Abstract: A method for improving charging information accuracy in a telecommunications network with respect to a target user equipment includes: in case of a handover or cell change procedure from a first base transceiver station to a second base transceiver station regarding the target user equipment, transmitting first non-delivered packet data volume information from the first base transceiver station to the second base transceiver station using at least one of the following: an X2 interface between the first base transceiver station and the second base transceiver station, an S1 interface between the first base transceiver station and the core network and the S1 interface between the core network and the second base transceiver station, and user equipment history information regarding the target user equipment; and determining, based on the first non-delivered packet data volume information, the second non-delivered packet data volume information.

Abstract: A method for improving charging information accuracy in a telecommunications network with respect to a target user equipment includes: in case of a handover or cell change procedure from the first base transceiver station to the second base transceiver station regarding the target user equipment, transmitting the first non-delivered packet data volume information from the first base transceiver station to the target user equipment and from the target user equipment to the second base transceiver station; and determining, based on the first non-delivered packet data volume information, the second non-delivered packet data volume information.

Abstract: Aspects of the disclosure provide systems and methods for facilitating dictation. Speech input may be provided to an audio input device of a computing device. A speech recognition engine at the computing device may obtain text corresponding to the speech input. The computing device may transmit the text to a remotely-located storage device. A login webpage that includes a session identifier may be accessed from a target computing device also located remotely relative to the storage device. The session identifier may be transmitted to the storage device and, in response, a text display webpage may be received at the target computing device. The text display webpage may include the speech-derived text and may be configured to automatically copy the text to a copy buffer of the target computing device. The speech-derived text may also be provided to native applications at target computing devices or NLU engines for natural language processing.

Abstract: A method for enhancing the use of proximity services in a public land mobile network includes: in a first step, a base transceiver station receives a first proximity services activation information related to a first user equipment of a plurality of user equipments; in a second step, the base transceiver station receives a second proximity services activation information related to a second user equipment of the plurality of user equipments; in a third step, a proximity services broadcast information is sent from the base transceiver station to the plurality of user equipments within the geographic area; and in a fourth step, the first user equipment starts to emit a first proximity services discovery signal and the second user equipment starts to emit a second proximity services discovery signal.

Abstract: A multi-modal user interface is described that hides response latency delays. A graphical user interface (GUI) supports different user input modalities including low delay inputs which respond to user inputs without significant delay, and high latency inputs which have a significant response latency after receiving a user input before providing a corresponding completed response. The GUI accepts user inputs in a sequence of mixed input modalities independently of response latencies without waiting for responses to high latency inputs, and responds to each user input in the sequence as if each preceding user input in the sequence had been performed without response latencies.

Abstract: A system and method for license detection is provided. Embodiments may include receiving, at a device, a recording of a speech signal associated with an audio recording application and determining if the audio recording application has an active license. Embodiments may also include inserting license reminder language to the speech signal if it is determined that the audio recording application does not have an active license and ignoring the license reminder language if it is determined that the audio recording application does have an active license. Embodiments may further include producing an output signal wherein the output signal includes the license reminder language if it is determined that the audio recording application does not have an active license and producing an output signal without the license reminder language if it is determined that the audio recording application does have an active license.

Abstract: A system and method for allowing user intervention in a speech recognition pipeline is presented. Embodiments may include receiving, at a computing device, a speech signal at a speech recognition engine, the speech signal being associated with an application. Embodiments may further include generating one or more suggested speech results at the speech recognition engine, the suggested speech results based upon, at least in part, the speech signal. Embodiments may also include displaying, at a graphical user interface associated with the computing device, the one or more suggested speech results prior to applying a final speech result. Embodiments may further include receiving a non voice-based selection of at least one of the one or more suggested speech results and applying the non voice-based selection to the application.

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: A method for improving charging information accuracy in a telecommunications network with respect to a target user equipment includes: in case of a handover or cell change procedure from a first base transceiver station to a second base transceiver station regarding the target user equipment, transmitting first non-delivered packet data volume information from the first base transceiver station to the second base transceiver station using at least one of the following: an X2 interface between the first base transceiver station and the second base transceiver station, an S1 interface between the first base transceiver station and the core network and the S1 interface between the core network and the second base transceiver station, and user equipment history information regarding the target user equipment; and determining, based on the first non-delivered packet data volume information, the second non-delivered packet data volume information.

Abstract: A method for improving charging information accuracy in a telecommunications network with respect to a target user equipment includes: in case of a handover or cell change procedure from the first base transceiver station to the second base transceiver station regarding the target user equipment, transmitting the first non-delivered packet data volume information from the first base transceiver station to the target user equipment and from the target user equipment to the second base transceiver station; and determining, based on the first non-delivered packet data volume information, the second non-delivered packet data volume information.

Abstract: At design time of a graphical user interface (GUI), a software component (VUIcontroller) is added to the GUI. At run time of the GUI, the VUIcontroller analyzes the GUI from within a process that executes the GUI. From this analysis, the VUIcontroller automatically generates a voice command set, such as a speech-recognition grammar, that corresponds to controls of the GUI. The generated voice command set is made available to a speech recognition engine, thereby speech-enabling the GUI. Optionally, a GUI designer may add properties to ones of the GUI controls at GUI design time, without necessarily writing a voice command set. These properties, if specified, are then used at GUI run time to control or influence the analysis of the GUI and the automatic generation of the voice command set.

