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    IEEE TRANSACTIONS ON WIRELESS COMMUNICATIONS, VOL. 4, NO. 6, NOVEMBER 2005 2777

    Transmission of Adaptive MPEG Video OverTime-Varying Wireless Channels: Modeling

    and Performance EvaluationLaura Galluccio, Giacomo Morabito, Member, IEEE , and Giovanni Schembra

     Abstract—Wireless channels are characterized by high time-varying bit-error rates (BERs). To cope with this problem, severaladaptive forward-error-correction (AFEC) schemes have beenproposed in the literature. They work locally at the wireless link,adding a variable amount of redundancy to the transmitted data inorder to maintain the packet error rate below an acceptable level.However, when such schemes are utilized, the bandwidth offered tothe applications changes when channel conditions change. In thispaper, the effects of these bandwidth variations are investigated

    in the case of real-time Motion Picture Experts Group (MPEG)video transmission. The MPEG encoder is controlled in order toadapt its emission rate to the current bandwidth offered by thewireless link. To this end, the encoding quality is diminished bythe source rate controller when the transmission rate has to bedecreased due to an increase in the channel BER, whereas it isimproved when the transmission rate can be increased due to adecrease in the channel BER. A Markov-based model, denoted asSBBP/SBBP/1/ K , has been introduced to model the scenario beingconsidered. The analytical framework allows evaluation of theperformance of the system and can be used to optimize the designof a video transmission system for wireless channels, providing theinstruments to derive the tradeoff between information corruptionin the wireless channel and MPEG video encoding quality.

     Index Terms—Forward error correction (FEC), Motion PictureExperts Group (MPEG), quality of service (QoS), switched batch

    Bernoulli process (SBBP), wireless channels.

    I. INTRODUCTION

    THE NEED for supporting multimedia applications in

    dynamic environments where users are equipped with

    wireless terminals is one of the most challenging research

    topics today. In fact, it is known that wireless channels are

    characterized by bit-error rates (BERs) that are several orders of 

    magnitude higher than the corresponding values for terrestrial

    networks. Accordingly, data packets may arrive at their desti-

    nation corrupted, thus becoming useless.

    To overcome this problem, one of the solutions most widely

    adopted today is using forward error correction (FEC). FEC

    algorithms introduce a chosen amount of redundancy: the

    Manuscript received August 15, 2003; revised September 3, 2004; ac-cepted September 13, 2004. The editor coordinating the review of thispaper and approving it for publication is V. K. Bhargava. The work of L. Galluccio and G. Morabito was supported by Ministero dell’Istruzione,dell’Università e della Ricerca (MIUR) under contract VICOM. The work of G. Schembra was supported by MIUR under contract TANGO.

    The authors are with the Dipartimento di Ingegneria Informatica e delleTelecomunicazioni (DIIT), University of Catania, 95124 Catania, Italy (e-mail:[email protected]; [email protected]; [email protected]).

    Digital Object Identifier 10.1109/TWC.2005.858028

    higher the BER, the higher the amount of redundancy intro-

    duced. However, in wireless channels, the BER is characterized

    by high time variability: There are periods when channel

    conditions are good, that is, the BER is low, and periods when

    channel conditions are bad, that is, the BER is high. In order to

    maintain a high level of resource efficiency while guaranteeing

    the information accuracy required by applications, several

    adaptive FEC (AFEC) schemes have been introduced in the

    recent past [1], [2], [6], [7]. According to these schemes, the

    amount of redundancy at any time depends on the channel

    conditions being low if channel conditions are good, and high

    if channel conditions are bad. One consequence is that AFEC

    schemes cause variations in the bandwidth offered to user

    applications, which therefore have to adapt their output rate

    accordingly.

    This paper focuses on video applications that are destined

    to become very common in wireless-communication scenarios.

    More specifically, the target of the paper is the definition of 

    an analytical framework for the design of a real-time Motion

    Picture Experts Group (MPEG) video transmission system over

    a wireless link that applies AFEC to keep the packet corrup-tion probability acceptable, i.e., below a given threshold. The

    MPEG encoder uses a rate controller that adapts the output

    rate by appropriately setting the quantizer scale parameter

    (QSP) [8], [12], [29] to follow the bandwidth variations, while

    maximizing encoding quality and stability. In order to achieve

    this target, the rate controller monitors the activity of the frame

    that is being encoded, its encoding mode, and the number of 

    bytes used to encode the previous frames. Then, it chooses the

    appropriate QSP in such a way that the transmission buffer at

    the sender site never saturates, even during periods with low

    available bandwidth. The whole system can be modeled by an

    emission process that feeds the transmission buffer. The serverof this buffer behaves according to the channel conditions

    estimated by the adaptive error controller: The serving rate

    is higher when channel conditions are good and lower when

    channel conditions are bad.

    Switched batch Bernoulli processes (SBBPs) are used to

    model both the MPEG source [4], [15], [17], and the server

    process of the transmission buffer that coincides with the time-

    varying bandwidth available in the wireless channel [20], [24]–

    [28]. Accordingly, an SBBP/SBBP/1/ K  model is introduced to

    describe the whole system.

    The analytical framework proposed in the paper is used to

    evaluate the performance in terms of the distortion introduced

    1536-1276/$20.00 © 2005 IEEE

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    2778 IEEE TRANSACTIONS ON WIRELESS COMMUNICATIONS, VOL. 4, NO. 6, NOVEMBER 2005

    Fig. 1. Mobile terminal system architecture.

    by the quantization mechanism in the encoding process, which

    are the loss and mean delay in the transmission buffer, at

    different target packet error probabilities (PEPs) achieved usingAFEC. Results obtained in the paper can be used to obtain

    the best tradeoff between encoding quality, which requires a

    high available bandwidth, and information correctness at the

    destination, which requires a high level of redundancy, thus

    causing bandwidth reduction.

