• Title/Summary/Keyword: 음성망

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An Improvement of Fast Handoff Protocol using Modified Local Registration in Mobile Computing Environment (이동 컴퓨팅 환경에서 수정된 지역 위치등록을 이용한 고속 핸드오프 프로토콜 개선)

  • Han, Seung-Jin;Choe, Seong-Yong;Lee, Jeong-Hyeon
    • The KIPS Transactions:PartC
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    • v.9C no.2
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    • pp.267-276
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    • 2002
  • By using wireless terminal, User that want to transmit multimedia traffic as well as simple text and voice have a tendency to increase. This paper proposes a fast handoff protocol that is suitable transmission for real-tim of multimedia traffic by using modified local registration. The proposed protocol solves the Triangle Routing Protocol that is one of existing opened issues without modifying protocol of CN, and we propose the method that MN is able to received a packet by real-time, even if MN is being handoff. We compare fast handoff protocol proposed in this paper with existing method n the registration cost and data packet transmission cost. As a result, we showed that fast handoff protocol proposed in this paper outperforms existing method.

LFMMI-based acoustic modeling by using external knowledge (External knowledge를 사용한 LFMMI 기반 음향 모델링)

  • Park, Hosung;Kang, Yoseb;Lim, Minkyu;Lee, Donghyun;Oh, Junseok;Kim, Ji-Hwan
    • The Journal of the Acoustical Society of Korea
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    • v.38 no.5
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    • pp.607-613
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    • 2019
  • This paper proposes LF-MMI (Lattice Free Maximum Mutual Information)-based acoustic modeling using external knowledge for speech recognition. Note that an external knowledge refers to text data other than training data used in acoustic model. LF-MMI, objective function for optimization of training DNN (Deep Neural Network), has high performances in discriminative training. In LF-MMI, a phoneme probability as prior probability is used for predicting posterior probability of the DNN-based acoustic model. We propose using external knowledges for training the prior probability model to improve acoustic model based on DNN. It is measured to relative improvement 14 % as compared with the conventional LF-MMI-based model.

Video Compression Standard Prediction using Attention-based Bidirectional LSTM (어텐션 알고리듬 기반 양방향성 LSTM을 이용한 동영상의 압축 표준 예측)

  • Kim, Sangmin;Park, Bumjun;Jeong, Jechang
    • Journal of Broadcast Engineering
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    • v.24 no.5
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    • pp.870-878
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    • 2019
  • In this paper, we propose an Attention-based BLSTM for predicting the video compression standard of a video. Recently, in NLP, many researches have been studied to predict the next word of sentences, classify and translate sentences by their semantics using the structure of RNN, and they were commercialized as chatbots, AI speakers and translator applications, etc. LSTM is designed to solve the gradient vanishing problem in RNN, and is used in NLP. The proposed algorithm makes video compression standard prediction possible by applying BLSTM and Attention algorithm which focuses on the most important word in a sentence to a bitstream of a video, not an sentence of a natural language.

Proposal of speaker change detection system considering speaker overlap (화자 겹침을 고려한 화자 전환 검출 시스템 제안)

  • Park, Jisu;Yun, Young-Sun;Cha, Shin;Park, Jeon Gue
    • The Journal of the Acoustical Society of Korea
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    • v.40 no.5
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    • pp.466-472
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    • 2021
  • Speaker Change Detection (SCD) refers to finding the moment when the main speaker changes from one person to the next in a speech conversation. In speaker change detection, difficulties arise due to overlapping speakers, inaccuracy in the information labeling, and data imbalance. To solve these problems, TIMIT corpus widely used in speech recognition have been concatenated artificially to obtain a sufficient amount of training data, and the detection of changing speaker has performed after identifying overlapping speakers. In this paper, we propose an speaker change detection system that considers the speaker overlapping. We evaluated and verified the performance using various approaches. As a result, a detection system similar to the X-Vector structure was proposed to remove the speaker overlapping region, while the Bi-LSTM method was selected to model the speaker change system. The experimental results show a relative performance improvement of 4.6 % and 13.8 % respectively, compared to the baseline system. Additionally, we determined that a robust speaker change detection system can be built by conducting related studies based on the experimental results, taking into consideration text and speaker information.

