Abstract: A system and method to process audio data received over the ARC or eARC interface of HDMI from audio sources are provided. A media device may receive compressed audio data in a number of data formats. The media device may convert between the audio formats provided by the audio sources and the audio formats supported by audio playback devices. The media device may inspect frames of audio data to determine if the frames are to be decoded. The frame may be decoded and subsequently encoded into the data formats supported by the audio playback devices. To reduce latency, the media device may enable a pass-through mode to bypass the decoding of the frames to allow the frames to be decoded at the audio playback devices. A bi-directional loopback application may route audio data received over the ARC or eARC interface from the audio sources to the audio playback devices.

Abstract: Implementations of the subject technology provide systems and methods for determining whether to interrupt a user of an audio device that is operating in a noise-cancelling mode of operation. For example, the user may desire to be interrupted by one or more pre-designated contacts that are identified at an associated electronic device as interrupt-authorized contacts, or by a person who speaks a designated keyword to the user.

Abstract: Techniques to use an embedded passcode within an audio ringtone to establish a secure connection for arbitrary phone relay are described. The use of an embedded passcode enables encrypted ad-hoc connections for the relay of audio of an incoming telephone call to a secondary device, such as a virtual assistant enabled smart speaker device.

Abstract: A system and method to mitigate the temporary loss of the input sampling clocks when receiving audio data over the ARC or eARC interface of HDMI are provided. A media device may substitute an externally generated clock derived from a local crystal oscillator of the media device for the missing input sampling clock. The external clock may be synchronized to the frequency of the input sampling clock. The media device may synchronize the external clock to the audio data when there is a loss of the input sampling clock. When the input sampling clock of the audio data reappears, the media device may switch back from the external clock to the input sampling clock. When transitioning between the input sampling clock and the external clock, the media device may insert zero padding into the audio data samples to mute any potential glitch in the sound from an audio playback device.

Abstract: A system and method to process audio data received over the ARC or eARC interface of HDMI from audio sources are provided. A media device may receive compressed audio data in a number of data formats. The media device may convert between the audio formats provided by the audio sources and the audio formats supported by audio playback devices. The media device may inspect frames of audio data to determine if the frames are to be decoded. The frame may be decoded and subsequently encoded into the data formats supported by the audio playback devices. To reduce latency, the media device may enable a pass-through mode to bypass the decoding of the frames to allow the frames to be decoded at the audio playback devices. A bi-directional loopback application may route audio data received over the ARC or eARC interface from the audio sources to the audio playback devices.

Abstract: A system and method to mitigate the temporary loss of the input sampling clocks when receiving audio data over the ARC or eARC interface of HDMI are provided. A media device may substitute an externally generated clock derived from a local crystal oscillator of the media device for the missing input sampling clock. The external clock may be synchronized to the frequency of the input sampling clock. The media device may synchronize the external clock to the audio data when there is a loss of the input sampling clock. When the input sampling clock of the audio data reappears, the media device may switch back from the external clock to the input sampling clock. When transitioning between the input sampling clock and the external clock, the media device may insert zero padding into the audio data samples to mute any potential glitch in the sound from an audio playback device.

Abstract: Techniques to use an embedded passcode within an audio ringtone to establish a secure connection for arbitrary phone relay are described. The use of an embedded passcode enables encrypted ad-hoc connections for the relay of audio of an incoming telephone call to a secondary device, such as a virtual assistant enabled smart speaker device.

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: 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.

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: An audio system embodiment includes a loudspeaker cabinet having at least one loudspeaker transducer and defining a longitudinal axis. Several microphones are distributed around the longitudinal axis, defining an array of microphones. A reference microphone is positioned in the loudspeaker cabinet, e.g., in a rear chamber of the at least one loudspeaker transducer. The audio system includes a processor and a memory having instructions that, when executed by the processor, cause the audio system to receive an audio signal from each distributed microphone and the reference microphone, and therefrom to estimate a direction, relative to the plurality of microphones, of a nearby, acoustically reflective surface. Responsive to the estimated direction, the audio system affects a mode of operation, e.g., beam forms an audio output in a selected direction corresponding to the estimated direction of the acoustically reflective surface. Related principles are described by way of reference to exemplary embodiments.

