Abstract: There is provided a clip-on microphone assembly comprising a clip-on mechanism (16) for attaching the microphone assembly (10) to the cloths of a user (11); at least three microphones (20, 21, 22) for capturing audio signals from the user's voice, the microphones defining a microphone plane; an acceleration sensor (32) for generating an orientation signal by sensing gravitational acceleration in at least two orthogonal dimensions, an audio signal processing unit (34) for producing an output audio signal (36) from the captured audio signals, comprising a beamformer unit (24) for processing the captured audio signals, in order to generate the output audio signal, in a manner so as to create an acoustic beam having a direction, a DOA unit (30) for determining the direction of arrival of sound by analyzing the captured audio signals; and a control unit (26) for controlling the beamforming unit, the control unit being adapted to determine an allowed angular sector (40) of the direction of the acoustic beam accordi

Abstract: The present invention provides a hearing protection earphone, a hearing protection method and a computer program storage medium, comprising a sound control system including a wearing-status sensing module for monitoring a current wearing status of the hearing protection earphone, sending a first monitoring signal when it is determined that the hearing protection earphone is in a wearing state, and sending a second monitoring signal when it is determined that the hearing protection earphone is in a non-wearing state; and a main control module for enabling a play function corresponding to a play mode which is adapted to the hearing protection earphone currently when the first monitoring signal sent from the wearing-status sensing module is received, and counting a stand-by time of the protection earphone and turning off the hearing protection earphone after a preset stand-by time elapses when the second monitoring signal sent from the wearing-status sensing module is received.

Abstract: The stethoscope with sound recognition capacity is an electric stethoscope. The stethoscope with sound recognition capacity comprises a stethoscope and a sound processing device. The stethoscope is a traditional medical device used to listen for audible sounds inside a patient. The stethoscope of the stethoscope with sound recognition capacity performs this traditional function. The sound processing device is an electrical circuit housed within the stethoscope. The sound processing device: a) converts the audible sounds captured by the stethoscope into an electrical signal; b) processes the electrical signal to extract diagnostic information; c) processes the extracted diagnostic information to identify one or more conditions of diagnostic relevance found in the patient; and, d) makes an audible spoken announcement of the one or more conditions of diagnostic relevance.

Abstract: A device and method capable of performing image following type audio control or image non-following type audio control are implemented. Images in different directions are selectively displayed on the display unit, and an output audio is controlled in accordance with an image display. A data processing unit executes image following type audio control of moving an audio source direction in accordance with movement of the display image of the display unit and image non-following type audio control of not moving the audio source direction in accordance with the movement of an image in units of individual controllable audio elements. The data processing unit acquires audio control information from an MP4 file or a media presentation description (MPD) file and executes either the image following type audio control or the image non-following type audio control in accordance with the acquired audio control information in units of individual controllable audio elements.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: A computer-implemented method for reducing undesired crosstalk signals on an inactive channel of a device comprising the steps of: (i) determining system volume levels required to achieve a range of desired audio output attenuation levels on an active channel of the device; (ii) determining a crosstalk compensation signal comprising a signal amplitude and associated phase shift required to reduce undesired crosstalk on the inactive channel of the device for each desired audio output attention level in the range of desired audio output attenuation levels; and (iii) generating a desired audio output attenuation level on the active channel of the device by generating a signal at the determined system volume level required to achieve said desired audio output attenuation level, and generating a contemporaneous crosstalk compensation signal on the inactive channel of the device by generating a signal at the determined signal amplitude and associated phase shift required to reduce the undesired crosstalk on the ina

Abstract: An electronic device is provided. The electronic device includes a first speaker, a first microphone, a storage unit storing a reference signal, and a controller configured to, when the reference signal is output through the first speaker, receive a first signal input through the first microphone and a second signal input through a second microphone of an external electronic device and determine whether to use the first microphone or the second microphone based on at least one of the first signal input through the first microphone and the second signal input through the second microphone.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: Audio processing architectures are described for implementing a multi-band audio compression algorithm with advanced surround processing. Exemplary architectures can improve the fidelity and perceived sound field spread of inexpensive, cabinet mounted, stereo speakers such as those that might be found in televisions, wireless speaker systems and sound bars. Embodiments of the present disclosure can improve inexpensive, cabinet mounted, stereo speakers by providing, e.g., (i) an Advanced Surround algorithm that adds depth and height to the left/right/center sound field images, (ii) a Soft Clip algorithm to minimize the perceived artifacts caused by compressor overshoot, (iii) configurable crossover filter order adjustment to allow better isolation between bands, (iv) a compressor maximum gain adjustment to reduce overshoot and minimize noise boost, and/or (v) a center gain adjustment to emphasize the perception of the center image (dialog) in high ambient noise situations.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: An analysis system has at least one monitoring assembly that includes at least one microphone to monitor an acoustic field proximate the monitoring assembly. The at least one microphone produces at least one microphone signal responsive to the acoustic field. The analysis system further includes a data storage device configured to buffer the at least one microphone signal and an audio analysis system configured to analyze a content of the data storage device where the audio analysis system is configured to analyze the content of the buffer to process at least a sound into a response or action.

