Abstract: Voice enabled dialog with web pages is provided. An Internet address of a web page is received including an area with which a user of a client device can specify information. The web page is loaded using the received Internet address of the web page. A task structure of the web page is then extracted. An abstract representation of the web is then generated. A dialog script, based on the abstract representation of the web page is then provided. Spoken information received from the user is converted into text and the converted text is inserted into the area.

Abstract: According to some aspects, a method of associating dictation with an electronic record in a system having a dictation system comprising a dictation application for capture of speech input and a separate electronic records system for managing electronic records is provided. The method comprises receiving, by the dictation application, speech input from a user corresponding to a dictation to be associated with an electronic record of the electronic records system, obtaining, by the dictation application, a job identifier associated with the dictation, providing, by the dictation application, the job identifier and audio data based on the speech input for transcription, obtaining, by the dictation application, a dictation marker comprising the job identifier and one or more delimiters, and causing the dictation marker to be inserted into the electronic record.

Abstract: In some embodiments, the recognition results produced by a speech processing system (which may include a top recognition result and one or more alternative recognition results) based on an analysis of a speech input, are evaluated for indications of potential significant errors. In some embodiments, the recognition results may be evaluated to determine whether a meaning of any of the alternative recognition results differs from a meaning of the top recognition result in a manner that is significant for the domain. In some embodiments, one or more of the recognition results may be evaluated to determine whether the result(s) include one or more words or phrases that, when included in a result, would change a meaning of the result in a manner that would be significant for the domain.

Abstract: A system and method for non-intrusive acoustic parameter estimation is included. The method may include receiving, at a computing device, a first speech signal associated with a particular user. The method may include extracting one or more short-term features from the first speech signal. The method may also include determining one or more statistics of each of the one or more short-term features from the first speech signal. The method may further include classifying the one or more statistics as belonging to one or more acoustic parameter classes.

Abstract: Techniques for adapting a trained neural network acoustic model, comprising using at least one computer hardware processor to perform: generating initial speaker information values for a speaker; generating first speech content values from first speech data corresponding to a first utterance spoken by the speaker; processing the first speech content values and the initial speaker information values using the trained neural network acoustic model; recognizing, using automatic speech recognition, the first utterance based, at least in part on results of the processing; generating updated speaker information values using the first speech data and at least one of the initial speaker information values and/or information used to generate the initial speaker information values; and recognizing, based at least in part on the updated speaker information values, a second utterance spoken by the speaker.

Abstract: A multi-mode speech communication system is described that has different operating modes for different speech applications. A signal processing module is in communication with the speech applications and includes an input processing module and an output processing module. The input processing module processes microphone input signals to produce a set user input signals for each speech application that are limited to currently active system users for that speech application. The output processing module processes application output communications from the speech applications to produce loudspeaker output signals to the system users, wherein for each different speech application, the loudspeaker output signals are directed only to system users currently active in that speech application.

Abstract: A user device provides dynamic speech processing services during variable network connectivity with a network server. The user device includes a monitor that monitors a level of network connectivity between the user device and the network server. A user device speech processor processes speech data and is initiated based on a determination that the level of network connectivity between the user device and the network server is impaired. The monitor determines when the level of network connectivity between the user device and the network server is no longer impaired.

Abstract: A mobile device is adapted for automatic speech recognition (ASR). A user interface for interaction with a user includes an input microphone for obtaining speech inputs from the user for automatic speech recognition, and an output interface for system output to the user based on ASR results that correspond to the speech input. A local controller obtains a sample of non-ASR audio from the input microphone for ASR-adaptation to channel-specific ASR characteristics, and then provides a representation of the non-ASR audio to a remote ASR server for server-side adaptation to the channel-specific ASR characteristics, and then provides a representation of an unknown ASR speech input from the input microphone to the remote ASR server for determining ASR results corresponding to the unknown ASR speech input, and then provides the system output to the output interface.

Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for generating domain-specific speech recognition models for a domain of interest by combining and tuning existing speech recognition models when a speech recognizer does not have access to a speech recognition model for that domain of interest and when available domain-specific data is below a minimum desired threshold to create a new domain-specific speech recognition model. A system configured to practice the method identifies a speech recognition domain and combines a set of speech recognition models, each speech recognition model of the set of speech recognition models being from a respective speech recognition domain. The system receives an amount of data specific to the speech recognition domain, wherein the amount of data is less than a minimum threshold to create a new domain-specific model, and tunes the combined speech recognition model for the speech recognition domain based on the data.

Abstract: A medical documentation system and a CDI system may be linked together, or integrated, so there is a tie between the two systems that allows for a much more efficient and effective CDI process. In one disclosed embodiment, a medical documentation system transmits to a CDI system a structured data set including at least some information relating to one or more medical facts the medical documentation system automatically extracted from text documenting a patient encounter.

