Abstract: An electronic device includes a processor configured to execute: accepting a character input; causing a display module to display a character or a character string, which has been input by the character input; causing the display module to display a plurality of words which are conversion candidates corresponding to the input character or character string; causing the display module to display, responding to designation by a user of a word of the plurality of words which are conversion candidates, at least one synonym corresponding to the designated word; and causing the display module to display, responding to selection by the user of a synonym of the displayed at least one synonym, the selected synonym in place of the input character or character string.
Abstract: An audio frame timing correction method and a wireless device are provided. A controller generates a reference clock for audio coding/decoding such that the reference clock runs fast and moved forward within an audio data sampling interval with the remaining time becoming a margin of the interval. An audio codec decodes demodulated data based on the reference clock, and codes an audio signal based on the reference clock. A demodulator detects wireless frame deviation and determines an adjustment timing whereat the wireless frame symbol timing and the audio frame timing are corrected based on the deviation and the margin. Upon the adjustment timing, the controller synchronizes audio sampling timing with the wireless frame symbol timing.
Abstract: A method, a computer readable medium, and a system for tagging Natural language application. In the method, utterances are analyzed using one or more rules, and a tag is assigned to the analyzed utterances based on the one or more rules. The analysis of the utterances may include determining a frequency of the utterances, and grouping the utterances by their determined frequency and their assigned tag. The frequency may represent a number of occurrences in natural language where the utterances share semantic meanings, for example. Further, the one or more rules may be prioritized, and each of the utterances may be analyzed using the prioritized rules. In this manner, meaning may be assigned to utterances such that groups of utterances may be tagged simultaneously.
Type:
Grant
Filed:
April 22, 2008
Date of Patent:
January 26, 2016
Assignee:
WEST CORPORATION
Inventors:
Kiyomi Murata, Steven John Schanbacher, Aaron Scott Fisher
Abstract: A system and method for parallel speech recognition processing of multiple audio signals produced by multiple microphones in a handheld portable electronic device. In one embodiment, a primary processor transitions to a power-saving mode while an auxiliary processor remains active. The auxiliary processor then monitors the speech of a user of the device to detect a wake-up command by speech recognition processing the audio signals in parallel. When the auxiliary processor detects the command it then signals the primary processor to transition to active mode. The auxiliary processor may also identify to the primary processor which microphone resulted in the command being recognized with the highest confidence. Other embodiments are also described.
Abstract: A voice quality conversion system includes: an analysis unit which analyzes sounds of plural vowels of different types to generate first vocal tract shape information for each type of the vowels; a combination unit which combines, for each type of the vowels, the first vocal tract shape information on that type of vowel and the first vocal tract shape information on a different type of vowel to generate second vocal tract shape information on that type of vowel; and a synthesis unit which (i) combines vocal tract shape information on a vowel included in input speech and the second vocal tract shape information on the same type of vowel to convert vocal tract shape information on the input speech, and (ii) generates a synthetic sound using the converted vocal tract shape information and voicing source information on the input speech to convert the voice quality of the input speech.
Abstract: A smart home interaction system is presented. It is built on a multi-modal, multithreaded conversational dialog engine. The system provides a natural language user interface for the control of household devices, appliances or household functionality. The smart home automation agent can receive input from users through sensing devices such as a smart phone, a tablet computer or a laptop computer. Users interact with the system from within the household or from remote locations. The smart home system can receive input from sensors or any other machines with which it is interfaced. The system employs interaction guide rules for processing reaction to both user and sensor input and driving the conversational interactions that result from such input. The system adaptively learns based on both user and sensor input and can learn the preferences and practices of its users.
Type:
Grant
Filed:
October 8, 2013
Date of Patent:
January 5, 2016
Assignee:
Nant Holdings IP, LLC
Inventors:
Farzad Ehsani, Silke Maren Witt-Ehsani, Walter Rolandi
Abstract: Systems and methods for efficient online domain adaptation are provided herein. Methods may include receiving a post-edited machine translated sentence pair, updating a machine translation model by adjusting translation weights for a translation memory and a language model while generating test machine translations of the machine translated sentence pair until one of the test machine translations approximately matches the post-edits for the machine translated sentence pair, and retranslating the remaining machine translation sentence pairs that have yet to be post-edited using the updated machine translation model.
Abstract: According to one embodiment, a machine translation apparatus includes a translation unit, an acquisition unit, a first calculation unit, a reverse translation unit, a second calculation unit and a selection unit. The acquisition unit acquires, at least one second forward-translated word different from the first forward-translated word, to obtain candidate words. The reverse translation unit obtains at least one reverse-translated word for each of the candidate words by reverse-translating each candidate word into the first language. The selection unit selects a corrected forward-translated word to be replaced with the first forward-translated word from among the candidate words based on the semantic similarity and fluency.
Abstract: A signal processing apparatus comprises a signal path for a signal, the signal path comprising a signal processing stage. An auxiliary stage is coupled to an input of the signal processing stage for, in response to a signal in the signal path at the input of the signal processing stage, generating a control signal indicative of the time of a crossing of a first threshold by the signal in the signal path at an output of the signal processing stage by detecting a crossing by the signal of a second threshold established by the auxiliary stage. The second threshold is substantially equal to the first threshold.
Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for approximating responses to a user speech query in voice-enabled search based on metadata that include demographic features of the speaker. A system practicing the method recognizes received speech from a speaker to generate recognized speech, identifies metadata about the speaker from the received speech, and feeds the recognized speech and the metadata to a question-answering engine. Identifying the metadata about the speaker is based on voice characteristics of the received speech. The demographic features can include age, gender, socio-economic group, nationality, and/or region. The metadata identified about the speaker from the received speech can be combined with or override self-reported speaker demographic information.
Type:
Grant
Filed:
March 19, 2013
Date of Patent:
November 17, 2015
Assignee:
Interactions LLC
Inventors:
Michael Johnston, Srinivas Bangalore, Junlan Feng, Taniya Mishra
Abstract: A questionnaire is presented to a user in a more efficient manner in which the user is more likely to participate. The questionnaire is sent electronically to the user's vehicle and presented audibly to the user. The user responds audibly to the questions in the questionnaire. The user's responses are converted to text and sent back to the provider server for tallying.
Abstract: Included are embodiments for multi-pass analytics. At least one embodiment of a method includes receiving data associated with a communication, performing first tier analytics on the received data, and performing second tier analytics on the received data, where the second tier analytics determines different characteristics of the received data than the first tier analytics.
Abstract: According to one embodiment, a machine translation apparatus includes a speech recognition unit, a translation unit, a detection unit and an addition unit. The speech recognition unit performs speech recognition of speech. The translation unit translates the plurality of source language strings into target language strings in a chronological order. The detection unit detects ambiguity in interpretation of the speech corresponding to a first target language string of the target language strings. The addition unit adds, an additional phrase being one of words and phrases to interpret uniquely a modification relationship, to the first target language string if ambiguity is detected.
Abstract: For respective sampling data of waveform data of sounds to be coded, a prediction residual value is calculated as sampling residual data, and an effective bit length is calculated from this residual waveform data. Then, for the effective bit length data, a maximum effective bit length among processing targets is generated as common effective actual data, and coded data in which this common effective actual data and information indicating the common effective bit length are arranged in a predetermined configuration format are generated. The information included in the coded data is analyzed and each of the plurality of the common effective bit information is extracted. Then, waveform data of the sounds are decoded by performing inverse linear prediction processing from an analysis result on the residual waveform data decompressed by performing bit extension which adds a portion other than the common effective bit length.
Abstract: A voice processing device includes a voice pitch converting unit that performs a voice pitch converting process with respect to an input voice signal and converts voice pitch of the input voice signal, an error detecting unit that detects an error between the number of samples of an output voice signal, which is expected, and the number of samples of the output voice signal, which is actually output, and a time length control unit that controls adjustment of the time length in such a manner that the time length of the output voice signal is corrected by the amount of the error.
Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for training recognition canonical representations corresponding to named-entity phrases in a second natural language based on translating a set of allowable expressions with canonical representations from a first natural language, which may be generated by expanding a context-free grammar for the allowable expressions for the first natural language.
Abstract: Statements of a computer program expressed using a first source natural language are made meaningful to a programmer familiar with a second target natural language. The first source natural language of the computer program is determined from the programmer, or through analysis, and the second target natural language desired by the programmer is selected. Textual constructs may be parsed, with reference to stored coding conventions to determine meaningful lexical tokens. Such tokens are translated with a translation engine, and displayed to the programmer, desirably using a graphical user interface feature of an integrated development, environment (IDE) for computer programming in a particular programming language.
Type:
Grant
Filed:
October 19, 2012
Date of Patent:
September 22, 2015
Assignee:
International Business Machines Corporation
Abstract: The present technology concerns improvements to smart phones and related sensor-equipped systems. Some embodiments relate to smart phone-assisted commuting, e.g., by bicycle. Some involve novel human-computer interactions, e.g., using tactile grammars—some of which may be customized by users. Others involve spoken clues, e.g., by which a user can assist a smart phone in identifying what portion of imagery captured by a smart phone camera should be processed, or identifying what type of image processing should be conducted. Some arrangements include the degradation of captured content information in accordance with privacy rules, which may be location-dependent, or based on the unusualness of the captured content, or responsive to later consultation of the stored content information by the user. A great variety of other features and arrangements are also detailed.
Type:
Grant
Filed:
December 30, 2010
Date of Patent:
September 22, 2015
Assignee:
Digimarc Corporation
Inventors:
Bruce L. Davis, Tony F. Rodriguez, Geoffrey B. Rhoads, William Y. Conwell
Abstract: An audio signal decoder configured to provide a decoded audio signal representation on the basis of an encoded audio signal representation including a sampling frequency information, an encoded time warp information and an encoded spectrum representation includes a time warp calculator and a warp decoder. The time warp calculator is configured to adapt a mapping rule for mapping codewords of the encoded time warp information onto decoded time warp values describing the decoded time warp information in dependence on the sampling frequency information. The warp decoder is configured to provide the decoded audio signal representation on the basis of the encoded spectrum representation and in dependence on the decoded time warp information.
Type:
Grant
Filed:
September 6, 2012
Date of Patent:
September 8, 2015
Assignees:
Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e. V., Dolby International AB
Inventors:
Stefan Bayer, Tom Baeckstroem, Ralf Geiger, Bernd Edler, Sascha Disch, Lars Villemoes
Abstract: An improved system and method are disclosed for peer-to-peer communications. In one example, the method enables an endpoint to send and/or receive audio speech translations to facilitate communications between users who speak different languages.