Abstract: Sound device receives a first sound signal from a first sound processor, generates a second sound signal based on the first sound signal, and transmits the second sound signal to a second sound processor. The second sound processor performs signal processing to the second sound signal to generate a third sound signal. The sound device receives the third sound signal from the second sound processor, checks a state of the second sound processor based on a signal received from the second sound processor, transmits a fourth sound signal based on the third sound signal to the first sound processor when determining that the state of the second sound processor is normal, and generates a fifth sound signal based on the first sound signal or the second sound signal to transmit it to the first sound processor when determining that the state of the second sound processor is abnormal.

Abstract: Example techniques relate to device spaces and default designations in a media playback system. A device space may create an association between a networked microphone device and one or more playback devices such that certain voice commands (e.g., playback commands) received by the networked microphone device are used to control the one or more playback devices (unless otherwise designated in the voice command). Furthermore, in bonded pairs and bonded groups of playback devices that include at least one NMD, certain playback devices within the bonded pair or group may be designated as default so as to avoid multiple responses to a voice input.

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: Methods, apparatuses, and non-transitory computer-readable storage mediums are provided for information processing. The method may be applied to an electronic equipment. The electronic equipment may collect environmental audio information when the electronic equipment plays multimedia. The electronic equipment may also perform noise detection on the environmental audio information to determine whether the environmental audio information represents a target noise scenario. The electronic equipment may also process a parameter of the multimedia played by the electronic equipment when the environmental audio information represents the target noise scenario.

Abstract: One or more accelerometers embedded with an earbud and/or a set of earphones are able to sense a moving pace of a user. Based on a moving pace of the user, a signal is sent to a remotely connected electronic device. The electronic device is able to separately increase and decrease a beat or rhythm of the audio from the electronic device based on a pace of the user. In some embodiments, an audio alert is sent to the user to inform the user of pace and whether the user has increased or decreased their pace. Additionally, in some embodiments, a program stored on the electronic device is used to compare the user's current progress and/or speed based on past runs and workouts.

Abstract: An example method of operation may include initiating an automated tuning procedure, detecting via one or more microphones a sound measurement associated with an output of one or more speakers at two or more locations, determining a number of speech transmission index (STI) values equal to a number of microphones, and averaging the speech transmission index values to identify a single speech transmission index value.

Abstract: Provided is an information processing device that controls and presents sound information in an appropriate form to a user who acts in an environment on the basis of situation recognition including recognition of the environment and recognition of the actions of the user.

Abstract: This application relates to audio driving circuitry (100), and in particular to audio driving circuitry for outputting first and second audio driving signals for driving a stereo audio load (106), which may be a stereo audio load of an accessory apparatus (102) removably coupled to the audio driving circuitry in use. A load monitor (111) is provided for monitoring to monitor, from a monitoring node (112), an indication of a common mode return current passing through a common return path, together with an indication of a common mode component of the first and second audio driving signals and to determine an impedance characteristic of the stereo audio load. The load monitor (111) can provide dynamic monitoring of any significant change in load impedance. In some embodiments the load monitor (111) comprises an adaptive filter (301) which adapts a parameter of the filter which is related to the load impedance so as to determine the indication of load impedance.

Abstract: One embodiment provides a computer-implemented method that includes receiving a trigger sound from a primary listening location. The trigger sound being received at multiple speakers in a synchronous network at different times. The trigger sound is recognized at the multiple speakers. A respective relative delay is determined based on a time differential function that determines time differences. Sound quality for the multiple speakers is improved based on the respective relative delay for each of the multiple speakers.

Abstract: A sound signal output method of outputting sound to a plurality of areas comprises obtaining a sound signal of a sound source, receiving a setting of reproduction information indicating whether or not to output the sound signal of the above-mentioned sound source to each of the plurality of areas where the above-mentioned sound signal is to be output, and outputting the above-mentioned reproduction information and the above-mentioned sound signal, or reproducing the above-mentioned sound signal based on the above-mentioned reproduction information.

Abstract: Processing acoustic signals to detect sound sources in a sound scene. The method includes: obtaining a plurality of signals representative of the sound scene, captured by a plurality of microphones of predefined positions; based on the signals captured by the microphones and on the positions of the microphones, applying a quantization of directional measurements of sound intensity and establishing a corresponding acoustic activity map in a sound source localization space, the space being of dimension N; constructing at least one vector basis of dimension less than N; projecting the acoustic activity map onto at least one axis of the vector basis; and searching for at least one local peak of acoustic activity in the map projection, an identified local peak corresponding to the presence of a sound source in the scene.

