Abstract: Methods and apparatus, including computer program products, for controlling a speaker, which is electrically powered with a low-power source and connected to a short-term energy storage. Audio is received for playback on the speaker. A time-resolved power analysis of the audio is acquired and a time-resolved speaker power requirement required by a speaker playing back the audio is calculated. The time-resolved speaker power requirement is compared with a combined capacity of the low-power source and the short-term energy storage. One or more of: a dynamic range, a frequency range, and an output gain of a digital signal processor are adjusted such that the speaker power requirement meets the combined capacity of the low-power source and the short-term energy storage for the duration of a playback of the received audio on the speaker.

Abstract: A system and method include reception of a first plurality of audio signals, generation of a second plurality of beamformed audio signals based on the first plurality of audio signals, each of the second plurality of beamformed audio signals associated with a respective one of a second plurality of beamformer directions, generation of a first TF mask for a first output channel based on the first plurality of audio signals, determination of a first beamformer direction associated with a first target sound source based on the first TF mask, generation of first features based on the first beamformer direction and the first plurality of audio signals, determination of a second TF mask based on the first features, and application of the second TF mask to one of the second plurality of beamformed audio signals associated with the first beamformer direction.

Abstract: Examples described herein relate to portable playback devices, such as smart headphones and earbuds, and ultra-portable devices having built-in voice assistants. Some example techniques relate to user interaction with voice assistants. Further example techniques relate to voice guidance played back by the headphones to guide the user under certain conditions.

Abstract: Array microphone systems and methods that can automatically focus and/or place beamformed lobes in response to detected sound activity are provided. The automatic focus and/or placement of the beamformed lobes can be inhibited based on a remote far end audio signal. The quality of the coverage of audio sources in an environment may be improved by ensuring that beamformed lobes are optimally picking up the audio sources even if they have moved and changed locations.

Abstract: A system in a first vehicle includes a plurality of sensors, a plurality of audio output devices and circuitry. The circuitry receives a plurality of signals from the plurality of sensors. The circuitry determines a relative position between a first device and the first vehicle based on the received plurality of signals or a geo-location of the first vehicle. The circuitry receives first information from the first device based on the determined relative position between the first device and the first vehicle, and at least one selection criterion. The circuitry generates a first three-dimensional (3D) audio object corresponding to the received first information. The circuitry controls, based on the determined relative position, a first set of audio output devices of the plurality of audio output devices to output the generated first 3D audio object, in a 3D space of the first vehicle at a first time instant.

Abstract: A system including circuitry communicably coupled to a plurality of sensors and a plurality of audio output devices are provided. The circuitry receives a plurality of signals from the plurality of sensors. The circuitry further detects a type of a first object and a position of the first object with respect to a first vehicle based on the received plurality of signals. The circuitry further generates metadata associated with the first object based on the detected type of the first object and the position of the first object. The metadata comprises a first three-dimensional (3D) audio object. The circuitry further controls a first set of audio output devices of the plurality of audio output devices to output the first 3D audio object in a 3D space of the first vehicle at a first time instant.

Abstract: A conversation support system supports conversation of passengers in an automobile. Seats of at least two or more rows are placed in the automobile, and the conversation support system includes a speech switching control part configured to designate a desired sound pickup and reproduce area based on designation by a first passenger sitting in a first seat, a first target speech emphasizing part configured to output a signal obtained by emphasizing speech emitted from the designated sound pickup and reproduce area to a loud speaker corresponding to the first seat, and a second target speech emphasizing part configured to output the signal obtained by emphasizing speech, which is collected with a microphone corresponding to the first seat and which is emitted from the first seat, to a loud speaker corresponding to the sound pickup and reproduce area.

Abstract: An audio mixing system including an on-site digital audio workstation and an off-site audio workstation may be provided. The on-site digital audio workstation may be configured to receive a number of audio data channels and may produce a multi-channel mix by mixing the number of audio data channels. The off-site audio workstation may be coupled, by a remote dynamic host configuration protocol server, to the digital audio workstation and may transmit mixing control data to the digital audio workstation, thereby changing properties of the multi-channel mix.

Abstract: A method and apparatus for detecting a photoacoustic light signal to prevent unauthorized voice commands for a microphone-controllable device are provided. The method includes receiving, by a processor, a signal, detecting, by the processor, that the signal is a photoacoustic signal generated by a thermal expansion and contraction of an object caused by at least one lightwave applied to the object, and activating, by the processor, a counter-measure to prevent the photoacoustic signal from reaching a microphone of a microphone-controllable device in response to detecting the photoacoustic signal.

Abstract: A sound sensing module for monitoring the operation of an appliance includes a microphone for monitoring sound generated during operation of the appliance and a controller operably coupled to the microphone. The controller can obtain a sound signal generated during operation of the appliance, analyze the sound signal to identify a sound signature corresponding to an operating event, and implement a responsive action, such as providing a user notification, providing troubleshooting instructions, ordering a replacement part, scheduling a maintenance visit.

Abstract: Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent.

