Abstract: Various embodiments relating to detecting at least one of conversation, the presence and the identity of others during presentation of digital content on a computing device. When another person is detected, one or more actions may be taken with respect to the digital content. For example, the digital content may be minimized, moved, resized or otherwise modified.

Abstract: A voice recognition device extracts, from a first voice signal of a user, a first string of phonemes included in the first voice signal, extracts, from a second voice signal of the user, a second string of phonemes included in the second voice signal, extracts a string of common phonemes from the first string and the second string, calculates, for each of a plurality of registered keywords, a degree of similarity between a string of phonemes corresponding to the keyword and the string of common phonemes, and selects, among the plurality of keywords, a prescribed number of keywords based on the degree of similarity for each keyword.

Abstract: Disclosed herein are embodiments of systems, methods, and products comprises an authentication server for authentication leveraging multiple audio channels. The server receives an authentication request regarding a user upon the user interacting with a first electronic device. The server requests the first device to transmit a first audio file of an audio sample to the server. The audio sample may be the user's audio command or a machine-generated audio signal. The server requests a second electronic device to transmit a second audio file that is the recording of the same audio sample to the server. The second electronic device is a trusted device in proximity of the first device and executes an authentication function to enable the recording and transmitting of the audio sample. The server determines a similarity score between the first audio file and the second audio file and authenticates the user based on the similarity score.

Abstract: A method to interactively convert a source language video/audio stream into one or more target languages in high definition video format using a computer. The spoken words in the converted language are synchronized with synthesized movements of a rendered mouth. Original audio and video streams from pre-recorded or live sermons are synthesized into another language with the original emotional and tonal characteristics. The original sermon could be in any language and be translated into any other language. The mouth and jaw are digitally rendered with viseme and phoneme morphing targets that are pre-generated for lip synching with the synthesized target language audio. Each video image frame has the simulated lips and jaw inserted over the original. The new audio and video image then encoded and uploaded for internee viewing or recording to a storage medium.

Abstract: The disclosed computer-implemented method for detecting suspicious voice calls may include (i) identifying an incoming voice call, (ii) extracting, from audio of the incoming voice call, a plurality of characteristics, (iii) calculating a trustworthiness score of the plurality of the characteristics based on a response by a recipient of the incoming voice call, and (iv) storing the trustworthiness score of the plurality of characteristics in a reputation database that (a) receives a request for the trustworthiness score, the request originating from an additional computing device and including an additional plurality of characteristics extracted from an additional incoming voice call, (b) determines that the additional plurality of characteristics matches the plurality of characteristics, and (c) enables the additional computing device to perform a security action on the additional incoming voice call by sending the trustworthiness to the additional computing device.

Abstract: Provided is a target speech signal extraction method for robust speech recognition including: (a) receiving information on a direction of arrival of the target speech source with respect to the microphones; (b) generating a nullformer by using the information on the direction of arrival of the target speech source to remove the target speech signal from the input signals and to estimate noise; (c) setting a real output of the target speech source using an adaptive vector w(k) as a first channel and setting a dummy output by the nullformer as a remaining channel; (d) setting a cost function for minimizing dependency between the real output of the target speech source and the dummy output using the nullformer by performing independent component analysis (ICA); and (e) estimating the target speech signal by using the cost function, thereby extracting the target speech signal from the input signals.

Abstract: Mechanisms are provided for clarifying homophone usage in natural language content. The mechanisms analyze natural language content to identify a homophone instance in the natural language content, the homophone instance being a first term having a first definition and a first pronunciation for which there is a second term having the first pronunciation and a second definition different from the first definition. The mechanisms, in response to identifying the homophone instance, analyze the natural language content to identify a third term that is a synonym for the second term. The third term has a third definition that is nearly the same as the second definition. The mechanisms, in response to the natural language content comprising the third term, perform a clarifying operation to modify the natural language content to clarify the homophone instance and generate a modified natural language content.

Abstract: A method and system of providing a portable voice-based control user interface for multiple types of appliances are disclosed. The method includes activating a built-in voice communication interface of a voice control apparatus; selecting a first target appliance to receive voice-based commands; receiving a first voice input; in accordance with a determination that the first target appliance is a first appliance of a first appliance type, processing the first voice input using a first NLP model corresponding to the first appliance type to obtain a first machine command, and sending the first machine command to the first appliance; and in accordance with a determination that the first target appliance is a second appliance of a second appliance type, processing the first voice input using a second NLP model corresponding to the second appliance type to obtain a second machine command, and sending the second machine command to the second appliance.

Abstract: Provided is a technology which improves reliability of the interaction between devices in a system where the devices communicate. In an information appliance system, multiple information appliances and a communication device such as a smart phone are in M2M communication. The communication device receives input of a voice from a user, and authenticates the user based on a voice signal, and sample voice data accumulated for user identification. The communication device performs a speech recognition process on the voice signal to determine an instruction of the user. When the user is authenticated and the instruction of the user is determined, the communication device transmits a command according to the instruction to an information appliance that is associated with the determined instruction of the user.

