• Title/Summary/Keyword: Data quantization

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A Blind Video Watermarking Technique Using Luminance Masking and DC Modulus Algorithm (휘도 마스킹과 DC Modulus 알고리즘을 이용한 비디오 워터마킹)

  • Jang Yong-Won;Kim, In-Taek;Han, Seung-Soo
    • The Transactions of the Korean Institute of Electrical Engineers D
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    • v.51 no.7
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    • pp.302-307
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    • 2002
  • Digital watermarking is the technique, which embeds an invisible signal including signal including owner identification and copy control information into multimedia data such as audio, video, and images for copyright protection. A new MPEG watermark embedding algorithm using complex block effect based on the Human Visual System(HVS) is introduced in this paper. In this algorithm, $8{\times}8$ dark blocks are selected, and the watermark is embedded in the DC component of the discrete cosine transform(DCT) by using quantization and modulus calculation. This algorithm uses a blind watermark retrieval technique, which detects the embedded watermark without using the original image. The experimental results show that the proposed watermark technique is robust against MPEG coding, bitrate changes, and various GOP(Group of Picture) changes.

A Simple Discrete Cosine Transform Systolic Array Based on DFT for Video Codec (DFT에 의한 비데오 코덱용 DCT의 단순한 시스톨릭 어레이)

  • 박종오;이광재;양근호;박주용;이문호
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.26 no.11
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    • pp.1880-1885
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    • 1989
  • In this paper, a new approach for systolic array realizing the discrete cosine transform (DCT) based on discrete Fourier transform (DFT) of an input sequence is presented. The proposed array is based on a simple modified DFT(MDFT) version of the Goertzel algorithm combined with Kung's approach and is proved perfectly. This array requires N cells, one multiplier and takes N clock cycles to produce a complete N-point DCT and also is able to process a continuous stream of data sequences. We have analyzed the output signal-to-noise ratio(SNR) and designed the circuit level layout of one-PE chip. The array coefficients are static adn thus stored-product ROM's can be used in place of multipliers to limit cost as eliminate errors due to coefficients quantization.

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Satellite Image Data Coding Using Wavelet Transform and Selectively Predictive Vector Quantization (웨이브릿 변환과 선택적 예측 벡터 양자화를 이용한 인공위성 화상데이터의 부호화)

  • 반성원;김병주;김경규;정원식;김영춘;신용달;김건일
    • Journal of the Korean Institute of Telematics and Electronics S
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    • v.36S no.4
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    • pp.38-44
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    • 1999
  • 본 논문에서는 웨이브릿 변환과 선택적 예측 벡터양자화를 이용한 인공위성 화상데이타 부호화 방법을 제안하였다. 이 방법에서는 대역내 중복성을 제거하기 위하여 각각의 대역을 웨이브릿 변환하고, 대역간 중복성을 제거하기 위해 에측하는 대역으로부터 생성된 임계치 지도를 이용하여 선택적 예측 벡터양자화를 행한다. 따라서 이 방법은 대역내 및 대역간 중복성을 효과적으로 제거하기 때문에 부호화 효율을 향상시킨다. 이 방법을 실제 Landsat TM 인공위성 화상데이타에 실험한 결과 기존의 방법에 비하여 부호화 효율이 향상됨을 확인하였다.

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Speaker Identification Using GMM Based on LPCA (LPCA에 기반한 GMM을 이용한 화자 식별)

  • Seo, Chang-Woo;Lee, Youn-Jeong;Lee, Ki-Yong
    • Speech Sciences
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    • v.12 no.2
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    • pp.171-182
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    • 2005
  • An efficient GMM (Gaussian mixture modeling) method based on LPCA (local principal component analysis) with VQ (vector quantization) for speaker identification is proposed. To reduce the dimension and correlation of the feature vector, this paper proposes a speaker identification method based on principal component analysis. The proposed method firstly partitions the data space into several disjoint regions by VQ, and then performs PCA in each region. Finally, the GMM for the speaker is obtained from the transformed feature vectors in each region. Compared to the conventional GMM method with diagonal covariance matrix, the proposed method requires less storage and complexity while maintaining the same performance requires less storage and shows faster results.

