Abstract: The present disclosure provides an information processing system and an information processing method capable of auditing the utterance data of an agent more flexibly. In one example, an information processing system includes: a storage section that stores utterance data of an agent; a communication section that receives request information transmitted from a client terminal and requesting utterance data of a specific agent from a user; and a control section that, when the request information is received through the communication section, replies to the client terminal with corresponding utterance data, and in accordance with feedback from the user with respect to the utterance data, updates an utterance probability level expressing a probability that the specific agent will utter utterance content indicated by the utterance data, and records the updated utterance probability level in association with the specific agent and the utterance content in the storage section.

Abstract: An electronic apparatus which registers a device to a server by using a voice, and a method therefor are provided. The electronic apparatus includes a communication circuit, a microphone, a memory for storing computer executable instructions, and at least one processor configured to execute the computer executable instructions to acquire, from a voice received through the microphone, information on an external device which a user wishes to register, based on an external device corresponding to the acquired information being searched through the communication circuit, control the communication circuit to transmit information on an access point to the external device to enable the external device to communicate with a server, and control the communication circuit to transmit a registration request with respect to the external device to the server.

Abstract: A system and method of a telephony application platform can include receiving a communication session request specifying a destination endpoint, connecting the communication session through a first application module according to the destination endpoint, receiving an instruction to transition communication control to a second application module, transitioning communication control of the communication session to the second application module, independently metering resource usage of the communication session of each application module, and accounting the metered usage of the communication session.

Abstract: Provided is an in-ear device and associated computational support system that leverages machine learning to interpret sensor data descriptive of one or more in-ear phenomena during subvocalization by the user. An electronic device can receive sensor data generated by at least one sensor at least partially positioned within an ear of a user, wherein the sensor data was generated by the at least one sensor concurrently with the user subvocalizing a subvocalized utterance. The electronic device can then process the sensor data with a machine-learned subvocalization interpretation model to generate an interpretation of the subvocalized utterance as an output of the machine-learned subvocalization interpretation model.

Abstract: Audio objects are associated with positional metadata. A received downmix signal comprises downmix channels that are linear combinations of one or more audio objects and are associated with respective positional locators. In a first aspect, the downmix signal, the positional metadata and frequency-dependent object gains are received. An audio object is reconstructed by applying the object gain to an upmix of the downmix signal in accordance with coefficients based on the positional metadata and the positional locators. In a second aspect, audio objects have been encoded together with at least one bed channel positioned at a positional locator of a corresponding downmix channel. The decoding system receives the downmix signal and the positional metadata of the audio objects. A bed channel is reconstructed by suppressing the content representing audio objects from the corresponding downmix channel on the basis of the positional locator of the corresponding downmix channel.

Abstract: An intercom device and a method of recording an audio stream at the intercom device. The intercom device includes a user interface, a memory, and an electronic processor. The electronic processor is communicatively coupled to the user interface and the memory and is configured to receive a user-selectable entry from an input mechanism of the user interface. The electronic processor sets an away mode of the intercom device in response to the user-selectable entry. An audio stream is received by the intercom device from another intercom device. Are cord signal indicative of a request to record the audio stream is also received from the another intercom device. The intercom device records the audio stream in the memory in response to receiving the record signal when operating in the away mode.

Abstract: A method includes receiving a designated event related to a second application while an execution screen of a first application is displayed on a display. The method also includes executing an artificial intelligent application in response to the designated event. The method further includes transmitting data related to the designated event to an external server, based on the executed artificial intelligent application. Additionally, the method includes sensing a user utterance related to the designated event for a designated period of time. The method also includes transmitting the user utterance to the external server. The method further includes receiving an action order for performing a function related to the user utterance from the external server. The method also includes executing the second application at least based on the received action order. The method further includes outputting a result of performing the function by using the second application.

Abstract: The present application provides a method of manufacturing a resonant panel (200) of a flat panel loudspeaker. The method comprises: pressing a resonant panel blank between a first pressing surface (302) and a second pressing surface of a press, whereby to form the resonant panel (200) of the flat panel loudspeaker. The second pressing surface substantially opposes the first pressing surface (302). The first pressing surface (302) comprises at least one tool relief region (306, 312, 314, 316, 318), whereby to form at least one corresponding respective panel relief region (206, 212, 214, 216, 218) in a surface of the resonant panel (200).

Abstract: Due to the high language use variability in real-life, manual construction of semantic resources to cover all synonyms is prohibitively expensive and may result in limited coverage. Described herein are systems and methods that automate the process of synonymy resource development, including both formal entities and noisy descriptions from end-users. Embodiments of a multi-task model with hierarchical task relationship are presented that learn more representative entity/term embeddings and apply them to synonym prediction. In model embodiments, a skip-gram word embedding model is extended by introducing an auxiliary task “neighboring word/term semantic type prediction” and hierarchically organize them based on the task complexity. In one or more embodiments, existing term-term synonymous knowledge is integrated into the word embedding learning framework.

Abstract: A method for computer control of online social interactions based on conversation processing. A portion of a conversation with a given user is recorded, stored in memory, and processed so as to influence subsequent interactions. These may include audio or other contextualized annunciations of subsequent telephone calls. Other modes addressing conversational processing for social purposes can result in interwoven conversation guided among threads of interest, advertising and incented participation of conversational content and placement. The invention is capable of implementation in telecommunications systems such as cellular, local exchange, and VOIP, and in combination with other forms of internet-based telecommunication, including smart phones and adaptive forums chat rooms.

