• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.5A
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• pp.726-742
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• 2000

A novel rte control algorithm consisting of two major components, i.e. a variable encoding frame rate method and a hybrid DCT/wavelet I-frame coding scheme, is proposed in this work for low bit rate video coding. Most existing rate control algorithms for low bit rate video focus on bit allocation at the macroblock level under a constant frame rate assumption. The proposed rate control algorithm is able to adjust the encoding frame rate at the expense of tolerable time-delay. Furthermore, an R-D optimized hybrid DCT/wavelet scheme is used for effective I-frame coding. The new rate-control algorithm attempts to achieve a good balance between spatial quality and temporal quality to enhance the overall human perceptual quality at low bit rates. It is demonstrated that the rate control algorithm achieves higher coding efficiency at low bit rates with a low additional computational cost. The variable frame rate method and hybrid I-frame coding scheme are compatible with the bi stream structure of H.263+.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.17 no.6
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• pp.154-159
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• 2003

In this paper, we proposed variable threshold dual rate ADPCM coding method which adapts two coding rates of the standard ADPCM of ITU G.726 for speech quality improvement at a comparably low coding rates. The ZCR(Zero Crossing Rate) is computed for speecd data and under the noisy environment, noise data dominant region showed higher ZCR and speech data dominant region showed lower ZCR. The speech data with the higher ZCR is encoded by low coding rate for reduced coded data and the speech data with the lower ZCR is encoded by high coding rate for speech quality improvements. For coded data, 2 bits are assigned for low coding rate of 16[Kbps] and 5 bits are is assigned for high coding rate of 40[Kbps]. Through the simulation, the proposed idea is evaluated and shown that the variable dual rate ADPCM coding technique shows the qood speech quality at low coding rate.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.8
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• pp.2875-2893
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• 2015

Existing coding aware routing algorithms focused on novel routing metric design that captures the characteristics of network coding. However, in packet coding algorithm, they use opportunistic coding scheme which didn't consider the queue state of the coding node and are equivalent to the conventional store-and-forward method in light traffic load condition because they never delay packets and there are no packets in the output queue of coding node, which results in no coding opportunity. In addition, most of the existing algorithms assume that all flows participating in the network have equal rate. This is unrealistic since multi-rate environments are often appeared. To overcome above problem and expand network coding to light traffic load scenarios, we present an enhanced coding-aware routing algorithm based on queue state and local topology (OQMCAR), which consider the queue state of coding node in packet coding algorithm where the control policy is of threshold-type. OQMCAR is a unified framework to merge single rate case and multiple rate case, including the light traffic load scenarios. Simulations results show that our scheme can achieve higher throughput and lower end-to-end delay than the current mechanisms using COPE-type opportunistic coding policy in different cases.

• ETRI Journal
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• v.31 no.5
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• pp.613-615
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• 2009

We propose a pilot-based coding (PBC) scheme for lossless bit rate reduction of parametric stereo (PS) in enhanced aacPlus. It uses PBC in addition to the existing frequency and time differential coding to encode and decode PS parameter indexes. We also design optimal Huffman codebooks (HCBs) for PBC in the proposed scheme. Experiments show that the proposed scheme is superior to the original coding scheme, where both the new coding structure and the optimal HCBs contribute to the bit rate reduction.

• IEIE Transactions on Smart Processing and Computing
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• v.1 no.3
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• pp.171-181
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• 2012

In the new video coding standard, called high efficiency video coding (HEVC), the coding unit (CU) is adopted as a basic unit of a coded block structure. Therefore, the rate control (RC) methods of H.264/AVC, whose basic unit is a macroblock, cannot be applied directly to HEVC. This paper proposes the largest CU (LCU) level RC method for hierarchical video coding in a HEVC. In the proposed method, the effective bit allocation is performed first based on the hierarchical structure, and the quantization parameters (QP) are then determined using the Cauchy density based rate-quantization (RQ) model. A novel method based on the linear rate model is introduced to estimate the parameters of the Cauchy density based RQ model precisely. The experimental results show that the proposed RC method not only controls the bitrate accurately, but also generates a constant number of bits per second with less degradation of the decoded picture quality than with the fixed QP coding and latest RC method for HEVC.

