• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.4
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• pp.484-490
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• 1990

In speech coding scheme, because information bits have different error sensitivities over channel errors, the channel coder for combining with speech coding should be realized by the variable coding rate considering the bit importance of speech information bits. In realizing the 4 kbps channel coder for 12kbps speech, this paper have chosen the channel coding method by analyzing the hard-decision post-decoding error rate of RCPC(Rate Compatible Punctured Convolutional) codes and bit error sensitivity of 12 kbps speech. Under the coherent QPSK and Rayleigh fading channel, the performance analysis has showed that 10dB gain was obtained in speech SEGSNR by 4-level uneuqal error protection, which was compared with the caseof no channel coding at 7dB channel SNR.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.3
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• pp.101-109
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• 2004

In this paper, we propose an efficient algerian for wavelet-based sti image coding method that utilizes the rate-distortion (R-D) theory. Since conventional tree-structured image coding schemes do not consider the rate-distortion theory properly, they show reduced coding performance. In this paper, we apply an rate-distortion optimized embedding (RDE) operation into the set partitioning in hierarchical trees (SPIHT) algorithm. In this algorithm, we use the rate-distortion slope as a criterion for the coding order of wavelet coefficients in SPIHT lists. We also describe modified set partitioning and rate-distortion optimized list scan methods. Experimental results demonstrate that the proposed method outperforms the SPIHT algorithm and the rate-distortion optimized embedding algerian with respect to the PSNR (peak signal-to-noise ratio) performance.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.92-97
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• 2000

An AMR(Adaptive Multi-Rate) speech coding algorithm has been adopted as a standard speech codec for IMT-2000. It is based on the algebraic CELP, and consists of eight speech coding modes having the bit rate from 4.75 kbit/s to 12.2 kbit/s. It also contains the VAD(Voice Activity Detector), SCR (Source Controlled Rate) operation, and error concealment scheme for robustness in a radio channel. The bit rate of AMR is changed on a frame basis depending on the channel condition. In this paper, we introduced AMR speech coding algorithm and performed the real-time implementation using TMS320C6201, i.e., a Texas Instrument's fixed-point DSP. With the ANSI C source code released from ETSI and 3GPP, we convert and optimize the program to make it run in real time using the C compiler and assembly language. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.1
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• pp.33-46
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• 1996

Inthis paper, we investigate the compatible coding technique, which receives much interest ever since the introduction of HDTV. First, attempts have been made to analyze the theoretical transform coding gains for various hierarchical decomposition techniques, namely subband, pyramid and DCT-based decomposition techniques. It is shown that the spatical domain techniques proide higher transform coding gains than the DCT-based coding technique. Secondly, we compare the performance of these spatial domain techniques, in terms of the PSNR versus various rate allocations to each layer. Based on these analyses, it is believed that the pyramid decomposition is more appropriate for the compatible coding. Also in this paper, we propose a hybrid prediction pyramid coding technique, by combining the spatio-temporal prediction in MPEG-2[3] and the adaptive MC(Motion Compensation)[1]. In the proposed coding technigue, we also employ an adaptive DCT coefficient scanning technique to exploit the direction information of the 2nd-layer signal. Through computer simulations, the proposed hybrid prediction with adaptive scanning technuque shows the PSNR improvement, by about 0.46-1.78dB at low 1st-layer rate(about 0.1bpp) over the adaptive MC[1], and by about 0.33-0.63dB at high 1st-layer rate (about 0.32-0.43bpp) over the spatio-temporal prediction[3].

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6C
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• pp.842-847
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• 2004

We present a new video coding technique to tradeoff frame rate and picture quality for low bit rate video coding. We show a model equation for selecting the optimal frame rate from the motion content of the source video. We can determine DCT quantization parameter (QP) using the frame rate and bit rate. For objective video quality measurement we propose a simple and effective error measure for skipped frames. The proposed method enhances the video quality up to 2 ㏈ over the H.263 TMN5 encoder.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2019.11a
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• pp.180-181
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• 2019

This paper proposes a rate control algorithm for intra coding frame in HEVC encoder using a deep learning approach. The proposed algorithm is designed for CTU level bit allocation in intra frame by considering visual features spatially and temporally. Our features are generated using visual geometry group (VGG-16) with deep convolutional layers, then it is used for bit allocation per each CTU within an intra frame. According to our experiments, the proposed algorithm can achieve -2.04% Luma component BD-rate gain with minimal bit accuracy loss against the HM-16.20 rate control model.

• ETRI Journal
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• v.29 no.1
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• pp.103-106
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• 2007

Pilot-based coding (PBC), which is used for lossless bit rate reduction of audio coding, has been recently proposed for MPEG Surround. We propose extended PBC for further lossless bit rate reduction of MPEG Surround. Extended PBC selects the number of pilots depending on the parameter band number and the type of spatial parameter. It then encodes the pilots and the relevant difference data. Experiments show that extended PBC is more effective than the original PBC, especially for high bit rate modes, with a negligible complexity increase on the decoder side.

• Journal of IKEEE
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• v.2 no.2 s.3
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• pp.233-238
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• 1998

In this paper, the implementation of dual rate ADPCM using G.726 16Kbps and 40Kbps speech codec algorithm is handled. For small signals, the low rate 16Kbps coding algorithm shows almost the same SNR as the high rate 40Kbps coding algorithm , while the high rate 40Kbps coding algorithm shows the higher SNR than the low rate 16Kbps coding algorithm fur large signal. To obtain the good trade-off between the data rate and synthesized speech quality, we applied low rate 16Kbps for the small signal and high rate 40Kbps for the large signal. Various threshold values determining the rate are applied for good trade-off between data rate and speech quality. The simulation result shows the good speech quality at a low rate comparing with 16Kbps & 40Kbps.

• The Transactions of the Korea Information Processing Society
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• v.4 no.10
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• pp.2629-2639
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• 1997

The video coding for very low bit-rate has recently received considerable attention, but the conventional coding schemes with block based transform suffer from the blocky effect for the constraints of limited bit-rate. In this paper, we present a video coding system based on wavelet transform and multiresolution motion estimation/compensation for very low bit-rate video. The proposed scheme uses the wavelet transform which is flexible to represent non-stationary image signals and adaptable to the human visual characteristics. The wavelet transformed coefficients are coded by various coding modes in accordance with the sum of absolute error after motion estimation/compensation in wavelet decomposed domain. And simple buffer control technique is applied to handle constant image quality. It is shown that the presented scheme has more acceptable image quality without blocky effects than conventional block based transform video coding.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.5
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• pp.417-425
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• 2003

H.264, which is the latest video coding standard of both ITU-T(International Telecommunication Union-Telecommunication standardization sector) and MPEG(Moving Picture Experts Group), adopts new video coding tools such as variable block size motion estimation, multiple reference frames, quarter-pel motion estimation/compensation(ME/MC), 4${\times}$4 Integer DCT(Discrete Cosine Transform), and Rate-Distortion Optimization, etc. These new video coding tools provide good coding of efficiency compared with existing video coding standards as H.263, MPEG-4, etc. However, these new coding tools require the increase of encoder complexity. Therefore, in order to apply H.264 to many real applications, fast algorithms are required for H.264 coding tools. In this paper, when encoder MacroBlock(MB) mode is decided by rate-distortion optimization tool, fast mode decision algorithm by skipping variable block size ME/MC and spatial-predictive coding, which occupies most encoder complexity, is proposed. In terms of computational complexity, the proposed method runs about 4 times as far as JM(Joint Model) 42 encoder of H.264, while the PSNR(peak signal-to-noise ratio)s of the decoded images are maintained.