Abstract: The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilizing high frequency reconstruction (HFR). It utilizes a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR unit.
Type:
Grant
Filed:
November 19, 2008
Date of Patent:
February 7, 2012
Assignee:
Coding Technologies AB
Inventors:
Kristofer Kjörling, Per Ekstrand, Holger Hörich
Abstract: Configurations herein provide a language processing mechanism operable to define a machine vocabulary and identify a machine language version of the words that preserves context and identifies the proper definition of the words by identifying and preserving context of a particular set of words, such as a sentence or paragraph. The machine vocabulary includes a definition section for each definition of a word. Each definition section includes a set of one or more definition elements. The definition elements include a predetermined format of definition fields, and each has a corresponding mask indicative of significant definition fields. The set of definition elements corresponding to a particular definition describe the usage of the word in a context matching that particular definition. Each definition element captures a characteristic of the definition according to fuzzy logic such that the definition elements collectively capture the context.
Type:
Grant
Filed:
September 1, 2009
Date of Patent:
January 31, 2012
Assignee:
Artificial Cognition Inc.
Inventors:
George H. Harvey, Donald R. Greenbaum, Charles H. Collins, Charles D. Harvey
Abstract: Systems and methods for improving the interaction between a user and a small electronic device such as a Bluetooth headset are described. The use of a voice user interface in electronic devices may be used. In one embodiment, recognition processing limitations of some devices are overcome by employing speech synthesizers and recognizers in series where one electronic device responds to simple audio commands and sends audio requests to a remote device with more significant recognition analysis capability. Embodiments of the present invention may include systems and methods for utilizing speech recognizers and synthesizers in series to provide simple, reliable, and hands-free interfaces with users.
Abstract: A speech enhancement system that improves the intelligibility and the perceived quality of processed speech includes a frequency transformer and a spectral compressor. The frequency transformer converts speech signals from the time domain to the frequency domain. The spectral compressor compresses a pre-selected portion of the high frequency band and maps the compressed high frequency band to a lower band limited frequency range.
Abstract: In one embodiment, a method of signal processing including includes encoding a low-frequency portion of a speech signal into at least an encoded narrowband excitation signal and a plurality of narrowband filter parameters; and generating a highband excitation signal based on a narrowband excitation signal. The encoded narrowband excitation signal includes a time warping, and the method includes applying a time shift to a high-frequency portion of the speech signal based on the information related to the time warping. The method also includes encoding the time-shifted high-frequency portion of the speech signal into at least one (A) a plurality of highband filter parameters and (B) a plurality of high band gain factors.
Type:
Grant
Filed:
April 3, 2006
Date of Patent:
December 13, 2011
Assignee:
QUALCOMM Incorporated
Inventors:
Koen Bernard Vos, Ananthapadmanabhan Aasanipalai Kandhadai
Abstract: Certain aspects and embodiments of the present invention are directed to systems and methods for monitoring and analyzing the language environment and the development of a key child. A key child's language environment and language development can be monitored without placing artificial limitations on the key child's activities or requiring a third party observer. The language environment can be analyzed to identify words, vocalizations, or other noises directed to or spoken by the key child, independent of content. The analysis can include the number of responses between the child and another, such as an adult and the number of words spoken by the child and/or another, independent of content of the speech. One or more metrics can be determined based on the analysis and provided to assist in improving the language environment and/or tracking language development of the key child.
Abstract: A translation device has a dictionary that stores a set of words and their corresponding meanings in plural languages; an input unit that inputs a document; a recognizing unit that recognizes text in the inputted document; an analyzing unit that devides the text recognized by the recognizing unit into words; a translating unit that translates each of the words obtained by the analyzing unit into a translated term by using the dictionary; and an output unit that outputs an output image containing the translated term for a key word.
Abstract: A system improves the perceptual quality of a speech signal by dampening undesired repetitive transient noises. The system includes a repetitive transient noise detector adapted to detect repetitive transient noise in a received signal. The received signal may include a harmonic and a noise spectrum. The system further includes a repetitive transient noise attenuator that substantially removes or dampens repetitive transient noises from the received signal. The method of dampening the repetitive transient noises includes modeling characteristics of repetitive transient noises; detecting characteristics in the received signal that correspond to the modeled characteristics of the repetitive transient noises; and substantially removing components of the repetitive transient noises from the received signal that correspond to some or all of the modeled characteristics of the repetitive transient noises.
Type:
Grant
Filed:
January 13, 2006
Date of Patent:
December 6, 2011
Assignee:
QNX Software Systems Co.
Inventors:
Phillip A. Hetherington, Shreyas A. Paranjpe
Abstract: A one-step correction mechanism for voice interaction is provided. Correction of a previous state is enabled simultaneously with recognition in a current or subsequent state. An application is decomposed into a set of tasks. Each task is associated with the collection of one piece of information. Each task may be in a different state. At any point during the interaction, while a task/state pair is active, the dialog manager may enable multiple other task/state pairs to be active in latent fashion. The application developer may then use those facilities or resources to the active task/state and the latent task/state pairs depending on contextual condition of the interaction state of the application.
Abstract: Methods and apparatus for characterizing media are described. In one example, a method of characterizing media includes capturing a block of audio; converting at least a portion of the block of audio into a frequency domain representation including a plurality of complex-valued frequency components; defining a band of complex-valued frequency components for consideration; determining a decision metric using the band of complex-valued frequency components; and determining a signature bit based on a value of the decision metric. Other examples are shown and described.
