Joint source-channel multistream coding and optical network ...

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Joint source-channel multistream coding and optical network ...

IEEE TRANSACTIONS ON MULTIMEDIA, VOL. 4, NO. 1, MARCH 2002 3Joint Source-channel Multistream Coding and OpticalNetwork Adapter Design for Video Over IPJie Chen, Senior Member, IEEE, and K. J. Ray Liu, Senior Member, IEEEAbstract—In this paper, we present a panorama pictureon how to achieve end-to-end video over IP service under thecommunication environments, consisting of backbone networks,hybrid access networks, and the end users. The paper consistsof three equally weighted subtopics which cover some novelthoughts in designing and implementing the video over IP systemin the different areas, namely, a Synchronous Optical NETwork(SONET) network adapter for backbone connections, the jointsource-channel multistream coding in hybrid access networks,and the content-based source coding in the transform domain. Wepropose to link the three different problems associated with thehybrid networks, which have different characteristics and designrequirements, to improve the critical performances in variousareas of video over IP systems. The goal is to deliver video over IPnetworks in more cost-effective and reliable manner.IP/ATM over SONET is currently a commonly used backbonetechnique. In the first part of this paper, we present a flexibledesign and implementation of a SONET network adapter to carryIP traffic via optical fiber. Unlike many conventional designs,our single-chip implementation supports different data rates(OC-3, OC-12, and OC-48), carries IP traffic directly over fiber,achieves more flexible for multivendor interoperability, andprovides bandwidth efficient designs at the lower system latency.Hybrid access-networks via wireline or wireless connections aremost likely needed for last-mile services. In the second part ofthis paper, we propose a joint source-channel multistream videocoding scheme to combat the transmission errors under the harshnetwork conditions. On top of traditional error control techniques,the simulation results demonstrate that our multistream designoutperform the conventional approaches by up to 5–7 dB underthe harsh network conditions. To support our multistream videocoding scheme, we need to access and manipulate video objectsrather than the frame of pixels. In the third part of this paper, wefocus on the coding of arbitrary shape video fully in the transformdomain.Index Terms—Content-based video coding, error-resilienttransmission, joint source-channel coding, optical network,SONET, streaming video, video over IP.I. INTRODUCTIONWITH the advances in digital compression technologyand the steady deployment of broad-band networkssuch as fiber optics, cable, xDSL, and the third-generationwireless CDMA systems, multimedia services such as packets(data/voice/video over IP) through broad-band networks haveManuscript received September 4, 2001. The associate editor coordinating thereview of this paper and approving it for publication was Dr. John A. Sorensen.J. Chen was with the Electrical and Computer Engineering Department, Institutefor Systems Research, University of Maryland, College Park, MD 20742USA. He is now with Flarion Technologies, Bedminster, NJ 07921 USA.K. J. R. Liu is with the Electrical and Computer Engineering Department, Institutefor Systems Research, University of Maryland, College Park, MD 20742USA.Publisher Item Identifier S 1520-9210(02)01462-1.Fig. 1.Hybrid networks are used to support multimedia services.been emerging as technologies for the new millennium. Thosestate-of-the-art technologies are changing our daily life andenabling not only e-commerce but also traditional businessesto enhance productivity, reduce costs, and increase businessagility. However, before we can realize the full potential ofthese multimedia services, we have to address the challengeof how to deliver multimedia services over networks costeffectively, ubiquitously, and with sufficient quality. Due tothe large variety of existing network technologies, it is mostlikely that hybrid networks are used to support multimediaservices, as shown in Fig. 1. However, different networkshave different characteristics. To optimize the performance ofmultimedia systems, we should consider some improvementsin designing and implementing multimedia over IP systemsuch as jointly considering video compression and deliveryschemes based on the network alternatives, capacities, andcharacteristics. This paper consists of three subtopics whichcover some novel thoughts in designing and implementingmultimedia over IP system in the different areas; namely, aSynchronous Optical NETwork (SONET) network adapter forbackbone connections, the joint source-channel multistreamvideo coding in hybrid access networks, and the content-basedsource coding in the transform domain. Although these threepieces of designs can be totally separated, here we try to putthem together as a system level paper. We do not intend toprovide the solutions to solve the end-to-end problems in thispaper. Instead, we propose the different ideas to improve thecritical performances in the different areas of video over IPsystems. The goal is to deliver video over IP networks in morecost-effective and reliable manner.1520-9210/02$17.00 © 2002 IEEE


4 IEEE TRANSACTIONS ON MULTIMEDIA, VOL. 4, NO. 1, MARCH 2002Fig. 2.Achitecture of a SONET network adapter served as the Layers 1 and 2 IP router to carry packets over SONET or directly over fiber.TABLE IDIFFERENT TECHNIQUES FOR ROBUST TRANSMISSIONA. Improvements in Designing a SONET Network Adapter forBackbone ConnectionsCompared to speech communications, video communicationsalways have enormous bandwidth requirements. Whenthe traffic volume is low between IP routers, bandwidth partitionsover a common interface made it attractive to carry IPpackets over ISDN, X.25, or frame relay connections. As thetraffic grows, it is becoming more desirable to carry packetsover the SONET [1] or directly over optical fiber because ofits reliability and broad bandwidth, at least in the backboneconnections or core networks which connect different internetservice providers (ISPs), as shown in Fig. 1.In the first part of our paper, we propose a flexible designand implementation of a SONET network adapter served asthe Layers 1 and 2 (the physical and data link layers) IP router,as shown in Fig. 2, which supports packets over SONET ordirectly over fiber. Because we emphasize on the design andimplementation of video over IP system in this paper, wehereby refer packet to the IP packet of video or simply asvideo/IP. Unlike the conventional packets over SONET [2] orover ATM then over SONET [3] design, our single-chip designsupports packets directly over fiber or over SONET at differenttransmission rates (OC-3, OC-12, and OC-48). In addition, ourhigh-speed (2.4 Gb/s) SONET network adapter design providesmore flexibilities for network security and multivendor interoperability.Moreover, we can reduce bandwidth requirements byapplying the point-to-point protocol (PPP) header compressionand the “strip PPP and CRC fields off” features.B. Joint Source-Channel Multistream Coding in HybridAccess NetworksAlthough the optical network is ideal for video transmissionbecause of its reliability and broad bandwidth, it is nowstill too expensive for average users to have the fiber connectedto the homes. As a result, access networks or last-mileservices of either wireline or wireless connections are usuallyneeded before we reach the core networks. Advanceddigital communication technologies such as cable, xDSL, andthe third-generation wireless CDMA are used to connect endusers to core networks. One inherent problem of any accessnetwork connections is that information may be alteredor lost during transmission due to channel noise/interference.The effect of such information loss can be devastating for thecompressed video because any damage to the compressed bitstream may lead to objectionable visual distortion at the decoder;it also causes the commonly used predictive coding topropagate errors in the reconstructed video to future frames inthe video sequence. Furthermore, in the commonly used variablelength coding (VLC), the boundary between video codewords is implicit. Transmission errors typically lead to an incorrectnumber of bits being used in VLC decoding, whichcauses loss of synchronization with the encoder. Many errorcontrol techniques therefore have been proposed for videocommunication, as summarized in Table I. On one hand, thetraditional error control and recovery schemes in data communicationshave been extended for video transmission. Onthe other hand, the signal-reconstruction and error-concealmenttechniques have been proposed to strive to obtain a closeapproximation of the original signal or attempt to make theoutput signal at the decoder least objectionable to human eyes.For data-oriented services, information reliability is moreimportant than system delay/latency. Therefore, the missingpacket retransmission is requested by the TCP layer at theend users at the cost of extra delay/latency. However, forreal-time multimedia over IP services, the extra delay/latencyis intolerable, and real-time protocol (RTP) user datagramprotocol (UDP) instead of TCP is used. The real-time and