Abstract: A method for enhancing the use of proximity services in a public land mobile network includes: in a first step, a base transceiver station receives a first proximity services activation information related to a first user equipment of a plurality of user equipments; in a second step, the base transceiver station receives a second proximity services activation information related to a second user equipment of the plurality of user equipments; in a third step, a proximity services broadcast information is sent from the base transceiver station to the plurality of user equipments within the geographic area; and in a fourth step, the first user equipment starts to emit a first proximity services discovery signal and the second user equipment starts to emit a second proximity services discovery signal.

Abstract: A method for synchronizing sound data and text data, said text data being obtained by manual transcription of said sound data during playback of the latter. The proposed method comprises the steps of repeatedly querying said sound data and said text data to obtain a current time position corresponding to a currently played sound datum and a currently transcribed text datum, respectively, correcting said current time position by applying a time correction value in accordance with a transcription delay, and generating at least one association datum indicative of a synchronization association between said corrected time position and said currently transcribed text datum. Thus, the proposed method achieves cost-effective synchronization of sound and text in connection with the manual transcription of sound data.

Abstract: A method for synchronizing sound data and text data, said text data being obtained by manual transcription of said sound data during playback of the latter. The proposed method comprises the steps of repeatedly querying said sound data and said text data to obtain a current time position corresponding to a currently played sound datum and a currently transcribed text datum, respectively, correcting said current time position by applying a time correction value in accordance with a transcription delay, and generating at least one association datum indicative of a synchronization association between said corrected time position and said currently transcribed text datum. Thus, the proposed method achieves cost-effective synchronization of sound and text in connection with the manual transcription of sound data.

Abstract: A method for synchronizing sound data and text data, said text data being obtained by manual transcription of said sound data during playback of the latter. The proposed method comprises the steps of repeatedly querying said sound data and said text data to obtain a current time position corresponding to a currently played sound datum and a currently transcribed text datum, respectively, correcting said current time position by applying a time correction value in accordance with a transcription delay, and generating at least one association datum indicative of a synchronization association between said corrected time position and said currently transcribed text datum. Thus, the proposed method achieves cost-effective synchronization of sound and text in connection with the manual transcription of sound data.

Abstract: In a method and a system (20) for creating or updating entries in a speech recognition (SR) lexicon (7) of a speech recognition system, said entries mapping speech recognition (SR) phoneme sequences to words, said method comprising entering a respective word, and in the case that the word is a new word to be added to the SR lexicon, also entering at least one associated SR phoneme sequence through input means (26), it is provided that the SR phoneme sequence associated with the respective word is converted into speech by phoneme to speech conversion means (4.4), and the speech is played back by playback means (28), to control the match of the phoneme sequence and the word.

Abstract: A method for preventing ping-pong handover in mobile radio networks having a multiplicity of stationary base stations at a distance from one another, and also stationary and/or moveable mobile radio stations, such as mobile radio telephones, which are in the transmission and reception range of the base stations. Repeated unwanted handover of a mobile radio station in the connected/active mode from a source base station to a destination base station is reduced or prevented. System messages are transmitted between the mobile radio stations and the base station, wherein the system messages for initiating the handover are used to transmit historical information relating to handovers which have already been performed, that information being used to automatically ascertain and set optimum operating parameters for the mobile radio stations and/or at least one of the base stations which are involved.

Abstract: A multi-modal user interface with increased responsiveness is described. A graphical user interface (GUI) supports multiple different user input modalities including low delay inputs which respond to user inputs without significant delay, and high latency inputs which have a significant response latency after receiving a user input before providing a corresponding completed response. The GUI accepts user inputs in a sequence of mixed input modalities independently of response latencies without waiting for responses to high latency inputs. The GUI also provides interim indication during response latencies of pending responses at a position in the GUI where the completed response will be presented.

Abstract: A multi-modal user interface is described that hides response latency delays. A graphical user interface (GUI) supports different user input modalities including low delay inputs which respond to user inputs without significant delay, and high latency inputs which have a significant response latency after receiving a user input before providing a corresponding completed response. The GUI accepts user inputs in a sequence of mixed input modalities independently of response latencies without waiting for responses to high latency inputs, and responds to each user input in the sequence as if each preceding user input in the sequence had been performed without response latencies.

Abstract: At design time of a graphical user interface (GUI), a software component (VUIcontroller) is added to the GUI. At run time of the GUI, the VUIcontroller analyzes the GUI from within a process that executes the GUI. From this analysis, the VUIcontroller automatically generates a voice command set, such as a speech-recognition grammar, that corresponds to controls of the GUI. The generated voice command set is made available to a speech recognition engine, thereby speech-enabling the GUI. Optionally, a GUI designer may add properties to ones of the GUI controls at GUI design time, without necessarily writing a voice command set. These properties, if specified, are then used at GUI run time to control or influence the analysis of the GUI and the automatic generation of the voice command set.