    The rest of the paper is organized as follows. Section II

    describes the wireless MPEG transmission system considered

    in this paper. Section III proposes an analytical framework of 

    the whole video transmission system, accounting for both the

    video source and the transmission channel. Section IV provides

    a derivation of the performance parameters. Section V applies

    the analytical framework to a case study in order to demon-

    strate the model’s capability of providing performance insights

    for the system design. Finally, Section VI concludes the paper.

    II. DESCRIPTION OF THE S YSTEM

    The architecture of the video transmission system in the

    mobile terminal considered in this paper is shown in Fig. 1.

    The adaptive rate source is an adaptive-rate MPEG video source

    over a User Datagram Protocol (UDP)/IP protocol suite. The

    video stream generated by the video source is encoded by

    the MPEG encoder according to the MPEG video standard

    [30], [31]. In the MPEG encoding standard, the frame, which

    corresponds to a single picture in a video sequence, is the basicdisplaying unit. Three encoding modes are available for each

    frame: intraframes (I), predictive frames (P), and interpolative

    frames (B). The basic idea behind MPEG video compression

    is to remove spatial redundancy within a video frame andtemporal redundancy between successive video frames. The

    encoder output is a deterministic period sequence in which the

    period is a group of pictures (GoPs) realized with three types of 

    encoded frames.

    1) I frames coded using only information present in the

    picture itself in order to provide potential random ac-

    cess points in the compressed video sequence. The cod-

    ing is based on the discrete-cosine transform according

    to the joint photographic experts group (JPEG) coding

    technique.

    2) P frames coded using a coding algorithm similar to the

    one used for I frames, but with the addition of motion

    compensation with respect to the previous I or P frame

    (forward prediction).

    3) B frames coded with motion compensation with respect

    to the previous I or P frame, and the next I or P frame, or

    an interpolation between them (bidirectional prediction).

    Typically, I frames require more bits than P frames, while B

    frames have the lowest bandwidth requirement.

    In encoding each frame, it is possible to tune the number

    of bits needed to represent the frame and, thus, its quality, by

    appropriately choosing the so-called QSP. Its value can range

    within the set [1, 31]: 1 being the value giving the best encodingquality but requiring the maximum number of bits to encode the

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    frame, and 31 being the value giving the worst encoding quality,

    but requiring the minimum number of bits.

    The QSP can be dynamically changed according to the

    feedback law implemented by the rate controller in order to

    achieve a given target. The MPEG encoder emits one frame

    every   ∆   seconds, and its output is packetized in the packe-

    tizer according to the UDP/IP protocol suite: the packetizerfragments the information flow into blocks of  U P bytes

    1; these

    blocks constitute the payloads for the UDP, which adds a header

    of 8 bytes; each UDP packet is then put in the payload field of 

    an IP packet.

    The IP packets are then sent to a transmission buffer whose

    service rate is time varying and depends on the channel con-

    dition estimated by the adaptive error controller, as will be

    explained below. The main target of the rate controller is to

    avoid buffer saturation, which causes losses and long delays,

    while maximizing the encoding quality and stability. To this

    end, it chooses the QSP parameter according to a feedback 

    law monitoring the activity of the frame being encoded, its

    encoding mode (I, P, or B), and the current number of packets

    in the transmission buffer. The model introduced in the paper

    is so general that it can be applied whatever the feedback law.

    The feedback law used in the paper was introduced in [4] and

    [17] and, for the sake of completeness, will be reported in

    Section V-A. It has been defined in such a way that a controlled

    number of packets are present in the transmission buffer at the

    end of each GoP, while pursuing a constant distortion level

    within the GoP. Packets leaving the transmission buffer enter

    the adaptive error controller. Its main target is to use FEC to

    partially solve the problem of wireless-link unreliability. The

    FEC block creator divides packets into sets of  k  blocks. These

    blocks are given as input to the AFEC encoder and encodedin sets of  m   blocks, with  m ≥  k. If any set of   k   or moreblocks belonging to the same packet is received correctly, then

    the original packet can be reconstructed properly. Obviously,

    the larger the value of  m, the higher the probability that the

    information can be reconstructed at the receiver station, but the

    lower the wireless-link bandwidth available at the video source.

    The value of  m   is chosen by the FEC controller in such a

    way that the PEP, i.e., the probability that a packet cannot be

    reconstructed at the receiver station, is no higher than a target

    value  P̂ (C )PEP. Given that wireless channel conditions change

    dynamically, AFEC encoding is applied, as proposed in [1],

    [2], [6], and [7]. This encoding technique requires knowledgeof the current BER on the link. This estimation is performed

    by the wireless channel estimator. The estimated BER value

    is given as input to the FEC controller, which evaluates  m  so

    that the requirement on the PEP is satisfied. The value of  m

    therefore changes in time and, as a consequence, the available

    link capacity c̃(t) also changes in time as

    c̃(t) =

      k

    m(t)

    · c   (1)

    1

    If Real-Time Protocol (RTP)/Real-Time Control Protocol (RTCP) protocolsare also used over the UDP/IP protocol suite, the related overhead should beconsidered.

    where  c   is the capacity (in packets/s) when FEC is not used.

    At any time, the service rate of the transmission buffer is set

    equal to   c̃(t). Accordingly, both the MPEG encoder outputprocess and the transmission-buffer service process are stochas-

    tic processes, the first depending on the behavior of the source

    and the rate controller, and the second on the BER behavior of 

    the wireless channel. These processes will be modeled with twodiscrete-time SBBP processes  Ỹ (n) and  Ñ (n), respectively, asdescribed in detail in Section III.

