• Title/Summary/Keyword: Viterbi algorithm.

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A Watermarking Method Based on the Informed Coding and Embedding Using Trellis Code and Entropy Masking (Trellis 부호 및 엔트로피 마스킹을 이용한 정보부호화 기반 워터마킹)

  • Lee, Jeong-Hwan
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.13 no.12
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    • pp.2677-2684
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    • 2009
  • In this paper, we study a watermarking method based on the informed coding and embedding by means of trellis code and entropy masking. An image is divided as $8{\times}8$ block with no overlapping and the discrete cosine transform(DCT) is applied to each block. Then the 16 medium-frequency AC terms of each block are extracted. Next it is compared with gaussian random vectors having zero mean and unit variance. As these processing, the embedding vectors with minimum value of linear combination between linear correlation and Watson distance can be obtained by Viterbi algorithm at each stage of trellis coding. For considering the image characteristics, we apply different weight value between the linear correlation and the Watson distance using the entropy masking. To evaluate the performance of proposed method, the average bit error rate of watermark message is calculated from different several images. By the experiments the proposed method is improved in terms of the average bit error rate.

A new spect of offset and step size on BER perfermance in soft quantization Viterbi receiver (연성판정 비터비 복호기의 최적 BER 성능을 위한 오프셋 크기와 양자화 간격에 관한 성능 분석)

  • Choi, Eun-Young;Jeong, In-Tak;Song, Sang-Seb
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.27 no.1A
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    • pp.26-34
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    • 2002
  • Mobile telecommunication systems such as IS-95 and IMT-2000 employ frame based communication using frames up to 20 msec in length and the receiving end has to store the whole frome before it is being processed. The size of the frame buffer ofter dominates those of the processing unit such as soft decision Viterbi decoder. The frame buffer for IMT-2000, for example, has to be increased 80 times as large as that of IS-95. One of the parameters deciding the number of bits in a frame will be obviously the number of bits in soft quantization. Start after striking space key 2 times. This paper has studied a new aspect of offset and quantization step size on BER performance and proposes a new 3-bit soft quantization algorithm which shows similar performance as that of 4-bit soft decision Viterbi receiver. The optimal offset values and step sizes for the other practical quantization levels ---16, 8, 4, 2--- have also been found. In addition, a new optimal symbol metric table has been devised which takes the accumulation value of various repeated signals and produces a rescaled 3-bit valu.tart after striking space key 2 times.

Design and Implementation of Wireless Modem for Indoor Data Communication (구내 데이터 통신용 무선모뎀 설계 및 구현)

  • Cho, Byung-Hak
    • Journal of The Institute of Information and Telecommunication Facilities Engineering
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    • v.11 no.1
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    • pp.16-22
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    • 2012
  • Wireless data communication is easy to be affected by channel noise and degrade reliability and stability by the multipath fading and ISI compared with wired data communication. In this paper, we designed and implemented indoor wireless modem adopted DQPSK modulation scheme for improvement of bandwidth efficiency, and convolutional encoding, Viterbi decoding and hybrid ARQ algorithm combinig FEC with CRC for efficient error control in indoor wireless channel. Testing the implemented wireless modem, we verified the proposed scheme is proper to efficient and reliable indoor wireless data communication.

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Text-Dependent Speaker Recognition Using DTW and State-Dependent Parameter Weighting Method of HMM (DTW 와 HMM의 상태별 파라미터 가중 기법을 이용한 문맥 종속형 화자인식)

  • 이철희;정성환;김종교
    • Proceedings of the IEEK Conference
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    • 2000.06d
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    • pp.77-80
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    • 2000
  • In this paper, the speaker-recognition process based on both DTW and discrete HMM was performed using the method to evaluate state-dependent parameter weighting from training data so as the personal audio-characteristics are to be well reflected. In the suggested method below, we found the optimal state sequence using the Viterbi algorithm. The optimal path could be evaluated after comparing the sequence of base pattern which already have, with that of the other patterns. After that the frame of which the pattern was matched with the base pattern in the same state are to be found so that the reference pattern can be gained by weighting on the numbers of matched frames.

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Reduced-state sequence estimation for TC 8PSK/OFDM with 2-stage IDFT/DFTs (두단계 IDFT/DFT를 갖는 TC 8PSK/OFDM를 위한 RSSE 방식)

  • Kang Hoon-Chul;Ko Sang-Bo;Jwa Jeong-Woo
    • Proceedings of the IEEK Conference
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    • 2004.06a
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    • pp.147-150
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    • 2004
  • In this paper, we propose a reduced-state sequence estimation (RSSE) for trellis coded modulation (TCM) in OFDM with two-stage IDFT/ DFTs, MMSE-LE, and interleaving on frequency selective Rayleigh fading channels. The Viterbi algorithm (VA) is used to search for the best path through the reduced-state trellis combined with equalization and TCM decoding. Computer simulations confirm the bit error probability of the proposed scheme.

