Intra- and Inter-Stream Synchronisation for Stored Multimedia Streams

Transcription

1 IEEE International Conferene on Multimedia Computing & Systems, June 17-23, 1996, in Hiroshima, Japan, p Intra- and Inter-Stream Synhronisation for Stored Multimedia Streams Ernst Biersak, Werner Geyer, Christoph Bernhardt Institut Euréom 2229 Route des Crêtes, Sophia ntipolis, FRNCE {erbi,geyer,bernhard}@eureomfr bstrat Multimedia streams suh as audio and video impose tight temporal onstraints due to their ontinuous nature Often, different multimedia streams must be presented in a synhronized way We introdue a sheme for the ontinuous and synhronous delivery of distributed stored multimedia streams aross a ommuniations network We propose a protool for the synhronized playbak, ompute the buffer requirement, and desribe the experimental results of our implementation The sheme is very general and does not require bounded jitter or synhronized loks and is able to ope with lok drifts and server drop outs 1 Introdution 11 Motivation dvanes in ommuniation tehnology lead to new appliations in the domain of multimedia Emerging highspeed, fiber-opti networks make it feasible to provide multimedia servies suh as Video On-Demand, Tele- Shopping or Distane Learning These appliations typially integrate different types of media suh as audio, video, text or images Customers of suh a servie retrieve the digitally stored media from a video server [1] for playbak 12 Multimedia Synhronization Multimedia refers to the integration of different types of data streams inluding both ontinuous media streams (audio and video) and disrete media streams (text, data, images) Between the information units of these streams a ertain temporal relationship exists Multimedia systems must maintain this relationship when storing, transmitting and presenting the data Commonly, the proess of maintaining the temporal order of one or several media streams is alled multimedia synhronization [4] Synhronization an be distinguished on different levels of abstration Event-based synhronization assures a proper orhestration of the presentation of distributed multimedia objets multimedia objet may be, for instane, a news ast onsisting of several subobjets like audio and video On a lower level ontinuous synhronization or stream synhronization, respetively, opes with the problem of synhronizing the playout of data streams [13] The lassial example of stream synhronization is the lip-synhronized presentation of audio and video [5] Continuous media are haraterized by a well-defined temporal relationship between subsequent data units Information is only onveyed when media quanta are presented ontinuously in time For video/audio the temporal relationship is ditated by the sampling rate The problem of maintaining ontinuity within a single stream is referred to as intra-stream synhronization Moreover, there exist temporal relationships between media-units of related streams, for instane, an audio and video stream The preservation of these temporal onstraints is alled interstream synhronization To solve the problem of stream synhronization, we have to regard both issues whih are tightly oupled One an distinguish between life synhronization for life media streams and syntheti synhronization for stored media streams [15] In the former ase, the apturing and playbak must be performed almost at the same time, while in the latter ase, samples are reorded, stored and played bak at a later point of time For life synhronization, eg in teleonferening, the tolerable end-to-end delay is in the order of a few hundred milliseonds only Consequently, the size of the elasti buffer must be kept small, trading-off requirements for jitter ompensation against low delay for interative appliations Syntheti synhronization of reorded media stream is easier to ahieve than life synhronization: higher end-to-end delays are tolerable, and the fat that soures an be influened proves to be very advantageous as will be shown later It is, for instane, possible to adjust playbak speed or to shedule the start-up times of streams as needed However, as resoures are limited, it is desirable for both kinds of synhronization to keep the required buffers as small as possi-

