International Business Machines Corporation

Nuance Communications, Inc.

Gary Robert Hanson

Peter J Guasti

Steven Gary Woodward

Harvey M Ruback

Amado Nassiff

Ronald E Vanbuskirk

Daniel E Badt

Huifang Wang

Edwin A Rodriguez

Carl A Smith

James Robert Lewis

Kerry Alan Ortega

Barbara E Ballard

Francis Fado

Arthur Keller

Werayuth T Charoenruengkit

Raimo Bakis

Ronald E VanBuskirk

Keith P Loring

John W Eckhart

Michael Florio

Lisa Abbott

William Russell Whitehead

Jon Palgon

An input from a manually initiated action within a computing system can be received. The system can be associated with a speech component. The input can be associated with a system function. The function can be an operation within the computing system and can be linked to a function identifier. The identifier can be translated to a command data. The command data can be associated with a command identifier, a command, and an alternative command. The command data can be a speech command registered within the speech component. The command data can be presented within a speech interface responsive to the translating. The speech interface can be associated with the speech component.

Permitting automated speech command discovery via manual event to command mapping

A speech processing method including the step of identifying prerecorded audio comprising an original speech segment and a corresponding original noise segment. An audio stream can be generated from the prerecorded audio. The audio stream can comprise a stream speech segment and a stream noise segment. The stream speech segment can have approximately a same duration as the original speech segment. The stream noise segment can have a longer duration than the original noise segment. The audio stream can be conveyed to a speech recognition engine. The speech recognition engine can automatically determine an end of utterance condition based upon the stream noise segment. The original noise segment can be of insufficient duration for the speech recognition engine to determine the end of utterance condition. Responsive to the determining of the end of utterance condition, the stream speech segment can be speech recognized.

Noise playback enhancement of prerecorded audio for speech recognition operations

An in-vehicle system that shares speech processing resources among multiple applications located within a vehicle. The system can include one or more software applications, each associated with different functionally independent in-vehicle consoles. Each application can have a console specific user interface. The system can also include a single in-vehicle speech processing system implemented separately from the in-vehicle consoles. The speech processing system can execute speech processing tasks responsive to requests received from the applications. That is, the in-vehicle speech processing system can provide speech processing capabilities for the applications. The provided speech processing capabilities can include text-to-speech capabilities and speech recognition capabilities.

Supporting multiple speech enabled user interface consoles within a motor vehicle

An automatic gain control method in accordance with the inventive arrangements can include the following steps. Initially, an audio signal can be provided to an audio device which has a range of permissible signal level settings and a signal level controller for establishing a particular signal level setting. In addition, an actual signal level can be measured for the audio signal at an established signal level setting. The measured actual signal level further can be stored in a volume map along with the corresponding established signal level setting. Following the storage of the measured actual signal level in the volume map, a different signal level setting can be established using the signal level controller. Subsequently, the actual signal level can be re-measured and the re-measured actual signal level and corresponding established different signal level setting can be stored in the volume map. Finally, the volume map can be used during an audio processing session to determine a signal level setting for the audio device, wherein the signal level setting corresponds to a desired actual audio signal level. In one aspect of the present invention, the method can also include detecting a hysteresis condition in the volume map.

Audio device characterization for accurate predictable volume control

A method for marking dictated text for deferred correction or review of dictated text in a speech recognition system proofreader, comprising the steps of: displaying previously dictated text; sequentially highlighting words in the text; selectively establishing a mark for different ones of the sequentially highlighted words responsive to user commands; and, storing the marks in an ordered list, each of the marks including a current position and length of a corresponding marked word, whereby the marked words can be later recalled for correction in accordance with the ordered list. The method can, further comprise the steps of: displaying the previously dictated text in a first display window; sequentially displaying in a second display window a portion of the previously dictated text including the sequentially highlighted word; and, sequentially displaying in a third display window within the second display window the sequentially highlighted word.

Text marking for deferred correction

A method of proofreading and correcting dictated text contained in an electronic document can include selecting proofreading criteria for identifying textual errors contained in the electronic document; playing back each word contained in the electronic document; and, marking as a textual error each played back word in nonconformity with at least one of the proofreading criteria.

Method and system for proofreading and correcting dictated text

A method and a system for improving recall of speech data in a computer speech system can include a plurality of speech cache management steps including providing a speech cache, receiving a speech system input and identifying a speech event in the received speech system input, the speech event comprising speech data. Subsequently, the speech data can be compared to pre-determined speech cache entry criteria; and, if the speech data meets one of the pre-determined entry criteria, at least one entry can be added to the speech cache, the at least one entry corresponding to the speech data. Additionally, the speech data can be compared to pre-determined speech cache exit criteria; and, if the speech data meets one of the pre-determined exit criteria, at least one entry can be purged from the speech cache, the at least one entry corresponding to the speech data. The entry criteria can include frequently used speech data, recently used speech data, and important speech data. Similarly, the exit criteria can include least frequently used speech data associated with each entry in the speech cache, least recently used speech data associated with each entry in the speech cache and least important speed data associated with each entry in the speech cache.

