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: Directional sound location is used to determine security events of a security system. A sound sensor reports a vector direction to a sound source of an acoustic signal. If the vector direction is historically observed at similar times, then perhaps the sound source is not considered a security event. However, if the vector direction is unknown or unfamiliar, the security system may infer the security event and notify emergency services.

Abstract: An earphone includes: a housing; a dynamic driver unit provided in the housing; and a sound conduit having a length of approximately 10 mm or more, the sound conduit being configured to transmit sound output from the dynamic driver unit.

Abstract: The present technology relates to a sound processing device, a sound processing method, and a program, which can collect a desired sound. A sound processing device includes: a sound collection unit configured to collect a sound; an application unit configured to apply a predetermined filter to a signal of the sound collected by the sound collection unit; a selection unit configured to select a filter coefficient of the filter applied by the application unit; and a correction unit configured to correct the signal from the application unit. The selection unit selects the filter coefficient on the basis of the signal of the sound collected by the sound collection unit. The selection unit creates, on the basis of the signal of the sound collected by the sound collection unit, a histogram which associates a direction where the sound occurs and a strength of the sound and selects the filter coefficient on the basis of the histogram. The present technology can be applied to a sound processing device.

Abstract: A speaker system is removably attached to a cellular telephone or telephone case to augment or supplant audio output of the phone. In one example, the system includes a protective telephone case with built in speakers. The attachment mechanism may alternatively or additionally be used as a universal attachment for cellular telephone accessories. The system may include a fastening mechanism configured to attach any accessory to a phone surface.

Abstract: An audio device includes an audio amplifier configured to receive an input signal and generate a differential output signal. A first signal combiner circuit is configured to generate a time-convolution signal of an analog current signal and an analog voltage signal. The analog current signal corresponds to a current at the differential output signal, and the analog voltage signal corresponds to a voltage across the differential output signal. A second signal combiner circuit is configured to subtract the generated time-convolution signal from the input signal.

Abstract: A method for determining equalization filter parameters for a headphone, the method includes determining a composite response curve based on an average of amplitude response values measured from a plurality of measurement locations, the plurality of measurement locations, in cumulative, substantially spanning at least the headphone transducer, and determining the equalization filter parameters based on the determined composite response curve.

Abstract: The invention relates to a subsidiary loudspeaker for vehicles and bikes, including a tail pipe, a host module and an eternal controller; the host module receives imitation commands from the external controller, a central processing unit operates and outputs a control signal according to the detected parameter values of power rotation speed and vehicle speed to adjust the imitated exhaust sound output by a speaker in the tail pipe; to thereby adjust the exhaust sound, to meet low noise requirements of environmental law and reduce occurrence of vehicle accidents.

Abstract: A system and method for controlling noise originating from a source external to a vehicle is disclosed. The method includes determining, by an active noise controller of a vehicle, characteristics of an unwanted noise. The unwanted noise originates from a source external to the vehicle. The method also includes determining an inverted noise based on the characteristics of the unwanted noise. The method also includes projecting the inverted noise. The projected inverted noise destructively interferes with the unwanted noise. The method also includes receiving a residual noise via an error microphone. The error microphone is configured to generate a signal based on the received residual noise.

Abstract: Techniques of performing binaural rendering involve separating symmetric and antisymmetric terms in the total output rendered in the ears of a listener. Along these lines, a sound field includes a set of sound field weights corresponding to spherical harmonic (SH) functions in a SH expansion of the sound field. In addition, an aggregate head-related transfer function (HRTF) includes a set of HRTF weights that correspond to a SH function. An HRTF weight may be generated from aggregating products of an HRTF at each of a set of loudspeaker positions and a SH function to which the HRTF weight corresponds at that loudspeaker position. The rendered sound field in one of the ears of the listener would be, when the sound field and HRTF is a function of frequency, a sum of the products of corresponding sound field weights and HRTF weights. One may save much computation by grouping the products into symmetric terms and antisymmetric terms.

Abstract: A filter generator (100) generates a filter on the basis of band information (frequency) and gain characteristics (gain value) set by a user. The filter generator (100) obtains weighting factor information on the basis of the band information selected by the user and calculates a gain difference between a gain value used in a preceding filtering process and the new gain value selected by the user. The filter generator (100) then obtains a correction gain by multiplying the weighting factor information by the gain difference and generates a filter by multiplying a coefficient of the filter used in the preceding filtering process by the correction gain.

