Patent classifications
H04R1/406
ESTIMATING USER LOCATION IN A SYSTEM INCLUDING SMART AUDIO DEVICES
Methods and systems for performing at least one audio activity (e.g., conducting a phone call or playing music or other audio content) in an environment including by determining an estimated location of a user in the environment in response to sound uttered by the user (e.g., a voice command), and controlling the audio activity in response to determining the estimated user location. The environment may have zones which are indicated by a zone map and estimation of the user location may include estimating in which of the zones the user is located. The audio activity may be performed using microphones and loudspeakers which are implemented in or coupled to smart audio devices.
Method and apparatus for an interactive user interface
A method, apparatus and computer program product are provided to facilitate user interaction with, such as modification of, respective audio objects. An example method may include causing a multimedia file to be presented that includes at least two images. The images are configured to provide animation associated with respective audio objects and representative of a direction of the respective audio objects. The method may also include receiving user input in relation to an animation associated with an audio object or the direction of the audio object represented by an animation. The method may further include causing replay of the audio object for which the user input was received to be modified.
Differential condenser microphone with double vibrating membranes
A dual-diaphragm differential capacitive microphone includes: a back plate, a first diaphragm, and a second diaphragm. The first diaphragm is insulatively supported on a first surface of the back plate, where the back plate and the first diaphragm form a first variable capacitor. The second diaphragm is insulatively supported on a second surface of the back plate, where the back plate and the second diaphragm form a second variable capacitor. The back plate is provided with at least one connecting hole. The second diaphragm is provided with a recess portion recessed towards the back plate, where the recess portion passes through the connecting hole and is connected to the first diaphragm. The dual-diaphragm differential capacitive microphone achieves a higher signal-to-noise ratio.
Wearable respiratory monitoring system based on resonant microphone array
A method for continuous acoustic signature recognition and classification includes a step of obtaining an audio input signal from a resonant microphone array positioned proximate to a target, the audio input signal having a plurality of channels. The target produces characterizing audio signals depending on a state or condition of the target. A plurality of features is extracted from the audio input signal with a signal processor. The plurality of features is classified to determine the state of the target. An acoustic monitoring system implementing the method is also provided.
Hearing device or system for evaluating and selecting an external audio source
A hearing system comprises a hearing device worn on the head, or fully or partially implanted in the head, of a user, and external audio transmitters. The hearing system allows wireless communication to be established between the hearing device and the audio transmitters. The hearing device comprises microphones providing respective electric input signals; a beamformer filter providing a beamformed signal from the electric input signals; and an output unit. The hearing system further comprises a selector/mixer for selecting and possibly mixing one or more of the electric input signals or the beamformed signal and external electric signals from the audio transmitters, and providing a current input sound signal based thereon for presentation to the user. The selector/mixer is controlled by a source selection processor, which determines the source selection control signal based on a comparison of the beamformed signal and the external electric sound signals or processed versions thereof.
AUTOMATIC LOUDSPEAKER ROOM EQUALIZATION BASED ON SOUND FIELD ESTIMATION WITH ARTIFICIAL INTELLIGENCE MODELS
One embodiment provides a computer-implemented method that includes acquiring, via at least one microphone, sound pressure data from a loudspeaker in a room. The sound pressure data is input into an artificial intelligence (AI) model. The AI model automatically estimates, without user interaction, at least one of energy average (EA) in a listening area or total sound power (TSP) produced by the loudspeaker. The AI model is trained prior to automatically estimating the at least one of the EA in the listening area or the TSP produced by the loudspeaker.
System and method for a voice-controllable apparatus
In accordance with an embodiment, an apparatus includes a millimeter wave radar sensor system configured to detect a location of a body of a person, where the detected location of the body of the person defines a direction of the person relative to the apparatus; and a microphone system configured to generate at least one audio beam as a function at least of the direction.
ELECTRONIC DEVICE FOR CONTROLLING BEAMFORMING AND OPERATING METHOD THEREOF
An electronic device is provided. The electronic device includes, for the purpose of determining a customized beamformer filter, an input module including a plurality of microphones configured to receive an external sound signal, a memory configured to store computer-executable instructions and an initial value of a voice parameter used to perform beamforming on the external sound signal, and a processor configured to execute the instructions by accessing the memory. The instructions may be configured to estimate a feature value of the external sound signal, calculate the initial value of the voice parameter used to perform beamforming based on the external sound signal received by the plurality of microphones, determine whether to store the calculated initial value according to the feature value, determine which one of the calculated initial value or an initial value stored in the memory used according to the feature value, and obtain a target voice parameter.
VEHICLE AUDIO ENHANCEMENT SYSTEM
The invention is an apparatus used to send sound exterior to the vehicle to a location on the interior of a vehicle which corresponds with the sound’s external location. The apparatus adjusts for gain by frequency and shifts frequencies of the audio signals received exterior to the vehicle to generate a modified audio signal. The adjustments include compensation for hearing loss, compensation for audio in the interior of the vehicle, and attenuation adjustments for unwanted exterior noise. The modified audio signal is generated by filtering unwanted audio from the exterior audio, adjusting for gain by frequency and shifting/compressing frequencies according to hearing deficiencies, and adjusting for gain by frequency according to audio in the interior of the vehicle. The modified audio signal may be limited to a maximum gain and then amplified in the interior of the vehicle.
Method and device for acute sound detection and reproduction
An electronic device or method for adjusting a gain on a voice operated control system can include one or more processors and a memory having computer instructions. The instructions, when executed by the one or more processors causes the one or more processors to perform the operations of receiving a first microphone signal, receiving a second microphone signal, updating a slow time weighted ratio of the filtered first and second signals, and updating a fast time weighted ratio of the filtered first and second signals. The one or more processors can further perform the operations of calculating an absolute difference between the fast time weighted ratio and the slow time weighted ratio, comparing the absolute difference with a threshold, and increasing the gain when the absolute difference is greater than the threshold. Other embodiments are disclosed.