Abstract: Input audio data portions of a common time window index value generated by multiple microphones at a location are received. Subband portions are generated from the input audio data portions. Peak powers, noise floors, etc., are individually determined for the subband portions. Weights for the subband portions are computed based on the peak powers, the noise floors, etc., for the subband portions. An integrated audio data portion of the common time window index is generated based on the subband portions and the weight values for the subband portions. An integrated signal may be generated based at least in part on the integrated audio data portion.

Abstract: The technology described in this document can be embodied in a method that includes receiving an input signal captured by one or more sensors associated with an active noise reduction (ANR) headphone, and determining one or more characteristics of a first portion of the input signal. Based on the one or more characteristics of the first portion of the input signal, a gain of a variable gain amplifier (VGA) disposed in an ANR signal flow path can be adjusted, and accordingly, a set of coefficients for a tunable digital filter disposed in the ANR signal flow path can be selected. The method further includes processing a second portion of the input signal in the ANR signal flow path using the adjusted gain and selected set of coefficients to generate a second output signal for the electroacoustic transducer of the ANR headphone.

Abstract: In a speaker sound cavity structure of a mobile terminal of the present disclosure, the speaker sound cavity structure includes an antenna supporting member arranged on a mobile terminal. The antenna supporting member includes a sound-guiding groove configured as a transmission channel for transmitting sound in the speaker sound cavity structure of the mobile terminal. A lead-out hole is configured to lead the sound in a speaker sound cavity into a new speaker sound cavity of the mobile terminal. A lead-in hole is configured to guide the sound in the speaker sound cavity of into the new speaker sound cavity of the mobile terminal. The speaker sound cavity structure is configured to guide the sound of the speaker sound cavity to the new speaker sound cavity of the mobile terminal, so that the space in the speaker sound cavity of the mobile terminal is expanded.

Abstract: A transducer module, comprising: a supporting substrate, having a first side and a second side; a cap, which extends over the first side of the supporting substrate and defines therewith a first chamber and a second chamber internally isolated from one another; a first transducer in the first chamber; a second transducer in the second chamber; and a control chip, which extends at least partially in the first chamber and/or in the second chamber and is functionally coupled to the first and second transducers for receiving, in use, the signals transduced by the first and second transducers.

Abstract: Phase effect interference is determined at a listening location between signals from at least two audio objects; and a modified position is computed for at least one of the audio objects such that the determined phase effect interference at the listening location is altered as a result of the modified position. For each audio object for which the position is modified at least phase of at least one frequency component of the respective signal is adjusted in correspondence with the modified position so as to eliminate the determined phase effect interference. The signals from the at least two audio objects are formed after this adjusting; and then the formed signals are provided to a sound system comprising multiple audio transducers so as to render the formed signals at the listening position where the phase effect interference is eliminated during rendering.

Abstract: A method and system of managing audio signals within a communication system is provided. An audio signal is received on two or more independent audio paths within an operating environment, such as a vehicular operating environment (106). A controller identifies an audio source ID associated with the audio signal. The audio source ID may be a talkgroup ID, an individual ID, or a channel ID associated with the audio signal. The received audio signal from the independent audio paths is mixed and played out at a spatially separated unique location within the vehicle from other audio signals based on the sound source ID, each sound source ID having a unique location within the vehicle.

Abstract: An embodiment as described herein includes a microelectromechanical system (MEMS) with a first MEMS transducer element, a second MEMS transducer element, and a semiconductor substrate. The first and second MEMS transducer elements are disposed at a top surface of the semiconductor substrate and the semiconductor substrate includes a shared cavity acoustically coupled to the first and second MEMS transducer elements.

Abstract: Personal audio systems and methods are disclosed. A personal audio system includes a class table storing processing parameters respectively associated with a plurality of annoyance noise classes, a controller, and a processor. The controller identifies an annoyance noise class of an annoyance noise included in an ambient audio stream and retrieves, from the class table, one or more processing parameters associated with the identified annoyance noise class. The processor to processes the ambient audio stream according to the one or more retrieved processing parameters class to provide a personal audio stream. The processor includes a pitch tracker to identify a fundamental frequency of the annoyance noise and a filter bank including a band reject filter tuned to the fundamental frequency.