Abstract: A method of performing a voice command function in an electronic device includes detecting voice of a user, acquiring one or more pieces of attribute information from the voice, and authenticating the user by comparing the attribute information with pre-stored authentic attribe information, using a recognition model. An electronic device includes a voice input module configured to detect a voice of a user, a first processor configured to acquire one or more pieces of attribute information from the voice and authenticate the user by comparing the attribute information with a recognition model, and a second processor configured to when the attribute information matches the recognition mode, activate the voice command function, receive a voice command of the user, and execute an application corresponding to the voice command. Other embodiments are also disclosed.

Abstract: A candidate selection apparatus utilizing voice recognition includes an association unit that associates target candidates with candidate numbers so that numerals of the target candidates coincide with numerals of the candidate numbers when the target candidates to be displayed in list form are character strings representing the numerals of the candidate numbers, and a display control unit that displays the target candidates and the candidate numbers in list form in accordance with the associations made between the target candidates and the candidate numbers.

Abstract: Method for controlling user access to a service available in a data network and/or to information stored in a user database, in order to protect stored user data from unauthorized access, such that the method comprises the following: input of a user's speech sample to a user data terminal, processing of the user's speech sample in order to obtain a prepared speech sample as well as a current voice profile of the user, comparison of the current voice profile with an initial voice profile stored in an authorization database, and output of an access-control signal to either permit or refuse access, taking into account the result of the comparison step, such that the comparison step includes a quantitative similarity evaluation of the current and the stored voice profiles as well as a threshold-value discrimination of a similarity measure thereby derived, and an access-control signal that initiates permission of access is generated only if a prespecified similarity measure is not exceeded.

Abstract: Methods, systems, and computer readable storage medium related to operating an intelligent digital assistant are disclosed. A text string is obtained from a speech input received from a user. The received text string is interpreted to derive a representation of user intent based at least in part on a plurality of words associated with a user and stored in memory associated with the user, the plurality of words including words from a plurality of user interactions with an automated assistant. At least one domain, a task, and at least one parameter for the task, are identified based at least in part on the representation of user intent. The identified task is performed. An output is provided to the user, where the output is related to the performance of the task.

Abstract: A machine, system and method for user-guided teaching and modifications of voice commands and actions to be executed by a conversational learning system. The machine includes a system bus for communicating data and control signals received from the conversational learning system to a computer system, a vehicle data and control bus for connecting devices and sensors in the machine, a bridge module for connecting the vehicle data and control bus to the system bus, machine subsystems coupled to the vehicle data and control bus having a respective user interface for receiving a voice command or input signal from a user, a memory coupled to the system bus for storing action command sequences learned for a new voice command and a processing unit coupled to the system bus for automatically executing the action command sequences learned when the new voice command is spoken.

Abstract: Systems and methods providing for secure voice print authentication over a network are disclosed herein. During an enrollment stage, a client's voice is recorded and characteristics of the recording are used to create and store a voice print. When an enrolled client seeks access to secure information over a network, a sample voice recording is created. The sample voice recording is compared to at least one voice print. If a match is found, the client is authenticated and granted access to secure information. Systems and methods providing for a dual use voice analysis system are disclosed herein. Speech recognition is achieved by comparing characteristics of words spoken by a speaker to one or more templates of human language words. Speaker identification is achieved by comparing characteristics of a speaker's speech to one or more templates, or voice prints. The system is adapted to increase or decrease matching constraints depending on whether speaker identification or speaker recognition is desired.

Abstract: Provided are embodiments of systems, computer medium and computer-implemented methods for authenticating users using voice biometrics. Methods including receiving a request to access a resource via a user device, receiving a credentials set from a user (the credentials set including candidate credentials and candidate voice stream), determining whether the candidate credentials are valid based on a comparison of the candidate credentials to existing user credentials, in response to determining that the candidate credentials are valid, determining whether the candidate voice stream is valid based on a comparison of the candidate voice stream to a voice biometric associated with the candidate credentials and, in response to determining that the candidate voice stream is valid, generating an authentication signal configured to enable access to the resource via the user device.

Abstract: Methods, systems, and computer readable storage medium related to operating an intelligent digital assistant are disclosed. A user request is received, the user request including at least a speech input received from the user. The user request is processed to obtain a representation of user intent, where the representation of user intent associates the user request with a task flow operationalizing a requested task, and the task flow is operable to invoke a plurality of services each supporting functions according to a respective plurality of service parameters. Based on the representation of user intent, one or more relevant task parameters are identified from a plurality of task parameters of the task flow. A subset of the plurality of services are selectively invoked during execution of the task flow, where the selectively invoked subset of the plurality of services support functions according to the identified one or more relevant task parameters.

