Abstract: An unknown content audio signal that is input to an audio power amplifier and speaker that have a known gain and speaker sensitivity, is buffered. A portion of the buffered signal is provided to an automatic content recognition (ACR) system, and then a content identifier, of the unknown content signal, as provided by the ACR system is received. Previously determined metadata is also received, for a reference audio content signal that has been assigned to the content identifier. The metadata includes a previously determined reference measure of an audio signal characteristic for the reference content. An input measure of the audio signal characteristic is computed, for the unknown content, and compared with the reference measure. On that basis, and also based on the known gain and speaker sensitivity, at least one of a scalar gain and spectral shaping filtering that is being applied to the unknown content is adjusted. Other embodiments are also described and claimed.

Abstract: An audio system is described that includes one or more speaker arrays that emit sound corresponding to one or more pieces of sound program content into associated zones within a listening area. Using parameters of the audio system (e.g., locations of the speaker arrays and the audio sources), the zones, the users, the pieces of sound program content, and the listening area, one or more beam pattern attributes may be generated. The beam pattern attributes define a set of beams that are used to generate audio beams for channels of sound program content to be played in each zone. The beam pattern attributes may be updated as changes are detected within the listening environment. By adapting to these changing conditions, the audio system is capable of reproducing sound that accurately represents each piece of sound program content in various zones.

Abstract: 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.

Abstract: An audio capture device generates two microphone beam patterns with different directivity indices. The audio capture device may determine the position of a user relative to the audio capture device based on sounds detected by the separate microphone beam patterns. Accordingly, the audio capture device allows the determination of the position of the user without the complexity and cost of using a dedicated listening device and/or a camera. In particular, the audio capture device does not need to be immediately proximate to the user (e.g., held near the ear of the user) and may be used to immediately provide other services to the user (e.g., audio/video playback, telephony functions, etc.). The position of the user may include the measured distance between the audio capture device and the user, the proximity of the user relative to another device/object, and/or the orientation of the user relative to the audio capture device.

Abstract: An audio emission device and an audio capture device that may respectively emit and capture sound within a listening area is described. The audio emission device may produce one or more primary audio beams in the listening area. Each of the primary audio beams may be formed by weighting a set of modal beam patterns. Separate orthogonal test signals may be injected into each modal beam pattern. Based on these separate orthogonal test signals, the individual modal beam patterns may be extracted from a detected sound signal, produced by the audio capture device, such that the contribution from each of these modal patterns in the detected sound signal may be determined. Utilizing the contributions from each modal beam pattern in the detected sound signal, the spatial relationship (e.g., distance and/or orientation/angle) between the audio emission device and the audio capture device may be determined.

Abstract: 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.

Abstract: A system and method for measuring the performance of a plurality of transducers integrated in one or more loudspeakers is described. The method simultaneously drives each transducer to emit sounds corresponding to distinct orthogonal test signals. A listening device senses sounds produced by the orthogonal test signals and analyzes the sensed audio signal to determine the performance of each transducer. By using orthogonal test signals, the multiple transducers may be measured and/or characterized simultaneously and with limited affect from extraneous noises.

Abstract: An unknown content audio signal that is input to an audio power amplifier and speaker that have a known gain and speaker sensitivity, is buffered. A portion of the buffered signal is provided to an automatic content recognition (ACR) system, and then a content identifier, of the unknown content signal, as provided by the ACR system is received. Previously determined metadata is also received, for a reference audio content signal that has been assigned to the content identifier. The metadata includes a previously determined reference measure of an audio signal characteristic for the reference content. An input measure of the audio signal characteristic is computed, for the unknown content, and compared with the reference measure. On that basis, and also based on the known gain and speaker sensitivity, at least one of a scalar gain and spectral shaping filtering that is being applied to the unknown content is adjusted. Other embodiments are also described and claimed.

Abstract: A video conferencing system is described that includes a near-end and a far-end system. The near-end system records both audio and video of one or more users proximate to the near-end system. This recorded audio and video is transmitted to the far-end system through the data connection. The video stream and/or one or more settings of the recording camera are analyzed to determine the amount of a video frame occupied by the recorded user(s). The video conferencing system may directly analyze the video frames themselves and/or a zoom setting of the recording camera to determine a ratio or percentage of the video frame occupied by the recorded user(s). By analyzing video frames associated with an audio stream, the video conferencing system may drive a speaker array of the far-end system to more accurately reproduce sound content based on the position of the recorded user in a video frame.

Abstract: A device displays a user interface that includes a content area and a deletion control. The device detects an input that includes a contact on the deletion control. In response, the device deletes content in the content area based on a duration and intensity of the contact, including: when the contact was maintained for a first time period without the intensity increasing above a threshold, deleting a first-type of sub-units of the content at a rate that does not vary based on the intensity; when the contact was maintained for a second time period without the intensity increasing above the threshold, deleting a second-type of sub-units of the content at a rate that does not vary based on the intensity; and when the intensity of the contact increased above the threshold, deleting sub-units of the content at a rate that varies based on the characteristic intensity of the contact.

Abstract: An audio system that adjusts one or more beam patterns emitted by one or more loudspeaker arrays based on the preferences of users/listeners is described. The audio system includes an audio receiver that contains a listener location estimator, a listener identifier, and a voice command processor. Inputs from the listener location estimator, the listener identifier, and the voice command processor are fed into an array processor. The array processor drives the one or more loudspeaker arrays to emit beam patterns into the listening area based on inputs from each of these devices. By examining the location, preferred usage settings, and voice commands from listeners, the generated beam patterns are customized to the explicit and implicit preferences of the listeners with minimal direct input. Other embodiments are also described.

Abstract: An audio capture device generates two microphone beam patterns with different directivity indices. The audio capture device may determine the position of a user relative to the audio capture device based on sounds detected by the separate microphone beam patterns. Accordingly, the audio capture device allows the determination of the position of the user without the complexity and cost of using a dedicated listening device and/or a camera. In particular, the audio capture device does not need to be immediately proximate to the user (e.g., held near the ear of the user) and may be used to immediately provide other services to the user (e.g., audio/video playback, telephony functions, etc.). The position of the user may include the measured distance between the audio capture device and the user, the proximity of the user relative to another device/object, and/or the orientation of the user relative to the audio capture device.

Abstract: A method for determining the orientation of a loudspeaker relative to a listening device is described. The method simultaneously drives each transducer to emit beam patterns corresponding to distinct orthogonal audio signals. The listening device senses sounds produced by the orthogonal audio signals and analyzes the sensed audio signal to determine the spatial orientation of the loudspeaker relative to the listening device. Other embodiments are also described.

Abstract: A directivity adjustment device that maintains a constant direct-to-reverberant ratio based on the detected location of a listener in relation to the speaker array is described. The directivity adjustment device may include a distance estimator, a directivity compensator, and an array processor. The distance estimator detects the distance between the speaker array and the listener. Based on this detected distance, the directivity compensator calculates a directivity index form a beam produced by the speaker array that maintains a predefined direct-to-reverberant sound energy ratio. The array processor receives the calculated directivity index and processes each channel of a piece of sound program content to produce a set of audio signals that drive one or more of the transducers in the speaker array to generate a beam pattern with the calculated directivity index.

Abstract: A system and method for measuring the performance of a plurality of transducers integrated in one or more loudspeakers is described. The method simultaneously drives each transducer to emit sounds corresponding to distinct orthogonal test signals. A listening device senses sounds produced by the orthogonal test signals and analyzes the sensed audio signal to determine the performance of each transducer. By using orthogonal test signals, the multiple transducers may be measured and/or characterized simultaneously and with limited affect from extraneous noises.