Abstract: A system for operating one or more devices using speech input including a receiver for receiving a speech input, a controller in communication with the receiver, software executing on the controller for converting the speech input into computer-readable data, software executing on the controller for generating a table of active commands, the table including a portion of all valid commands of the system, software executing on the controller for identifying at least one active command represented by the data, and software executing on the controller for transmitting the active command to at least one device operable by the active command.

Abstract: A method is shown for supporting a multichannel audio extension at an encoding end of a multichannel audio coding system. In order to improve the audio quality over a large frequency range, the method comprises transforming each channel of a multichannel audio signal into the frequency domain and dividing a bandwidth of the frequency domain signals into a first region of lower frequencies and at least one further region of higher frequencies. Then, the frequency domain signals are encoded in each of the frequency regions with another type of coding to obtain parametric multichannel extension information for the respective frequency region. The invention relates equally to a method for supporting in a corresponding manner a multichannel audio extension at a decoding end. Also shown are a corresponding encoder, a corresponding decoder, and corresponding devices, systems and software program products.

Abstract: A structured document processing apparatus includes an acquisition unit configured to acquire a structured document, a storage unit configured to store a structure model tree which indicates a typical structure of the acquired structured document, a parsing unit configured to parse the acquired structured document, an updating unit configured to update the structure model tree to match a structure of the parsed structured document therewith, a division unit configured to divide the acquired structured document into a plurality of lexical items, and a calculation unit configured to calculate frequency-of-occurrence information indicating locations of each of the lexical items in the acquired structured document.

Abstract: Coding free text documents, especially in medicine, has become an urgent priority as electronic medical records (EMR) mature, and the need to exchange data between EMRs becomes more acute. However, only a few automated coding systems exist, and they can only code a small portion of the free text against a limited number of codes. The precision of these systems is low and code quality is not measured. The present invention discloses a process and system which implements semantic coding against standard lexicon(s) with high precision. The standard lexicon can come from a number of different sources, but is usually developed by a standard's body. The system is semi-automated to enable medical coders or others to process free text documents at a rapid rate and with high precision.

Abstract: A method, system and apparatus for processing multi-lingual syndicated content feeds is provided. In an embodiment of the invention, the system can include a syndicated content aggregator configured for coupling to multiple syndicated content subscribers and syndicated content feed sources. The system also can include a translation server coupled to the aggregator. The translation server can include logic to translate syndicated content feeds, such as RSS or Atom feeds, which are received from the feed sources into target lingual languages specified by the aggregator. In this regard, the translation server can include a machine translator configured to translate content from a first lingual language to a second lingual language.

Abstract: A method for text-to-pronunciation conversion includes a process for searching grapheme-phoneme segments and a three-stage process of text-to-pronunciation conversion. This method looks for a sequence of grapheme-phoneme pairs, which is referred to as a chunk, via a trained pronouncing dictionary, performs grapheme segmentation, chunk marking and a decision process on an input text, and determines a pronouncing sequence for the text. With the chunk marking, the method greatly reduces the search space on the associated phoneme graph, and thereby efficiently enhances the search speed for the candidate chunk sequences. The method keeps a high word-accuracy as well as saves computing time.

Abstract: This invention is a combination of software and hardware components and methodologies that enable voice recognition for multiple users simultaneously. It introduces the concept of a “conversational voice log” and how voice logs are combined to represent the spoken words of a meeting or group conversations. It defines the components needed, command set for control, text output features, and usage of such a system.

Abstract: A method for performing a search of a codebook is provided. The codebook includes a plurality of tracks each having a plurality of even pulse positions. The method includes partitioning a codevector having a plurality of pulses into a first subset of pulses and a second subset of pulses. Each pulse is assignable to a pulse position in the codevector, and each pulse is associated with a shift bit for indicating an odd position. The method also includes performing a first search for determining a first set of possible pulse positions for the pulses in the codevector. The method further includes performing a second search for determining a second set of possible pulse positions for the pulses in the codevector. In addition, the method includes forming the codevector using the first and second sets of possible pulse positions.

Abstract: A pitch detection method and apparatus, the pitch detection apparatus includes: a data rearrangement unit which rearranges voice data on the basis of a center peak of the voice data included in a single frame; a decomposition unit which decomposes rearranged voice data into even symmetrical components on the basis of a center peak; a pitch determination unit which obtains a segment correlation value between a reference point and at least one or more local peaks in relation to even symmetrical components, and determines the location of a local peak corresponding to a maximum segment correlation value among the obtained segment correlation values, as a pitch period.

Abstract: A synthesis subband filter apparatus is provided. The apparatus is used for processing 18 sets of signals which each includes 32 subband sampling signals in accordance with a specification providing 512 window coefficients. The apparatus includes a processor for processing the 18 sets of signals in sequence. The processor further includes a converting module and a generating module. The converting module is used for converting the 32 subband sampling signals of the set of signals being processed into 32 converted vectors by use of 32-points discrete cosine transform (DCT), and writing the 32 converted vectors into 512 default vectors with a first-in, first-out queue. The generating module is used for generating 32 pulse code modulation (PCM) signals, relative to the set of signals being processed according to a set of synthesis formulae proposed in this invention.

