• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.416-422
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• 2005

Speech recognition in digital communication systems has very low performance due to the spectral distortion caused by speech codecs. In this paper, the spectral distortion by speech codecs is analyzed and a pre-processing method which compensates for the spectral distortion is proposed for performance enhancement of speech recognition. Three standard speech codecs. IS-127 EVRC. ITU G.729 CS-ACELP and IS-96 QCELP. are considered for algorithm development and evaluation, and a single method which can be applied commonly to all codecs is developed. The performance of the proposed method is evaluated for three codecs, and by using the speech features extracted from the compensated spectrum. the recognition rate is improved by the maximum of $15.6\%$ compared with that using the degraded speech features.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.113-116
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• 2002

Windows provides Directshow for efficient multimedia streaming processings such as multimedia capture, storage, display and etc. Presently, many motion picture codecs and audio codecs are made to be used in Directshow framework and Windows also supports many codecs (MPEG4, H,263, WMV, WMA, ASF, etc.) in addition to a lot of useful tools for multimedia streaming processing. Therefore, Directshow can be effectively utilized for developing windows-based multimedia streaming applications such as visual monitoring systems which needs to store real-time video data for later retrieval. In this paper, we present our efforts for developing a Directshow Filter System supporting storage of motion pictures in various motion picture codecs. Our Directshow Filter system also provides an additional functionality of motion detection.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.49 no.5
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• pp.22-29
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• 2012

By using the cyclic properties of symmetric balance incomplete block design(BIBD) codes and arrayed-waveguide grating(AWG) routers, a compact optical CDMA network coder-decoder(codec) can be constructed. It can be observed that the various code families obtained by BIBD improve the BER performance compared to M-sequence code.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2248-2251
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• 2003

Communication between speech networks employing different speech codecs requires interoperability. The cascade connection of two different codecs, called tandem coding, not only degrades speech quality, but also produces high computational loads. These Problems can be solved by using the transcoding algorithm. This paper presents an effective algorithm for transcoding from G.729A to EVRC and its simulation results.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2020.07a
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• pp.510-512
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• 2020

In this paper, we reviewed the method on the codec-agnostic design of MPEG V-PCC. The codec-agnostic approach designed V-PCC can use any video codec for compression. Thus, adoption with other codecs beside HEVC has not been systematically discussed. Through the analysis of the design issues that related to MPEG EVC and JPEG. We provided a strategy on choosing and targeting different video codecs for V-PCC.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.6A
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• pp.405-413
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• 2003

This paper presents new DSP (Digital Signal Processor) instructions and their hardware architecture to efficiently implement RS (Reed-Solomon) codecs, which is one of the most widely used FEC (Forward Error Control) algorithms. The proposed DSP architecture can implement various primitive polynomials by program, and thus, hardwired codecs can be replaced. The new instructions and their hardware architecture perform GF (Galois Field) operations using the proposed GF multiplier and adder. Therefore, the proposed DSP architecture can significantly reduce the number of clock cycles compared with existing DSP chips. It can perform RS decoding rate of up to 228.1 Mbps on 130MHz DSP chips.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.466-469
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• 2002

Using intelligent information contained within the speech to identify the specific hidden data in the watermarked multimedia data is considered to be an efficient method to achieve the speech digital watermarking. This paper presents the performance comparison between various types of speech codec in order to determine an appropriate one to be used in digital watermarking applications. In the experiments, the speech signal encoded by four different types of speech codec, namely CELP, GSM, SBC and G.723.1codecs is embedded into a grayscale image, and theirs performance in term of speech recognition are compared. The method for embedding the speech signal into the host data is borrowed from a watermarking method based on the zerotrees of wavelet packet coefficients. To evaluate efficiency of the speech codec used in watermarking applications, the speech signal after being extracted from the attacked watermarked image will be played back to the listeners, and then be justified whether its content is intelligible or not.

• Electronics and Telecommunications Trends
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• v.35 no.5
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• pp.102-111
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• 2020

An increase in high-quality video service continually leads to the standardization of high-performance video codecs such as the versatile video coding standard. Although such codecs have improved coding efficiency in terms of high fidelity, a tremendous increase in the amount of video data is required for more efficient compression, especially for efficiently recognizing and analyzing the target within the millions of objects/events captured every day, such as those by surveillance systems. Therefore, newly established MPEG standardization efforts have studied the new generation of video compression standards for machine vision-oriented video. This paper presents the standardization trends in video coding for machines and discusses further directions for improvement.

• The Journal of the Acoustical Society of Korea
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• v.28 no.1E
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• pp.29-34
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• 2009

MPEG4 High Efficiency Advanced Audio Coding (HE-AAC) and MPEG Surround are one of the most efficient combinations for low bit rate multi-channel audio coding. Based on the fact that these two codecs have identical quadrature mirror filter (QMF) analysis and synthesis structures, we propose a dual-domain connection scheme for the codecs. Specifically two time-domain connection methods are analyzed and compared to the QMF subband-domain connection method. Experimental results show that both the time-domain connection methods cause no subjective sound quality degradation compared to the QMF subband-domain connection method, which verifies that one can select either of them depending on application scenarios.

• The Journal of the Acoustical Society of Korea
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• v.18 no.6
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• pp.71-76
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• 1999

In digital cellular communication systems, tandem coding occurs in communications between mobile phones with different speech codecs, resulting in poor voice quality, high computational load, and long transmission delay. In this paper, voice packet conversion technique is proposed to solve the tandem coding problems, and packet conversion algorithm from 13kbps QCELP to 8kbps QCELP is developed. Simulations using various speech data show that the proposed packet conversion method produces voice quality which is equivalent to that by the conventional tandem coding method with shorter transmission delay using about 33% computational load.