A loudspeaker cabinet has a number of pairs of microphones, each pair includes the same internal microphone and a different external microphone. For each pair of microphones, a process (i) receives a first audio signal of sound captured by the internal microphone and a second audio signal of sound captured by the different external microphone, (ii) estimates, using first and second audio signals, a radiation impedance, and (iii) computes a detection value based on the radiation impedance in a frequency band. A difference between (i) a currently computed detection value associated with a given pair of microphones and (ii) a previously computed detection value associated with said given pair, is computed. The sound produced by the cabinet is adjusted, in response to the computed difference meeting a threshold. Other embodiments are also described and claimed.

A distributed voice controlled system has a primary assistant and at least one secondary assistant. The primary assistant has a housing to hold one or more microphones, one or more speakers, and various computing components. The secondary assistant is similar in structure, but is void of speakers. The voice controlled assistants perform transactions and other functions primarily based on verbal interactions with a user. The assistants within the system are coordinated and synchronized to perform acoustic echo cancellation, selection of a best audio input from among the assistants, and distributed processing.

A signal from a sensor may be received indicative of an acceleration of a moving component of a transducer at a location where the sensor is mounted. A position the moving component may be determined based on the acceleration. The position of the moving component may be compared with a reference to output a measure of distortion associated with the transducer. Nonlinearities in audio output by the transducer may be corrected based on the measure of distortion.

Example techniques may involve performing aspects of a spatial calibration. An example implementation may include detecting a trigger condition that initiates calibration of a media playback system including multiple audio drivers that form multiple sound axes, each sound axis corresponding to a respective channel of multi-channel audio content The implementation may also include causing the multiple audio drivers to emit calibration audio that is divided into constituent frames, the multiple sound axes emitting calibration audio during respective slots of each constituent frame. The implementation may further include recording the emitted calibration audio. The implementation may include causing delays for each sound axis of the multiple sound axes to be determined, the determined delay for each sound axis based on the slots of recorded calibration audio corresponding to the sound axes and causing the multiple sound axes to be calibrated.

Systems and methods for holistically modelling audio feedback and removing the entire feedback signal corresponding thereto. The systems can operate at a much larger loop-gain (and hence with a much higher loudspeaker volume), than those conventional systems which seek to remove singing frequencies with PEQs. The systems are an improvement over traditional audio feedback elimination systems which attempt to reduce the effect of the audio feedback by simply scaling down the audio volume of the signal frequencies that are prone to howling, and those feedback elimination systems which simply employ adaptive notch filtering to detect and notch the so-called singing or howling frequencies as they occur in real-time. Such devices may typically have several knobs and buttons needing tuning, for example: the number of adaptive parametric equalizers (PEQs) versus fixed PEQs; attack and decay timers; and/or PEQ bandwidth. The systems set forth herein obviate the need for tuning knobs or buttons, making set up easy.

An audio system is provided that efficiently detects speaker arrays and configures the speaker arrays to output sound. In this system, a computing device may record the addresses and/or types of speaker arrays on a shared network while a camera captures video of a listening area, including the speaker arrays. The captured video may be analyzed to determine the location of the speaker arrays, one or more users, and/or the audio source in the listening area. While capturing the video, the speaker arrays may be driven to sequentially emit a series of test sounds into the listening area and a user may be prompted to select which speaker arrays in the captured video emitted each of the test sounds. Based on these inputs from the user, the computing device may determine an association between the speaker arrays on the shared network and the speaker arrays in the captured video.

A first signal may be received indicative of audio to be played by a speaker. A second signal may be received which comprises (i) a voice input received by a microphone and (ii) at least a portion of the audio played by the speaker at a same time that the microphone receives the voice input. Based on the first signal, nonlinearities output by the speaker which played the audio may be determined. At least the nonlinearities from the second signal may be removed to output a third signal comprising substantially the voice input received at the microphone.

Examples of the disclosure relate to enabling external control of noise control processing. In examples of the disclosure a location of a listener is determined. The listener is listening to audio content and the audio content is processed using noise control processing. An acoustic response at the location of the listener is determined and a control signal is provided. The control signal indicates an adjustment for one or more parameters of the noise control processing, and the adjustment is based, at least in part, on the acoustic response at the location of the listener.

The invention relates to a hearing aid a cochlear implant comprising a) at least one input transducer for capturing incoming sound and for generating electric audio signals which represent frequency bands of the incoming sound, b) a sound processor which is configured to analyze and to process the electric audio signals, c) a transmitter that sends the processed electric audio signals, d) a receiver/stimulator, which receives the processed electric audio signals from the transmitter and converts the processed electric audio signals into electric pulses, e) an electrode array embedded in the cochlear comprising a number of electrodes for stimulating the cochlear nerve with said electric pulses, and f) a control unit configured to control the distribution of said electric pulses to the number of said electrodes. The control unit is configured to distribute said electric pulses to the number of said electrodes by applying one out of a plurality of different coding schemes, and wherein the applied coding scheme is selected according to characteristics of the incoming sound.

A distributed voice controlled system has a primary assistant and at least one secondary assistant. The primary assistant has a housing to hold one or more microphones, one or more speakers, and various computing components. The secondary assistant is similar in structure, but is void of speakers. The voice controlled assistants perform transactions and other functions primarily based on verbal interactions with a user. The assistants within the system are coordinated and synchronized to perform acoustic echo cancellation, selection of a best audio input from among the assistants, and distributed processing.