• Journal of Information Processing Systems
• /
• v.19 no.6
• /
• pp.817-829
• /
• 2023

Variable wireless channel is a big challenge for real-time video applications, and the rate adaptation of realtime video streaming becomes a hot topic. Intra-video coding is important for high-quality video communication and industrial video applications. In this paper, we proposed a novel adaptive scheme for real-time video transmission with intra-only coding over a wireless network. The key idea of this scheme is to estimate the instantaneous remaining capacity of the network to adjust the quality of the next several video frames, which not only can keep low queuing delay and ensure video quality, but also can respond to bandwidth changes quickly. We compare our scheme with three different schemes in the video transmission system. The experimental results show that our scheme has higher bandwidth utilization and faster bandwidth change response, while maintaining low queuing delay.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.43 no.3 s.309
• /
• pp.54-60
• /
• 2006

This paper presents an effective algorithm for transcoding between the Enhanced Variable Rate Codec(EVRC) and G.729A. The simplest way to communicate between heterogeneous speech networks is the cascade connection of two different codecs, called tandem coding. However, tandem coding not only produces high computational loads, but also makes long delay, These problems can be solved by using the transcoding algorithm. The proposed algorithm consists of LSP (Line Spectral Pair) conversion, pitch delay conversion and algorithm for reduction of delay. Experimental results show the proposed algorithm produces lower computational complexity, shorter algorithm delay, and similar speech quality when compared with the tandem algorithm.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1994.06c
• /
• pp.210-213
• /
• 1994

In this paper, we present a combination of trellis coded vector quantization and code-excited linear prediction coding, termed trellis excitation coding, for an efficient 4.8 kbps speech coding system. A training sequence-based algorithm is developed for designing an otimized codebook subject to the TEC structure. Also, we discuss the trellis symbol release rules that avoid excessive encoding delay. Finally, simulation results for the TEC coder are given at bit rate of 4.8 kbps.

• Journal of Broadcast Engineering
• /
• v.16 no.4
• /
• pp.624-634
• /
• 2011

Recently, the JCT-VC is developing the next generation video coding standard that is called HEVC. HEVC has adopted many coding technologies increasing coding efficiency. For chroma interpolation, DCT-based interpolation filter showing better performance than the linear filter in H.264/AVC was adopted in HEVC. In this paper, a combined filter that utilizes the FIR filter and the linear filter in H.264/AVC is proposed to increase coding efficiency. When the proposed method is compared with DCT-based interpolation filter, the experimental results for various sequences show that the average BD-rate improvements on chroma U and V components are 0.9% and 1.1%, respectively, in the high efficiency case of random access structure, those on U and V components are 1.1% and 1.1%, respectively, in the low complexity case of random access structure, those on U and V components are 0.9% and 1.4%, respectively, in the high efficiency case of low delay structure, and those on U and V components are 1.8% and 1.8%, respectively, in the low complexity case of low delay structure.

• ETRI Journal
• /
• v.39 no.5
• /
• pp.672-682
• /
• 2017

Video compression exploits statistical, spatial, and temporal redundancy, as well as transform and quantization. In particular, the transform in a frequency domain plays a major role in energy compaction of spatial domain data into frequency domain data. The high efficient video coding standard uses the type-II discrete cosine transform (DCT-II) and type-VII discrete sine transform (DST-VII) to improve the coding efficiency of residual data. However, the DST-VII is applied only to the Intra $4{\times}4$ residual block because it yields relatively small gains in the larger block than in the $4{\times}4$ block. In this study, after rearranging the data of the residual block, we apply the DST-VII to the inter-residual block to achieve coding gain. The rearrangement of the residual block data is similar to the arrangement of the basis vector with a the lowest frequency component of the DST-VII. Experimental results show that the proposed method reduces the luma-chroma (Cb+Cr) BD rates by approximately 0.23% to 0.22%, 0.44% to 0.58%, and 0.46% to 0.65% for the random access, low delay B, and low delay P configurations, respectively.

• The Journal of the Acoustical Society of Korea
• /
• v.18 no.6
• /
• pp.71-76
• /
• 1999

In digital cellular communication systems, tandem coding occurs in communications between mobile phones with different speech codecs, resulting in poor voice quality, high computational load, and long transmission delay. In this paper, voice packet conversion technique is proposed to solve the tandem coding problems, and packet conversion algorithm from 13kbps QCELP to 8kbps QCELP is developed. Simulations using various speech data show that the proposed packet conversion method produces voice quality which is equivalent to that by the conventional tandem coding method with shorter transmission delay using about 33% computational load.

• The Journal of the Acoustical Society of Korea
• /
• v.26 no.6
• /
• pp.243-249
• /
• 2007

We propose an effective voice-based gender identification using group delay(GD) Generally, features for speech recognition are composed of magnitude information rather than phase information. In our approach, we address a difference between male and female for GD which is a derivative of the Fourier transform phase. Also, we propose a novel way to incorporate the features fusion scheme based on a combination of GD and magnitude information such as mel-frequency cepstral coefficients(MFCC), linear predictive coding (LPC) coefficients, reflection coefficients and formant. The experimental results indicate that GD is effective in discriminating gender and the performance is significantly improved when the proposed feature fusion technique is applied.

• Journal of electromagnetic engineering and science
• /
• v.9 no.3
• /
• pp.124-129
• /
• 2009

Although the maximum sum-rate capacity of multiple-input multiple output(MIMO) broadcast channels(BCs) can be achieved by dirty-paper coding(DPC), the results were obtained without fairness considerations in uncorrelated MIMO channels. In this paper, we propose new multiuser scheduling algorithms, which find a best user set for approaching the maximum sum-rate capacity while maintaining fairness among users. We analyze the performance of the proposed algorithms using zero-forcing dirty paper coding(ZF-DPC) in the correlated MIMO BCs for throughput and delay fairness, respectively. Numerical results demonstrate that a large time window can reduce the average throughput difference between users, but it increases head-of-line(HOL) delay jitters in the case of delay fairness.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
• /
• v.16 no.2
• /
• pp.69-76
• /
• 2023

A high-quality audio/voice codec using the MDCT/IMDCT process can perfectly restore the current frame through an overlap-add process with the previous frame. In the overlap-add process, an algorithm delay equal to the frame length occurs. In this paper, we propose a MDCT/IMDCT process that reduces algorithm delay by using a variable phase shift in MDCT/IMDCT process. In this paper, a low-delay audio/speech codec was proposed by applying the low delay MDCT/IMDCT algorithm to the ITU-T standard codec G.729.1 codec. The algorithm delay in the MDCT/IMDCT process can be reduced from 20 ms to 1.25 ms. The performance of the decoded output signal of the audio/speech codec to which low-delay MDCT/IMDCT is applied is evaluated through the PESQ test, which is an objective quality test method. Despite of the reduction in transmission delay, it was confirmed that there is no difference in sound quality from the conventional method.

• Journal of Broadcast Engineering
• /
• v.22 no.3
• /
• pp.321-326
• /
• 2017

High Efficiency Video Coding (HEVC) adopted the Discrete Cosine Transform-II (DCT-II) based interpolation filter to improve coding efficiency in motion compensation and estimation. In HEVC, the interpolation filters based on the DCT-II are composed of 8-point for half-pixel and 7-point for 1/4-pixel and 3/4-pixel. In this paper, a DST-VII based interpolation filter is used improve motion compensation and estimation. The experimental results which applied the DST-VII interpolation filter are presented. They show the 0.45% of average bitrate reduction in Random Access configuration and 0.5% of average bitrate reduction in Low Delay B configuration, respectively.