Method and system for enhancing acoustic performances in an adverse acoustic environment, where the system comprises: an array of acoustic sensors having different directivities; and an analysis module being configured for optimizing signal enhancement of at least one source, by correlating the sensors according to respective position of the at least one source in respect to the directivity of the acoustic sensors, based on reflections from reverberating surfaces in the specific acoustic environment, wherein the optimization and sensors directivity allows maintaining the sensor array in compact dimensions without affecting signal enhancement and source separation.

The invention relates to a hearing aid comprising a first microphone configured to receive a first acoustic signal and to convert the first acoustic signal to a first electrical audio signal, a speaker configured to emit an acoustic output signal into an ear of a user of the hearing aid device, a first analog-to-digital converter for converting the first electrical audio signal into a first time-domain input signal, a first input unit comprising a first analysis filter bank which is configured to convert the first time-domain input signal to a number NI,1 of first input frequency bands wherein the number NI,1 of first input frequency bands is determined by said first analysis filter bank, a first frequency band bundling and allocation unit which is configured to bundle adjacent first input frequency bands and to allocate first frequency bands to be processed to a number NP,1 of first processing channels, a memory unit which is configured to store data indicating which of the first NI,1 input frequency bands are subject to a likelihood of feedback that is above a threshold, a signal processing unit is configured to process the first frequency bands to be processed in the number NP,1 of first processing channels, and wherein the number NP,1 of first processing channels is smaller than the number NI,1 of first input frequency bands, and wherein the first frequency band bundling and allocation unit is configured to generate a first bundling and allocation scheme which determines the bundling of the first NI,1 input frequency bands and the allocation of the first frequency bands to be processed to the first NP,1 processing channels wherein said first bundling and allocation scheme depends on the likelihood of feedback to occur in at least one of the first NI,1 input frequency bands.

The present invention is a system that detects the location and head orientation of a live performer on stage in front of an audience and adjusts the individual elements of the mix either by changing them at the listener's position or by feeding the location and orientation information back to the monitor mixer or a combination of both. The adjustments would include left/right panning, relative levels, equalization, transient response, reverberation levels, panning, and time delay, as well as other possible modifications to the signal so that the performer senses that he/she is actually listening to the various instruments, vocals, acoustic space and audience, in their real or virtual locations.

An audio processing method may involve receiving output signals from each microphone of a plurality of microphones in an audio environment, the output signals corresponding to a current utterance of a person and determining, based on the output signals, one or more aspects of context information relating to the person, including an estimated current proximity of the person to one or more microphone locations. The method may involve selecting two or more loudspeaker-equipped audio devices based, at least in part, on the one or more aspects of the context information, determining one or more types of audio processing changes to apply to audio data being rendered to loudspeaker feed signals for the audio devices and causing one or more types of audio processing changes to be applied. In some examples, the audio processing changes have the effect of increasing a speech to echo ratio at one or more microphones.

An example implementation may involve a control device displaying a prompt to initiate a calibration sequence that involves calibration of a playback device for a given environment in which the playback device is located. The example implementation may also involve the control device displaying (i) a prompt to prepare a playback device for calibration within a given environment, (ii) a prompt to prepare the given environment for calibration of the playback device, and/or (iii) a prompt to prepare the control device for calibration of the playback device. The example implementation may also involve a control device displaying a selectable control, that, when selected, initiates calibration of the playback device. The example implementation may further involve a control device initiating calibration of the playback device.

Systems and methods for calibrating a playback device include (i) outputting first audio content; (ii) capturing audio data representing reflections of the first audio content within a room in which the playback device is located; (iii) based on the captured audio data, determining an acoustic response of the room; (iv) connecting to a database comprising a plurality of sets of stored audio calibration settings, each set associated with a respective stored acoustic room response of a plurality of stored acoustic room responses; (v) querying the database for a stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located; and (vi) applying to the playback device a particular set of stored audio calibration settings associated with the stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located.

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 calibrating a playback device include (i) outputting first audio content; (ii) capturing audio data representing reflections of the first audio content within a room in which the playback device is located; (iii) based on the captured audio data, determining an acoustic response of the room; (iv) connecting to a database comprising a plurality of sets of stored audio calibration settings, each set associated with a respective stored acoustic room response of a plurality of stored acoustic room responses; (v) querying the database for a stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located; and (vi) applying to the playback device a particular set of stored audio calibration settings associated with the stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located.

An audio processing method may involve receiving output signals from each microphone of a plurality of microphones in an audio environment, the output signals corresponding to a current utterance of a person and determining, based on the output signals, one or more aspects of context information relating to the person, including an estimated current proximity of the person to one or more microphone locations. The method may involve selecting two or more loudspeaker-equipped audio devices based, at least in part, on the one or more aspects of the context information, determining one or more types of audio processing changes to apply to audio data being rendered to loudspeaker feed signals for the audio devices and causing one or more types of audio processing changes to be applied. In some examples, the audio processing changes have the effect of increasing a speech to echo ratio at one or more microphones.

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.