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Interactive Cellular and Cordless Video Telephony: State-of-the-Art System Design Principles and Expected Performance
Abstract
Second-generation (2G) mobile radio standards have not been designed with video communications in mind, although the employment of error-resilient, constant-bit-rate proprietary video codecs over these systems is realistic. The third-generation (3G) systems are capable of providing higher rates and better communications integrity in support of video applications. This paper advocates the employment of burst-by-burst adaptive transceivers, which are capable of accommodating the time-variant channel quality fluctuation of wireless channels. This paper is concluded with a range of performance figures and system design guidelines for wireless systems.
... In addition, Internet protocol (IP) based architecture for 3G wireless systems promises to provide nextgeneration wireless services such as voice, high-speed data, Internet access, audio and video streaming on an all IP network [2]. However, wireless video [3,4,5] has bandwidth, delay, and loss requirements, many existing mobile networks cannot provide a guaranteed quality of service because temporally high bit error rates are unavoidable during fading periods. ...
... The evolving next-generation video compression standard certainly will be based on wavelets. Therefore the research on networked wavelets-based compressed video data becomes very important [3,4,5]. ...
... In addressing a topic as broad as wireless video [3,4,5], which could encompass source coding, channel coding, modulation, channel equalization, and multiuser detection, ideally an end-to-end approach should be taken that considers all components together. In practice, however, depending on their background, researchers typically focus on one or two areas (e.g., error-resilient video coding [53], channel coding [54], joint source-channel coding (JSCC) [20].) ...
Thesis (Ph. D.)--University of Aston in Birmingham, 1995.
... In addition, Internet protocol (IP) based architecture for 3G wireless systems promises to provide nextgeneration wireless services such as voice, high-speed data, Internet access, audio and video streaming on an all IP network [2]. However, wireless video [3, 4, 5] has bandwidth, delay, and loss requirements, many existing mobile networks cannot provide a guaranteed quality of service because temporally high bit error rates are unavoidable during fading periods. Video sequence requires huge amount of data to process whereas the network bandwidth and the capacity of storage media are limited, thus video compression plays a central role in modern multimedia communications. ...
... The evolving next-generation video compression standard certainly will be based on wavelets. Therefore the research on networked wavelets-based compressed video data becomes very important [3, 4, 5]. ...
... In addressing a topic as broad as wireless video [3, 4, 5], which could encompass source coding, channel coding, modulation, channel equalization, and multiuser detection , ideally an end-to-end approach should be taken that considers all components together. In practice, however, depending on their background, researchers typically focus on one or two areas (e.g., error-resilient video coding [53], channel coding [54], joint source-channel coding (JSCC) [20].) ...
Compressed video bitstream transmissions over wireless networks are addressed in this work. We first consider error control and power allocation for transmitting wireless video over CDMA networks in conjunction with multiuser detection. We map a layered video bitstream to several CDMA fading channels and inject multiple source/parity layers into each of these channels at the transmitter. We formulate a combined optimization problem and give the optimal joint rate and power allocation for each of linear minimum mean-square error (MMSE) multiuser detector in the uplink and two types of blind linear MMSE detectors, i.e., the direct-matrix-inversion (DMI) blind detector and the subspace blind detector, in the downlink. We then present a multiple-channel video transmission scheme in wireless CDMA networks over multipath fading channels. For a given budget on the available bandwidth and total transmit power, the transmitter determines the optimal power allocations and the optimal transmission rates among multiple CDMA channels, as well as the optimal product channel code rate allocation. We also make use of results on the large-system CDMA performance for various multiuser receivers in multipath fading channels. We employ a fast joint source-channel coding algorithm to obtain the optimal product channel code structure. Finally, we propose an end-to-end architecture for multi-layer progressive video delivery over space-time differentially coded orthogonal frequency division multiplexing (STDC-OFDM) systems. We propose to use progressive joint source-channel coding to generate operational transmission distortion-power-rate (TD-PR) surfaces. By extending the rate-distortion function in source coding to the TD-PR surface in joint source-channel coding, our work can use the ??equal slope?? argument to effectively solve the transmission rate allocation problem as well as the transmission power allocation problem for multi-layer video transmission. It is demonstrated through simulations that as the wireless channel conditions change, these proposed schemes can scale the video streams and transport the scaled video streams to receivers with a smooth change of perceptual quality.
... In recent years the concept of intelligent multi-mode, multimedia transceivers (IMMT) has emerged in the context of wireless systems [1] and the range of various existing solutions that have found favour in existing standard systems was summarised in the excellent overview by Nanda [1]. Due to lack of space in this treatise we have to limit our discourse to a small subset of the associated wireless video transceiver design issues, referring the reader for a deeper exposure to the literature cited [3][13]. A further advantage of the IMMTs of the near future is that due to their flexibility they are likely to be able to reconfigure themselves in various operational modes in order to ensure backwards compatibility with existing, so-called second generation standard wireless systems, such as the Japanese Digital Cellular [14], the Pan-American IS-54 [15] and IS-95 [16] systems, as well as the Global System of Mobile Communications (GSM) [17] standards. ...
... In order to provide wireless videophony services in the context of proprietary or second-generation (2G) wireless systems , an additional speech channel has to be allocated for the transmission of the video information. Here we briefly review some of the associated fixed but programmable-rate video coding aspects with reference to some of the recent literature on the topic [3][13]. We note, however that a substantial amount of further work has to be carried out in the area of intelligent transducers and interfaces, such as appropriate miniaturized video cameras, low powerconsumption video displays and in particular in the field of ergonomic hand-held portable multi-media communicator construction and design. ...
... $Figure 17: Video PSNR versus channel SNR comparision of the fixed modulation modes of BPSK, 4QAM and 16QAM, and the burst-by-burst AQAM/CDMA modem, supporting 2 users with the aid of joint detection. These results were recorded for the Miss-America video sequence at SQCIF resolution (128x96 pels) over the COST 207 Bad Urban channel model at a FER of 5% Hanzo et al. [3] ...
The design trade-offs of interactive wireless video systems are
discussed and performance comparisons are provided both in the context
of second- and third-generation wireless videophone systems. We commence
our discussions by a comparative study of arbitrarily programmable, but
fixed-rate, videophone codecs using quarter common intermediate format
(QCIF) video sequences scanned at 10 frames/s. These proprietary codecs
were designed to allow direct replacement of mobile radio voice codecs
in second generation wireless systems, such as the Pan-European GSM, the
American IS-54 and IS-95 as well as the Japanese systems, operating at
13, 8, 9.6 and 6.7 kbps, respectively, although better video quality is
maintained over higher-rate, 32 kbps cordless systems, such as the
Japanese PHS and the European DECT and CT2 systems. From the range of
codecs investigated, best overall performance was achieved by our
vector-quantised codecs, followed by the discrete cosine transformed and
the quadtree-based schemes. The associated video peak signal-to-noise
ratio (PSNR) was around 30 dB, while the subjective video quality can be
assessed under http://www-mobile.ecs.soton.ac.uk. A range of multimode
wireless transceivers is also proposed. The second part of the paper is
dedicated to burst-by-burst (BbB) adaptive wireless video transceivers
employing the standard H.263 codec. It is demonstrated that the proposed
BbB adaptive transceivers provide an improved video performance in
comparison to their statically reconfigured counterparts in the context
of both wideband BbB adaptive quadrature amplitude modulation (AQAM)
transceivers and the joint-detection based code division multiple access
(CDMA) transceivers of the third generation systems
... In [10] Irregular-Variable length coding (IVLC) is presented which endow its performance to near capacity joint source coding. Cordless video telephony and interactive cellular use burst by burst adoptive transceivers and a principles scheme is designed in [16]. Sphere Packing modulation aided Differential Space Time Spreading (SP-DSTS) is briefly presented in [3][4][5]. ...
