Abstract: Telephone calls can be validated using blockchains. In one example, a computing node can generate a distributed ledger indicating relationships between telephone numbers and unique identifiers assigned to the telephone numbers. The distributed ledger can be accessible to a telecommunication system for validating telephone calls.

Abstract: Concepts and technologies are disclosed herein for communications network security for handling proxy voice calls that employ a voicebot. According to one aspect disclosed herein, a call handling system can intercept, from a communications network, a call request that is directed to a called target device. The call handling system can determine that the call request was generated by a voicebot on behalf of a user equipment. The call handling system can suspend the call request from being routed to the called target device. The call handling system can generate a voicebot confirmation request that identifies the voicebot and the user equipment. The call handling system can provide the voicebot confirmation request to the called target. The call request can be suspended while the voicebot confirmation request is provided to the called target device.

Abstract: A call analytics system and associated methods that can be used to rapidly analyze call data and provide conversational insights. The call analytics system receives audio call data of a phone call between a customer and an agent of a business, and converts the call data into one or more messages for handling by a distributed stream-processing platform. In some embodiments, the stream-processing platform is the Apache Kafka platform. The distributed platform processes the messages and communicates with various software modules to generate a variety of conversational insights. When processed by a stream-processing platform, certain analyses can occur in parallel which allows conversational insights to be provided to the businesses shortly (e.g., within seconds) after the call data is received.

Abstract: Post-filtering may be performed for ratio masks as part of audio enhancement. Audio data may be received. A machine learning model may be applied to generate gain values for different spectrum bands of the audio data. The gain values may then be modified using an envelope post-filter according to a monotonically increasing function applied to the gain values to produce modified gain values used to generate an enhanced version of the audio data.

Abstract: Aspects of the disclosure relate to voiceprint tracking and anomaly detection. A computing platform may detect voice information from a call management system. The computing platform may establish voiceprints for employees and clients of an enterprise. The computing platform may detect a call between an employee and a caller attempting to access a client account. The computing platform may identify a first voiceprint corresponding to the employee and a second voiceprint corresponding to the caller. The computing platform may compare the second voiceprint to a known voiceprint corresponding to the client. Based on the comparison of the second voiceprint to the known voiceprint, the computing platform may determine that the second voiceprint does not match the known voiceprint. The computing platform may identify that the second voiceprint corresponds to another employee of the enterprise, and may send a security notification indicating potential unauthorized account access to an enterprise computing device.

Abstract: A wearable electronic device (WED) includes one or more sensors and cameras that determine a location of a physical object in a zone where the user is located and that track movement of an electronic device that moves to define a boundary of the zone. The WED includes a processor that generates binaural sound and a display that displays a virtual image of the boundary of the zone and a visual warning that notifies the user of the physical object.

Abstract: The present disclosure discloses an interactive voice-control method and apparatus, a device and a medium. The method includes: obtaining a sound signal at a voice interaction device and recognized information that is recognized from the sound signal; determining an interaction confidence of the sound signal based at least on at least one of an acoustic feature representation of the sound signal and a semantic feature representation associated with the recognized information; determining a matching status between the recognized information and the sound signal; and providing the interaction confidence and the matching status for controlling a response of the voice interaction device to the sound signal.

Abstract: The example embodiments provides a processing system (10) including: an acquisition unit (11) that acquires target speech data in which a target speech is recorded or a target feature value that indicates a feature of the target speech; an inference unit (12) that infers a language of the target speech, based on an inference model for inferring a language of a speech from speech data or a speech feature value and the target speech data or the target feature value; a result output unit (13) that outputs an inference result by the inference unit (12); a determination unit (14) that determines whether the inference result is correct; and a learning data output unit (15) that outputs the inference result determined to be correct by the determination unit (14) and the target speech data or the target feature value, as learning data for generating the inference model.

Abstract: Disclosed is a voice message exchange system and method for improving communication between an inmate and a third party by enabling the inmate to leave a message when a call is not answered and further allowing the third party who receives the message to reply with a message to the inmate. Additionally, outside parties that meet the institution's requirements can leave messages for inmates at any time. The present invention can be used as an add-on to legacy inmate call management systems or incorporated internally into an inmate call management system. The system also provides monitoring, controlling, recording, and billing means.

Abstract: Methods and systems of authoring audio signal(s) into haptic data file(s) are disclosed. An audio analysis module analyses the audio signal(s) using filterbank(s) or by performing a spectrogram analysis. Transients are detected in the audio signal. If present, the transients are processed to determine a transient score and a transient binary. A database stores device specific information and actuator specific information. A haptic perceptual bandwidth of an electronic computing device having an embedded actuator is determined by using information from the database. A user interface allows modification of time-amplitude values and transient values based on the determined haptic perceptual bandwidth. Authored time amplitude values are aggregated in authored audio descriptor data, which is passed to a transformation module that fits the data into the haptic perceptual bandwidth and implements algorithms to produce transformed audio descriptor data.