Abstract: The present invention is a method and apparatus in a data processing system that includes a Voice over Internet Protocol (VoIP) communication system for improving transmit and receive data paths. The communication system includes a digital signal processing unit. The digital signal processing unit includes a mandatory coder/decoder (codec) that does not include an internal packet loss concealment (PLC) function, an internal voice activity detection (VAD) function, an internal comfort noise generation (CNG) function, or an internal discontinuous transmission generation (DTX) function. The digital signal processing unit also includes an enhanced codec that includes any combination of the following modules all internal to the enhanced codec: internal packet loss concealment (PLC) function, a voice activity detection (VAD) function, a comfort noise generation (CNG) function, and a discontinuous transmission generation (DTX) function.

Abstract: A method of identifying duplicate voice recording by receiving digital voice recordings, selecting one of the recordings; segmenting the selected recording, extracting a pitch value per segment, estimating a total time that voice appears in the recording, removing pitch values that are less than and equal to a user-definable value, identifying unique pitch values, determining the frequency of occurrence of the unique pitch values, normalizing the frequencies of occurrence, determining an average pitch value, determining the distribution percentiles of the frequencies of occurrence, returning to the second step if additional recordings are to be processed, otherwise comparing the total voice time, average pitch value, and distribution percentiles for each recording processed, and declaring the recordings duplicates that compared to within a user-definable threshold for total voice time, average pitch value, and distribution percentiles.

Abstract: An electronic circuit includes storage circuitry and a speech coder coupled with the storage circuitry to have a codebook with sets of track location numbers for respective pulses, the speech coder operable to identify a group of track location numbers in the codebook substantially equally spaced from each other by a pitch lag amount, and make a selection from the group of track location numbers of a selected track location number. Other electronic circuits, processes, methods, devices and systems are disclosed and claimed.

Abstract: An audio information processing system, which when incorporated in home audio video systems, provides independent volume control capability, independent equalization setting capability and independent special effects capability of voice and background sound, to the home audio-video system. The audio information processing system receives an audio signal and extracts there from a voice signal and a background signal based upon correlation of language tracks, correlation of a center channel with surround sound channels, via a voice detection circuit, or via other means. Once the voice signal and background signal are determined, separate processing is performed, and combining of the separately processed voice and background signals may be performed.

Abstract: A system and method for natural language processing comprises a blackboard data structure for providing a shared knowledge repository over which a collection of natural language agents can execute processes on the processable data form, each agent being capable of providing a processing resource usable for serving requests to execute a natural language process on the processable data form, and determining, based on their respective capabilities and examination of the blackboard, what requests for processing they can best serve; and a dispatcher for coordinating the work of registered agents, maintaining a high-level description of tasks to be completed to provide a solution to a given natural language engineering problem, and determining the registered agents that best provide a solution to the given natural language engineering problem.

Abstract: An apparatus for automatically switching language input modes including a first unit, to determine whether to turn on an automatic language input mode switching function; a second unit to bypass a current keystroke input via a predetermined input device when the control signal is an off signal, the second unit either bypasses the current keystroke or deletes previous keystrokes and converts the previous keystroke(s) and the current keystroke into their respective language counterparts according to recognizing a language input mode of a scan code of the current keystroke when the control signal is an on signal; and a third unit to recognize the language input mode of the scan code of the current keystroke by referring to language dictionaries and provide a current language input mode, the recognized language input mode of the current keystroke, keystroke deletion range information, and keystroke conversion range information to the second unit.

Abstract: An apparatus for providing flexible text based language identification includes an alphabet scoring element, an n-gram frequency element and a processing element. The alphabet scoring element may be configured to receive an entry in a computer readable text format and to calculate an alphabet score of the entry for each of a plurality of languages. The n-gram frequency element may be configured to calculate an n-gram frequency score of the entry for each of the plurality of languages. The processing element may be in communication with the n-gram frequency element and the alphabet scoring element. The processing element may also be configured to determine a language associated with the entry based on a combination of the alphabet score and the n-gram frequency score.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: A method for distributing voice-recognition grammars includes receiving match data from a first remote element. The match data includes information associated with an attempt by the remote element to match received audio information to first stored audio data. The method also includes generating a grammar entry based on the match data. The grammar entry includes second stored audio data and a word identifier associated with the second stored audio data. Additionally, the method includes transmitting the grammar entry to a second remote element.

Abstract: Transmitters and receivers in multiple description coding systems use correlating and decorrelating transforms to generate and process multiple descriptions of elements of an input signal. The multiple descriptions include groups of correlating transform coefficients that permit recovery of an inexact facsimile of the signal if some of the correlating transform coefficients are lost or corrupted during transmission. Noiseless implementations of the correlating and decorrelating transforms are described that allow the signal elements to be quantized with different quantizing resolutions. Implementations using the Fast Hadamard Transform are described that reduce the resources needed to perform the transforms.