... Mutual information is also called relative entropy where it represents the distance between two probability distributions. For a transmitted symbol Y and with the channel output Z, the mutual information can be expressed as with the following equation [15,16]. ...
With the ever growth of Internet users, video applications, and massive data traffic across the network, there is a higher need for reliable bandwidth-efficient multimedia communication. Versatile Video Coding (VVC/H.266) is finalized in September 2020 providing significantly greater compression efficiency compared to Highest Efficient Video Coding (HEVC) while providing versatile effective use for Ultra-High Definition (HD) videos. This article analyzes the quality performance of convolutional codes, turbo codes and self-concatenated convolutional (SCC) codes based on performance metrics for reliable future video communication. The advent of turbo codes was a significant achievement ever in the era of wireless communication approaching nearly the Shannon limit. Turbo codes are operated by the deployment of an interleaver between two Recursive Systematic Convolutional (RSC) encoders in a parallel fashion. Constituent RSC encoders may be operating on the same or different architectures and code rates. The proposed work utilizes the latest source compression standards H.266 and H.265 encoded standards and Sphere Packing modulation aided differential Space Time Spreading (SP-DSTS) for video transmission in order to provide bandwidth-efficient wireless video communication. Moreover, simulation results show that turbo codes defeat convolutional codes with an averaged Eb/N0 gain of 1.5 dB while convolutional codes outperform compared to SCC codes with an Eb/N0 gain of 3.5 dB at Bit Error Rate (BER) of 10−4. The Peak Signal to Noise Ratio (PSNR) results of convolutional codes with the latest source coding standard of H.266 is plotted against convolutional codes with H.265 and it was concluded H.266 outperform with about 6 dB PSNR gain at Eb/N0 value of 4.5 dB.
... Os sistemas móveis atuais, conhecidos como sistemas móveis de segunda geração (Second-Generation (2G) Wireless Systems), apesar de possuírem uma tecnologia bastante amadurecida e difundida, sofrem dificuldades quando o seu uso se volta para transmissão a altas taxas de dados [1]. Em função disso, buscou-se a utilização de padrões que possibilitassem o emprego de transferência de dados a taxas maiores. ...
... A potência do sinal nesta distância é a média da intensidade do campo recebido com uma distribuição gaussiana [1]. A confiabilidade de 90 % (R 90 ) na cobertura da célula é aceita como uma meta de serviço nos sistemas celulares e pode ser relacionada a R 50 por [3] ...
This paper presents some results from the analysis of microcell system integrated by optical links. It is used four propagation models (Tonguz-Jung, Okumura-Hata, Maciel-Bertoni-Xia, and Cost 231 Walfisch-Ikegami) with link optical model, where systems parameters like optical modulation index, carrier-to-noise ratio, height of transmission antenna, and power of transmission are under analysis. It is show that the Maciel-Bertoni-Xia and Cost 231 Walfisch-Ikegami models are the best to be used. The problems found with the four models are highlighted. The numerical simulation results for the optical-cellular system with diferent propagation models are presented.
... Nasruminallah and Lajos Hanzo, Fellow, IEEE 1 PSNR is the most widely used and simplest form of objective video quality measure that represents the ratio of the peak-to-peak signal to the root-meansquared noise [1]. days of wireless video communications [3]- [6], substantial further advances have been made in the field of both proprietary and standard-based solutions [7], [8]. Furthermore, the joint optimization of different functions such as joint source and channel decoding (JSCD) has gained considerable attention. ...
... On the other hand, a novel irregular variable length coding (IrVLC) scheme designed for near-capacity joint source and channel coding was proposed in [17]. Likewise, Hanzo et al. [7] advocated the employment of state-of-the-art high-speed-packet-access-style [18] burst-byburst adaptive transceivers for interactive cellular and cordless video telephony, which are capable of accommodating 0018-9545/$26.00 © 2009 IEEE time-variant channel quality fluctuation of wireless channels. An iterative-source-and-channel-decoding-aided irregular convolutional coded videophone scheme using reversible variablelength codes and the maximum a posteriori (MAP) [19] detection algorithm was proposed in [20]. ...
In this paper, we propose a family of short block codes (SBCs) designed for guaranteed convergence in soft-bit-assisted iterative joint source and channel decoding, which facilitate improved iterative soft-bit source decoding (SBSD) and channel decoding. Data-partitioned (DP) H.264 source-coded video is used to evaluate the performance of our system using SBC-assisted SBSD, in conjunction with recursive systematic convolution (RSC) codes for transmission over correlated narrow-band Rayleigh fading channels. The effect of different SBC schemes having diverse minimum Hamming distances d H,min and code rates on the attainable system performance is demonstrated, when using iterative SBSD and channel decoding, while keeping the overall bit rate budget constant by appropriately partitioning the total available bit rate budget between the source and channel codecs to improve the overall bit error rate (BER) performance and to enhance the objective video quality expressed in terms of peak signal-to-noise ratio (PSNR). EXtrinsic Information Transfer (EXIT) charts were used to analyze the attainable system performance. Explicitly, our experimental results show that the proposed error protection scheme using rate-1/3 SBCs having d H,min = 6 outperforms the identical-rate SBCs having d H,min = 3 by about 2.25 dB at the PSNR degradation point of 1 dB. Additionally, an Eb / N 0 gain of 9 dB was achieved, compared with the rate-5/6 SBC having d H,min = 2 and an identical overall code rate. Furthermore, an Eb / N 0 gain of 25 dB is attained at the PSNR degradation point of 1 dB while using iterative soft-bit source and channel decoding with the aid of rate-1/3 SBCs relative to the identical-rate benchmarker.
... In addition, Internet protocol (IP)-based architecture for 3G wireless systems promises to provide next-generation wireless services such as voice, high-speed data, Internet access, and audio and video streaming on an all IP network [3]. However, wireless video [4]- [6] has bandwidth, delay, and loss requirements, many existing mobile networks cannot provide a guaranteed quality of service because temporally high bit error rates (BERs) are unavoidable during fading periods. This paper presents an approach to providing robust and efficient transmission of video over personal communications networks employing wireless CDMA access. ...
... In addressing a topic as broad as wireless video [4]- [6], which could encompass source coding, channel coding, modulation, channel equalization, and multiuser detection, ideally an end-to-end approach should be taken that considers all components together. In practice, however, depending on their background, researchers typically focus on one or two areas (e.g., error-resilient video coding [21], channel coding [22], joint source-channel coding (JSCC) [23]). ...