Abstract: A Bluetooth earphone combined with a pair of eyeglasses includes an earphone body and an engaging member. The earphone body includes a recess in which two axle holes and ratchets are designed. The engaging member includes a coupling portion, a ratchet wheel provided at the bottom of the engaging member, and two shafts provided on the ratchet wheel. The shafts are engaged in the axle holes, and the ratchet wheel are coupled with the ratchets, such that the engaging member can rotate at a proper angle relative to the earphone body with the shafts as a center of rotation and can be sustained in an adequate angle. As such, the Bluetooth earphone is flexibly combined with a pair of eyeglasses owned by a user in usage.

Abstract: An electronic component with a self-damping MLCC is provided. The electronic component comprising a pulse signal generator and a substrate comprising first traces and second traces. An MLCC is provided comprising a first capacitive couple between two first external terminations and a second capacitive couple between two second external terminations wherein each first external termination is in electrical contact with a different first trace and each second external termination is in electrical contact with a different second trace. The pulse signal generator provides a first pulse to the first traces and a second pulse to the second traces wherein the first pulse and second pulse are not in phase.

Abstract: There is provided a signal processing apparatus advantageous in terms of sound source separation performance. The signal processing apparatus includes a dividing unit configured to divide audio signal acquired by a plurality of audio acquisition units into components of a plurality of different frequency bands, and a processing unit configured to form, based on the audio signal, a plurality of directional beams having different directivities in accordance with a target direction and a target width. Each of the plurality of directional beams has directivities in different directions for the respective components of the frequency bands divided by the dividing unit.

Abstract: A method for forming a complete ring network of a plurality of Bluetooth® speakers, the method including populating a configurable speaker register of each of the plurality of Bluetooth® speakers with an address of an upstream Bluetooth® speaker that is in the plurality of Bluetooth® speakers, populating the configurable speaker register of each of the plurality of Bluetooth® speakers with an address of a downstream Bluetooth® speaker that is in the plurality of Bluetooth® speakers, and coupling an audio source to one Bluetooth® speaker of the plurality of Bluetooth® speakers.

Abstract: An apparatus including: a router configured to route a first audio signal to a local loudspeaker apparatus; and a switch configured to route at least one further audio signal, wherein in a first mode of operation the switch is configured to route the at least one further audio signal to at least one remote loudspeaker apparatus dependent on determining the presence of at least one remote loudspeaker apparatus, and in a second mode of operation the switch is configured to route the at least one further audio signal to the local loudspeaker apparatus.

Abstract: The present disclosure relates to a method for estimating one or more compensation parameters from impedance measurements of an acoustic load. The method comprises a series of steps including providing a probe assembly measurement setup and the disclosure also relates to a system to perform impedance measurements. The compensation parameters are estimated through a minimization process, preferably minimizing an error estimate of a first real and second imaginary part of the reflectance. The resulting compensation parameters are used to restore causality of the reflectance and accordingly provide more accurate impedance measurements.

Abstract: A magnetic sensor mounted on a headband of the personal audio delivery device may output a sensor signal indicative of an interaction between a magnetic field of a transducer of a personal audio delivery device and the magnetic sensor. A head size of a head on which the personal audio delivery device is worn is calculated based on the sensor signal from the magnetic sensor. Based on the head size, a non-linear transfer function is identified which characterizes how sound is transformed via the head with the calculated head size. An output signal is generated indicative of one or more audio cues to facilitate spatialization of sound associated with the output signal based on the identified non-linear transfer function. The sound associated with the output signal is output by the transducer of the personal audio delivery device.

Abstract: A method and an apparatus for separating speech data from background data in an audio communication are suggested. The method comprises: applying a speech model to the audio communication for separating the speech data from the background data of the audio communication; and updating the speech model as a function of the speech data and the background data during the audio communication.

Abstract: In some examples, sound verification may include a speaker device that may be configured to transmit sound at a dynamic volume level and a listening device that may be configured to receive the sound and provide feedback to the speaker device based on the received sound. The primary transceiver device may be further configured to adjust the dynamic volume level based on the feedback provided by the secondary transceiver device.