Unequal error protection for MPEG-2 video transmission over wireless channels

This paper proposes an unequal error protection (UEP) method for MPEG-2 video transmission. Since the source and channel coders are normally concatenated, if the channel is noisy, more bits are allocated to channel coding and fewer to source coding. The situation is reversed when the channel conditions are more benign. Most of the joint source channel coding (JSCC) methods assume that the video source is subband coded, the bit error sensitivity of the source code can be modeled, and the bit allocations for different subband channels will be calculated. The UEP applied to different subbands is the rate compatible punctured convolution channel coder. However, the MPEG-2 coding is not a subband coding, the bit error sensitivity function for the coded video can no longer be applied. Here, we develop a different method to find the rate-distortion functions for JSCC of the MPEG-2 video. In the experiments, we show that the end-to-end distortion of our UEP method is smaller than the equal error protection method for the same total bit-rate.     

Requantization for transcoding of MPEG-2 intraframes

An investigation on requantization for transcoding of video signals is carried out. Specifically, MPEG-2 compatible discrete cosine transform (DCT) intraframe coding is addressed. The aim of this work is twofold: first, to provide a theoretical analysis of the transcoding problem, and second, to derive quantization methods for efficient transcoding based on the results of the analysis. The mean squared error (MSE) cost function is proposed for designing a quantizer with minimum distortion resulting in up to 1.3 dB gain compared with the quantizer used in the MPEG-2 reference coder TM5. However, the MSE quantizer leads in general to a larger bit rate and may therefore only be applied locally to blocks of sensitive image content. A better rate-distortion performance can be provided by the maximum a posteriori (MAP) cost function. In critical cases, the MAP quantizer gives a 0.4 dB larger signal-to-noise-ratio (SNR) at the same bit rate compared with the TM5 quantizer. The results are not limited to MPEG-2 and can be adapted to other coding schemes such as H.263 or JPEG.     

Speech and channel coding for digital land-mobile radio

The joint development of a medium bit-rate speech coder along with an effective channel coding technique to provide a robust, spectrally efficient, high-quality mobile communication system is described. A subband coder operating at 12 kb/s is used; in the absence of channel errors, it provides speech quality comparable to current analog land-mobile radio systems. The coder design incorporates a unique coding of the side information to facilitate the use of forward-error-correction coding without the need to code the entire bit stream. The use of excessive overhead for redundancy is avoided while the harsh effects of frequent channels are mitigated. These techniques have been used in an experimental FDMA (frequency-division multiple access) digital land-mobile radio system. The combined speech and channel coder operates at 15 kb/s and provides intelligible speech at fading channel error rates up to 8%     

Le codeur mpeg-2 AAC expliqué aux traiteurs de signaux

The MPEG- 2 Advanced Audio Coder is the latest issue of the MPEG audio encoders/decoders family, whose most popular version is known as MP3. It gathers many of the latest highly efficient sound compression techniques in a quite classically structured coder. The main part is based on a Discrete Cosine Transform with variable resolution. The output from this filterbank is compressed by the combination of an adaptive bit allocation module, according to frequency subbands, and a set of noiseless Huffman codebooks. Bit allocation is controlled by a psychoa-coustic model which determines an audibility threshold for signal distortion in the frequency domain. This article intends to explain the ISO standard without replacing it, and also to be a general introduction to perceptual audio coding.     

Image coding with robust channel-optimized trellis-codedquantization

This paper presents a wavelet-based image coder that is optimized for transmission over the binary symmetric channel (BSC). The proposed coder uses a robust channel-optimized trellis-coded quantization (COTCQ) stage that is designed to optimize the image coding based on the channel characteristics. A phase scrambling stage is also used to further increase the coding performance and robustness to nonstationary signals and channels. The resilience to channel errors is obtained by optimizing the coder performance only at the level of the source encoder with no explicit channel coding for error protection. For the considered TCQ trellis structure, a general expression is derived for the transition probability matrix. In terms of the TCQ encoding rat and the channel bit error rate, and is used to design the COTCQ stage of the image coder. The robust nature of the coder also increases the security level of the encoded bit stream and provides a much more visually pleasing rendition of the decoded image. Examples are presented to illustrate the performance of the proposed robust image coder     

Forward error control for MPEG-2 video transport in a wireless ATM LAN

The possibility of providing multimedia services to mobile users has led to interest in designing broadband wireless networks that can guarantee quality of service for traffic flows. However, a fundamental problem in these networks is that severe losses may occur due to the random fading characteristics of the wireless channel. Error control algorithms which compensate for these losses are required in order to achieve reasonable loss rates. In this paper, the performance of error control based on forward error correction (FEC) for MPEG-2 video transmission in an indoor wireless ATM LAN is studied. A random bit error model and a multipath fading model are used to investigate the effect of errors on video transport. Combined source and channel coding techniques that employ single-layer and scalable MPEG-2 coding to combat channel errors are compared. Simulation results indicate that FEC-based error control in combination with 2-layer video coding techniques can lead to acceptable quality for indoor wireless ATM video.Work performed while the author was at AT&T Bell Laboratories on a D.O.E. fellowship program.     