Error control and power allocation for transmitting wireless video
over CDMA networks are considered in conjunction with multiuser
detection. We map a layered video bitstream to several CDMA fading
channels and inject multiple source/parity layers into each of these
channels at the transmitter. At the receiver, we employ a linear minimum
mean-square error (MMSE) multiuser detector in the uplink and two types
of blind linear MMSE detectors, i.e., the direct-matrix-inversion blind
detector and the subspace blind detector, in the downlink, for
demodulating the received data. For given constraints on the available
bandwidth and transmit power, the transmitter determines the optimal
power allocation among different CDMA fading channels and the optimal
number of source and parity packets to send that offer the best video
quality. We formulate a combined optimization problem and give the
optimal joint rate and power allocation for each of these three
receivers. Simulation results show a performance gain of up to 3.5 dB
with joint optimization over with rate optimization only
... The signal is sent through two antennas while the receiver has a single antenna. Figure 2 shows the iterative decoder whose statistical independence is given by the EXIT charts, and it employs that it is always related to its length [7]. As a result, rather than performing the ISCD independently on the various frames slices, the bits generated by the MBs are concatenated in single stream and processed so as to achieve improvement in performance without giving in the form of video delay. ...
The introduction of 5G with excessively high speeds and ever-advancing cellular device capabilities has increased the demand for high data rate wireless multimedia communication. Data compression, transmission robustness and error resilience are introduced to meet the increased demands of high data rates of today. An innovative approach is to come up with a unique setup of source bit codes (SBCs) that ensure the convergence and joint source-channel coding (JSCC) correspondingly results in lower bit error ratio (BER). The soft-bit assisted source and channel codes are optimized jointly for optimum convergence. Source bit codes assisted by iterative detection are used with a rate-1 precoder for performance evaluation of the above mentioned scheme of transmitting sata-partitioned (DP) H.264/AVC frames from source through a narrowband correlated Rayleigh fading channel. A novel approach of using sphere packing (SP) modulation aided differential space time spreading (DSTS) in combination with SBC is designed for the video transmission to cope with channel fading. Furthermore, the effects of SBC with different hamming distances d(H,min) but similar coding rates is explored on objective video quality such as peak signal to noise ratio (PSNR) and also the overall bit error ratio (BER). EXtrinsic Information Transfer Charts (EXIT) are used for analysis of the convergence behavior of SBC and its iterative scheme. Specifically, the experiments exhibit that the proposed scheme of error protection of SBC d(H,min) = 6 outperforms the SBCs having same code rate, but with d(H,min) = 3 by 3 dB with PSNR degradation of 1 dB. Furthermore, simulation results show that a gain of 27 dB Eb/N0 is achieved with SBC having code rate 1/3 compared to the benchmark Rate-1 SBC codes.
... Coupled with the notion of ubiquitous computing [18] and nomadic communications [19,20], mobile networking practices enable tangible new possibilities for novel kinds of multimedia applications 'on the move': navigation [21,22,23], personal locator services [24] interactive audio/video [25,5], network games [26]. ...
This paper presents an architecture and protocol in support of seamless mobility for future IP Radio Access Networks (IP-RANs). It encompasses a novel approach for seamless handoff and proactive relocation of IP roaming state. The latter establishes a generic substrate for proactive state relocation of different context classes relating to the state of IP connectivity for a mobile node (MN). To address such form of IP mobility, the proposed model iden-tifies a tentative mobility matrix (TMM), which represents an ac-curate mapping between a mobility neighbourhood vector (MNV), surrounding the current point of attachment of an MN and the cor-rect underlying routing neighbourhood vector (RNV), over arbi-trary routing topologies. Sustained IP connectivity is achieved by introducing a 1-neighbour-lookahead (1-NL) view of IP roaming state derived from the established TMM component; seamlessness is pursued through mapping of the 1-NL component to some proactive care-of address (PCoA) onto the IP Multicast domain; this allows ab-stracting a plurality of candidate care-of address instantiations of the MN onto a single handoff routing identifier.
... Among the Internet-based applications , video streaming is particularly hard for two reasons: frequent packet losses and lack of processing power of handsets . Unlike the fixed-line networks, the wireless networks have frequent packet losses as well as high RTT (round-trip time) [7]. The packet loss is quite persistent regardless of time, and the rate is relatively high (5–10%) by noise. ...
This paper presents a case of video streaming system for mobile phone which has actually been implemented and deployed for commercial services in CDMA2000 1X cellular phone networks. As the computing environment and the network connection of cellular phones are significantly different from the wired desktop environment, the traditional desktop streaming method is not applicable. Therefore, a new architecture is required to suit the successfully streaming in the mobile phone environment. We have developed a very lightweight video player for use in mobile phone and the related authoring tool for the player. The streaming server has carefully been designed to provide high efficiency, reliability and scalability. Based on a specifically-designed suite of streaming protocol, the server employs an adaptive rate control mechanism which transmits the media packets appropriately into the network according to the change in network bandwidth.
... Concerning communication links for practical UVS systems, high-speed wireless links (Keller and Hanzo 2000;Hanzo et al. 2000) are becoming more and more important for connecting cameras to VIP units to achieve ubiquity in video surveillance as they allow higher flexibility and lower costs in installation. The most widely used wireless surveillance cameras generally adopt H.264 source coding and robust wideband transmission techniques (e.g., direct sequence spread spectrum) at high frequencies (e.g., 2.4-GHz ISM band). ...
Video surveillance systems are playing an important role to protect lives and assets of individuals, enterprises and governments. Due to the prevalence of wired and wireless access to Internet, it would be a trend to integrate present isolated video surveillance systems by applying distributed computing environment and to further gestate diversified multimedia intelligent surveillance (MIS) applications in ubiquity. In this paper, we propose a distributed and secure architecture for ubiquitous video surveillance (UVS) services over Internet and error-prone wireless networks with scalability, ubiquity and privacy. As cloud computing, users consume UVS related resources as a service and do not need to own the physical infrastructure, platform, or software. To protect the service privacy, preserve the service scalability and provide reliable UVS video streaming for end users, we apply the AES security mechanism, multicast overlay network and forward error correction (FEC), respectively. Different value-added services can be created and added to this architecture without introducing much traffic load and degrading service quality. Besides, we construct an experimental test-bed for UVS system with three kinds of services to detect fire and fall-incident features and record the captured video at the same time. Experimental results showed that the proposed distributed service architecture is effective and numbers of services on different multicast islands were successfully connected without influencing the playback quality. The average sending rate and the receiving rates of these services are quite similar, and the surveillance video is smoothly played.
... There has been an increasing interest in multimedia transmission over wireless channels [1] [2]. However, in wireless systems the data rate is limited by both the available bandwidth and transmission power. ...
We propose a novel class of short block codes (SBCs) designed for guaranteed convergence to an infinitesimally low bit error ratio (BER), which relies on the joint optimisation of soft-bit assisted iterative source and channel decoding. An iterative detection aided combination of SBC assisted soft-bit source decoding (SBSD) and a rate-1 precoder was used to evaluate the attainable performance of the proposed scheme for transmission of data-partitioned (DP) H.264 source coded video over correlated narrowband Rayleigh fading channels. Additionally, we demonstrated the effects of different SBCs having diverse minimum Hamming distances [dH,min] but identical coding rates on both the overall BER performance as well as on the objective video quality expressed in terms of the Peak Signal-to-Noise Ratio (PSNR). The convergence behaviour of the iterative decoding scheme using SBCs as a function of dH,min is analysed by utilising extrinsic information transfer (EXIT) charts. Explicitly, our experimental results show that the proposed error protection scheme using SBCs associated with dH,min = 6 outperforms the identical-code-rate SBCs having dH,min = 3 by about 3 dB at the PSNR degradation point of 1 dB. Additionally, an Eb/No gain of 27 dB is attained using iterative soft-bit source and channel decoding with the aid of rate-1/3 SBCs relative to the identical-rate benchmarker.
