Abstract: Disclosed is a noise suppression technology for suppressing various types of noise including unknown noise without storing a large number of noise information in advance. Specifically disclosed is an auxiliary device connectable to an information processing apparatus. The information processing apparatus is provided with: means for suppressing noise in a noisy signal, generating the noise information, and suppressing the noise in the noisy signal by using the generated noise information; and noise information generation means for updating the noise information on the basis of the result of suppression of the noise in the noisy signal. The auxiliary device is provided with a mechanism unit for generating noise to be suppressed by the noise suppression means and a mechanism control unit for controlling the mechanism unit so that the noise occurs at a timing at which the noise suppression means performs a noise suppression process.

Abstract: There is provided a method of operating a device, the device comprising a plurality of audio sensors and being configured such that when a first audio sensor of the plurality of audio sensors is in contact with a user of the device, a second audio sensor of the plurality of audio sensors is in contact with the air, the method comprising obtaining respective audio signals representing the speech of a user from the plurality of audio sensors; and analyzing the respective audio signals to determine which, if any of the plurality of audio sensors is in contact with the user of the device.

Abstract: A matching system receives probe audio samples for comparison to references of a data store. Comparisons are generated to determine a sufficient match for a portion or a first amount of the probe sample. Ranking scores are assigned to the resulting match references. The match references are retained, unless meeting a score threshold. Comparisons are continually generated with second amounts of the probe sample and the retained references are updated with further matching references assigned ranking scores. The retained results are merged and determined to satisfy a score threshold for release as outputted results for matching references.

Abstract: An audio device, in at least one embodiment, includes a device package and a plurality of microphones. The device package defines a component cavity and a plurality of vias including a first via and a second via. The vias comprise openings in the device package extending between the component cavity and an exterior. The microphones are located within the component cavity. The microphones are configured to generate electrical signals in accordance with acoustic pressure in the respective vias. A disclosed audio apparatus includes an apparatus housing and a speaker and one or more noise cancellation microphones within the frame. The speaker includes a speaker diaphragm configured to vibrate in accordance with an audio signal. The apparatus housing may define a speaker cavity configured to mechanically support the speaker diaphragm.

Abstract: There is provided a wireless audio transmission system, in particular a wireless microphone system, for wireless audio transmission. The system has at least one wireless microphone unit for detecting audio signals and for wirelessly transmitting the detected audio signals, and a central unit. The central unit has a wireless receiving unit for receiving the wirelessly transmitted audio signals from the wireless microphone units, a pilot tone transmitting unit for transmitting a first pilot tone on a pilot channel, and a pilot tone receiving unit for receiving and processing a second pilot tone signal. The pilot tone signal has information in respect of the wireless transmission between the microphone units and the central unit.

Abstract: A diaphragm is arranged on the upper surface of a silicon substrate so as to cover a chamber in the silicon substrate. Multiple anchors are provided on the upper surface of the silicon substrate, and the lower surfaces of corner portions of the diaphragm are supported by the anchors. Also, a fixed electrode plate is provided above the diaphragm with an air gap therebetween. In a view from a direction perpendicular to the upper surface of the silicon substrate, the entire length of the outer edge of the diaphragm located between adjacent anchors is located outward of a line segment that circumscribes the edges of the adjacent anchors on the side distant from the center of the diaphragm. Also, one or two or more through-holes are formed in the diaphragm in the vicinity of the anchors.

Abstract: An acoustic device captures sound using two or more microphones and filters the sound into a plurality of sub-bands. For each of the plurality of sub-bands, the device identifies sound captured by the microphones from at least two directions and attenuates the sound captured by the microphones, within each of the sub-bands, in substantially one of the directions.

Abstract: Apparatus comprising: at least one transducer configured to generate at least one lower frequency acoustic signal for output by a surface when in contact with the apparatus and at least one higher frequency acoustic signal for output by air conduction.

Abstract: A computer-implemented communications method is discussed. The method includes receiving a typed user input in a texting conversation on an electronic communication channel between two or more users of electronic communications devices; analyzing the user input to identify whether the user input contains a reserved word for triggering a search; submitting a search query received in association with the reserved word to a search engine, if the user input contains a reserved word; receiving one or more search results from the search engine; and providing, in the electronic communication channel for incorporation into the texting conversation, at least one of the one or more search results.

Abstract: The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In various embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible.

