Disclosed embodiments provide glitch prediction based on machine learning algorithms in mixed analog and digital systems, particularly directed to digital microelectromechanical (MEMS) multipath acoustic sensors or microphones, which allow seamless, low latency gain changes without audible artifacts or interruptions in the audio output signal.

An example method of operation may include detecting an acoustic stimulus via active beams associated with at least one microphone disposed in a defined space, detecting loudspeaker characteristic information of at least one loudspeaker providing the acoustic stimulus, transmitting acoustic stimulus information based on the acoustic stimulus to a central controller, and modifying, via a central controller, at least one control function associated with the at least one microphone and the at least one loudspeaker to minimize acoustic feedback produced by the loudspeaker.

An integrated cinema amplifier comprises a power supply stage that distributes power over a plurality of channels for rendering immersive audio content in a surround sound listening environment. The amplifier automatically detects maximum and net power availability and requirements based on audio content by decoding audio metadata and dynamically adjusts gains to each channel or sets of channels based on content and operational/environmental conditions. A power supply stage provides power to drive a plurality of channels corresponding to speaker feeds to a plurality of speakers. The amplifier has a front panel having an LED array with each LED associated with a respective channel or group of channels of the multi-channel amplifier, and a control unit configured to light the LEDs according to display patterns based on operating status or error conditions of the amplifier.

A method for speech level adaptation, the method includes: (A) Receiving, by a voice interactive intelligent personal assistant, multiple input audio signals that includes first and second groups of input audio signals. (B) Buffering the multiple input audio signals. (C) Searching for a voice trigger in the first group of input audio signals. When finding the voice trigger then (D) Determining a linear gain factor to be applied on the second group of buffered audio signals, (E) Applying the linear gain factor on the second group of buffered audio signals to provide the output audio signals; and (F) applying a speech recognition process on the output audio signals to detect an audio command embedded in the output audio signals.

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.

Exemplary multipath digital microphones described herein can comprise exemplary embodiments of automatic gain control and multipath digital audio signal digital signal processing chains, which allow low power and die size to be achieved as described herein, while still providing a high DR digital microphone systems. Further non-limiting embodiments can facilitate switching between multipath digital audio signal digital signal processing chains while minimizing audible artifacts associated with either the change in the gain automatic gain control amplifiers switching between multipath digital audio signal digital signal processing chains.

A networked speaker system automatically adjusts certain audio speaker system settings based on whether the system is inside a building or has been moved outside as may be sensed by one or capacitors on a speaker. Also, techniques are described for adjusting speaker driver direction based on walls or other barriers in a room or based on the location of a listener.

Proposed is an acoustic output device that obtains enough distance attenuation to achieve localization of a sound field by controlling the driving of a loudspeaker array by use of a more compact shape. The acoustic output device 10 comprises: a loudspeaker array 20 that includes a plurality of loudspeakers 20-1, 20-2, . . . , 20-N arranged in a two-dimensional plane; and an amplifier array 40 that includes a plurality of amplifiers 40-1, 40-2, . . . , 40-N that control the amplitude and phase of the drive signals for each loudspeaker according to the eigenvectors of the predetermined radiation mode of the loudspeaker array 20.

Embodiments of the invention provide for methods for automatically controlling loudness of an audio signal. Embodiments of the invention also provide for a device for automatically controlling loudness of an audio signal. In particular, the invention relates to controlling loudness of an audio signal to be replayed within a vehicle, such as a car.

Example embodiments provide a process that includes one or more of receiving an audio signal at a feedback compressor circuit, receiving an auxiliary attenuation signal from an auxiliary attenuation source, determining a threshold power level based on a value of the auxiliary attenuation signal, determining an output power level of the audio signal exceeds the threshold power level, combining the audio signal with the auxiliary attenuation signal from the auxiliary attenuation source and a compressed attenuation signal from the feedback compressor circuit to create a combination signal, and generating an audio output signal of the feedback compressor circuit based on the combination signal.