... Since the early days of wireless video communications [74][75][76] substantial further advances have been made both in the field of proprietary and standard-based solutions [77,78]. Furthermore, the discovery of turbo codes [52,79] made it practical to achieve transmission close to the Shannon limit at a moderate computational complexity and delay. ...
In this tutorial, a unified treatment of the topic of near capacity multimedia communication systems is offered, where we focus our attention not only on source and channel coding but also on their iterative decoding and transmission schemes. There is a paucity of up-to-date surveys and review articles on the unified treatment of the topic of near capcity multimedia communication systems using iterative detection aided joint source-channel decoding employing sophisticated transmission techniques - even though there is a plethora of papers on both iterative detection and video telephony. Hence this paper aims to fill the related gap in the literature.
... To illustrate the video processings we also have implemented a Discrete Cosine Transform, which is working on 8x8 pixels MacroBlocs, since this kind of algorithm is nearly systematic in video compression's standards like MPEGx or H.26x [7]. The 2 loop kernels of this algorithm are based on a Multiplication-ACcumulation. ...
Within the framework of third generation telecommunication domain, reconfigurable architectures are becoming more and more popular thanks to both their flexibility and performances. In order to define a system that combines high-performance and low-energy consumption, a dynamically reconfigurable architecture, designed with energy awareness is proposed. This paper presents the main features of the DART architecture along with results from the application domain implementations. These results validate the architectural choices and demonstrate the adequacy between DART and next generation telecommunication applications.
... These processings have been chosen because they are representative of three main kinds of critical processings in this application domain : the video processing, the speech processing and the W-CDMA. To illustrate video processing we have implemented a Discrete Cosine Transform, which is working on 8x8 pixel macroblocs, since this kind of algorithm is nearly systematic in video compression standards like MPEGx or H.26x [11]. The 2 loop kernels of this algorithm are based on Multiplication-ACcumulation. ...
In addition to the high performance requirements inher ent to multimedia processings or to W-CDMA, future gen eration mobile telecommunicatio ns brings new constraints to the semiconductor design world. Infact. to support these processings, a system will have to be very flexible, in or der to support the various algorithms allowed by the norm and the addition of new services, while keeping an energy consumption level compatible with the portability notion of this system. In order to associate high performances and low energy consumption in a flexible system, we developed a dynamically reconfigurable architecture called DART. The aim of this paper is to present this architecture and to esti mate its level of peiformance and its adequacy with foture generation mobile telecommunicatio n systems.
... Finally, for completeness we also mention that the authors of [16]- [22] have conceived a range of near-instantaneously adaptive video transceivers, where the channel quality experienced was used for instructing a VBR video codec to represent the current video frame using as many source-coded bits, as many the transceiver was capable of delivering by the current transmission burst. The benefit of this solution is that it is a frame-by-frame encoding regime, which has a fixed video frame-rate, avoids buffering more than a single videoframe and hence does not impose any extra delays. ...
Adaptive media playout (AMP) control is proposed in order to compensate for the bit-ratefluctuation of networks, which may result in playout interruptions in video streaming application. Most AMP algorithms found in the literature trigger playout-rate adjustments based on the buffer fullness or its variation. However, the challenge of the threshold based methods is to select the appro- priate threshold for triggering a playout-rate adjustment owing to the unknown fluctuation of the channel quality and the video bitrate. We conceive an adaptive media playout regime based on underflow probability estimation, which requires no significant statistical knowledge of the previous tele-traffic load. To achieve this, we present an underflow probability estimation model based on large deviation theory relying on the buffer fullness and on its variation. We will then directly use the underflow proba- bility to trigger the actions of playout control, instead of using indirect methods based on a buffer fullness threshold or buffer fullness variation threshold. Experiments based on MPEG-4 Variable Bit-Rate encoded video and VBR channels associated with Adaptive Modulation and Coding are conducted in order to investigate the achievable performance of the proposed algorithm. Our simulation results demonstrate an improved performance in comparison to other recent AMP algorithms.
... Moreover, a irregular variable length coding (IrVLC) scheme designed for near-capacity joint source and channel coding was presented in [7]. Likewise, [8] advocates the employment of state-of-the-art system design principles and the performance of burst-by-burst adaptive transceivers designed for interactive cellular and cordless video telephony. Instead of the traditional serial concatenation of the classic Variable Length Codes (VLC) with a channel code, a parallel concatenated coding scheme was presented in [9] where the VLCs were combined with a turbo code. ...
In this paper we evaluate the performance of a Data- Partitioned (DP) H.264 coded video transmission system using Unequal Error Protection (UEP) IrRegular Convolutional Codes (IRCC). Using UEP, perceptually more important bits are provided with more strong protection relative to less important bits. An iterative detection aided combination of IRCC and a rate-1 precoder was used to improve the overall BER performance and to enhance the objective video quality expressed in terms of Peak Signal-to-noise Ratio (PSNR). The effect of different error protection schemes on the attainable system performance is demonstrated, while keeping the overall bit-rate budget constant for the transmission of DP H.264 source coded video over correlated narrowband Rayleigh fading channels. In this paper we exploited the high design flexibility of IRCCs, which constitutes a family of different rate subcodes, while maintaining an excellent iterative decoding convergence performance. Additionally, due to the use of different-rate subcodes, IRCCs have the capability of providing UEP for the H.264 coded video stream. An EXIT chart matching procedure was used for the design of our specific IRCC. Additionally, EXIT charts were used for analysing the attainable system performance of various error protection schemes employed. Explicitly, our experimental results show that the proposed UEP scheme using IRCC outperforms its Equal Error Protection (EEP) counterpart employing regular convolutional codes by about 0.5 dB Eb/N0 at the PSNR degradation point of 1 dB. I. MOTIVATION AND BACKGROUND
... The emergence of the MPEG-4 standard has brought the area of video segmentation and contour tracking to the forefront of research [1], [2]. Mobile technology and video conferencing type multimedia applications usually generate and display headand-shoulder type images. ...
A new method is proposed to automatically segment out a person's face from a given sequence of images that consists of a head-and-shoulder view, using the fast marching level set approach. The method proposed involves a fast, reliable and computationally efficient algorithm, which exploits the colour information in an image to segment the face region. The colour information is derived and used to classify pixels in the image into different levels depending on the value of the colour information for that pixel. Once this is achieved, a priority-labelled fast marching level set algorithm is utilised to locate the face region. The performance of the fast marching face-segmentation algorithm is illustrated by some simulation results carried out on head-and-shoulder test images.
... Other efforts are in the field of incorporating turbo codes in practical speech and video systems. A range of application examples can be found in the context of interactive and broadcast video systems as well as local area networks in [4] and [5], and in [42]- [52]. In short, an exciting era at the 50th anniversary of Shannonian information theory, witnessing the first practical systems performing close to information theoretical limits, stimulating the research community to aspire to similar performance over dispersive, fading wireless channels. ...
