Abstract: An apparatus and method for performing asynchronous speech recognition using multiple microphones are disclosed. The apparatus includes a microphone selection unit, a signal-to-noise ratio measurement unit, a speech recognition and verification unit, and a final recognition result output unit. The microphone selection unit selects two or more microphones responsive to a user's voice from among a plurality of microphones distributed around the user. The signal-to-noise ratio measurement unit measures the signal to noise ratios of inputs of the selected two or more microphones. The speech recognition and verification unit performs speech recognition using the input of the microphone having a highest signal to noise ratio, and verifies the speech recognition using the inputs of the remaining microphones. The final recognition result output unit outputs the final recognition results of the user's voice based on the results of the speech recognition and verification unit.

Abstract: Provided is a mobile communication terminal including: a camera module which captures an image of a set area; a microphone module which, when a sound including a voice of a user is input, extracts a sound level corresponding to the sound and a sound generating position; and a control module which estimates a position of a lip of the user from the image, extracts a voice level from the sound level corresponding to the position of the lip of the user and a voice generating position from the sound generating position, and recognizes the voice of the user based on at least one of the voice level and the voice generating position.

Abstract: Disclosed herein is a voice-based CAPTCHA method and apparatus which can perform a CAPTCHA procedure using the voice of a human being. In the voice-based CAPTCHA) method, a plurality of uttered sounds of a user are collected. A start point and an end point of a voice from each of the collected uttered sounds are detected and then speech sections are detected. Uttered sounds of the respective detected speech sections are compared with reference uttered sounds, and then it is determined whether the uttered sounds are correctly uttered sounds. It is determined whether the uttered sounds have been made by an identical speaker if it is determined that the uttered sounds are correctly uttered sounds.

Abstract: An evaporator for a looped heat pipe (LHP) system, in which a working fluid circulates to cool heat generating electronic components that generate heat during operation, the evaporator including: a body comprising an inlet through which the working fluid enters and an outlet through which the working fluid is discharged; a sintered wick that is included in the body, wherein the sintered wick is formed by sintering a metal powder, and a plurality of pores are formed in the sintered wick; and an additional layer that is formed on a vaporization surface of the sintered wick where evaporation of the working fluid occurs, wherein a plurality of through holes are formed in the additional layer such that the working fluid changed into a vapor state passes through the additional layer after passing the sintered wick.

Abstract: An evaporator for a looped heat pipe (LHP) system, in which a working fluid circulates to cool a heat generating electronic component that generates heat during operation, the evaporator including: a body including an inlet through which the working fluid enters and an outlet through which the working fluid is discharged; a sintered wick that is included in the body, wherein the sintered wick is formed by sintering a copper powder, and a plurality of pores are formed in the sintered wick; and an additional layer that is formed on a surface of the sintered wick, wherein the additional layer is formed by sintering copper particles having a size smaller than that of the copper powder forming the sintered wick, and the working fluid moved from the sintered wick is changed in a vapor state to be discharged.

Abstract: Disclosed is apparatus and system for user interface. The apparatus for user interface comprises a body unit including a groove which is corresponding to a structure of an oral cavity and operable to be mounted on upper part of the oral cavity; a user input unit receiving a signal from the user's tongue in a part of the body unit; a communication unit transmitting the signal received from the user input unit; and a charging unit supplying an electrical energy generated from vibration or pressure caused by movement of the user's tongue.

Abstract: Method of the present invention may include receiving speech feature vector converted from speech signal, performing first search by applying first language model to the received speech feature vector, and outputting word lattice and first acoustic score of the word lattice as continuous speech recognition result, outputting second acoustic score as phoneme recognition result by applying an acoustic model to the speech feature vector, comparing the first acoustic score of the continuous speech recognition result with the second acoustic score of the phoneme recognition result, outputting first language model weight when the first coustic score of the continuous speech recognition result is better than the second acoustic score of the phoneme recognition result and performing a second search by applying a second language model weight, which is the same as the output first language model, to the word lattice.

Abstract: Disclosed are a distributed environment rescoring method and apparatus. A distributed environment rescoring method in accordance with the present invention includes generating a word lattice by performing voice recognition on received voice, converting the word lattice into a word confusion network formed from the temporal connection of confusion sets clustered based on temporal redundancy and phoneme similarities, generating a list of subword confusion networks based on the entropy values of the respective confusion sets included in the word confusion network, and generating a modified word confusion network by modifying a list of the subword confusion networks through distributed environment rescoring.

Abstract: Disclosed are a personalized advertisement device based on speech recognition SMS services and a personalized advertisement exposure method based on speech recognition SMS services. The present invention provides a personalized advertisement device based on speech recognition SMS services and a personalized advertisement exposure method based on speech recognition SMS services capable of maximizing an effect of advertisement by grasping user's intention, an emotion state, and positional information from speech data uttered by a user during a process of providing speech recognition SMS services, configuring advertisements based thereon, and exposing the configured advertisements to a user.

Abstract: An evaporator for a looped heat pipe (LHP) system, in which a working fluid circulates to cool heat generating electronic components that generate heat during operation, the evaporator including: a body comprising an inlet through which the working fluid enters and an outlet through which the working fluid is discharged; a sintered wick that is included in the body, wherein the sintered wick is formed by sintering a metal powder, and a plurality of pores are formed in the sintered wick; and an additional layer that is formed on a vaporization surface of the sintered wick where evaporation of the working fluid occurs, wherein a plurality of through holes are formed in the additional layer such that the working fluid changed into a vapor state passes through the additional layer after passing the sintered wick.

