Abstract: An acoustical performance evaluation method includes a reference sound pressure decision step and a deviation calculation step. In the reference sound pressure decision step, frequency response of sound pressure level, obtained by considering only an influence of a first factor of a plurality of factors causing a deterioration in frequency response of sound pressure level of a loudspeaker to be evaluated, are decided as a reference sound pressure. The reference sound pressure is frequency response of sound pressure level of the loudspeaker in the anechoic chamber calculated through a simulation or measured through a measurement experiment. In the deviation calculation step, a deviation between the reference sound pressure and a target sound pressure is calculated as an evaluation index of acoustical performance. The target sound pressure is frequency response of sound pressure level of the loudspeaker in the anechoic chamber calculated through a simulation or measured through a measurement experiment.

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 head band structure of headset includes a retractable adjusting block and a head band group. The retractable adjusting block includes an upper cover, a sliding slice, an elastic piece and a lower cover. The sliding slice has at least one blocking portion. A substantial middle of the elastic piece is arched upward to form a convex surface. The head band group is connected with the retractable adjusting block. The head band group has a head band, and two end covers disposed to two opposite ends of the head band. A surface of each end cover has a waveform locking portion, and at least one stopping frame disposed at one end of the waveform locking portion. The convex surface keeps contacting with the waveform locking portion.

Abstract: At least one exemplary embodiment is directed to a method of suppressing cable acoustic noise or reducing microphone squeal.

Abstract: Embodiments of wireless audio systems and methods for wirelessly communicating audio are disclosed herein. In one example, a method for generating a 3D audio representation of an audio is disclosed. The method includes collecting, by a first wireless headphone, a first audio signal of the audio and generating, by the first wireless headphone, a first synchronizing signal based on a local clock of the first wireless headphone. The method also includes collecting, by a second wireless headphone, a second audio signal of the audio and generating, by the second wireless headphone, a second synchronizing signal based on a local clock of the second wireless headphone. The method yet includes generating, by a user equipment, the 3D audio representation of the audio based on the first and the second audio signals and the first and the second synchronizing signals.

Abstract: A vehicle engine sound control system identifies a vehicle driver by a driver smartphone or a driver biometric information detecting sensor and analyzes the music to which the identified driver listens with the driver smartphone or a vehicle infotainment system. A traveling pattern of the driver is analyze by applying any one among a vehicle, a GPS, a road, and weather as a condition. A driver propensity engine sound pattern is generated as a result value by learning at least any one information among a driver identifying unit, a music analyzing unit, and a travel analyzing unit. The engine sound is adjusted and output based the result value.

Abstract: A speaker device includes a first output section that outputs a first signal and a second output section that outputs a second signal distinguishable from the first signal.

Abstract: A method and system directed to controlling Active Noise Reduction (ANR) audio devices with active noise reduction. The system generates one or more control signals, using a controller, to set one or more ANR parameters of a first and a second wearable audio device to a first ANR state; detects at least one of: whether the first wearable audio device is engaged with or removed from a first ear of a user, using a first sensor of the first wearable audio device; or whether a second wearable audio device is engaged with or removed form a second ear of a user, using a second sensor of the second wearable audio device; and automatically adjusts the one or more ANR parameters of the first and/or second wearable audio device to a second ANR state when either the first wearable audio device or the second wearable audio device, or both, are removed from an ear of the user, wherein the second ANR state comprises a reduction in a level of ANR at least at some frequencies compared to the first ANR state.

Abstract: Various implementations include speaker systems. In some particular cases, a speaker system includes: at least two distinct loudspeakers each capable of outputting stereo audio on its own, wherein each loudspeaker comprises: a plurality of transducers, and a controller coupled with the plurality of transducers, where the controller in each loudspeaker is configured to: receive a left channel audio input signal and a right channel audio input signal; extract a center channel signal from the left channel audio input signal and the right channel audio input signal; and provide an audio output signal to at least one of the plurality of transducers for outputting the center channel signal.

Abstract: A private audio system for a 3d like sound experience for a vehicle passenger comprising: at least one acoustic sound emitter configured to transmit sound from an audio source in an audio frequency; at least one ultrasonic speaker configured to transmit an audio beam from an audio source, wherein the audio beam is orientable relative to the passenger; a passenger monitoring unit configured to track a location of a passenger's head; a signal processor configured to detect the location of the passenger's ears and to control the orientation of the ultrasound audio beam based on information provided by the passenger monitoring unit.