We provide an overview of the novel class of channel codes
referred to as turbo codes, which have been shown to be capable of
performing close to the Shannon limit. We commence with a discussion on
turbo encoding, and then move on to describing the form of the iterative
decoder most commonly used to decode turbo codes. We then elaborate on
various decoding algorithms that can be used in an iterative decoder,
and give an example of the operation of such a decoder using the
so-called soft output Viterbi (1996) algorithm (SOVA). Lastly, the
effect of a range of system parameters is investigated in a systematic
fashion, in order to gauge their performance ramifications
Covering the full range of channel codes from the most conventional through to the most advanced, the second edition of Turbo Coding, Turbo Equalisation and Space-Time Coding is a self-contained reference on channel coding for wireless channels. The book commences with a historical perspective on the topic, which leads to two basic component codes, convolutional and block codes. It then moves on to turbo codes which exploit iterative decoding by using algorithms, such as the Maximum-A-Posteriori (MAP), Log-MAP and Soft Output Viterbi Algorithm (SOVA), comparing their performance. It also compares Trellis Coded Modulation (TCM), Turbo Trellis Coded Modulation (TTCM), Bit-Interleaved Coded Modulation (BICM) and Iterative BICM (BICM-ID) under various channel conditions. The horizon of the content is then extended to incorporate topics which have found their way into diverse standard systems. These include space-time block and trellis codes, as well as other Multiple-Input Multiple-Output (MIMO) schemes and near-instantaneously Adaptive Quadrature Amplitude Modulation (AQAM). The book also elaborates on turbo equalisation by providing a detailed portrayal of recent advances in partial response modulation schemes using diverse channel codes. A radically new aspect for this second edition is the discussion of multi-level coding and sphere-packing schemes, Extrinsic Information Transfer (EXIT) charts, as well as an introduction to the family of Generalized Low Density Parity Check codes. This new edition includes recent advances in near-capacity turbo-transceivers as well as new sections on multi-level coding schemes and of Generalized Low Density Parity Check codes. Comparatively studies diverse channel coded and turbo detected systems to give all-inclusive information for researchers, engineers and students. Details EXIT-chart based irregular transceiver designs. Uses rich performance comparisons as well as diverse near-capacity design examples.
We present a multiple-channel video transmission scheme in wireless CDMA networks over multipath fading channels. We map an embedded video bitstream, which is encoded into multiple independently decodable layers by 3D-ESCOT video coding techniques, to multiple CDMA channels. Each video source layer is protected by a product channel code structure. For a given budget on the available bandwidth and total transmit power, the transmitter determines the optimal power allocations and the optimal transmission rates among multiple CDMA channels, as well as the optimal product channel code rate allocation. We make use of results on the large-system CDMA performance for various multiuser receivers in multipath fading channels. Simulation results show that the proposed framework allows the video quality to degrade gracefully as the fading worsens or the bandwidth decreases, and it offers improved video quality at the receiver.
A framework of joint antenna selection and source coding strategy for scalable video transmission over wireless systems was proposed. Each layer of a bit-stream could be switched among multiple transmit antennas in order to achieve unequal error protection (UEP) for prioritized video layered bit-streams. At first wavelet-based 3D ESCOT codec generates layered bit streams that need prioritized delivery. Then we obtain the ordering of transmit antennas with channel strengths as partial CSI. Finally by the proposed joint antenna selection and layered source coding algorithm, we are able to transmit higher priority layers of a bit stream into antennas with higher channel strengths. Therefore we can achieve UEP for layered scalable video coding transmission over MIMO OFDM system. Simulation results show that the proposed framework improved the system performance significantly. The proposed joint antenna selection and layered source coding algorithm enables us to achieve UEP for layered video transmission over MIMO OFDM system with low complexity of algorithm at the transmit side as well as achieve different QoS at receive side.
A scalable video delivery transmission framework over MIMO OFDM wireless channels by combining power allocation and antenna selection scheme has been proposed. The framework consisting of independently decodable layers (3D ESCOT), UEP channel coding structure, and antenna selection scheme provide more system error resilience even during deep fading period in wireless channels. A new algorithm is proposed to obtain the optimal power allocation and optimal transmission rate allocation among multiple layers, subject to constraints on the total transmission rate and the total power level. This proposed joint power allocation and antenna selection algorithm in MIMO OFDM system enables us to both overcome the challenge of the full CSI that was assumed in the existing approaches and utilize the estimated CSI from the receiver so as to achieve the optimal solution for a video bitstream in terms of its total expected distortion. Generally, as the wireless channel conditions change, the framework we proposed can scale the video streams and transport the scaled video streams to receivers with a smooth change of perceptual quality.
Against the backdrop of the emerging 3G wireless personal communications standards and broadband access network standard proposals, this volume covers a range of coding and transmission aspects for transmission over fading wireless channels. It presents the most important classic channel coding issues and also the exciting advances of the last decade, such as turbo coding, turbo equalisation and space-time coding. It endeavours to be the first book with explicit emphasis on channel coding for transmission over wireless channels.
Divided into 4 parts:
Part 1 - explains the necessary background for novices. It aims to be both an easy reading text book and a deep research monograph.
Part 2 - provides detailed coverage of turbo conventional and turbo block coding considering the known decoding algorithms and their performance over Gaussian as well as narrowband and wideband fading channels.
Part 3 - comprehensively discusses both space-time block and space-time trellis coding for the first time in literature.
Part 4 - provides an overview of turbo equalisations, also referred to as turbo demodulation.
The book systematically converts the lessons of Shannon's information theory into design principles applicable to practical wireless systems. It provides overall design performance studies, giving cognizance to the contradictory design requirements of bit error rate, implementational complexity, coding and interleaving delay, effective throughput, coding rate and other related systems design aspects in a comprehensive manner.
Turbo coding has opened an exciting new chapter in the design of iterative detection assisted communication systems. Similar dramatic advances have been achieved with the advent of space time coding, when communicating over dispersive fading wireless channels. By assuming no prior knowledge in the field of channel coding, the authors provide a self-contained reference on these stimulating hot topics, concluding at an advanced level. This essential volume is divided into five key parts: 1. Convolutional and Block Coding Introduces the family of convolutional codes, hard and soft-decision Viterbi algorithms and the most prominent classes of block codes, namely Reed-Solomon (RS) and Bose-Chaudhuri-Hocquenghem (BCH) codes, as well as their algebraic and trellis-decoding. 2. Turbo Convolutional and Turbo Block Coding Introduces turbo convolutional codes and details the Maximum A-Posteriori (MAP), Log-MAP and Max-Log-MAP as well as the Soft Output Viterbi Algorithm (SOVA). Investigates the effects of the various turbo codec parameters. Studies the super-trellis structure of turbo codes and characterises turbo BCH codes. Portrays Redundant Residue Number System (RRNS) based codes and their turbo decoding. 3. Coded Modulation: TCM, TTCM, BICM, BICM-ID Studies Trellis Coded Modulation (TCM), Turbo Trellis Coded Modulation (TTCM), Bit-Interleaved Coded Modulation (BICM), Iterative BICM (BICM-ID) and compares them under various channel conditions. 4. Space-Time Block and Space-Time Trellis Coding Introduces space-time codes and studies their performance using numerous channel codecs providing guidelines for system designers. Studies Multiple-Input Multiple-Output (MIMO) based schemes and the concept of near-instantaneously Adaptive Quadrature Amplitude Modulation (AQAM) combined with near-instantaneously adaptive turbo channel coding. 5. Turbo Equalisation Covers the principle in detail, provides theoretical performance bounds for turbo equalisers and includes a study of various turbo equaliser arrangements. Also addresses the problem of reduced implementation complexity and covers turbo equalised space-time trellis codes. If you are looking for a comprehensive treatment covering both classic channel coding techniques and recent advances in this field, then this is the book for you. Researchers, practising engineers and advanced students will all find it both informative and stimulating.
