Mobile Audiovisual Terminal: System Design and Subjective Testing in DECT and UMTS networks

It is anticipated that there will shortly be a requirement for multimedia terminals that operate via mobile communications systems. This paper presents a functional specification for such a terminal operating at 32 kb/s in a digital European cordless telecommunications (DECT) and universal mobile telecommunications system (UMTS) radio network. A terminal has been built, based on a PC with digital signal processor (DSP) boards for audio and video coding and decoding. Speech coding is by a phonetically driven code-excited linear prediction (CELP) speech coder and video coding by a block-oriented hybrid discrete cosine transform (DCT) coder. Separate channel coding is provided for the audio and video data. The paper describes the techniques used for audio and video coding, channel coding, and synchronization. Methods of subjective testing in a DECT network and in a UMTS network are also described. These consisted of subjective tests of first impressions of the mobile audio–visual terminal (MAVT) quality, interactive tests, and the completion of an exit questionnaire. The test results showed that the quality of the audio was sufficiently good for comprehension and the video was sufficiently good for following and repeating simple mechanical tasks. However, the quality of the MAVT was not good enough for general use where high-quality audio and video was needed, especially when transmission was in a noisy radio environment

Fast algorithms and hardware architectures for H.264/AVC

制度:新 ; 文部省報告番号:甲2460号 ; 学位の種類:博士(工学) ; 授与年月日:2007/6/25 ; 早大学位記番号:新456

A modified H.263 algorithm using bit allocation buffer control algorithm

Buffer control is an important problem in very low bitrate video coding. In a recent work [ 111, the authors had proposed a new buffer control algorithm for motion-compensated hybrid DPCMiDCT coding. The algorithm is based on the use of bit allocation algorithm to determine the quantization scale factors in such coder to meet a given target bit rate. Simulation results showed that, using the proposed algorithm, the H.261 coder can achieve a higher PSNR and better visual quality than the coder using traditional buffer control algorithm. In this paper, we apply this buffer control algorithm to a modified version of the H.263 algorithm for very low bit-rate video coding. Comparing the performance of the modified H.263 codec with the TMN5 model also shows that better visual quality can be obtained at comparable PSNR values.published_or_final_versio

Complexity management of H.264/AVC video compression.

The H. 264/AVC video coding standard offers significantly improved compression efficiency and flexibility compared to previous standards. However, the high computational complexity of H. 264/AVC is a problem for codecs running on low-power hand held devices and general purpose computers. This thesis presents new techniques to reduce, control and manage the computational complexity of an H. 264/AVC codec. A new complexity reduction algorithm for H. 264/AVC is developed. This algorithm predicts "skipped" macroblocks prior to motion estimation by estimating a Lagrange ratedistortion cost function. Complexity savings are achieved by not processing the macroblocks that are predicted as "skipped". The Lagrange multiplier is adaptively modelled as a function of the quantisation parameter and video sequence statistics. Simulation results show that this algorithm achieves significant complexity savings with a negligible loss in rate-distortion performance. The complexity reduction algorithm is further developed to achieve complexity-scalable control of the encoding process. The Lagrangian cost estimation is extended to incorporate computational complexity. A target level of complexity is maintained by using a feedback algorithm to update the Lagrange multiplier associated with complexity. Results indicate that scalable complexity control of the encoding process can be achieved whilst maintaining near optimal complexity-rate-distortion performance. A complexity management framework is proposed for maximising the perceptual quality of coded video in a real-time processing-power constrained environment. A real-time frame-level control algorithm and a per-frame complexity control algorithm are combined in order to manage the encoding process such that a high frame rate is maintained without significantly losing frame quality. Subjective evaluations show that the managed complexity approach results in higher perceptual quality compared to a reference encoder that drops frames in computationally constrained situations. These novel algorithms are likely to be useful in implementing real-time H. 264/AVC standard encoders in computationally constrained environments such as low-power mobile devices and general purpose computers

An adaptive multiresolution modification of the H.263 video coding algorithm

An adaptive multiresolution approach for video coding is presented. The algorithm uses the information content to determine the resolution of the video to be encoded. An important advantage of the algorithm is that the codec can maintain a very stable frame rate with reasonable image quality during scene change and provide better quality video when the motion is less rapid. Simulation results show that the modified H.263 coder, using the proposed algorithm, can maintaining better image quality and a more steady frame rate than the TMN 5 algorithm at low bit-rate.published_or_final_versio

Improved quality block-based low bit rate video coding.

