Abstract: Mechanisms are provided for latent ambiguity handling in natural language processing. The method may include: providing an annotated semantic graph based on a knowledge base in which nodes representing semantic concepts are linked by semantic relationships, wherein one or more nodes are annotated to indicate a latent ambiguity of the semantic concept; processing a natural language input by activation of the annotated semantic graph; during processing when encountering an annotated node indicating a latent ambiguity of the semantic concept, applying a pre-defined latent ambiguity process to the node; and replacing any annotated nodes indicating a latent ambiguity of the concept remaining in an output of the processing of the natural language input with an alternative node from the semantic graph.

Abstract: According to one embodiment of the present invention, a system for converting a display from a source spoken language to a target spoken language includes at least one processor. The at least one processor may be configured to determine the source spoken language of content within a selected area of the display. The at least one processor may be further configured to translate the content from the source spoken language to a selected target spoken language. In addition, the at least one processor may be configured to evaluate the translated content and remap the translated content to the selected area based on the evaluation. Finally, the at least one processor may be configured to present the translated content within the selected area on the display.

Abstract: In accordance with embodiments of the present disclosure, an integrated circuit for implementing at least a portion of an audio device may include an audio output configured to reproduce audio information by generating an audio output signal for communication to at least one transducer of the audio device, a microphone input configured to receive an input signal indicative of ambient sound external to the audio device and a processor configured to implement an impulsive noise detector. The impulsive noise detector may include a sudden onset detector for predicting an occurrence of a signal burst event of the input signal and an impulsive detector for determining whether the signal burst event comprises a speech event or a noise event.

Abstract: A friend addition method, device and medium are provided. The method includes: when a voiceprint-based friend addition request is received, target voice information is acquired; user information of a user to be added is determined according to the target voice information; and the user to be added is added as a friend according to the user information.

Abstract: Systems and methods are disclosed for customizing a neural network for a custom dataset, when the neural network has been trained on data from a general dataset. The neural network may comprise an output layer including one or more nodes corresponding to candidate outputs. The values of the nodes in the output layer may correspond to a probability that the candidate output is the correct output for an input. The values of the nodes in the output layer may be adjusted for higher performance when the neural network is used to process data from a custom dataset.

Abstract: A voice starting end detection apparatus includes a first detector that detects a starting end of a voice segment from input signals that are input in a time series, a first transmitting unit that transmits, when the starting end is detected, input signals subsequent to the starting end, and a first receiving unit that receives a terminal end detection signal indicating that a terminal end of the voice segment has been detected. The voice terminal end detection apparatus includes a second receiving unit that receives input signals subsequent to the starting end, a second detector that detects the terminal end from the received input signals, a second transmitting unit that transmits, when the terminal end is detected, the terminal end detection signal. The first transmitting unit stops transmitting the input signals when the first receiving unit receives the terminal end detection signal.

Abstract: An envelope sequence is provided that can improve approximation accuracy near peaks caused by the pitch period of an audio signal. A periodic-combined-envelope-sequence generation device according to the present invention takes, as an input audio signal, a time-domain audio digital signal in each frame, which is a predetermined time segment, and generates a periodic combined envelope sequence as an envelope sequence. The periodic-combined-envelope-sequence generation device according to the present invention comprises at least a spectral-envelope-sequence calculating part and a periodic-combined-envelope generating part. The spectral-envelope-sequence calculating part calculates a spectral envelope sequence of the input audio signal on the basis of time-domain linear prediction of the input audio signal.

Abstract: Approaches provide for using voice information to influence the importance of search result categories for a search query. For example, various embodiments may provide search results for a search query based on a most relevant search result category to the search query. Voice information associated with a subsequent user interaction may be analyzed to identify whether the search result category is correct or if search results from a different category should be provided. Additionally, the voice information may be used to update the relevance score of the search result category to the search query to improve the category matching of future queries.

Abstract: A speech recognition system prompts a user to provide a first utterance, which is recorded. Speech recognition is performed on the first user utterance to yield a recognition result. The user is prompted to provide a second user utterance, which is recorded, processed and compared to the first utterance to detect a plurality of acoustic differences for each acoustic parameter. The acoustic model used by the speech recognition engine is modified as a function of the acoustic difference.

Abstract: A method for selecting text created in a mobile terminal by word and correcting it or changing it to another word, and a mobile terminal implementing the same are disclosed. The mobile terminal includes: a display unit to display one or more words of text, and to display tags for each of the one or more words; an input unit to select at least one of the tagged one or more words as selected one word; and a controller to display candidate words having a similar pronunciation to that of the word selected via the input unit, select one of the candidate words as selected one candidate word, and change the selected one word from the text to the selected one candidate word.