Design and Implementation of Vehicle Control Network Using WiFi Network System (WiFi 네트워크 시스템을 활용한 차량 관제용 네트워크의 설계 및 구현)

  • Yu, Hwan-Shin
    • Journal of the Korea Academia-Industrial cooperation Society
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    • v.20 no.3
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    • pp.632-637
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    • 2019
  • Recent researches on autonomous driving of vehicles are becoming very active, and it is a trend to assist safe driving and improve driver's convenience. Autonomous vehicles are required to combine artificial intelligence, image recognition capability, and Internet communication between objects. Because mobile telecommunication networks have limitations in their processing, they can be easily implemented and scale using an easily expandable Wi-Fi network. We propose a wireless design method to construct such a vehicle control network. We propose the arrangement of AP and the software configuration method to minimize loss of data transmission / reception of mobile terminal. Through the design of the proposed network system, the communication performance of the moving vehicle can be dramatically increased. We also verify the packet structure of GPS, video, voice, and data communication that can be used for the vehicle through experiments on the movement of various terminal devices. This wireless design technology can be extended to various general purpose wireless networks such as 2.4GHz, 5GHz and 10GHz Wi-Fi. It is also possible to link wireless intelligent road network with autonomous driving.

Analysis of deep learning-based deep clustering method (딥러닝 기반의 딥 클러스터링 방법에 대한 분석)

  • Hyun Kwon;Jun Lee
    • Convergence Security Journal
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    • v.23 no.4
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    • pp.61-70
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    • 2023
  • Clustering is an unsupervised learning method that involves grouping data based on features such as distance metrics, using data without known labels or ground truth values. This method has the advantage of being applicable to various types of data, including images, text, and audio, without the need for labeling. Traditional clustering techniques involve applying dimensionality reduction methods or extracting specific features to perform clustering. However, with the advancement of deep learning models, research on deep clustering techniques using techniques such as autoencoders and generative adversarial networks, which represent input data as latent vectors, has emerged. In this study, we propose a deep clustering technique based on deep learning. In this approach, we use an autoencoder to transform the input data into latent vectors, and then construct a vector space according to the cluster structure and perform k-means clustering. We conducted experiments using the MNIST and Fashion-MNIST datasets in the PyTorch machine learning library as the experimental environment. The model used is a convolutional neural network-based autoencoder model. The experimental results show an accuracy of 89.42% for MNIST and 56.64% for Fashion-MNIST when k is set to 10.

Region of Interest Extraction and Bilinear Interpolation Application for Preprocessing of Lipreading Systems (입 모양 인식 시스템 전처리를 위한 관심 영역 추출과 이중 선형 보간법 적용)