The aim of this research is to develop algorithms for enhancing the subjective quality and coding efficiency of standard block-based video coders. In the past few years, numerous video coding standards based on motion-compensated block-transform structure have been established where block-based motion estimation is used for reducing the correlation between consecutive images and block transform is used for coding the resulting motion-compensated residual images. Due to the use of predictive differential coding and variable length coding techniques, the output data rate exhibits extreme fluctuations. A rate control algorithm is devised for achieving a stable output data rate. This rate control algorithm, which is essentially a bit-rate estimation algorithm, is then employed in a bit-allocation algorithm for improving the visual quality of the coded images, based on some prior knowledge of the images. Block-based hybrid coders achieve high compression ratio mainly due to the employment of a motion estimation and compensation stage in the coding process. The conventional bit-allocation strategy for these coders simply assigns the bits required by the motion vectors and the rest to the residual image. However, at very low bit-rates, this bit-allocation strategy is inadequate as the motion vector bits takes up a considerable portion of the total bit-rate. A rate-constrained selection algorithm is presented where an analysis-by-synthesis approach is used for choosing the best motion vectors in term of resulting bit rate and image quality. This selection algorithm is then implemented for mode selection. A simple algorithm based on the above-mentioned bit-rate estimation algorithm is developed for the latter to reduce the computational complexity. For very low bit-rate applications, it is well-known that block-based coders suffer from blocking artifacts. A coding mode is presented for reducing these annoying artifacts by coding a down-sampled version of the residual image with a smaller quantisation step size. Its applications for adaptive source/channel coding and for coding fast changing sequences are examined

Region-Based Transform-Domain Video Scrambling

In this paper, we address the problem of scrambling regions of interest in a video sequence. We target applications such as video surveillance preserving privacy, anonymous video communications, or TV news safeguarding the anonymity of a source. We propose an efficient solution based on transform-domain scrambling. More specifically, the sign of selected transform coefficients is flipped during encoding. We address the two cases of Motion JPEG 2000 and MPEG-4. Simulation results show that it can be successfully applied to conceal information in regions of interest in the scene while providing with a good level of security. Furthermore, the scrambling is flexible and allows adjusting the amount of distortion introduced. Finally, this is achieved with a small impact on coding performance and negligible computational complexity increase

Video Quality Measurement for 3G Handset

The quality of video has become a decisive factor for the consumer of 3G video services to choose his mobile operator. It is, therefore, critical for 3G network operator, equipment provider and service provider to measure and hence maintain the video quality of video services they offer. A project has been proposed in University of Plymouth to develop a test platform to evaluate video quality for 3G handset using Asterisk PBX server. For this purpose, support for 3G-324M protocol and all the audio and video codecs (i.e. H.263 baseline level 10 and MPEG-4 simple profile @ level 0) mandated and recommended by 3G- 324M standard should be added in to Asterisk®. The purpose of this thesis is to identify the correct software implementation of H.263 baseline level 10 and MEPG-4 simple profile @ level 0 video codecs so that they can then be incorporated in to Asterisk®. This is the part of the above mentioned project. Open source FFmpeg-libavcodec is believed to support both MPEG-4 and H.263 codecs. Similarly Telenor H.263 codec is also free to use. This project tests both the capabilities and suitability of the above mentioned software packages/codecs for adding in to Asterisk to perform the required encoding and decoding. Experiments showed that FFmpeg-libavcodec can neither decode nor encode to MPEG-4 simple profile @ level 0. It seems that FFmpeg requires some major modifications in its source code to support MPEG-4 simple profile @ level 0 codec. Although FFmepg can decode and encode to H.263 baseline level 10, but it does not offer a fine control over bitrate while encoding, and reports very high muxing overhead while decoding, H.263 baseline level 10. Telenor H.263 codec can decode and encode to H.263 baseline level 10.without any problem. Telenor H.263 codec is, therefore, more suitable for incorporating in to Asterisk® than FFmpeg for decoding and encoding to H.263 baseline level 10 bitstreams. ISchool of Computing, Communication and Electronic