An impulse response is computed between i) an audio signal that is being output as sound by a loudspeaker that is integrated in a loudspeaker enclosure, and ii) a microphone signal from a microphone that is recording the output by the loudspeaker and that is also integrated in the loudspeaker enclosure. A reverberation spectrum is extracted from the impulse response. Sound power spectrum at the listening distance is computed, based on the reverberation spectrum, and an equalization filter is determined based on i) the estimated sound power spectrum and ii) a desired frequency response at the listening distance. Other aspects are also described and claimed.

An exemplary microphone capture simulation system accesses a captured set of audio signals captured by a plurality of directional microphones disposed at a plurality of locations on a perimeter of a capture zone of a real-world scene. The system identifies a location within the capture zone that corresponds to a virtual location at which a user is virtually located within a virtual reality space that is based on the capture zone. Based on the captured set of audio signals and the identified location, the system generates a simulated set of audio signals representative of a simulation of a full-sphere multi-capsule microphone capture at the identified location. The system processes the simulated set of audio signals to form a renderable set of audio signals configured to be rendered to simulate full-sphere sound for the virtual location while the user is virtually located at the virtual location.

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.

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.

Techniques for simulating a microphone array and generating synthetic audio data to analyze the microphone array geometry. This reduces the development cost of new microphone arrays by enabling an evaluation of performance metrics (False Rejection Rate (FRR), Word Error Rate (WER), etc.) without building device hardware or collecting data. To generate the synthetic audio data, the system performs acoustic modeling to determine a room impulse response associated with a prototype device (e.g., potential microphone array) in a room. The acoustic modeling is based on two parametersa device response (information about acoustics and geometry of the prototype device) and a room response (information about acoustics and geometry of the room). The device response can be simulated based on the microphone array geometry, and the room response can be determined using a specialized microphone and a plane wave decomposition algorithm.

An example playback device is configured to receive a first stream of audio comprising source audio content to be played back by the playback device and record, via one or more microphones of the playback device, an audio signal output by the playback device based on the playback device playing the source audio content. The playback device is also configured to determine a transfer function between a frequency-domain representation of the first stream of audio and a frequency-domain representation of the recorded audio signal, and then determine an estimated frequency response of the playback device based on a difference between (i) the transfer function and (ii) a self-response of the playback device, where the self-response of the playback device is stored in a memory of the playback device. Based on the estimated frequency response, the playback device is configured to determine an acoustic calibration adjustment and implement the acoustic calibration adjustment.

Presented is an acoustic low frequency horn for corner placement, radiating backward into the corner cube, from where the waves are reflected to the room, guided by 3 plains out of the corner ideally forming the biggest possible conical horn. Providing a smooth expansion of the wave path avoids the formation of a horn mouth and the problems therefrom. In consequence it has a high efficiency and needs only a smaller speaker that fits into an enclosure of about one magnitude smaller size than comparable corner horns. Thereby retaining the excellent reproduction characteristics of an over sized bass horn, but with reduced enclosure size, material and overall cost. It comes without cutoff frequency effect and can play down to the limit of human audibility without changing its characteristics at lowest frequencies.

According to an embodiment, a noise cancellation method comprises updating a first far-end channel filter according to a first variance yielded based on an input signal and a first far-end signal included in the input signal, providing a first output signal obtained by removing a first far-end noise signal from the input signal through the first far-end channel filter, updating a second far-end channel filter according to a second variance yielded based on the first output signal and a second far-end signal included in the input signal, and providing a second output signal obtained by removing a second far-end noise signal from the first output signal through the second far-end channel filter.

A hearing-assist system for use in a venue in which the system includes circuitry inserted in a signal path between a program source feed of sound and hearing-assist units of users in that venue which improves the quality of sound heard by the users via the hearing-assist units.

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.