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Patents

 
System and method for interacting with live agents in an automated call center
Tue, 27 Jan 2015 08:00:00 EST
Embodiments of an interface system that enables a call center agent to access and intervene in an interaction between an automated call center system and a caller whenever necessary for complex application tasks is described. The system includes a user interface that presents the agent with one or more categories of information, including the conversation flow, obtained semantic information, the recognized utterances, and access to the utterance waveforms. This information is cross-linked and attached with a confidence level for better access and navigation within the dialog system for the generation of appropriate responses to the caller.
Apparatus, system, and method for natural language processing
Tue, 27 Jan 2015 08:00:00 EST
Various embodiments are described for searching and retrieving documents based on a natural language input. A computer-implemented natural language processor electronically receives a natural language input phrase from an interface device. The natural language processor attributes a concept to the phrase with the natural language processor. The natural language processor searches a database for a set of documents to identify one or more documents associated with the attributed concept to be included in a response to the natural language input phrase. The natural language processor maintains the concepts during an interactive session with the natural language processor. The natural language processor resolves ambiguous input patterns in the natural language input phrase with the natural language processor. The natural language processor includes a processor, a memory and/or storage component, and an input/output device.
Speech coding of principal-component channels for deleting redundant inter-channel parameters
Tue, 27 Jan 2015 08:00:00 EST
Disclosed is an audio encoding device which removes unnecessary inter-channel parameters from the subject to be encoded, improving the encoding efficiency thereby. In this audio encoding device, a principal component analysis unit (301) converts an inputted left signal {Lsb(f)} and an inputted right signal {Rsb(f)} into a principal component signal {PCsb(f)} and an ambient signal {Asb(f)} and calculates for each sub-band, a rotation angle which indicates the degree of conversion; a monophonic encoding unit (303) encodes the principal component signal {Pcsb(f)}; a rotation angle encoding unit (302) encodes the angle of rotation {θb}; a local monophonic decoding unit (603) creates a decoded principal component signal; and a redundant parameter elimination unit (604) identifies the redundant parameters by analyzing the encoding quality of the decoded principal component signal and eliminates the redundant parameters from the signal to be encoded.
Efficient temporal envelope coding approach by prediction between low band signal and high band signal
Tue, 27 Jan 2015 08:00:00 EST
This invention provides a more efficient way to quantize temporal envelope shaping of high band signal by benefiting from energy relationship between low band signal and high band signal; if low band signal is well coded or it is coded with time domain codec such as CELP, temporal envelope shaping information of low band signal can be used to predict temporal envelope shaping of high band signal; the temporal envelope shaping prediction can bring significant saving of bits to precisely quantize temporal envelope shaping of high band signal. This prediction approach can be combined with other specific approach to further increase the efficiency and save mores bits.
Identifying qualified audio of a plurality of audio streams for display in a user interface
Tue, 27 Jan 2015 08:00:00 EST
A clear picture of who is speaking in a setting where there are multiple input sources (e.g., a conference room with multiple microphones) can be obtained by comparing input channels against each other. The data from each channel can not only be compared, but can also be organized into portions which logically correspond to statements by a user. These statements, along with information regarding who is speaking, can be presented in a user friendly format via an interactive timeline which can be updated in real time as new audio input data is received.
Determining user intent based on ontologies of domains
Tue, 27 Jan 2015 08:00:00 EST
Methods, systems, and computer readable storage medium related to operating an intelligent digital assistant are disclosed. A plurality of predefined domains each representing a respective area of service offered by an intelligent automated assistant are stored. A text string derived from a user request is obtained, the text string including at least one or more words derived from a speech input received from a user. From the plurality of predefined domains, a relevant domain for the user request is identified based at least on respective degrees of match between the text string derived from the user request and a respective plurality of words associated with each predefined domain. A task is executed in accordance with steps specified in a task flow associated with the relevant domain, and in accordance with one or more task parameters derived from the user request.
Centralized method and system for clarifying voice commands
Tue, 27 Jan 2015 08:00:00 EST
A method and system for facilitating centralized interaction with a user includes providing a recognized voice command to a plurality of application modules. A plurality of interpretations of the voice command are generated by at least one of the plurality of application modules. A centralized interface module visually renders the plurality of interpretations of the voice command on a centralized display. An indication of selection of an interpretation is received from the user.
Method and system for controlling external output of a mobile device
Tue, 27 Jan 2015 08:00:00 EST
A method and system is provided that controls an external output function of a mobile device according to control interactions received via the microphone. The method includes, activating a microphone according to preset optional information when the mobile device enters an external output mode, performing an external output operation in the external output mode, detecting an interaction based on sound information in the external output mode, and controlling the external output according to the interaction.