CHEN AND LIU: JOINT SOURCE-CHANNEL MULTISTREAM CODING AND OPTICAL NETWORK 5interactivity requirements may exclude the deployment of somewell-known error-recovery techniques such as the AutomaticRepeat-reQuest (ARQ) retransmission. In addition, issues suchas audio-visual synchronization and multipoint communicationsfurther complicate the problem of error recovery. Asa result, the self-recovery transmission mechanism (or jointsource-channel transmission mechanism) is more desirable.Our multistream video over IP in the second part of this paper isactually designed for that purpose. As the video segmentationtechniques become more mature [13], it enables us to view,access, and manipulate video objects rather than the frame ofpixels with great error robustness at a large range of bit-rates.On top of the traditional error control techniques such asforward error correction (FEC), we propose improvementsin robust transmission in the second part of our paper bytaking advantage of both content-based video coding and jointsource-channel coding.C. The Coding of Arbitrary Shape Video Fully in theTransform DomainTo support the content-based multistream video codingscheme proposed in the second part of this paper, we need toperform motion estimation and compensation for arbitrarilyshaped video objects. Now the question becomes “What thecontent-based source coding scheme should we use?” Surely,we can adopt the MPEG-4 content-based video coding design tomeet our requirements. However, in the third part of our paper,we propose an alternative coding of arbitrary shape video fullyin transform domain instead. Unlike the conventional MPEG-4modified block matching approaches [14], [15], the motionestimation and compensation in our coding scheme work fullyin the discrete cosine transform (DCT) instead of the spatialdomain. It is important to recognize that the DCT-based natureof our improved design enables the efficient combination ofboth DCT and motion estimation/compensation units, whichconsume most of computing power in a video coder [16]–[18],into one single unit. Therefore, our emphasis in this designis on optimizing the hardware complexity by minimizing thecomputational units along the data path. Simulation resultsdemonstrate the comparable performances between our transformdomain approach and the conventional MPEG-4 design[14], [15]. Notice that, if the original input image sequences arenot decomposed into several video objects of arbitrary shapes,the proposed scheme simply degenerates into supporting thevideo coding of conventional image sequences with rectangularshapes. Therefore, this approach can be viewed as a logicalextension of the DCT-based motion estimation/compensationschemes [19]–[22] toward arbitrarily shaped video sequences.D. Paper OutlineIn this paper, we present novel concepts in combining the designand implementation of video over IP systems from variousaspects of core network design, data transmission in access network,and source coding. Our contributions include:1) a flexible single-chip design and implementation of aSONET network adapter served as the Layers 1 and 2Fig. 3. SONET can transport many different digital signals at manydifferent data rates. Here, stratum is used as the reference clock for networksynchronization.IP router to deliver video over IP over SONET/SDH ordirectly over fiber, which provides reliable, high-speed,and multivendor compatible backbone connectionsbetween ISPs;2) a joint source-channel multistream video coding schemeto combat the transmission errors in hybrid access networksunder the harsh network conditions; and3) a content-based video coding fully in DCT-domain to reducethe hardware complexity.The rest of paper is organized as follows. In Section II, wepresent our flexible design of the SONET network adapterto deliver video over IP over SONET or directly over fiber.Then we describe our joint source-channel multistream videocoding scheme to combat the transmission errors in Section III.In Section IV, an efficient DCT-domain content-based videocoding mechanism is proposed. Finally, we conclude our workin Section V.II. A FLEXIBLE DESIGN AND IMPLEMENTATION OFSONET NETWORK ADAPTERAs we have pointed out in the introduction, video overoptical network is a leading expedient solution to providehigh-capacity backbone connections between ISPs. Althoughour design (as shown in Fig. 2) can support both IP andATM traffic, here we only emphasize the IP design in thispaper (please refer to [23] for ATM over SONET or directlyover fiber design). Before we proceed with our discussion,we feel the need to answer some common questions. “Fiberoptics provides abundant bandwidth, how does the video informationbe encapsulated in it?” SONET [or its counterpart,synchronous digital hierarchy (SDH)] [1] is a standard whichprovides different rates, formats, and optical parameter specificationsfor optical interfaces ranging from 51 Mb/s (OC-1) to9.8 Gb/s (OC-192) capacities. The major attribute of SONETis its ability to transport many different (asynchronous or synchronous)digital signals using a standard synchronous transportsignal (STS) format, as shown in Fig. 3. The video signalscan vary from low bit-rate (DS0) in H.263, to medium bit-rate(DS-1, CEPT-1) in MPEG-1, up to high bit-rate (DS-2, DS-3)


6 IEEE TRANSACTIONS ON MULTIMEDIA, VOL. 4, NO. 1, MARCH 2002Fig. 4.Pipelined architecture of the payload processor: (a) payload processor at the receiver site and (b) payload processor at the transmit site.in MPEG-2 and HDTV applications. The mapping of tributarysignals (DS-1, CEPT-1, and DS-2) into an STS is accomplishedthrough the use of virtual tributaries and payloadpointers. The pointers allow the flexible alignment of payloadwithin the transport signal by indicating where the asynchronousor synchronous payload begins [1]. DS0 signals at64 kb/s are not addressed as a SONET format, but they areprepackaged in DS1s via switches, channel banks, etc. Anothercommon question is: “What is novel in our design?”Unlike many conventional designs, we support the following:1) different data rates (OC-3, OC-12, and OC-48) and differenttraffics (IP and ATM) on a single device;2) IP traffic not only over SONET but also directly overfiber (there are many designs of IP over SONET thenover fiber, but not IP directly over fiber);3) more flexible design for multivendor interoperabilityand lower system latency (we provide many choices forISPs to configure the device as we will discuss later);4) bandwidth reduction by applying our novel PPP headercompression and the “strip PPP header and CRC fieldsoff” features;5) two levels of scrambling to prevent malicious networkattacks.Packet over SONET, or more accurately Video/IP/PPP/HDLC/SONET, is described in [2]. Compared to the packetover SONET design, the packet directly over fiber design ismore cost-effective and bandwidth efficient. Because the dataengine or the core processor in our SONET network adapterdesign is the “payload processor,” we derive the fully pipelinedarchitecture in the payload processor, as shown in Fig. 4.The difference between the packet directly over fiber andover SONET design is whether or not to bypass the “SONEToverhead processor” block, as shown in Fig. 2. In addition, thedesigns of the HDLC stuffing and destuffing blocks in “payloadprocessor,” as shown in Fig. 4 are different between thesetwo modes. Here, we provide a provisionable mode-controlregister to control the operating modes of the device (0: packetover fiber, 1: packet over SONET). On the other hand, what iscommon between these two operating modes is that they sharethe same “payload processor” andnetwork interface” blocks[24]. In what follows, we will explain our SONET networkadapter design and implementation in more detail. Furthermore,we will focus on the design of each block at the receive sideof the payload processor, and the mirrored approach can beapplied to the transmit side.A. The Design of HDLC Stuffing and Destuffing BlocksHDLC stuffing and destuffing blocks (shown in Fig. 4)work differently for the packet over SONET versus the packetdirectly over fiber mode. For the video/IP over SONET design,the “HDLC destuffing” block delineates a HDLC packet usingthe HDLC character (0x7E) at the transmit side, then the flagsare removed at the receive side. The unescaped destuffing isperformed (0x7D5D is unescaped to 0x7D and 0x7D5E is unescapedto 0x7E). However, for our novel packet directly overfiber design, the HDLC byte boundary is meaningless becausethe synchronization mechanism used in packet over SONETdesign is no longer valid. Therefore, the bit-synchronousHDLC has to be used. For the bit-synchronous HDLC, the“HDLC stuffing” block at transmit side inserts a zero bitafter any run of five consecutive logic-one bits of transmittedpackets. This distinguishes the real data from the HDLCend-of-packet mark, which is “01 111 110” (0x7E) in binary ora run of six consecutive logic-ones, which cannot by definitionbe a part of the encoded user data. The “HDLC destuffing”block at the receive side can easily decode this bit stream bycounting consecutive logic-ones. If the counter reaches five andthe next bit is a zero, then that zero is a stuffing bit and shouldbe removed. If the bit is one, the end-of-packet is encountered.In addition, no escape characters, i.e., 0x7E is escaped to0x7D5E, are needed in the bit-synchronous HDLC thus beingmore bandwidth efficient compared to the byte-synchronousHDLC. The optional CRC-32 or CRC-16 calculation on thewhole received packet is performed after HDLC destuffing.The result after the CRC checking should be all zeros and adifferent value indicates an error.B. The Design of PPP DetachCreativity and flexibility have been built in our “PPP detach”design to improve bandwidth efficiency. The procedure of PPPdetach is listed in Table II. Here, PayloadType [1:0] register isused to distinguish the different types of data traffics becauseour device is capable of handling both ATM and IP traffics. Inthe packet over SONET design, the EOPMarker indicates theend of packets (EOP). As we have mentioned in the introduction,any networks may experience information loss or distortion.It is better, from the performance point of view, not to discardthe error packets in real-time multimedia services becausethe decoder may still be able to utilize those packets to recoverthe original signals or some useful information by applying the


