Abstract: An electronic device includes one or more microphones that generate audio signals and a wind noise detection subsystem. The electronic device may also include a wind noise reduction subsystem. The wind noise detection subsystem applies multiple wind noise detection techniques to the set of audio signals to generate corresponding indications of whether wind noise is present. The wind noise detection subsystem determines whether wind noise is present based on the indications generated by each detection technique and generates an overall indication of whether wind noise is present. The wind noise reduction subsystem applies one or more wind noise reduction techniques to the audio signal if wind noise is detected. The wind noise detection and reduction techniques may work in multiple domains (e.g., the time, spatial, and frequency domains).

Abstract: Techniques and systems are described for generating and using a sound localization model. A described technique includes obtaining for a building a sound sensor map indicating locations of first and second sound sensor devices in respective first and second rooms of the building; causing an autonomous device to navigate to the first room and to emit, during a time window, sound patterns at one or more frequencies within the first room; receiving sound data including first and second sound data respectively from the first and second sound sensor devices that are observed during the time window; and generating and storing a sound localization model based on the sound sensor map, autonomous device location information, and the received sound data, the model being configured to compensate for how sounds travels among rooms in at least a portion of the building such that an origin room of a sound source is identifiable.

Abstract: A method for automatic for sound recognition, comprising a) raw spectrogram generation from a sound signal spectrum; b) wide-band spectrum determination; c) wide-band continuous spectrum determination; d) tonal and time-transient spectrum determination; wide-band continuous spectrogram and tonal and time-transient spectrogram determination; and) spectrogram image generation.

Abstract: A system and method for routing the audio output from a selected one of a plurality of independent audio sources to an audio system for amplification/playback; the routing being based upon the movement of a user's head, and in particular the user's line-of-sight inferred from the movement. The invention employs a spatial mapping of the plurality of audio sources and routes the respective audio output of each of the sources to the audio amplification/playback system upon making the determination that the user's line-of-sight is being directed to a particular one of the audio sources.

Abstract: Systems and methods of presenting an output audio signal to a listener located at a first location in a virtual environment are disclosed. According to embodiments of a method, an input audio signal is received. For each sound source of a plurality of sound sources in the virtual environment, a respective first intermediate audio signal corresponding to the input audio signal is determined, based on a location of the respective sound source in the virtual environment, and the respective first intermediate audio signal is associated with a first bus. For each of the sound sources of the plurality of sound sources in the virtual environment, a respective second intermediate audio signal is determined. The respective second intermediate audio signal corresponds to a reverberation of the input audio signal in the virtual environment.

Abstract: A system and method can provide nose, such as wind noise, mitigation and/or microphone blending. Some methods may include sampling a sound signal from a plurality of microphones to generate a frame comprising a plurality of time-frequency tiles of the sound signal, each time-frequency tile including respective values of at least one feature from the plurality of microphones, comparing the respective values of the at least one feature to determine whether each time-frequency tile satisfies a similarity threshold, and flagging each time-frequency tile as noise if it fails to satisfy the similarity threshold, grouping the plurality of time-frequency tiles into sets of frequency-adjacent time-frequency tiles, and for each set of frequency-adjacent time-frequency tiles in the frame: counting a number of flagged time-frequency tiles, and attenuating all of the time-frequency tiles in the each set if the number exceeds a noise bin count threshold to thereby reduce noise in the sound signal.

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 device includes one or more processors configured to receive, via wireless transmission from a streaming device, encoded ambisonics audio data representing a sound field. The one or more processors are also configured to perform decoding of the ambisonics audio data to generate decoded ambisonics audio data. The decoding of the ambisonics audio data includes base layer decoding of a base layer of the encoded ambisonics audio data and selectively includes enhancement layer decoding in response to an amount of movement of the device. The one or more processors are further configured to adjust the decoded ambisonics audio data to alter the sound field based on data associated with at least one of a translation or an orientation associated with the movement of the device. The one or more processors are also configured to output the adjusted decoded ambisonics audio data to two or more loudspeakers for playback.

Abstract: A system for facilitating configuration of an audio system tests the existing audio system configuration by playing reference audio clips of various different genres of programming, the audio of which is received by a microphone of a remote control device or other handheld mobile device at various different listening points in the room. The system then compares the audio signal representing audio of the reference audio clip received at the microphone to a reference audio signal representing the reference audio clip and determines other audio characteristics. The system then determines, based on the comparison and other characteristics of the room (e.g., furniture layout, room construction materials, wall treatments and current speaker positioning) suggested changes to a configuration of the audio system to increase audio quality of the audio system for playing audio associated with the various different genres of programming.

Abstract: A system and method for quieting unwanted sound. As a non-limiting example, various aspects of this disclosure provide a system and method, for example implemented in a premises-based or home audio system, for quieting unwanted sound at a particular location.