    III. SYSTEM M ODEL

    In this section, we derive a discrete-time analytical model of 

    the system described in the previous section. We will set the

    slot duration ∆ equal to the video-frame interval.As a first step, Sections III-B and III-C will describe the mod-

    els of the noncontrolled MPEG encoder output and the available

    capacity of the channel as SBBPs [9]. Then, the whole system

    will be modeled as an SBBP/SBBP/1/ K  queueing system in

    Section III-D, where K  is the maximum number of packets thetransmission buffer can contain. For the sake of completeness,

    Section III-A provides a brief outline of SBBP processes.

     A. Switched Batch Bernoulli Processes (SBBPs)

    An SBBP   Y (n)   is a discrete-time emission processmodulated by an underlying Markov chain [9], and represents

    a special case of the family of the hidden Markov model

    processes [19].

    Each state of the Markov chain is characterized by an emis-

    sion probability density function (pdf): The SBBP emits data

    units according to the pdf of the current state of the underlyingMarkov chain. Therefore, the SBBP  Y (n)   is fully describedby the state space  (Y ) of the underlying Markov chain, themaximum number of data units the SBBP can emit in one

    slot   r(Y )MAX, and the matrix set   (Q

    (Y ), B(Y ))), where   Q(Y )

    is the transition probability matrix of the underlying Markov

    chain, while  B(Y ) is the emission probability matrix whose

    rows contain the emission pdfs for each state of the underlying

    Markov chain.

    If we indicate the state of the underlying Markov chain in the

    generic slot n  as  S (Y )(n), the generic elements of the matricesQ(Y ) and B(Y ) are defined as follows:

    Q(Y )[sY  ,s

    Y ] = ProbS (Y )(n + 1) = sY  |S 

    (Y )(n) = sY 

    ∀sY  , sY   ∈

    (Y ) (2)

    B(Y )

    [sY  ,r] = ProbY (n) = r|S (Y )(n) = sY 

    ∀sY   ∈

    (Y ), ∀r ∈

    0, r(Y )MAX

    .   (3)

    We will introduce an extension to the meaning of the

    SBBP to model not only a source emission process, but also

    a video-sequence activity process, and an available wireless-

    channel-capacity process. In the latter cases, we will indi-

    cate them as an activity SBBP and a transmission-channelSBBP, respectively, and their matrices   B(Y ) as the activity

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    probability matrix and the channel-transmission probability

    matrix, respectively.

     B. Noncontrolled MPEG Source Model

    The noncontrolled MPEG video source is part of the

    adaptive-rate source shown in Fig. 1 comprising the videosource, the MPEG encoder, and the packetizer. We denote it as

    noncontrolled because we are assuming it works with a constant

    QSP q  not controlled by the rate controller.

    The first step in modeling the whole video transmission

    system shown in Fig. 1 is the derivation of the SBBP process

    Ỹ q(n), modeling the emission of the noncontrolled MPEGvideo source at the packetizer output for each QSP  q .

    This model was calculated by the authors in [4] and [17].

    Here, for the sake of brevity, we will refer to those works

    in order to define the notation. The model captures two dif-

    ferent components: the activity-process behavior and the ac-

    tivity/emission relationships. As input, it takes the first- and

    second-order statistics of the activity process, and the threefunctions, one for each encoding mode (I, P, or B), charac-

    terizing the activity/emission relationships. The state of the

    underlying Markov process of   Ỹ q(n)   is a double variable,

    S (Ỹ )(n) = (S (G)(n), S (F )(n)), where  S (G)(n) ∈ (G) is thestate of the underlying Markov chain of the activity process

    G(n), and S (F )(n) ∈  J  is the frame to be encoded in the GoP atthe slot n. The state set (G) represents the set of activity levelsto be captured. For example, according to [5], we have  (G) ={Very Low, Low, High, Very High}. Set  J , on the other hand,represents the set of frames in GoP and depends on the GoP

    structure. For example, if the movie is encoded with the GoP

    structure IBBPBB, set J  is defined as  J  = {I,B,B,P,B,B}.As demonstrated in [4] and [17], the underlying Markov

    chain of  Ỹ q(n) is independent of  q . Therefore, we will indicate

    its transition probability matrix as   Q(Ỹ ) instead of   Q(Ỹ q),

    and set  (Q(Ỹ ), B(Ỹ q)), for each  q  ∈  [1, 31], defines the SBBPemission process modeling the output flow of the noncontrolled

    MPEG encoder, when it uses a constant QSP value q .

    C. Service SBBP Model

    The target of this section is to derive the SBBP model of 

    the process  Ñ (n), which represents the service process of the

    transmission buffer when AFEC is employed. As said so far,it closely depends on the amount of redundancy the AFEC

    encoder introduces to achieve the target maximum PEP  P̂ (C )PEP

    due to the wireless channel.

    As usual, (e.g., [14], [24], and [26]), we assume that the

    channel behavior can be described by means of an  M -states

    Markov process. Accordingly, channel statistical behavior can

    be described by an M  × M   transition probability matrix  Q(C )

    and by BERi, the BERs for each state of the process i ∈  [1,M ].Thus, the service SBBP model is represented by the follow-

    ing parameters:

    1) the maximum number of packets that can be transmitted

    in a time slot r( Ñ )MAX;

    2) the state space ( Ñ );

    3) the matrix set   (Q( Ñ ), B(

     Ñ ))   containing the transitionprobability matrix and the channel-emission probability

    matrix.

    Obviously, the transition probability matrix  Q( Ñ ) of the un-

    derlying Markov chain of the process  Ñ (n)   coincides withthe channel-transition probability matrix  Q(C ), as calculated

    in [26]. The state space  ( Ñ ) coincides with the channel state

    space, i.e.,  ( Ñ ) = [1,M ]. Instead, in order to derive  B(

     Ñ ),

    we have to calculate the bandwidth reduction due to the AFEC

    redundancy for each state i  of the channel SBBP. This depends

    on the BER characterizing the state BERi.

