Abstract: An information processing device is provided to present a sensation to a user taking a delay occurring before sound is perceived by the user appropriately into consideration. The information processing device includes a generation section that generates, on the basis of a difference between a first timing at which a sound is to be outputted from an acoustic presentation section and a second timing at which the sound is to be perceived by a first user, first data to be used for presentation of a sensation to the first user by a sensation presentation section worn by the first user.

Abstract: An example operation may include one or more of receiving a data file comprising a list of interactive voice response (IVR) prompts, identifying an IVR prompt with incorrect content that will cause an error during one or more of reading and playing of the IVR prompt, modifying the incorrect content via addition or removal of an element from text content within the IVR prompt, and storing the modified IVR prompt in memory.

Abstract: Conformable sealing eartips having ribs formed on an interior surface of one or more eartip flanges are provided. The eartip may include a body and one or more flanges. The body may include an ear insertion end, an earphone insertion end opposite the ear insertion end, and a central opening extending between the ear insertion end and the earphone insertion end to form an inner sound channel through the body. The one or more flanges may extend from the body and may include an outer sealing surface and an inner surface. At least one of the one or more flanges may include a plurality of ribs protruding from the inner surface of the at least one of the one or more flanges.

Abstract: In some examples, loudness enhancement based on multiband range compression may include determining, based on variations in compression parameters that include compression thresholds and compression ratios, corresponding variations in loudness levels for a specified loudness standard. A learning model may be trained based on the variations in the compression parameters and the corresponding variations in the loudness levels. A specified loudness level for a device may be ascertained, for example, from a user of the device. The compression parameters for the specified loudness level may be determined based on the trained learning model. Further, sub-band compression of an input audio signal may be performed, based on the determined compression parameters, by processing the input audio signal using a perfect reconstruction filterbank.

Abstract: Various acoustic manipulation and microphone gain adjustment techniques are enabled. For instance, a method comprises receiving, by a voice-enabled device comprising a processor and from a microphone, an input comprising environmental audio, determining, by the voice-enabled device, a level of the environmental audio, determining, by the voice-enabled device, a gain-adjust value based on the environmental audio, and adjusting, by the voice-enabled device, a gain of the microphone by the gain-adjust value.

Abstract: Various embodiments of a speaker and a method of forming such speaker are disclosed. The speaker includes a frame having a first major surface, a second major surface, and an opening disposed between the first and second major surfaces; a cone disposed adjacent the first major surface of the frame; and an adhesive layer disposed on at least a portion of the second major surface of the frame. The adhesive layer occludes the opening of the frame.

Abstract: A system and method for providing an interactive voice response dialogue by using a cloud computing system coupled to a mobile computing device. The cloud computing system may receive a request from a mobile computing device to initiate communication with an interactive voice response system having a tree structure including audio files in the cloud computing system. An interactive voice response dialogue may be conducted using the cloud computing system when the mobile computing device does not have the capacity to receive the one or more files. When the mobile computing device does have the capacity to receive the one or more files, one or more files may be transmitted from the cloud computing system to the mobile computing device, so that he mobile computing device may conduct the interactive voice response dialogue.

Abstract: This invention is a gunshot detection device that provides very reliable inside and outside real-time situational awareness of gunshot events, while reducing Gunshot Detection False Positives and Negatives.

Abstract: An electronic device is disclosed. In addition, various embodiments recognized through the specification are possible. The electronic device includes a sensor, a processor operatively connected with the sensor, and a memory, operatively connected with the processor, including instructions. The instructions, when executed by the processor, cause the processor to perform biometric authentication for a user of the electronic device using the sensor, while a call connection procedure is performed with an ARS server, receive a voice signal for requesting private information of the user from the ARS server and convert the voice signal into text, and determine private information requested by the ARS server among at least one private information of the user, the at least one private information being previously stored in the memory, based on the converted text, and transmit the determined private information to the ARS server.

Abstract: Apparatuses, methods, systems, and program products are disclosed for detecting and correcting audio echo. An apparatus includes a processor and a memory that stores code executable by the processor. The code is executable by the processor to detect audio that is received through an audio channel of a first user during a conference call being echoed through an audio channel of a second user in the conference call. The code is executable by the processor to identify the second user associated with the audio channel that is echoing the audio of the first user. The code is executable by the processor to perform a corrective action associated with the second user to prevent audio that is received from the first user from being echoed through the audio channel of the second user.

Abstract: An MEMS microphone is provided, comprising a first substrate and a vibration diaphragm supported above the first substrate by a spacing portion, the first substrate, the spacing portion, and the vibration diaphragm enclosing a vacuum chamber, and a static deflection distance of the vibration diaphragm under an atmospheric pressure being less than a distance between the vibration diaphragm and the first substrate, wherein: one of the vibration diaphragm and the first substrate is provided with a magnetic film, and the other one of the vibration diaphragm and the first substrate is provided with a magnetoresistive sensor cooperating with the magnetic film, the magnetoresistive sensor being configured to sense a change in a magnetic field of the magnetic film during a vibration of the vibration diaphragm and output a varying electrical signal.

