Abstract: Apparatuses, systems, and techniques for assigning execution of applications to various processing units using machine learning are disclosed herein. Usage data for an application to be executed using a computing system including an integrated processing unit and a discrete processing unit is identified. At least a portion of operations of the application to be executed using the integrated processing unit or the discrete processing unit based on the usage data and in view of at least one of one or more system performance metrics or one or more user experience metrics associated with executing the application using the integrated processing unit and the discrete processing unit.

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 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 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 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 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 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 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 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 loudspeaker that measures the impulse response of a listening area is described. The loudspeaker may output sounds corresponding to a segment of an audio signal. The sounds are sensed by a listening device proximate to a listener and transmitted to the loudspeaker. The loudspeaker includes an adaptive filter that estimates the impulse response of the listening area based on the signal segment. An error unit analyzes the estimated impulse response together with the sensed audio signal received from the listening device to determine the accuracy of the estimate. New estimates may be generated by the adaptive filter until an accuracy level is achieved for the signal segment. A processor may utilize one or more estimated impulse responses corresponding to various signal segments that cover a defined frequency spectrum for adjusting the audio signal to compensate for the impulse response of the listening area. Other embodiments are also described.

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: Methods and apparatus for processing media signals. In one embodiment, a data processing device processes fixed and variable rate data using a first and second processing unit. The processing comprises real-time processing of audio/video signals by a graphics processing unit (GPU) and/or central processing unit (CPU). The processing units process data efficiently by establishing one processor as always processing variable rate data, and using one or more schemes for determining processor will process fixed rate data. A shared memory enables the processors to communicate with one another in order to determine which will process the fixed rate data. In one scheme for determining which of the processors will process the fixed rate data the second processor need merely be unlocked. In another embodiment, the second processor must be unlocked and immediately available.

Abstract: A loudspeaker that measures the impulse response of a listening area is described. The loudspeaker may output sounds corresponding to a segment of an audio signal. The sounds are sensed by a listening device proximate to a listener and transmitted to the loudspeaker. The loudspeaker includes an adaptive filter that estimates the impulse response of the listening area based on the signal segment. An error unit analyzes the estimated impulse response together with the sensed audio signal received from the listening device to determine the accuracy of the estimate. New estimates may be generated by the adaptive filter until an accuracy level is achieved for the signal segment. A processor may utilize one or more estimated impulse responses corresponding to various signal segments that cover a defined frequency spectrum for adjusting the audio signal to compensate for the impulse response of the listening area. 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: 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: 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 apparatus induces a first wireless device to transmit audio-related data to both the apparatus and a second wireless device by providing the first wireless device with a false indication of the apparatus being capable of performing an audio function that the apparatus is incapable of performing so as to enable the second wireless device to receive the audio-related data simultaneously with the apparatus, and wherein the second wireless device cooperates with the apparatus to leave the first wireless device unaware of the simultaneous receipt of the audio-related data by the second wireless device.

Abstract: A dictation computer that includes a fan speed regulator is described. The fan speed regulator monitors a speech recognition unit to determine when the speech recognition unit is activated. Upon detection that the speech recognition unit is activated, the fan speed regulator ducks the speed of a cooling fan embedded within the dictation computer to an optimized speed of rotation over a delay time interval. The fan speed regulator may include components to adapt the optimized speed and delay time to the characteristics of the dictation computer and the user. Other embodiments are also described.

Abstract: Methods and apparatus for processing media signals. In one embodiment, a data processing device processes fixed and variable rate data using a first and second processing unit. The processing comprises real-time processing of audio/video signals by a graphics processing unit (GPU) and/or central processing unit (CPU). The processing units process data efficiently by establishing one processor as always processing variable rate data, and using one or more schemes for determining processor will process fixed rate data. A shared memory enables the processors to communicate with one another in order to determine which will process the fixed rate data. In one scheme for determining which of the processors will process the fixed rate data the second processor need merely be unlocked. In another embodiment, the second processor must be unlocked and immediately available.