Abstract: A method of recognizing speech is provided. The method includes the operations of (a) dividing first speech that is input to a speech recognizing apparatus into frames; (b) converting the frames of the first speech into frames of second speech by applying conversion rules to the divided frames, respectively; and (c) recognizing, by the speech recognizing apparatus, the frames of the second speech, wherein (b) comprises converting the frames of the first speech into the frames of the second speech by reflecting at least one frame from among the frames that are previously positioned with respect to a frame of the first speech.

Abstract: A multi-layered speech recognition apparatus and method, the apparatus includes a client checking whether the client recognizes the speech using a characteristic of speech to be recognized and recognizing the speech or transmitting the characteristic of the speech according to a checked result; and first through N-th servers, wherein the first server checks whether the first server recognizes the speech using the characteristic of the speech transmitted from the client, and recognizes the speech or transmits the characteristic according to a checked result, and wherein an n-th (2?n?N) server checks whether the n-th server recognizes the speech using the characteristic of the speech transmitted from an (n?1)-th server, and recognizes the speech or transmits the characteristic according to a checked result.

Abstract: Disclosed herein are systems, methods, and computer readable-media for answering a communication notification. The method for answering a communication notification comprises receiving a notification of communication from a user, converting information related to the notification to speech, outputting the information as speech to the user, and receiving from the user an instruction to accept or ignore the incoming communication associated with the notification. In one embodiment, information related to the notification comprises one or more of a telephone number, an area code, a geographic origin of the request, caller id, a voice message, address book information, a text message, an email, a subject line, an importance level, a photograph, a video clip, metadata, an IP address, or a domain name. Another embodiment involves notification assigned an importance level and repeat attempts at notification if it is of high importance.

Abstract: The multi-party conversation analyzer of the present invention allows users to search a database of recorded phone calls to find calls which fit user-defined criteria for “suspicious calls”. Such criteria may include indications that a call included a 3-way call event, presence of an unauthorized voiced during the call, presence of the voice of an individual known to engage in identity theft, etc. A segment of speech within a call may be graphically selected and a search for calls with similar voices rapidly initiated. Searches across the database for specified voices are speeded by first searching for calls which contain speech from cohort speakers with similar voice characteristics.

Abstract: A system enables voice authentication via a network. The system may include an intelligent voice response engine operatively coupled to the network for receiving transaction or access requests from a plurality of telecommunications devices over the network. A speech recognition and verification services engine may be operatively coupled to the network and a database may be operatively coupled to the speech recognition and verification services engine for storing user voice print profiles. The speech recognition and verification services engine may receive a speaker verification call from the intelligent voice response engine and perform speaker verification on the received speaker verification call based on the stored user voice print profiles. The speech recognition and verification services engine may generate a verification score based upon results of the speaker verification.

Abstract: A method, device, system, and computer medium for providing interactive advertising are provided. For example, a device may request an advertisement from a remote server, receive the advertisement, receive a response from a user who is listening and/or watching the advertisement, and transmit the response to the server for further action. The user may input a response by speaking. A server may receive an advertisement request from the device, select an advertisement based on pre-defined one or more criteria, transmit the selected advertisement to the device for play, receive from the device a response to the selected advertisement, and then perform an action corresponding to the received response.

Abstract: During a communication technique, an individual is identified based on a signal that includes vocal sounds of the individual and a voice print of the individual. For example, the voice print may include features characteristic of the individual's voice. Alternatively or additionally, the identification may be based on context information associated with a conversation that includes the individual and/or based on pronunciation of the individual's name. After the individual is identified, content in a social graph, which is associated with the individual, may be accessed and provided. This content may include business information, such as: contact information, education information, a job title, an organization associated with the individual, and/or connections of the individual to other individuals in the social graph.

Abstract: Exemplary embodiments provide systems, devices and methods that allow creation and management of lists of items in an integrated manner on an interactive graphical user interface. A user may speak a plurality of list items in a natural unbroken manner to provide an audio input stream into an audio input device. Exemplary embodiments may automatically process the audio input stream to convert the stream into a text output, and may process the text output into one or more n-grams that may be used as list items to populate a list on a user interface.

Abstract: A speech recognition method includes receiving input speech from a user, processing the input speech to obtain at least one parameter value, and determining an experience level of the user using the parameter value(s). The method can also include prompting the user based upon the determined experience level of the user to assist the user in delivering speech commands.