Multiuser detection assisted, multiple transmit antenna based OFDM arrangements are studied in the context of HIPERLAN 2-like systems. It is demonstrated that the system user capacity can be improved with the aid of unique spatial user signatures, hence supporting a multiplicity of users. The maximum likelihood sequence estimation (MLSE) detection algorithm outperformed the minimum mean square error (MMSE) scheme by about 5 dB in terms of the required signal-to-noise ratio (SNR) and this performance gain manifested itself also in terms of the system improved video performance.
We present a multiple-channel video transmission scheme in wireless CDMA networks over multipath fading channels. We map an embedded video bitstream, which is encoded into multiple independently decodable layers by 3D-ESCOT video coding technique, to multiple CDMA channels. One video source layer is transmitted over one CDMA channel. Each video source layer is protected by a product channel code structure. A product channel code is obtained by the combination of a row code based on rate compatible punctured convolutional code (RCPC) with cyclic redundancy check (CRC) error detection and a source-channel column code, i.e., systematic rate-compatible Reed-Solomon (RS) style erasure code. For a given budget on the available bandwidth and total transmit power, the transmitter determines the optimal power allocations and the optimal transmission rates among multiple CDMA channels, as well as the optimal product channel code rate allocation, i.e., the optimal unequal Reed-Solomon code source/parity rate allocations and the optimal RCPC rate protection for each channel. In formulating such an optimization problem, we make use of results on the large-system CDMA performance for various multiuser receivers in multipath fading channels. The channel is modeled as the concatenation of wireless BER channel and a wireline packet erasure channel with a fixed packet loss probability. By solving the optimization problem, we obtain the optimal power level allocation and the optimal transmission rate allocation over multiple CDMA channels. For each CDMA channel, we also employ a fast joint source-channel coding algorithm to obtain the optimal product channel code structure. Simulation results show that the proposed framework allows the video quality to degrade gracefully as the fading worsens or the bandwidth decreases, and it offers improved video quality at the receiver.
The fundamental advantage of burst-by-burst (BbB) adaptive
intelligent multimode multimedia transceivers (IMMTs) is
that-irrespective of the propagation environment encountered-when the
mobile roams across different environments subject to path loss; shadow-
and fast-fading; co-channel-, intersymbol-, and multiuser interference,
while experiencing power control errors, the system will always be able
to configure itself in the highest possible throughput mode, while
maintaining the required transmission integrity. Finding a specific
solution to a distributive or interactive video communications problem
has to be based on a compromise in terms of the inherently contradictory
constraints of video quality, bit rate, delay, robustness against
channel errors, and the associated implementational complexity.
Considering some of these tradeoffs and proposing a range of attractive
solutions to various video communications problems is the basic aim of
this overview. The article portrays a range of proprietary video codecs
and compares them to some of the existing standard video codecs. A
number of multimode video transceivers are also characterized. Systems
employing the standard H.263 video codec in the context of wideband BbB
adaptive video transceivers are examined, and the concept of
BbB-adaptive video transceivers is then extended to CDMA-based systems
Burts-by-burst (BbB) adaptive high-speed downlink packet access (HSDPA) style multicarrier systems are reviewed, identifying their most critical design aspects. These systems exhibit numerous attractive features, rendering them eminently eligible for employment in next-generation wireless systems. It is argued that BbB-adaptive or symbol-by-symbol adaptive orthogonal frequency division multiplex (OFDM) modems counteract the near instantaneous channel quality variations and hence attain an increased throughput or robustness in comparison to their fixed-mode counterparts. Although they act quite differently, various diversity techniques, such as Rake receivers and space-time block coding (STBC) are also capable of mitigating the channel quality variations in their effort to reduce the bit error ratio (BER), provided that the individual antenna elements experience independent fading. By contrast, in the presence of correlated fading imposed by shadowing or time-variant multiuser interference, the benefits of space-time coding erode and it is unrealistic to expect that a fixed-mode space-time coded system remains capable of maintaining a near-constant BER.
In the past few years, the development of complex surveillance
systems has captured the interest of both the research and industrial
worlds. Strong and challenging requirements of modern society are
involved in this problem, which aims to increase safety and security in
several application domains such as transport, tourism, home and bank
security, military applications, etc. At the same time, fast
improvements in microelectronics, telecommunications, and computer
science make it necessary to consider new perspectives in this field.
The main objective of this paper is to investigate, discuss, and
evaluate the impact of distributed processing and new communication
techniques on multimedia surveillance systems, which represent the
so-called third-generation surveillance systems (3 GSSs). In particular,
aspects related to the distribution of intelligence among
multiple-processing and wide-bandwidth resources are discussed in
detail. It is shown how distribution of intelligence can be obtained by
a hierarchical architecture that partitions, in a dynamic way, the main
logical processing tasks (i.e., representation, recognition, and
communication) performed in a 3 GSS physical architecture made up of
intelligent cameras, hubs, and central control rooms. The advantages of
this solution are pointed out in terms of 1) increased flexibility and
reconfigurability and 2) optimal allocation of available processing and
bandwidth resources. Finally, a case study is analyzed that allows one
to gain a deeper insight into a distributed surveillance system
Bridging the gap between the video compression and communication communities, this unique volume provides an all-encompassing treatment of wireless video communications, compression, channel coding, and wireless transmission as a joint subject. WIRELESS VIDEO COMMUNICATIONS begins with relatively simple compression and information theoretical principles, continues through state-of-the-art and future concepts, and concludes with implementation-ready system solutions. This book's deductive presentation and broad scope make it essential for anyone interested in wireless communications. It systematically converts the lessons of Shannon's information theory into design principles applicable to practical wireless systems. It provides in a comprehensive manner "implementation-ready" overall system design and performance studies, giving cognizance to the contradictory design requirements of video quality, bit rate, delay, complexity error resilience, and other related system design aspects. Topics covered include • information theoretical foundations • block-based and convolutional channel coding • very-low-bit-rate video codecs and multimode videophone transceivers • high-resolution video coding using both proprietary and standard schemes • CDMA/OFDM systems, third-generation and beyond adaptive video systems. WIRELESS VIDEO COMMUNICATIONS is a valuable reference for postgraduate researchers, system engineers, industrialists, managers and visual communications practitioners. © 2001 by the Institute of Electricaland Electronics Engineers, Inc. All rights reserved.
