Abstract: An audio data transmitting/receiving apparatus for realizing a high-quality frame compensation in audio communications. In an audio data transmitting apparatus (10), a delay part (104) subjects multi-channel audio data to a delay process that delays the L-ch encoded data relative to the R-ch encoded data by a predetermined delay amount. A multiplexing part (106) multiplexes the audio data as subjected to the delay process. A transmitting part (108) transmits the audio data as multiplexed. In an audio data receiving apparatus (20), a separating part (114) separates, for each channel, the audio data received from the audio data transmitting apparatus (10). A decoding part (118) decodes, for each channel, the audio data as separated. If there has occurred a loss or error in the audio data as separated, then a frame compensating part (120) uses one of the L-ch and R-ch encoded data to compensate for the loss or error in the other encoded data.

Abstract: Disclosed are systems, methods and computer-readable media for controlling a computing device to provide contextual responses to user inputs. The method comprises receiving a user input, generating a set of features characterizing an association between the user input and a conversation context based on at least a semantic and syntactic analysis of user inputs and system responses, determining with a data-driven machine learning approach whether the user input begins a new topic or is associated with a previous conversation context and if the received question is associated with the existing topic, then generating a response to the user input using information associated with the user input and any previous user input associated with the existing topic.

Abstract: Methods and mobile devices are provided for asymmetric independent processing of audio streams in a system on a chip (SOC). More specifically, independent audio paths are provided for processors performing audio processing on the SOC and mixing of decoded audio samples from the processors is performed digitally on the SOC by a hardware digital mixer.

Abstract: A system for facilitating the translation of messages from an original language to a language preferred by an end user is provided. The system allows distributors, end users or others to create translated versions of messages used by an application program, while protecting against changes that might affect operation of the application program. More particularly, a new message editor application ensures that proper formatting of messages is utilized, and protects verified messages against unauthorized editing using a data conformance stamp. Before applying a new message, the application program recreates the data conformance stamp to ensure that the new message has not been altered outside of the message editor application.

Abstract: A voice recognition device is provided with a sentence selecting unit for selecting sentences each having a predetermined number or less of acoustic units as sentences to be recognized from candidates for the sentences to be recognized. According to an exemplary embodiment, the sentence selecting unit may define the selected sentences as sentences to be recognized. According to another exemplary embodiment, when a candidate for a sentence to be recognized has a number of acoustic units exceeding the predetermined number, one or more acoustic units may be trimmed from the candidate, and the trimmed candidate may be defined as one of the sentences to be recognized.

Abstract: Cross-channel interference is eliminated and multi-channel sources are separated by estimating a source absence probability for a current frame of a first channel output, and determining an interference elimination coefficient for matching a secondary signal of the first channel output with a primary signal of a second channel output by using the source absence probability, generating an interference signal by multiplying the second channel output by an over-subtraction factor and the interference elimination coefficient, wherein a partial differentiation is performed for a v-norm value of a spectral amplitude difference, between the first channel output and the second channel output multiplied by the interference elimination coefficient and a result of multiplication of the source absence probability, by using the interference elimination coefficient to determine an update amount of the interference elimination coefficient for a next frame.

Abstract: A method and computer system for translating sentences between languages from an intermediate language-independent semantic representation is provided. On the basis of comprehensive understanding about languages and semantics, exhaustive linguistic descriptions are used to analyze sentences, to build syntactic structures and language independent semantic structures and representations, and to synthesize one or more sentences in a natural or artificial language. A computer system is also provided to analyze and synthesize various linguistic structures and to perform translation of a wide spectrum of various sentence types. As result, a generalized data structure, such as a semantic structure, is generated from a sentence of an input language and can be transformed into a natural sentence expressing its meaning correctly in an output language.

Abstract: A distributed intelligibility testing system provides standardized audio tests to a plurality of remotely located client systems. The testing system includes a test manager that records a plurality of audio test words based on established intelligibility standards and generates a test protocol corresponding to the audio test words. A database receives and stores the audio test words and the test protocol. The audio test words are stored as a plurality of audio test files. Respective client systems in communication with the database receive and play the audio test files in accordance with the test protocol. The client systems record test responses when the audio test files are played. The test responses are stored in a database, and then evaluated.

Abstract: A compressor device for a compander system has a level detecting/control device and a pre-emphasis device for carrying out an adaptive pre-emphasis filtering. The invention is also directed to an expander device for a compander system with a level detecting/control device and a de-emphasis device for carrying out an adaptive de-emphasis filtering.

Abstract: A system and method for providing an augmented version of a Low-Complexity Sub-band Coder (LC-SBC) is described herein. In accordance with the method, a series of input audio samples representative of the frame are received. A series of sub-band samples is generated for each of a plurality of frequency sub-bands based on the input audio samples. A determination is made as to whether the frame is a voice frame or a noise frame. Responsive to a determination that the frame is a noise frame, an index representative of a previously-processed series of sub-band samples stored in a history buffer for at least one of the frequency sub-bands is encoded instead of encoding the series of sub-band samples generated for the frequency sub-band.