Abstract: Systems, methods, and devices for location-based context driven voice recognition are disclosed. A mobile or stationary computing device can include position locating functionality for determining the particular physical location of the computing device. Once the physical location of the computing device determined, a context related to that particular physical location. The context related to the particular physical location can include information regarding objects or experiences a user might encounter while in that particular physical location. The context can then be used to determine delimited or constrained voice recognition vocabulary subset based on the range of experiences a user might encounter within a particular context. The voice recognition vocabulary subset can then be referenced or used by a voice recognizer to increase the speed, accuracy, and effectiveness in receiving, recognizing, and acting in response to voice commands received by the user while in that particular physical location.

Abstract: Device, system, and method of liveness detection using voice biometrics. For example, a method comprises: generating a first matching score based on a comparison between: (a) a voice-print from a first text-dependent audio sample received at an enrollment stage, and (b) a second text-dependent audio sample received at an authentication stage; generating a second matching score based on a text-independent audio sample; and generating a liveness score by taking into account at least the first matching score and the second matching score.

Abstract: A speech detection apparatus and method are provided. The speech detection apparatus and method determine whether a frame is speech or not using feature information extracted from an input signal. The speech detection apparatus may estimate a situation related to an input frame and determine which feature information is required for speech detection for the input frame in the estimated situation. The speech detection apparatus may detect a speech signal using dynamic feature information that may be more suitable to the situation of a particular frame, instead of using the same feature information for each and every frame.

Abstract: In some implementations, audio data for an utterance is provided over a network. At a client device and over the network, information is received that indicates candidate transcriptions for the utterance and semantic information for the candidate transcriptions. A semantic parser is used at the client device to evaluate each of at least a plurality of the candidate transcriptions. One of the candidate transcriptions is selected based on at least the received semantic information and the output of the semantic parser for the plurality of candidate transcriptions that are evaluated.

Abstract: A recognition dictionary creation device identifies the language of a reading of an inputted text which is a target to be registered and adds a reading with phonemes in the language identified thereby to the target text to be registered, and also converts the reading of the target text to be registered from the phonemes in the language identified thereby to phonemes in a language to be recognized which is handled in voice recognition to create a recognition dictionary in which the converted reading of the target text to be registered is registered.

Abstract: Systems and methods for controlling access to resources using spoken Completely Automatic Public Turing Tests To Tell Humans And Computers Apart (CAPTCHA) tests are disclosed. In these systems and methods, entities seeking access to resources are required to produce an input utterance that contains at least some audio. That utterance is compared with voice reference data for human and machine entities, and a determination is made as to whether the entity requesting access is a human or a machine. Access is then permitted or refused based on that determination.

Abstract: A voice control method and a mobile terminal apparatus are provided. The mobile terminal apparatus includes a voice receiving module, a voice outputting module, a voice wake-up module and a language recognition module. When the voice wake-up module determined that a first voice signal matches to identification information, the voice receiving module is turned on. When the voice receiving module receives a second voice signal after the first voice signal, the language recognition module parses the second voice signal and obtains a voice recognition result. When the voice recognition result includes an executing request, the language recognition module executes a responding operation, and the voice receiving module is turned off from receiving a third voice signal. When the voice recognition result does not include the executing request, the language recognition module executes a speech conversation mode.

Abstract: The present disclosure relates to systems and methods that are related to projection.

Abstract: A method and a processing device for managing an interactive speech recognition system is provided. Whether a voice input relates to expected input, at least partially, of any one of a group of menus different from a current menu is determined. If the voice input relates to the expected input, at least partially, of any one of the group of menus different from the current menu, skipping to the one of the group of menus is performed. The group of menus is different from the current menu include menus at multiple hierarchical levels.

Abstract: A universal pattern processing system receives input data and produces output patterns that are best associated with said data. The system uses input means receiving and processing input data, a universal pattern decoder means transforming models using the input data and associating output patterns with original models that are changed least during transforming, and output means outputting best associated patterns chosen by a pattern decoder means.

Abstract: An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

Abstract: A method for receiving processed information at a remote device is described. The method includes transmitting from the remote device a verbal request to a first information provider and receiving a digital message from the first information provider in response to the transmitted verbal request. The digital message includes a symbolic representation indicator associated with a symbolic representation of the verbal request and data used to control an application. The method also includes transmitting, using the application, the symbolic representation indicator to a second information provider for generating results to be displayed on the remote device.

Abstract: Techniques for training and applying prosody models for speech synthesis are provided. A speech recognition engine processes audible speech to produce text annotated with prosody information. A prosody model is trained with this annotated text. After initial training, the model is applied during speech synthesis to generate speech with non-standard prosody from input text. Multiple prosody models can be used to represent different prosody styles.

