Abstract: A voice quality preference learning device according to an embodiment includes a storage, a user interface system, and a learning processor. The storage stores a plurality of acoustic models. The user interface system receives an operation input indicating a voice quality preference of a user for voice quality. The learning processor learns a preference model corresponding to the voice quality preference of the user based at least in part on the operation input, the operation input associated with a voice quality space, wherein the voice quality space is obtained by dimensionally reducing the plurality of acoustic models.

Abstract: A computer-implemented method includes detecting a first set and a second set of citations to a legal case in a plurality of legal documents and a first legal document distinct from the plurality of legal documents, respectively. The computer-implemented method further includes determining tones corresponding to each citation in the first and second sets of citations. The computer-implemented method further includes determining a score for each tone in the first and second sets of tones. The computer-implemented method further includes aggregating a first and subset and a second of the first and second sets of citations, respectively. The computer-implemented method further includes generating an average score for the first and second subsets. The computer-implemented method further includes determining a degree of similarity between the first and second subsets based, in part, on a comparison of average scores. A corresponding computer program product and computer system are also disclosed.

Abstract: Aspects of the disclosure provide systems and methods for facilitating dictation. Speech input may be provided to an audio input device of a computing device. A speech recognition engine at the computing device may obtain text corresponding to the speech input. The computing device may transmit the text to a remotely-located storage device. A login webpage that includes a session identifier may be accessed from a target computing device also located remotely relative to the storage device. The session identifier may be transmitted to the storage device and, in response, a text display webpage may be received at the target computing device. The text display webpage may include the speech-derived text and may be configured to automatically copy the text to a copy buffer of the target computing device. The speech-derived text may also be provided to native applications at target computing devices or NLU engines for natural language processing.

Abstract: A speech extraction method based on the supervised learning auditory attention includes: converting an original overlapping speech signal into a two-dimensional time-frequency signal representation by a short-time Fourier transform to obtain a first overlapping speech signal; performing a first sparsification on the first overlapping speech signal, mapping intensity information of a time-frequency unit of the first overlapping speech signal to preset D intensity levels, and performing a second sparsification on the first overlapping speech signal based on information of the preset D intensity levels to obtain a second overlapping speech signal; converting the second overlapping speech signal into a pulse signal by a time coding method; extracting a target pulse from the pulse signal by a trained target pulse extraction network; converting the target pulse into a time-frequency representation of the target speech to obtain the target speech by an inverse short-time Fourier transform.

Abstract: Methods and apparatus are provided for improving wake-up word (or trigger word) detection by an audio device. After initially detecting a WUW, an audio device validates the detected WUW using inputs from one or more other systems such as voice activity detection (VAD), on-head detection, or other headphone data. Other headphone data includes inputs received via sensors on the audio device that provide contextual information associated with a state of the user. Based on the inputs from other systems, the audio device is able to identify unintended WUW activations and increase WUW detection accuracy.

Abstract: Utterance-based user interfaces can include activation trigger processing techniques for detecting activation triggers and causing execution of certain commands associated with particular command pattern activation triggers without waiting for output from a separate speech processing engine. The activation trigger processing techniques can also detect speech analysis patterns and selectively activate a speech processing engine.

Abstract: Techniques are disclosed for generating anomaly scores for a neuro-linguistic model of input data obtained from one or more sources. According to one embodiment, generating anomaly scores includes receiving a stream of symbols generated from an ordered stream of normalized vectors generated from input data received from one or more sensor devices during a first time period. Upon receiving the stream of symbols, generating a set of words based on an occurrence of groups of symbols from the stream of symbols, determining a number of previous occurrences of a first word of the set of words, determining a number of previous occurrences of words of a same length as the first word, and determining a first anomaly score based on the number of previous occurrences of the first word and the number of previous occurrences of words of the same length as the first word.

Abstract: A voice tonal control system is provided to achieve a target perceived cognitive state of a user's voice. For this purpose a computer-implemented method includes receiving, by a computer device, user input defining a target perceived cognitive state of a user's voice, determining, by the computer device, an actual perceived cognitive state of the user's voice based on cognitively analyzing a spoken sample of the user's voice, and providing, by the computer device, an alert in real time to the user based on the actual perceived cognitive state of the user's voice differing from the target perceived cognitive state of the user's voice.

Abstract: Systems and methods of network-based learning models for natural language processing are provided. Information may be stored information in memory regarding user interaction with network content. Further, a digital recording of a vocal utterance made by a user may be captured. The vocal utterance may be interpreted based on the stored user interaction information. An intent of the user may be identified based on the interpretation, and a prediction may be made based on the identified intent. The prediction may further correspond to a selected workflow.

Abstract: The present disclosure provides projection neural networks and example applications thereof. In particular, the present disclosure provides a number of different architectures for projection neural networks, including two example architectures which can be referred to as: Self-Governing Neural Networks (SGNNs) and Projection Sequence Networks (ProSeqoNets). Each projection neural network can include one or more projection layers that project an input into a different space. For example, each projection layer can use a set of projection functions to project the input into a bit-space, thereby greatly reducing the dimensionality of the input and enabling computation with lower resource usage. As such, the projection neural networks provided herein are highly useful for on-device inference in resource-constrained devices. For example, the provided SGNN and ProSeqoNet architectures are particularly beneficial for on-device inference such as, for example, solving natural language understanding tasks on-device.

