• Title/Summary/Keyword: Rate adaptation

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N- gram Adaptation Using Information Retrieval and Dynamic Interpolation Coefficient (정보검색 기법과 동적 보간 계수를 이용한 N-gram 언어모델의 적응)

  • Choi Joon Ki;Oh Yung-Hwan
    • MALSORI
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    • no.56
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    • pp.207-223
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    • 2005
  • The goal of language model adaptation is to improve the background language model with a relatively small adaptation corpus. This study presents a language model adaptation technique where additional text data for the adaptation do not exist. We propose the information retrieval (IR) technique with N-gram language modeling to collect the adaptation corpus from baseline text data. We also propose to use a dynamic language model interpolation coefficient to combine the background language model and the adapted language model. The interpolation coefficient is estimated from the word hypotheses obtained by segmenting the input speech data reserved for held-out validation data. This allows the final adapted model to improve the performance of the background model consistently The proposed approach reduces the word error rate by $13.6\%$ relative to baseline 4-gram for two-hour broadcast news speech recognition.

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Analysis of Adaptation and Self-Consciousness between Supervised and Unsupervised Children (가족구조에 따른 자기보호아동과 성인보호아동의 학교적응 및 자의식 정서)

  • Lee, Jung-Sook;Kim, Eun-Kyung
    • Journal of the Korean Home Economics Association
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    • v.46 no.2
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    • pp.85-96
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    • 2008
  • Given the evolving nature of the family unit, a large number of children are being left unsupervised after school. The purpose of this study is to understand the adaptation ability and emotional capacity of these children. To achieve this objective, we investigated the different characteristics of 708 middle-school students in Seoul, dividing them into two categories adult-care children, for whom adults provide care after school, and self-care children for whom no adult supervision was present. In particular, we examined children's adaptation to the school environment and possible self-consciousness difference between self-care and adult-care children, in consideration of their family characteristic; divorced, separated, widowed parent, remarried parents, ordinary families. The results showed that self-care children tend to have a higher rate of shame-proneness and guilt-proneness compared to adult-care children. Furthermore, self-care children exhibited lower school adaptation rate than adult-care children. There was no significant difference in schoolmate relationships between the two groups. In relation to specific family structures, children from reorganized families showed no significant differences in school adaptation and self-conscious, while self-care children from ordinary families revealed low school adaptation and high self-conscious characteristics. The results of this study are critical in the effective analysis and understanding of children's adaptive and emotional behaviors arising from changes in their family structure.

Utility-Based MPEG-21 Video Adaptation for Universal Multimedia Access (UMA를 위한 유틸리티 기반 MPEG-21 비디오 적응)

  • 김재곤;강경옥;김진웅;김형명
    • Proceedings of the IEEK Conference
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    • 2003.07d
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    • pp.1491-1494
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    • 2003
  • Video adaptation in response to dynamic resource conditions and user preferences is required as a key technology to enable universal multimedia access (UMA) through heterogeneous networks by a multitude of devices in a seamless way. Although many adaptation techniques exist, selections of appropriate adaptations among multiple choices are often ad hoc. To provide a systematic solution, we present a general conceptual framework to model video entity, adaptation, resource, utility, and relations among them. It allows for formulation of various adaptation problems as resource-constrained utility maximization. We apply the framework to a practical case of dynamic bit rate adaptation. Furthermore, we present a description tool, which has been accepted as a part of the MPEG-21 Digital Item Adaptation (DIA), along with a brief overview of the .elated descriptors to support terminal and network quality of service (QoS).

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A Simulation Study on Improvements of Speech Processing Strategy of Cochlear Implants Using Adaptation Effect of Inner Hair Cell and Auditory Nerve Synapse (청각신경 시냅스의 적응 효과를 이용한 인공와우 어음처리 알고리즘의 개선에 대한 시뮬레이션 연구)

  • Kim, Jin-Ho;Kim, Kyung-Hwan
    • Journal of Biomedical Engineering Research
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    • v.28 no.2
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    • pp.205-211
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    • 2007
  • A novel envelope extraction algorithm for speech processor of cochlear implants, called adaptation algorithm, was developed which is based on a adaptation effect of the inner hair cell(IHC)/auditory nerve(AN) synapse. We achieved acoustic simulation and hearing experiments with 12 normal hearing persons to compare this adaptation algorithm with existent standard envelope extraction method. The results shows that speech processing strategy using adaptation algorithm showed significant improvements in speech recognition rate under most channel/noise condition, compared to conventional strategy We verified that the proposed adaptation algorithm may yield better speech perception under considerable amount of noise, compared to the conventional speech processing strategy.

SVM Based Speaker Verification Using Sparse Maximum A Posteriori Adaptation

  • Kim, Younggwan;Roh, Jaeyoung;Kim, Hoirin
    • IEIE Transactions on Smart Processing and Computing
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    • v.2 no.5
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    • pp.277-281
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    • 2013
  • Modern speaker verification systems based on support vector machines (SVMs) use Gaussian mixture model (GMM) supervectors as their input feature vectors, and the maximum a posteriori (MAP) adaptation is a conventional method for generating speaker-dependent GMMs by adapting a universal background model (UBM). MAP adaptation requires the appropriate amount of input utterance due to the number of model parameters to be estimated. On the other hand, with limited utterances, unreliable MAP adaptation can be performed, which causes adaptation noise even though the Bayesian priors used in the MAP adaptation smooth the movements between the UBM and speaker dependent GMMs. This paper proposes a sparse MAP adaptation method, which is known to perform well in the automatic speech recognition area. By introducing sparse MAP adaptation to the GMM-SVM-based speaker verification system, the adaptation noise can be mitigated effectively. The proposed method utilizes the L0 norm as a regularizer to induce sparsity. The experimental results on the TIMIT database showed that the sparse MAP-based GMM-SVM speaker verification system yields a 42.6% relative reduction in the equal error rate with few additional computations.