CHEN AND LIU: JOINT SOURCE-CHANNEL MULTISTREAM CODING AND OPTICAL NETWORK 7TABLE IIPROCEDURE OF PPP DETACH. HERE, POS STANDS FOR PACKET OVER SONETOR DIRECTLY OVER FIBERerror control schemes, as listed in Table I. Therefore, we introduceour novel BadMarker and AbortMarker to distinguish anerror packet from a good packet ending with EOPMarker.• BadMarker: This is used to mark the packet as a bad oneonce a checksum mismatch occurs, and the “CRC check”block at the receiver side can change EOPMarker to Bad-Marker.• AbortMarker: This indicates where the packet getsaborted, i.e., due to a service disruption.Notice that every packet can only end up with either EOP-Marker, BadMarker, or AbortMarker, but not both. Next, wewill discuss our novel design in the kernel (Step 3) of the PPPdetach procedure listed in Table II.1) Malformed Packets Handling: According to the InternetEngineering Task Force (IETF) 1619 Standard [2]:Any malformed packets must be silently dropped.Here, the malformed packet stands for the packet that its PPPheader fields mismatch the provisioned values. Provided that weonly allow IP packet with the protocol number 0x0021 to pass,packets with different protocols other than 0x0021 are consideredas malformed packets. As a result, we can silently dropthe whole malformed packet and increase the malformed packetcounter by one.Many different protocols can be encapsulated in PPP suchas RTP/IP/PPP or UDP/IP/PPP for real-time video/audio applications,and TCP/IP/PPP [25] for non-real-time data applications.Those protocols are numbered according to IETF 1700[26] such as 0x0021 for IP, 0x8021 for the network negotiationand IP control protocol, and 0xC021 for the link control protocol(LCP). Because PPP can support multiprotocol encapsulation,we provide 12 programmable 16-bit registers, PPP-protocol, , to store the allowablePPP protocols which can be provisioned by different ISPs. (Thetwo commonly used protocols: IP (protocol number: 0x0021)and IP control protocol (protocol number: 0x8021) are definedas default). The “PPP detach” block checks the PPP header todetermine any malformed packet.Fig. 5. Different PPP formats: (a) packet with uncompressed PPP header and(b) packet with compressed PPP header.2) Optimize Bandwidth Usage: Due to the explosive growthof Internet traffics, bandwidth efficiency as a de facto is one ofthe most important design criteria of many network systems.There have been many approaches to achieve this design goal atmany different levels ranging from application down to physicallayer or from system down to chip design. Here, we provideour novel “PPP header compression” and “strip PPP header andCRC fields off” features to achieve bandwidth saving at the datalink level based on the system setup and the characteristics ofincoming traffics.a) PPP header compression: Except for the commonly useduncompressed PPP format for data transmission, we proposeour novel bandwidth efficient “PPP header compression”in our design. The uncompressed PPP formats isshown in Fig. 5(a). Here, the address field stands for thedestination to accept the packet (default value: 0xFF isfor all stations to accept the packet), and the control fieldstands for packet number (default value: 0x03 is for unnumberedpacket). Since PPP supports multiprotocol encapsulation,the protocol field defines the types of datatraffic in the payload such as IP, IPv6, AppleTalk, etc. Forvideo over IP, the PPP payload field contains the IP paradigmfor video service such as resynchronization marker,macroblock (MB) number, encoding mode, quantizationstep size, DCT and motion information, MB-based videotexture information, etc. To accommodate the contentbasedvideo coding in the third part of our paper, we alsoinclude the shape information of the associated video objectsin IP paradigm. The payload size of PPP is variable,up to some negotiated maximum size prior to a connectionsetup (default value: 1500 bytes). The checksum(CRC) is normally 2 bytes (CRC-16: )but a 4-byte CRC (CRC-32:)can be negotiated. Here, the CRC polynomial is gener-


8 IEEE TRANSACTIONS ON MULTIMEDIA, VOL. 4, NO. 1, MARCH 2002ated on the whole packet including the PPP header but notHDLC Flag, 0x7E. Unlike the conventional PPP design,to reduce the bandwidth usage, we introduce “PPP headercompression” or packets with compressed PPP header [asshown in Fig. 5(b)] in our design because the address andcontrol fields of point-to-point network are normally setto be 0xFF03. Therefore, those fields can be omitted (twoout of five PPP header bytes can be saved for each packet).We can check that the incoming packet is in the compressedor the uncompressed PPP header format based onthe setting of control register, PPP-header-check.b) Strip PPP header and CRC fields off: Because PPP supportsmultiprotocol encapsulation, we can classify the incomingdata stream embedded in the PPP packets intotwo groups: control signals and real datum. Those twotypes of traffics can be distinguished based on the protocolnumbers associated with the data packets.— Control signals: Prior to real data traffic flowingthrough the networks, PPP has to go through threebasic phases of negotiations: the LCP negotiation, theauthentication and link quality management, and thenetwork control protocol (NCP) negotiation [27]. Bothtransmitter and receiver need to negotiate and agree oncertain network parameters such as maximum packetsize, routing information (IP address), etc. Also, thosecontrol signals are embedded in PPP. Therefore, weare not allowed to strip those PPP header and CRCfields off to save bandwidth because they are criticalfor establishing the connections and not retrievableonce they are discarded.— Real datum: Once the connections established, thevideo bit-stream can be carried over PPP through thenetworks. The duration of network occupancy can lastfrom minutes to hours depending upon the applications.Furthermore, the PPP packet sizes vary frommedium to long (up to 64 kbytes) and are embedded inIP packets. Due to the reliability (BER below 10 )of optical fiber, both PPP header and CRC fields (4 upto 8 bytes) therefore can be stripped off from the IPpacket to save bandwidth. Based on the recent studiesof data traffics over Internet [28], nearly half of datastreams have the packet size of 40 up to 44 bytes. Withthe average IP packet size of 40 bytes, we can save upto 20% of bandwidth.C. The Scrambler Design for Network SecurityAs network security becomes increasingly more important,we include the self-synchronous [29] scrambler/unscramblerin our SONET network adapter design. This self-synchronizingscramblerworks like aninfinite impulse response (IIR) filter. The pseudorandom outputof the 43th registeris xored with the inputand is transmitted . The purpose of this scrambler is notonly to ensure an adequate number of transitions (zeros to onesand ones to zeros) for SONET clock recovery but also to randomizethe bits going out to the networks to prevent maliciousattacks. The scrambler operates continuously throughout thebytes of the synchronous payload envelope (SPE) [1] but bypassingthe SONET overhead bytes. On the top of includingself-synchronous in our design, there is another issue:“Where should we place the scrambler/unscrambler?” From theperspective of the SPE mapping, the placement of the scramblerbefore or after the HDLC stuffing block at the transmitside does not matter because the SPE mapping is transparent tothe SONET network. The agreement in the IETF standard is toplace the scrambler after the HDLC stuffing block for pragmaticreasons [2]. However, using this approach is not with its problems,because malicious users can transmit packets filled witheither the HDLC flag pattern (0x7E) or the escape sequences,i.e., 0x7D5E so that the link bandwidth is halved. This couldrender sophisticated quality of service (QoS) control mechanismsuseless [30]. Unlike the conventional design, we providethe great flexibility for ISPs to choose the location of the scrambler,which is indicated by “Payload-control register” at transmitside (00: No scrambler, 01: Post-scrambler, 10: Pre-scrambler,11: Undefined). Our default setting Payload-control [1:0]=“00”is for interworking with the old equipment design based on RFC1619 [2].Based on the previous discussion, in our improved SONETnetwork design, there are two levels of scrambling: the SONETscrambling and the packet scrambling (prescrambling andpostscrambling). Note that only the prescrambling can avoid(not definitely) malicious network attacks by sending in alot of7E/7Ds. As the counterpart of the scrambler, the unscramblerfunctioning as a FIR filter is placed in the mirrored position correspondingto the location of the scrambler.D. The Performance of SONET DeviceIn this part of the paper, we propose many new ideas in designingand implementing the SONET network adapter in amore flexible, reliable, and bandwidth efficient way. The clockrate of our single-chip design, as shown in Fig. 2, is 78 MHzs MHz for single OC-48 linkand four OC-12 links, or 20 MHzsMHz for four OC-3 lines. The calculation is based on thefollowing:each OC-48 frame, for instance, consists of 38 880 bytes(1440 bytes overhead 37 440 bytes payload) which areprocessed in parallel via four different logical data channelsin our design (the time span to process those bytes is125 s).In terms of data throughput rate, our design can support up to2.4 Gb/s.The latency of each block in the payload processor, shown inFig. 4, is listed in Table III. These result in negligible overallsystem latency, i.e., 324 ns in the transmit direction and 252 nsin the receive direction for each OC-48 connection.In our loopback bit error rate (BER) test (Data streamSONET network adapter at transmit side fiber SONETnetwork adapter at receive side Data stream), we observethat our design has negligible bit errors.Inaddition, we use the same loopback setup to perform the networksecurity test by maliciously sending alot of 7E/7Ds. Withthe two levels of scrambling, we can recover all sending bits.