Abstract: An audio receiver receives one or more input audio signals representing one or more channels of a sound content and applies a first beam pattern to the input audio signals to generate a first set of beam-formed audio signals. The audio receiver determines a second beam pattern that is less directional than the first beam pattern. The audio receiver determines that driving of a loudspeaker array using the first set of beam-formed audio signals will cause one or more transducers of the loudspeaker array to operate beyond an operational threshold, and in response applies the second beam pattern to the input audio signals to generate a second set of beam-formed audio signals. The audio receiver drives the loudspeaker array using the second set of beam-formed audio signals. Other embodiments are also described and claimed.

Abstract: An audio system includes one or more loudspeaker cabinets, each having loudspeakers. The system outputs an omnidirectional sound pattern to determine the acoustic environment. Sensing logic determines an acoustic environment of the loudspeaker cabinets. The sensing logic may include an echo canceller. A playback mode processor adjusts an audio program according to a playback mode determined from the acoustic environment of the audio system. 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: An audio system embodiment includes a loudspeaker cabinet having at least one loudspeaker transducer and defining a longitudinal axis. Several microphones are distributed around the longitudinal axis, defining an array of microphones. A reference microphone is positioned in the loudspeaker cabinet, e.g., in a rear chamber of the at least one loudspeaker transducer. The audio system includes a processor and a memory having instructions that, when executed by the processor, cause the audio system to receive an audio signal from each distributed microphone and the reference microphone, and therefrom to estimate a direction, relative to the plurality of microphones, of a nearby, acoustically reflective surface. Responsive to the estimated direction, the audio system affects a mode of operation, e.g., beam forms an audio output in a selected direction corresponding to the estimated direction of the acoustically reflective surface. Related principles are described by way of reference to exemplary embodiments.

Abstract: An audio system includes one or more loudspeaker cabinets, each having loudspeakers. The system outputs an omnidirectional sound pattern to determine the acoustic environment. Sensing logic determines an acoustic environment of the loudspeaker cabinets. The sensing logic may include an echo canceller. A playback mode processor adjusts an audio program according to a playback mode determined from the acoustic environment of the audio system. 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: 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 loudspeaker cabinet has a number of pairs of microphones, each pair includes the same internal microphone and a different external microphone. For each pair of microphones, a process (i) receives a first audio signal of sound captured by the internal microphone and a second audio signal of sound captured by the different external microphone, (ii) estimates, using first and second audio signals, a radiation impedance, and (iii) computes a detection value based on the radiation impedance in a frequency band. A difference between (i) a currently computed detection value associated with a given pair of microphones and (ii) a previously computed detection value associated with said given pair, is computed. The sound produced by the cabinet is adjusted, in response to the computed difference meeting a threshold. Other embodiments are also described and claimed.

Abstract: A loudspeaker cabinet has a number of pairs of microphones, each pair includes the same internal microphone and a different external microphone. For each pair of microphones, a process (i) receives a first audio signal of sound captured by the internal microphone and a second audio signal of sound captured by the different external microphone, (ii) estimates, using first and second audio signals, a radiation impedance, and (iii) computes a detection value based on the radiation impedance in a frequency band. A difference between (i) a currently computed detection value associated with a given pair of microphones and (ii) a previously computed detection value associated with said given pair, is computed. The sound produced by the cabinet is adjusted, in response to the computed difference meeting a threshold. Other embodiments are also described and claimed.

Abstract: An audio receiver receives one or more input audio signals representing one or more channels of a sound content and applies a first beam pattern to the input audio signals to generate a first set of beam-formed audio signals. The audio receiver determines a second beam pattern that is less directional than the first beam pattern. The audio receiver determines that driving of a loudspeaker array using the first set of beam-formed audio signals will cause one or more transducers of the loudspeaker array to operate beyond an operational threshold, and in response applies the second beam pattern to the input audio signals to generate a second set of beam-formed audio signals. The audio receiver drives the loudspeaker array using the second set of beam-formed audio signals. Other embodiments are also described and claimed.

Abstract: A loudspeaker cabinet has a number of pairs of microphones, each pair includes the same internal microphone and a different external microphone. For each pair of microphones, a process (i) receives a first audio signal of sound captured by the internal microphone and a second audio signal of sound captured by the different external microphone, (ii) estimates, using first and second audio signals, a radiation impedance, and (iii) computes a detection value based on the radiation impedance in a frequency band. A difference between (i) a currently computed detection value associated with a given pair of microphones and (ii) a previously computed detection value associated with said given pair, is computed. The sound produced by the cabinet is adjusted, in response to the computed difference meeting a threshold. Other embodiments are also described and claimed.