• Journal of Broadcast Engineering
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• v.2 no.1
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• pp.24-35
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• 1997

Basic and transmission video quality testing was performed in real-time mode using hardware video codec based on MPEG-2 MP@ML standard including subsequent transmitting and receiving satellite simulator unit. The double-stimulus Impairment scale method and the double-stimulus continous quality scale method based on CCIR 500-5 were used as an evaluation method. The whole digital broadcasting satellite system consisting of MPEG-2 codec, system mux/demux, channel codec, channel, modem, antenna, etc. was put into the overall video quality testbed and the basic and transmission error quality assessment was performed at various bitrates and BER for an integrated system performance evaluation. In transmission error video quality testing, transmission error video quality maintained on average above 3.9 point on the 5-point scale. The low-bit rate quality such as film mode(@2Mbps) highly depended on the statitical characteristics of video source and maintained on average around 2.7 point.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.7B
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• pp.1393-1399
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• 1999

In this paper, a VLSI design of the convolutional codec chip of code rate r=l/3, and constraint length K=9 is presented, which is able to correct errors of the received data when transmitted data is corrupted in channels. The circuit design mainly aimed for simple implementation. In the decoder, Viterbi algorithm with 3-bit soft-decision is employed. For information sequence updating and storage, the register exchange method is employed, where the register length is 5$\times$K(45 stages). The codec chip is designed using VHDL language and Design Analyzer and VHDL Simulator of Synopsys are used for simulation and synthesis. The chip is composed of ENCODER block, ALIGN block, BMC block, ACS block, SEL_MIN block and REG_EXCH block. The operation of the codec chip is verified though the logic simulations, where several error conditions are assumed. As a result of the timing simulation after synthesis, the decoding speed of 325.5Kbps is achieved, and 6,894 gates is used.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.8
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• pp.3076-3082
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• 2010

The techniques and the products which use various video compression codec are come out from army or civil field. In existing high-end PC environment, process of the video compression codec does not become a problem, but in embedded system environments which limited system resources, because the system load due to the high-resolution images compressed by high-density, issues of performance and utilization are highlighted. This paper proposes the DirectShow Filter interfaces which are a hardware method in order to solve the problem existing software algorithms for image compression performance and peripheral interfaces.

• Proceedings of the KIEE Conference
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• 2003.11b
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• pp.295-298
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• 2003

In this paper, a novel maximum a posterion (MAP) estimation for the channel decoding of H.264 codes in the presence of transmission error is presented. Arithmetic codes with a forbidden symbol and trellis search techniques are employed in order to estimate the best transmitted. And, there has been growing interest of communication, the research about transmission of exact data is increasing. Unlike the case of voice transmission, noise has a fatal effect on the image transmission. The reason is that video coding standards have used the variable length coding. So, only one bit error affects the all video data compressed before resynchronization. For reasons of that, channel needs the channel codec, which is robust to channel error. But, usual channel decoder corrects the error only by channel error probability. So, designing source codec and channel codec, Instead of separating them, it is tried to combine them jointly. And many researches used the information of source redundancy In received data. But, these methods do not match to the video coding standards, because video ceding standards use not only one symbol but also many symbols in same data sequence. In this thesis, We try to design combined source-channel codec that is compatible with video coding standards. This MAP decoder is proposed by adding semantic structure and semantic constraint of video coding standards to the method using redundancy of the MAP decoders proposed previously. Then, We get the better performance than usual channel coder's.

• The Journal of the Acoustical Society of Korea
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• v.32 no.2
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• pp.157-166
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• 2013

This paper proposes a non-intrusive speech quality estimation method considering the effects of packet loss to perceptual quality. Packet loss is a major reason of quality degradation in a packet based speech communications network, whose effects are different according to the input speech characteristics or the performance of the embedded packet loss concealment (PLC) algorithm. For the quality estimation system that involves packet loss effects, we first observe the packet loss of G.729 codec which is one of narrowband codec in VoIP system. In order to quantify the lost packet affects, we design a classification algorithm only using speech parameters of G.729 decoder. Then, the degradation values of each class are iteratively selected that maximizes the correlation with the degradation PESQ-LQ scores, and total quality degradation is modeled by the weighted sum. From analyzing the correlation measures, we obtained correlation values of 0.8950 for the intrusive model and 0.8911 for the non-intrusive method.