CHEN AND LIU: JOINT SOURCE-CHANNEL MULTISTREAM CODING AND OPTICAL NETWORK 9TABLE IIILATENCY OF EACH BLOCK IN OUR DESIGNTherefore, the optical network is the leading expedient solutionfor providing reliable, high-speed video over IP service.III. JOINT SOURCE–CHANNEL ROBUST MULTISTREAMVIDEO TRANSMISSIONAs we have pointed out in the introduction, although it is idealto carry video over optical networks, the cost of providing theend-to-end optical connections is beyond the reach of the averageusers. Therefore, optical networks are most likely usedin the core networks connecting different ISPs. In addition, theubiquitous requirement of multimedia services excludes the useof optical networks. Therefore, some types of wireline or wirelessaccess networks (last-mile services) are most likely neededto connect the end-users to the backbone ISPs, as shown inFig. 1. However, any access networks may experience moreor less transmission errors. Transmission errors can be roughlyclassified into two categories: 1) random bit errors and 2) erasureerrors. How to provide an error-resilient transmission is achallenge for us, which leads to our work in the second part ofthe paper.The random bit errors are caused by the imperfections ofphysical networks, which result in bit inversion, bit insertion,and bit deletion. Depending on the coding methods andthe affected information contents, the impact of random biterrors can range from negligible to objectionable. When afixed-length coding is used, a random bit error only affects onecode word and the damage is generally acceptable. But if VLCi.e., Huffman coding, is used, random bit errors can cause theloss of synchronization so that many following bits are not decodeableuntil the next synchronization code word is received.In some cases, even after the synchronization is obtained,the decoded information can still be useless because there isno way for decoder to determine which spatial or temporallocations corresponding to the decoded information. Erasureerrors, on the other hand, can be caused by the packet loss inpacket networks, or system failures for a short time. Randombit errors in VLC can also cause erasure errors since a singlebit error can lead to many following bits being undecodeableand hence useless. The effect of erasure errors is much moredestructive than the random bit errors due to the loss or thedamage of a contiguous segment of bits. Since almost all thestate-of-the-art video compression techniques use VLC, there isno need to treat random bit errors and erasure errors separately.We therefore use the generic term “transmission errors” in thispaper to refer to both random bit errors and erasure errors.With the development of video segmentation techniques[13], we can view, access, and manipulate video objects ratherthan frames of pixels, which is used in MPEG-1, MPEG-2, andFig. 6. Illustration of the transport coder with transport prioritization and itsmultistream video over IP design. Here, the drawing is not proportional to thereal packet size.H.263 video coding systems. By taking the advantage of content-basedvideo coding, in this part of our paper, we proposea novel joint source-channel multistream video transmissionscheme to combat transmission errors in access networks.A. What is Unique in the Multistream Video Coding?In our design, the video objects are encoded into differentvideo streams, as shown in Fig. 6. Each video object is encodeddifferently based on its perceptual importance. The perceptualimportance of the video streams can be determined based on theoutputs of segmentation.• Easy case: For the specific applications where the inputsignals come from a controlled environment or where thefeatures to be analyzed are simple, the perceptual importanceof video streams can be easily determined. For example,the video sequences are created by combining aforeground object that has been filmed over a blue screenwith a background sequence that has been taken independentlysuch as “Akiyo” test sequence. In the sequence, wesimply refer theprimary video stream to the video streamof the foreground video object, i.e., the person “Akiyo,”and refer the secondary video stream to the stream of thebackground video object.• Difficult case: The determination of perceptual importancecan be difficult for generic images or videosequences because the perceptual importance of differentobjects dynamically varies at different occasions.Therefore, the determination depends very much on thesegmentation techniques used, which still have to be improved[13]. Our multistream scheme works based on the


10 IEEE TRANSACTIONS ON MULTIMEDIA, VOL. 4, NO. 1, MARCH 2002Fig. 7.Video quality of different coding schemes: (a) single stream and (b) multistream.assumption that we can segment the video sequence anddistinguish the significance of different video streams.On the top of traditional error control techniques, we proposea joint source-channel multistream video coding schemeto combat the transmission errors under the harsh networkconditions. The basic idea behind this multistream scheme isto send the same packets of the primary video streams twicein the consecutive order under the harsh network conditions.Most likely, these dual description packets end up goingthrough the different routes/paths of the networks. Because thechance that all paths simultaneously experience informationlosses is small, the decoder has better chance to reconstruct theoriginal video or to recover some useful information based onthe dual descriptions of primary video stream than the singledescription approach, as shown in Fig. 7. In this figure, allerrors are isolated as MB-errors because we adopt MB-basedscheme for the multistream design. In other words, insteadof grouping consecutive MBs in the scanning order, an MBbelongs to exactly one slice in our design (this type of design isquite common [31]). One reason why we adopt the MB-basedinstead of the slice-based scheme is that we try to minimizethe impact of transmission error propagation due to the harshnetwork condition and isolate the error only to the affectedMB. For example, in wireless communication, signal fadingdue to time-variant multipath propagation often causes thesignal to fall below the noise level, thus resulting in a largenumber of errors. As a result, the bursty transmission errormay render the consecutive video MB useless if we groupthem together. Therefore, we use MB-based coding and adoptthe MB-based IP packet instead of the slice-based IP packetfor transmission. The same argument also holds true for thewireline channels. For the wireline packet-switch networks,the routers i.e., CISCO routers are designed and configuredbased on the open shortest path first (OSPF) protocol for equalcost load balance. It routes the consecutive IP packets from thesame source via different paths to the destination. As a result,we try to keep the IP payload as small as possible (MB-basedversus slice-based). The other more important reason is that wetry to keep the repeated information as small as possible anduse the MB-based IP packet instead of the slice-based IP packetso that it is easier for us to keep the constant bandwidth in ourmultistream design. Furthermore, our multistream is designedto carry not only just bursty traffic, but also constant-bit-ratereal-time video traffic. A long IP payload would have a negativeimpact on this type of traffic flow. Therefore, our MB-baseddesign is also chosen for minimizing the system delay.A common question is: “The idea of using object-basedcoding for error-resilience is not new. It was one of themain reasons that MPEG-4 develops the object-based codingscheme. What is new in our design?” The conventional content-baseddesign codes the video into different video streamsand transmits each video stream only once no matter whetherthe channel condition is good or bad. However, our multistreamdesign distinguishes the primary video streams among theother video streams (the secondary video streams). We transmitthe video streams according to the significance of the videostreams under the adverse channel conditions. Sometimes,we may stop transmitting those secondary video streams andallocate their bandwidth resource to transmit the primary videostreams again. It is important to point out that, under thenormal channel condition, we still use the same approach asthe conventional single stream video approach by transmittingall video streams once regardless of the primary or secondaryvideo streams. Another common concern is: “In the case ofadverse network, it may increase the bandwidth requirementand worsen the network traffic condition without improving theQoS by throwing in more packets in the multistream design.”Instead of simply throwing in more packets in the communicationpipes and dynamically increasing the bandwidthrequirement, our multistream scheme keeps the same bandwidthrequirement by adjusting the quantization step, adoptingtransport prioritization and sometimes transmitting only theprimary video streams. The term transport prioritization hererefers to various mechanisms to provide different quality ofdifferent video streams in transport, including using unequalerror protection [31], [32] and assigning different priorities todifferent video streams. Our goal is to ensure that the criticalinformation can reach the decoder while still keeping the