Method of speech synthesis
Tue, 27 Jan 2015 08:00:00 EST
The present invention relates to a method of text-based speech synthesis, wherein at least one portion of a text is specified; the intonation of each portion is determined; target speech sounds are associated with each portion; physical parameters of the target speech sounds are determined; speech sounds most similar in terms of the physical parameters to the target speech sounds are found in a speech database; and speech is synthesized as a sequence of the found speech sounds. The physical parameters of said target speech sounds are determined in accordance with the determined intonation. The present method, when used in a speech synthesizer, allows improved quality of synthesized speech due to precise reproduction of intonation.
Semiconductor integrated circuit device and electronic instrument
Tue, 27 Jan 2015 08:00:00 EST
A semiconductor integrated circuit device including: a storage section which temporarily stores a command and text data input from the outside; a speech synthesis section which synthesizes a speech signal corresponding to the text data based on the command and the text data stored in the storage section, and outputs the synthesized speech signal to the outside; and a control section which controls a timing at which the command and the text data stored in the storage section are transferred to the speech synthesis section based on a speech synthesis start control signal. The control section controls an output of a speech output start notification signal which notifies in advance a start of outputting the synthesized speech signal to the outside based on occurrence of a speech synthesis start event, and then controls a start of outputting the synthesized speech signal to the outside at a given timing.
Natural language call router
Tue, 27 Jan 2015 08:00:00 EST
A natural language call router forwards an incoming call from a caller to an appropriate destination. The call router has a speech recognition mechanism responsive to words spoken by a caller for producing recognized text corresponding to the spoken words. A robust parsing mechanism is responsive to the recognized text for detecting a class of words in the recognized text. The class is defined as a group of words having a common attribute. An interpreting mechanism is responsive to the detected class for determining the appropriate destination for routing the call.
Method of navigating in a sound content
Tue, 27 Jan 2015 08:00:00 EST
A method of navigating in a sound content wherein at least one key word is stored in association with at least two positions representative of said key word in the sound content, and wherein the method comprises: a step of displaying a representation of the sound content; during playback of the sound content, a step of detecting a current extract representative of a key word stored at a first position; a step of determining at least one second extract representative of said key word and a second position as a function of the stored positions; and a step of highlighting the position of the extracts in the representation of the sound content. The invention also relates to a system adapted to implement the navigation method.
Acoustic processing apparatus and method
Tue, 27 Jan 2015 08:00:00 EST
An acoustic processing apparatus is provided. The acoustic processing apparatus including a first extracting unit configured to extract a first acoustic model that corresponds with a first position among positions set in a speech recognition target area, a second extracting unit configured to extract at least one second acoustic model that corresponds with, respectively, at least one second position in proximity to the first position, and an acoustic model generating unit configured to generate a third acoustic model based on the first acoustic model, the second acoustic model, or a combination thereof.
Parameter learning in a hidden trajectory model
Tue, 27 Jan 2015 08:00:00 EST
Parameters for distributions of a hidden trajectory model including means and variances are estimated using an acoustic likelihood function for observation vectors as an objection function for optimization. The estimation includes only acoustic data and not any intermediate estimate on hidden dynamic variables. Gradient ascent methods can be developed for optimizing the acoustic likelihood function.
System and method for speech recognition using pitch-synchronous spectral parameters
Tue, 27 Jan 2015 08:00:00 EST
The present invention defines a pitch-synchronous parametrical representation of speech signals as the basis of speech recognition, and discloses methods of generating the said pitch-synchronous parametrical representation from speech signals. The speech signal is first going through a pitch-marks picking program to identify the pitch periods. The speech signal is then segmented into pitch-synchronous frames. An ends-matching program equalizes the values at the two ends of the waveform in each frame. Using Fourier analysis, the speech signal in each frame is converted into a pitch-synchronous amplitude spectrum. Using Laguerre functions, the said amplitude spectrum is converted into a unit vector, referred to as the timbre vector. By using a database of correlated phonemes and timbre vectors, the most likely phoneme sequence of an input speech signal can be decoded in the acoustic stage of a speech recognition system.
Method and device for noise reduction control using microphone array
Tue, 27 Jan 2015 08:00:00 EST
The present invention provides a noise reduction control method using a microphone array and a noise reduction control device using a microphone array wherein the method comprises the steps of: S1: collecting, by the microphone array, acoustic signals; S2: estimating incidence angles of all acoustic signals of the microphone array; S3: conducting a statistics on signal components according to incidence angles; S4: determining a parameter α from a ratio of noise components according to the statistical result and using the parameter α as a control parameter for controlling an adaptive filter. With the present invention, space position information of the sound is obtained directly with the microphone array to control update of the adaptive filter more accurately, so as to eliminate noise, enhance SNR and protect speech quality well at the same time.
