Abstract: Improving speech capabilities of a multimodal application including receiving, by the multimodal browser, a media file having a metadata container; retrieving, by the multimodal browser, from the metadata container a speech artifact related to content stored in the media file for inclusion in the speech engine available to the multimodal browser; determining whether the speech artifact includes a grammar rule or a pronunciation rule; if the speech artifact includes a grammar rule, modifying, by the multimodal browser, the grammar of the speech engine to include the grammar rule; and if the speech artifact includes a pronunciation rule, modifying, by the multimodal browser, the lexicon of the speech engine to include the pronunciation rule.

Abstract: A method and system for automated code generation in a call flow builder (10) can include a display coupled to a processor. The processor can be programmed to select a real code (database connection) or a prototype code using a graphical interface (20) to provide a selected code and develop a call flow using the selected code. The processor can be programmed to select the prototype code as the selected code, test the call flow in a local development environment and further enable the switching of the selected code from the prototype to the real code to complete a database connection. The processor can be further programmed to enable specification of a default or range of values. Additionally, the processor can be programmed to use a database connection code that replaces a prototype assignment of values to variables when the real code is the selected code.

Abstract: The present invention includes a telecommunication method. The method can include the step of a sending device operating in a first communication mode initializing a communication to a receiving device operating in a second communication mode. The receiving device can be queried to automatically determine that the receiving device is operating in the second communication mode. In one embodiment, the first and second communication modes can include a speech mode and a text mode. The communication can be established between the sending and receiving devices, where the sending device communicates in the first communication mode, and where the receiving device communicates in the second communication mode. During the communication, a format of messages conveyed between the sending device and the receiving device can be automatically changed in accordance with the first communication mode and the second communication mode.

Abstract: A method (100) and system (30) to enable automatic generation and testing of voice applications includes generating (102) a test driver application (TDA) (32) and generating (104) a modified original voice application (34) to be tested by the TDA within a call flow builder (10). The modified application can include or generate (106) “test hooks” or more particularly DTMF tones and DTMF grammars that can be used to synchronize the modified original voice application with the TDA. The TDA can test (110) all possible paths of the modified original voice application. Note the TDA and the modified original voice application can be generated and/or tested (112) in a test environment within the call flow builder or a telephony environment. The TDA can be automatically generated (108) to exercise all possible flows where the DTMF tones define the current state and location of the modified application.

Abstract: The present invention discloses a method for a voice-enabled computing environment. The method includes a step of receiving a series of voice commands related to an abstraction, wherein the voice command specifies an abstraction type. The method further includes responding to the voice command to replay the commands, executing at least one programmatic action related to the abstraction, where the programmatic action is executed demonstrating characteristics specific to the abstraction type associated with the voice command. The present method applies, but is not limited to, the execution of collections of commands that may have diverging timing characteristics.

Abstract: A method for creating and editing an XML-based speech synthesis document for input to a text-to-speech engine is provided. The method includes recording voice utterances of a user reading a pre-selected text and parsing the recorded voice utterances into individual words and periods of silence. The method also includes recording a synthesized speech output generated by a text-to-speech engine, the synthesized speech output being an audible rendering of the pre-selected text, and parsing the synthesized speech output into individual words and periods of silence. The method further includes annotating the XML-based speech synthesis document based upon a comparison of the recorded voice utterances and the recorded synthesized speech output.

Abstract: Methods, systems, and products for testing a grammar used in speech recognition for reliability in a plurality of operating environments having different background noise that include: receiving recorded background noise for each of the plurality of operating environments; generating a test speech utterance for recognition by a speech recognition engine using a grammar; mixing the test speech utterance with each recorded background noise, resulting in a plurality of mixed test speech utterances, each mixed test speech utterance having different background noise; performing, for each of the mixed test speech utterances, speech recognition using the grammar and the mixed test speech utterance, resulting in speech recognition results for each of the mixed test speech utterances; and evaluating, for each recorded background noise, speech recognition reliability of the grammar in dependence upon the speech recognition results for the mixed test speech utterance having that recorded background noise.

Abstract: Methods, apparatus, and products are disclosed for dynamically publishing directory information for a plurality of interactive voice response (‘IVR’) systems through an IVR directory service that include: providing a description of a web services publication interface for the IVR directory service; receiving, on behalf of one or more IVR systems, web services publication requests through the publication interface; determining, in response to the web services publication requests, directory information for each IVR system requesting publication; adding the directory information for each IVR system to an IVR system directory; generating a voice mode user interface to reflect the directory information for each IVR system added to the IVR system directory; and interacting, using the voice mode user interface, with a caller to identify a particular IVR system in dependence upon the IVR system directory and query information provided by the caller and to connect the caller with the identified IVR system.

Abstract: Methods, apparatus, and products are disclosed for signaling correspondence between a meeting agenda and a meeting discussion that include: receiving a meeting agenda specifying one or more topics for a meeting; analyzing, for each topic, one or more documents to identify topic keywords for that topic; receiving meeting discussions among participants for the meeting; identifying a current topic for the meeting in dependence upon the meeting agenda; determining a correspondence indicator in dependence upon the meeting discussions and the topic keywords for the current topic, the correspondence indicator specifying the correspondence between the meeting agenda and the meeting discussion; and rendering the correspondence indicator to the participants of the meeting.

