Abstract: A method, and system, of digital room correction for a device, such as a smart speaker, including a loudspeaker. The method comprises capturing audio from an environment local to the device, for example from one or more microphones of a smart speaker. The captured audio is then processed to recognize one or more categories of sound. A digital room correction procedure may then be controlled dependent upon recognition and/or analysis of at least one of the categories of sound.

Abstract: A multi-zone media system and method for providing multi-zone media. As a non-limiting example, various aspects of this disclosure provide a system and method that flexibly presents media content (e.g., audio content) to a user as the user moves throughout a premises.

Abstract: An apparatus with acoustic enhancement and corresponding frequency response is disclosed. The apparatus includes a driver with a primary bass port having a primary bass port chamber and a secondary bass port having a secondary bass port chamber. The secondary bass port can be coupled to the primary bass port at one end and have substantially unimpeded air flow at another end. The apparatus may further include an acoustic chamber that is separated or isolated from either the primary bass port chamber, secondary bass port chamber, or both primary and secondary bass port chambers. A switch may also be included to dynamically control an air flow resistor at the latter end of the secondary bass port. The primary bass port chamber, secondary bass port chamber, acoustic chamber, and the air flow resistor can separately or collectively be used for tuning the frequency response according to the acoustic enhancement desired.

Abstract: A microphone including a light emitting part can be miniaturized. The microphone includes: a microphone unit; an impedance converter that converts output impedance of the microphone unit; a light source that notifies an operation state of the microphone unit; a conversion substrate on which the impedance converter is mounted; a light source substrate on which the light source is mounted; and a connection substrate to which a signal line for transmitting a signal from the impedance converter and a power line for transmitting power to the light source are connected. The conversion substrate, the light source substrate, and the connection substrate are three-dimensionally connected to one another to constitute one substrate unit.

Abstract: In some implementations, the disclosed technology determines that a microphone is likely closer to a sound source compared to other microphones partially based on determining a spectral standard deviation associated with an audio signal captured by the microphone. In some implementations, the disclosed technology determines a spectral standard deviation of audio signals associated with different microphones and determines that the microphone with the lowest spectral standard deviation is likely the closest microphone to a sound source. The disclosed technology can determine the closest microphone generally independent of input signal level, independent of microphone sensitivity, independent of microphone design, independent of microphone manufacturer, and independent of the sound source intensity (e.g., “loud” or “soft”).

Abstract: A vehicle and an audio processing method for the same are provided to allow a sound level output through a speaker to be maintained at a preset sound level by adjusting a volume of a power amplifier to a constant value based on modulation information of a received broadcast signal. The audio processing method includes receiving a broadcast signal and obtaining modulation information of the broadcast signal and determining volume information of the broadcast signal based on the modulation information. A volume of a power amplifier is adjusted based on the determined volume information to adjust an output sound level of the broadcast signal to correspond to a preset output sound level.

Abstract: Embodiments described herein provide a combined multi-source time difference of arrival (TDOA) tracking and voice activity detection (VAD) mechanism that is applicable for generic array geometries, e.g., a microphone array that lies on a plane. The combined multi-source TDOA tracking and VAD mechanism scans the azimuth and elevation angles of the microphone array in microphone pairs, based on which a planar locus of physically admissible TDOAs can be formed in the multi-dimensional TDOA space of multiple microphone pairs. In this way, the multi-dimensional TDOA tracking reduces the number of calculations that was usually involved in traditional TDOA by performing the TDOA search for each dimension separately.

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: An acoustic apparatus and method produces an acoustic signal in response to an electrical input signal applied to an acoustic receiver. The acoustic signal is converted to an electrical output signal that is proportional to a sound pressure of the acoustic signal, using an electro-acoustic transducer. In some embodiments the apparatus and method determine whether there is a change in the acoustic signal indicative of a change in an acoustic load coupled to the receiver by comparing the electrical output signal to reference information. The change in acoustic load, in one example, is attributable to ear wax accumulation in an output of the acoustic receiver or acoustic passage in the ear canal of a user or is attributable to seal leakage.

Abstract: A sound pickup device according to this embodiment includes a microphone part placed at a sound pickup position in close proximity to an ear hole without blocking an ear canal, and positioned so as to face outside of the ear canal, and a wire having flexibility and extending to the sound pickup position. Further, in this embodiment, at least part of the wire is formed along an auricle.

Abstract: A filter determination system, a filter determination device, a filter determination method and a program with which a filter can be appropriately determined are provided. A processing device according to this embodiment includes a frequency response acquisition unit that acquires a frequency response of a filter for an audio signal, a smoothing unit that smoothes the frequency response and obtains a smoothed response, a candidate point determination unit that determines candidate split points based on a bottom position of the smoothed response, and a split point determination unit that determines one or more band split points from the candidate split points.

