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-channel audio system that can provide a variable sub-sample delay between two or more audio channels. In one embodiment, a variable timing clock generator generates multiple clock signals where each may have different phase, and the clock generator can vary the phase difference, in accordance with a sub-sample delay setting input. These clock signals are used by respective digital-to-analog converters (DACs) to convert the digital audio channels into analog form. In another embodiment, a variable delay block is added to an oversampling DAC, on a per channel basis. 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: Systems and methods for controlling echo in audio communications between a near-end system and a far-end system are described. The system and method may intelligently assign a plurality of microphone beams to a limited number of echo cancellers for processing. The microphone beams may be classified based on generated statistics to determine beams of interest (e.g., beams with a high ratio of local-voice to echo). Based on this ranking/classification of microphone beams, beams of greater interest may be assigned to echo cancellers while less important beams may temporally remain unprocessed until these beams become of higher importance/interest. Accordingly, a limited number of echo cancellers may be used to intelligently process a larger number of microphone beams based on interest in the beams and properties of echo cancellation performed for each beam.

Abstract: In one embodiment, a process for suppressing reverberation begins with a device of a user obtaining a reverberant speech signal from a voice of the user. The device determines a first estimated reverberation component of the reverberant speech signal. The device generates a first de-reverberated output signal with a first reverberation suppression based on the reverberant speech signal and the first estimated reverberation component. Then, the device generates a second improved reverberation component using the first de-reverberated output signal. The device generates a second de-reverberated output signal with a second reverberation suppression based on the reverberant speech signal and the second improved reverberation component.

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: A diaphragm for an audio speaker includes an outer shell and an inner shell. Each shell has a face portion and an edge portion that is formed to be substantially perpendicular to the face portion. The inner shell is inserted into the outer shell such that at least a part of the edge portion of each shell is in contact with at least a part of the edge portion of the other shell and a space is formed between the face portions of each shell. A cellular core fills the space formed between the face portions of each shell and is bonded to the face portion of each shell. The outer and inner shells may have identical sizes and shapes such that there is an interference fit between the contacting edge portions. A voice coil may be supported by the edge portions.

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 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: 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: In one embodiment, a process for suppressing reverberation begins with a device of a user obtaining a reverberant speech signal from a voice of the user. The device determines a first estimated reverberation component of the reverberant speech signal. The device generates a first de-reverberated output signal with a first reverberation suppression based on the reverberant speech signal and the first estimated reverberation component. Then, the device generates a second improved reverberation component using the first de-reverberated output signal. The device generates a second de-reverberated output signal with a second reverberation suppression based on the reverberant speech signal and the second improved reverberation component.

Abstract: An electronic audio device including an enclosure having an acoustic output opening and a speaker positioned within the enclosure. The speaker and the acoustic output opening are acoustically coupled by an acoustic output pathway. The acoustic output pathway includes a damping chamber to dampen a resonance frequency of the acoustic output pathway. The speaker is between the damping chamber and the acoustic output opening.

Abstract: Devices and methods are disclosed that allow for selective acoustic near-field nulls for microphone arrays. One embodiment may take the form of an electronic device including a speaker and a microphone array. The microphone array may include a first microphone positioned a first distance from the speaker and a second microphone positioned a second distance from the speaker. The first and second microphones are configured to receive an acoustic signal. The microphone array further includes a complex vector filter coupled to the second microphone. The complex vector filter is applied to an output signal of the second microphone to generate an acoustic sensitivity pattern for the array that provides an acoustic null at the location of the speaker.

Abstract: An electronic device having an enclosure having a top panel and a bottom panel. An electromagnet is mounted within the enclosure, the electromagnet having a core portion attached to the top panel and a coil connected to the core portion. An attractor plate is attached to the bottom panel, the attractor plate forming part of a magnetic circuit of the electromagnet such that when an electrical audio signal is applied to the electromagnet, the bottom panel vibrates and produces a sound. A permanent magnet is further attached to the core portion, the permanent magnet is configured to create a bias in the magnetic circuit so as to modify a distortion in the sound.

Abstract: A method and apparatus for outputting audio based on an orientation of an electronic device, or video shown by the electronic device. The audio may be mapped to a set of speakers using either or both of the device and video orientation to determine which speakers receive certain audio channels.

Abstract: Devices and methods are disclosed that allow for selective acoustic near-field nulls for microphone arrays. One embodiment may take the form of an electronic device including a speaker and a microphone array. The microphone array may include a first microphone positioned a first distance from the speaker and a second microphone positioned a second distance from the speaker. The first and second microphones are configured to receive an acoustic signal. The microphone array further includes a complex vector filter coupled to the second microphone. The complex vector filter is applied to an output signal of the second microphone to generate an acoustic sensitivity pattern for the array that provides an acoustic null at the location of the speaker.

Abstract: A method and apparatus for outputting audio based on an orientation of an electronic device, or video shown by the electronic device. The audio may be mapped to a set of speakers using either or both of the device and video orientation to determine which speakers receive certain audio channels.

Abstract: A diaphragm for an audio speaker includes an outer shell and an inner shell. Each shell has a face portion and an edge portion that is formed to be substantially perpendicular to the face portion. The inner shell is inserted into the outer shell such that at least a part of the edge portion of each shell is in contact with at least a part of the edge portion of the other shell and a space is formed between the face portions of each shell. A cellular core fills the space formed between the face portions of each shell and is bonded to the face portion of each shell. The outer and inner shells may have identical sizes and shapes such that there is an interference fit between the contacting edge portions. A voice coil may be supported by the edge portions.