2 ble [9] 13 Context of the Synhronization Problem The synhronization problem addressed in this paper is motivated by our work on salable video servers We have designed and implemented a video server, alled Server rray, onsisting of n server nodes video is distributed over all server nodes using a tehnique alled sub-frame striping: Eah video frame f i is partitioned into n equal size parts i, j, alled sub-frames, that are stored on the n different servers If F i = { i, 1,, in, } denotes the set of sub-frames for f, then: i i, j = f i j = 1 n The server array with the synhronization mehanisms presented in this paper has been suessfully implemented in our video server prototype [1] During playbak, eah server node is ontinuously transmitting its sub-frames to the lient The transfer is sheduled suh that all striping bloks that are part of the same frame are ompletely reeived by the lient at the deadline of the orresponding frame The lient reassembles the frame by ombining the sub-frames from all server nodes nother example for inter-stream synhronization of stored multimedia streams is given by Cen et al [2] They desribe a distributed MPEG player with the audio server and the video server being at different loations in the Internet environment 2 Synhronization Sheme 21 Overview We propose a synhronization sheme for the delivery of stored media that ahieves both, suitable intra- and interstream synhronization The sheme is reeiver-based and does not assume global loks Resynhronization is done by skipping/pausing, and furthermore, we apply the onept of a buffer level ontrol To initiate the playbak of a stream in a synhronized manner we introdue a novel start-up protool Our protool has been influened by the ideas of Ishibashi et al [8] who ahieve inter-stream synhronization by providing intra-stream synhronization for eah stream involved and by Santoso [14] who provides onditions for a smooth playout For re-synhronization, we adopt a sheme similar to the one desribed by Koehler et al and Rothermel et al [9], [13] We derive our synhronization sheme by step-wise refinement: First we develop a solution for the ase of zero jitter and then relax this assumption requiring bounded jitter only Finally, we over synhronization problems not introdued by jitter In eah step we derive the buffer requirements and playout deadlines to assure inter- and intra-stream synhronization We present three models Model 1 solves the problem of different, but fixed delays on the network onnetions for eah substream We propose a synhronization protool that ompensates for these delays by omputing well-defined starting times for eah server The protool allows to initiate the synhronized playbak of a media stream that is omposed of several substreams Model 2 takes into aount the jitter experiened by media-units travelling from the soure to the destination Jitter is assumed to be bounded To smoothen out jitter, elasti buffers are required Our sheme guarantees a smooth playbak of the stream and has very low buffer requirements Model 2 overs intra-stream synhronization as well as inter-stream synhronization Model 3 solves the problems of lok drift, hanging network onditions and server drop outs by employing a buffer level ontrol with a feedbak loop to the servers so as to regain synhronization in the ase of disturbanes gain, buffer requirements are regarded with respet to the results of models 1 and 2 The behavior of a filtering funtion is examined Filters are neessary to identify whether a problem is of long-term or short-term effet The tuning of some parameters is disussed For the proposed synhronization sheme, we assume that a lient D is reeiving sub-streams from different servers 1 Client and servers are interonneted via a network (see figure 1) S S S Network Figure 1 Distributed arhiteture for the synhronization sheme Eah of the servers denoted by S delivers an independent substream of media-units (sometimes also referred to as frames) The prodution rate is driven by the server lok rriving media-units are buffered in FIFO queues at the destination D The playout of the entire stream, omposed of all the substreams, is driven by the destination s lok 1 It is also possible that a single server sends multiple substreams to a lient Our model is more general and overs this ase too D 2

3 22 Soures of synhrony Several soures of asynhrony exist in the onfiguration desribed in the previous setion These are: Different delays, network jitter, end-system jitter, lok drift, alteration of the average delay, and server drop outs 23 ssumptions Our synhronization mehanism uses time stamps Eah media-unit 2 sheduled by a server is stamped with the urrent loal time to enable the lient to alulate statistis, suh as for the roundtrip delay, jitter, or inter-arrival times Moreover, we assume that eah media-unit arries a sequene number for determining media-unit order We ould use for our purposes a protool suh as RTP, whih is urrently designed by the IETF, and whih provides fields for both, time stamp and sequene number In ontrast to other approahes, buffer requirements or fill levels are always stated in terms of media-units or time, instead of the amount of alloated memory This seems reasonable beause media-unit sizes vary due to enoding algorithms like JPEG or MPEG [9] However, notie that a mapping of media-units to the alloation of bytes must be arried out for implementation purposes Taking the largest media-unit of a stream as an estimate wastes a lot of memory, espeially when using MPEG ompression Sophistiated solutions of mapping are subjet of future work In the following, we will use the term buffer slot to denote the buffer spae for one media-unit Sine proessing time, eg for protool ations does not onern the atual synhronization problem, we will neglet Finally, we assume that ontrol messages are reliably transferred Table 1 shows the parameters that we used to desribe our model set of media-units that needs to be played out at the same time is referred to as synhronization group We assume that media-units are distributed in a round robin fashion aross the involved server nodes Hene, we an identify the storage loation of a media-unit by its media-unit number 3, ie Server S i mod n stores media-unit i (1) This leads to the following formulation of the synhronization problem: The lient must playout the media-units of all subsets I j, with j mod n = 0, at the same time 2 We will also use the abbreviation mu for media-unit 3 This implies that eah substream will send media-units at the same rate n extension of the sheme to different media-unit rate, eah one being the integer multiple of a base rate is straight forward Symbol Desription Unit n N i, j, υ number of server nodes in the server array number of media-units of a stream media-unit index ( ijυ,, { 0,, N 1} ) k server index ( k { 0,, n 1} ) I j index set of n subsequent mediaunits starting with media-unit j S k denotes server node k providing substream k D s i s i a i denotes the destination or lient node initial sending time of media-unit i in server time synhronized sending time of media-unit i in server time arrival time of media-unit i in lient time a The roundtrip delay omprises the delay for a ontrol message that requests a media-unit and the delay for delivering the media-unit 24 Model 1: Start-Up Synhronization d i roundtrip delay a for media-unit i measured at lient site d max maximum roundtrip delay t start starting time of the synhronization protool t ref t 0 t i δ ij referene time for the start-up alulation earliest possible playout time of the first media-unit expeted arrival of the media-unit i at the lient site arrival time differene between media-unit i and j Table 1 Model parameters 241 Introdution Under the assumption of onstant delay and zero jitter, we solve the synhronization problem by assuring that the first n media-units, whih onstitute a synhronization group, arrive at the same time at the lient We therefore need 3