Joint video coding of MPEG-2 video programs for digitalbroadcasting services

In digital broadcasting services such as digital satellite TV, cable TV, and digital terrestrial TV, several video programs are compressed by MPEG-2 and then simultaneously transmitted over a conventional CBR (constant bit rate) broadcasting channel. In this environment, the picture quality of the aggregated video programs should be kept as equal as possible to provide fair video services for all the viewers. We propose a joint video coding scheme for ensuring that the picture quality of all the programs are nearly the same. This objective is achieved by simultaneously controlling the video encoders to generate the VBR (variable bit rate) compressed video streams. This paper contributes in two ways. First, we propose a rate-distortion estimation method for MPEG-2 video, which enables us to predict the amount of bits and the distortion generated from an encoded picture at a given quantization step size and vice versa. The most attractive features of the proposed estimation method are its accuracy and a computational complexity low enough to be applied to real-time video coding applications. Second, this paper presents an efficient and accurate joint video coding scheme using the rate-distortion estimation results. The experimental results show that our coding scheme gives a higher and more stable picture quality and a more efficient channel utilization than an independent coding scheme that encodes each program independently     

Progressive transmission of images using MAP detection overchannels with memory

We propose a new maximum a posteriori (MAP) detector, without the need for explicit channel coding, to lessen the impact of communication channel errors on compressed image sources. The MAP detector exploits the spatial correlation in the compressed bitstream as well as the temporal memory in the channel to correct channel errors. We first present a technique for computing the residual redundancy inherent in a compressed grayscale image (compressed using VQ). The performance of the proposed MAP detector is compared to that of a memoryless MAP detector. We also investigate the dependence of the performance on memory characteristics of the Gilbert-Elliott channel as well as average channel error rate. Finally, we study the robustness of the proposed MAP detector's performance to estimation errors.     

Low complexity LMMSE turbo equalization for combined error control coded and linearly precoded OFDM

The turbo equalization approach is studied for Orthogonal Frequency Division Multiplexing （OFDM） system with combined error control coding and linear precoding. While previous literatures employed linear precodcr of small size for complexity reasons, this paper proposes to use a linear precoder of size larger than or equal to the maximum length of the equivalent discrete-time channel in order to achieve full frequency diversity and reduce complexities of the error control coder/decoder. Also a low complexity Linear Minimum Mean Square Error （LMMSE） turbo equalizer is derived for the receiver. Through simulation and performance analysis, it is shown that the performance of the proposed scheme over frequency selective fading channel reaches the matched filter bound; compared with the same coded OFDM without linear precoding, the proposed scheme shows an Signal-to-Noise Ratio （SNR） improvement of at least 6dB at a bit error rate of 10 6 over a multipath channel with exponential power delay profile. Convergence behavior of the proposed scheme with turbo equalization using various type of linear precoder/transformer, various interleaver size and error control coder of various constraint length is also investigated.     

Time-Varying and Frequency-Selective Channel Estimation with Unequally Spaced Pilot Symbols

In this paper, an accurate and computationally efficient algorithm is proposed for estimating time-varying and frequency-selective fading channel with unequally spaced pilot symbols. By employing the time-varying coefficient polynomial interpolation method, it is proved that the time-varying channel impulse response can be estimated by the product of a constant matrix and the fading information at pilot symbol positions. Furthermore, a least square off-line training algorithm is presented to optimally calculate the constant matrix, taking into consideration of the statistics of channel fading and noise. The new algorithm can also be applied for estimating flat fading channel with equally spaced pilot symbols as a special case. Simulation results indicate that our new channel estimation algorithm leads to small mean square error for fading estimation and provides bit error rate performance close to that of the perfect channel estimation.     

Double differential space-time block coding for time-selectivefading channels

Most existing space-time coding schemes assume time-invariant fading channels and offer antenna diversity gains relying on accurate channel estimates at the receiver. Other single differential space-time block coding schemes forego channel estimation but are less effective in rapidly fading environments. Based on a diagonal unitary matrix group, a novel double differential space-time block coding approach is derived in this paper for time-selective fading channels. Without estimating the channels at the receiver, information symbols are recovered with antenna diversity gains regardless of frequency offsets. The resulting transceiver has very low complexity and is applicable to an arbitrary number of transmit and receive antennas. Approximately optimal space-time codes are also designed to minimize bit error rate. System performance is evaluated both analytically and with simulations     

OFDM-based turbo-coded hierarchical and non-hierarchicalterrestrial mobile digital video broadcasting

