Abstract: A process for reproducing sound using a loudspeaker array that is housed in a loudspeaker cabinet includes the selection of a number of sound rendering modes and changing the selected sound rendering mode based on changes in one or both of sensor data and a user interface selection. The sound rendering modes include a number of mid-side modes and at least one direct-ambient mode. Other embodiments are also described and claimed.

Abstract: A process for reproducing sound using a loudspeaker array that is housed in a loudspeaker cabinet includes the selection of a number of sound rendering modes and changing the selected sound rendering mode based on changes in one or both of sensor data and a user interface selection. The sound rendering modes include a number of mid-side modes and at least one direct-ambient mode. Other embodiments are also described and claimed.

Abstract: A binaural sound reproduction system, and methods of using the binaural sound reproduction system to dynamically re-center a frame of reference for a virtual sound source, are described. The binaural sound reproduction system may include a head-mounted device, e.g., headphones, having a device sensor to provide device orientation data corresponding to a direction of the head-mounted device. The system may use the orientation data to determine whether the head-mounted device has moved over an angle within a range of movement, and may adjust an audio output to render the virtual sound source in an adjusted source direction based on the range of movement. Other embodiments are also described and claimed.

Abstract: A differential pressure gradient micro-electro-mechanical system (MEMS) microphone for measuring an acoustic characteristic of a loudspeaker. The microphone includes a MEMS microphone housing and a compliant membrane mounted in the MEMS microphone housing, the compliant membrane dividing the MEMS microphone housing into a first chamber and a second chamber. The first chamber includes a primary port open to a first side of the compliant membrane and the second chamber includes a secondary port open to a second side of the compliant membrane, and the primary port and the secondary port are tuned with respect to one another to control a pressure difference between the first side and the second side of the compliant membrane such that at least 10 dB of attenuation is observed in a microphone signal output relative to a microphone having a sealed first or second chamber.

Abstract: A binaural sound reproduction system, and methods of using the binaural sound reproduction system to dynamically re-center a frame of reference for a virtual sound source, are described. The binaural sound reproduction system may include a reference device, e.g., a mobile device, having a reference sensor to provide reference orientation data corresponding to a direction of the reference device, and a head-mounted device, e.g., headphones, having a device sensor to provide device orientation data corresponding to a direction of the head-mounted device. The system may use the reference orientation data to determine whether the head-mounted device is being used in a static or dynamic use case, and may adjust an audio output to render the virtual sound source in an adjusted source direction based on the determined use case. Other embodiments are also described and claimed.

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: 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 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 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 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: 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: 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.