Abstract: An interference filtering method applied to the voice commands of a user of a device includes audio acquisition unit of device taking a first audio signal including user voice from the environment and a second audio signal from an audio output unit of a device creating competing noise. A first background audio signal is obtained by filtering a speech sound region in first audio signal, and a second background audio signal is obtained by filtering a speech sound region in second audio signal. A time difference T and a sound amplified parameter X are obtained by comparison. A third audio signal is obtained by performing time compensation, amplification, and an inverting operation on second audio signal. First audio signal and third audio signal are synthesized to produce fourth audio signal for feeding to voice recognition unit of the original user device.

Abstract: One or more embodiments present positional information associated with a text or music to a user. In one embodiment, a determination is made that at least one line from a digital representation of text or music has been selected. Another determination is made that the line is associated with a set of positional information. The set of positional information is presented on a digital representation of a venue along with the presentation of the line of text or music.

Abstract: The present teaching relates to obtaining information from a user via a bot. In one example, a request is obtained to collect information in connection with a user. A statement is generated to be expressed to the user for facilitating a conversation between the user and the bot based on the request. Information is received in connection with the user and collected during the conversation. The collected information characterizes the user in a plurality of modalities. The collected information is automatically analyzed in the plurality of modalities to obtain an assessment of one or more human traits of the user. A plurality of result summaries are generated based on the assessment. The plurality of result summaries are provided in response to the request.

Abstract: A Mobile Voice Self Service (MVSS) mobile system that includes an MVSS mobile device, on which a VoiceXML browser is implemented directly. The VoiceXML browser may request a VoiceXML application from a VoiceXML application server and process it. A client system may include the VoiceXML application server that the VoiceXML application is requested from. Upon request, the VoiceXML application may deliver the requested VoiceXML application to the VoiceXML application browser. A vendor media resource system may provide advanced Media Resource Control Protocol (MRCP) services, such as Automatic Speech Recognition (ASR) or Text-To-Speech (TTS), to the VoiceXML application that is being processed by the VoiceXML application browser. A call data manager may also be implemented on the MVSS mobile device and may provide call data that, in conjunction with data from the VoiceXML application server, may authorize access to advanced Media Resource Control Protocol (MRCP) services.

Abstract: An augmented reality (AR) device can be configured to monitor ambient audio data. The AR device can detect speech in the ambient audio data, convert the detected speech into text, or detect keywords such as rare words in the speech. When a rare word is detected, the AR device can retrieve auxiliary information (e.g., a definition) related to the rare word from a public or private source. The AR device can display the auxiliary information for a user to help the user better understand the speech. The AR device may perform translation of foreign speech, may display text (or the translation) of a speaker's speech to the user, or display statistical or other information associated with the speech.

Abstract: The present disclosure provides a method and apparatus of discovering a bad case based on artificial intelligence, a device and a storage medium, wherein the method comprises: performing named entity recognition for a to-be-recognized query, and respectively obtaining a confidence level of each character in the query; respectively obtaining a probability value of each character of forming a word with a neighboring character in the query; determining whether there is a bad case according to the confidence level and the probability value. The solution of the present disclosure may be applied to save man power costs, and improve the processing efficiency and enhance a discovery rate of bad cases.

Abstract: Aspects migrate an unstructured document to a specific document type definition Darwin Information Typing architecture wherein processors are configured to calculate a verb to noun ratio of an unstructured document by dividing a of plurality verbs of the unstructured document by a plurality of nouns of the unstructured document, assign a first weight to the unstructured document based on the calculated verb to noun ratio, and migrate the unstructured document to a specific document type definition Darwin Information Typing Architecture based on the first weight.

Abstract: Social data of a conversation partner is analyzed who is physically situated relative to a user to have an in-person conversation with the user. From the analysis, a list of topics and a sentiment corresponding to each topic on the list of topics are computed. An evaluation is made that a first value of a first sentiment corresponding to a first topic in the list of topics exceeds a threshold. The user is provided a notification about the first topic and the first sentiment, causing the user to discuss the first topic with the partner in the in-person conversation. When a second topic has a second sentiment below the threshold, the user is caused to drop the second topic from the in-person conversation.

Abstract: Systems, apparatuses, and methods are described for controlling source tracking and delaying beamforming in a microphone array system. A source tracker may continuously determine a direction of an audio source. A source tracker controller may pause the source tracking of the source tracker if a user may continue to speak to the system. The source tracker controller may resume the source tracking of the source tracker if the user may cease to speak to the system, or when one or more pause durations have been reached.

Abstract: Disclosed are a display apparatus, a voice acquiring apparatus and a voice recognition method thereof, the display apparatus including: a display unit which displays an image; a communication unit which communicates with a plurality of external apparatuses; and a controller which includes a voice recognition engine to recognize a user's voice, receives a voice signal from a voice acquiring unit, and controls the communication unit to receive candidate instruction words from at least one of the plurality of external apparatuses to recognize the received voice signal.

Abstract: The present disclosure provides a method and a device for processing a speech based on artificial intelligence. The method includes: grading a current frame included in a speech packet to be decoded by using an acoustic model to obtain a grading result; identifying whether the current frame is a quasi-silent frame according the grading result; and skipping the current frame and not decoding the current frame if the current frame is the quasi-silent frame. In the present disclosure, before the current frame included in the speech pocket to be decoded is decoded, it is identified whether to decode the current frame according to the grading result obtained with the acoustic model. When there is no need to decode the current frame, the current frame is skipped. Thus, a redundancy decoding may be avoided, a speed of decoding is improved and recognition of the speech packet to be decoded is expedited.