4 t i = t 0 i The major problem addressed by model 1 is the ompensation for different delays due to the independene of the different substreams For instane, the geographial distane from server to lient may be different for eah server Thus, starting transmission of media-units in a synhronized order would lead to different arrival times at the lient with the result of asynhrony Usually, this is ompensated by delaying media-units at the lient [5] Depending on the loation of the soures, large buffers may be required In order to avoid buffering to ahieve the equalization of different delays, we take advantage of the fat that stored media offers more flexibility: The idea is to initiate playout at the servers suh that media-units arrive at the sink in a synhronous manner This is performed by shifting the starting times of the servers on the time axis in orrelation to the network delay of their onnetion to the lient The proposed start-up protool onsists of two phases In the first phase, alled evaluation phase, roundtrip delays for eah substream are alulated, while In the seond phase, alled synhronization phase, the starting time for eah server is alulated and transmitted bak to the servers The model is based on the assumption of a onstant end-to-end delay without any jitter For the moment we do not onsider hanging network onditions, server dropouts, and lok drift In suh a senario, synhronization needs to be done one at the beginning and is maintained afterwards automatially We need to introdue some more notation to express interdependenies between the parameters of the model We then give a desription of the start-up protool flow and prove its orretness We lose the setion with an example of the protool The starting time t start of the protool equals the beginning of the first phase Without loss of generality let t start = 0 To begin with, we regard the first n media-units of a stream given by that are distributed aross the n servers The seond phase of the protool begins at time t ref, determined by the last of the first n media-units that arrives: t ref = max { a i i } The differene δ ij = a i a j ij, between the arrival times of arbitrary media-units i and j is needed to alulate the starting times of the servers 242 Start-Up Protool The synhronization protool for starting playbak at the server sites is launhed after all involved parties are (2) ready for playbak and onsists of an evaluation phase and synhronization phase During start-up, the lient sends two different kinds of ontrol messages to the servers: Eval_Request(i): Client D requests media-unit i from server S i, i Syn_Request(i, s ): Client D transmits the starting i time s i to server S i (a) Evaluation Phase t loal time t start, lient D sends an Eval_Request(i) to servers S i, i Server S i reeives the Eval_Request(i) at loal time s i, i Server S i sends media-unit i time-stamped with s i immediately bak to lient D, i t loal time a i, lient D reeives media-unit i from Server S i, i t loal time t ref, lient D has reeived the last mediaunit The roundtrip delays d i = a i t start, i and the maximum round trip time d max = max { d i i }, are omputed (b) Synhronization Phase t loal time t ref, lient D omputes the earliest playout time t 0 = max { t ref + d i i }, the index υ that determines t 0 as υ = { j t ref + d j = t 0 }, and the delay differenes as δ υi = a υ a i, i With these results the starting time of Server S i is alulated in server time s i = s i + d max + δ υi, i Client D sends a Syn_Request(i, s i ) to server S i, i t loal time s i + d i + ( t ref a i ), server S i reeives the Syn_Request(i, s i ), i I 0 t loal time s i, server S i starts sheduling of the substream by sending media-unit i, i t loal time t i, lient D reeives media-unit i, i t any time, only one synhronization group of n media-units must be buffered at the lient; after the omplete reeption the media-units are played out immediately 25 Model 2: Intra- and Inter-Stream Synhronization 251 Introdution Model 1 shows how to ope with different but onstant delays for eah substream However, synhronization is performed under the assumption that jitter does not exist Model 2 loosens this assumption and takes into aount 4