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Development of a lipsync algorithm based on A/V corpus (코퍼스 기반의 립싱크 알고리즘 개발)

  • 하영민;김진영;정수경
    • Proceedings of the IEEK Conference
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    • 2000.09a
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    • pp.145-148
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    • 2000
  • 이 논문에서는 2차원 얼굴 좌표데이터를 합성하기 위한 음성과 영상 동기화 알고리즘을 제안한다. 영상변수의 획득을 위해 화자의 얼굴에 부착된 표시를 추적함으로써 영상변수를 획득하였고, 음소정보뿐만 아니라 운율정보들과의 영상과의 상관관계를 분석하였으며 합성단위로 시각소에 기반한 코퍼스를 선택하고, 주변의 음운환경도 함께 고려하여 연음현상을 모델링하였다. 입력된 코퍼스에 해당되는 패턴들을 lookup table에서 선택하여 주변음소에 대해 기준패턴과의 음운거리를 계산하고 음성파일에서 운율정보들을 추출해 운율거리를 계산한 후 가중치를 주어 패턴과의 거리를 얻는다. 이중가장 근접한 다섯개의 패턴들의 연결부분에 대해 Viterbi Search를 수행하여 최적의 경로를 선택하고 주성분분석된 영상정보를 복구하고 시간정보를 조절한다.

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Automatic Speech Database Verification Method Based on Confidence Measure

  • Kang Jeomja;Jung Hoyoung;Kim Sanghun
    • MALSORI
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    • no.51
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    • pp.71-84
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    • 2004
  • In this paper, we propose the automatic speech database verification method(or called automatic verification) based on confidence measure for a large speech database. This method verifies the consistency between given transcription and speech using the confidence measure. The automatic verification process consists of two stages : the word-level likelihood computation stage and multi-level likelihood ratio computation stage. In the word-level likelihood computation stage, we calculate the word-level likelihood using the viterbi decoding algorithm and make the segment information. In the multi-level likelihood ratio computation stage, we calculate the word-level and the phone-level likelihood ratio based on confidence measure with anti-phone model. By automatic verification, we have achieved about 61% error reduction. And also we can reduce the verification time from 1 month in manual to 1-2 days in automatic.

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Maximum Likelihood Receivers for DAPSK Signaling

  • Xiao Lei;Dong Xiaodai;Tjhung Tjeng T.
    • Journal of Communications and Networks
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    • v.8 no.2
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    • pp.205-211
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    • 2006
  • This paper considers the maximum likelihood (ML) detection of 16-ary differential amplitude and phase shift keying (DAPSK) in Rayleigh fading channels. Based on the conditional likelihood function, two new receiver structures, namely ML symbol-by-symbol receiver and ML sequence receiver, are proposed. For the symbol-by-symbol detection, the conventional DAPSK detector is shown to be sub-optimum due to the complete separation in the phase and amplitude detection, but it results in very close performance to the ML detector provided that its circular amplitude decision thresholds are optimized. For the sequence detection, a simple Viterbi algorithm with only two states are adopted to provide an SNR gain around 1 dB on the amplitude bit detection compared with the conventional detector.

The Performance Improvement of BASK System of Giga-Bit MODEM Using the Fuzzy System (퍼지 시스템을 이용한 Giga-Bit MODEM의 BASK 시스템 성능 개선)

  • Eom, Ki-Hwan;Lee, Kyu-Yun
    • Journal of Institute of Control, Robotics and Systems
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    • v.13 no.5
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    • pp.462-466
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    • 2007
  • This paper proposes an automatic bandwidth control method for the performance improvement of Binary Amplitude Shift Keying (BASK) system for Giga-Bit Modem in millimeter band. In order to improve the performance of the BASK system with a fixed bandwidth, the proposed method is to adjust a bandwidth of low pass filter in receiver using the fuzzy system. The BASK system consists of a high speed shutter of the transmitter and a counter and a repeater of receiver. The repeater consists of four stage converters, and a converter is constructed with a low pass filter and a limiter. The inputs to the fuzzy system are the reminder and integral of remainder of counter, and output is a bandwidth. We used a Viterbi algorithm to find the optimum detection from the output of the counter. Simulation results showed that the proposed system improves the performance compared to the fixed bandwidth.

The Voice Dialing System Using Dynamic Hidden Markov Models and Lexical Analysis (DHMM과 어휘해석을 이용한 Voice dialing 시스템)

  • 최성호;이강성;김순협
    • Journal of the Korean Institute of Telematics and Electronics B
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    • v.28B no.7
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    • pp.548-556
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    • 1991
  • In this paper, Korean spoken continuous digits are ercognized using DHMM(Dynamic Hidden Markov Model) and lexical analysis to provide the base of developing voice dialing system. After segmentation by phoneme unit, it is recognized. This system can be divided into the segmentation section, the design of standard speech section, the recognition section, and the lexical analysis section. In the segmentation section, it is segmented using the ZCR, O order LPC cepstrum, and Ai, parameter of voice speech dectaction, which is changed according to time. In the standard speech design section, 19 phonemes or syllables are trained by DHMM and designed as a standard speech. In the recognition section, phomeme stream are recognized by the Viterbi algorithm.In the lexical decoder section, finally recognized continuous digits are outputed. This experiment shiwed the recognition rate of 85.1% using data spoken 7 times of 21 classes of 7 continuous digits which are combinated all of the occurence, spoken by 10 man.

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