Abstract: A method for voice recording and an electronic device thereof are provided. The method includes determining a voice recording mode from among a plurality of voice recording modes, determining a voice beamforming direction according to the determined voice recording mode, and recording voice signals based on the determined voice beamforming direction.

Abstract: A hearing assistance system includes an adaptive binaural beamformer based on a multichannel Wiener filter (MWF) optimized for noise reduction and speech quality criteria using a priori spatial information. In various embodiments, the optimization problem is formulated as a quadratically constrained quadratic program (QCQP) aiming at striking an appropriate balance between these criteria. In various embodiments, the MWF executes a low-complexity iterative dual decomposition algorithm to solve the QCQP formulation.

Abstract: The invention is a filter circuit for noise cancelling earphones wherein one end of the filter circuit for noise cancelling earphones is connected to the circuit for noise cancelling microphones. In the circuit starting from one end of the said noise cancelling microphones circuit connected to the resistor R0 set the first current node. The other end of the said noise cancelling microphones circuit is connected to the ground while the other end of the said resistor R0 is connected to the high level.

Abstract: An audio device is described which comprises a housing holding electronic components and having a face being substantially covered by a layer of transparent material. The audio device has a display comprising a layer of display material supported under at least part of the layer of transparent material; and at least one display actuating element in the housing configured to flex the display and the layer of transparent material to generate sound waves. There is at least one air cavity formed between the layer of display material and the housing; and an entrance formed in the layer of transparent material to allow air to pass between the air cavity and an exterior of the device.

Abstract: A method and system for a passive headset with dynamically controlled LEDs, where the method comprises, in a passive headset comprising speakers, light emitting diodes (LEDs), and LED driver circuitry: receiving an electrical signal that includes an audio signal and an LED control signal, filtering out the LED control signal from the received electrical signal and communicating a resulting output audio signal to the speakers, filtering out the audio signal from the received electrical signal and communicating a resulting output LED control signal to the LED driver circuitry, generating a bias voltage for each of the one or more LEDs utilizing the LED driver circuitry and the output LED control signal, and configuring a light output of the LED utilizing the generated bias voltage. The amplifier may include a mixer that sums an audio signal with a control signal.

Abstract: The various embodiments set forth a headphone apparatus that includes a first earcup, a sensor included in the first earcup, and a controller. The first earcup is coupled to a headband, and includes a loudspeaker system and a thermal control subsystem. The controller is configured to detect an output generated by the sensor, and, based on the output, transmit a signal to the thermal control subsystem included in the first earcup to modify a temperature associated with at least the first earcup.

Abstract: A system for producing an encoded digital audio recording has an audio encoder that encodes a digital audio recording having a number of audio channels or audio objects. An equalization (EQ) value generator produces a sequence of EQ values which define EQ filtering that is to be applied when decoding the encoded digital audio recording, wherein the EQ filtering is to be applied to a group of one or more of the audio channels or audio objects of the recording independent of any downmix. A bitstream multiplexer combines the encoded digital audio recording with the sequence of EQ values, the latter as metadata associated with the encoded digital audio recording. Other embodiments are also described including a system for decoding the encoded audio recording.

Abstract: Provided is an earset. The earset includes an internal microphone provided inside an earphone worn on a user's ear, and configured to receive a first sound transferred from the user's mouth through an external auditory meatus via an Eustachian tube and convert the first sound into a first sound signal, at least one external microphone provided outside the earphone, and configured to receive a second sound provided from the user's mouth and convert the second sound into a second sound signal, and a controller configured to filter a noise of the second sound signal based on the first sound signal, remove the noise, and generate a third sound signal.

Abstract: There is provided audio equipment capable of being miniaturized by sharing a plurality of types of input terminals and output terminals and reducing footprints of the terminals. The audio equipment has a CPU, ROM, RAM, a USB connector, an SD card, a DAC (D/A converter), a changeover section, an amplifier, a PHONE (headphone) terminal, a DIGITAL I/O LINE-IN terminal, a switch, and a display section. The DIGITAL I/O LINE-IN terminal is a terminal which shares a digital audio optical input terminal, a digital audio coaxial input terminal, a digital audio coaxial output terminal, and an analog audio input terminal. The switch sets the terminal by switching.

Abstract: Provided are methods and systems for enhancing the intelligibility of an audio (e.g., speech) signal rendered in a noisy environment, subject to a constraint on the power of the rendered signal. A quantitative measure of intelligibility is the mean probability of decoding of the message correctly. The methods and systems simplify the procedure by approximating the maximization of the decoding probability with the maximization of the similarity of the spectral dynamics of the noisy speech to the spectral dynamics of the corresponding noise-free speech. The intelligibility enhancement procedures provided are based on this principle, and all have low computational cost and require little delay, thus facilitating real-time implementation.