The feasibility of terrestrial digital video broadcast (DVB) to mobile receivers is studied and turbo coded performance enhancements are proposed. Initially, the MPEG-2 codec is subjected to a rigorous bit error sensitivity investigation, in order to assist in designing various error protection schemes for wireless DVB transmission. The turbo codec is shown to provide signal-to-noise ratio (SNR) performance advantages in excess of 5-6 dB over conventional convolutional coding both in terms of bit error rate and video quality. Our experiments suggested that-despite our expectations-multi-class data partitioning did not result in error resilience improvements, since a high proportion of relatively sensitive video bits had to be relegated to the lower integrity subchannel, when invoking a powerful low-rate channel codec in the high-integrity protection class. Nonetheless, DVB transmission to mobile receivers is feasible, when using turbo-coded OFDM transceivers at realistic power-budget requirements under the investigated highly dispersive fading channel conditions. It is interesting to note furthermore that the 5-6 dB SNR improvement due to turbo coding allows us to invoke for example the double-throughput 16-level quadrature amplitude modulation (16-QAM) mode instead of the standard convolutional-coded 4-QAM mode. This facilitates doubling the bit rate and hence improving the video quality     

一种基于MPEG-4的感兴趣区域视频编码新方法

本文提出了一种基于MPEG-4的感兴趣区域视频编码新方法.通过对感兴趣区边沿宏块采用强制帧内编码及宏块内预测编码时参考位置的自适应选择,该方法有效地抑制了数据噪声的扩散和传播.通过采用自适应宏块大小的运动估计/补偿算法,提高了感兴趣区特别是图像运动复杂的感兴趣区的编码效率及质量.在码率分配部分,本文方法通过计算不同区域的图像复杂性和能量,依据用户可设定的感兴趣权重因子不等重地分配可用码率资源.实验证明,本文方法较大程度改善了感兴趣区视频编码的压缩效率,提高了码率分配地灵活性和有效性.     

比特交织编码调制技术分析

为了在不降低频带利用率和功率利用率的情况下以设备的复杂化来换取编码增益,提出了一种将编码和调制方式结合在一起的比特交织编码调制(Bit Interleaved Coded Modulation,BICM)技术。该技术针对网格编码调制(TrellisCoded Modulation,TCM)灵活性差和衰落信道中性能差的缺点,在理想交织时将编码器和调制器分离设计,具有高度的灵活性,并可以得到近似网格编码调制的性能,并且对衰落信道具有很好的鲁棒性。介绍了比特交织编码调制的构成模式、关键技术,包括信道编码、比特交织和调制,给出了BCIM理论分析,并对不同的交织深度、交织方法和高阶调制等进行了性能仿真分析。仿真结果表明比特交织编码调制可以取得良好的误码率性能。     

An integrated temporal error concealment for H.264/AVC based on spatial evaluation criteria

Owing to error-prone transmission networks, the compressed video bit stream is prone to packet loss in the transmission channel. This loss causes serious distortion and the distortion will propagate to successive frames, especially in highly compressed video coding standard. Therefore, it is very important to efficiently enhance the restored result. In this paper, an integrated temporal error concealment technique for H.264/AVC is proposed. The technique could effectively restore the corrupted data by adaptively integrating error concealment approaches with the adaptive weight-based switching algorithm. The integrated mechanism is based on spatial evaluation criteria, judged by boundary distortion estimation and texture intensity. Experimental results show that the technique could effectively enhance the performance of error concealment.     

单载波频域均衡信道估计分析

针对高速率信息传输系统中多径衰落对单载波频域均衡(SC-FDE)信道估计造成的影响,在SC-FDE原理基础上,将无线通信中信噪比估计和SC-FDE中信道估计结合起来,对现有信噪比估计算法进行优化。分析和仿真结果表明,此方法能有效地改善多径衰落信道中信道估计的有效性,提高了频域均衡的效果,从而改善了SC-FDE系统的误码率(BER)性能。     

Motion-compensated 3-D subband coding of video

This paper describes a video coding system based on motion-compensated three-dimensional (3-D) subband/wavelet coding (MC-3DSBC), which can overcome the limits of both 3-D SBC and MC prediction-based coding. In this new system, spatio-temporal subbands are generated by MC temporal analysis and a spatial wavelet transform, and then encoded by 3-D subband-finite state scalar quantization (3DSB-FSSQ). The rate allocation from the GOP level to each class of subbands is optimized by utilizing the structural property of MC-3DSBC that additive superposition approximately holds for both rate and distortion. The proposed video coding system is applied to several test video clips. Its performance exceeds that of both a known MPEG-1 implementation and a similar subband MC predictive coder while maintaining modest computational complexity and memory size.     

Power allocation for multi-antenna systems with space-time block coding in the presence of estimation error and delayed feedback

Based on imperfect channel state information with channel estimation error at the receiver and delayed feedback at the transmitter, a suboptimal power allocation (PA) scheme to minimize bit error rate (BER) under a power constraint is developed for beamforming multi-antenna systems with space-time block coding. The proposed scheme is based on a so-called compressed signal-to-noise ratio criterion, where a single compressed factor is utilized, and it can be used to generalize some existing schemes by setting the compressed factor to different forms. A closed-form compressed factor is derived to minimize the BER, and the resultant close-form expression of power allocation is attained. This closed-form expression is computational efficient and can obtain the BER performance close to the existing optimal approach which requires numerical search. Simulation results show that the proposed scheme can provide BER lower than the equal power allocation scheme. However, due to the impact of both estimation error and delayed feedback, it has performance degradation when compared to the PA scheme with estimation error or delayed feedback only.