Abstract: The present invention relates to a home entertainment system, which includes four major parts: entertainment play equipment, a Bluetooth transmitter, a set of rear audio play devices, and a set of front audio play devices with a subwoofer. The rear audio play devices are composed of a main speaker, a sub-speaker, wire, and a Bluetooth receiver; the front Bluetooth speakers are composed of a main speaker, a sub-speaker, a subwoofer, wire, and a Bluetooth receiver, and each Bluetooth receiver is integrated in each main speaker. This home entertainment system uses a Bluetooth transmitter module to connect the entertainment play equipment with the rear speakers and the front speakers through wireless radio waves, and can simultaneously emit 4-channel audio of left front, left rear, right front, and right rear to achieve stereo surround sound effects. When the existing home entertainment audio is installed, it is necessary to drill, wire, and construct on the wall.

Abstract: A signal processing device includes a switching receptor, a storage, and a signal processor. The switching receptor receives switching of a speaker serving as a supply destination of a signal. The storage stores an optimal setting, which is obtained by measuring characteristics of the speaker selected by the switching, in association with the switching of the speaker. The signal processor reads out the optimal setting, which is associated with the switching received by the switching receptor, from the storage, and uses the optimal setting to process the signal to be supplied to the speaker.

Abstract: Methods and systems for vocal interaction preservation for teleported audio. A method includes determining spatial parameters of a first space including at least one sound source and at least one audio source, wherein the at least one sound source emits sound within the first space, wherein the at least one audio source captures audio data based on sounds emitted within the first space, wherein the spatial parameters of the first space characterize sound characteristics of the first space; determining vocal spatial parameters of each of the at least one sound source, wherein the vocal spatial parameters of each sound source define characteristics of the sound source which affect sound waves emitted by the sound source; and generating, for each sound source, a respective clean version of the audio data based on the spatial parameters of the first space and the vocal spatial parameters of the sound source.

Abstract: A digital audio array circuit is provided. The digital audio array circuit includes at least two digital audio units and a system master unit. Each of the digital audio units is configured to transform a received sound wave to a digital audio signal. Each of the digital audio units includes a left/right channel configuration input terminal. The system master unit is connected to the at least two digital audio units in time division multiplexing to receive the digital audio signals. The left/right channel configuration input terminal of each of the digital audio units is configured to receive a same synchronizing signal.

Abstract: Embodiments disclosed herein extending an audio track by joining similar portions. Audio features (e.g., spectral features, modulation features) may be extracted from the audio track. The audio track may be segmented, e.g., based on the audio features, and each segment may be slid through the audio track using a timestep. In each timestep, the sliding segment may be compared to the underlying portion of the audio track and a similarity score (e.g., a cross-correlation) may be generated. A self-similarity matrix may be generated based on the comparisons involving all the segments. The self-similarity matrix may be analyzed for peak values and segments corresponding to the peak values may be joined to extend the audio track. The embodiments may be applied to any kind of audio including music, ambient noise, speech, etc.

Abstract: An audio-based repetition counter system and method are disclosed. In some embodiments, the audio-based repetition sound counter system includes a computing device connected to a microphone. The computing device includes a processor and one or more memory devices, and a repetition counter module is stored on the memory device and executed by the processor. The repetition counter module processes audio data received from a microphone and provide a number of repetitions for one or more series of one or more activities; for example, jumping rope. Further, an audio-based method of counting repetitions for a physical activity is provided.

Abstract: A system configured to improve audio processing by performing dereverberation and noise reduction during a communication session. The system may apply a two-channel dereverberation algorithm by calculating coherence-to-diffuse ratio (CDR) values and calculating dereverberation (DER) gain values based on the CDR values. While the DER gain values may be calculated at a first stage within the pipeline, the device may apply the DER gain values at a second stage within the pipeline. For example, the device may calculate the DER gain values prior to performing residual echo suppression (RES) processing but may apply the DER gain values after performing RES processing, in order to avoid excessive attenuation of the local speech. In addition to removing reverberation, the DER gain values also remove diffuse noise components, reducing an amount of noise reduction required. Thus, the device may soften noise reduction when the DER gain values are applied.

Abstract: A method includes receiving sound by a first audio unit installed in an electrical outlet, routing audio data corresponding to the received sound from the first audio unit to a second audio unit installed in a second electrical outlet, and sending the audio data to a mobile device using a wireless link between the mobile device and the second audio unit. Routing the audio data may include receiving the audio data from the first audio unit by a third audio unit and routing the audio data to the second audio unit by the third audio unit serving as a router. The data may be routed using table driven routing, on-demand routing or some other appropriate routing protocol. The method may also include performing voice recognition on the audio data and detecting a command word and routing command word data to the second audio unit.

Abstract: In some embodiments, a system comprises a host computing device and an audio device including at least one speaker and a plurality of microphones, the audio device being wirelessly and communicatively coupled to the host computing device. The host computing device can include one or more processors and one or more machine-readable, non-transitory storage mediums that include instructions configured to cause the one or more processors of the host computing device to perform operations including: receiving user environment data by one or more sensors of the host computing device; determining a characterization profile of a surrounding environment of the user based on the user environment data; and sending the characterization profile to the audio device, the characterization profile configured to cause the audio device to adapt an audio cardioid pattern of the plurality of microphones on the audio device based on the characterization profile.