CHEN AND LIU: JOINT SOURCE-CHANNEL MULTISTREAM CODING AND OPTICAL NETWORK 11same bandwidth requirement throughout the service becausethe bandwidth requirement is normally negotiated before thecommunication pipe is setup. During the multimedia service, itis impractical to expand the network bandwidth dynamically toaccommodate the transmission of multiple copies of the videostreams because it may affect the other users and only worsenthe network traffic condition without improving the QoS.Overall, the multistream scheme requires the same bandwidthas the conventional singlestream approach. Next, we willexplain our design in more detail.B. The Design of Multistream Video CodingThe video objects are lossily encoded so that it reduces thebit-rate by representing the original video using some transformedvariables, then applying quantization. In order to facilitatethe rate-control transmission, we apply tight control on theencoded bit-rate so that different video objects may have differentquantization step sizes [33], [34]. At the decoder side,the quality of the reconstructed video object is highly correlatedwith the quantizer step size used by the encoder. For instance,the larger the step sizes, the lower the bit-rate thus the lowerthe quality of the secondary video streams. The commonly usedvariable length entropy coder, i.e., Huffman coder is then employedafter quantization. After the entropy coding, the overheadinformation such as resynchronization markers, data partitioningmarkers, etc., as shown in Fig. 5, are attached to thevideo streams. It leads to a common question: “MPEG-2 alsohas video scalability mechanism built for error-resilience andrate-control transmission, why don’t we consider it?” The differencebetween our approach and the MPEG-2 design is thatwe manipulate video objects rather than the frame of pixels. InMPEG-2, all video objects within a frame are treated equallyimportant and the bandwidth is allocated evenly to transmit thecoded information. To keep the same bandwidth requirement,our multistream design can not only adjust the quantization steplike the MPEG-2 approach, but also adopt transport prioritizationby sacrificing the secondary video information for theprimary video information. The transport coder with transportprioritization refers to as an ensemble of devices performingchannel coding, packetization and/or modulation, as shown inFig. 6. For instance, the router will make the best effort to deliverthose high priority packets associated with the primaryvideo stream in congested networks. It results in less objectionablevisual distortion than MPEG-2 design especially when thesystem throughput rate is low. The “MUX controller” in Fig. 6controls whether the encoded IP packets are transmitted in eitherthe “normal mode” or “adverse network mode” based onthe network conditions. Here, the feedback information aboutthe network condition can be obtained by using the delay andloss-rate statistics at the decoder [11], [35]. Under the normalmode, those encoded IP packets are multiplexed and sent outalternatively. In other words, we transmit all video streams regardlessof the primary or secondary video streams similar tothe single stream design. When the network condition deterioratesthen “the MUX controller” switches to “adverse networkmode”, and the IP packets of the primary video stream are repeatedonce and then sent out, as shown in Fig. 6. Compared tothe processing of the primary video stream, the rest of secondaryvideo streams are more coarsely quantized, less error protectedcoded and transmitted at low priority. Providing that the motionof those secondary video objects is zero or low at that momentsuch as in the “Akiyo” test sequence, we can even stop transmittingthose background secondary video streams. The othercommon question is: “The retransmission is usually taken careof by the ARQ and TCP. Why should we design the multistreamvideo transmission?” The reason to introduce the multistreamvideo transmission is because sometimes it may not be feasibleor cost effective in the certain applications to guarantee the losslessvideo transmission. For instance, the real-time and interactivityrequirements exclude the deployment of some well-knownerror-recovery techniques such as the ARQ retransmission. Inaddition, issues such as audio-visual synchronization and multipointcommunications further complicate the problem of errorrecovery. As a result, the self-recovery transmission mechanism(or joint source-channel transmission mechanism) is more desirable.If the system delay is not an issue such as in non-real-timedata-oriented services, then the “resend” is taken care of by theTCP layer, and ARQ in the link layer or the media access control(MAC) layer. Under such a circumstance, we can send themultiple copies of the video streams regardless of the primaryor secondary video streams to the receiver to reduce the transmissionerrors under the harsh network conditions. However, itis not the case for the real-time video services because dynamicallychanging the overall system delay is not tolerant.Our multistream transmission scheme works based on thefact that there are several parallel paths/routes between thesource and destination. Each path may be temporarily downor suffer from bursty errors. However, the error events ofdifferent paths are independent so that the probability thatall paths simultaneously experiencing losses is small. Thesewireline/wireless paths can be physically distinct between thesource and destination. For example, in wireline packet-switchnetworks, the routers (like CISCO routers) are designed andconfigured based on the OSPF protocol for equal cost load balance.It routes the IP packets from the same source via differentpaths to the destination. In addition, it intelligently redirects thetraffic flow once adverse channel condition encountered insteadof keeping throw the packets in the already congested path.Therefore, our multistream design outperforms the conventionaldesign under the harsh network condition because the chanceof the dual descriptions of the video streams via different pathssimultaneously experiencing information loss is small. As forthe wireless communication, the mobile is moving during thetransmission. Therefore, the channel condition at time mayvery well be different than that at time . Furthermore, thesignals can be deflected by surrounding infrastructures. Underthe multipath fading condition, there are many paths betweentransmitter and receiver. In our multistream design, the dualdescriptions of video streams at time and independentlytravel through the different paths thus introducing the diversity.Because the chance of all paths simultaneously experiencinginformation loss is small, the sum of signal level at the receiverantenna in our multistream design is stronger than that in thesingle stream design; thus achieving better picture quality. Evenwhen only one single physical path exists between the sourceand destination, the path can be divided into several virtual


12 IEEE TRANSACTIONS ON MULTIMEDIA, VOL. 4, NO. 1, MARCH 2002Fig. 8.Transmission errors affect the video quality by using different coding schemes.channels by using the time interleaving scheme, as shownin Fig. 6. Overall, under the adverse channel conditions, thedecoder can still reconstruct the original video sequence orrecover some useful information depending on the stream whichis received correctly or least distorted. The resolution of thereconstructed video at the decoder side is gracefully improveddepending on the number of different descriptions receivedcorrectly. The other question is: “What is the difference betweenour design and the conventional multiple-description coding?”Unlike the conventional multiple-description coding [36], [37],we apply the content-based video coding instead of the layeredcoding. In the layered video coding approach, all video objectswithin a frame are treated equally important and the bandwidthis allocated evenly to transmit those coded information. On theother hand, in content-based video coding, video information ispartitioned into more than one video object and encoded intomultiple video streams. Combining different video streamswith different transport priorities, our emphasis is on the robusttransmission of the primary video streams. Under the harshnetwork condition, we allocate more resource to error protectthose important video information. In principle, we trade thequality of the least significant video objects for the quality ofthe most significant ones. As a result, we can reconstruct a closeapproximation of the original video (at least for the primaryvideo object) and make the output video at the decoder leastobjectionable to human eyes.C. Simulation ResultsTo test the performance of our proposed multistream techniqueto combat the transmission errors in the hybrid accessnetworks, we performed several experiments by taking someMPEG-4 test sequences in CIF or QCIF formats as input (Eachsequence contains 300 video frames and we assume that the significanceof different video objects is known in advance). Thesegmented video objects in each test sequence are differentlycoded into different video streams using our novel compresseddomain content-based video coding scheme proposed in SectionIV. The video streams are then quantized and transportedwith transport prioritization to achieve the heterogeneous videoqualities for different video objects (the primary and secondaryvideo streams). The syntax of both the YUV MB and its associatedbinary alpha block (BAB) supports the communicationof information relating to the decoded video quality. Furthermore,the same quantization functionality for the YUV block isalso applied to the compressed BAB. In our experiments, thequantization parameters and channel coding rates are chosenso that the combined output rates of video streams remain constantthus the same required bandwidth throughout the service.