Abstract: Methods, apparatus, and products are disclosed for adjusting a speech engine for a mobile computing device based on background noise, the mobile computing device operatively coupled to a microphone, that include: sampling, through the microphone, background noise for a plurality of operating environments in which the mobile computing device operates; generating, for each operating environment, a noise model in dependence upon the sampled background noise for that operating environment; and configuring the speech engine for the mobile computing device with the noise model for the operating environment in which the mobile computing device currently operates.

Abstract: Methods, apparatus, and products are disclosed for record disambiguation in a multimodal application operating on a multimodal device, the multimodal device supporting multiple modes of interaction including at least a voice mode and a visual mode, that include: prompting, by the multimodal application, a user to identify a particular record among a plurality of records; receiving, by the multimodal application in response to the prompt, a voice utterance from the user; determining, by the multimodal application, that the voice utterance ambiguously identifies more than one of the plurality of records; generating, by the multimodal application, a user interaction to disambiguate the records ambiguously identified by the voice utterance in dependence upon record attributes of the records ambiguously identified by the voice utterance; and selecting, by the multimodal application for further processing, one of the records ambiguously identified by the voice utterance in dependence upon the user interaction.

Abstract: A software method for allocating application tasks between a client and a server can include the step of detecting client-based computing resources for executing at least one application task. At least one indicator of the detected client-based computing resources can be conveyed to a remotely located application server, the application server can determine whether to allocate at least one application task to the client or to a server component based upon at least one indicator.

Abstract: A method of adjusting music length to expected waiting time while a caller is on hold includes choosing one or more media selections based upon their play duration and matching the selection(s) to the expected waiting time.

Abstract: The present invention discloses a solution for customizing synthetic voice characteristics in a user specific fashion. The solution can establish a communication between a user and a voice response system. A data store can be searched for a speech profile associated with the user. When a speech profile is found, a set of speech output characteristics established for the user from the profile can be determined. Parameters and settings of a text-to-speech engine can be adjusted in accordance with the determined set of speech output characteristics. During the established communication, synthetic speech can be generated using the adjusted text-to-speech engine. Thus, each detected user can hear a synthetic speech generated by a different voice specifically selected for that user. When no user profile is detected, a default voice or a voice based upon a user's speech or communication details can be used.

Abstract: The present invention discloses a system and a method for creating a reduced script, which is read by a voice talent to create a concatenative text-to-speech (TTS) voice. The method can automatically process pre-recorded audio to derive speech assets for a concatenative TTS voice. The pre-recording audio can include sets of recorded phrases used by a speech user interface (Sill). A set of unfulfilled speech assets needed for foil phonetic coverage of the concatenative TTS voice can be determined. A reduced script can be constructed that includes a set of phrases, which when read by a voice talent result in a reduced corpus. When the reduced corpus is automatically processed, a reduced set of speech assets result. The reduced set includes each of the unfulfilled speech assets. When this reduced corpus is combined with existing speech assets the result will be a voice with a complete set of speech assets.

Abstract: A method of adjusting music length to expected waiting time while a caller is on hold includes choosing one or more media selections based upon their play duration and matching the selection(s) to the expected waiting time.

Abstract: The present invention discloses a concept and a use of audio anchors within voice-enabled interfaces. Audio anchors can be user configurable points from which audio playback occurs. In the invention, a user can identify an interface position at which an audio anchor is to be established. The computing device can determine an anchor direction setting, with values that include forward playback and backward playback. Interface items can then be audibly enumerated from the audio anchor in a direction indicated by the anchor direction setting. For example, if a set of interface items are alphabetically ordered items and if an audio anchor is set at a first item beginning with a letter “G” and an anchor direction is set to indicate backward playback, then the interface items beginning with letters “A-F” can be audibly played in reverse alphabetical order. Additionally, a rate of audio playback can be user adjustable.

Abstract: Methods, apparatus, and computer program products are described for automatic speech recognition (‘ASR’) that include accepting by the multimodal application speech input and visual input for selecting or deselecting items in a selection list, the speech input enabled by a speech recognition grammar; providing, from the multimodal application to the grammar interpreter, the speech input and the speech recognition grammar; receiving, by the multimodal application from the grammar interpreter, interpretation results including matched words from the grammar that correspond to items in the selection list and a semantic interpretation token that specifies whether to select or deselect items in the selection list; and determining, by the multimodal application in dependence upon the value of the semantic interpretation token, whether to select or deselect items in the selection list that correspond to the matched words.

Abstract: A method for prompting user input for a multimodal interface including the steps of providing a multimodal interface to a user, where the interface includes a visual interface having a plurality of input regions, each having at least one input field; selecting an input region and processing a multi-token speech input provided by the user, where the processed speech input includes at least one value for at least one input field of the selected input region; and storing at least one value in at least one input field.

Abstract: A wizard that from a fixed design can create various audio interfaces. The generated interfaces can be speech only, DTMF only, or various mixed speech and DTMF UIs. When specifying both speech and DTMF prompts, a number of combinations of these interfaces could be automatically generated. Robust speech recognition systems can be built by automatically generating a “shadow” DTMF application. The DTMF application will perform the same task as the primary speech application; however the transfer to a DTMF application could be done explicitly by the user, or could be transferred automatically (either a temporary or permanent transition) at a point in the call flow where there was a problem with the speech recognition.