Abstract: Methods, systems, and apparatus for an automated calling system are disclosed. Some implementations are directed to using a bot to initiate telephone calls and conduct telephone conversations with a user. The bot may be interrupted while providing synthesized speech during the telephone call. The interruption can be classified into one of multiple disparate interruption types, and the bot can react to the interruption based on the interruption type. Some implementations are directed to determining that a first user is placed on hold by a second user during a telephone conversation, and maintaining the telephone call in an active state in response to determining the first user hung up the telephone call. The first user can be notified when the second user rejoins the call, and a bot associated with the first user can notify the first user that the second user has rejoined the telephone call.

Abstract: The method for adapting a sound converter to a reference sound converter includes the sound converter having a first linear transfer function with a first frequency response, a second linear transfer function with a second frequency response, and a trivial nonlinearity. The sound converter has a non-linear transfer function corresponding to the frequency response from combination of the first linear transfer function, the trivial nonlinearity, and the second linear transfer function. A first frequency spectrum of the reference sound converter is determined at a low input level. A second frequency spectrum of the reference sound converter is determined at a high input level. The second determined frequency spectrum is used as the second frequency response in the second linear transfer function, and the division of the first frequency spectrum by the second frequency spectrum is used as the first frequency response in the first linear transfer function.

Abstract: Methods for improved fraudster detection in a call center. A subset of a plurality of voiceprints from a plurality of interactions between callers and agents at a call center can be used as the basis for fraudster detection. A plurality of connected components that represents one or more voiceprints can be determined based on the subset of the plurality of voiceprints.

Abstract: A call screening computing system is described that is configured to perform voice captcha and real-time monitoring of calls into a contact center of an organization. The call screening computing system includes a chat bot configured to operate as an AI-based call screener. The chat bot is configured to perform voice captcha by sending a random question to a user device placing a call into the contact center, and analyzing the received answer to determine whether a user of the user device is human or a robot. The chat bot is configured to, based on the user being human, determine whether the user is a legitimate customer of the organization by generating and presenting authentication challenges to the user device. The chat bot may be configured to monitor and interact with a conversation between the user and an agent of the organization during the call into the contact center.

Abstract: Introduced here is a platform for actively managing communication channels. In particular, the communication channels are managed to increase the probability of success of a call center agent to meaningfully engage with a client. In addition, the platform provides identification information and other information to facilitate a conversation between the parties, while also reducing the pre-call preparation by an agent. The platform parses search results to identify contact information for prospect by utilizing keyword identification algorithm. During the call, the platform applies voice detection and speaker detection algorithms to ensure that the agent is speaking with a live voice, rather than a pre-recorded message. By applying these algorithms, the platform aims to increase call center efficiency and conversation rates, while reducing pre-call preparation and skepticism of a prospect.

Abstract: Various embodiments are disclosed for (possibly simultaneously) applying EQ and DRC to audio signals. In an embodiment, a method comprises: dividing an input audio signal into n frames, where n is a positive integer greater than one; dividing each frame of the input audio signal into Nb frequency bands, where Nb is a positive integer greater than one; for each frame n: computing an input level of the input audio signal in each band f, resulting in a input audio level distribution for the input audio signal; computing a gain for each band f based at least in part on a mapping of one or more properties of the input audio level distribution to a reference N audio level distribution computed from one or more reference audio signals; and applying each computed gain for each band f to each corresponding band f of the input audio signal.

Abstract: [Problem] Provided is a relaying device that can track and record a communicated voice of a specific communication terminal using a mixing function of the relaying device. [Solution] When a communication terminal has sent a call packet to call another at least one communication terminal, a communication session in which this communication terminal that has made a call and a communication terminal that has been called are participating terminals is established. In the communication session, voice communication is relayed by transferring a voice signal which is transmitted from the participating terminals to another at least one communication terminal. A virtual device is registered as a participating terminal of the communication session and the voice signal which is transmitted from the communication session to the virtual device is recorded in the relaying device.

Abstract: Data analysis, filter, and presentation techniques are described for an example call management system. An example method for a data management system includes receiving, from a user device, an account identifier of a first person, receiving, from the user device, a first message related to a topic of conversation to be discussed during a telephone call with the first person, determining, based on the account identifier and within a pre-determined time period, a presence of a set of data that describes conversation(s) from prior telephone call(s)/chat(s) with the first person, and sending, to a computer, at least some data from the set of data and a second message that indicates that the telephone call is mapped to a second person, where the computer is configured to display on a screen the second message and a presentation option that presents the at least some data via the computer.

Abstract: A method for detecting an audio input includes: acquiring audio input signals received by at least two input signal channels of an audio input module; for each of the audio input signals, filtering the audio input signal according to a preset audio output signal of an electronic device where the audio input module is located to obtain a target signal; for each of the audio input signals, determining a comparison parameter value according to the target signal and the audio input signal; and determining a performance state of the audio input module according to the comparison parameter values.

Abstract: The present invention provides a display device and a television device with which it is possible to reduce the thickness of the device profile while also realizing desirable acoustic characteristics. This display device includes: a first plate-shaped object; a display module including a display element and a second plate-shaped object that supports the display element, the display module being disposed so as to face the first plate-shaped object; a liquid layer formed by sealing up a liquid between the first plate-like object and the display module; and a vibrator provided to the first plate-shaped object and/or the display module. The loss coefficient of the display device at 25° C. is 1×10?2 or greater and a thickness ratio between the first plate-shaped object and the display module is from 1:10 to 10:1.