Abstract: A hypothesis stitcher for speech recognition of long-form audio provides superior performance, such as higher accuracy and reduced computational cost. An example disclosed operation includes: segmenting the audio stream into a plurality of audio segments; identifying a plurality of speakers within each of the plurality of audio segments; performing automatic speech recognition (ASR) on each of the plurality of audio segments to generate a plurality of short-segment hypotheses; merging at least a portion of the short-segment hypotheses into a first merged hypothesis set; inserting stitching symbols into the first merged hypothesis set, the stitching symbols including a window change (WC) symbol; and consolidating, with a network-based hypothesis stitcher, the first merged hypothesis set into a first consolidated hypothesis.

Abstract: A system and method are disclosed for generating a teleconference space for two or more communication devices using a computer coupled with a database and comprising a processor and memory. The computer generates a teleconference space and transmits requests to join the teleconference space to the two or more communication devices. The computer stores in memory identification information, and audiovisual data associated with one or more users, for each of the two or more communication devices. The computer stores audio transcription data, transmitted to the computer by each of the two or more communication devices and associated with one or more communication device users, in the computer memory. The computer merges the audio transcription data from each of the two or more communication devices into a master audio transcript, and transmits the master audio transcript to each of the two or more communication devices.

Abstract: Methods and devices for aggregating hardware loopback streams of a plurality of display devices in communication with a computer device may include a plurality of hardware loopback streams with rendered audio data from the plurality of display devices in communication with the computer device. The methods and devices may include combining the rendered audio data from the plurality of hardware loopback streams into a loopback buffer to create aggregated loopback audio data. The methods and devices may include providing the loopback buffer with the aggregated loopback audio data to one or more applications executing on the computer device.

Abstract: Attributes of electronic content from a meeting are identified and evaluated to determine whether sub-portions of the electronic content should or should not be attributed to a user profile. Upon determining that the sub-portion should be attributed to a user profile, attributes of the sub-portion of electronic content are compared to attributes of stored user profiles. A probability that the sub-portion corresponds to at least one stored user profile is calculated. Based on the calculated probability, the sub-portion is attributed to a stored user profile or a guest user profile.

Abstract: A pharmacy management system for automated sig code translation using machine learning includes a processor and a memory storing instructions that, when executed by the one or more processors, cause the pharmacy management system to train a machine learning model to analyze sig codes, receive a sig code, analyze the sig code utterance and generate an output corresponding to the sig code utterance. A computer-implemented method includes training a machine learning model to analyze sig codes, receiving a sig code, analyzing the sig code utterance, and generating an output corresponding to the sig code utterance. A non-transitory computer readable medium containing program instructions that when executed, cause a computer to: train a machine learning model to analyze sig codes, receive a sig code, analyze the sig code utterance, and generate an output corresponding to the sig code utterance.

Abstract: A vehicular apparatus having at least one of a voice calling function and a voice recognition function, the apparatus comprising a voice input unit including a plurality of microphones, the voice input unit being disposed between a driver's seat and a passenger seat with respect to a vehicle width direction; and a control unit configured to control a directionality direction and a gain level of each of the plurality of microphones, wherein the control unit controls the directionality directions of the plurality of microphones in two directions, the two directions being a driver's seat side and a passenger seat side, and controls a gain level on the passenger seat side to be lower than a gain level on the driver's seat side.

Abstract: Techniques are described for generating a custom voice for a virtual agent. In one implementations, a method includes receiving information identifying a customer contacting a call center. The method includes selecting a voice for a virtual agent based on information about the customer. The method also includes assigning the voice to the virtual agent during communications with the customer.

Abstract: Voice biometrics scoring is performed on received asynchronous audio outputs from microphones distributed at ad hoc locations to generate confidence scores that indicate a likelihood of an enrolled user speech utterance in the output, a subset of the outputs is selected based on the confidence scores, and the subset is spatially processed to provide audio output for voice application use. Alternatively, asynchronous spatially processed audio outputs and corresponding biometric identifiers are received from corresponding devices distributed at ad hoc locations, audio frames of the outputs are synchronized using the biometric identifiers, and the synchronized frames are coherently combined. Alternatively, uttered speech associated with respective ad hoc distributed devices is received and non-coherently combined to generate a final output of uttered speech.

Abstract: A system and method may identify a fraud ring based on call or interaction data by analyzing by a computer processor interaction data including audio recordings to identify clusters of interactions which are suspected of involving fraud each cluster including the same speaker; analyzing by the computer processor the clusters, in combination with metadata associated with the interaction data, to identify fraud rings, each fraud ring describing a plurality of different speakers, each fraud ring defined by a set of speakers and a set of metadata corresponding to interactions including that speaker; and for each fraud ring, creating a relevance value defining the relative relevance of the fraud ring.

Abstract: A real-time messaging system platform receives from, a communication application installed on a first user communication device, a first user identifier, an indication that a messaging service message composed by the first user is being directed to a first destination associated with a second user, and an identification of data present in the messaging service message. The identification of the data and a message transmission history of the first user is used to enable selection of a message of a first entity. The selected message of the first entity is caused to be displayed by the communication application within a message framework that frames the first user messaging service message. The first entity message is caused to be included in the messaging service message transmitted to the second user device, wherein the first entity is not provided with access to the identity of the first user or the second user.