Abstract: In an aspect, in general, a system includes a first input for receiving a first data representing an interaction among a plurality of parties, the first data identifying a plurality of parts of the interaction and identifying a plurality of segments associated with each part of the plurality of parts, a second input for receiving a second data associating each of one or more labels with one or more corresponding query phrases, a searching module for searching the first data to identify putative instances of the query phrases, and a classifier for labeling the parts of the interaction associated with the identified putative instances of the query phrases with the labels corresponding to the identified query phrases.

Abstract: Methods, systems, and computer programs encoded on a computer storage medium for real-time acoustic adaptation using stability measures are disclosed. The methods include the actions of receiving a transcription of a first portion of a speech session, wherein the transcription of the first portion of the speech session is generated using a speaker adaptation profile. The actions further include receiving a stability measure for a segment of the transcription and determining that the stability measure for the segment satisfies a threshold. Additionally, the actions include triggering an update of the speaker adaptation profile using the segment, or using a portion of speech data that corresponds to the segment. And the actions include receiving a transcription of a second portion of the speech session, wherein the transcription of the second portion of the speech session is generated using the updated speaker adaptation profile.

Abstract: A system and method may be provided to segment and/or classify an audio signal from transformed audio information. Transformed audio information representing a sound may be obtained. The transformed audio information may specify magnitude of a coefficient related to energy amplitude as a function of frequency for the audio signal and time. Features associated with the audio signal may be obtained from the transformed audio information. Individual ones of the features may be associated with a feature score relative to a predetermined speaker model. An aggregate score may be obtained based on the feature scores according to a weighting scheme. The weighting scheme may be associated with a noise and/or SNR estimation. The aggregate score may be used for segmentation to identify portions of the audio signal containing speech of one or more different speakers. For classification, the aggregate score may be used to determine a likely speaker model to identify a source of the sound in the audio signal.

Abstract: A method and system for using conversational biometrics and speaker identification and/or verification to filter voice streams during mixed mode communication. The method includes receiving an audio stream of a communication between participants. Additionally, the method includes filtering the audio stream of the communication into separate audio streams, one for each of the participants. Each of the separate audio streams contains portions of the communication attributable to a respective participant. Furthermore, the method includes outputting the separate audio streams to a storage system.

Abstract: A display device with a voice recognition capability may be used to allow a user to speak voice commands for controlling certain features of the display device. As a means for increasing operational efficiency, the display device may utilize a plurality of voice recognition units where each voice recognition unit may be assigned a specific task.

Abstract: The voice recognition device according to the present disclosure includes a communication interface that communicates with an external device, a first microphone that collects sound to produce audio data, and a controller that analyzes the audio data produced by the first microphone, determines contents of a designation corresponding to an analysis result, and then controls its own device based on a determination result, and yet controls its own device to urge a user to use the external device when the contents of the designation corresponding to the analysis result cannot be determined.

Abstract: A system includes one or more computer processors that are configured to receive data relating to a composition of a target group, receive logged communications of the target group, extract textual information from the logged communications, analyze the textual information using statistical and linguistic sentiment analysis techniques, identify an individual or sub-group from the target group as a function of the analysis of the textual information, and display on a user interface or transmit to another processor the identified individual or sub-group of the target group and to display on the user interface or transmit to another processor a sentiment assessment of the identified individual or sub-group as a function of the statistical and linguistic sentiment analysis.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for speech endpointing based on a voice profile. In one aspect, a method includes the actions of receiving audio data corresponding to an utterance spoken by a particular user. The actions further include generating a voice profile for the particular user using at least a portion of the audio data. The actions further include determining in the audio data a beginning point or an ending point of the utterance based at least in part on the voice profile for the particular user. The actions further include based on the beginning point, the ending point, or both the beginning point and the ending point, outputting data indicating the utterance.

Abstract: The present invention provides a communication monitoring and analysis method and system. More specifically, the present invention provides a method for analyzing problems within a communication network and further identifying sources of such problems based on voice sample substitutions. The voice sample substitutions used may correspond to canned speech samples used to highlight possible communication network problems.

Abstract: Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing in a downlink path of a communication device. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify the identity of a far-end speaker participating in a voice call with a user of the communication device. Knowledge of the identity of the far-end speaker is then used to improve the performance of one or more downlink speech processing algorithms implemented on the communication device.

Abstract: Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify a user of the communication device and/or the identity of a far-end speaker participating in a voice call with a user of the communication device. Knowledge of the identity of the user and/or far-end speaker is then used to improve the performance of one or more speech processing algorithms implemented on the communication device.

