Abstract: A method, computer program product, and computing system for obtaining, by a computing device, encounter information of a patient encounter, wherein the encounter information may include audio encounter information obtained from at least a first encounter participant. The audio encounter information obtained from at least the first encounter participant may be processed. A user interface may be generated displaying a plurality of layers associated with the audio encounter information obtained from at least the first encounter participant. A user input may be received from a peripheral device to navigate through each of the plurality of layers associated with the audio encounter information displayed on the user interface.

Abstract: A system and method for authoring a voice application is provided. The voice recognition process uses a dynamic grammar which obtains data from a backend data source based upon an input in order to create expected results that a speech engine can recognize. The process can retrieve data from at least one of a plurality of back-end data sources and can build a grammar based on the data using at least in part a dynamic grammar builder. The grammar is loaded into the voice recognition application using at least in part the reusable dialog component. A data access service, XSLT processor, or other data accessing framework can be used to facilitate access and manipulation of data in heterogeneous environments.

Abstract: Method and apparatus for multi-pass speech recognition. An input device receives spoken input. A processor performs a first pass speech recognition technique on the spoken input and forms first pass results. The first pass results include a number of alternative speech expressions, each having an assigned score related to the certainty that the corresponding expression correctly matches the spoken input. The processor selectively performs a second pass speech recognition technique on the spoken input according to the first pass results. Preferably, the second pass attempts to correctly match the spoken input to only those expressions which were identified during the first pass. Otherwise, if one of the expressions identified by the first pass is assigned a score higher than a predetermined threshold (e.g., 95%), the second pass is not performed.

Abstract: Method and apparatus for multi-pass speech recognition. An input device receives spoken input. A processor performs a first pass speech recognition technique on the spoken input and forms first pass results. The first pass results include a number of alternative speech expressions, each having an assigned score related to the certainty that the corresponding expression correctly matches the spoken input. The processor selectively performs a second pass speech recognition technique on the spoken input according to the first pass results. Preferably, the second pass attempts to correctly match the spoken input to only those expressions which were identified during the first pass. Otherwise, if one of the expressions identified by the first pass is assigned a score higher than a predetermined threshold (e.g., 95%), the second pass is not performed.

Abstract: The present invention introduces a system and method for improved robustness in a speech system. In one embodiment, a method comprises receiving an utterance from an intended talker at a speech recognition system; computing a speaker verification score with a voice characteristic model associated and with the utterance; computing a speech recognition score associated with the utterance; and selecting a best hypothesis associated with the utterance and based on both the speaker verification score and the speech recognition score.

Abstract: The present invention introduces a system and method for automatically optimizing recognition configuration parameters for speech recognition systems. In one embodiment, a method comprises receiving an utterance at a speech recognizer, wherein the speech recognizer has a learning mode. The speech recognizer is run in a learning mode to automatically generate tuned configuration parameters. Subsequent utterances are recognized with the tuned configuration parameters to generate future recognition results.

Abstract: A method and system for barge-in acknowledgement are disclosed. A prompt is attenuated upon detection of speech. The speech is accepted and the prompt is terminated if the speech corresponds to an allowable response.

Abstract: A computer-implemented graphical design tool allows a developer to graphically author a dialog flow for use in a voice response system and to graphically create an operational link between a hypermedia page and a speech object. The hypermedia page may be a Web site, and the speech object may define a spoken dialog interaction between a person and a machine. Using a drag-and-drop interface, the developer can graphically define a dialog as a sequence of speech objects. The developer can also create a link between a property of any speech object and any field of a Web page, to voice-enable the Web page, or to enable a speech application to access Web site data.

Abstract: A system and method provides universal access to voice-based documents containing information formatted using MIME and HTML standards using customized extensions for voice information access and navigation. These voice documents are linked using HTML hyper-links that are accessible to subscribers using voice commands, touch-tone inputs and other selection means. These voice documents and components in them are addressable using HTML anchors embedding HTML universal resource locators (URLs) rendering them universally accessible over the Internet. This collection of connected documents forms a voice web. The voice web includes subscriber-specific documents including speech training files for speaker dependent speech recognition, voice print files for authenticating the identity of a user and personal preference and attribute files for customizing other aspects of the system in accordance with a specific subscriber.

Abstract: A method and apparatus are provided for performing prosody based endpoint detection of speech in a speech recognition system. Input speech represents an utterance, which has an intonation pattern. An end-of-utterance condition is identified based on prosodic parameters of the utterance, such as the intonation pattern and the duration of the final syllable of the utterance, as well as non-prosodic parameters, such as the log energy of the speech.

Abstract: A network comprises a number of speech-enabled sites maintaining a number of voice pages. A central server on the network executes a voice browser which provides users with access to the sites using voice-activated hyperlinks. The server also maintains and brokers information associated with the users based on spoken dialogs between the users and the sites. In response to a user accessing a given ASR site, information about that user is provided by the server for use by that ASR site. The information is used by the ASR site to optimize a spoken dialog between the user and the ASR site by reducing the amount of information the user is required to provide during the dialog. Information about the user can thereby be shared between separate speech enabled sites, in a manner which is transparent to the user, in order to expedite the user's interaction with those sites.