Abstract: A sound bar system has couplings adaptable to irregular mounting challenges. An arcuate panel extends from a top edge of a front rectangular frame to a rear edge of a bottom rectangular frame. The arcuate panel has a plurality of parallel sets of three unique slots for receiving corresponding three-legged nuts for attaching clamps or similar connectors having 360° of rotational freedom. Single-legged sliders, each with a threaded rod extension, may also be used in one slot of said three unique slots. End brackets with 360° of rotational freedom and adjustable extension support the sound bar by the arcuate panel, not the end panels. More than one set of slots may be used in a single installation.

Abstract: Disclosed is an AI voice interaction device, including a main body and an earphone detachably connected with the main body. User's voice is picked up by a voice pick-up, digital voice signals are obtained through analog-to-digital conversion, and are further transmitted to a first communication module through an earphone communication module. A body processor implements communication connection with a cloud server, and transmits to the cloud server for voice recognition and semantic analysis. Afterwards, the cloud server implements the functions such as making phone calls, sending short messages through the cellular transceiver, or calls corresponding network data, and transmits the network data back to the wristband-type AI voice interaction device to perform corresponding voice broadcasting according to the network data. Further, a TWS earphone and a wristband-type AI voice interaction system are disclosed.

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 method of processing an audio signal is disclosed. According to embodiments of the method, magnitude response information of a prototype filter is determined. The magnitude response information includes a plurality of gain values, at least one of which includes a first gain corresponding to a first frequency. The magnitude response information of the prototype filter is stored. The magnitude response information of the prototype filter at the first frequency is retrieved. Gains are computed for a plurality of control frequencies based on the retrieved magnitude response information of the prototype filter at the first frequency, and the computed gains are applied to the audio signal.

Abstract: The various implementations described herein include methods, devices, and systems for automatic audio equalization. In one aspect, a method is performed at an electronic device that includes speakers, microphones, processors and memory. The electronic device outputs audio user content from the speakers and automatically equalizes subsequent audio output of the device without user input. The automatic equalization includes: (1) obtaining audio content signals, including receiving outputted audio content at each microphone; (2) determining from the audio content signals phase differences between microphones; (3) obtaining a feature vector based on the phase differences; (4) obtaining a frequency correction from a correction database based on the obtained feature vector; and (5) applying the obtained frequency correction to the subsequent audio output.

Abstract: This disclosure relates to a microphone system for a motor vehicle, having a first, second and third microphone and a signal processing device, which is configured to process respective signals provided by the microphones; wherein the signal processing device is configured to process the signals of the first and second microphones in such a manner that a driver directivity oriented towards a position of a driver seat is provided, or to process the signals of the second and third microphones in such a manner that a passenger directivity oriented towards a position of a passenger's seat is provided, and to provide a driver interfering noise directivity by processing the signals of two microphones together, and to further process a driver useful signal associated with the driver directivity in dependence on a driver interfering signal associated with the driver interfering noise directivity or to provide a passenger interfering noise directivity by processing the signals of two microphones together, and to furth

Abstract: The disclosure relates to a method and apparatus for acquiring spatial division information. The method includes controlling a sound source device to play a first sound signal; obtaining a second sound signal that is a sound signal collected by a sound collecting device when the first sound signal is propagated to the sound collecting device; obtaining direct intensity information based on the second sound signal, wherein the direct intensity information indicates an intensity of a direct sound signal in the second sound signal, wherein the direct sound signal is a sound signal that is generated by the sound source device and reaches the sound collecting device without physical reflection; and obtaining spatial division information based on the direct intensity information, wherein the spatial division information indicates whether the sound source device and the sound collecting device are in a same spatial zone.

Abstract: Techniques for simulating a microphone array and generating synthetic audio data to analyze the microphone array geometry. This reduces the development cost of new microphone arrays by enabling an evaluation of performance metrics (False Rejection Rate (FRR), Word Error Rate (WER), etc.) without building device hardware or collecting data. To generate the synthetic audio data, the system performs acoustic modeling to determine a room impulse response associated with a prototype device (e.g., potential microphone array) in a room. The acoustic modeling is based on two parameters—a device response (information about acoustics and geometry of the prototype device) and a room response (information about acoustics and geometry of the room). The device response can be simulated based on the microphone array geometry, and the room response can be determined using a specialized microphone and a plane wave decomposition algorithm.

Abstract: An encoder includes a signal detection circuit and a signal encoding circuit. The signal encoding circuit is configured to encode the Nth-frame downmixed signal when the signal detection circuit detects that an Nth-frame downmixed signal includes a speech signal, or when the signal detection circuit detects that the Nth-frame downmixed signal does not include a speech signal, encode the Nth-frame downmixed signal when the signal detection circuit determines that the Nth-frame downmixed signal satisfies a preset audio frame encoding condition, or skip encoding the Nth-frame downmixed signal when the signal detection circuit determines that the Nth-frame downmixed signal does not satisfy a preset audio frame encoding condition.