Patent classifications
H04R3/002
Signal processing based on audio context
Described herein are systems, methods, and apparatus for determining audio context between an audio source and an audio sink and selecting signal profiles based at least in part on that audio context. The signal profiles may include noise cancellation which is configured to facilitate operation within the audio context. Audio context may include user-to-user and user-to-device communications.
Sound masking system
A sound masking system according to the invention is disclosed in which one or more sound masking loudspeaker assemblies are coupled to one or more electronic sound masking signal generators. The loudspeaker assemblies in the system of the invention have a low directivity index and preferably emit an acoustic sound masking signal that has a sound masking spectrum specifically designed to provide superior sound masking in an open plan office. Each of the plurality of loudspeaker assemblies is oriented to provide the acoustic sound masking signal in a direct path into the predetermined area in which masking sound is needed. In addition, the sound masking system of the invention can include a remote control function by which a user can select from a plurality of stored sets of information for providing from a recipient loudspeaker assembly an acoustic sound masking signal having a-selected sound masking spectrum.
Acoustic transducer aging compensation with life indicator
Acoustic transducer aging compensation is effective for an acoustic transducer that is driven with an adjustable drive power to output a signal. A microphone can measure the amplitude of the transmitted signal corresponding to a transmitted sound pressure level (SPL). A controller can periodically compare the transmitted SPL to the drive power or a previous SPL, and determine if the received SPL has declined with respect to the input drive power over time, whereupon the controller can direct an increase in drive power to the SPL-declined acoustic transducer to compensate for the decline in received SPL. If drive power is at a maximum, the controller can further instruct a mobile device receiver to lower its receiver detection threshold for the signal from the SPL-declined acoustic transducer to further compensate for the decline in SPL from that acoustic transducer. A life indicator can be provided to inform the system operator of the degraded speaker so as to provide an early warning indicator for servicing of that transducer.
MICROMECHANICAL PIEZOELECTRIC ACTUATORS FOR IMPLEMENTING LARGE FORCES AND DEFLECTIONS
A MEMS includes a diaphragm, a stroke structure coupled to the diaphragm, and at least two piezoelectric actuators coupled to a plurality of mutually spaced-apart contact points of the stroke structure via a plurality of mutually spaced-apart connecting elements, the at least two piezoelectric actuators being configured to cause a stroke movement of the stroke structure so as to deflect the diaphragm.
ESTIMATING VOLTAGE ON SPEAKER TERMINALS DRIVEN BY A CLASS-D AMPLIFIER
A system includes an audio amplifier, a duty cycle detector, a channel equalizer, and a sample-and-hold circuit. The audio amplifier is configured to amplify an analog audio signal to produce an amplified audio signal. The duty cycle detector is configured to generate a saturation detect signal at a first state upon detection that the amplified audio signal produced by the audio amplifier is clipped. The channel equalizer is configured to generate an initial estimate of a speaker terminal voltage. The sample-and-hold circuit is configured to sample and hold the initial estimate of the speaker terminal voltage as a final estimate of the speaker voltage when the saturation detect signal is in the first state.
Apparatus for and method of wind detection
A method, comprising: obtaining one or more accelerometer signals derived from an accelerometer; and determining one or more parameters of wind at the accelerometer based on the one or more accelerometer signals.
Methods, systems, and media for ambient background noise modification based on mood and/or behavior information
Methods, systems, and media for ambient background noise modification are provided. In some implementations, the method comprises: identifying at least one noise present in an environment of a user having a user device, an activity the user is currently engaged in, and a physical or emotional state of the user; determining a target ambient noise to be produced in the environment based at least in part on the identified noise, the activity the user is currently engaged in, and the physical or emotional state of the user; identifying at least one device associated with the user device to be used to produce the target ambient noise; determining sound outputs corresponding to each of the one or more identified devices, wherein a combination of the sound outputs produces an approximation of one or more characteristics of the target ambient noise; and causing the one or more identified devices to produce the determined sound outputs.
METHOD AND DEVICE FOR ACUTE SOUND DETECTION AND REPRODUCTION
Earpieces and methods for acute sound detection and reproduction are provided. A method can include measuring an external ambient sound level (xASL), monitoring a change in the xASL for detecting an acute sound, estimating a proximity of the acute sound, and upon detecting the acute sound and its proximity, reproducing the acute sound within an ear canal, where the ear canal is at least partially occluded by an earpiece. Other embodiments are disclosed.
TOUCHLESS INTERACTION USING AUDIO COMPONENTS
The present teachings relate to an electronic device comprising: a first module for generating an audio signal; a second module for generating an ultrasonic signal; a mixer for generating a combined signal; a transmitter for outputting an acoustic signal dependent upon the combined signal; and, a processing means for controlling the ultrasonic signal; wherein, in response to receiving a first instruction signal for initiating the ultrasonic signal, the processing means is configured to increase the amount of the ultrasonic signal in the combined signal from an essentially zero value to a predetermined value over a predetermined enable time-period. The present teachings also relate to an electronic device configured to decrease the amount of the ultrasonic signal in the combined signal from an essentially zero value to a predetermined value over a predetermined disable time-period, and to an electronic device configured to remove the audio signal from the combined signal whilst preventing pop-noise, and to an electronic device capable of replacing the ultrasonic signal whilst minimizing the processing time. The present teachings further relate to a method for reducing the occurrence of pop noise in an acoustic signal associated with: initiating the ultrasonic signal in the combined signal, terminating the ultrasonic signal in the combined signal, terminating the audio signal in the combined signal, and replacing the ultrasonic signal in the combined signal. The present teachings also relate to a computer software product for implementing any of the method steps disclosed herein, and to a computer storage medium storing the computer software herein disclosed.
FORMING VIRTUAL MICROPHONE ARRAYS USING DUAL OMNIDIRECTIONAL MICROPHONE ARRAY (DOMA)
A dual omnidirectional microphone array noise suppression is described. Compared to conventional arrays and algorithms, which seek to reduce noise by nulling out noise sources, the array of an embodiment is used to form two distinct virtual directional microphones which are configured to have very similar noise responses and very dissimilar speech responses. The only null formed is one used to remove the speech of the user from V.sub.2. The two virtual microphones may be paired with an adaptive filter algorithm and VAD algorithm. to significantly reduce the noise without distorting the speech, significantly improving the SNR of the desired speech over conventional noise suppression systems.