Abstract: Various embodiments of the invention provide methods, systems, and computer-program products for handling a call reaching a party's voicemail. A predictive model is used to generate a first probability of an outcome occurring as a result of leaving a voicemail message and a second probability of the outcome occurring as a result of sending a text message. If the first probability is greater, an IVR uses DTMF tone(s) to bypass a greeting on the voicemail so that a message can be recorded with minimal time. If the second probability is greater, the call is ended and an email is sent to the party's service provider using an email address comprising a mobile number for the party and a domain portion specific to the provider. As a result, the provider delivers a text message to the party with the body of the message including the content of the email.

Abstract: When a new communication is received at a server, the system can generate machine readable data pertaining to the communication, e.g., a transcript for the communication. Using the transcript, the system can utilize natural language processing techniques to find at least one request in the communication. The system can search summaries of prior communications, and if there is a communication including a similar request, the system can display a record of the prior communication.

Abstract: Methods for generating an object based audio program which is renderable in a personalizable manner, e.g., to provide an immersive, perception of audio content of the program. Other embodiments include steps of delivering (e.g., broadcasting), decoding, and/or rendering such a program. Rendering of audio objects indicated by the program may provide an immersive experience. The audio content of the program may be indicative of multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects, and typically also a default set of objects which will be rendered in the absence of a selection by a user) and a bed of speaker channels. Another aspect is an audio processing unit (e.g., encoder or decoder) configured to perform, or which includes a buffer memory which stores at least one frame (or other segment) of an object based audio program (or bitstream thereof) generated in accordance with, any embodiment of the method.

Abstract: Systems and methods are provided for acoustic echo canceling. In one embodiment, a system for canceling acoustic echoes comprises one or more microphones configured to pick up sound generated by a sound source and transferred from the sound source to the one or more microphones via one or more unknown transfer paths having one or more unknown transfer functions, and to provide one or more electrical microphone signals therefrom, one or more adaptive filters for approximating the one or more unknown transfer functions with one or more estimated transfer functions and filtering one or more electrical signals representative of the sound generated by the sound source with the one or more estimated transfer functions to provide one or more estimated signals therefrom, and a microphone calibration block configured to individually attenuate or amplify the one or more electrical microphone signals dependent on the one or more estimated transfer functions.

Abstract: Provided are a voice denoising method and apparatus, a server and a storage medium. The voice denoising method comprises: acquiring voice signals synchronously collected by an acoustic microphone and a non-acoustic microphone (S100); carrying out voice activity detection according to the voice signal collected by the non-acoustic microphone to obtain a voice activity detection result (S110); and according to the voice activity detection result, denoising the voice signal collected by the acoustic microphone to obtain a denoised voice signal (S120). The effect of denoising can be enhanced, and the quality of voice signals can be improved.

Abstract: Electronic glasses display a virtual image for a telephone call before a first user wearing the electronic glasses answers the telephone call. The electronic glasses display the virtual image at a location in an environment of the first user where a voice of a second user will externally localize in empty space outside the head of the first user when the first user answers the telephone call.

Abstract: A loudspeaker system can have first and second loudspeakers selectively operable in a single-channel mode or in a multi-channel mode. In the single-channel mode, the first and the second loudspeakers are configured to simultaneously reproduce a substantially identical signal. In the multi-channel mode, the first loudspeaker reproduces a first-channel signal and the second loudspeaker reproduces a second-channel signal. The first-channel signal and the second-channel signal can constitute respective portions of a multi-channel signal. Such loudspeaker systems can also have a mode selector configured to select one of the single-channel mode and the multi-channel mode. In some embodiments, such selection can occur in response to one or more detected proximities of another loudspeaker system. Multi-zone loudspeaker systems are also disclosed.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, relating to synthetic call status updates. In some implementations, a method includes determining, by a task manager module, that a triggering event has occurred to provide a current status of a user call request. The method may then determine, by the task manager module, the current status of the user call request. A representation of the current status of the user call request is generated. Then, the generated representation of the current status of the user call request is provided to the user.

Abstract: Methods, systems, and computer-readable media for interactive voice response using a cloud-based service are disclosed. Voice input indicating interaction with an entity is received at a service provider environment. The entity is associated with an IVR data structure comprising one or more options associated with contacting the entity. The IVR data structure is selected based (at least in part) on analysis of the voice input by the service provider environment. The voice input is sent from a voice-capturing device to the service provider environment via a non-telephonic network. A contact point is selected from the IVR data structure based (at least in part) on analysis of the voice input and/or contextual information. A connection is initiated with the contact point.