Abstract: A graphical user interface is described for human guided audio source separation in a multi-speaker automated transcription system receiving audio signals representing speakers participating together in a speech session. A speaker avatar for each speaker is distributed about a user interface display to suggest speaker positions relative to each other during the speech session. There also is a speaker highlight element on the interface display for visually highlighting a specific speaker avatar corresponding to an active speaker in the speech session to aid a human transcriptionist listening to the speech session to identify the active speaker. A speech signal processor performs signal processing of the audio signals to isolate an audio signal corresponding to the highlighted speaker avatar.

Abstract: Systems, methods, and apparatus for implementing at least one virtual agent. In some embodiments, the at least one virtual agent is programmed to analyze first information regarding a first person; analyze second information regarding a second person different from the first person; and make a joint recommendation for a plurality of persons based at least in part on the first and second information, wherein the plurality of persons comprises the first person and the second person.

Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for communicating information about transcription progress from a unified messaging (UM) server to a UM client. In one embodiment, the transcription progress describes speech to text transcription of speech messages such as voicemail. The UM server authenticates and establishes a session with a UM client, then receives a get message list request from a UM client as of a first time, responds to the get message list request with a view of a state of messages and available transcriptions for transcribable messages in a list of messages associated with the get message list call at the first time, and, at a second time subsequent to the first time, transmits to the UM client a notification that provides an indication of progress for at least one transcription not yet complete in the list of messages. The messages can include video.

Abstract: According to some aspects, a method for improving understandability of audio corresponding to dictation to assist a transcriptionist in transcribing the dictation is provided. The method comprises presenting a user interface to the transcriptionist, the user interface including at least one control that can be selectively set to one of a plurality of settings, receiving a selection of one of the plurality of settings via the at least one control, and compressing a dynamic range of at least a portion of the audio using at least one parameter value associated with the selected setting.

Abstract: Technology for improving the predictive accuracy of input word recognition on a device by dynamically updating the lexicon of recognized words based on the word choices made by similar users. The technology collects users' vocabulary choices (e.g., words that each user uses, or adds to or removes from a word recognition dictionary), associates users who make similar choices, aggregates related vocabulary choices, filters the words, and sends words identified as likely choices for that user to the user's device. Clusters may include, for example, users in a particular location (e.g., sets of people who use words such as “Puyallup,” “Gloucester,” or “Waiheke”), users with a particular professional or hobby vocabulary, or application-specific vocabulary (e.g., word choices in map searches or email messages).

Abstract: A method of providing hybrid speech recognition between a local embedded speech recognition system and a remote speech recognition system relates to receiving speech from a user at a device communicating with a remote speech recognition system. The system recognizes a first part of speech by performing a first recognition of the first part of the speech with the embedded speech recognition system that accesses private user data, wherein the private user data is not available to the remote speech recognition system. The system recognizes the second part of the speech by performing a second recognition of the second part of the speech with the remote speech recognition system. The final recognition result is a combination of these two recognition processes. The private data can be such local information as a user location, a playlist, frequently dialed numbers or texted people, user contact list information, and so forth.

Abstract: Methods and systems for providing help prompts to a user of an automated dialog system are presented. In some embodiments, a computing device may receive a help request from the user of an automated dialog system. The help request may comprise a user request for information about one or more capabilities of the automated dialog system. The computing device may identify information expected to be input by the user to request that the automated dialog system perform its one or more capabilities. A natural language help prompt may be generated to provide guidance to the user to provide the identified information expected to be input.

Abstract: Utterance data that includes at least a small amount of manually transcribed data is provided. Automatic speech recognition is performed on ones of the utterance data not having a corresponding manual transcription to produce automatically transcribed utterances. A model is trained using all of the manually transcribed data and the automatically transcribed utterances. A predetermined number of utterances not having a corresponding manual transcription are intelligently selected and manually transcribed. Ones of the automatically transcribed data as well as ones having a corresponding manual transcription are labeled. In another aspect of the invention, audio data is mined from at least one source, and a language model is trained for call classification from the mined audio data to produce a language model.

Abstract: Methods and apparatus for reducing latency in speech recognition applications. The method comprises receive first audio comprising speech from a user of a computing device, detecting an end of speech in the first audio, generating an ASR result based, at least in part, on a portion of the first audio prior to the detected end of speech, determining whether a valid action can be performed by a speech-enabled application installed on the computing device using the ASR result, and processing second audio when it is determined that a valid action cannot be performed by the speech-enabled application using the ASR result.

Abstract: According to some aspects, a method of searching for content in response to a user voice query is provided. The method may comprise receiving the user voice query, performing speech recognition to generate N best speech recognition results comprising a first speech recognition result, performing a supervised search of at least one content repository to identify one or more supervised search results using one or more classifiers that classify the first speech recognition result into at least one class that identifies previously classified content in the at least one content repository, performing an unsupervised search of the at least one content repository to identify one or more unsupervised search results, wherein performing the unsupervised search comprises performing a word search of the at least one content repository, and generating combined results from among the one or more supervised search results and the one or more unsupervised search results.