CHEN AND LIU: JOINT SOURCE-CHANNEL MULTISTREAM CODING AND OPTICAL NETWORK 13Fig. 9.Simulation setup for testing the performance of our multistream video coding scheme under the wireless channel conditions.Fig. 10.Performance comparison between our proposed multistream design and the conventional single stream design.Due to our joint source-channel coding design, the rate-compatiblepunctured convolutional (RCPC) [38] code is employed forchannel coding in our simulations. It provides an efficient meansin implementing a variable-rate error control for different videostreams so that only a single encoder/decoder pair is needed. Inaddition to that, the orthogonal frequency division multiplexing(OFDM) is used for modulation because it increases the robustnessagainst frequency selective fading or narrow-band interference(For detailed information about OFDM design, pleaserefer to [39]). At the decoder side, the soft-decision Viterbi decodingscheme is utilized to improve the error-correction capabilitiesbecause it outperforms the hard-decision one by 2 dB.We test the performance of our proposed scheme under both thewireline and wireless access channels. Our simulations are performedon the added white Gaussian noise (AWGN) channel.In addition, for our wireless channel simulation, we adopt theRayleigh fading channel to model the bursty channel conditioncaused by multipath fading [40]–[42] (Under the urban transmissionenvironment, there are up to six different paths/rays inour multipath fading model, which is the same as the commonlyused urban channel models).In our wireline simulation, we assume that there are twodistinct routes between the source and destination. The router,which acts like the CISCO router, at the encoder side distributesthe video IP packets via the two paths independently. The transmissionerrors affect the video quality by using different codingschemes, as shown in Fig. 8. For the simple cases such as“Akiyo” and “Mother and Daughter” test sequences, wherethe foreground scenes change while the background scenesremain the same, we can allocate all bandwidth under the harshnetwork condition to transmit the primary video object, becauseit dominates the whole video scene. Our multistream worksextremely well for those simple test cases. For the complicatecases such as the “Coastguard” test sequence in which boththe foreground and background scenes change, we allocatemore bandwidth to transmit the primary video streams thanthe secondary ones when the network condition deteriorates.Compared to the conventional layered coding mechanism, ourmultistream works better in terms of error resilience. From thesimulation results, we observe that the multistream approachoutperforms the single stream approach by 3–5 dB under thenoisy channel conditions.Under the wireless multipath fading condition (we use thecarrier frequency of 900 MHz, the mobile velocity of 10–20mi/h with up to six propagation paths and the system throughputrate is around 128 kb/s), our simulation with the setup, as shownin Fig. 9, demonstrates that our multistream works better thanthe single stream design by up to 5–7 dB, as shown in Fig. 10.


14 IEEE TRANSACTIONS ON MULTIMEDIA, VOL. 4, NO. 1, MARCH 2002Fig. 11.Our fully DCT-based coder structure. Here, TD-ME stands for estimates motion in the transform domain.Fig. 12.Padding technique is applied to the VOP of “News” sequence: (a) repetitive padding of VOP and (b) extended padding of VOP.The reason is that the IP packets of primary video streams are repeatedonce and then sent out, thus introducing the signal diversityunder the harsh network condition. As a result, the dual descriptionsof primary video stream independently travel throughthe different fading paths. Because the chance that all paths simultaneouslyexperience information loss is small, the sum ofthe signal level at the receiver antenna in our multistream designis stronger than that in the single stream design (In our design,the OFDM is used to increase the robustness against thefrequency selective fading or the narrow-band interference). Asa result, the decoder can reconstruct a better original video sequencedepending on the stream which is received correctly orleast distorted by taking advantage of the multiple paths betweensource and destination.We also perform tests to compare our multistream designwith the MPEG-2 and multidescription rate-control, error-resilientcoding schemes. Under the adverse channel condition,we transmit every frame instead of the primary video streamstwice. Provided that the system throughput is low kb/s asin video phone applications, we can only use the coarse quantizationstep to keep the constant bandwidth during the service.Based on the simulation results, the coarse quantization causesmore objectionable visual distortion. Therefore, to support ourmultistream video coding scheme, it is better to adopt the transportprioritization and to access and manipulate video objectsrather than the frame of pixels.IV. CODING OF ARBITRARY SHAPE VIDEOFULLY IN THE TRANSFORM DOMAINIn the second part of this paper, we have discussed thejoint source-channel multistream coding. Now, the questionbecomes “What content-based source coding scheme should weuse?” Surely, we can adopt the MPEG-4 content-based videocoding design to meet our multistream design requirement.However, in this part of our paper, we propose “Arbitrarilyshaped video coding in the DCT-domain,” instead. Instead oftreating the moving picture as a single entity in the MPEG-1and MPEG-2 design, our multistream design treats the movingpicture as an organized collection of visual objects and encodesthem into different video streams. Furthermore, an imageor a video sequence may be constructed by the compositionof one or more independently coded visual objects. Unlikethe conventional MPEG-4 arbitrarily shaped visual codingapproaches performing the compression in DCT-domain andthe motion estimation/compensation in the spatial domain, itis worth pointing out that we achieve our content-based videocompression, motion estimation, and compensation all in theDCT-domain. The advantage of such an approach is that wecan optimize the hardware complexity by combining both DCTand motion estimation/compensation units into a single unit,thus reducing the computational units along the data path.A. Compressed Domain Content-Based Video CodingBesides the spatial domain motion estimation, manipulatingvideo data in the DCT-domain has actually long been recognizedas more efficient in many advanced video applications[43]. In [43], Chang and Messerschmitt derive a complete setof algorithms for the manipulations of compressed video signalssuch as overlap translation, scaling, linear filtering, rotation, andpixel multiplication in the DCT-domain. In addition, they proposethe idea of performing compression using DCT withoutmotion compensation (MC) in the spatial domain. However,


CHEN AND LIU: JOINT SOURCE-CHANNEL MULTISTREAM CODING AND OPTICAL NETWORK 15TABLE IVMACROBLOCK-BASED REPETITIVE PADDING PROCEDURESTABLE VCOMPUTATIONAL COMPLEXITY OF STEP 2.1–2.5 IN OUR DESIGN FOR A MACROBLOCK SIZE OF N 2 N (HERE WE USE N =16AS AN EXAMPLE)nothing is mentioned about how to achieve the combined designof DCT and motion estimation units, which consume mostof the computing power in a video coder. In [19], Koc and RayLiu propose a detailed framework concerning how to performmotion estimation in the DCT-domain. The scheme is proposedas an alternative cost-effective solution to outperform the thenstate-of-the-art coding standards such as H.261, MPEG-1, andMPEG-2 for rectangular frame video coding. Nothing about thearbitrarily shaped video coding is mentioned.In recent years, many communication/interactive applicationshave appeared on the Internet. Unlike MPEG-1 and MPEG-2,wherein the emphasis is primarily on coding efficiency, the content-basedvideo coding is the main theme of MPEG-4. Althoughthe new or modified techniques such as the shape codingare introduced in MPEG-4, the DCT and block-based motionestimation are still the fundamental techniques in the standard.The difference between our work in this part of paper and that in[19] bears some analogy to the difference between video codingscheme in MPEG-4 and that in MPEG-1 and MPEG-2. In otherwords, the DCT pseudophase technique is still the kernel ofour design. Our fully DCT-based coder structure is shown inFig. 11. We combine both the DCT coder and the motion estimation/compensationprocessors into one unit, and remove theinverse IDCT unit from the feedback loop. All cost-effective,high throughput advantages (motion estimation/compensationin the DCT-domain versus the MPEG approaches) mentionedin [19] are also applied in our design. In fact, we can view themotion compensated video coding of rectangular frames in theDCT-domain as the special case of our compressed domain arbitrarilyshaped video coding. In what follows, we will focus onthe difference between our newly proposed scheme and that in[19].1) MB-Based Repetitive Padding: The MB-based repetitivepadding is required to estimate motion for the contour MBswhich reside on the boundary of the video object and containpartial video information, as shown in Fig. 12. The proceduresof padding are listed in Table IV.2) Binary Shape Coding: The arbitrary shape of the videoobjects can be described in either the binary or gray mode. In thispaper, we are only interested in the binary mode because it formsthe simplest class of objects. The video objects and shapes areseparately coded and the suitable shape coding methods include:1) vertex-based coding [44] and chain coding [45] in contour-basedapproaches;2) modified-READ (MR) method [46] and context-basedarithmetic encoding (CAE) [47] in bitmap-based approaches;3) Chroma-keying [48].


