Abstract: A portable electronic device including an enclosure having an enclosure wall that forms an interior chamber. A speaker module is positioned within the interior chamber and includes a speaker and a module wall forming a back volume chamber of the speaker. The back volume chamber includes an acoustic vent port formed through the module wall to acoustically couple the back volume chamber to the interior chamber. The device further including an electromechanical valve for regulating the acoustic coupling of the back volume chamber to the interior chamber. The electromechanical valve is operable to transition between an open configuration in which the acoustic vent port is open to the interior chamber and a closed configuration in which the acoustic vent port is closed off from the interior chamber.

Abstract: A method is described which decodes a downmix matrix for mapping a plurality of input channels of audio content to a plurality of output channels, the input and output channels being associated with respective speakers at predetermined positions relative to a listener position, wherein the downmix matrix is encoded by exploiting the symmetry of speaker pairs of the plurality of input channels and the symmetry of speaker pairs of the plurality of output channels. Encoded information representing the encoded downmix matrix is received and decoded for obtaining the decoded downmix matrix.

Abstract: Active acoustic filter systems and methods are disclosed. An active acoustic filter system includes a memory storing one or more stored sound profiles and a respective set of processing parameters associated with each of the one or more stored sound profiles. A processor coupled to the memory generates processed sound by processing ambient sound in accordance with a set of processing parameters retrieved from the memory based on results from comparing an ambient sound profile to at least one of the one or more stored sound profiles.

Abstract: Devices and techniques are generally described for automatic volume leveling of an audio signal. An audio signal may be received and a first portion of the audio signal may be stored in a buffer. A root mean square (RMS) estimate of a level of the first portion of the audio signal may be determined based at least in part on a second portion of the audio signal previous to the first portion. A modified audio signal may be generated by multiplying the first portion of the audio signal in the buffer by a variable gain. The variable gain may be based at least in part on the RMS estimate. The modified audio signal may be sent to a loudspeaker.

Abstract: System, method, and apparatus including receiving an identification of content played a playback system and an identification of location of the playback system. A playlist of content is associated with the location. A request is received to retrieve the playlist of content and the requested playlist of content is sent to a playback system.

Abstract: An audio and seating system is disclosed for one or more chairs comprising at least one wireless speaker.

Abstract: Various implementations include systems and approaches for binaural testing. In one implementation, a system includes a binaural test dummy including a body having: a head-and-neck region; and a set of head-mounted microphones coupled with the head-and-neck region at anatomically correct ear locations; and a control system coupled with the binaural test dummy for incrementally modifying a position of the binaural test dummy across a range of motion.

Abstract: Systems, methods, and non-transitory computer-readable media can receive, from a first entity, training data for training an intent model associated with a first intent of a plurality of intents. A first intent model associated with the first intent is generated based on the training data. The first intent model is made available in an intent marketplace for access by a second entity.

Abstract: Example techniques involve noise-robust acoustic echo cancellation. An example implementation may involve causing one or more speakers of the playback device to play back audio content and while the audio content is playing back, capturing, via the one or more microphones, audio within an acoustic environment that includes the audio playback. The example implementation may involve determining measured and reference signals in the STFT domain. During each nth iteration of an acoustic echo canceller (AEC): the implementation may involve determining a frame of an output signal by generating a frame of a model signal by passing a frame of the reference signal through an instance of an adaptive filter and then redacting the nth frame of the model signal from an nth frame of the measured signal. The implementation may further involve determining an instance of the adaptive filter for a next iteration of the AEC.

Abstract: Systems and methods for information extraction from natural language texts using a combination of classifier models.

Abstract: A multi-channel decorrelator for providing a plurality of decorrelated signals on the basis of a plurality of decorrelator input signals is configured to premix a first set of N decorrelator input signals into a second set of K decorrelator input signals, wherein K<N. The multi-channel decorrelator is configured to provide a first set of K? decorrelator output signals on the basis of the second set of K decorrelator input signals. The multi-channel decorrelator is further configured to upmix the first set of K? decorrelator output signals into a second set of N? decorrelator output signals, wherein N?>K?. The multi-channel decorrelator can be used in a multi-channel audio decoder. A multi-channel audio encoder provides complexity control information for the multi-channel decorrelator.