Abstract: A technique for identifying one or more items from amongst a plurality of items in response to a spoken utterance is used to improve call routing and information retrieval systems which employ automatic speech recognition (ASR). An automatic speech recognizer is used to recognize the utterance, including generating a plurality of hypotheses for the utterance. A query element is then generated for use in identifying one or more items from amongst the plurality of items. The query element includes a set of values representing two or more of the hypotheses, each value corresponding to one of the words in the hypotheses. Each value in the query element is then weighted based on hypothesis confidence, word confidence, or both, as determined by the ASR process. The query element is then applied to the plurality of items to identify one or more items which satisfy the query.

Abstract: A method and apparatus for generating a noise-reduced feature vector representing human speech are provided. Speech data representing an input speech waveform are first input and filtered. Spectral energies of the filtered speech data are determined, and a noise reduction process is then performed. In the noise reduction process, a spectral magnitude is computed for a frequency index of multiple frequency indexes. A noise magnitude estimate is then determined for the frequency index by updating a histogram of spectral magnitude, and then determining the noise magnitude estimate as a predetermined percentile of the histogram. A signal-to-noise ratio is then determined for the frequency index. A scale factor is computed for the frequency index, as a function of the signal-to-noise ratio and the noise magnitude estimate. The noise magnitude estimate is then scaled by the scale factor.

Abstract: The present invention introduces a system and method for unsupervised, on-line, adaptation in speaker verification. In one embodiment, a method for adapting a speaker model to improve the verification of a speaker's voice, comprises detecting a channel of a verification utterance; learning vocal characteristics of the speaker on the detected channel; and transforming the learned vocal characteristics of the speaker from the detected channel to the speaker model of a second channel.

Abstract: A speech-enabled distributed processing system forming a Voice Web includes a gateway, one or more voice content sites coupled to the gateway over a wide area network, and a browser coupled to the gateway over a network, which may or may not be the wide area network. The gateway receives telephone calls from one or more users over telephony connections and performs endpointing of speech of each user. The browser provides the gateway with information enabling the gateway to selectively direct the endpointed speech to a voice content site via the wide area network. The gateway outputs the endpointed speech in the form of application protocol requests onto the wide area network to the appropriate site, as specified by the browser, or to the browser. The gateway receives prompts in the form of application protocol responses from the browser or a voice content site and plays the prompts to the appropriate user over the telephony connection.

Abstract: A technique for adaptation of a speech recognizing system across multiple remote communication sessions with a speaker. The speaker can be a telephone caller. An acoustic model is utilized for recognizing the speaker's speech. Upon initiation of a first remote session with the speaker, the acoustic model is speaker-independent. During the first session, the speaker is uniquely identified and speech samples are obtained from the speaker. In the preferred embodiment, the samples are obtained without requiring the speaker to engage in a training session. The acoustic model is then modified based upon the samples thereby forming a modified model. The model can be modified during the session or after the session is terminated. Upon termination of the session, the modified model is then stored in association with an identification of the speaker. During a subsequent remote session, the speaker is identified and, then, the modified acoustic model is utilized to recognize the speaker's speech.

Abstract: The present invention introduces a system and method for dynamically improving speech recognition in a speech recognition or other speech processing system. The method comprises dynamically adjusting the system, which comprises estimating the utilization of resources in the system; and improving the performance of the system according to the availability of resources.

Abstract: A voice authentication system having a cognitive recall mechanism for password verification is provided. A user is enrolled for password verification by receiving a first voice input from the user representing the password prompt and a second voice input representing a correct response to the password prompt. The first and second voice inputs may be stored as waveforms, as voiceprints, recognized speech data, or a combination thereof. During verification, the identity of the user is verified by outputting the user-provided password prompt and evaluating a response to password prompt against the correct response. Thus, the user is able to select his own password prompt to facilitate cognitive recall of the password during a subsequent verification phase.

Abstract: A computerized method for building and running natural language understanding systems, wherein a natural language understanding system takes a sentence as input and returns some representation of the possible meanings of the sentence as output (the “interpretation”) using a run-time interpreter th assigns interpretations to sentences and a compiler that produces (in a computer memory) an internal specification needed for the run-time interpreter from a user specification of the semantics of the application. The compiler builds a natural language system, while the run-time interpreter runs the system.

Abstract: A technique for recognizing telephone numbers and other information embedded in voice messages stored in a telephone voice messaging system. A voice recognition system is coupled to the telephone voice messaging system. A voice message stored in the voice messaging system is transferred to the voice recognition system. The voice recognition system segments the voice message and then searches the segments for a predetermined speech reference model (grammar) which is expected to contain information of importance to the recipient of the message. In a preferred embodiment, the predetermined is a numeric grammar which specifies a sequence of numbers occurring in the voice message. In alternate embodiments, the grammar specifies a date, a time, an address, and so forth, and can specify more than one such type of information. The grammar can be modified or selected by the recipient of the voice message so that the voice recognition system searches for information of particular interest to the recipient.