    The FEC redundancy to be introduced to achieve the target

    value for the maximum PEP  P̂ (C )PEP   should be such that the

    resulting PEP for any state   i   of the channel  P (C )PEP,i   is lower

    than or equal to the target one, i.e.,

    P (C )PEP,i ≤

     P̂ (C )PEP.   (4)

    According to the notation introduced in Section II, indicating

    the size of each block expressed in bits as  R, and assuming

    that losses introduced by the wireless channel are independent

    and uniformly distributed within a block,2 the PEP, when the

    channel is in the generic state  i, can be calculated as follows:

    P (C )PEP,i =

    ml=m−k+1

    m

    l

    · (1 − P BEP,i)

    m−l · (P BEP,i)l (5)

    where  P BEP,i   represents the probability that a block is cor-

    rupted when the channel is in state  i, and can be evaluated as

    follows:

    P BEP,i = 1 − (1 − BERi)R.   (6)

    Now, substituting (5) in (4), we can numerically find the

    minimum value of  m  verifying the inequality in (4) for each

    value  i  of the channel state. Let us indicate this value as  mi.

    Accordingly, the capacity  Ñ i  (in packets/s), which is actuallyavailable for the transmission of data to obtain a PEP lower

    than  P̂ (C )PEP   in the wireless channel when its state is  i, can be

    calculated as follows:

    Ñ i =  kmi

    · c   (7)

    where   c   is the channel capacity when no FEC encoding is

    applied (in [packets/s]).

    In general, from (7), we obtain a noninteger value for  Ñ i.However, we can assume that, when the channel state is  i, in

    each slot, the channel is able to transmit either Di =    Ñ i pack-ets with a probability of   pDi  = 1 − ( Ñ i − Di), or   (Di + 1)packets with a probability of  pDi+1 = 1 − pDi , where we haveindicated the largest integer no greater than  x  as x.

    2This assumption is accurate if interleaving is utilized, which is usual inwireless communications [28].

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    In summary, the emission probability matrix of the SBBP

    modeling the channel is  B ( Ñ ) ∈ [M X r

    ( Ñ )MAX], and its generic

    element can be calculated as follows:

    B( Ñ )

    [i,d] =  pDi,   if  d  =  Di pDi+1,   if  d  =  Di + 1

    0,   otherwise

    (8)

    where  r( Ñ )MAX   is the maximum number of packets that can be

    transmitted in one slot, i.e.,

    r( Ñ )MAX = maxi

    {Di + 1}.   (9)

    The transition probability matrix and the state space, together

    with the channel-emission probability matrix and the maximum

    number of packets that can be transmitted in one slot defined in

    (8) and (9), completely characterize the channel SBBP model.

     D. Video-Transmission-System Model

    The adaptive-rate source pursues a given target by imple-

    menting a feedback law in the rate controller, which calculates

    the value  q  of the QSP to be used by the MPEG encoder for

    each frame. The target of this section is to model the video

    transmission system as a whole, indicated here as  Σ. To thisaim, we use a discrete-time queueing system model.

    Let  K  represent the maximum number of packets that can

    be contained in the queue of the transmission buffer and its

    server. The server capacity of this queueing system, that is, the

    number of packets that can leave the queue at each time slot,

    is a stochastic process that has been modeled with the channelSBBP process  Ñ (n).

    The input of the queue system is the emission process of 

    the adaptive-rate source, indicated here as  Ỹ (n). Therefore, atslot  n, the transmission-buffer queue size is incremented by

    Ỹ (n), and decremented by  Ñ (n). Both the input and the outputprocesses can be modeled by means of two SBBP processes,

    as discussed above, where the slot duration is the frame

    duration ∆.To model the queueing system, we assume a late-arrival-

    system-with-immediate-access time diagram [3], [11]: Packets

    arrive in batches, and can enter the service facility if it is

    free, with the possibility of them being ejected almost instan-taneously. Note that in this model, a packet service time is

    counted as the number of slot boundaries from the point of entry

    to the service facility up to the packet departure time. Therefore,

    even though we allow the arriving packet to be ejected almost

    instantaneously, its service time is counted as 1, not 0.

    A complete description of   Σ   at the   nth slot requires athree-dimensional Markov process, whose state is defined as

    S (Σ)(n) = (S (Q)(n), S ( Ñ )(n), S 

      ˜(Y )(n)), where:

    1)   S (Q)(n) ∈  [0,K ]  is the transmission-buffer queue statein the  nth slot, i.e., the number of packets in the queue

    and in the service facility at the observation instant;

    2)   S ( Ñ )(n)   is the state of the underlying Markov chain of the channel SBBP  Ñ (n);

    3)   S   ˜(Y )(n) is the state of the underlying Markov chain of 

    Ỹ (n), which coincides with that of  Ỹ q(n), for any   q  ∈[1, 31].