Abstract: Apparatus and methods are disclosed for simplifying and improving the reliability of communication between end users and access control systems such as those that include door and/or gate controllers, tenant access to buildings, and the like. In certain embodiments using an analog modem at the controller end of the system, the corresponding remote analog modem (which communicates with that analog modem at the controller end) is not within the direct control of the end user, and the end user's communicates instead via the user's standard hardware/software for communicating generally, such as via the Internet. Depending on the application, the control data can be maintained and/or synched at any selected combination or sub-combination of several locations: (1) the user's local system; (2) an intermediate server system; and/or (3) the controller end system.

Abstract: A system and method for improved directory assistance searches involving feedback from previous calls made to telephone numbers in a vendor database, in order to improve the quality of customer service and satisfaction by the vendors associated with the telephone numbers, and to find the vendors associated with the telephone numbers that have provided quality customer service in the past. The system and method automatically rates a telephone number according to a plurality of parameters. In one embodiment of the system, dirty numbers (numbers reassigned to a second party from a previous party) are screened.

Abstract: A call calculator and method thereof used to automate testing of a voice self service platform. The call calculator may have functional logic to calculate an estimated call load. The type of load may be least one of cyclic, peak, burst, choppy, and sustained. The call calculator may use a set of input parameters of an algorithm to calculate the estimated call load, wherein the algorithm may be based on an integral calculus model.

Abstract: A sound processing device includes a first calculation unit configured to calculate a suppression gain of noise by using respective input signals input from a plurality of microphones; an integration unit configured to obtain an integration gain by using a suppression gain of an acoustic echo and the suppression gain of the noise; an application unit configured to apply the integration gain to one input signal among the plurality of input signals; and a second calculation unit configured to calculate the suppression gain of the acoustic echo by using signals to which the integration gain is applied, output signals that are output to a replay device, and the one input signal.

Abstract: A method for performing a surround audio compatibility assessment on a plurality of original surround channels is described herein. A surround audio compatibility assessment system accepting original surround signals is also described herein.

Abstract: A sound signal processing apparatus includes a sliding operation detecting section to which a sound signal collected by a microphone is inputted, and which determines start and end of a sliding operation by a determination process using a sliding sound signal component in the inputted sound signal, the sliding sound signal component being produced by the sliding operation on the microphone itself or its vicinity, and a control section that performs a predetermined control process that is set with respect to the sliding operation, during a period from the start to the end of the sliding operation determined by the sliding operation detecting section.

Abstract: A feedback circuit provides a feedback signal to a transducer. The feedback circuit includes an ADC that generates digital representations of a feedback signal, digital controller that identifies adjustments for the feedback, and DAC that generates an analog output of the adjusted feedback signal. The digital controller performs speculative computation to identify adjustments for the feedback signal output for each output value from the ADC prior to receiving the output from the ADC. The ADC and DAC include sigma-delta modulators that operate with a zero clock cycle delay in a forward path. The ADC, digital controller, and DAC generate adjustments to the feedback output signal with reduced delay that reduce phase lag and improve phase margin to maintain stability in the transducer.

Abstract: The invention relates to a method for operating a public address system with a control center to which loudspeakers are connected via a two-wire loop system that is routed via isolator modules. A fault can be located on the loop system if the control center measures the impedance of the loop system successively from isolator module to isolator module and the impedance of the entire loop system, inputs the measured values as target values into an impedance table, periodically measures the impedance of the entire loop system during operation, compares said impedance with the corresponding target value from the impedance table, generates a fault message upon determining a deviation, ascertains the fault location by comparing the measured value with the individual target values in the impedance table, and displays said fault location.

Abstract: A periodogram-based wireless microphone signal detection method, which solves the problem of incapability of distinguishing wireless microphone signal from narrow-band interference, includes: acquiring a time-domain digital signal for detection through antenna module, RF front-end module, ADC module and time-domain signal preprocessing module; and performing detection of microphone signal in frequency domain, that is, calculating an average value of periodogram of M segments of the time-domain digital signal for detection, shifting the average value to obtain a shifted vector, estimating a mean and a covariance matrix of the shifted vector, locating a maximum point of the vector, and taking several points on each side of the point as a center to form an information vector; calculating a decision statistical quantity of the information vector using the decision theory, simulating or calculating a threshold in a predefined method, and deciding the signal is narrow-band interference if the decision statistical q

Abstract: Complex recording triggers are provided. As an example of a system and method, the volume of call records can be reduced by complex recording triggers, thereby reducing the storage space allocated for call records.