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A Study on Design and Implementation of Embedded System for speech Recognition Process

  • Kim, Jung-Hoon;Kang, Sung-In;Ryu, Hong-Suk;Lee, Sang-Bae
    • Journal of the Korean Institute of Intelligent Systems
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    • v.14 no.2
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    • pp.201-206
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    • 2004
  • This study attempted to develop a speech recognition module applied to a wheelchair for the physically handicapped. In the proposed speech recognition module, TMS320C32 was used as a main processor and Mel-Cepstrum 12 Order was applied to the pro-processor step to increase the recognition rate in a noisy environment. DTW (Dynamic Time Warping) was used and proven to be excellent output for the speaker-dependent recognition part. In order to utilize this algorithm more effectively, the reference data was compressed to 1/12 using vector quantization so as to decrease memory. In this paper, the necessary diverse technology (End-point detection, DMA processing, etc.) was managed so as to utilize the speech recognition system in real time

A study of broad board classification of korean digits using symbol processing (심볼을 이용한 한국어 숫자음의 광역 음소군 분류에 관한 연구)

  • Lee, Bong-Gu;Lee, Guk;Hhwang, Hee-Yoong
    • Proceedings of the KIEE Conference
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    • 1989.07a
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    • pp.481-485
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    • 1989
  • The object of this parer is on the design of an broad board classifier for connected. Korean digit. Many approaches have been applied in speech recognition systems: parametric vector quantization, dynamic programming and hiden Markov model. In the 80's the neural network method, which is expected to solve complex speech recognition problems, came bach. We have chosen the rule based system for our model. The phoneme-groups that we wish to classify are vowel_like, plosive_like fricative_like, and stop_like.The data used are 1380 connected digits spoken by three untrained male speakers. We have seen 91.5% classification rate.

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Fuzzy Neural Network Model Using A Learning Rule Considering the Distances Between Classes (클래스간의 거리를 고려한 학습법칙을 사용한 퍼지 신경회로망 모델)

  • Kim Yong-Soo;Baek Yong-Sun;Lee Se-Yul
    • Journal of the Korean Institute of Intelligent Systems
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    • v.16 no.4
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    • pp.460-465
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    • 2006
  • This paper presents a new fuzzy learning rule which considers the Euclidean distances between the input vector and the prototypes of classes. The new fuzzy learning rule is integrated into the supervised IAFC neural network 4. This neural network is stable and plastic. We used iris data to compare the performance of the supervised IAFC neural network 4 with the performances of back propagation neural network and LVQ algorithm.

A New Proposal of Extended BTC for Picture Data Compression (영상압축을 위한 확장된 BTC의 새로운 제안)

  • 고형화;이충웅
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.25 no.1
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    • pp.81-87
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    • 1988
  • This paper proposes a new EBTC(extended block truncation coding) algorithm extended from the BTC for image compression. The EBTC has a capability to eliminate the defects of BTC, such as the deterioration of resolution or blocky effect,and to make a real-time processing like BTC. It shows better performances than the DPCM and the transform coding. Especially, it is a suitable coding method for the high quality picture transmission. It may be adequate to the system of transmission rate of 30-50 Mbits/sec. The picture quality has been scarecely degraded with a vector quantization to the EBTC output at the bit rate of 1.25 bits/pel. The bit rate of the scalar quantized EBTC method is 2.6-3.7 bits/pel.

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On a Improvement of Pitch Search Time for Real Time Implementation in IMBE Vocoder (IMBE Vocoder 실시간 처리를 위한 피치 검색 시간 개선에 관한 연구)

  • Jang KyungA;KIM JeongJin;Min So Yeon;Bae MyungJin
    • Proceedings of the Acoustical Society of Korea Conference
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    • spring
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    • pp.24-27
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    • 1999
  • IMBE(Improved Multi-Band Excitation) vocoders exhibit good performance at low data rates. The major drawback to IMBE coders is their large computational requirements. In this paper, thus, we propose a new pitch search method that preserves the quality of the IMBE vocoder with reduced complexity. The basic idea is to skip unnecessary range of the pitch searching by using the quantization error. Applying the proposed method to the IMBE vocoder, we can get approximately $45.88\%$ processing time reduction and there is no difference in voice quality between conventional IMBE and proposed IMBE.

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Hight throughput CORDIC-based Direct Digital Frequency Synthesizer (고속 CORDIC에 기반한 직접 디지털 주파수 합성기)

  • Park, Minkyoung;Park, Sungsoo;Kim, Kiseon;Lee, Jeong-A
    • Proceedings of the IEEK Conference
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    • 1999.11a
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    • pp.784-787
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    • 1999
  • This paper describes a direct digital frequency synthesizer using the CORDIC algorithm, which can be implemented efficiently for a digital sinusoid synthesis. To optimize the hardware design parameters, we perform numerical analysis of the quantization effects for the CORDIC-based architecture. A pipelined architecture is employed to obtain a high data throughput,. We estimate and summarize its hardware costs for a variable accuracy, and a CORDIC-based architecture for 9 bit accuracy is emulated in FPGA.

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