    According to the late-arrival-system-with-immediate-access

    time diagram, the transmission-buffer state in slot  (n + 1)  canbe obtained through the Lindley equation [13]

    sQ  = max

    minsQ + r,K 

    − d, 0

      (10)

    where sQ   is the transmission-buffer state in the generic slot n,

    while r  and d  are the server capacity and the number of arrivals

    at slot n + 1, respectively.The channel SBBP  Ñ (n), modeled in Section III-C, can

    be equivalently characterized through the set of transition

    probability matrices M ( Ñ )(d), which are transition probability

    matrices including the probability that the server capacity is

    d   (in packets/slot). These matrices can be obtained from the

    parameter set (Q

    ( Ñ )

    , B

    ( Ñ )

    ) as follows:M (

     Ñ )(d)

    sÑ 

    ,sÑ 

    ≡ Prob

      Ñ (n + 1) =  d,S (

     Ñ )(n + 1) =  sÑ 

    S ( Ñ )(n) = sÑ 

    =Q(

     Ñ )

    sÑ 

    ,sÑ 

     · B( Ñ )sÑ 

    ,d

    ∀d ∈

    0, r˜(N )

    MAX

    .   (11)

    The adaptive-rate source emission process is modeled byan SBBP whose emission probability matrix depends on the

    transmission-buffer state. In order to model this process, we

    use the SBBP models of the noncontrolled MPEG video source

    described in Section III-B,  Ỹ q(n), for each q  ∈  [1, 31]. So, we

    have a parameter set  (Q(Ỹ ), B(Ỹ 1), B(Ỹ 2), . . . , B(Ỹ 31)), which

    represents an SBBP whose transition matrix is   Q(Ỹ ), and

    whose emission process is characterized by a set of emission

    matrices  {B(Ỹ q)}q=1,2,...,31. Consequently, at each time slot,the emission of the MPEG video source is characterized by an

    emission probability matrix chosen according to the QSP value

    defined by the feedback law q  =  φ(sQ,a , j).

    More concisely, as in (11), for the channel SBBP, wecharacterize the emission process of the adaptive-rate source

    through the set of matrices  {M (Ỹ )

    sQ

    (r)},  ∀r ∈  [0, r(Ỹ )MAX], each

    matrix representing the transition probability matrix including

    the probability of   r   packets being emitted when the buffer

    state is   sQ. Accordingly, the generic element of the ma-

    trix  M (Ỹ )sQ

    (r)   can be obtained from the above parameter set

    (Q(Ỹ ), B(Ỹ 1), B(Ỹ 2), . . . , B(Ỹ 31)) as follows:M 

    (Ỹ )sQ

    (r)[(i,j),(i,j)]

    =

    a∈(Act)

    Q(Ỹ )[(i,j),(i,j)]

    ·B(Ỹ q)(r)[(i,j),r]

    · f Act(a|i, j)   (12)

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    where the following hold:

    1)   q  is the QSP chosen when the frame to be encoded is the

     j th in the GoP, the activity is  a , and the transmission-

    buffer state before encoding this frame is  sQ. The value

    of  q  is determined by the feedback law  =  φ(·), i.e.,

    = φs

    Q, a

    , j

    .   (13)

    2)   f Act(a|i, j ) is the probability that the generic frame j

    in the GoP has an activity  a when its activity level is  i.

    This function, as demonstrated in [15], [16], and [21], is a

    Gamma pdf, whose mean value and variance characterize

    the video trace.

    3)   (Act) is the set of all the possible activities.

    Finally, we can model the video transmission system as a

    whole. If we indicate two generic states of the system as  sΣ =(sQ, s

    Ñ , s

    Ỹ  )   and   sΣ = (s

    Q, s

    Ñ , s

    Ỹ  ), the generic element of 

    the transition matrix of the video transmission system as a

    whole Q(Σ) can be calculated, due to (11) and (12), as follows:

    Q(Σ)

    sQ

    ,sÑ 

    ,sỸ 

    ,

    sQ

    ,sÑ 

    ,sỸ 

    ≡ Prob

    S (Q)(n + 1) = sQ,

    S ( Ñ )(n + 1) = s

    Ñ ,

    S (Ỹ )(n + 1) = sỸ ,

    S (Q)(n) = sQ,

    S ( Ñ )(n) = s

    Ñ ,

    S (Ỹ )(n) = sỸ ,

    =

    d( Ñ )MAXd=0

    r(Ỹ  )MAXr=0

    M (

     Ñ )(d)

    sÑ 

    ,sÑ 

    · M (Ỹ )s

    Q

    (r)s

    Ỹ  ,s

    Ỹ  · ψ sQ, sQ, . . . , r , d   (14)

    where ψ(sQ, sQ, . . . , r , d) is a Boolean condition for the queue

    state behavior, and is defined as follows:

    ψsQ, s

    Q,K,r,d

    =

    1,   if  max

    minsQ + r,K 

    − d, 0

     =  sQ0,   otherwise

      .   (15)

    Once the matrix Q(Σ) is known, we can calculate the steady-

    state probability array of the system  Σ   as the solution of thefollowing linear system

    π(Σ) · Q(Σ) = π(Σ)

    π(Σ) · 1 = 1  (16)

    where   1   is a column array whose elements are equal to 1,and  π(Σ) is the steady-state probability array, whose generic

    element is

    π(Σ)

    [(sQ,s Ñ ,sỸ )] =  ProbS (Q)(n) = sQ, S 

    ( Ñ )(n) = sÑ ,

    S (Ỹ )(n) = sỸ 

    .   (17)

    A direct solution of the system in (16) may be difficultsince the number of states grows explosively as the maximum

    transmission buffer size K  increases. Nevertheless, many algo-

    rithms, e.g., [10], [18], and [23], enable us to calculate the array

    π(Σ), while maintaining a linear dependence on  K .

    IV. QUANTIZATION-D ISTORTION A NALYSIS

    In this section, we evaluate both the static and time-varying

    statistics of the quantization distortion, represented by the pro-

    cess PSNR(n).More specifically, we will quantize the PSNR process with

    a set of   L   different levels of distortion,   {µ1, µ2, . . . , µL},

    each representing an interval of distortion values where thequality perceived by the users can be considered constant.

    As an example, for the movie Evita, from a subjective

    analysis obtained with 300 tests, the following   L = 5   lev-els of distortion were envisaged:  µ1 = [31.2, 34.2]   dB,  µ2 =[34.2, 35.0]   dB,   µ3 = [35.0, 36.2]   dB,   µ4 = [36.2, 38.4]   dB,and µ5 = [38.4, 52.1] dB.