Abstract: A support device, program and support method for supporting generation of text from speech data. The support device includes a confirmed rate calculator, a candidate obtaining unit and a selector. The confirmed rate calculator calculates a confirmed utterance rate which is an utterance rate of a confirmed part having already-confirmed text in the speech data. The candidate obtaining unit obtains multiple candidate character strings resulting from a speech recognition of an unconfirmed part having unconfirmed text in the speech data. The selector preferentially selects, from among the plurality of candidate character strings, a candidate character string whose utterance time consumed in uttering the candidate character string at the confirmed utterance rate is closest to an utterance time of the unconfirmed part of the speech data.

Abstract: The example embodiments relate to a method for assisting in the translational testing of computer software. A translated panel, which may include a translatable portion and a continuously dynamic portion, may be provided. An overlay that includes only the continuously dynamic portions of the panel may be produced. The overlay may be applied to first and second versions of the panel to produce first and second iterations of the panel, respectively, wherein each iteration does not include the continuously dynamic portion of the panel. The translatable portions of the first and second iterations may be compared to determine whether the translatable portions are different. If the translatable portions are different, then the panel may be retested for translational accuracy; if the translatable portions are not different, then the program may indicate that the second iteration is not different from the first iteration.

Abstract: An audio encoding method and apparatus, and an audio decoding method and apparatus, for processing a death sinusoid and a general continuation sinusoid. Using the unique characteristic of a death sinusoid, in that the death sinusoid has a tendency such that an amplitude component of the death sinusoid is less than that of a previous sinusoid, a method of adding an encoding syntax by distinguishing a general continuation sinusoid from a death sinusoid is provided. That is, when difference coding of the amplitude component of a death sinusoid is performed, the number of bits used when a negative number is coded is less than the number of bits used when a positive number is coded, in a Huffman table. By using this method, a bit rate in an entire coding decreases.

Abstract: A method and apparatus for estimating the pitch period of a signal. The method includes identifying a first candidate pitch period by performing a search only over a first range of potential pitch periods. The method further includes determining a second candidate pitch period by dividing the first candidate pitch period by an integer, wherein the second candidate pitch period is outside the first range of potential pitch periods. The method further includes selecting as the estimate of the pitch period of the signal the smaller of the candidate pitch periods that is such that portions of the signal separated by that candidate pitch period are well correlated.

Abstract: The invention presents a method to improve the recovering from packet loss, frame erasure or jitter concealment during signal communication, especially for VoIP (Voice Over Internet Protocol) applications. A variable delay concept (instead of constant delay) is introduced to guarantee the continuity and periodicity of signal after recovering lost frames, adding frames or removing frames. During the recovering of lost frames or the adding of extra frames, the copy of previous signal from history buffer into missing frame(s) is based on the frame length, onset, and offset information.

Abstract: To search out a new word that should be newly registered in a dictionary contained in a segmentation device for segmenting a text into words. This system inputs a training text into the segmentation device to cause the segmentation device to segment the training text into words, and thereby generates a plurality of segmentation candidates in association with certainty factors of the results of the segmentation, the segmentation candidates respectively containing mutually different combinations of words as results of the segmentation of the training text. Then, this system computes a likelihood that the each word is a new word by summing up some of the certainty factors that are respectively associated with some of the plurality of segmentation candidates that contain the each word.

Abstract: A system and method for automatically adjusting floor controls for a conversation is provided. Audio streams are received, which each originate from an audio source. Floor controls for a current configuration including at least a portion of the audio streams are maintained. Conversational characteristics shared by two or more of the audio sources are determined. Possible configurations for the audio streams are identified based on the conversational characteristics. An analysis of the current configuration and the possible configurations is performed. A change threshold is applied to the analysis. When the analysis satisfies the change threshold, the floor controls are automatically adjusted. The audio streams are mixed into one or more outputs based on the adjusted floor controls.

Abstract: Translating information, such as an object type and content, in a first data object from an initial language into a target language includes searching a database to locate a second data object having the object type, attempting to obtain, from the second data object, the object type and the content in the target language, and storing, in the first data object, the object type and content in the target language.

Abstract: A solution is proposed for the data-handling system-assisted comprehension of texts (particularly, when they are written in a foreign language). For this purpose, a difficulty index is associated with each known word (for example, according to its frequency of use in standard texts). The words of a current document opened using the data-handling system are sorted according to their difficulty indexes. The words having the difficulty indexes higher than a threshold value (for example, based on the skill of a reader) are selected as difficult. A translation of each difficult word is then extracted from a dictionary; this translation is displayed close to each occurrence of the corresponding difficult word.

Abstract: A recording and playback system is provided. The system includes an audio capturing device configured to receive an analog input and an encoder coupled to the audio capturing device configured to generate a digital signal based on the analog input. The system further includes a recognition engine coupled to the audio capturing device and configured to generate text data based on the analog input, wherein the encoder and the recognition engine simultaneously generate the digital signal and the text data such that the digital signal and the text data can be provided in a synchronized manner.