• Proceedings of the KIPE Conference
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• 2002.07a
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• pp.727-729
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• 2002

In this paper, we proposed quad variable rates ADPCM coding method and its implementation on C6000 DSP, which is modified from the standard ADPCM of ITU G.726 for speech quality improvement considering the environmental noise Four coding rates, 16Kbps, 24Kbps, 32Kbps and 40Kbps are used for speech window samples and the rate decision threshold is decided by the environmental noise level. The object of the proposed method is to reduce the coding rate while retaining the speech quality and the speech quality is considerably close to 40Kbps single rate coder with the coding rate close to 16Kbps single rate coder under the environmental noise. The environmental noise level affects the coding rate and the noise level is calculated per every speech window samples. At high noise level, more samples are coded at higher rates to enhance the quality, but at low noise level, only the big speech signals are coded at higher rates and more speech samples are coded at lower coding rates to reduce the coding rates. The influence of the noise on tile speech signal is considerably high for small signals and the small signal has the higher ZCR (zero crossing rate). The method is simulated in PC and to be implemented on C6000 floating point DSP board in real time operations.

• IEIE Transactions on Smart Processing and Computing
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• v.1 no.3
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• pp.152-160
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• 2012

This paper proposes a frame-level rate control algorithm for low delay video applications to reduce the fluctuations in the bitrate. The proposed algorithm minimizes the bitrate fluctuations in two ways with minimal coding loss. First, the proposed rate control applies R-Q model to all frames including the first frame of every group of pictures (GOP) except for the first one of a sequence. Conventional rate control algorithms do not use any R-Q models for the first frame of each GOP and do not estimate the generated-bit. An unexpected output rate result from the first frame affects the remainder of the pictures in the rate control. Second, a rate-distortion (R-D) cost is calculated regardless of the hierarchical coding structure for low bitrate fluctuations because the hierarchical coding structure controls the output bitrate in rate distortion optimization (RDO) process. The experimental results show that the average variance of per-frame bits with the proposed algorithm can reduce by approximately 33.8% with a delta peak signal-to-noise ratio (PSNR) degradation of 1.4dB for a "low-delay B" coding structure and by approximately 35.7% with a delta-PSNR degradation of 1.3dB for a "low-delay P" coding structure, compared to HM 8.0 rate control.

• Proceedings of the IEEK Conference
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• 2000.06a
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• pp.17-20
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• 2000

This paper addresses a novel algorithm for variable rate channel coding with interleaver punctured convolutional code for wireless communication. In other to increase the coding performance and achieve the variable channel coding rate, serially concatenated convolutional coding scheme will be applied. In this paper, we characterize the effect of interleaver puncturing on the effectiveness of the proposed scheme some simulation results are presented, in which the channel model of additive Gaussian noise is assumed.

• ETRI Journal
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• v.38 no.1
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• pp.185-194
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• 2016

In this paper, a novel rate control scheme for low delay video coding of High Efficiency Video Coding (HEVC) is proposed. The proposed scheme is developed by considering a new temporal prediction structure of HEVC. In the proposed scheme, the relationship between bit rate and quantization step is exploited firstly to formulate an accurate quadratic rate-quantization (R-Q) model. Secondly, a method of determining the quantization parameters (QPs) for the first frames within a group of pictures is proposed. Thirdly, an accurate frame-level bit allocation method is proposed for HEVC. Finally, based on the proposed R-Q model and the target bit allocated for the frame, the QPs are predicted for coding tree units by using rate-distortion (R-D) optimization. We compare our scheme against that of three other state-of-the-art rate control schemes. Experimental results show that the proposed rate control scheme can increase the Bjøntegaard delta peak signal-to-noise ratio by 0.65 dB and 0.09 dB on average compared with the JCTVC-I0094 and JCTVC-M0036 schemes, respectively, both of which have been implemented in an HEVC test model encoder; furthermore, the proposed scheme achieves a similar R-D performance to Wang's scheme, as well as obtaining the smallest bit rate mismatch error of all the schemes.

• Proceedings of the IEEK Conference
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• 2003.11a
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• pp.45-48
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• 2003

In general, speech coding methods are classified into the following three categories: the waveform coding, the source coding and the hybrid coding. Fast speaking is possible to encode with a few information compared with slow speaking rate. In case of speaking rate, low frequency band is more important than high frequency band while listening. Speech vocoding technique is developing to way with low bit rate and complexity and high sound quality. the CELP type of vocoder support very good sound quality with low bit rate but these vocoders don't consider about the speaking rate. When we consider speaking rate and encode the frame depending on the speaking rate, the bit rate is able to reduce the bit rate than the conventional vocoder. We propose the technique to estimate the speaking rate and applied PSOLA technique in case of the frame of slow speaking rate. As a result of simulation bit rate can be reduced about 300 bps.