    The pdf  f PSNR( p)  can be easily calculated from the transi-tion probability matrix and the steady-state probability array of 

    the whole system, which have been derived in (14) and (16),

    respectively [see (18) at the bottom of the page], where the

    following hold:

    1)   ψ[s

    Q,a

    ,j

    ]( p) is a Boolean condition defined as follows

    ψ[sQ,a,j]( p) =

    1,   if  F (j

    )φsQ, a

    , j

     =  p0,   otherwise

      .

    (19)

    2)   F (j)(q ) in (19) is the so-called distortion curve [5], [17],

    [22] for the generic frame  j, which is the curve linking

    the average PSNR to the QSP value,  q , used to encode

    the frame.

    Now, in order to calculate the statistics of the quantized PSNR

    process, let us define the array   γ (ζ ) in which the generic

    element   γ (ζ )l   , for each  l ∈  [1, L], is the QSP range giving a

    distortion belonging to the   lth level for a frame encoded withencoding mode   ζ  ∈ {I, P, B}. Of course, by so doing, weassume that a variation of  q   within the interval  γ 

    (ζ )l   does not

    cause any appreciable distortion. From the distortion curves

    for the movie   Evita, we have calculated the following QSP

    f PSNR( p) ≡  Prob {PSNR(n) = p} =K 

    sQ=0

    sÑ 

    ∈( Ñ )

    i∈(G)

    j∈J 

    K sQ=0

    sÑ 

    ∈( Ñ )

    i∈(G)

    a∈(Act)

    f Act(a|i, j )

    · Q(Σ)sQ

    ,sÑ 

    ,(i,j)

    ,

    sQ

    ,sÑ 

    ,(i,j) · π(Σ)

    sQ

    ,sÑ 

    ,(i,j) · ψ[sQ,a,j]( p)   (18)

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    ranges corresponding to the above distortion levels  µl, for each

    l ∈  [1, 5].

    1) For I frames:   γ (I) = [[16, 31], [13, 15], [10, 12], [6, 9],[1, 5]].

    2) For P frames:   γ (P) = [[15, 31], [13, 14], [10, 12], [6, 9],

    [1, 5]].3) For B frames:  γ (B) = [[17, 31], [14, 16], [11, 13], [7, 10],

    [1, 6]].

    Let   q  =  φ(sQ, a, j )   be the feedback law, linking the

    transmission-buffer state at the beginning of a generic slot

    n,   sQ ∈  [0,K ], the activity of the frame in the same slot,

    a ∈ (G), and the position in the GoP of the frame tobe encoded,   j ∈ J , to the QSP to be used to encode the

    current frame. Moreover, for each   a and   j , let   θ(a,j)l   =

    {∀sQ   such that  φ(sQ, a

    , j ) ∈  γ (ζ )

    l   }   be the range of valuesof the transmission-buffer state for which the rate controller

    chooses QSP values belonging to the level µl, according to theadopted feedback law. By definition, it follows that a variation

    of the transmission-buffer state within  θ(a,j)l   does not cause

    any appreciable distortion variation.

    Let us now calculate the probability that the value of the

    process PSNR(n)   is in the generic interval  µl,  π(PSNR)[l]

      , and

    the pdf  f δl(m)  of the stochastic variable   δ l, representing theduration of the time the process PSNR(n)   remains in thegeneric interval µl without interruption. They are defined as

    π(PSNR)[l]

      = Prob {PSNR(n) ∈  µl}   (20)

    and (21), shown at the bottom of the page. The term π(PSNR)[l]   in

    (20) can be calculated from the pdf  f PSNR( p) obtained in (18)as follows:

    π(PSNR)[l]   =

     p∈µl

    f PSNR( p).   (22)

    In order to calculate the pdf   f δl(m)   in (21), let us indicatethe matrix containing the one-slot probabilities of transition

    towards system states in which the distortion level is   µl   as

    Q(Σ)→µl . It can be obtained from the transition probability matrix

    of the system Q(Σ), as in (23), shown at the bottom of the page.

    Therefore, the pdf f δl(m) can be calculated as the probability

    that the system Σ, starting from a distortion level  µl, remains in

    the same level for  (m − 1)  consecutive slots, and leaves thislevel at the mth slot, that is

    f δl(m) = π(Σ1,µl) ·Q(Σ)→µl

    m−1· Q(Σ)→(=µl) · 1

    where :   π(Σ1,µl) =  π(Σ,µl) · Q(Σ)→µl

    π(Σ,=µl) · Q(Σ)→µl · 1T 

    .   (24)

    The array π(Σ1,µl) in (24) is the steady-state probability array in

    the first slot of a period in which the distortion level is  µl. The

    array π(Σ,=µl), on the other hand, is the steady-state probability

    array in a generic slot in which the distortion level is other than

    µl, and is defined as

    π(Σ,=µl) =  π(Σ) · Q(Σ)→µl

    π(Σ) · Q(Σ)→µl · 1T .   (25)

    V. CAS E S TUDY

     A. System Characterization

    We analyzed the statistical characteristics of 1 hour of MPEG

    video sequences of the movie Evita. To encode this movie, we

    used a frame rate of  F   = 25 frames/s, and a frame size of 180macroblocks. The GoP structure IBBPBB was used, selecting a

    ratio of total frames to intraframes of  GI  = 6, and the distancebetween two successive P frames or between the last P frame in

    the GoP and the I frame in the next GoP as  GP = 3. The sizeof the transmission buffer has been set to  K  = 60 packets. Thegross link capacity assigned to the video application is 2 Mb/s.

    The IP packets at the wireless terminal are divided into 40 bytes

    blocks, as usual in the universal mobile telecommunications

    system (UMTS) environment. The AFEC module encodes sets

    of  k  = 16 blocks into sets of  m.In this case study, we use the eight-state finite-state Markov

    channel (FSMC) model introduced in [26] for the wireless

    channel and consider two different cases.