  • Jae Hyeok Han;Yong Ki Kim;Mi Hye Kim
    • The Transactions of the Korea Information Processing Society
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    • v.13 no.4
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    • pp.189-198
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    • 2024
  • Lipreading is one of the important parts of speech recognition, and several studies have been conducted to improve the performance of lipreading in lipreading systems for speech recognition. Recent studies have used method to modify the model architecture of lipreading system to improve recognition performance. Unlike previous research that improve recognition performance by modifying model architecture, we aim to improve recognition performance without any change in model architecture. In order to improve the recognition performance without modifying the model architecture, we refer to the cues used in human lipreading and set other regions such as chin and cheeks as regions of interest along with the lip region, which is the existing region of interest of lipreading systems, and compare the recognition rate of each region of interest to propose the highest performing region of interest In addition, assuming that the difference in normalization results caused by the difference in interpolation method during the process of normalizing the size of the region of interest affects the recognition performance, we interpolate the same region of interest using nearest neighbor interpolation, bilinear interpolation, and bicubic interpolation, and compare the recognition rate of each interpolation method to propose the best performing interpolation method. Each region of interest was detected by training an object detection neural network, and dynamic time warping templates were generated by normalizing each region of interest, extracting and combining features, and mapping the dimensionality reduction of the combined features into a low-dimensional space. The recognition rate was evaluated by comparing the distance between the generated dynamic time warping templates and the data mapped to the low-dimensional space. In the comparison of regions of interest, the result of the region of interest containing only the lip region showed an average recognition rate of 97.36%, which is 3.44% higher than the average recognition rate of 93.92% in the previous study, and in the comparison of interpolation methods, the bilinear interpolation method performed 97.36%, which is 14.65% higher than the nearest neighbor interpolation method and 5.55% higher than the bicubic interpolation method. The code used in this study can be found a https://github.com/haraisi2/Lipreading-Systems.

Korean Sentence Generation Using Phoneme-Level LSTM Language Model (한국어 음소 단위 LSTM 언어모델을 이용한 문장 생성)

  • Ahn, SungMahn;Chung, Yeojin;Lee, Jaejoon;Yang, Jiheon
    • Journal of Intelligence and Information Systems
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    • v.23 no.2
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    • pp.71-88
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    • 2017
  • Language models were originally developed for speech recognition and language processing. Using a set of example sentences, a language model predicts the next word or character based on sequential input data. N-gram models have been widely used but this model cannot model the correlation between the input units efficiently since it is a probabilistic model which are based on the frequency of each unit in the training set. Recently, as the deep learning algorithm has been developed, a recurrent neural network (RNN) model and a long short-term memory (LSTM) model have been widely used for the neural language model (Ahn, 2016; Kim et al., 2016; Lee et al., 2016). These models can reflect dependency between the objects that are entered sequentially into the model (Gers and Schmidhuber, 2001; Mikolov et al., 2010; Sundermeyer et al., 2012). In order to learning the neural language model, texts need to be decomposed into words or morphemes. Since, however, a training set of sentences includes a huge number of words or morphemes in general, the size of dictionary is very large and so it increases model complexity. In addition, word-level or morpheme-level models are able to generate vocabularies only which are contained in the training set. Furthermore, with highly morphological languages such as Turkish, Hungarian, Russian, Finnish or Korean, morpheme analyzers have more chance to cause errors in decomposition process (Lankinen et al., 2016). Therefore, this paper proposes a phoneme-level language model for Korean language based on LSTM models. A phoneme such as a vowel or a consonant is the smallest unit that comprises Korean texts. We construct the language model using three or four LSTM layers. Each model was trained using Stochastic Gradient Algorithm and more advanced optimization algorithms such as Adagrad, RMSprop, Adadelta, Adam, Adamax, and Nadam. Simulation study was done with Old Testament texts using a deep learning package Keras based the Theano. After pre-processing the texts, the dataset included 74 of unique characters including vowels, consonants, and punctuation marks. Then we constructed an input vector with 20 consecutive characters and an output with a following 21st character. Finally, total 1,023,411 sets of input-output vectors were included in the dataset and we divided them into training, validation, testsets with proportion 70:15:15. All the simulation were conducted on a system equipped with an Intel Xeon CPU (16 cores) and a NVIDIA GeForce GTX 1080 GPU. We compared the loss function evaluated for the validation set, the perplexity evaluated for the test set, and the time to be taken for training each model. As a result, all the optimization algorithms but the stochastic gradient algorithm showed similar validation loss and perplexity, which are clearly superior to those of the stochastic gradient algorithm. The stochastic gradient algorithm took the longest time to be trained for both 3- and 4-LSTM models. On average, the 4-LSTM layer model took 69% longer training time than the 3-LSTM layer model. However, the validation loss and perplexity were not improved significantly or became even worse for specific conditions. On the other hand, when comparing the automatically generated sentences, the 4-LSTM layer model tended to generate the sentences which are closer to the natural language than the 3-LSTM model. Although there were slight differences in the completeness of the generated sentences between the models, the sentence generation performance was quite satisfactory in any simulation conditions: they generated only legitimate Korean letters and the use of postposition and the conjugation of verbs were almost perfect in the sense of grammar. The results of this study are expected to be widely used for the processing of Korean language in the field of language processing and speech recognition, which are the basis of artificial intelligence systems.