Abstract: Microphone signals are received from microphones of a computer device. Each microphone signal of the microphone signals is acquired by a respective microphone of the microphones. A previously unselected microphone is selected from the microphones as a reference microphone, which generates a reference microphone signal. An adaptive filter is used to create, based on microphone signals of the microphones other than the reference microphone, predicted microphone signals for the reference microphone. Based on the predicted microphone signals for the reference microphone, an enhanced microphone signal is outputted for the reference microphone. The enhanced microphone signal may be used as microphone signal for the reference microphone in subsequent audio processing operations.

Abstract: Determination of a composite acoustic parameter value for a headset is presented herein. A directionally enhanced audio signal is generated based on audio signals from an acoustic sensor array and a spatial signal enhancement filter that is directed for enhancement of a sound source. A SNR improvement value is determined based on a SNR value of the directionally enhanced audio signal and a SNR value of an audio signal from an acoustic sensor of the acoustic sensor array. The SNR improvement value is input into a model that maps SNR improvement values to spatial acoustic parameters to determine a spatial acoustic parameter. A temporal acoustic parameter is determined based on the audio signals. The composite acoustic parameter value is determined based on the spatial acoustic parameter and a temporal acoustic parameter value. Audio content presented to a user is adjusted based in part on the composite acoustic parameter value.

Abstract: Data-analysis-based processing is provided to validate audio content delivery via embedded inaudible sound signals. The processing obtains, by a validation system, an indication of detection of an inaudible sound signal embedded in an audio output of provided content initiated by a user on a user-device. The inaudible sound signal is inaudible to the user, and the audio output is from the user-device. Based on obtaining the indication of detection of the inaudible sound signal embedded in the audio output, the validation system validates that the audio output of the provided content is ON with respect to the user-device.

Abstract: One or more sensors are configured to contextualize a series of user generated movements to control one or more electronic devices. For example, a set of earphones comprises one or more sensors for sensing a location of the earphones. The one or more sensors enable earphones such as a pair of bluetooth or earphones wirelessly coupled to a bluetooth enabled electronic device, the capability to understand the configuration of use of the earphones. Based on a location and use or non-use of the earphones, one or more contextual responses is able to be applied for a given action.

Abstract: A method for generating an acoustic stimulus for use in an ear biometric process on a user, the method comprising: receiving an indication of stimulation frequencies for use in the ear biometric process; grouping the stimulation frequencies into bands of a psychoacoustic scale; generating the acoustic stimulus, the acoustic stimulus comprising a masked bandpass component within each band of the psychoacoustic scale that comprises one or more of the stimulation frequencies.

Abstract: An amplifier having one or more channels where each channel includes a two half bridges (a master and slave sub-channel). The sub-channels can be connected either in parallel or in a full-bridge configuration via internal switches that route signals to a pair of speaker jacks. One switch in the amplifier has a first position that selectively connects the outputs of the master and slave sub-channel to the same input of the speaker load so that the two sub-channels will drive the speaker load in parallel and a second position where the output of the slave sub-channel is connected to another input of the speaker load so that the master sub-channel and the slave sub-channel will drive the speaker load in a Full-bridge configuration.

Abstract: An electronic device can include a housing defining an aperture and a display positioned in the aperture. The display and the housing can define an internal volume in which a speaker assembly is positioned. The speaker assembly can include a speaker module and a speaker enclosure in fluid communication, with the speaker enclosure at least partially defining a speaker volume.

Abstract: A method for testing an audio signal system having multiple audio component connection sockets, each socket having at least one audio signal interface connected to a ground interface of the socket via a series circuit including a resistor and a capacitor, including feeding, via a signal processing device, a time domain reflectometry test signal into a wired network of the audio signal system, in which the sockets are connected to the signal processing device via differently lengthed electrical lines, detecting in a temporally resolved manner, via the signal processing device, test return signals reflected at the multiple sockets, comparing the detected test return signals with a temporally resolved reference pattern for test return signals, which reference pattern was created under predefined boundary conditions, and determining a faulty connection of a socket if comparing the detected test return signals with the reference pattern reveals a deviation above a predefined threshold value.

Abstract: A sound source estimation system includes a microphone that detects sound which is generated from an object including a rotary device, a rotation speed acquiring unit that acquires a rotation speed of the rotary device, a frequency analyzing unit that generates frequency sound data indicating change of a frequency spectrum of sound detected by the microphone, a degree calculating unit that calculates a degree based on a loudest sound frequency indicating loudest sound out of frequencies of the sound and the rotation speed, and a degree comparing unit that determines a component with a degree closest to the degree as a sound source candidate of the noise.