    1) Pedestrian: The mobile user’s velocity is 5 km/h.

    2) Driver: The mobile user’s velocity is 55 km/h.

    Assuming that wireless transmission is performed in the 2-GHz

    band, which is the value used in UMTS, the maximum Doppler

    frequency is  f m = 10 Hz in the first case and  f m  = 100 Hz inthe second.

    The values that characterize Q(C ) are given in Table I for the

    pedestrian and driver cases. The above matrices were calculated

    f δl(m) = Prob

    PSNR(n + 1) ∈  µl, . . . ,PSNR(n + m − 1) ∈  µl,PSNR(n + m) ∈ µl

    PSNR(n − 1) ∈ µlPSNR(n) ∈  µl

      (21)

    Q(Σ)→µl

    sQ,s

    Ñ ,(i,j)

    ,

    sQ,s

    Ñ ,(i,j) = a∈(Act) Q

    (Σ)

    sQ,s˜N 

    ,(i,j),sQ,s˜N 

    ,(i,j)f Act(a|i, j),   if  sQ ∈  θ(a,j)l

    0,   otherwise(23)

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    TABLE IQ(C ) PARAMETERS IN THE PEDESTRIAN CAS E (f m  = 10 Hz)  A ND  DRIVER CAS E (f m  = 100 Hz)

    TABLE IIREDUNDANCY BLOCKS AND NET  LIN K CAPACITY OFFERED TO THE APPLICATION FOR DIFFERENT  CHANNEL STATES AND  TARGET ERROR

    PROBABILITIES   ˆ

    (C )

    PEP  IN THE

     DRIVER

     CAS E

    , WHEN THE

     GROSS

     LIN K

     CAPACITY IS

     c  = 2 Mb/s

    assuming that the video-frame rate is 25 frames/s and therefore,

    the slot duration is ∆ = 40 ms.The target error-probability values considered are  P̂ 

    (C )PEP =

    10−5,   P̂ (C )PEP = 10

    −4,   P̂ (C )PEP = 10

    −3, and   P̂ (C )PEP = 10

    −2.

    Table II lists, for each state   i  of the server SBBP model, the

    values of  mi   and the resulting available link capacities  c̃i   for

    these

      ˆP 

    (C )

    PEP values in the driver case, taken as an example.In this case study, we will consider a feedback law obtained

    from the statistics of the movie Evita, expressed in terms of rate

    and distortion curves [5], [17], [23]. The rate curves  Ra,j (q )give the expected number of packets which will be emitted

    when the j th frame in the GoP has to be encoded, if its activity

    value is  a, and is encoded with a QSP value  q . The distortion

    curves  F (j)(q )   give the expected encoding PSNR, and havebeen defined in Section IV. The rate and the distortion curves

    for the movie Evita are shown in Fig. 2.

    The considered feedback law aims to maintain the number

    of packets in the transmission-buffer queue lower than a given

    threshold K θ at the end of each GoP interval, while maintaining

    stable the PSNR during the whole GoP. In this case, both therate curves Ra,j (q ) and the distortion curves  F 

    (j)(q ) are used.

    More specifically, if we indicate the transmission-buffer queue

    length and the channel available capacity when the  jth frame in

    the GoP has to be encoded as  sQ   and sÑ , respectively, and  a

    being the activity of this frame, the QSP is chosen assuming the

    following.

    1) The activity will remain constant during the rest of the

    GoP, that is, Act(n) = a, for each frame h  ∈  [ j + 1, GI ].2) The channel behavior, and therefore the available network 

    bandwidth  Ñ (n), remains constant during the rest of theGoP, that is, for each frame  h  ∈  [ j + 1,GI ].

    Under these assumptions, the QSP is chosen as the minimum

    QSP q̄ , such that it is possible to find a set of QSP values for thenext frames of the GoP,  [q j+1, . . . , q  GI ], so that the followinghold.

    1) The PSNR of those frames is constant, and equal to the

    value that should be achieved for frame j .

    2) The number of emitted packets expected for the next

    frames of the GoP, if these QSP values are used, added

    to the current queue, minus the number of packets that

    will leave the queue until the end of the GoP, results tolower than the given threshold  K θ.

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    Fig. 2. Rate-distortion curves for I, P, and B frames. Rate curves for (a) frame I, (b) frame B, and (c) frame P. (d) Distortion curves.

    In other words, the feedback law works by choosing the QSP

    as in (26), shown at the bottom of the page.

     B. Numerical Results

    Fig. 3 shows the pdfs of the transmission-buffer queuesize for the two values of the Doppler frequency  f m   and for

    a given value of the target error probability among those being

    considered. The values shown have been calculated as follows:

    ProbS (Q)(n) = sQ

    =

    s Ñ ∈( Ñ )

    sỸ 

     ∈(Ỹ  )

    π(Σ)

    [(sQ,s  Ñ ,sỸ  )].   (27)

    We can observe that the curves are basically Gamma distri-

    butions and are very similar to each other independently of 

    the  P̂ (C )PEP   value. This is the evidence that the feedback law

    works properly. This is further demonstrated in Fig. 4 where

    we show the average queue size as well as the mean delay in

    the transmission buffer. The value of the average queue size

    does not change significantly when the  P̂ (C )PEP  changes and ishigher in the driver case. This can be explained by the fact that

    in the driver case, the wireless medium quality is lower and

    therefore, the transmission-buffer service rate is lower. Similar

    discussions can be carried out concerning Fig. 5, where the

    performance in terms of loss probability in the transmission

    buffer is shown and calculated as in [4].

    q  =  φ(sQ,a ,j) = minq̄∈[1,31]

    q̄  such that ∃ [q j+1, . . . , q  GI ]  for which :F (k)

    (q k) = F (j)

    (q̄ )   ∀k ∈  [ j + 1, . . . , GI ]sQ + Ra,j(q̄ ) +GI

    k=j Ra,j(q k) − (GI  −  j + 1) · Ñ (n) ≤  K θ

    (26)

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    Fig. 3. Transmission-buffer size pdf for  P̂ (C )PEP = 10

    2 (a) in the pedestrian case and (b) in the driver case.

