Abstract: An image capture device includes a processor for wind noise processing. The processor receives signals from a first microphone, a first plurality of microphones, and a second plurality of microphones. The processor may segment each signal into low frequency bins and high frequency bins. The processor may select a minimum level signal bin for each low frequency bin. For the high frequency bins, the processor may select a minimum level signal bin for a first group of microphones or a second group of microphones. The processor may generate a composite signal by combining the selected minimum level signal bins for each low frequency bin and the selected minimum level signal bins for each high frequency bin.

Abstract: A hearing device, e.g. a hearing aid, configured to play sound into an ear canal of a user, the hearing device comprising at least one input transducer (ITq) for picking up sound sq at said at least one input transducer from a sound field S around the hearing device and providing corresponding at least one electric input signal(s) INq representing said sound sq, i=1, . . .

Abstract: A microphone assembly includes: at least three microphones for capturing audio signals from the user's voice, the microphones defining a microphone plane; an acceleration sensor for sensing gravitational acceleration in at least two orthogonal dimensions so as to determine a direction of gravity; a beamformer unit for processing the captured audio signals in a manner so as to create a plurality of N acoustic beams, a unit for selecting a subgroup of M acoustic beams from the N the acoustic beams; an audio signal processing unit having M independent channels for producing an output audio signal for each of the M acoustic beams; a unit for estimating the speech quality of the audio signal in each of the channels; and an output unit for selecting the signal of the channel with the highest estimated speech quality as the output signal of the microphone assembly.

Abstract: One or more sensors are configured to contextualize a series of user generated movements to control one or more electronic devices. For example, a set of earphones comprises one or more sensors for sensing a location of the earphones. The one or more sensors enable earphones such as a pair of bluetooth or earphones wirelessly coupled to a bluetooth enabled electronic device, the capability to understand the configuration of use of the earphones. Based on a location and use or non-use of the earphones, one or more contextual responses is able to be applied for a given action.

Abstract: A head-mounted wearable device provides many functions to a user. The functions may be provided based in part on whether the device is being worn (donned) or not worn (doffed). Force sensing resistors in the device provide output indicative of forces associated with wearing the device. A time series of values indicative of the magnitude of a force may be determined. An increase or decrease in force values measured at different times may indicate that the device has been donned or doffed, respectively. If a subsequent force value does not differ substantially from a previous force value, this may indicate that the device has remained donned or doffed.

Abstract: The disclosure of the present specification aims to provide a directivity control system for speakers, which is capable of easily controlling directivity of acoustic data to provide desired sound fields with a plurality of speakers, and freely producing sound fields with high accuracy. A plurality of physically connected speaker units are virtually arrayed at respective directivity angles according to control signals that define the directivity angles. Virtual delay time data is calculated for the case where data is outputted with the virtual array, and output is performed according to output data affected by the delay time data, so that desired sound fields can be formed with the speaker units remaining fixed, without the necessity of mechanically rotating the speaker units.

Abstract: An exemplary passive acoustic proximity detection system is configured to determine a first acoustic spectrum of a first signal representative of audio detected and output by a first microphone configured to be positioned at an ear canal entrance of a user. The detection system is further configured to determine a second acoustic spectrum of a second signal representative of audio detected and output by a second microphone configured to be located away from the ear canal entrance. Based on a comparison of the first acoustic spectrum and the second acoustic spectrum, the detection system is configured to generate a proximity indicator indicative of a proximity of an object to the first microphone. Based on the proximity indicator, the detection system is configured to select a signal processing program for execution by the passive acoustic detection system.

Abstract: An electronic device is disclosed. The electronic device comprises a communication unit, a speaker, and a processor for: outputting a test sound through the speaker when a preset signal is received from an external terminal device through the communication unit; acquiring, on the basis of sound data, reverberation time information for each frequency of the test sound and size information on a space in which the electronic device is positioned when sound data acquired by recording a test sound in a terminal device is received through the communication unit; acquiring, on the basis of the reverberation time information for each frequency and the size information on a space, a sound absorption rate of an object arranged in the space; and identifying information on the object on the basis of the sound absorption rate.

Abstract: An audio signal processing apparatus includes; a first correlation component separating unit configured to predict a first signal from a second signal in a predetermined period to generate a first correlation component signal and to separate the first non-correlation component signal from the first signal by using the first correlation component signal; a second correlation component separating unit configured to predict a second signal from the first signal in the predetermined period to generate a second correlation component signal and to separate the second non-correlation component signal from the second signal by using the second correlation component signal; a correlation component synthesizing unit configured to synthesize the first correlation component signal and the second correlation component signal to generate a synthesized correlation component signal; a first gain multiplying unit configured to multiply the synthesized correlation component signal by a gain to generate a correlation component

Abstract: A specialized audio/instrument cable with built-in digital signal processing capabilities that adds user-defined audio effects (such as reverb, delay, chorus and/or distortion) from within the cable itself to affect the sound generated from an instrument or microphone such that the cable is the only connection needed between the instrument or microphone and an output device (such as an amplifier, PA, powered speaker, music mixer, or a recording device). The audio effects used by the cable can be changed via (i) an app from a smartphone, tablet, computer or other electronic device; (ii) a wireless controller that attaches to the instrument; (iii) a pedal, and/or (iv) any other type of wireless controller that has the ability to communicate with a smartphone/tablet/computer or other electronic device.