16 IEEE TRANSACTIONS ON MULTIMEDIA, VOL. 4, NO. 1, MARCH 2002TABLE VISUMMARY OF ARBITRARY SHAPE VIDEO CODING IN THE DCT-DOMAINIn our design, we select the block-based methods of CAE for ourshape coding. The adopted block-based syntax has allowed thecompressed BABs to be blended seamlessly into the traditionalvideo syntax structures, as shown in Fig. 5. This in turn easesthe task of supporting the important features such as the error-resilient,bit-allocation, and rate-control operations [33], [34]. Justas with the YUV encoding, a BAB may be intra-coded usingthe CAE. Furthermore, it may be inter-coded using MC andCAE, or it may merely be reconstructed by the MC without CAEwhich analogous to the “not-coded” MB mode in the MPEG-4video standard. Both the YUV and binary alpha decoding requirethe use of motion estimation/compensation to exploit thespatial redundancy.3) Arbitrary Shape Video Coding Algorithm: In terms of arbitraryshape motion estimation, we can treat the contour MBssame as the regular ones except for pixels outside are paddedbased on the video content inside the video boundary by followingthe procedures in Table IV. In what follows, the videoobject plane (VOP) refers to the instances of video objects at agiven time. Because our content-based video coding is the extensionof the work in [19], please refer to [19] for the detailedexplanation of the equations and definitions such as the “TypeI” and “Type II” DCT, , , etc. Our proposed scheme is summarizedin Tables VI and VII.Now let us take a look at the overall computational complexity.To process each VOP, Step 1 of the proposed approach(VOP formation and padding) is only needed to be executedonce. Therefore, the overall computational complexity of ourdesign is determined by the complexity of Steps 2.1–2.5, whichserves as the computing engine of the whole design. The com-


CHEN AND LIU: JOINT SOURCE-CHANNEL MULTISTREAM CODING AND OPTICAL NETWORK 17TABLE VIISUMMARY OF ARBITRARY SHAPE VIDEO CODING IN THE DCT-DOMAIN (CONTINUATION)plexity of each step is listed in Table V. The reason we call ourdesign the content-based video coding is that the motion estimation/compensationis performed only for those MBs containingthe video information. Overall, the scheme requires thecomputational complexity of . Here, stands for theMB size. As for our fully DCT-based design versus the conventionaldesign in terms of computation saving, we indeed haveperformed some measurements. Compared to the fully searchblock matching scheme, our DCT-based design can save up to60–75% computation depending upon the characteristic of incomingvideo sequence and other factors. (Although there aremany fast search motion estimation schemes, we are not in theposition to compare our design against all other approaches becauseit is not the main theme of this paper). For the large motionsgoing beyond the block boundary, we will use motionvector (0,0), instead.To facilitate the explanation of our proposed scheme, let ususe “News” in CIF format with the frame size ofasinput video sequence. The panorama scene is shown in Fig. 13.The “News” sequence consists of four VOPs and three correspondingbinary alpha planes (the background VOP has no alphaplane). Here, we apply our design to , the third VOP inFig. 14(a), as an example to illustrate our design because it is theforeground VOP and most importantly the location and shapeof , as shown in Fig. 14(b), vary with time. The reason tointroduce the VOP formation, which is the first step of our design,is to achieve high data compression rate because we do notneed to estimate motions for those MBs containing no video information.After the VOP formation, the is now boundedby the tightest rectangle containing the video object. However,this tightest rectangle may not consist of multiples of MB of sizeTherefore, we need to extend the bottom right coordinateof window in Fig. 14(a) to satisfy that requirement.The final bounded with the window size ofFig. 13.Panorama scene of “News” in CIF format.and its corresponding alpha plane are shown in Fig. 14(c) and14(d), respectively. Then, the bounded VOP is padded, as shownin Fig. 12. The bounded VOP window is further divided intononoverlapped MBs. The motion estimation process starts fromthe top left MB in the bounded VOP window to the top right one,and then to the next row, and so on for every MB in the boundedVOP window. The transform domain nature of the approach enablesus to directly extract motion vectors out of the consecutiveVOPs of a moving object. To envision how it works, we presentan example to estimate motion of a contour MB by followingSteps 2.1–2.4 in our design, as shown in Fig. 15. The peak positionamong and indicates the integer-pel motionvector of (3, 2). The peak position among andimplies the half-pel motion vector of (3.5, 2.5).


18 IEEE TRANSACTIONS ON MULTIMEDIA, VOL. 4, NO. 1, MARCH 2002Fig. 14. VOP formation on VOP of “News”: (a) VOP ; (b) alpha plane of VOP ; (c) VOP in bounded window; and (d) shape of bounded VOP .After motion estimation, the current block of sizein the current frame can be best predicted by the blockdisplaced from the previous block position with the estimatedmotion vector . Based on the derivation in [21], [43],the DCT of the motion-compensated residual in terms of thedisplaced block difference (DBD) is given byIn other words, the DCT of the motion-compensated residualcan be expressed as the difference between the DCT of the displacedblock and the DCT of the current block. As a result,we can perform MC in the DCT-domain, as shown in Fig. 11,which serves the purpose of building a fully DCT-based motioncompensated video without converting back to the spatial domain.Most importantly, we can efficiently combine the designof the DCT coder and the DCT-domain motion estimation/compensationunit into one unit thus significantly reducing the hardcomplexity.Now the question becomes: “How do we extract the displacedDCT block in the DCT-domain or how to compute?” For the integer pixel and subpixel MC inDCT-domain, we adopt the bilinear interpolation methodproposed in [21], [43].B. Simulation ResultsSimulations have been performed on the “News” sequencein the CIF format. The bounded previous and current VOPs areshown in Fig. 16(a) and (b), respectively. The reconstructedVOP using our proposed compressed domain coding schemeis shown in Fig. 16(d). The simulation results demonstratethe comparable video quality between the reconstructed andcurrent VOPs. Considering that the motion compensated videocoding of the rectangular frame is the special case of our arbitrarilyshaped video coding, it is easy to see that our proposeddesign backward compatible to code regular images. In otherwords, we can view our approach as a logical extension ofthose DCT-based motion estimation schemes [19], [20] towardcoding video sequences of arbitrary shape.Due to its lower computational complexity as comparedto other difference measures, the sum of absolute difference(SAD) is adopted in the MPEG-4 standards to measurethe prediction errors [14], [15]. Simulations have also beenperformed to compare our design with the modified blockmatching (or polygon matching) method used in MPEG-4 interms of prediction errors. Here, the MPEG-4 video referencesoftware (MoMuSys) is used as reference in simulating theperformance of modified block matching approach. The resultsare shown in Fig. 17. Furthermore, the simulation resultsdemonstrate that the comparable performance of both our de-


CHEN AND LIU: JOINT SOURCE-CHANNEL MULTISTREAM CODING AND OPTICAL NETWORK 19Fig. 15.Motion estimation of a contour MB by following Steps 2.1–2.4 in our compressed domain design.sign and the one used in MPEG-4 in terms of prediction errors.Compared to the conventional arbitrarily shaped videocoding design, we optimize the hardware complexity by minimizingthe computational units along the data path thus morecost effective.Other than the “News” test sequence, the simulations arealso performed for “Foreman” and “Mother and Daughter” sequences,etc. In order to show that our design is also backwardcompatible to handle the rectangular frame of video, here wetreat “Mother and Daughter” sequence as the regular frame ofpixels. The simulation results, as shown in Fig. 18, demonstratethe comparable video quality between our compresseddomain design and the conventional approach used in videostandards.