Abstract: One method of operation includes beamforming a plurality of microphone outputs to obtain a plurality of virtual microphone audio channels. Each virtual microphone audio channel corresponds to a beamform. The virtual microphone audio channels include at least one voice channel and at least one noise channel. The method includes performing voice activity detection on the at least one voice channel and adjusting a corresponding voice beamform until voice activity detection indicates that voice is present on the at least one voice channel. Another method beamforms the plurality of microphone outputs to obtain a plurality of virtual microphone audio channels, where each virtual microphone audio channel corresponds to a beamform, and with at least one voice channel and at least one noise channel. The method performs voice recognition on the at least one voice channel and adjusts the corresponding voice beamform to improve a voice recognition confidence metric.

Abstract: Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing in an uplink path of a communication device. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify the identity of a near-end speaker. Knowledge of the identity of the near-end speaker is then used to improve the performance of one or more uplink speech processing algorithms implemented on the communication device.

Abstract: A method and apparatus for voice recognition performed in a voice recognition block comprising a plurality of voice recognition stages. The method includes receiving a first plurality of voice inputs, corresponding to a first phrase, into a first voice recognition stage of the plurality of voice recognition stages, wherein multiple ones of the voice recognition stages includes a plurality of voice recognition modules and multiples ones of the voice recognition stages perform a different type of voice recognition processing, wherein the first voice recognition stage processes the first plurality of voice inputs to generate a first plurality of outputs for receipt by a subsequent voice recognition stage. The method further includes, receiving by each subsequent voice recognition stage a plurality of outputs from a preceding voice recognition stage, wherein a plurality of final outputs is generated by a final voice recognition stage from which to approximate the first phrase.

Abstract: A method includes obtaining a speech sample from a pre-processing front-end of a first device, identifying at least one condition, and selecting a voice recognition speech model from a database of speech models, the selected voice recognition speech model trained under the at least one condition. The method may include performing voice recognition on the speech sample using the selected speech model. A device includes a microphone signal pre-processing front end and operating-environment logic, operatively coupled to the pre-processing front end. The operating-environment logic is operative to identify at least one condition. A voice recognition configuration selector is operatively coupled to the operating-environment logic, and is operative to receive information related to the at least one condition from the operating-environment logic and to provide voice recognition logic with an identifier for a voice recognition speech model trained under the at least one condition.

Abstract: Embodiments of the present invention provide a method, device, and system for classifying voice conference minutes. The method is: performing voice source locating according to audio data of the conference site so as to acquire a location of a voice source corresponding to the audio data, writing the location of the voice source into additional field information of the audio data, writing a voice activation flag into the additional field information, packaging the audio data as an audio code stream, and sending the audio code stream and the additional field information of the audio code stream to a recording server, so that the recording server classifies the audio data according to the additional field information and writes a participant identity that corresponds to the location of the voice source corresponding to the audio data into the additional field information of the audio code stream.

Abstract: A method of processing a call in a voice-command platform includes a step of transferring the call from the voice-command platform to a second voice-command platform. The method continues with the step of transmitting, either directly or indirectly, grammar information from the voice command platform to the second voice-command platform for use by a voice command application executing in the second voice-command platform in processing the call. The grammar information could be logic defining application-level grammar or system-level grammar. Alternatively, the grammar information could be a network address (e.g., URI or URL) where the grammar is stored in a file, e.g., a VXML document. The features of this invention enhance the user experience by preserving and using grammars used initially in the first voice command platform in other, downstream, voice command platforms.

Abstract: An interactive robot may audibly communicate with a user, determine whether reasoning is performed according to a reasoning weight of a schedule if there is a schedule to be recommended by the interactive robot during conversation between the interactive robot and the user, perform reasoning using information collected in a database if reasoning is necessary, and provide schedule information to the user.

Abstract: Transferring metadata is disclosed. Information about a network interaction is processed to generate metadata describing the network interaction. Based on the metadata it is determined whether the metadata is to be transferred to an aggregator. In the event that the metadata is to be transferred, one or more aggregators are determined to which the metadata is to be transferred. The metadata is transferred to the one or more aggregators.

Abstract: Systems, apparatus and methods of reducing or eliminating device loss are described herein. A computing device may receive a user input. The user input may include a proximity preference. The computing device may generate an alert signal upon detecting that a distance between the computing device and the user has increased beyond the first proximity preference. The detecting may be based on sensing a characteristic of the user, such as a voice characteristic or a facial characteristic, or upon detecting that a signal between a user headset and the computing device has diminished in strength.