Abstract: Methods and systems for leveraging a topic divergence model to generate dynamic sidebar chat conversations are disclosed. A method includes: collecting, by a computing device, data from a first group chat; determining, by the computing device, a main topic of the first group chat and a current topic of the first group chat using the data collected from the first group chat; determining, by the computing device, a first divergence score representing a degree of divergence between the main topic of the first group chat and the current topic of the first group chat; determining, by the computing device, that the first divergence score exceeds a predetermined threshold; and in response to determining that the first divergence score exceeds the predetermined threshold, the computing device creating a second group chat; determining an efficacy based on an emotive analysis which increases as the divergence score decreases.

Abstract: Example embodiments of systems and methods for providing a video call between a first party and a second party. The method includes detecting an attempt to establish the video call by monitoring a voice call between the first party and the second party; authenticating the first and second parties; receiving from the first and second parties respective voice acceptances to invitations to establish the video call; retrieving video call identifications of the first and second parties; and establishing the video call without revealing the video call identification of the first party to the second party and the video call identification of the second party to the first party. A server uses credentials of a first party to request if that party would allow access of his identifying information to a second party and use that credential to further authenticate the first party.

Abstract: Systems and methods are provided for improving audio processing by receiving an external input sound message during playing a first audio message; matching the external input sound message with a receiving message to obtain a matching result, wherein the receiving message is associated with the first audio message in content, wherein the matching is based on a proportion of sound units in the sound message that hit sound units in the receiving message; determining whether the matching result meets a threshold; and upon determining that the matching result meets the threshold, stop playing the first audio message.

Abstract: Described herein are methods, computer-readable media, techniques, and methodologies for collaborative control of an artificial intelligence (AI) voice response system and an AI voice response system configured for collaborative interaction and control. A user command submitted by a first user to an AI voice response client device is evaluated against a user profile associated with the first user to determine whether the user command requires authorization from a second user prior to execution. If authorization is required, the user command or those portion(s) of the user command requiring authorization are sent to a user device of the second user. The second user is provided with the capability to approve, reject, or modify the user command. If approved or modified, the user command or the modified command are executed and the execution result is presented to the first user via the AI voice response client device or another device.

Abstract: A computer-implemented system for improving computerized analysis and grading for one or more populated sentence structure diagrams is shown. The system includes a sentence diagramming interface which in turn includes a natural language input for receiving a natural language sentence to be diagrammed, a sentence construction toolbox including a plurality of sentence construction components used in generating a constructed sentence structure, and a sentence construction workspace configured to receive and allow manipulation of at least one selected sentence construction component to generate the constructed sentence structure and receive and allow manipulation of each word in the natural language sentence such that each word is assigned to a position in the constructed sentence structure to generate a constructed and populated sentence structure.

Abstract: An apparatus for processing an encoded audio signal, which includes a sequence of access units, each access unit including a core signal with a first spectral width and parameters describing a spectrum above the first spectral width, has a demultiplexer generating, from an access unit of the encoded audio signal, the core signal and a set of the parameters, an upsampler upsampling the core signal of the access unit and outputting a first upsampled spectrum and a timely consecutive second upsampled spectrum, the first upsampled spectrum and the second upsampled spectrum, both, having a same content as the core signal and having a second spectral width being greater than the first spectral width of the core spectrum, a parameter converter converting parameters of the set of parameters of the access unit to obtain converted parameters, and a spectral gap filling processor processing the first upsampled spectrum and the second upsampled spectrum using the converted parameters.

Abstract: A system of reducing transmissions of packetized data in a voice activated data packet based computer network environment is provided. A natural language processor component can parse an input audio signal to identify a request and a trigger keyword. Based on the input audio signal, a direct action application programming interface can generate a first action data structure, and a content selector component can select a content item. An interface management component can identify candidate interfaces and determine if prior instances of the packetized data was transmitted to the candidate interfaces. The interface management component can prevent the transmission of the packetized data if determined to be redundant, such as having previously received the data, and instead transmit it to a separate client device of a different device type.

Abstract: A dynamic speech processing system and method is provided. The system includes a receiver configured to receive a plurality of audio files. The audio files include sample training audio files and run-time audio files. The system further includes a speech processor coupled to the receiver and configured to compute a variable value for a specific audio file. The speech processor is configured to dynamically select a set of relevant speech recognition engines for a specific run-time audio file based on the variable value.

Abstract: Disclosed is a system for context-based abbreviation disambiguation, the system comprising: an ontological databank represented into a multi-dimensional space, a synonym databank, a glossary databank, and a server arrangement. The server arrangement is configured to obtain a text comprising abbreviations and concept phrases, extract a target abbreviation from the abbreviations, obtain potential expansions for the target abbreviation, calculate a synonym match score for potential expansions, using synonyms of the potential expansions and the concept phrases, calculate a concept match score, using concepts relating to the potential expansions and the concept phrases, calculate a context match score for the potential expansions using a comparison module, and determine one of the potential expansions as a valid expansion of the target abbreviation based on at least one of the: synonym match score, concept match score and context match score.

Abstract: A method of providing real-time speech analysis to a user includes capturing a speech input, performing a real-time recognition of the speech input including converting the speech input to a text, analyzing the recognized speech input to identify an error in a voice of the user, the analyzing including comparing a voice of a correct text generated by an automated speech generation system with the captured speed input, and processing the text to extract a context dialog prompt.