The past several years have been exciting for wireless communications. The public appetite for new services and equipment continues to grow. The Second Generation systems that have absorbed our attention during recent years will soon be commercial realities. In addition to these standard systems, we see an explosion of technical alternatives for meeting the demand for wireless communications. The debates about competing solutions to the same problem are a sign of the scientific and technical immaturity of our field. Here we have an application in search of technology rather than the reverse. This is a rare event in the information business. Happily, there is a growing awareness that we can act now to prevent the technology shortage from becoming more acute at the end of this decade. By then, market size and user expectations will surpass the capabilities of today's emerging systems. Third Generation Wireless Information Networks will place even greater burdens on technology than their ancestors. To discuss these issues, Rutgers University WINLAB plays host to a series of Workshops on Third Generation Wireless Information Networks. The first one, in 1989, had the flavor of a gathering of committed enthusiasts of an interesting niche of telephony. Presentations and discussions centered on the problems of existing cellular systems and technical alternatives to alleviating them. Although the more distant future was the announced theme of the Workshop, it drew only a fraction of our attention.
Statistical Packet Assignment Multiple Access (SPAMA) is proposed, based on a statistical allocation of bandwidth resources to terminals which share a slotted, framed channel. The statistical nature of the centralized slot assignment scheme allows an accurate matching of bitrate requirements for different multimedia services with a minimal amount of signalling, while maintaining a throughput of up to 93%.
The video performance of the Median wireless asynchronous transfer mode (WATM) system is evaluated for a range of application scenarios using the H.263 video codec and a novel packetisation and acknowledgement scheme. The video resolutions and system parameters used are summarised in Tables 1 and 2. The required channel signal-to-noise ratio for near-unimpaired video quality is about 16dB over the dispersive worst-case channel used.
Following the launch of the Pan-European digital mobile radio (GSM) system its salient features are surnrnarised in this tutorial review [I, 81. Time Division Multiple Access (TDMA) with eight users per carrier is used at a multi-user rate of 27 1 kbit/s, demanding a channel equaliser to combat dispersion. The error protected chip-rate of the full-rate traffic channels is 22.8 kbitls, while in half-rate channels is 11.4 kbitls. There are two speech traffic channels, five different-rate data traffic channels and 14 various control and signalling channels to support the system's operation. A moderately complex, 13 kbitls Regular Pulse Excited speech codec with long term predictor (LTP) is used, combined with an embedded three-class error correction codec and multi-layer interleaving to provide sensitivity-matched unequal error protection for the speech bits.An overall speech delay of 57.5 ms is maintained. Slow frequency hopping at 217 hops/s yields substantial performance gains for slowly moving pedestrians.
Adaptive modulation is applied in conjunction with a Decision Feedback Equalizer (DFE) in order to mitigate the eects of the slowly varying wideband multi-path Rayleigh fading channel in a noise-limited environment. Turbo BCH coding was applied in order to improve the BER and BPS performance and subsequently, the Soft-Decision (SD) based error detection capability of the turbo codec was utilized by the proposed blind modulation detection algorithm in order to detect the various modulation modes.
Radio transmission technologies for IMT-2000 are being studied and standardized all over the world. Several national/regional standardization bodies are developing proposals on radio transmission technologies to meet the cut off date of June 30, 1998 set by ITU. This paper shows necessity to develop and standardize IMT-2000 in Japan, principle attitude on IMT-2000 standardization, selection process of W-CDMA, and the current situation of study on IMT-2000 in ARIB (Association of Radio Industries and Businesses).
In video coding applications the main objective is to remove the vast amount of redundancy which normally exists in the spatial domain (within a frame) as well as in the temporal direction (frame-to-frame). Attempts to minimize the temporal redundancies can be accomplished by interframe coding techniques [1]. In addition, many applications of high compression video coding involve the use of hybrid coding [2]. This method, which is a combination of DPCM and transform coding, is presently considered the most effective coding for video teleconferencing applications [3,4]. The main disadvantage of this method however, is the subjective degradation in which the viewers perceive the outlines of the transform blocks. This type of distortion, which appears as discontinuities at the edges of blocks, can be very objectionable to the viewer. As a result, in this Chapter we present a different approach which is not only free of block distortion but also extremely efficient in terms of compression and hardware complexity. The method is based on bandwidth splitting using quadrature mirror filtering (QMF) which has been extensively investigated in recent years for still image applications [5–14].
This paper gives an overview of worldwide air interface research activities towards the third generation mobile communications (UMTS and FPLMTS/IMT-2000). The status of standardization for each main region (Europe, Asia, USA) is discussed. Third generation research activities are described focusing into the air interface concept developments. Details of different CDMA, TDMA, OFDM and hybrid air interface designs are given. Furthermore, the implications of Time Division Duplex operation are covered.
This chapter presents discrete cosine transform. The development of fast algorithms for efficient implementation of the discrete Fourier transform (DFT) by Cooley and Tukey in 1965 has led to phenomenal growth in its applications in digital signal processing (DSP). The discovery of the discrete cosine transform (DCT) in 1974 has provided a significant impact in the DSP field. While the original DCT algorithm is based on the FFT, a real arithmetic and recursive algorithm, developed by Chen, Smith, and Fralick in 1977, was the major breakthrough in the efficient implementation of the DCT. A less well-known but equally efficient algorithm was developed by Corrington. Subsequently, other algorithms, such as the decimation-in-time (DIT),decimation-in-frequency (DIF), split radix, DCT via other discrete transforms such as the discrete Hartley transform (DHT) or the Walsh-Hadamard transform (WHT), prime factor algorithm (PFA), a fast recursive algorithm, and planar rotations, which concentrate on reducing the computational complexity and/or improving the structural simplicity, have been developed. The dramatic development of DCT-based DSP is by no means an accident.
There is no better way to quantize a single vector than to use VQ with a codebook that is optimal for the probability distribution describing the random vector. However, direct use of VQ suffers from a serious complexity barrier that greatly limits its practical use as a complete and self-contained coding technique
The paper presents twenty-three manuscripts that reflect the recent research in the area of video compression and an overview of their contents. The following aspects of image sequence compression were addressed: motion estimation and compensation; spatiotemporal subsampling and interpolation; bit allocation and rate control; multiresolution compression; low-bit rate compression; and cell-loss concealment.
We investigate the average information rates attained by adapting the transmit power and the information rate relative to channel variations in code division multiple access communication systems, Our results show that the rate adaptation provides a higher average information rate than the power adaptation for a given average transmit power, and the rate increase when using rate adaptation is more significant for channels with a faster decaying multipath intensity profile and weaker line-of-sight component.
Towards location aware mobile ad hoc sensors A Systems Engineering Approach to Wireless Information Networks The Second Edition of this internationally respected textbook brings readers fully up to date with the myriad of developments in wireless communications. When first published in 1995, wireless communications was synonymous with cellular telephones. Now wireless information networks are the most important technology in all branches of telecommunications. Readers can learn about the latest applications in such areas as ad hoc sensor networks, home networking, and wireless positioning. Wireless Information Networks takes a systems engineering approach: technical topics are presented in the context of how they fit into the ongoing development of new systems and services, as well as the recent developments in national and international spectrum allocations and standards. The authors have organized the myriad of current and emerging wireless technologies into logical categories: * Introduction to Wireless Networks presents an up-to-the-moment discussion of the evolution of the cellular industry from analog cellular technology to 2G, 3G, and 4G, as well as the emergence of WLAN and WPAN as broadband ad hoc networks * Characteristics of Radio Propagation includes new coverage of channel modeling for space-time, MIMO, and UWB communications and wireless geolocation networks * Modem Design offers new descriptions of space-time coding, MIMO antenna systems, UWB communications, and multi-user detection and interference cancellation techniques used in CDMA networks * Network Access and System Aspects incorporates new chapters on UWB systems and RF geolocations, with a thorough revision of wireless access techniques and wireless systems and standards Exercises that focus on real-world problems are provided at the end of each chapter. The mix of assignments, which includes computer projects and questionnaires in addition to traditional problem sets, helps readers focus on key issues and develop the skills they need to solve actual engineering problems. Extensive references are provided for those readers who would like to explore particular topics in greater depth. With its emphasis on knowledge-building to solve problems, this is an excellent graduate-level textbook. Like the previous edition, this latest edition will also be a standard reference for the telecommunications industry.
