and 180 degrees or approximately 180 degrees out of phase with respect to each other.

Abstract: Methods, systems, computer-readable media, and apparatuses for gesture recognition are disclosed for audio control. The gestures can be used for audio control in virtualized environments. The audio control can include generating a sound that mimics an effect of a modification of a sound wave that would occur if the gesture were performed in a non-virtualized environment.

Abstract: A method for reducing the latency period of a filter bank for filtering an audio signal. A large number of signal blocks in the time domain are formed from the audio signal, wherein for at least a plurality of the signal blocks in each instance a filter function is predetermined, at least one partial interval of the signal block is predetermined as a prediction period, signal components of the signal block in the at least one partial interval are estimated for the prediction period, and a predicted signal block is generated from the signal components estimated for the prediction period and from the signal components of the signal block outside the prediction period. The predicted signal block, filtered with the predetermined filter function, is transformed into the frequency domain to form a transformed signal block. Signal components of the transformed signal block are output for further processing.

Abstract: Methods and apparatuses for addressing open space noise are disclosed. In one example, a method for masking open space noise includes receiving a microphone output signal from a microphone, the microphone one of a plurality of microphones in an open space. The method includes detecting a presence of a noise source from the microphone output signal, and determining whether the noise source is capable of being masked with a noise masking sound. The method further includes increasing a volume of a noise masking sound output from a loudspeaker responsive to a determination that the noise source is capable of being masked, the loudspeaker located in a same geographic sub-unit of the open space as the microphone. The loudspeaker is one of a plurality of loudspeakers in the open space.

Abstract: The invention relates to a method and a loudspeaker for storing equalization settings in an active loudspeaker including a loudspeaker cabinet, at least one driver connected to the cabinet and an amplifier electronics including an equalizer, in which method the loudspeaker is calibrated in controlled conditions for forming factory calibration parameters for controlling the equalizer, the factory calibration parameters controlling the equalizer are stored in the amplifier electronics. In accordance with the invention the factory calibration parameters are also stored in a memory fixedly mounted to the loudspeaker.

Abstract: A terminal device includes: a first housing including a first opening and a second opening through which sound of a speaker is output; a second housing which is turnably coupled to the first housing and closes the first opening based on a turn angle relative to the first housing; and a closing member which closes the second opening in a state where the first opening is opened by the second housing.

Abstract: Methods and apparatuses for addressing open space noise are disclosed. In one example, a method for masking open space noise includes receiving a plurality of mobile device microphone data from a plurality of mobile devices. A location data associated with each mobile device in the plurality of mobile devices is received. A plurality of stationary microphone data is received from a plurality of stationary microphones. A sound masking noise output is adjusted at one or more loudspeakers responsive to the plurality of mobile device microphone data and the plurality of stationary microphone data.

Abstract: The sound reproduction includes generating from a customized audio signal an acoustically isolated sound field at a position dependent on a sound field position control signal; providing a listening position signal representing a position of a listener and a listener identification signal representing the identity of the listener; and processing the listening position signal, the listener identification signal, and an audio signal. The sound reproduction further includes controlling, via the sound field position control signal, the position of the sound field dependent on the listening position signal so that the position of the sound field is at the position of the listener; and processing the audio signal according to an audio setting dependent on the identity of the listener to provide the customized audio signal.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: Techniques are provided for QR Decomposition (QRD) based minimum variance distortionless response (MVDR) adaptive beamforming. A methodology implementing the techniques according to an embodiment includes receiving signals from microphone array, identifying signal segments that include a combination of speech and noise, and identifying signal segments that include noise in the absence of speech. The method also includes calculating a QRD and an inverse QRD (IQRD) of the spatial covariance of the noise components. The method further includes estimating a relative transfer function (RTF) associated with the source of the speech, based on the noisy speech signal segments, the QRD, and the IQRD. The method further includes estimating a multichannel speech-presence-probability (SPP) on whitened input signals based on the IQRD. The method further includes calculating beamforming weights, for the microphone array, based on the RTF and the IQRD, to steer a beam in the direction associated with the speech source.

Abstract: Devices connected with a computer system are interrogated to detect whether they are configured and functioning. In an audio device implementation, a known audio sample is output to a loudspeaker. If the loudspeaker is properly configured and functional, an attached microphone will capture the sound corresponding to the audio sample generated by the loudspeaker. If sound corresponding to the audio sample is detected, an indication is provided to the user that the microphone and loudspeaker are operational. If no sound corresponding to the audio sample is detected, an indication is provided to the user that the loudspeaker is not enabled and additional configuration is required.