• The Journal of the Acoustical Society of Korea
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• v.36 no.1
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• pp.70-77
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• 2017

Bandwidth extension is a technique to improve speech quality, intelligibility and naturalness, extending from the 300 ~ 3,400 Hz narrowband speech to the 50 ~ 7,000 Hz wideband speech. In this paper, an Artificial Bandwidth Extension (ABE) module embedded in the Opus audio decoder is designed using the information of narrowband speech to reduce the computational complexity of LPC (Linear Prediction Coding) and LSF (Line Spectral Frequencies) analysis and the algorithm delay of the ABE module. We proposed a spectral envelope extension method using DBN (Deep Belief Network), one of deep learning techniques, and the proposed scheme produces better extended spectrum than the traditional codebook mapping method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.2
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• pp.386-392
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• 2015

Data in a image compression codec are processed with a specific regular block size. The processing order of block sized data is changed in specific function blocks and the data is packed in memory and read by a new sequence. To maintain a regular throughput rate, double buffering is normally used that interleaving two block sized memory to do concurrent read and write operations. Single buffering using only one block sized memory can be adopted to the simple data reordering, but when a complicate reordering occurs, irregular address changes prohibit from implementing adequate address generating for single buffering. This paper shows that there is a predictable and recurring regularity of changing address access orders within a finite updating counts and suggests an effective method to implement. The data reordering function using suggested idea is designed with HDL and implemented with TSMC 0.18 CMOS process library. In various scan blocks, it shows more than 40% size reduction compared with a conventional method.

• ETRI Journal
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• v.31 no.4
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• pp.408-418
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• 2009

Adopting an encryption function in voice over Wi-Fi service incurs problems such as additional power consumption and degradation of communication quality. To overcome these problems, a partial encryption (PE) algorithm for compressed speech was recently introduced. However, from the security point of view, the partial encryption sets (PESs) of the conventional PE algorithm still have much room for improvement. This paper proposes a new selection method for finding a smaller PES while maintaining the security level of encrypted speech. The proposed PES selection method employs the perceptual evaluation of the speech quality (PESQ) algorithm to objectively measure the distortion of speech. The proposed method is applied to the ITU-T G.729 speech codec, and content protection capability is verified by a range of tests and a reconstruction attack. The experimental results show that encrypting only 20% of the compressed bitstream is sufficient to effectively hide the entire content of speech.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.4
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• pp.891-898
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• 2014

The DVC (Distributed Video Coding) provides a theoretical basis for the implementation of light video encoder. Conventionally, lots of studies have been focused on the codec scheme of Stanford University that has a feedback channel to control the bit rate finely. However, the codec scheme can not evaluate the qualities of the frames reconstructed by the received parity bits at the decoder side. This paper presents an efficient method of estimating distortion by correcting the virtual channel noises in side information and then facilitating the measurements of the visual qualities. Through several simulations, it is shown that the proposed method is very efficient in estimating the visual qualities of the reconstructed WZ frames.

• Proceedings of the KIEE Conference
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• 2000.07d
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• pp.3166-3168
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• 2000

This paper proposed variable threshold dual rate ADPCM coding method which is modified from the standard ADPCM of ITU G.726 for speech quality improvement. The speech quality of variable threshold dual rate ADPCM is better than single rate ADPCM at noisy environment without increasing the complexity by using ZCR(Zero Crossing Rate). In this case, ZCR is used to divide input signal samples into two categories(noisy & speech). The samples with higher ZCR is categorized as the noisy region and the samples with lower ZCR is categorized as the speech region. Noisy region uses higher threshold value to be compressed by 16Kbps for reduced bit rates and the speech region uses lower threshold value to be compressed by 40Kbps for improved speech quality. Comparing with the conventional ADPCM, which adapts the fixed coding rate. the proposed variable threshold dual rate ADPCM coding method improves noise character without increasing the bit rate. For real time applications, ZCR calculation was considered as a simple method to obtain the background noise information for preprocess of speech analysis such as FFT and the experiment showed that the simple calculation of ZCR can be used without complexity increase. Dual rate ADPCM can decrease the amount of transferred data efficiently without increasing complexity nor reducing speech quality. Therefore result of this paper can be applied for real-time speech application such as the internet phone or VoIP.