This paper describes picture information transmission for portable
multimedia terminals. The radio links used in portable multimedia
terminals have narrower channel capacity and higher transmission error
rates than wired links such as those used in ISDN. To transmit
multimedia information of satisfactory quality over radio links,
robustness against radio link errors must be improved, because picture
deterioration is much more apparent than audio deterioration. First, the
effects of transmission errors on picture quality are analyzed using the
H.261 coding system used for ISDN picture communication. Second, the
relationship among bit error rate, terminal velocity, and picture
quality is analyzed and the deterioration mechanisms of picture quality
are discussed. Three techniques for improving picture quality against
radio link errors are proposed.
The most recent addition to William C. Y. Lee's acclaimed series on mobile and cellular communications, Mobile Communications Design Fundamentals, Second Edition offers designers, researchers, and students an up-to-date, invaluable guide to the theoretical framework of mobile radio communications and how such systems are designed. With an abundance of new material, this Second Edition covers leading-edge Personal Communications Service (PCS), microcell, and CDMA systems, providing all the theoretical and design knowledge and know-how needed to design with both present and future technology. Useful as a professional handbook or as a senior/graduate level text, the book provides complete coverage of the differences between fixed and wireless radio systems, up to and including the new FCC-promoted PCS systems; an authoritative description of the mobile radio environment that gives engineers the necessary technical background to confidently select the appropriate radio technology; definitive, clearly presented design parameters for both the base and mobile units; troubleshooting approaches that help you anticipate the problems associated with each system and solve them when they arise; comprehensive guidelines for how to develop the system design and frequency plan and how to tackle all capacity issues, and new information on CDMA, a hot broadband radio technology...boosting microcell technology capacity with system planning...built in prediction...analyzing digital communication systems...and covering noncellular mobile radio systems, including those for data communication. With more than half of the material in this new edition based on the author's own widely recognized research work, Mobile Communications Design Fundamentals is a book no one interested in the new wave in mobile communications can afford to miss.
Wideband wireless access based on direct sequence code division multiple access, called coherent multi-rate W-CDMA in this paper, designed for next generation mobile communications systems is introduced. It employs inter-cell asynchronous operation with a fast cell search algorithm, orthogonal multi-spreading factor (SF) forward links, and pilot symbol assisted coherent reverse and forward links. Inter-cell asynchronous cell site operation facilitates the continuous deployment of the system from outdoors to indoors. An orthogonal multi-SF forward link allows flexible offering of different multi-rate services to users without losing orthogonality. Gains in the radio link capacity and coverage are obtained from the use of coherent Rake combining and fast transmit power control (TPC) in both forward and reverse links. Computer simulation and field experiment results of the coherent multi-rate W-CDMA radio link performance are presented. Also presented are interference cancellation and adaptive antenna array techniques that can significantly improve the link capacity and coverage.
Very low bit rate video coding has received considerable attention in academia and industry in terms of both coding algorithms and standards activities. In addition to the earlier ITU-T efforts on H.320 standardization for video conferencing from 64 kbps to 1.544 Mbps in ISDN environment, the ITU-T/SG15 has formed an expert group on low bit coding (LBC) for visual telephone below 64 kbps. The ITU-T/SG15/LBC work consists of two phases: the near-term and long-term. The near-term standard H.32P/N, based on existing compression technologies, mainly addresses the issues related to visual telephony at below 28.8 kbps, the V.34 modem rate used in the existing Public Switched Telephone Network (PSTN). H.32P/N will be technically frozen in January '95. The long-term standard H.32P/L, relying on fundamentally new compression technologies with much improved performance, will address video telephony in both PSTN and mobile environment. The ISO/SG29/WG11, after its highly visible and successful MPEG 1/2 work, is starting to focus on the next- generation audiovisual multimedia coding standard MPEG 4. With the recent change of direction, MPEG 4 intends to provide an audio visual coding standard allowing for interactivity, high compression, and/or universal accessibility, with high degree of flexibility and extensibility. This paper briefly summarizes these on-going standards activities undertaken by ITU-T/LBC and ISO/MPEG 4 as of December 1994.
A comparative study of arbitrarily programmable, but fixed-rate videophone codecs using quarter common intermediate format (QCIF) video sequences scanned at 10 frames/s is offered. These codecs were designed to allow direct replacement of mobile radio voice codecs in second generation wireless systems, such as the Pan-European GSM, the American IS-54 and IS-95 as well as the Japanese systems, operating at 13, 8, 9.6 and 6.7 kbps, respectively, although better video quality is maintained over higher-rate, 32kbps cordless systems, such as the Japanese PHS and the European DECT and CT2 systems. Best overall performance was achieved by our vector-quantised codecs, followed by the discrete cosine transformed and the quadtree-based schemes, which were characterised by the bit allocation schemes of Table 1. The associated video Peak Signal-to-Noise Ratio (PSNR) was around 30 dB, while the subjective quality can be viewed under http://www-mobile.ecs.soton.ac.uk. A range of multimode wireless transceivers were also proposed, which are characterised by Table 2.
The paper gives an overview on trends in mobile communications which is currently one of the most emerging markets. The demand for new and extended services is a driving force for the development of future mobile networks. Especially the provision of video communications is a great challenge. The current status of MPEG and ITU-T/SG15 concerning mobile communications is described and related to current projects. In the second part some aspects for the design of video codecs for mobile video telephony are outlined. Based on the coming standard H.26P and advanced video codec is described and its suitability for mobile networks is proven by some simulation results.
Statistical Packet Assignment Multiple Access (SPAMA) is proposed, based on a statistical allocation of bandwidth resources to terminals which share a slotted, framed channel. The statistical nature of the centralized slot assign- ment scheme allows an accurate matching of bitrate requirements for different multimedia ser- vices with a minimal amount of signalling, while maintaining a throughput of up to 93%.
In the future satellite broadcasting system in 21 GHz band, the rainfall attenuation is a most significant problem. To solve this problem, the hierarchical transmission systems have been studied. This paper analyzes the performance of the hierarchical modulation scheme from the view point of power assignment in the presence of the rainfall attenuation. This paper shows an optimum power assignment ratio to maximize the spectral efficiency and the signal-to-noise ratio of received image, and these optimum ratio is varied with the measure of system performance.
Fundamentals of VHF and UHF propagation propagation over irregular terrain propagation in built-up areas area coverage and planning tools characterisation of multipath phenomena wideband channel characterisation other mobile radio channels and methods of characterisation sounding sampling and simulation man-made noise and interference multipath mitigation techniques.