Abstract: An amplifier having one or more channels where each channel includes a two half bridges (a master and slave sub-channel). The sub-channels can be connected either in parallel or in a full-bridge configuration via internal switches that route signals to a pair of speaker jacks. One switch in the amplifier has a first position that selectively connects the outputs of the master and slave sub-channel to the same input of the speaker load so that the two sub-channels will drive the speaker load in parallel and a second position where the output of the slave sub-channel is connected to another input of the speaker load so that the master sub-channel and the slave sub-channel will drive the speaker load in a Full-bridge configuration.

Abstract: Example techniques involve a calibration state variable. An example implementation receives, via a network interface, an indication that the first playback device is calibrated. Based on receiving the indication that the first playback device is calibrated, the example implementation updates a calibration state variable to indicate that the first playback device is calibrated, wherein the calibration state variable is stored in the data storage. The example implementation sends, via the network interface, an indication of the updated calibration state variable to a second device.

Abstract: A speaker driving device includes a first calculation unit, a first driving signal generation unit and a third calculation unit. The first calculation unit outputs a first calculation signal obtained from a first input signal based on response characteristics according to a first parameter defining an equivalent circuit of a first speaker unit. The first driving signal generation unit generates a first driving signal based on a second driving signal and the first calculation signal. The first driving signal drives a first output speaker unit. The third calculation unit generates a third calculation signal from a second input signal based on response characteristics according to a third parameter defining an equivalent circuit of a third speaker unit. The second driving signal generation unit generates the second driving signal based on the first driving signal and the third calculation signal. The second driving signal drive a second output speaker unit.

Abstract: In one embodiment, a portable computing device is described. The portable computing device includes a first surface comprising at least one user interface. The portable computing device also includes a second surface opposite the first surface. Further, the portable computing device includes at least one speaker port in the first surface. Further yet, the portable computing device includes a collapsible speaker chamber configured on the second surface opposite the at least one speaker port. Moreover, the portable computing device includes a speaker configured in the portable computing device between the speaker port and the collapsible speaker chamber.

Abstract: A system for facilitating configuration of an audio system tests the existing audio system configuration by playing reference audio clips of various different genres of programming, the audio of which is received by a microphone of a remote control device or other handheld mobile device at various different listening points in the room. The system then compares the audio signal representing audio of the reference audio clip received at the microphone to a reference audio signal representing the reference audio clip and determines other audio characteristics. The system then determines, based on the comparison and other characteristics of the room (e.g., furniture layout, room construction materials, wall treatments and current speaker positioning) suggested changes to a configuration of the audio system to increase audio quality of the audio system for playing audio associated with the various different genres of programming.

Abstract: A double-talk state can be accurately detected, and based on a detection result, echo can be appropriately suppressed. When a sound is output from a speaker and only the output sound is input to a microphone, a comparison is made, for each of different frequency bands, between a frequency mask generated based on a power spectrum or an amplitude spectrum for a learning signal transmitted through a transmitting signal path and a value of a power spectrum or an amplitude spectrum for an input signal input from the microphone, to detect whether there is a double-talk state. In a case of detecting that no signal is being transmitted through the transmitting signal path and that a signal is being transmitted through the receiving signal path, an echo suppressor is used to execute processing of suppressing an echo in the input signal.

Abstract: An audio capture apparatus comprises a first beamformer (303) which is arranged to generate a beamformed audio output signal. An adapter (305) adapts beamform parameters of the first beamformer and a detector (307) detects an attack of speech in the beamformed audio output signal. A controller (309) controls the adaptation of the beamform parameters to occur in a predetermined adaptation time interval determined in response to the detection of the attack of speech. The beamformer (303) may generate noise reference signal(s) and the detector (309) may be arranged to detect the attack of speech in response to a comparison of a signal level of the beamformed audio output signal relative to a signal level of the at least one noise reference signal.

Abstract: A speaker with an integrated air pressure and vibration mitigation system is used to produce high fidelity acoustic and vibratory output. The speaker has a main cabinet a set of ventilation holes, an acoustic driver, a vibration unit, and a dampening unit. The main cabinet is an enclosure that retains the acoustic driver, the vibration unit and the dampening unit in desired orientations. The ventilation holes traverse through the main cabinet and provide exhaust ports for unwanted air pressure force and vibrations to be expelled into the external environment. The acoustic driver is mounted to a first face of the main cabinet. The dampening unit is a filter that is mounted in between the vibration unit and the main housing and prevents the transmission of unwanted vibrations and air pressure between the vibration unit and the acoustic driver. Thus, the acoustic driver can oscillate without hinderance and produce distortion-free audio.

Abstract: The present invention provides a speaker, which includes a basket, a diaphragm fixed to the basket, and a voice coil located below the diaphragm and used for driving the diaphragm to vibrate and sound. The diaphragm includes a dome and a folded ring, and the dome includes first side edges arranged oppositely and second side edges connected with the first side edges and arranged oppositely. The dome includes two or more opening portions disposed separately along the first side edge or the second side edge, and an extended portion is arranged between two adjacent opening portions.

Abstract: The disclosure provides an audio signal processing circuit and an audio signal processing method, adapted for processing the input signal of a speaker configured with a rated power. The audio signal processing circuit includes, but not limited to, an audio signal generator and a power adjuster. The audio signal generator provides an audio signal including a high-frequency signal and a middle and low-frequency signal. The power adjuster is electronically coupled to the audio signal generator, and adjusts the power of the high-frequency signal according to an enhanced power larger than the rated power without adjusting the power of the middle and low-frequency signal. The output signal of the power adjuster can be inputted into the speaker. Accordingly, the transmission range for the high-frequency signal can be improved.