Front-End Processing for Speech Recognition in the Telephone Network (전화망에서의 음성인식을 위한 전처리 연구)

  • Jun, Won-Suk;Shin, Won-Ho;Yang, Tae-Young;Kim, Weon-Goo;Youn, Dae-Hee
    • The Journal of the Acoustical Society of Korea
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    • v.16 no.4
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    • pp.57-63
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    • 1997
  • In this paper, we study the efficient feature vector extraction method and front-end processing to improve the performance of the speech recognition system using KT(Korea Telecommunication) database collected through various telephone channels. First of all, we compare the recognition performances of the feature vectors known to be robust to noise and environmental variation and verify the performance enhancement of the recognition system using weighted cepstral distance measure methods. The experiment result shows that the recognition rate is increasedby using both PLP(Perceptual Linear Prediction) and MFCC(Mel Frequency Cepstral Coefficient) in comparison with LPC cepstrum used in KT recognition system. In cepstral distance measure, the weighted cepstral distance measure functions such as RPS(Root Power Sums) and BPL(Band-Pass Lifter) help the recognition enhancement. The application of the spectral subtraction method decrease the recognition rate because of the effect of distortion. However, RASTA(RelAtive SpecTrAl) processing, CMS(Cepstral Mean Subtraction) and SBR(Signal Bias Removal) enhance the recognition performance. Especially, the CMS method is simple but shows high recognition enhancement. Finally, the performances of the modified methods for the real-time implementation of CMS are compared and the improved method is suggested to prevent the performance degradation.

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Home Health Care Service Using Routine Vital Sign Checkup and Electronic Health Questionnaires (주기적인 생리변수 측정과 전자건강설문을 이용한 재택건강관리서비스)

  • 박승훈;우응제;이광호;김종철
    • Journal of Biomedical Engineering Research
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    • v.22 no.5
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    • pp.469-477
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    • 2001
  • In this Paper. we describe a home health care service using electronic health questionnaires and routine checkup of vital signs Including ECG (Electrocardiography) , blood pressure. and SpO$_2$ (Oxygen Saturation) . This system is for patients at home with chronic diseases, discharged Patients, or any normal people for the Prevention of disease The service requires a home health care terminal and a PC with Interned connection installed at Patient home. The distance health care management center is equipped with a vital-sign and questionnaire interpreter as well as database, Web, and notification servers with UMS (Unified Messaging System). Participating Physician can access the servers at the center using a Web browser running on a PC available to them at any time. These components are linked together through various kinds of data and voice communication channels including PSTN (Public Switched Telephone Network) . CATV(Community Antenna TV) . Interned. and mobile communication network. Following the Physician's direction given to a Patient. he or she uses the home health care terminal to collect vital signs and fill out the questionnaire. When the terminal automatically transmits these data to the management center. the data interpreter and servers at the center process the information fo1lowing the Protocol implemented on the system. Physicians can retrieve and review data corresponding to their Patients and send back their diagnostic reports to the center. UMS at the center delivers the physician 's recommendation to the corresponding patient through the notification server. Patients can also reprieve and review their own records as well as diagnostic reports from physicians. The system Provides a new way of collecting diagnostic information and delivering doctor's recommendation to patients at home for their health management. Future works are needed in the development of new technology for measurements and interpretations of various vital signs .

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