• Proceedings of the KIEE Conference
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• 1988.07a
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• pp.225-228
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• 1988

A new transmission scheme is presented for transmitting motion video at very low data rates. In this scheme, two-level cartoons are extracted from gray-level images, and their interframe differences are coded using 2D run-length coding. Experimental results show that the proposed scheme yields compression ratio as high as 216:1.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.42 no.3 s.303
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• pp.53-62
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• 2005

In the decoding process of interframe Wavelet coding, the Wavelet transform requires huge computational complexity. Since the decoder may need to be used in various devices such as PDAs, notebooks, or PC, the decoder's complexity should be adapted to the processor's computational power. So, it is natural that the low complexity codec is also required for scalable video coding. In this paper, we develop a method of controlling and lowering the complexity of the spatial Wavelet transform while sustaining the same coding efficiency as the conventional spatial Wavelet transform. In addition, the proposed method may alleviate the ringing effect for slowly changing image sequences.

• Journal of Broadcast Engineering
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• v.2 no.2
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• pp.114-124
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• 1997

In object based coding, binary shape ccx:ling plays an important role by ccx:ling the outer shape of object. Here we propose a new shape ccx:ling tool, which enccx:les the outline of shape from a baseline. Different from 2-D (Vertex) shape ccx:ling algorithms. the proposed method encodeds the data that are extracted in a I-D fashion. The enccx:led data consist of the starting position, distance lists, and turning point lists. In the lossless ccx:ling mode, every contour pixel is input for ccx:ling, whereas variable sampling has been employed to enccx:le fewer contour pixels while preserving reasonable distortion. For interframe ccx:ling, a fast motion compensation was achieved by use of distance and turning point lists. Subjective viewing tests proved that the proposed method outperforms the current shape ccx:ling standard, CAE, in MPEG-4. In objective results for compression efficiency, the proposed method was significantly better in intraframe coding than CAE, whereas CAE was better in interframe ccx:ling.

• Proceedings of the IEEK Conference
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• 1998.06a
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• pp.677-680
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• 1998

In this paper, we proposed a lwo bit rate subband coding with adaptive vector quantization using the correlation between motion vector and block energy in subband. In this method, the difference between the input signal and the motion compensated interframe prediction signal is decomposed into several narrow bands using quadrature mirror filter (QMF) structure. The subband signals are then quantized by adaptive vector quantizers. In the codebook generating process, each classified region closer to the block value in the same region after the classification of region by the magnitude of motion vector and the variance values of subband block. Because codebook is genrated considering energy distribution of each region classified by motion vector and variance of subband block, this technique gives a very good visual quality at low bit rate coding.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.9
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• pp.124-132
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• 1994

An image data compression method for a digital VCR must satisfy special requirements such as high speed playback. various edting capabilities and error concealment to provide immunity to tape dropouts. Taking these requirements requirements into consideration, this paper proposes a new interframe subband coding algorithm for a digital VCR. In the proposed method, continuous input images are fist partitioned into four frequency bands. The lowest frequency subband is coded with 3-D block adaptive quantization that removes the level redundancy within each level. The other higher frequency subbands are coded by an intraframe coding method using the property of the human visual system. To keep reasonable image quality in high speed palyback, a segment forming method in the frequency domaing is also proposed Computer simulation results demonstrate that the proposed algorithm has the potential of achieving virtually lossless compression in normal play and produces an image with less mosaic errors in high speed play.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.7
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• pp.1487-1504
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• 1997

In this paper, the two-stage variable block-size multiresolution motion algorithm is proposed for an interframe coding scheme in the wavelet decomposition. An optimal bit allocagion between motion vectors and the prediction error in sense of minimizing the total bit rate is obtained by the proposed algorithm. The proposed algorithm consists of two stages for motion estimatation and only the first stage can be separated and run on its own. The first stage of the algorithm introduces a new method to give the lower bit rate of the displaced frame difference as well as a smooth motion field. In the second stage of the algorithm, the technique is introduced to have more accurate motion vectors in detailed areas, and to decrease the number of motion vectors in uniform areas. The algorithm aims at minimizin gthe total bit rate which is sum of the motion vectors and the displaced frame difference. The optimal bit allocation between motion vectors and displaced frame difference is accomplished by reducing the number of motion vectors in uniform areas and it is based on a botom-up construction of a quadtree. An entropy criterion aims at the control of merge operation. Simulation resuls show that the algorithm lends itself to the wavelet based image sequence coding and outperforms the conventional scheme by up to the maximum 0.28 bpp.

