Patent classifications
H04R3/005
DIGITAL STETHOSCOPE
A digital stethoscope includes a stethoscope housing defining a housing edge. The digital stethoscope also includes a surface region secured to the stethoscope housing at the housing edge, and a number of microphones. The digital stethoscope also includes a processing device disposed within the stethoscope housing and in communication with the microphones. The processing device receives the digital audio data from the microphones.
Direction-dependent single-source forward cancellation
Active noise cancellation systems, components, and methods are provided with single-source forward cancellation using a direction-dependent filter response. One illustrative active sound cancelling device includes: a primary external microphone that produces a primary receive signal; a secondary external microphone that produces a secondary receive signal, the primary and secondary receive signals representing ambient audio that potentially includes sound having a predominate direction of arrival; a speaker that converts an output signal into internal audio to at least partly cancel said sound, the output signal including a forward cancellation signal; a forward filter that operates solely on the primary receive signal to produce the forward cancellation signal; and a direction finder that operates on the primary and secondary receive signals to derive an estimate of said predominate direction of arrival, the direction finder adjusting the forward filter to implement a filter response corresponding to said estimate.
Dynamic Player Selection for Audio Signal Processing
In one aspect, a first playback device is configured to (i) receive a set of voice signals, (ii) process the set of voice signals using a first set of audio processing algorithms, (iii) identify, from the set of voice signals, at least two voice signals that are to be further processed, (iv) determine that the first playback device does not have a threshold amount of computational power available, (v) receive an indication of an available amount of computational power of a second playback device, (vi) send the at least two voice signals to the second playback device, (vii) cause the second playback device to process the at least two voice signals using a second set of audio processing algorithms, (viii) receive, from the second playback device, the processed at least two voice signals, and (ix) combine the processed at least two voice signals into a combined voice signal.
Microphone Array Beamforming Control
Systems, apparatuses, and methods are described for controlling source tracking and delaying beamforming in a microphone array system. A source tracker may continuously determine a direction of an audio source. A source tracker controller may pause the source tracking of the source tracker if a user may continue to speak to the system. The source tracker controller may resume the source tracking of the source tracker if the user may cease to speak to the system, or when one or more pause durations have been reached.
OPTIMIZATION OF NETWORK MICROPHONE DEVICES USING NOISE CLASSIFICATION
Systems and methods for optimizing network microphone devices using noise classification are disclosed herein. In one example, individual microphones of a network microphone device (NMD) detect sound. The sound data is analyzed to detect a trigger event such as a wake word. Metadata associated with the sound data is captured in a lookback buffer of the NMD. After detecting the trigger event, the metadata is analyzed to classify noise in the sound data. Based on the classified noise, at least one performance parameter of the NMD is modified.
NETWORKED MICROPHONE DEVICES, SYSTEMS, & METHODS OF LOCALIZED ARBITRATION
A first playback device is configured to perform functions comprising: detecting sound, identifying a wake word based on the sound as detected by the first device, receiving an indication that a second playback device has also detected the sound and identified the wake word based on the sound as detected by the second device, after receiving the indication, evaluating which of the first and second devices is to extract sound data representing the sound and thereby determining that the extraction of the sound data is to be performed by the second device over the first device, in response to the determining, foregoing extraction of the sound data, receiving VAS response data that is indicative of a given VAS response corresponding to a given voice input identified in the sound data extracted by the second device, and based on the VAS response data, output the given VAS response.
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.
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.