Abstract: The present disclosure relates to a method of determining parameters for use by a first dynamic range control, DRC, model. The method comprises feeding a first audio signal to a second DRC model and receiving a second audio signal from the second DRC model, the second audio signal being a dynamic range controlled version of the first audio signal, rule-based selecting one or more pairs of samples of the first audio signal and corresponding samples of the second audio signal, and determining parameters of a first set of parameters among the parameters for use by the first DRC model based on the one or more selected pairs of samples. The present disclosure further relates to a method of reversing DRC of a dynamic range controlled audio signal and to a method of declipping a clipped audio signal that has been clipped by a DRC model. The present disclosure yet further relates to corresponding apparatus and computer-readable media.

Abstract: Natural language commands are automatically mapped to application programming interface (API) functions of web-based services. Grammatical rules may be applied to the APIs in order to classify the actions of the APIs, in some embodiments, generating mappings between the natural language commands and the APIs. When a natural language input is received and a natural language command identified form the natural language input, the corresponding API indicated in the mapping may be invoked.

Abstract: A sound adjustment method includes the following steps: obtaining a sound signal including frequency bands with a corresponding original sound pressure level value; converting the original sound pressure level value of each frequency band into a corresponding loudness level value; adjusting each loudness level value by a preset loudness level value to obtain an adjusted loudness level value of each frequency band; converting each adjusted loudness level value into a corresponding adjusted sound pressure level value; calculating a target sound pressure level value of each frequency band according to the original sound pressure level of each frequency band and each adjusted sound pressure level value; adjusting the original sound pressure level value of each frequency band by each target sound pressure level value to obtain an adjusted sound signal; outputting the adjusted sound signal.

Abstract: An earbud system is described. The earbud system may include a primary earbud having a primary antenna. The earbud system may further include a secondary earbud having a secondary antenna. The primary antenna may have a higher effective radiation area than the secondary antenna. The secondary antenna may have a higher gain than the primary antenna.

Abstract: A signal processing device comprises: input transducer(s) configured to convert input(s) to an input signal; output transducer(s) configured to convert an output signal to output(s); a signal processing circuit configured to at least subtract a feedback estimation signal from the input signal to produce a feedback compensated signal; and an adaptive feedback estimator.

Abstract: The present disclosure provides a hearing aid charger case. The hearing aid charger case comprises a housing, a lining and a charger module. The housing comprises a lid body and a lower housing; the lid body and the lower housing body are pivoted to each other through a hinge; a back surface of the lower housing body is disposed with a USB interface that connects a power supply. The lining and the charger module are arranged in the lower housing body, and the lining is located above the charger module and is fixedly connected to the charger module, and the inside of the lining are provided with two charging bays. The bottom of each charging bay is disposed with two charging pins at intervals. The hearing aid charger case can charge and store two hearing aids at the same time, which is greatly convenient for the users.

Abstract: A method performed by a wearable audio output device worn by a user is provided for controlling the reproduction of external noise and/or noise masking. A first a first lower-power detector determines if the user is in transport or a change in the user's transport. In response to the determination, a second higher-power detector exits a power saving state and determines a mode of the transport based. The wearable audio output device adjust a level of attenuation applied by the wearable audio output device to the external noise or adjusting a level of noise masking output by the wearable audio output device based on the determined mode of the transport.

Abstract: Apparatus, computer readable media and methods for processing a multi-channel, spatial audio format input signal. For example, one such method comprises determining object location metadata based on the received spatial audio format input signal; and extracting object audio signals based on the received spatial audio format input signal, wherein the extracting object audio signals based on the received spatial audio format input signal includes determining object audio signals and residual audio signals.

Abstract: A headset includes: a microphone, a first transmission line that transmits a first electrical signal generated by the microphone to an external device; an operation switch that receives an operation for switching between (i) a non-mute state in which the first electrical signal is transmitted to the external device and (ii) a mute state in which the first electrical signal is not transmitted to the external device; an electronic switch that switches between (i) a conductive state in which the first electrical signal is transmitted and (ii) a non-conductive state in which the first electrical signal is not transmitted, on the basis of a control voltage which changes according to the state of the operation switch; and a voltage generator that generates the control voltage.

Abstract: The present invention is directed to methods and apparatus for translating a first plurality of audio input channels to a second plurality of audio output channels. This includes determining that there is pair-wise coding among any of the first plurality of audio input channels, determining an input/output-mapping matrix for mapping at least a first set of the first plurality of audio input channels to at least a second set of the second plurality of audio output channels; and deriving the second plurality of audio output channels based on first plurality of audio input channels, the input/output-mapping matrix and the determined pair-wise coding. The first plurality of audio input channels represent the same soundfield represented by the second plurality of audio output channels.