Abstract: Disclosed herein are device, system and method embodiments for implementing a voice endpoint to chatbot bridge interface system. A bridge interface device operates by receiving query text corresponding to audio information captured at a voice endpoint, generating a bot agent request based on the query text and a bot agent associated with the query text, and sending the bot agent request to the bot agent. Further, the bridge interface device receives a bot agent response including response information associated with the query text, and sends a query response to the voice endpoint based on the bot agent response.

Abstract: A device monitors audio data for a predetermined and/or user-defined wakeword. The device detects an error in detecting the wakeword in the audio data, such as a false-positive detection of the wakeword or a false-negative detection of the wakeword. Upon detecting the error, the device updates a model trained to detect the wakeword to create an updated trained model; the updated trained model reduces or eliminates further errors in detecting the wakeword. Data corresponding to the updated trained model may be collected by a server from a plurality of devices and used to create an updated trained model aggregating the data; this updated trained model may be sent to some or all of the devices.

Abstract: Systems, methods, and computer program products of audio processing based on Adaptive Intermediate Spatial Format (AISF) are described. The AISF is an extension to ISF that allows spatial resolution around an ISF ring to be adjusted dynamically with respect to content of incoming audio objects. An AISF encoder device adaptively warps each ISF ring during ISF encoding to adjust angular distance between objects, resulting in increase in uniformity of energy distribution around the ISF ring. At an AISF decoder device, matrices that decode sound positions to the output speaker take into account the warping that was performed at the AISF encoder device to reproduce the true positions of sound sources.

Abstract: According to some aspects, disclosed methods and systems may include having a user input one or more speech commands into an input device of a user device. The user device may communicate with one or more components or devices at a local office or headend. The local office or the user device may transcribe the speech commands into language transcriptions. The local office or the user device may determine a mood for the user based on whether any of the speech commands may have been repeated. The local office or the user device may determine, based on the mood of the user, which content asset or content service to make available to the user device.

Abstract: One embodiment provides a computer program product for improving accuracy of a transcript of a spoken interaction. The computer program product comprises a computer readable storage medium having program instructions embodied therewith. The program instructions are executable by a processor to cause the processor to identify a plurality of patterns in the transcript. The plurality of patterns are indicative of a group of acoustically similar words in the transcript and a corresponding local, sequential context of the group of acoustically similar words. The program instructions are further executable by the processor to cause the processor to predict conditional probabilities for the group of acoustically similar words based on a predictive model and the plurality of patterns, detect one or more transcription errors in the transcript based on the conditional probabilities, and correct the one or more transcription errors by applying a multi-pass correction on the one or more transcription errors.

Abstract: A communication apparatus includes: a first type communication unit configured to perform communication with a portable device in a near field communication mode; a display unit; and a control device configured to perform: a receiving process of receiving a radio wave for connection with the portable device in the near field communication mode, from the portable device through the first type communication unit; and a display process of controlling the display unit to display a notice for prompting a user to perform operation for permitting the portable device to transmit information to the communication apparatus in the near field communication mode, in response to receipt of the radio wave in the receiving process.

Abstract: Methods and devices can enhance language processing in an autism spectrum disorder (ASD) individual through auditory manipulation of an auditory stream. The auditory stream is received and includes an acoustic stimulus perceptually representing an object. An acoustic manipulation parameter for a predetermined acoustic detail characteristic is selected. The predetermined acoustic detail characteristic is associated with the ASD individual and is based on a measured language processing capability of the ASD individual. The auditory stream is modified based on the selected parameter, to reduce the predetermined acoustic detail characteristic while preserving a lexicality of the stimulus, such that the reduced acoustic detail characteristic enhances perception of the object by the ASD individual even when the stimulus includes two or more acoustically distinct stimuli each perceptually representing the object. The modified auditory stream is output to the ASD individual via at least one loudspeaker.

Abstract: A method and a virtual agent system services a user request from a user. The virtual agent system includes: (a) a conversational user interface receiving the user request and communicating with two or more virtual agents; and (b) a dialog manager including a natural language processing module, that directs operations of the conversational user interface, wherein the dialog manager (i) receives and analyzes the user request from the conversation user interface using the natural language processing module, (ii) causes the conversational user interface to request and to receive a response to the user request from each of the virtual agents, and (iii) integrates the received responses to the user request into an integrated response based on the natural language processing module and causes the conversational user interface to provide the integrated response to the user.

Abstract: On a touch-panel display of an image forming apparatus, which is divided to five areas, that is, a system area, a function selection area, a preview area, an action panel area and a task trigger area, pieces of information are displayed. Even if an operational mode is switched, the same or similar information is always displayed in the area arranged at the same position. In the task trigger area, software buttons operated by the user for actually operating the image forming apparatus are displayed.

Abstract: Systems, methods, and apparatuses are disclosed for adaptively generating a summary of web-based content based on an attribute of a mobile communication device having transmitted a request for the web-based content. By adaptively generating the summary based on an attribute of the mobile communication device such as an amount of visual space available or a number of characters permitted in the interface, a display of the web-based content may be controlled on the mobile communication device in a way that was not previously available. This enables control of displaying web-based content that has been adaptively generated to be displayed on limited display screens based on a learned attribute of the mobile communication device requesting the web-based content.