Abstract: A multi-channel audio decoder for providing at least two output audio signals on the basis of an encoded representation is configured to render a plurality of decoded audio signals, which are obtained on the basis of the encoded representation, in dependence on one or more rendering parameters, to obtain a plurality of rendered audio signals. The multi-channel audio decoder is configured to derive one or more decorrelated audio signals from the rendered audio signals, and to combine the rendered audio signals, or a scaled version thereof, with the one or more decorrelated audio signals, to obtain the output audio signals. A multi-channel audio encoder provides a decorrelation method parameter to control an audio decoder.

Abstract: Disclosed are an electronic device and a method of processing an audio signal by the electronic device. The electronic device includes: a processor functionally connected to a speaker and a microphone; and a memory electrically connected to the processor. The memory includes instructions to cause the processor, when executed, to output a first audio signal through the speaker; identify that a second audio signal detected by the microphone corresponds to the first audio signal when the first audio signal is outputted through the speaker; and control output of the first audio signal based on the identification. Further, various other embodiments may be possible.

Abstract: Low-latency audio networking is disclosed. In one embodiment, an example playback device includes a processor and memory having stored thereon instructions executable by the processor. The example instructions are to cause the first playback device to perform functions comprising: receiving audio information; selecting a first frequency channel of a first spectrum based on a threshold latency associated with the audio information; transmitting to the second playback device via a second frequency channel of a second spectrum, control information that identifies the first frequency channel of the first spectrum; and transmitting to the second playback device via the first frequency channel of the first spectrum, the audio information to be played by the second playback device.

Abstract: A mobile communication environment (100) can include a mobile device (160) to measure and send sound pressure level data. The mobile device (160) can initiate the collection of audio information responsive to detecting a trigger event. Mobile device (160) can measure or calculate the sound pressure level from the audio information. Metadata including time information and geographic location information can be captured with the collected audio information. Mobile device (160) can send the sound pressure level data and metadata through a wired or wireless communication path to a database (614).

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: Example embodiments disclosed herein relate to a estimation of reverberant energy components from audio sources. A method of estimating a reverberant energy component from an active audio source (100) is disclosed. The method comprises determining a correspondence between the active audio source and a plurality of sample sources by comparing one or more spatial features of the active audio source with one or more spatial features of the plurality of sample sources, each of the sample sources being associated with an adaptive filtering model (101); obtaining an adaptive filtering model for the active audio source based on the determined correspondence (102); and estimating the reverberant energy component from the active audio source over time based on the adaptive filtering model (103). Corresponding system (800) and computer program product (900) are also disclosed.

Abstract: A method and system for a headset with programmable modes, where the headset comprises a speaker and a beamforming microphone: sensing an ambient sound level near the headset, and configuring the headset in one of a plurality of modes by configuring a beam pattern of the beamforming microphone based on at least the sensed ambient noise level. A user of the headset may configure the headset in the one of the plurality of modes or it may be automatically configured. The beamforming microphone may comprise an array of sound sensing elements. The headset may be configured in a quiet mode when the sensed ambient sound level is below that of a desired sound source and may be configured with a wide beam pattern. The headset may be configured in a loud mode when the sensed ambient sound level is above that of a desired sound source.

Abstract: The present invention discloses a speaker module, comprising: a speaker assembly, a module shell and a front cover. The module shell is configured to bear the speaker assembly, and comprises a first shell and a second shell, wherein the first shell is doped with a thermally conductive filler. The front cover is configured to cooperate with the module shell to encapsulate the speaker assembly. The speaker module provided by the present invention can quickly discharge heat generated by the speaker assembly during operation through the module shell to prevent overheat of the speaker assembly, thereby avoiding performance loss of a speaker due to high temperature.

Abstract: An externally coupled speaker assembly having a horizontal ledge extending along a top side of a housing having an opening therein. A screen is positioned at the opening of the housing. A horizontal portion of the screen is retained between a bottom surface of a speaker and the horizontal ledge of the housing. A vertical portion of the screen is retained in a channel approximate to the opening of the housing.