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 appliance can include a microphone transducer, a processor, and a memory storing instructions. The appliance is configured to receive an audio signal at the microphone transducer and to detect an utterance in the audio signal. The appliance is further configured to classify a speech mode based on the utterance. The appliance is further configured to determine conditions of an environment of the appliance. The appliance is further configured to select at least one of a playback volume or a speech output mode from a plurality of speech output modes based on the classification, and the conditions of the environment of the appliance. The appliance is further configured to adapt the playback volume and/or mode of played-back speech according to the speech output mode. The appliance may be configured to synthesize speech according to the speech output mode, or to modify synthesized speech according to the speech output mode.

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 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: 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 appliance can include a microphone transducer, a processor, and a memory storing instructions. The appliance is configured to receive an audio signal at the microphone transducer and to detect an utterance in the audio signal. The appliance is further configured to classify a speech mode based on the utterance. The appliance is further configured to determine conditions of an environment of the appliance. The appliance is further configured to select at least one of a playback volume or a speech output mode from a plurality of speech output modes based on the classification, and the conditions of the environment of the appliance. The appliance is further configured to adapt the playback volume and/or mode of played-back speech according to the speech output mode. The appliance may be configured to synthesize speech according to the speech output mode, or to modify synthesized speech according to the speech output mode.

Abstract: A method performed by an audio system within a room that includes a first loudspeaker that has a first beamforming array and a second loudspeaker that has a second beamforming array. The method obtains a sound program as several input audio channels. The method performs a beamforming algorithm based on the input audio channels to cause each of the arrays to produce a front beam pattern and a side beam pattern when the loudspeakers are close to an object within the room, where the front beam patterns are directed away from the object and the side beam patterns are directed towards the object and the beam patterns contain different portions of the sound program. The method performs a cross-talk cancellation (XTC) algorithm based on a subset of the input audio channels to produce several XTC output signals for driving the arrays when the loudspeakers are far away from the object.

Abstract: 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.

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 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: 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 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.

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: A method for using a loudspeaker array that is housed in a loudspeaker cabinet to present audio content to a listener in a room includes receiving (1) an audio channel that includes audio content and (2) acoustical characteristics of the room. The method also produces (1) a first beamformer input signal from the audio channel and (2) a second beamformer input signal and a third beamformer input signal by decorrelating the audio channel and adjusting the audio channel in accordance with the acoustical characteristics of the room. The second and third beamformer input signals are different de-correlated versions of the audio channel. The method also generates driver signals from the first, second, and third beamformer input signals to drive the loudspeaker array to produce a main beam, a first ambient beam, and a second ambient beam, respectively. Other embodiments are also described and claimed.

Abstract: A microphone equalization system determines an equalization filter which is used to spectrally shape a signal from a microphone. A loudspeaker has an enclosure, a back volume, a driver diaphragm, and a sensor located inside the back volume. An external microphone is located outside the loudspeaker. A processor determines the equalization filter based on comparison of a first signal from an output of the sensor and a second signal from an output of the external microphone both of which are produced while the driver diaphragm produces acoustic pressure waves produced by the driver diaphragm. The processor is to then spectrally shape a third signal from the output of the external microphone that is responsive to further acoustic pressure waves from a source other than the loudspeaker. Other aspects 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 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 includes one or more loudspeaker cabinets, each having loudspeakers. Sensing logic determines an acoustic environment of the loudspeaker cabinets. The sensing logic may include an echo canceller. A low frequency filter corrects an audio program based on the acoustic environment of the loudspeaker cabinets. The system outputs an omnidirectional sound pattern, which may be low frequency sound, to determine the acoustic environment. The system may produce a directional pattern superimposed on an omnidirectional pattern, if the acoustic environment is in free space. The system may aim ambient content toward a wall and direct content away from the wall, if the acoustic environment is not in free space. The sensing logic automatically determines the acoustic environment upon initial power up and when position changes of loudspeaker cabinets are detected. Accelerometers may detect position changes of the loudspeaker cabinets.

Abstract: A transition between a stereophonic presentation and a monophonic presentation of a stereophonic input signal that includes a left channel signal and a right channel signal extracts content that is present at similar levels but not in-phase between the left and right channel signals to produce at least one of a left enhancement signal and a right enhancement signal. The left channel signal, the right channel signal, and only one of the left and right enhancement signals are combined to produce a monophonic signal for the monophonic presentation. Cross-fading between the left channel signal and the monophonic signal and between the right channel signal and the monophonic signal may be used to transition between the stereophonic presentation and the monophonic presentation. The stereophonic input signal may be up-mixed to produce enhancement signal. A similar transition between a multichannel presentation and a monophonic presentation of a multichannel signal is described.

Abstract: An electronic device having a device housing includes a loudspeaker and several microphones within the device housing. A control circuit is electrically coupled to the loudspeaker and microphones. The loudspeaker produces speech and/or music. The control circuit determines a statistical measure for a first data set representing individual impulse responses from the plurality of microphones and compares that to a predetermined statistical measure for a second data set representing individual object-free impulse responses from the plurality of microphones to determine if an object is near the device. The statistical measure may be variance and may be computed in the time domain. Variance may be calculated using differences between the individual impulse responses and a mean impulse response that is a linear combination of the impulse responses for the plurality of microphones. The control circuit may include echo cancellers to mitigate common signals and/or other acoustic sources.