Abstract: An audio renderer can have a machine learning model that jointly processes audio and visual information of an audiovisual recording. The audio renderer can generate output audio channels. Sounds captured in the audiovisual recording and present in the output audio channels are spatially mapped based on the joint processing of the audio and visual information by the machine learning model. Other aspects are described.

Abstract: A synchronization method performed for a first piece of audio equipment connected to a wireless network via a first radio point that is identified by a first BSSID, the method comprising the steps of: detecting a second piece of audio equipment connected to the same wireless network and forming part of the same audio group; acquiring a second BSSID of a second radio point via which the second piece of audio equipment is connected to the wireless network, and then determining whether the first BSSID and the second BSSID are identical; if not, performing a selection stage to select an optimum radio point that is identified by an optimum BSSID; if the optimum BSSID is different from the first BSSID, connecting the first piece of audio equipment to the wireless network via the optimum radio point.

Abstract: In one aspect, a device may include at least one processor and storage accessible to the at least one processor. The storage may include instructions executable by the at least one processor to identify at least one characteristic associated with audio as sensed at a first location, with the audio being produced at a second location that is different from the first location. The instructions may also be executable to, based on the at least one identified characteristic, adjust a first volume level of a first component of the audio in a first frequency and/or first frequency band but not a second volume level of a second component of the audio in a second frequency and/or second frequency band of the audio.

Abstract: Audio data is generated for a vehicle audio system using a portable computing device. Vehicle parameter data is received at the portable computing device from a vehicle. Sound parameter data is generated from the vehicle parameter data. Audio data is generated using a synthesizer based on the sound parameter data. The generated audio data is transmitted to the vehicle audio system from the portable computing device.

Abstract: Generating, rendering and outputting one or more audio signals that have one or more audio sequences and/or spectral modulations of an audio file including a plurality of audio-based user prompts in an audio interference processing system. An audio interference processing system and method incudes initiating an instance of an audio interference processing session, configuring one or more session parameters, rendering/outputting a training audio output to one or more transducers, rendering/outputting one or more audio discrimination and audio navigation signals to the one or more transducers, receiving two or more sensor inputs in response to rendering/outputting one or more audio discrimination and audio navigation signals to the one or more transducers, processing the two or more sensor inputs, and providing a feedback output to a user via one or more modalities.

Abstract: A system and method for dynamically controlling multiple wireless speakers in response to the sensed location of one or more listeners and predefined audio preferences associated therewith. The position of a device associated with a given user is continuously monitored and the state of the wireless speakers adjusted accordingly so as to provide the user with a customized, yet consistent audio experience as they move with respect to the wireless speakers. Prioritization among multiple user devices is also provided for, thereby ensuring that higher-priority users are provided with the aural environment they have specified, or placing the wireless speaker system into an operational state that provides a suitable audio environment for all users.

Abstract: A method and apparatus for handling both echo cancellation and noise in a given space using a single adaptive filter. The proposed method utilizes just a single adaptive filter and without any control logic, the system is capable of handling both the AEC and the noise in a given space. The technology as proposed and claimed herein proposes using a single adaptive filter to cancel referenced noise AEC and unreferenced noise ANC in a given physical space using a beamformer. By adding the far end reference and a beam pointed dynamically on the noise source in the given physical space this can be achieved. This invention also reduces the processing load and memory.

Abstract: In one aspect, a device may include at least one processor and storage accessible to the at least one processor. The storage may include instructions executable by the at least one processor to determine an operative microphone of the device from among plural available microphones and take one or more actions at the first device based on determining the operative microphone. The one or more actions may include, for example, presenting an indication at the device that a first microphone is the operative microphone, and/or switching the operative microphone from one available microphone to another available microphone at which audible input is already being received.

Abstract: Example techniques involve a calibration state variable. An example implementation receives, via a network interface, an indication that the first playback device is calibrated. Based on receiving the indication that the first playback device is calibrated, the example implementation updates a calibration state variable to indicate that the first playback device is calibrated, wherein the calibration state variable is stored in the data storage. The example implementation sends, via the network interface, an indication of the updated calibration state variable to a second device.

Abstract: The invention relates to an amplifier unit for a MEMS sound transducer, which is operable as a microphone and as a loudspeaker, comprising at least one audio amplifier for sound reproduction and/or sound recording. According to the invention, the amplifier unit is designed in such a way that the MEMS sound transducer provided therefor is simultaneously operable as a loudspeaker and as a microphone. Moreover, the invention relates to sound-generating unit comprising a MEMS sound transducer, which is operable as a microphone and as a loudspeaker, and an amplifier unit coupled to the sound transducer for sound reproduction and/or sound recording.