• The Journal of Korean Institute of Communications and Information Sciences
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• v.42 no.2
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• pp.432-443
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• 2017

The primary goal of the MPEG-H 3D Audio standard is to provide immersive audio environments for high-resolution broadcasting services such as UHDTV. This standard incorporates a wide range of technologies such as encoding/decoding technology for multi-channel/object/scene-based signal, rendering technology for providing 3D audio in various playback environments, and post-processing technology. The reference software decoder of this standard is a structure combining several modules and can operate in various modes. Each module is composed of independent executable files and executed sequentially, real time decoding is impossible. In this paper, we make DLL library of the core decoder, format converter, object renderer, and binaural renderer of the standard and integrate them to enable frame-based decoding. In addition, by measuring the computation complexity of each mode of the MPEG-H 3D-Audio decoder, this paper also provides a reference for selecting the appropriate decoding mode for various hardware platforms. As a result of the computational complexity measurement, the low complexity profiles included in Korean broadcasting standard has a computation complexity of 2.8 times to 12.4 times that of the QMF synthesis operation in case of rendering as a channel signals, and it has a computation complexity of 4.1 times to 15.3 times of the QMF synthesis operation in case of rendering as a binaural signals.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.93-96
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• 2008

This paper describes an efficient design of a multi-standard Viterbi decoder that supports multiple constraint lengths and code rates. The Viterbi decode. is parameterized for the code rates 1/2, 1/3 and constraint lengths 7, 9, thus it has four operation modes. In order to achieve low hardware complexity and low power, an efficient architecture based on hardware sharing techniques is devised. Also, the optimization of ACCS (Accumulate-Subtract) circuit for the one-point trace-back algorithm reduces its area by about 35% compared to the full parallel ACCS circuit. The parameterized Viterbi decoder core has 79,818 gates and 25,600 bits memory, and the estimated throughput is about 105 Mbps at 70 MHz clock frequency.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.05a
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• pp.132-135
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• 2011

This paper describes a multi-mode LDPC decoder which supports three block lengths(648, 1296, 1944) and four code rates(1/2, 2/3, 3/4, 5/6) of IEEE 802.11n WLAN standard. To minimize hardware complexity, it adopts a block-serial (partially parallel) architecture based on the layered decoding scheme. A novel memory reduction technique devised using the min-sum decoding algorithm reduces the size of check-node memory by 47% as compared to conventional method. The designed LDPC decoder is verified by FPGA implementation, and synthesized with a $0.18-{\mu}m$ CMOS cell library. It has 219,100 gates and 45,036 bits RAM, and the estimated throughput is about 164~212 Mbps at 50 MHz@2.5v.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.827-830
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• 2000

3GPP and ETSI adopted AMR(Adaptive Multi-Rate) as a standard for next generation IMT-2000 service. In this paper, we analyzed algorithm about AMR and optimized ANSI C source on the C complier and assembly language of Texas Instrument . The implemented AMR speech codec requires 28.2MIPS of complexity for encoder and 5.5MIPS for decoder. we performed real-time implementation of AMR speech codec using 82% of TMS320C5402 with 40 MIPS specification. We give proof that the output speech of the implemented speech codec on DSP board is identical with result of C source program simulation. Also the reconstructed speech is verified in the real-time environment consisted of microphone and speaker.

• The Journal of the Acoustical Society of Korea
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• v.16 no.3E
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• pp.56-61
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• 1997

In this paper, we describe a real-time implementation of a multi-channel auido codec system that is based on the MPEG-1 audio algorithm. The major feature of this system is that it has a flexible multi-DSP system that can be adapted for various applications with using up to four TMS320C40 DSPs. The purpose of this paper is to present the problems of the system and is to describe the optimized methods to solve the problems in the view of hardware and software. Our audio codec is composed of an encoder an a decoder system and the bit rate of bitstream is up to 384 kbps. Fast input/output interfaces, DSP overloads, and inter-DSP communications methods with high speed are considered in multi-DSP H/W. Also, to run real-time in S/W, optimizing methods of algorithm are considered. After implementation of system, the subjective assessment method, and 'triple stimulus/hidden reference/double blind' that recommended by ITU-R TG10/3 is adopted for the quality of our system. All test items except one are awarded difference grades(diffgrade) better than 1-. Form the results, multi-channel audio system can be used for HDTV service.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.494-496
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• 2002