Method of managing a speech cache

A computer system has a notification manager for playing a message to a user by selecting one of a plurality of audio notifications. The method includes the step of setting a priority level for each notification arriving into a queue. The notification is inserted into a position in the queue based upon the priority level of the notification, such that the audio notifications at the queue top have a generally higher priority than audio notifications at the queue bottom. The notification at the top of the queue can be selected if the priority level of the notification is greater than a predetermined gate level. Once a notification is selected, a message corresponding to the selected notification is played to the user.

Audio notification management system

A method for adjusting audio input signal gain in a speech system can include seven steps. First, an upper and a lower threshold can be predetermined in which the upper and lower threshold define an optimal range of audio data signal amplitude measurements. Second, a frame of unpredicted digital audio data samples can be received. Each sample can indicate an amplitude measurement of the audio data signal at a particular point in time. Third, a maximum signal amplitude can be calculated for a configurable measurement percentile of the unpredicted digital audio data samples. Fourth, the audio input signal gain can be incrementally adjusted downward if the maximum signal amplitude exceeds the upper threshold. Conversely, fifth, the audio input signal gain can be incrementally adjusted upward if the maximum signal amplitude falls below the lower threshold. Sixth, additional frames of unpredicted digital audio data samples can be received. Finally, seventh, each of the third through the sixth steps can be repeated with the received additional frames until the calculated maximum signal amplitude falls within the optimal range of audio signal amplitude.

Method for dynamic adjustment of audio input gain in a speech system

A method of proofreading and correcting dictated text contained in an electronic document comprises the steps of: selecting proofreading criteria for identifying textual errors contained in the electronic document; playing back each word contained in the electronic document; and, marking as a textual error each played back word in nonconformity with at least one of the proofreading criteria. The method can further comprise the step of editing each the marked textual error identified in the marking step. In particular, the editing step can include reviewing each the marked textual error identified in the marking step; accepting user specified changes to each marked textual error reviewed in the reviewing step; and, unmarking each marked textual error corrected by the user in the accepting step. Also, the reviewing step can include highlighting each the word in the electronic document corresponding to the marked textual error marked in the marking step; and, displaying an explanation for each marked textual error in a user interface. Moreover, the reviewing step can further include suggesting a recommended change to the marked textual error; displaying the recommended change in the user interface; and, accepting a user specified preference to substitute the recommended change for the marked textual error.

Method of marking previously dictated text for deferred correction in a speech recognition proofreader

A method for inserting a delay between the playback of individual words or phrases by a speech recognition system, comprises the steps of: (A) waiting for a playback command; (B) measuring a delay upon occurrence of the playback command; (C) initiating playback of only one of the individual words or phrases upon expiration of the delay; (D) waiting for a subsequent playback command; and, (E) upon occurrence of the subsequent playback command, repeating the steps (B), (C) and (D) for playing subsequent ones of the individual words or phrases, one at a time. The method can further comprise the steps of: (F) comparing a user requested delay to a predetermined delay; (G) changing from one at a time playback to continuous playback whenever the user requested delay is not greater than the predetermined delay; and, (H) changing from continuous playback to one at a time playback whenever the user requested delay is greater than the predetermined delay.

Variable speed audio playback in speech recognition proofreader

In a TTS system external to a TTS engine, a method for dynamically altering text-to-speech (TTS) attributes of a TTS engine not inherently capable of dynamic attribute alteration is disclosed. The method of the invention can include several steps beginning with the step of representing in the external system a resettable start word position value by a first variable. Second, a current word position offset value can be represented in the external system by a second variable. Third, the second variable can be updated in the external system each time a word of the specified text is played back. Fourth, the TTS playback can be stopped in response to a user request to alter the TTS attribute. Fifth, after stopping the TTS playback, the TTS engine attribute can be altered. Also, after stopping the TTS playback, the first variable can be replaced in the external system with a new start word position value corresponding to the next successive unplayed word of the specified text. Finally, playback of the text can be restarted in accordance with the new word start position value to play back the next successive unplayed word.

Method for dynamically altering text-to-speech (TTS) attributes of a TTS engine not inherently capable of dynamic attribute alteration

A method for managing a speech application, comprising the steps of: categorizing text from a sequential list of playable elements recorded in a dictation session into segments of only dictated playable elements and segments of only non-dictated playable elements; and, playing back the list of playable elements audibly on a segment-by-segment basis, the segments of dictated playable elements being played back from previously recorded audio and the segments of non-dictated playable elements being played back with a text-to-speech engine. The list of playable elements can be played back without having to determine during the playing back, on a playable-element-by-playable-element basis, whether previously recorded audio is available. The list of playable elements can be simultaneously played back audibly and displayed whether the playable elements are dictated or non-dictated.

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Palm Beach Gardens, FL, United States of America

Gary Robert Hanson