Abstract: Various embodiments are directed to cooperation among communications devices having microphones to employ their microphones in unison to provide voice detection with noise reduction for voice communications. A first communications device comprises a processor circuit; a first microphone; an interface operative to communicatively couple the processor circuit to a network; and a storage communicatively coupled to the processor circuit and arranged to store a sequence of instructions operative on the processor circuit to store a first detected data that represents sounds detected by the first microphone; receive a second detected data via the network that represents sounds detected by a second microphone of a second communications device; subtractively sum the first and second data to create a processed data; and transmit the processed data to a third communications device. Other embodiments are described and claimed herein.

Abstract: A computing system with multiple audio devices local to an environment to provide synchronized audio output to a user. A remote system receives a local audio signal representing sound captured from an environment from a local audio device and a remote audio signal representing sound captured from the environment from the remote audio device. The server determines a delay associated with the local audio signal and a delay associated with the remote audio signals and, based on the delays, synchronizes the audio being output by the local and remote devices.

Abstract: Provided is a speaker device configured so that a time lag between a sound signal and a noise cancellation signal can be prevented, worsening of high-frequency characteristics can be avoided, and acoustic characteristics can be improved. A plane diaphragm 11 includes a flexible circuit board 20. A sound voice coil pattern 21 to which drive current corresponding to a sound signal is supplied and a noise cancellation voice coil pattern 22 to which drive current corresponding to a noise cancellation signal is supplied are formed on the flexible circuit board 20 and are formed corresponding to a formed magnetic field of a magnet 13.

Abstract: Embodiments are provided for equalizing audio data for output by a speaker of an electronic device based on a local position or orientation of the electronic device. According to certain aspects, the electronic device can determine (858, 868) its local position based on various sensor data, and identify (870, 872) an appropriate equalization setting. In some cases, the electronic device can modify (876, 880) the equalization setting based on acoustic and/or optical data. The electronic device can apply (882) the modified or unmodified equalization setting to an audio signal and cause the speaker to output (886) the audio signal with the applied equalization setting.

Abstract: According to an embodiment, a method of operating a speaker with an acoustic pump includes generating a carrier signal having a first frequency by exciting the acoustic pump at the first frequency and generating an acoustic signal having a second frequency by adjusting the carrier signal. In such embodiments, the first frequency is outside an audible frequency range and the second frequency is inside the audible frequency range. Adjusting the carrier signal includes performing adjustments to the carrier signal at the second frequency. Other embodiments include corresponding systems and apparatus, each configured to perform corresponding embodiment methods.

Abstract: An audio signal processing apparatus that includes a band division section, an analysis section, a gain adjustment amount calculation section, and a gain adjustment section. The band division section is configured to generate a resonance in-band signal by band division on an audio input signal. The analysis section is configured to extract an amount of features from each of the resonance in-band signal and the input signal. The gain adjustment amount calculation section is configured to calculate a gain adjustment amount for a resonance frequency band in the input signal, the gain adjustment amount being calculated based on the amount of features of the resonance in-band signal and the amount of features of the input signal. The gain adjustment section is configured to perform a gain adjustment on the resonance frequency band in the input signal based on the gain adjustment amount.

Abstract: Example methods and systems for displaying one or more indications that indicate (i) the direction of a source of sound and (ii) the intensity level of the sound are disclosed. A method may involve receiving audio data corresponding to sound detected by a wearable computing system. Further, the method may involve analyzing the audio data to determine both (i) a direction from the wearable computing system of a source of the sound and (ii) an intensity level of the sound. Still further, the method may involve causing the wearable computing system to display one or more indications that indicate (i) the direction of the source of the sound and (ii) the intensity level of the sound.

Abstract: For protecting a speaker, an input signal is received, and an excursion of the speaker that would be caused by the input signal is predicted. In response to the predicted excursion exceeding a threshold, a targeted excursion of the speaker is determined by compressing the predicted excursion. The targeted excursion is translated into an output signal, which is output to the speaker.

Abstract: A method for suppressing interference noise in an acoustic system with a microphone that generates an input signal and a loudspeaker that generates an acoustic signal which partially feeds back to the microphone. A first intermediate signal is formed along a primary signal path as a function of the input signal, and an output signal is formed via a frequency distortion. The output signal is coupled into a signal feedback path. A second intermediate signal is formed in the signal feedback path via a decorrelation and used as an input value for an adaptive filter. The adaptive filter generates a compensation signal which compensates the input signal. A third intermediate signal is formed from the input signal and/or compensated input signal, which is used as an input value for the adaptive filter. The output signal is fed to the loudspeaker for reproduction.

Abstract: Systems and methods discussed herein involve designating a default playback device in a local playback network. One method may involve determining that a networked microphone device is available for assignment to one or more playback zones within a local playback network, displaying a prompt to indicate at least one playback zone to which the network microphone device is to be assigned, receiving an input indicating a particular playback zone in the local playback system to which the network microphone device is to be assigned, and storing data indicating an assignment of the network microphone device to the particular playback zone.