Abstract: An audio system that maintains an identical or similar direct-to-reverberant ratio for sound produced from a first speaker array and sound produced by a second speaker array at the location of a listener is described. The audio system may determine characteristics of the first and second speaker arrays, including the distance between the first speaker array and the listener and the second speaker array and the listener. Based on these characteristics, beam patterns are selected for one or more of the speaker arrays such that sound produced by each of the speaker arrays maintains a preferred direct-to-reverberant ratio at the location of the listener.

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: 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: Loudspeakers are described that may reduce comb filtering effects perceived by a listener by either 1) moving transducers closer to a sound reflective surface (e.g., a baseplate, a tabletop or a floor) through vertical (height) or rotational adjustments of the transducers or 2) guiding sound produced by the transducers to be released into the listening area proximate to the reflective surface through the use of horns and openings that are at a prescribed distance from the reflective surface. The reduction of this distance between the reflective surface and the point at which sound emitted by the transducers is released into the listening area may lead to a shorter reflected path that reduces comb filtering effects caused by reflected sounds that are delayed relative to the direct sound. Accordingly, the loudspeakers shown and describe may be placed on reflective surfaces without sever audio coloration caused by reflected sounds.

Abstract: Loudspeakers are described that may reduce comb filtering effects perceived by a listener by either 1) moving transducers closer to a sound reflective surface (e.g., a baseplate, a tabletop or a floor) through vertical (height) or rotational adjustments of the transducers or 2) guiding sound produced by the transducers to be released into the listening area proximate to the reflective surface through the use of horns and openings that are at a prescribed distance from the reflective surface. The reduction of this distance between the reflective surface and the point at which sound emitted by the transducers is released into the listening area may lead to shorter reflected path that reduces comb filtering effects caused by reflected sounds that are delayed relative to the direct sound. Accordingly, the loudspeakers shown and described may be placed on reflective surfaces without severe audio coloration caused by reflected sounds.

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: A system and method for driving a loudspeaker array across directivities and frequencies to maintain timbre constancy in a listening area is described. In one embodiment, a frequency independent room constant describing the listening area is determined using the directivity index of a first beam pattern, the direct-to-reverberant ratio DR at the listener's location in the listening area, and an estimated reverberation time T60 for the listening area at a designated frequency. On the basis of this room constant, an offset may be generated for a second beam pattern. The offset describes the decibel difference between first and second beam patterns to achieve constant timbre and may be used to adjust the second beam pattern at multiple frequencies. Maintaining constant timbre improves audio quality regardless of the characteristics of the listening area and the beam patterns used to represent sound program content. Other embodiments are also described.

Abstract: A directivity pattern generator for producing sound patterns using a modal architecture is described. The directivity pattern generator may include a beam pattern mixing unit, which defines sound patterns to be emitted by an audio system in terms of a set of frequency invariant modes or modal patterns. The beam pattern mixing unit produces a set of modal gains representing the level or degree each of the predefined modal patterns is to be applied to a set of audio streams. Modal filters may be used to modal amplitudes that compensate for inefficiencies of the each modal pattern at low frequencies. The directivity pattern generator may include a modal decomposition unit for generating driving signals for each transducer in one or more loudspeaker arrays based on weighted values for the modal gains/amplitudes.

Abstract: An audio receiver that performs crosstalk cancellation using a speaker array is described. The audio receiver detects the location of a listener in a room and processes a piece of sound program content to be output through the speaker array using one or more beam pattern matrices. The beam pattern matrices are generated according to one or more constraints. The constraints may include increasing a right channel and decreasing a left channel at the right ear of the listener, increasing a left channel and decreasing a right channel at the left ear of the listener, and decreasing sound in all other areas of the room. These constraints cause the audio receiver to beam sound primarily towards the listener and not in other areas of the room such that crosstalk cancellation is achieved with minimal effects due to changes to the frequency response of the room. Other embodiments are also described.

Abstract: A system for enhancing audio including a computer and an output device. The computer includes a sensor configured to determine a user location relative to the computer. The sensor is also configured to gather environment data corresponding to an environment of the computer. The computer also includes a processor in communication with the sensor and configured to process the user location and the environment data and adjust at least one of an audio output or a video output. The output device is in communication with the processor and is configured to output at least one of the audio output or the video output.

Abstract: A loudspeaker system includes a driver in an enclosure that provides a back volume which is sealed with respect to acoustic pressure waves produced by a driver diaphragm. An external microphone is located outside the back volume. An internal microphone located inside the back volume. A computational unit is coupled to the external microphone and the internal microphone and configured to determine a transfer function for an equalization filter. The transfer function determination is responsive to the external microphone and the internal microphone. A digital signal processor is coupled to a signal source, the driver, and the computational unit. The digital signal processor is configured to implement the equalization filter as determined by the computational unit, create a filtered audio signal from the signal source, and provide the filtered audio signal to the driver.

Abstract: A multi-way speaker array is disclosed that includes rings of transducers of different types. The rings of transducers may encircle the cabinet of the speaker array such that the speaker array is rotationally symmetric. The distance between rings of transducers may be based on a logarithmic scale. By separating rings of transducers using logarithmic spacing, denser transducer spacing at short wavelengths is achieved while limiting the number of transducers needed for longer wavelengths by spacing them in larger and larger logarithmic increments. Transducers with overlapping frequency ranges may be used in the speaker array to avoid initial dips or shortfalls in directivity for corresponding beam patterns.

Abstract: A multi-way speaker array is disclosed that includes rings of transducers of different types. The rings of transducers may encircle the cabinet of the speaker array such that the speaker array is rotationally symmetric. The distance between rings of transducers may be based on a logarithmic scale. By separating rings of transducers using logarithmic spacing, denser transducer spacing at short wavelengths is achieved while limiting the number of transducers needed for longer wavelengths by spacing them in larger and larger logarithmic increments. Transducers with overlapping frequency ranges may be used in the speaker array to avoid initial dips or shortfalls in directivity for corresponding beam patterns.

Abstract: A system and method are described for transforming stereo signals into mid and side components xm and xs to apply processing to only the side-component xs and avoid processing the mid-component. By avoiding alteration to the mid-component XM, the system and method may reduce the effects of ill-conditioning, such as coloration that may be caused by processing a problematic mid component xM while still performing crosstalk cancellation and/or generating virtual sound sources. Additional processing may be separately applied to the mid and side components xM and xs and/or particular frequency bands of the original stereo signals to further reduce ill-conditioning.

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 loudspeaker array has a cabinet in which is formed a continuously open circumferential horn for controlling sound produced by a number of transducers which are positioned in the cabinet, at a throat of the horn. The continuously open circumferential horn may 1) improve the power efficiency of the transducers without unwanted aliasing effects in audible frequency ranges and 2) provide vertical control for sound emitted by the transducers by flaring.

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.