Abstract: A method for separating a sound source from a mixed signal, includes Transforming a mixed signal to channel signals in frequency domain; and grouping several frequency bands for each channel signal to form frequency clusters. Further, the method for separating the sound source from the mixed signal includes separating the frequency clusters by applying a blind source separation to signals in frequency domain for each frequency cluster; and integrating the spectrums of the separated signal to restore the sound source in a time domain wherein each of the separated signals expresses one sound source.

Abstract: A microphone-array-based speech recognition system using a blind source separation (BBS) and a target speech extraction method in the system are provided. The speech recognition system performs an independent component analysis (ICA) to separate mixed signals input through a plurality of microphone into sound-source signals, extracts one target speech spoken for speech recognition from the separated sound-source signals by using a Gaussian mixture model (GMM) or a hidden Markov Model (HMM), and automatically recognizes a desired speech from the extracted target speech. Accordingly, it is possible to obtain a high speech recognition rate even in a noise environment.

Abstract: Provided are an apparatus and method for generating a noise adaptive acoustic model including a noise adaptive discriminative adaptation method. The method includes: generating a baseline model parameter from large-capacity speech training data including various noise environments; and receiving the generated baseline model parameter and applying a discriminative adaptation method to the generated results to generate an migrated acoustic model parameter suitable for an actually applied environment.

Abstract: A dialogue system include learning initiation unit which receives conversation education domain and target completion condition inconversation education domain and receives user's utterance, voice recognition unit which converts user's utterance into utterance text based on utterance information, language understanding unit which determines user's dialogue act based on converted utterance text and generates logical expression using slot expression corresponding to determined dialogue act and slot expression defined in conversation education domain, dialogue/progress management unit which determines utterance vertex with logical expression similar to that of utterance patterns of plurality of utterance vertices connected to system's final utterance vertex in dynamic dialogue graph and determines utterance vertices connected to determined utterance vertex as next utterance, system dialogue generation unit which retrieves utterance patterns connected to utterance vertex corresponding to determined next utterance

Abstract: Method of the present invention may include receiving speech feature vector converted from speech signal, performing first search by applying first language model to the received speech feature vector, and outputting word lattice and first acoustic score of the word lattice as continuous speech recognition result, outputting second acoustic score as phoneme recognition result by applying an acoustic model to the speech feature vector, comparing the first acoustic score of the continuous speech recognition result with the second acoustic score of the phoneme recognition result, outputting first language model weight when the first coustic score of the continuous speech recognition result is better than the second acoustic score of the phoneme recognition result and performing a second search by applying a second language model weight, which is the same as the output first language model, to the word lattice.

Abstract: Disclosed herein are an apparatus and method for preprocessing speech signals to perform speech recognition. The apparatus includes a voiced sound interval detection unit, a preprocessing method determination unit, and a clipping signal processing unit. The voiced sound interval detection unit detects a voiced sound interval including a voiced sound signal in a voice interval. The preprocessing method determination unit detects a clipping signal present in the voiced sound interval. The clipping signal processing unit extracts signal samples adjacent to the clipping signal, and performs interpolation on the clipping signal using the adjacent signal samples.

Abstract: A method for separating a sound source from a mixed signal, includes Transforming a mixed signal to channel signals in frequency domain; and grouping several frequency bands for each channel signal to form frequency clusters. Further, the method for separating the sound source from the mixed signal includes separating the frequency clusters by applying a blind source separation to signals in frequency domain for each frequency cluster; and integrating the spectrums of the separated signal to restore the sound source in a time domain wherein each of the separated signals expresses one sound source.

Abstract: A metadata search apparatus using speech recognition includes a metadata processor for processing contents metadata to obtain allomorph of target vocabulary required for speech recognition and search; a metadata storage unit for storing the contents metadata; a speech recognizer for performing speech recognition on speech data uttered by a user by searching the allomorph of the target vocabulary; a query language processor for extracting a keyword from the vocabulary speech-recognized by the speech recognizer; and a search processor for searching the metadata storage unit to extract the contents metadata corresponding to the keyword. An IPTV receiving apparatus employs the metadata search apparatus to provide IPTV services through the functions of speech recognition.

Abstract: A microphone-array-based speech recognition system using a blind source separation (BBS) and a target speech extraction method in the system are provided. The speech recognition system performs an independent component analysis (ICA) to separate mixed signals input through a plurality of microphone into sound-source signals, extracts one target speech spoken for speech recognition from the separated sound-source signals by using a Gaussian mixture model (GMM) or a hidden Markov Model (HMM), and automatically recognizes a desired speech from the extracted target speech. Accordingly, it is possible to obtain a high speech recognition rate even in a noise environment.

Abstract: Provided are an apparatus and method for recognizing speech, in which reliability with respect to phoneme-recognized phoneme sequences is calculated and performance of speech recognition is enhanced using the calculated results. The method of recognizing speech includes the steps of: determining a boundary between phonemes included in character sequences that are phonetically input to detect each phoneme interval; calculating reliability according to a probability that a phoneme indicated by the detected phoneme interval corresponds to a phoneme included in a predefined phoneme model; calculating a phoneme alignment cost with respect to the character sequences based on the calculated reliability and a pre-trained and stored phoneme recognition probability distribution; and performing phoneme alignment based on the calculated phoneme alignment cost to perform speech recognition on the input character sequences.