Type:
Grant
Filed:
February 20, 2008
Date of Patent:
November 15, 2011
Assignee:
The Nielsen Company (US), LLC
Inventors:
Alexander Topchy, Venugopal Srinivasan, Arun Ramaswamy
Abstract: An improved audio coding technique encodes audio having a low frequency transient signal, using a long block, but with a set of adapted masking thresholds. Upon identifying an audio window that contains a low frequency transient signal, masking thresholds for the long block may be calculated as usual. A set of masking thresholds calculated for the 8 short blocks corresponding to the long block are calculated. The masking thresholds for low frequency critical bands are adapted based on the thresholds calculated for the short blocks, and the resulting adapted masking thresholds are used to encode the long block of audio data. The result is encoded audio with rich harmonic content and negligible coder noise resulting from the low frequency transient signal.
Abstract: An associated-information storage unit stores a name of associated information and a display position in association with each other. An example storage unit stores a semantic class, an example in a source language, and an example in a target language in association with each other. A dictionary storage unit stores the name of associated information and the semantic class in association with each other. An acquiring unit acquires the name of the associated information corresponding to the display position of the selected associated information from the associated-information storage unit, and acquires a semantic class corresponding to the acquired name of the associated information from the dictionary storage unit. A translation unit acquires an example in the target language corresponding to the acquired semantic class and a speech recognition result from the example storage unit, thereby translating the recognition result.
Abstract: Voice recognition methods and systems are disclosed. A voice signal is obtained for an utterance of a speaker. The speaker is categorized as a male, female, or child and the categorization is used as a basis for dynamically adjusting a maximum frequency fmax and a minimum frequency fmin of a filter bank used for processing the input utterance to produce an output. Corresponding gender or age specific acoustic models are used to perform voice recognition based on the filter bank output.
Abstract: A system, method, apparatus, signal-bearing medium, and means for transmitting speech activity in a distributed voice recognition (VR) system. The distributed voice recognition system includes a local VR engine in a subscriber unit (102) and a server VR engine on a server (160). The local VR engine comprises a voice activity detection (VAD) module (106) that detects voice activity within a speech signal, and comprises an advanced feature extraction (AFE) module (104) that extracts features from a speech signal. The detected voice activity information is transmitted over a first wireless communication channel to the server (160). The feature extraction information is transmitted over a second wireless communication channel, separate from the first wireless communication channel, to the server (160). The server (160) processes the received information to determine a linguistic estimate of the electrical speech signal, and transmits the linguistic estimate to the subscriber unit (102).
Abstract: An administration method and system. The method includes receiving by a computing system, a telephone call from an administrator. The computing system presents an audible menu associated with a plurality of computers to the administrator. The computing system receives from the administrator, an audible selection for a computer from the audible menu. The computing system receives from the administrator, an audible verbal command for performing a maintenance operation on the computer. The computing system executes the maintenance operation on the computer. The computing system receives from the computer, confirmation data indicating that the maintenance operation has been completed. The computing system converts the confirmation data into an audible verbal message. The computing system transmits the second audible verbal message to the administrator.
Type:
Grant
Filed:
May 1, 2008
Date of Patent:
November 1, 2011
Assignee:
International Business Machines Corporation
Abstract: A language processing unit identifies a word by performing language analysis on a text supplied from a text holding unit. A synthesis selection unit selects speech synthesis processing performed by a rule-based synthesis unit or speech synthesis processing performed by a pre-recorded-speech-based synthesis unit for a word of interest extracted from the language analysis result. The selected rule-based synthesis unit or pre-recorded-speech-based synthesis unit executes speech synthesis processing for the word of interest.
Abstract: A system for use in speech recognition includes an acoustic module accessing a plurality of distinct-language acoustic models, each based upon a different language; a lexicon module accessing at least one lexicon model; and a speech recognition output module. The speech recognition output module generates a first speech recognition output using a first model combination that combines one of the plurality of distinct-language acoustic models with the at least one lexicon model. In response to a threshold determination, the speech recognition output module generates a second speech recognition output using a second model combination that combines a different one of the plurality of distinct-language acoustic models with the at least one distinct-language lexicon model.
Abstract: A device including a display screen for displaying m-words of data, a text entry device for entering data, a processor receiving data from the text entry device and causing it to be displayed on the display screen. Upon activation the processor initializes a precursor to a predefined value. The device further includes a non-volatile memory storing a dictionary containing a plurality of entries, each entry including an index, a candidate word, and a score. The processor selects a list of n-number of candidate words from the dictionary whose index matches the precursor, and causes m-number of candidate words from the list of candidate words to be displayed on the display screen. The processor causes the display to prompt the user to select one of the displayed candidate words or enter a desired word using the text entry device.
Abstract: Methods of defining ontologies, word disambiguation methods, computer systems, and articles of manufacture are described according to some aspects. In one aspect, a word disambiguation method includes accessing textual content to be disambiguated, wherein the textual content comprises a plurality of words individually comprising a plurality of word senses, for an individual word of the textual content, identifying one of the word senses of the word as indicative of the meaning of the word in the textual content, for the individual word, selecting one of a plurality of event classes of a lexical database ontology using the identified word sense of the individual word, and for the individual word, associating the selected one of the event classes with the textual content to provide disambiguation of a meaning of the individual word in the textual content.
Type:
Grant
Filed:
November 4, 2005
Date of Patent:
October 11, 2011
Assignee:
Battelle Memorial Institute
Inventors:
Antonio P. Sanfilippo, Stephen C. Tratz, Michelle L. Gregory, Alan R. Chappell, Paul D. Whitney, Christian Posse, Robert L. Baddeley, Ryan E. Hohimer
Abstract: A coded signal conveys encoded audio information and metadata that may be used to control the loudness of the audio information during its playback. If the values for these metadata parameters are set incorrectly, annoying fluctuations in loudness during playback can result. The present invention overcomes this problem by detecting incorrect metadata parameter values in the signal and replacing the incorrect values with corrected values.