Noise suppression in a Mel-filtered spectral domain
Tue, 27 Jan 2015 08:00:00 EST
Techniques are described herein that suppress noise in a Mel-filtered spectral domain. For example, a window may be applied to a representation of a speech signal in a time domain. The windowed representation in the time domain may be converted to a subsequent representation of the speech signal in the Mel-filtered spectral domain. A noise suppression operation may be performed with respect to the subsequent representation to provide noise-suppressed Mel coefficients.
Method and system for determining device settings at device initialization
Tue, 27 Jan 2015 08:00:00 EST
A device searches for an available network. The device automatically sends a request message to an identified available network. After the device receives a response message from the identified available network, the device selects a language and/or other device setting based on contents of the response message.
Content page URL translation
Tue, 27 Jan 2015 08:00:00 EST
The present technology may translate a content of a web page such as content locator (e.g., a uniform resource locator (URL)) from a source language to a target language. The content locator may be associated with a content page. The translation may involve dividing the content locator into segment tokens in a first language, followed by translating, transliterating or not changing a segment token. The processed tokens are then reassembled in a second language. The translation may be provided by a translation module through a content page provided by a network browser.
Systems and methods for data loss prevention in bilingual text messages with a data loss policy which recognizes only one primary language
Tue, 27 Jan 2015 08:00:00 EST
A computer-implemented method for data loss prevention may include 1) identifying a network configured with a data loss prevention system including at least one data loss prevention policy directed to textual data expressed in a primary natural language, 2) identifying a textual object subject to a data loss prevention assessment within the network, 3) determining that the textual object includes a textual component that is not expressed in the primary natural language, 4) in response to determining that the textual object includes the textual component, translating the textual component from a secondary natural language that the data loss prevention policy does not recognize into the primary natural language that the data loss prevention policy does recognize, and 5) after translating the textual component into the primary natural language, applying the data loss prevention policy to a modified textual object including the translated textual component. Various other methods, systems, and computer-readable media are also disclosed.
Automated project localization into multiple languages by using machine translation
Tue, 27 Jan 2015 08:00:00 EST
Applications can be localized by localization experts to allow them to be used by a broader customer base. The localization can be done given resource files containing localization resources. A localization resource may contain a programming component and non-programming component. The non-programming component can be sent to a machine localizer. The machine localizer may provide a plurality of localizations corresponding to a plurality of languages based on the non-programming component of the localization resource. A plurality of localized applications can be complied based on the localized non-programming components and the original programming component.
Humanoid robot equipped with a natural dialogue interface, method for controlling the robot and corresponding program
Tue, 27 Jan 2015 08:00:00 EST
A humanoid robot equipped with an interface for natural dialog with an interlocutor is provided. Previously, the modalities of dialog between humanoid robots equipped moreover with evolved displacement functionalities and human beings are limited notably by the capabilities for voice and visual recognition processing that can be embedded onboard said robots. The present disclosure provides robots are presently equipped with capabilities to resolve doubt on a several modalities of communication of the messages that they receive and for combining these various modalities which make it possible to greatly improve the quality and the natural character of dialogs with the robots' interlocutors. This affords simple and user-friendly means for carrying out the programming of the functions making it possible to ensure the fluidity of these multimodal dialogs.
Real-time quality monitoring of speech and audio signals in noisy reverberant environments for teleconferencing systems
Tue, 27 Jan 2015 08:00:00 EST
A method for real-time monitoring of audio signals reception quality includes receiving output signals from a plurality of microphone clusters, each microphone cluster having at least two microphone units to receive audio signals from at least two distinct directions and output corresponding electrical signals; identifying comparative features of output signals for each of the microphone clusters; and selecting at least one microphone cluster based on the identified features. A system for real-time monitoring of audio signals reception quality includes a plurality of microphone clusters, each microphone cluster having at least two microphone units to receive audio signals from at least two distinct directions and output corresponding electrical signals; and a main audio unit to identify comparative features of output signals for each of the microphone clusters and to select at least one microphone cluster based on the identified features.
System and method for three-way call detection
Tue, 27 Jan 2015 08:00:00 EST
A system for detecting three-way calls in a monitored telephone conversation includes as speech recognition processor that transcribes the monitored telephone conversation and associates Characteristics of the monitored telephone conversation with a transcript thereof, a database to store the transcript and the characteristics associated therewith, and a three-way Call detection processor to analyze the characteristics of the conversation and to detect therefrom the addition of one or more parties to the conversation. The system preferably includes at least one domain-specific language model that the speech recognition processor utilizes to transcribe the conversation. The system may operate in real-time or on previously recorded conversations. A query and retrieval system may be used to retrieve and review call records from the database.

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