Abstract: An integrated circuit for implementing at least a portion of a personal audio device may include an output for providing an output signal to a transducer, wherein the output signal includes a pilot signal, a microphone input for receiving a microphone signal from a microphone indicative of an output of the transducer, and a processing circuit. The processing circuit may be configured to implement a pilot signal control to apply an adjustment to the pilot signal as necessary to maintain the pilot signal at a substantially constant magnitude regardless of proximity of the transducer to a pinna and implement a proximity determination block configured to determine proximity of the transducer to the pinna based on the adjustment.

Abstract: The present invention provides a mobile terminal, including: a housing including an upper cover, a lower cover covering the upper cover to form an accommodating space, and a sound channel penetrating through the upper cover; and a speaker unit accommodated in the accommodating space of the housing; the speaker unit dividing the accommodating space into a front sound cavity and a rear cavity; the sound channel communicating the front sound cavity with an outside; and where in, the speaker unit is fixed on the upper cover and enclosed together with the upper cover to from the front sound cavity; the sound channel is filled with a sound absorbing material.

Abstract: The technology provides for a pair of earbuds. For instance, a first earbud and a second earbud may each include one or more sensors. The pair of earbuds may further include one or more processors configured to receive one or more first sensor signals from a first sensor in the first earbud and one or more second sensor signals from a second sensor in the second earbud. The one or more processors may compare the first sensor signals with the second sensor signals. Based on the comparison, the one or more processors may detect whether the first earbud and the second earbud are being worn by different users. Based on detecting that the first earbud and the second earbud are being worn by different users, the one or more processors may control the first earbud and the second earbud to use a shared mode.

Abstract: Mechanisms are provided for conducting a natural language dialogue between the automatic dialogue system and a user of a client computing device. An automatic dialogue system receives natural language text corresponding to a user input from the user via the client computing device, the natural language text having an ambiguous portion of natural language text. The automatic dialogue system analyzes user profile information corresponding to the user to identify an anomaly in the user profile information and predicts a user intent associated with the anomaly. The automatic dialogue system disambiguates the ambiguous portion of the natural language text based on the predicted user intent and generates a response to the user input based on the disambiguated natural language text which is output to the client computing device to thereby conduct the natural language dialogue.

Abstract: An audio device including a housing having a side facing outwardly of the housing, the housing further having a first mounting position disposed on the side. The side has a first portion and a second portion, the first portion opposite the second portion relative to the first mounting position. The audio device also includes a first omnidirectional microphone disposed on the first portion. The audio device also includes a second microphone disposed on the first mounting position. The second microphone may be a second-order microphone. The second-order microphone has a first-order microphone disposed within a first platform mounted a distance from the first mounting position. The audio device also includes a first speaker disposed on the second portion. The first speaker is further disposed such that an audio output of the first speaker falls within a null of the second microphone.

Abstract: A method is presented for estimating and suppressing reverberation from a digital reverberant signal. A method for changing a first reverberation estimation according to another reverberation estimation is further provided. A method for controlling the reverberation suppression rate is also presented.

Abstract: A method for performing DRC on a HOA signal comprises transforming the HOA signal to the spatial domain, analyzing the transformed HOA signal, and obtaining, from results of said analyzing, gain factors that are usable for dynamic compression. The gain factors can be transmitted together with the HOA signal. When applying the DRC, the HOA signal is transformed to the spatial domain, the gain factors are extracted and multiplied with the transformed HOA signal in the spatial domain, wherein a gain compensated transformed HOA signal is obtained. The gain compensated transformed HOA signal is transformed back into the HOA domain, wherein a gain compensated HOA signal is obtained. The DRC may be applied in the QMF-filter bank domain.

Abstract: A power supply circuit is a power supply circuit that generates a direct current (DC) voltage from an alternating current (AC) voltage, and includes a rectifying circuit that rectifies the AC voltage, a capacitor that smooths the rectified voltage, a first switching element disposed on a path for charging the capacitor, a comparison circuit that compares an index voltage depending on the smoothed voltage with a reference voltage, and a driving circuit that sets the first switching element to an ON state, in a state in which the index voltage is lower than the reference voltage, and sets the first switching element to an OFF state, in a state in which the index voltage exceeds the reference voltage.

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: A hearing assistive system includes a personal communication device and a head-worn device. The personal communication device has a user interface for user interaction, a processor controlling the user interface and for running an application program, and input and output transducers. The head-worn device has an input transducer adapted for converting sound into an electric signal applied to a processor outputting a modified audio signal via an output transducer. The application program generates a data packet modulated on an audio carrier for output via the output transducer, decodes a data packet modulated on an audio carrier and received via the input transducer. The head-worn device has an audio signaling block having a detector block and a response block. The detector block detects and decodes the data packet received by the input transducer. The response block generates a data packet modulated on an audio carrier for output via the output transducer.