Abstract: A handheld electronic device includes a reduced QWERTY keyboard and is enabled with disambiguation software that is operable to disambiguate text input. In addition to identifying and outputting representations of language objects that are stored in the memory and that correspond with a text input, the device provides a learning function which facilitates providing proposed corrected output by the device in certain circumstances of erroneous input.

Abstract: A system and method are disclosed that improve automatic speech recognition in a spoken dialog system. The method comprises partitioning speech recognizer output into self-contained clauses, identifying a dialog act in each of the self-contained clauses, qualifying dialog acts by identifying a current domain object and/or a current domain action, and determining whether further qualification is possible for the current domain object and/or current domain action. If further qualification is possible, then the method comprises identifying another domain action and/or another domain object associated with the current domain object and/or current domain action, reassigning the another domain action and/or another domain object as the current domain action and/or current domain object and then recursively qualifying the new current domain action and/or current object. This process continues until nothing is left to qualify.

Abstract: The invention provides a multi-language communication method, comprising: providing to each of the persons access to a plurality of language options; receiving from each of one or more of the persons an indication of a respective choice of language chosen by the respective person from the language options; storing the language choices in a database; and making the language choices accessible by one or more applications so that the applications can when operated selectively provide output to each respective person in the respective language choice.

Abstract: Certain aspects of a method and system for a dual mode subband acoustic echo canceller with integrated noise suppression may include splitting an input signal into a lowband component and a highband component. The subbands of each of the lowband component and the highband component may be processed in order to reduce an echo associated with the input signal and to suppress the noise associated with the input signal.

Abstract: A system and method for automated dictionary population is provided to facilitate the entry of textual material in dictionaries for enhancing word prediction. The automated dictionary population system is useful in association with a mobile device including at least one dictionary which includes entries. The device receives a communication which is parsed and textual data extracted. The text is compared to the entries of the dictionaries to identify new words. Statistical information for the parsed words, including word usage frequency, recency, or likelihood of use, is generated. Profanities may be processed by identifying profanities, modifying the profanities, and asking the user to provide feedback. Phrases are identified by phrase markers. Lastly, the new words are stored in a supplementary word list as single words or by linking the words of the identified phrases to preserve any phrase relationships. Likewise, the statistical information may be stored.

Abstract: Apparatus and methods are provided for using automatic speech recognition to analyze a voice interaction and verify compliance of an agent reading a script to a client during the voice interaction. In one aspect of the invention, a communications system includes a user interface, a communications network, and a call center having an automatic speech recognition component. In other aspects of the invention, a script compliance method includes the steps of conducting a voice interaction between an agent and a client and evaluating the voice interaction with an automatic speech recognition component adapted to analyze the voice interaction and determine whether the agent has adequately followed the script. In still other aspects of the invention, respective data for specific interactions, including voice records or panels, transcribed text equivalents to such voice records or panels, and/or script text governing such voice records or panels, can be forwarded to one or more respective clients.

Abstract: In one embodiment of the method, a block of digital audio data is stored in a buffer, and a set of parcor values are calculated for the block of digital audio data. Then, the parcor values are entropy encoded for transmission. The entropy coding includes selecting a table including offset values and entropy parameters from a plurality of tables based on a sampling rate of the block of digital audio data, wherein each offset is associated with a coded-coefficient and is different for each coded-coefficient in the first twenty coefficients, calculating parcor residual values based on offset values from the selected table, and encoding the parcor residual values using entropy codes defined by entropy parameters from the selected table.

Abstract: Enhancing noisy speech recognition accuracy by receiving geotagged audio signals that correspond to environmental audio recorded by multiple mobile devices in multiple geographic locations, receiving an audio signal that corresponds to an utterance recorded by a particular mobile device, determining a particular geographic location associated with the particular mobile device, selecting a subset of geotagged audio signals and weighting each geotagged audio signal of the subset based on whether the respective audio signal was manually uploaded or automatically updated, generating a noise model for the particular geographic location using the subset of weighted geotagged audio signals, where noise compensation is performed on the audio signal that corresponds to the utterance using the noise model that has been generated for the particular geographic location.

Abstract: A text input for a messaging application is performed on the basis of the language stored respective to the input or the selected contact item in the memory.

Abstract: A rule-based end-pointer isolates spoken utterances contained within an audio stream from background noise and non-speech transients. The rule-based end-pointer includes a plurality of rules to determine the beginning and/or end of a spoken utterance based on various speech characteristics. The rules may analyze an audio stream or a portion of an audio stream based upon an event, a combination of events, the duration of an event, or a duration relative to an event. The rules may be manually or dynamically customized depending upon factors that may include characteristics of the audio stream itself, an expected response contained within the audio stream, or environmental conditions.