    Fig. 4. Average transmission-buffer size and mean delay versus the target error probability P̂ (C )PEP.

    Fig. 6 shows the performance related to the encoding quality.

    In particular, it can be observed that, for high values of  P̂ (C )PEP,due to the high amount of available bandwidth, the most likely

    PSNR level is the highest. On the contrary, for low values of 

    P̂ (C )PEP, as a result of the large amount of redundancy introduced

    by AFEC, the available bandwidth is low, and therefore, the

    video source reduces the encoding quality. For this reason, the

    lower the value of the target PEP in the wireless link  P̂ (C )PEP,

    the greater the probability of poorer PSNR levels. In order to

    better quantify the influence of the choice of the target value

    P̂ (C )PEP   on the encoding performance, in Fig. 6, the average

    PSNR level is shown. As expected, the worst case for the av-

    erage PSNR level is given when the AFEC has a very stringent

    target for the maximum PEP  P̂ (C )PEP. When a less stringent targetvalue for the PEP is required, the encoding quality increases.

    Obviously, the average PSNR value, and thus encoding quality,

    is higher in the pedestrian case.

    VI. CONCLUSION

    In this paper, we have defined an analytical framework for

    the evaluation of the performance of real-time MPEG video

    transmission over a wireless link that applies AFEC to keep the

    PEP below a given threshold.

    The MPEG encoder uses a rate controller that adapts the

    output rate by appropriately setting the QSP to follow the

    bandwidth variations while maximizing encoding quality and

    stability. The whole system has been modeled by an emission

    process that feeds the transmission buffer; the server of thisbuffer behaves according to the channel conditions, i.e., the

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    Fig. 5. Packet loss probability in the transmission buffer versus the target error probability  P̂ (C )PEP.

    Fig. 6. Average PSNR level versus the target error probability  P̂ (C )PEP.

    service rate is higher when channel conditions are good and

    lower when channel conditions are bad.

    SBBPs have been used to model both the MPEG video source

    [4], [15], [17] and the server process of the transmission buffer

    that coincides with the time-varying available bandwidth in the

    network. Accordingly, the whole system has been modeled as

    an SBBP/SBBP/1/ K  process.

    The analytical framework proposed in the paper has been

    used to evaluate the performance in terms of the distortion

    introduced by the quantization mechanism in the encodingprocess, which are the loss and mean delay in the transmission

    buffer. Numerical results show that our system is very robust

    and reliable due to the implemented feedback law that main-

    tains almost constant the mean delay and the loss probabil-

    ity in the output buffer. Moreover, the corruption probability

    in the wireless channel is also limited in spite of possible

    variations in time in the wireless-channel BER. The proposed

    model allows the designer to evaluate the introduced encoding

    quality variation that represents the cost of using this ap-

    proach. The results obtained in the paper can be used to obtain

    the best tradeoff between encoding quality and informationcorrectness.

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    Laura Galluccio received the Laurea degree in elec-trical engineering and the Ph.D. degree in electrical,

    computer and telecommunications engineering, bothfrom the University of Catania, Catania, Italy, in2001 and 2005, respectively.

    Since 2002, she has been with the Italian NationalConsortium of Telecommunications (CNIT), whereshe is working as a Research Fellow within the Vir-tual Immersive Communications (VICOM) Project.From May to July 2005, she was a Visiting Scholarat the COMET Group, Columbia University, New

    York, NY. Her research interests include ad hoc and sensor networks, protocolsand algorithms for wireless networks, and network performance analysis.

    Dr. Galluccio served and will serve in the Program Committee of the4th Academic Network for Wireless Internet Research in Europe (ANWIRE)International Workshop on Wireless Internet and Reconfigurability, the 20thInternational Symposium on Computer and Information Sciences (ISCIS 05),and Networking 2006.

    Giacomo Morabito   (M’02) received the Laureadegree in electrical engineering and the Ph.D.degree in electrical, computer, and telecommunica-tions engineering from the University of Catania,Catania, Italy, in 1996 and 2000, respectively.

    From November 1999 to April 2001, he was withthe Broadband and Wireless Networking Laboratoryof the Georgia Institute of Technology as a ResearchEngineer. Since May 2001, he has been with theSchool of Engineering at Enna of the University of Catania, where he is currently an Assistant Professor.

    He is serving as a Guest Editor on the editorial board of Computer Networksand Mobile Networks and Applications (MONET). He is also a Member of thetechnical program committee of several conferences. Moreover, he has beenthe Technical Program Co-Chair of Med-Hoc-Net 2004. His research interestsinclude mobile and satellite networks, self-organizing networks, quality of service (QoS), and traffic management.

    Dr. Morabito is serving on the Editorial Board of  IEEE Wireless Communi-cations Magazine.

    Giovanni Schembra   received the degree in elec-trical engineering from the University of Catania,Catania, Italy, in 1991. Working in the telecom-munications area, he received the Master’s degreefrom CEFRIEL, Milan, Italy, in 1992, with his thesisfocusing on the analytical performance evaluation inan ATM network. He received the Ph.D. degree inelectronics, computer science, and telecommunica-tions engineering with a dissertation on multimediatraffic modeling in a broadband network.

    He is currently an Assistant Professor in Telecom-munications at the University of Catania.