20 IEEE TRANSACTIONS ON MULTIMEDIA, VOL. 4, NO. 1, MARCH 2002Fig. 16. Illustration of the performance of our content-based video coding: (a) bounded previous VOP; (b) bounded current VOP; (c) bounded alpha plane forprevious VOP; and (d) reconstructed VOP using our proposed design.consider video compression and delivery schemes based on thenetwork alternatives, capacities, and characteristics. This paperdiscusses three subtopics.1) A SONET network adapter design served as the Layers 1and 2 (the physical and data link layers) IP router to deliverpacket video over SONET/SDH or directly over fiberto provide reliable, high-capacity backbone connections.2) A joint source-channel multistream video coding schemeto combat the transmission errors in hybrid access networksunder the harsh network conditions.3) Compressed domain content-based video coding to reducethe hardware complexity and to improve the encoderperformance.Fig. 17. Comparing the performance between MPEG-4 and our videocoding approaches in terms of prediction errors using “News” testingsequence. Here, total sum of absolute differences is the summation of SADfor all MBs within each frame: (a) “Foreman” using MPEG-4 MoMuSysreference software; (b) “Foreman” using our design,; (c) “Mother andDaughter” using MPEG-2; and (d) “Mother and Daughter” using our design.V. CONCLUSIONTo optimize the performance of the multimedia over IPsystems with the given QoS requirements, we should jointlyAlthough these three subtopics can be separately presented, wetry to put them together as a system level paper. The reason isthat, under the current communication environment, the videoover IP services most likely go through hybrid networks (corenetworks, access networks) with different characteristics. Ourdesign targets at the different problems associated with hybridnetworks, i.e., robustness transmission in access networks,multivendor interoperability in the design of backbone devices,etc. Also, we present some novel thoughts in designing andimplementing video over IP systems. The simulation resultshave confirmed our ideas. Overall, our goal is to improve thequality of video over IP service under the current communicationenvironment.


CHEN AND LIU: JOINT SOURCE-CHANNEL MULTISTREAM CODING AND OPTICAL NETWORK 21Fig. 18.Comparing the video quality of MPEG-2 and our video coding approaches in terms of video quality.REFERENCES[1] M.-C. Chow, Understanding SONET/SDH: Standards and Applications:Andan Publisher, 1995.[2] “PPP over SONET/SDH,”, IETF RFC 1619, 1994.[3] T. C. Kwok, “PPP over ATM,” IEEE Commun. Mag., vol. 37, pp. 84–89,May 1999.[4] M. Ghanbari, “Two-layer coding of video signals for VBR networks,”IEEE J. Select. Areas Commun., vol. 7, pp. 801–806, June 1989.[5] Y.-Q. Zhang, Y. J. Liu, and R. L. Pickholtz, “Layered image transmissionover cellular radio channels,” IEEE Trans. Veh. Technol., vol. 43, pp.786–796, Aug. 1994.[6] M. Khansari and M. Vitterli, “Layered transmission of signals overpower-constrained wireless channels,” in Proc. IEEE Int. Conf. ImageProcessing, Washington, DC, Oct. 1995, pp. 380–383.[7] Q.-F. Zhu, Y. Wang, and L. Shaw, “Coding and cell loss recovery forDCT-based packet video,” IEEE Trans. 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22 IEEE TRANSACTIONS ON MULTIMEDIA, VOL. 4, NO. 1, MARCH 2002[46] F. McConnell, D. Bodson, and R. Schaphorst, FAX: Digital FacsimileTechnology and Applications. Norwood, MA: Artech House, 1992.[47] G. Langdon and J. Rissanen, “Compression of black-white images witharithmetic coding,” IEEE Trans. Commun., vol. COM-6, pp. 158–67,June 1981.[48] T. Chen, C. Swain, and B. Haskell, “Coding of subregions for contentbasedscalable video,” IEEE Trans. Circuits Syst. Video Technol., vol. 7,pp. 256–260, Feb. 1997.Jie Chen (S’95–M’97–SM’02) received the B.S.degree in electrical engineering and the M.S. degreein physics from Fudan University, Shanghai, China,in 1987 and 1990, respectively, and the M.S. andPh.D. degrees in electrical engineering from theUniversity of Maryland, College Park, in 1992 and1998, respectively.He is currently a Principle System Engineer ofFlarion Technologies, a Bell Lab spin-off start-upcompany, working on digital audio broadcastand wireless broadband network. He was withBell Laboratories, Lucent Technologies, Murray Hill, NJ, in research anddevelopment of optical network integrated circuits. In addition, he also servedas a member of the ITU-T working group for developing the third-generationCDMA2000 standard. His research interests span the broad aspects of computerengineering and microelectronics (with emphasis on high-performance,low-power architecture, CAD and VLSI design) with the applications forwireless communication, networking (with emphasis on computer and opticalnetworking, and network management) and multimedia signal processing (withemphasis on image and video processing). He is the author of book Designof Digital Video Coding Systems: A Complete Compressed Domain Approach(New York: Marcel Dekker 2001). He has published over 20 refereed journaland conference papers and holds four U.S. patents.Dr. Chen has received numerous awards, including the Bell Lab divisionaward for contributions in the optical network device design, the Hughes divisionaward for contributions in multimedia over wireless local-loop systemdesign, and the student paper award of the International Conference on Circuitsand Systems’97 (ISCAS’97). He serves as an Associate Editor for IEEETRANSACTIONS ON MULTIMEDIA, an Editor for EURASIP Journal of AppliedSignal Processing, and was a Guest Editor for a Special Issue of Multimedia overIP of IEEE TRANSACTIONS ON MULTIMEDIA, chaired several special sessionsin the international conferences, and held tutorial session of “UnderstandingEmerging and Future Communication Technologies” at ISCAS’2001. He currentlyserves as the member of both multimedia and video technologies committeesfor IEEE Circuits and Systems Society.K. J. Ray Liu (S’86–M’86–SM’93) received the B.S.degree from the National Taiwan University, Taipei,Taiwan, R.O.C., and the Ph.D. degree from the Universityof California, Los Angeles, both in electricalengineering.He is Professor of the Electrical and ComputerEngineering Department and Institute for SystemsResearch, University of Maryland, College Park.He was a Visiting Professor at Stanford University,Stanford, CA; IT University, Copenhagen, Denmark;and Hong Kong Polytechnic University, Hong Kong.His research interests span broad aspects of signal processing architectures;multimedia signal processing; wireless communications and networking;information security; and bioinformatics, in which he has published over 230refereed papers, of which over 70 are in archival journals.Dr. Liu is the recipient of numerous awards, including the 1994 NationalScience Foundation Young Investigator, the IEEE Signal Processing Society’s1993 Senior Award (Best Paper Award), the IEEE Vehicular TechnologyConference Best Paper Award, Amsterdam, 1999. He also received the GeorgeCorcoran Award in 1994 for outstanding contributions to electrical engineeringeducation and the Outstanding Systems Engineering Faculty Award in 1996in recognition of outstanding contributions in interdisciplinary research, bothfrom the University of Maryland. He is Editor-in-Chief of EURASIP Journalon Applied Signal Processing, and has been an Associate Editor of IEEETRANSACTIONS ON SIGNAL PROCESSING, a Guest Editor of Special Issues onMultimedia Signal Processing of PROCEEDINGS OF THE IEEE, a Guest Editorof a Special Issue on Signal Processing for Wireless Communications of IEEEJOURNAL OF SELECTED AREAS IN COMMUNICATIONS, a Guest Editor of aSpecial Issue on Multimedia Communications over Networks of IEEE SIGNALPROCESSING MAGAZINE, a Guest Editor of a Special Issue on Multimedia overIP of IEEE TRANSACTIONS ON MULTIMEDIA, and an Editor of the Journalof VLSI Signal Processing Systems. He currently serves as the Chair ofMultimedia Signal Processing Technical Committee of IEEE Signal ProcessingSociety and the Series Editor of the Marcel Dekker series on Signal Processingand Communications.

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