• MALSORI
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• no.54
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• pp.27-43
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• 2005

Existing standard speech coders can provide speech communication of high quality while they degrade the performance of speech recognition systems that use the reconstructed speech by the coders. The main cause of the degradation is that the spectral envelope parameters in speech coding are optimized to speech quality rather than to the performance of speech recognition. For example, mel-frequency cepstral coefficient (MFCC) is generally known to provide better speech recognition performance than linear prediction coefficient (LPC) that is a typical parameter set in speech coding. In this paper, we propose a speech coder using MFCC instead of LPC to improve the performance of a server-based speech recognition system in network environments. However, the main drawback of using MFCC is to develop the efficient MFCC quantization with a low-bit rate. First, we explore the interframe correlation of MFCCs, which results in the predictive quantization of MFCC. Second, a safety-net scheme is proposed to make the MFCC-based speech coder robust to channel error. As a result, we propose a 8.7 kbps MFCC-based CELP coder. It is shown from a PESQ test that the proposed speech coder has a comparable speech quality to 8 kbps G.729 while it is shown that the performance of speech recognition using the proposed speech coder is better than that using G.729.

• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.3
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• pp.21-30
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• 1990

This paper describes a hardware implementation of the interframe monochrome video CODEC using a MC-CVQ(Motion Compensated and Classified Vector Quantization) algorithm. The specifications of this CODEC are (1) the resolution of image is $128{\times}128$ pixels, and (2) the transmission rates are about 10frames/sec at the 64Kbps channel. In order to design the CODEC under these conditions, it is implemented by a multiprocessor system composed of MC unit, CVQ nuit and decoder unit, which are controlled by microprogramming technique. And the 3~stage pipelined ALU(Arithmetic and Logic Unit) is adopted to calculate the minimum error distance in the MC unit and CVQ nuit. The realized system shows that the transmission rates are 6-15 frames/sec according to the relative motion of the video signal.

• ETRI Journal
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• v.33 no.6
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• pp.871-879
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• 2011

Because a wide variety of multimedia services are provided through personal wireless communication devices, the demand for efficient bandwidth utilization becomes stronger. This demand naturally results in the introduction of the variable bitrate speech coding concept. One exemplary work is the selectable mode vocoder (SMV) that supports speech/music classification. However, because it has severe limitations in its classification performance, a couple of works to improve speech/music classification by introducing support vector machines (SVMs) have been proposed. While these approaches significantly improved classification accuracy, they did not consider correlations commonly found in speech and music frames. In this paper, we propose a novel and orthogonal approach to improve the speech/music classification of SMV codec by adaptively tuning SVMs based on interframe correlations. According to the experimental results, the proposed algorithm yields improved results in classifying speech and music within the SMV framework.

• The KIPS Transactions:PartB
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• v.9B no.6
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• pp.715-726
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• 2002

In this paper, we construct real-time moving picture encoder based on fractal theory by using general purpose digital signal processors. The constructed encoder is implemented using two fixed-point general DSPs (ADSP2181) and performs image encoding by three stage pipeline structure. In the first pipeline stage, the image grabber acquires image data from NTSC standard image signals and stores digital image into frame memory. In the second stage, the main controller encode image dada using fractal algorithm. The last stage, output controller perform Huffman coding and result the coded data via RS422 port. The performance tests of the constructed encoder shows over 10 frames/sec encoding speed for QCIF data when all the frames are encoded. When we encode the images using the interframe and redundency based on the proposed algorithms, encoding speed increased over 30 frames/sec in average.