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Rapid Speaker Adaptation Based on Eigenvoice Using Weight Distribution Characteristics (가중치 분포 특성을 이용한 Eigenvoice 기반 고속화자적응)

  • 박종세;김형순;송화전
    • The Journal of the Acoustical Society of Korea
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    • v.22 no.5
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    • pp.403-407
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    • 2003
  • Recently, eigenvoice approach has been widely used for rapid speaker adaptation. However, even in the eigenvoice approach, Performance improvement using very small amount of adaptation data is relatively small in comparison with that using somewhat large adaptation data because the reliable estimation of weights of eigenvoice is difficult. In this paper, we propose a rapid speaker adaptation method based on eigenvoice using the weight distribution characteristics to improve the performance on a small adaptation data. In the Experimental results on vocabulary-independent word recognition task (using PBW 452 database), the weight threshold method alleviates the problem of relatively low performance for a tiny small adaptation data. When single adaptation word is used, word error rate is reduced about 9-18% by the weight threshold method.

Efficient Power and Rate Adaptation Strategy for Improved Spectral Efficiency in Multi-Carrier DS-CDMA Communications (다중 반송파 부호분할 다중접속 통신에서의 주파수 효율 향상을 위한 효율적인 전력 및 전송률 적응화 기법)

  • Lee, Ye Hoon;Kim, Dong Ho
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.38A no.8
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    • pp.697-703
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    • 2013
  • We propose an efficient frequency-time domain resource allocation scheme in multi-carrier (MC) direct-sequence code-division multiple-access (DS/CDMA) communications. We consider, as a power allocation strategy in the frequency domain, transmitting each user's DS waveforms over the user's sub-band with the largest channel gain. We then consider rate adaptation in the time domain, where the data rate is adapted such that a desired transmission quality is maintained. We analyze the achievable average data rate of the proposed scheme with fixed average transmission power, and compare the performance to single carrier DS/CDMA systems with power and rate adaptations.

Queueing Theoretic Approach to Playout Buffer Model for HTTP Adaptive Streaming

  • Park, Jiwoo;Chung, Kwangsue
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.12 no.8
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    • pp.3856-3872
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    • 2018
  • HTTP-based adaptive streaming (HAS) has recently been widely deployed on the Internet. In the HAS system, a video content is encoded at multiple bitrates and the encoded video content is segmented into small parts of fixed durations. The HAS client requests a video segment and stores it in the playout buffer. The rate adaptation algorithm employed in HAS clients dynamically determines the video bitrate depending on the time-varying bandwidth. Many studies have shown that an efficient rate adaptation algorithm is critical to ensuring quality-of-experience in HAS systems. However, existing algorithms have problems estimating the network bandwidth because bandwidth estimation is performed on the client-side application stack. Without the help of transport layer protocols, it is difficult to achieve accurate bandwidth estimation due to the inherent segment-based transmission of the HAS. In this paper, we propose an alternative approach that utilizes the playout buffer occupancy rather than using bandwidth estimates obtained from the application layer. We start with a queueing analysis of the playout buffer. Then, we present a buffer-aware rate adaptation algorithm that is solely based on the mean buffer occupancy. Our simulation results show that compared to conventional algorithms, the proposed algorithm achieves very smooth video quality while delivering a similar average video bitrate.

A Noble Decoding Algorithm Using MLLR Adaptation for Speaker Verification (MLLR 화자적응 기법을 이용한 새로운 화자확인 디코딩 알고리듬)

  • 김강열;김지운;정재호
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.2
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    • pp.190-198
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    • 2002
  • In general, we have used the Viterbi algorithm of Speech recognition for decoding. But a decoder in speaker verification has to recognize same word of every speaker differently. In this paper, we propose a noble decoding algorithm that could replace the typical Viterbi algorithm for the speaker verification system. We utilize for the proposed algorithm the speaker adaptation algorithms that transform feature vectors into the region of the client' characteristics in the speech recognition. There are many adaptation algorithms, but we take MLLR (Maximum Likelihood Linear Regression) and MAP (Maximum A-Posterior) adaptation algorithms for proposed algorithm. We could achieve improvement of performance about 30% of EER (Equal Error Rate) using proposed algorithm instead of the typical Viterbi algorithm.

Acoustic and Pronunciation Model Adaptation Based on Context dependency for Korean-English Speech Recognition (한국인의 영어 인식을 위한 문맥 종속성 기반 음향모델/발음모델 적응)

  • Oh, Yoo-Rhee;Kim, Hong-Kook;Lee, Yeon-Woo;Lee, Seong-Ro
    • MALSORI
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    • v.68
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    • pp.33-47
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    • 2008
  • In this paper, we propose a hybrid acoustic and pronunciation model adaptation method based on context dependency for Korean-English speech recognition. The proposed method is performed as follows. First, in order to derive pronunciation variant rules, an n-best phoneme sequence is obtained by phone recognition. Second, we decompose each rule into a context independent (CI) or a context dependent (CD) one. To this end, it is assumed that a different phoneme structure between Korean and English makes CI pronunciation variabilities while coarticulation effects are related to CD pronunciation variabilities. Finally, we perform an acoustic model adaptation and a pronunciation model adaptation for CI and CD pronunciation variabilities, respectively. It is shown from the Korean-English speech recognition experiments that the average word error rate (WER) is decreased by 36.0% when compared to the baseline that does not include any adaptation. In addition, the proposed method has a lower average WER than either the acoustic model adaptation or the pronunciation model adaptation.

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