A bi-directional and multi-channel wireless telemetry capsule, 11mm in diameter, is presented that can transmit video images from inside the human body and receive a control signal from an external control unit. The proposed telemetry capsule includes transmitting and receiving antennas, a demodulator, decoder, four LEDs, and CMOS image sensor, along with their driving circuits. The receiver demodulates the received signal radiated from the external control unit. Next, the decoder receives the stream of control signals and interprets five of the binary digits as an address code. Thereafter, the remaining signal is interpreted as four bits of binary data. Consequently, the proposed telemetry module can demodulate external signals so as to control the behavior of the camera and four LEDs during the transmission of video images. The proposed telemetry capsule can simultaneously transmit a video signal and receive a control signal determining the behavior of the capsule itself. As a result, the total power consumption of the telemetry capsule can be reduced by turning off the camera power during dead time and separately controlling the LEDs for proper illumination of the intestine.

• Journal of Broadcast Engineering
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• v.6 no.1
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• pp.66-71
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• 2001

This paper Introduces an implementation of MPEG-2 AAC codec system for digital audio broadcasting. This system consists of the encoder and the decoder. This system includes MPEG-2 system multiplexing and demultiplexing modules for Interfacing to the ETRI-DAB system. Four DSPs are adopted for the encoder and three DSPs for 7he decoder. Each DSP Processes system control. 1/0 control, audio signal processing. multiplexing and demultiplexing. This Paper also discusses some near future estimations relaxed to the DAB system and it\`s services. Currently a stereo audio codec is available but multi-channel audio codec and MPEG-4 audio cosec wall be also Implemented.

• Journal of Electrical Engineering and Technology
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• v.9 no.4
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• pp.1411-1417
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• 2014

An MPEG-2 AAC Huffman decoding method based on the fixed length compacted codeword tables, where each codeword can contain multiple number of Huffman codes, was proposed. The proposed method enhances the searching efficiency by finding multiple symbols in a single search, i.e., a direct memory reading of the compacted codeword table. The memory usage is significantly saved by separately handling the Huffman codes that exceed the length of the compacted codewords. The trade-off relation between the computational complexity and the amount of memory usage was analytically derived to find the proper codeword length of the compacted codewords for the design of MPEG-2 AAC decoder. To validate the proposed algorithm, its performance was experimentally evaluated with an implemented MPEG-2 AAC decoder. The results showed that the computational complexity of the proposed method is reduced to 54% of that of the most up-to-date method.

• 한국정보통신설비학회:학술대회논문집
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• 2005.08a
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• pp.417-421
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• 2005

In the recent digital communication systems, the performance of Turbo Code using the error correction coding depends on the interleaver influencing the free distance determination and the recursive decoding algorithms that is executed in the turbo decoder. However, performance depends on the interleaver depth that needs many delays over the reception process. Moreover, turbo code has been known as the robust coding methods with the confidence over the fading channel. International Telecommunication Union(ITU) has recently adopted it as the standardization of the channel coding over the third generation mobile communications(IMT-2000). Therefore, in this paper, we proposed the interleaver that has the better performance than existing block interleaver, and modified turbo decoder that has the parallel concatenated structure using MAP algorithm. In the real-time voice and video service over third generation mobile communications, the performance of the proposed two methods was analyzed and compared with the existing methods by computer simulation in terms of reduced decoding delay using the variable decoding method over AWGN and fading channels for CDMA environments.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.18 no.1
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• pp.15-29
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• 2024

A fetoscope is an optical endoscope, which is often applied in fetoscopic laser photocoagulation to treat twin-to-twin transfusion syndrome. In an operation, the clinician needs to observe the abnormal placental vessels through the endoscope, so as to guide the operation. However, low-quality imaging and narrow field of view of the fetoscope increase the difficulty of the operation. Introducing an accurate placental vessel segmentation of fetoscopic images can assist the fetoscopic laser photocoagulation and help identify the abnormal vessels. This study proposes a method to solve the above problems. A novel encoder-decoder network with a dual-path structure is proposed to segment the placental vessels in fetoscopic images. In particular, we introduce a channel attention mechanism and a continuous convolution structure to obtain multi-scale features with their weights. Moreover, a switching connection is inserted between the corresponding blocks of the two paths to strengthen their relationship. According to the results of a set of blood vessel segmentation experiments conducted on a public fetoscopic image dataset, our method has achieved higher scores than the current mainstream segmentation methods, raising the dice similarity coefficient, intersection over union, and pixel accuracy by 5.80%, 8.39% and 0.62%, respectively.