• 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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• 2008.06a
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• pp.807-808
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• 2008

In H.264, sub-pel motion estimation (ME) has strong effect when coding video sequences. 1/4-pel performs better at low bitrate while 1/8-pel gives better results at high bitrate. In this paper, a variable sub-pixel motion vector resolution based on block mode motion estimation method is proposed. Experiment results show that the proposed method can take the advantage of 1/4-pel at low bitrate and 1/8-pel at high bitrate. In addition to that, time is reduced from 14% to 53% compared to KTA1.3 with 1/8-pel motion vector (MV) resolution.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.8
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• pp.1091-1098
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• 1995

This paper proposes an efficient noiseless encoding method of vector quantization(VQ) index using a dynamic index mapping. Using high interblock correlation, the proposed index mapper transforms an index into a new one with lower entropy. In order to achieve good performance with low computational complexity, we adopt 'the sum of differences in pixel values on the block boundaries' as the cost function for index mapping. Simulation results show that the proposed scheme reduces the average bitrate by 40 - 50 % in ordinary VQ system for image compression. In addition, it is shown that the proposed index mapping method can be also applied to mean-residual VQ system, which allows the reduction of bitrate for VQ index by 20 - 30 %(10 - 20 % reduction in total bitrate). Since the proposed scheme is one for noiseless encoding of VQ index, it provides the same quality of the reconstructed image as the conventional VQ system.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.7
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• pp.1-5
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• 1998

In this paper, recovery method with high-bitrate and low-bitrate coder was implemented in order to recover consecutive packet loss over the Internet. LPC was used as redundant audio data for recover of lost packets and RTP parcket format was modified for accommodation of redundant data. In measuring results using random packet loss rate with three redundant datra in every packet, it has shown that recovery rate was 80% in los rate of 50%. Since the processing delay for recovery of the lost packet was 200ms, this recovery method can be applied to real-time Internet sevice such as Internet phone.

• Journal of Korea Multimedia Society
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• v.7 no.9
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• pp.1246-1254
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• 2004

This paper presents a image coding method that is well adaptive to close-range video transmission because of its low titrate and simple coding procedure. At first, it reduces temporal redundancies by performing image DPCM between previous frame and current frame, and makes wavelet transformed image of this difference image. Then, the coefficients are quantized selectively by using the coefficient values of base level and mid-frequency level because inter-level redundancies are widely exists in multi-resolution images. Finally quantized coefficients are made iron the function that implies the target bitrate, the average coefficient energy, and the value of the level. The proposed method shows the effective Performance in the experiments using the continuous motion images and transition images.

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.328-338
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• 2005

Technology for compressing low-bitrate multichannel audio coding is being standardized owing to the increasing need of consumer for multichannel audio contents. In this paper we propose the sound source location cue coding (SSLCC) for extremely compressing multichannel audio to be suitable at the narrow bandwidth transmission environment. To improve the compression capability of the conventional binaural cue coding(BCC), the SSLCC adopts the virtual source location information (VSLI) as a spatial cue parameter, a symmetric uniform quantizer, and Huffman coder. The objective and subjective assessment results show that the SSLCC provides lower bitrate and better audio quality than conventional BCC method.

• IEIE Transactions on Smart Processing and Computing
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• v.5 no.5
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• pp.303-309
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• 2016

This paper presents a channel-adaptive rate control algorithm for low delay video coding. The main goal of the proposed method is to adaptively use the unknown available channel bandwidth while reducing the end-to-end delay between encoder and decoder. The key idea of the proposed algorithm is for the status of the encoder buffer to indirectly reflect the mismatch between the available channel bandwidth and the generated bitrate. Hence, the proposed method fully utilizes the unknown available channel bandwidth by monitoring the encoder buffer status. Simulation results show that although the target bitrate mismatches the available channel bandwidth, the encoder efficiently adapts the given available bandwidth to improve the peak signal-to-noise ratio.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.218-223
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• 2008

In this paper, we implement several methods to improve an audio collaboration system over IP networks, and then evaluate the performance of the implemented methods. In general, speech and audio quality degrades depending on the characteristics of IP networks such as jitter and packet loss. In order to reduce this quality degradation, we propose a lower bit rate audio delivery scheme using the MPEG-2 AAC (Advanced Audio Coding) audio codec in a viewpoint that a packet loss rate could be reduced by a smaller packet size. In addition, iLBC (Internet Low-Bitrate Codec) and the G.711 packet loss concealment algorithm defined by IEFT and ITU-T, respectively, are applied to a audio collaboration system. RAT (Robust-Audio Tool)[7] is used as a baseline platform for the implementation of the proposed methods. It is shown from the implementation that the implemented MPEG-2 AAC audio codec with a bitrate of 256 kbit/s performs as similar as the uncompressed audio quality with a bitrate of 512 kbit/s, and that iLBC and the G.711 packet loss concealment algorithm can improve speech quality when a packet loss rate is 2~10%.

• Journal of Broadcast Engineering
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• v.23 no.4
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• pp.549-558
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• 2018

We designed and implemented streaming server based on MEPG DASH, which is able to provide high quality video with low-latency live streaming service like 360 VR video on the existing cable network via low-spec media service devices such as IPTV and OTT(Over the Top) SettopBox. We also designed and applied management process which is cable of supporting services by cashing streaming video file(MPD, Segment Files) to reduce the server response delay time. Further more, we confimred that it is also able to provide high quality of tiled video streaming with over 50,000kbps bitrate and 8K@60P through the experiment.

• Journal of Korea Society of Industrial Information Systems
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• v.12 no.1
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• pp.54-60
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• 2007

The Digital Multimedia Broadcasting(DMB) system is developed to offer high quality multimedia content to the mobile environment. The system adopts the MPEG-4 standard for the main video, audio and other media format. For providing interactive contents, it also adopts the MPEG-4 scene description that refers to the spatio-temporal specifications and behaviors of individual objects. With more interactive contents, the scene description also needs higher bitrate. However, the bandwidth for allocating meta data, such as scene description is restrictive in the mobile environment. On one hand, the DMB terminal renders each media stream according to the scene description. Thus the binary format for scene(BIFS) stream corresponding to the scene description should be decoded and parsed in advance when presenting media data. With this reasoning, the transmission delay of the BIFS stream would cause the delay in transmitting whole audio-visual scene presentations, although the audio or video streams are encoded in very low bitrate. This paper presents the effective optimization technique in adapting the BIFS stream into the expected bitrate without any waste in bandwidth and avoiding transmission delays inthe initial scene description for interactive DMB content.