Abstract: The invention discloses a method for noise reduction in a binaural hearing aid, said binaural hearing aid comprising a first local unit and a second local unit, wherein the method comprises the following steps: generating a first main signal and a first auxiliary signal in the first local unit from an environment sound, and a second main signal in the second local unit from the environment sound, estimating a direction of arrival of a useful sound signal in the environment sound, assigning a first frequency range and a second frequency range, generating a first range beamformer signal in the first frequency range from the first main signal, the first auxiliary signal and the second main signal by imposing at least one spatial condition related to the estimated direction of arrival on the directional characteristic of the first range beamformer signal, generating a second range beamformer signal in the second frequency range from the first main signal and the second main signal by imposing at least one spatial

Abstract: A robotic system that can be used to treat a patient. The robotic system includes a mobile robot that has a camera. The mobile robot is controlled by a remote station that has a monitor. A physician can use the remote station to move the mobile robot into view of a patient. An image of the patient is transmitted from the robot camera to the remote station monitor. A medical personnel at the robot site can enter patient information into the system through a user interface. The patient information can be stored in a server. The physician can access the information from the remote station. The remote station may provide graphical user interfaces that display the patient information and provide both a medical tool and a patient management plan.

Abstract: A method and apparatus of determining the position of a mobile device in a region of a vehicle cabin are described. The mobile device has a speaker and at least one microphone. The vehicle has an audio system comprising at least two speakers. The mobile device detects first and second acoustic signals respectively transmitted via the first and second vehicle speaker. The acoustic signals comprise a respective detection pattern. The detection patterns are mutually orthogonal. The detected acoustic signals may be compared or correlated with the detection patterns and a respective matched acoustic signal generated. The location of the mobile device within a region of the vehicle cabin may be determined based on the time difference of arrival of the first matched acoustic signal and the second matched acoustic signal. The mode of operation of the mobile device may be set or changed dependent on the location of the mobile device.

Abstract: The present disclosure describes systems, devices, and techniques related to a wearable audio device, such as an earbud or other device that is configured to detect inputs and change the operation of the wearable audio device in accordance with the inputs. In some embodiments, the wearable audio device is disposed in a structure and detects signals propagating through or within the structure. Various inputs may cause one or more signals to propagate through or within the structure, outside the structure, or some combination thereof. The wearable audio device may determine whether a detected signal was generated by an input and, if so, change its operation in accordance with the input.

Abstract: Wireless-enabled loudspeaker includes a wooden capacitive touch user interface. The loudspeaker may comprise at least one electroacoustic transducer, a processor in communication with the at least one electroacoustic transducer, and a wooden exterior surface comprising a capacitive touch user interface that allows a user to control operation of the loudspeaker. The wooden exterior surface acts a dielectric for the capacitive touch user interface. The loudspeaker may comprise a wireless transceiver circuit for receiving and transmitting wireless communication signals via a wireless network. The wireless transceiver circuit may receive wirelessly audio content from streaming audio content servers that are connected to the Internet. The capacitive touch user interface comprises a plurality of user control icons etched in the wooden exterior surface, and a plurality of capacitive sense electrodes located under the wooden exterior surface.

Abstract: A sound or thermal baffling device comprising an enclosure containing a variable density fluid and a force generating means for preserving and creating the structure and form of the enclosure, the shape and composition of the enclosure crafted to vary the baffling characteristics of the enclosure, and a further embodiment showing how a cellular material containing a variable density fluid may be created and used, and a still further embodiment showing improvements to ear protectors and head phone sets, including latching means for attaching these and other devices to the ears and head. Various applications involving previous as well as new uses are set out, including a description of how dynamic sound baffling may be implemented.

Abstract: Embodiments are described for a method of rendering audio for playback through headphones comprising receiving digital audio content, receiving binaural rendering metadata generated by an authoring tool processing the received digital audio content, receiving playback metadata generated by a playback device, and combining the binaural rendering metadata and playback metadata to optimize playback of the digital audio content through the headphones.

Abstract: The present disclosure relates to a method for playing audio, an apparatus for playing audio, a playing device, and storage medium, and belongs to the field of electronic technology. The method includes detecting an amount of audio sampling points in a buffer area that is configured to store the audio sampling points received from an electronic device based on a first sampling frequency, wherein the audio sampling points are obtained by decoding audio data from the electronic device; when a first quantity of the sampling points based on the first sampling frequency is out of a default quantity range, adjusting a preset first frequency-division parameter to obtain a second frequency-division parameter; determining a second sampling frequency based on the second frequency-division parameter; and extracting the audio sampling points from the buffer area based on the second sampling frequency for playing audio data.

Abstract: There are two representations for Higher Order Ambisonics denoted HOA: spatial domain and coefficient domain. The invention generates from a coefficient domain representation a mixed spatial/coefficient domain representation, wherein the number of said HOA signals can be variable. An aspect of the invention further relates to methods and apparatus decoding multiplexed and perceptually encoded HOA signals, including transforming a vector of PCM encoded spatial domain signals of the HOA representation to a corresponding vector of coefficient domain signals by multiplying the vector of PCM encoded spatial domain signals with a transform matrix and de-normalizing the vector of PCM encoded and normalized coefficient domain signals, wherein said de-normalizing comprises.

Abstract: An apparatus comprising: an input configured to receive at least two audio signals, the at least two audio signals having a relative displacement between them; an orientation determiner configured to determine an audio capture orientation based on a device orientation and information concerning microphone configuration in the apparatus; and an audio output generator configured to output the at least two audio signals based on the capture orientation, such that playback of the at least two audio signals is performed based on the capture orientation.