Abstract: A method comprising receiving first data representative of a query received by an electronic device. Device characteristic data indicative of at least one characteristic of the electronic device is received. It is determined, using the device characteristic data, that the electronic device is authorized to access a first portion of at least one knowledge graph, which is an access-controlled portion. The at least one knowledge graph also includes a second portion which is a non-access-controlled portion. The first data is sent to at least the first portion of the at least one knowledge graph. Second data is received from the first portion of the at least one knowledge graph. The second data is representative of an answer to the query. Answer data representative of the answer to the query is generated using the second data.

Abstract: Embodiments of the present invention provide a system that that can be used to determine whether a sentiment analysis model is portable between two data sets. During operation, the system analyzes the text of a respective review in a data set (e.g., a set of reviews) using the sentiment analysis model to determine a sentiment expressed in the review. The system then computes a confidence score, which indicates the accuracy of a respective sentiment. The system subsequently determines a confidence score distribution for various sentiments, as determined by the sentiment analysis model. The system determines the significance of changes between the confidence score distribution and a benchmark confidence score distribution, which is associated with a benchmark data set for which the sentiment analysis model yields a high accuracy. The system can then determine whether the sentiment analysis model is portable to the data set based on the significance of changes.

Abstract: An audio playback device receives an instruction from a user to select a target voice model from a plurality of voice models and assigns the target voice model to a target character in a text. The audio playback device also transforms the text into a speech, and during the process of transforming the text into the speech, transforms sentences of the target character in the text into the speech of the target character according to the target voice model.

Abstract: A robot apparatus including an input unit to receive a voice command from a user, a determination unit to determine whether a voice command is repeated a predetermined number of times, and a control unit to register a shortcut command to shorten a voice command if it is determined a voice command is repeated a predetermined number of times. A shortcut command to shorten a voice command of a user is generated, and thus user convenience is enhanced.

Abstract: The present disclosure discloses a method for processing a voice in interior environment of a vehicle, an electronic device and a storage medium. The method includes the following. A reference audio is acquired, and the reference audio is recorded to obtain a recorded audio. A pure voice is acquired. Noise data for each part or period of the recorded audio satisfying a target signal-to-noise ratio condition pertaining to that part is selected from the recorded audio, and the noise data is superimposed to the pure data to obtain a noisy voice. The noisy voice and the reference audio are inputted to an acoustic echo canceller (AEC) module as inputted data. The AEC module is configured to perform an echo cancellation operation on the inputted data to obtain training data having AEC residual noise.

Abstract: A method for determining whether a text complies with a query, an apparatus and a computer program product, the method comprising: obtaining a macro comprising a plurality of representations of a query; and executing the macro against a database comprising a simplified representation of at least one phrase within the text, to determine 10 whether the text complies with any of the representations of the query. The simplified representation is presented in the form of: conferrer (carrier; bearer; substrate; descriptor) and provides information pertaining to who, what, when and how with respect to the text associated with the simplified representation.

Abstract: Techniques are disclosed for generating ASR training data. According to an embodiment, impactful ASR training corpora is generated efficiently, and the quality or relevance of ASR training corpora being generated is increased by leveraging knowledge of the ASR system being trained. An example methodology includes: selecting one of a word or phrase, based on knowledge and/or content of said ASR training corpora; presenting a textual representation of said word or phrase; receiving a speech utterance that includes said word or phrase; receiving a transcript for said speech utterance; presenting said transcript for review (to allow for editing, if needed); and storing said transcript and said audio file in an ASR system training database. The selecting may include, for instance, selecting a word or phrase that is under-represented in said database, and/or based upon an n-gram distribution on a language, and/or based upon known areas that tend to incur transcription mistakes.

Abstract: Systems, methods, and computer-readable media are disclosed for voice-based content searching. The systems, methods, and computer-readable media described herein may improve user experiences by providing voice-based content searching using mobile devices. In an example method described herein, a computer system may receive a voice search initiation signal from a mobile device over a WiFi protocol connection or a Bluetooth protocol connection. The computer system may monitor a WiFi receiver and a Bluetooth receiver of the computer system for first voice data. The computer system may determine that the first voice data is being received at the WiFi receiver. The computer system may determine that second voice data is received at the Bluetooth receiver while the first voice data is being received at the WiFi receiver, and discard the second voice data.

Abstract: A signal processor comprising: an input terminal, configured to receive an input-signal; a voicing-terminal, configured to receive a voicing-signal representative of a voiced speech component of the input-signal; an output terminal; a delay block, configured to receive the input-signal and provide a filter-input-signal as a delayed representation of the input-signal; a filter block, configured to: receive the filter-input-signal; and provide a noise-estimate-signal by filtering the filter-input-signal; a combiner block, configured to: receive a combiner-input-signal representative of the input-signal; receive the noise-estimate-signal; and combine the combiner-input-signal with the noise-estimate-signal to provide an output-signal to the output terminal; and a filter-control-block, configured to: receive the voicing-signal; receive signalling representative of the input-signal; and set filter coefficients of the filter block in accordance with the voicing-signal and the input-signal.

Abstract: A method, performed by a dialogue processing device, of processing dialogue associated with a user based on dialog act information, the method comprises receiving speech information, corresponding to speech of the user, including a plurality of sentence units; identifying a first sentence unit and a second sentence unit, of the plurality of sentence units, based on receiving the speech information; extracting a first dialog act indicative of an intention of the first sentence unit and extracting a second dialog act indicative of an intention of the second sentence unit; extracting a first dialog act indicative of an intention of the first sentence unit and extracting a second dialog act indicative of an intention of the second sentence unit; processing the first sentence unit and the second unit in a sequence according to respective priority orders assigned based on a number of empty slots of dialogue frames of the sentence units.