5 end-system jitter and network jitter We onsider the umulative jitter and assume the jitter to be bounded Due to jitter, media-units will not arrive in a synhronized manner although they have been sent in a timely manner The temporal relationship within a single substream is destroyed and time gaps between arriving mediaunits vary aording to the ourred jitter Thus, an isohronous playbak annot be ahieved if arriving media-units of a substream would be played out immediately Furthermore, jitter may distort the relationship between mediaunits of a synhronization group Hene, intra-stream synhronization as well as inter-stream synhronization is disturbed To smoothen out the effets of jitter, media-units must be delayed at the sink suh that a ontinuous playbak an be guaranteed For this purpose, playout buffers are required The main point addressed by model 2 is intra- and interstream synhronization and the alulation of the required buffer spae First, we regard the synhronization of a single substream Based on a rule of Santoso [14], we formulate a theorem that states a well defined playout time 4 for a substream suh that intra-stream synhronization an be guaranteed Smooth playout annot be guaranteed if starting before the playout deadline Starting at a later time would require more buffer spae fterwards, we will extend our onsiderations to the synhronization of multiple substreams The main idea in order to ahieve interstream synhronization is to maintain intra-stream synhronization for eah substream [8] We begin with an extension of the model parameters used so far (f Table 2) Throughout this paper, we assume bounded jitter and we use the definition of jitter given by Rangan et al [12] who define jitter as the differene between the maximum delay and the minimum delay k max min = d k d k, k max = max { k k { 0 n 1} } 252 Synhronized Playout for a Single Substream To guarantee the timely presentation of a single stream subjet to jitter, it is neessary to buffer arriving mediaunits at the lient to ompensate the jitter The buffer is emptied at a onstant rate Santoso et al [14] have already shown that the temporal relationship within one ontinuous media stream an be 4 The playout time or playout deadline is defined as the time elapsed at the lient between arrival and playout of the first media-unit of a substream (3) (4) Symbol Desription Unit m r d k max d k min d k k max k + k - max+ preserved by delaying the output of the first media-unit for max min d k d k seonds Based on this theorem, the playout deadline is derived The deadline given by Santoso (ase (a)) an be lowered in some situations (ase (b)) Theorem 1: Consider a single substream k in ase of bounded jitter k given by (3) Then smooth playout an be guaranteed whenever either one of the following starting onditions holds true max min (a) d k d k = k seonds elapsed after the arrival of the first media-unit, or (b) the ( k r + 1 )-th media-unit has arrived Proof: See [7] When using the shifting strategy, we need to provide for sub-stream k a total buffer b k of (for the derivation see [7]) 3 Model 3: Resynhronization 31 Introdution substream or server index ( m { 0,, n 1} ) requested display ate of eah substream at the lient maximum delay for substream k minimum delay for substream k average delay for substream k jitter for substream k maximum jitter of all substreams maximum upper deviation from d k due to jitter for substream k maximum lower deviation from d k due to jitter for substream k maximum upper deviation of all substreams Table 2 Model parameters b k 2 k max+ + = + k r [mu/se] Models 1 and 2 assured both intra-stream synhronization and inter-stream synhronization under the assumption that jitter is bounded In TM based networks, this assumptions typially holds true at least for the network beause we an express the aeptable QoS in parameters (5) 5

6 like throughput, delay, jitter or ell losses [3] If the endsystem is not using a real-time operating system, bounded jitter an not be guaranteed When jitter is unbounded, an appliation needs to make ertain assumptions on the amount of jitter sine buffer spae may be limited or the inrease in end-to-end delay by too large a buffer is unaeptable [3] To avoid buffer overflow in ase of unbounded jitter, we introdue model 3 Model 3 an be haraterized as a sheme for resynhronization We apply the onept of a buffer level ontrol to detet asynhrony To reover from asynhrony, we use feedbak messages to the servers Model 3 opes with asynhronies introdued by: lteration of the average delay Clok drift Server drop outs n alteration of the average delay leads to a gap 5 or a onentration in the ontinuous media stream gap ours when the average delay beomes longer, a onentration an be observed when the average delay beomes smaller The result of lok drift is very similar to the result of a hange in delay, but arises muh more slowly Clok drift introdues a skew mehanism is needed to adapt to hanging onditions in order to preserve synhronization without alloating additional buffer spae Solving the problem by additional buffering based on worst ase estimates might turn out to be a diffiult task beause hanging onditions are unpreditable Even if we sueed to get worst ase estimates, we have to be aware that, first, resoures are limited and that, seond, large playout buffers inrease the overall endto-end delay whih is not desired Furthermore, unontrolled buffering ompensates the problems to a ertain amount but will not resolve them over a long period of time Sine all the desribed disturbing fators affet the buffer level, the buffer level an be regarded as an indiator for upoming synhronization problems One a sink has disovered a problem, it has to take measures to restore synhronization Sine asynhrony is basially a shifting in the media stream, we only need to orret this shifting Corretive ations must be feed bak either to the soure or to the sink in order to restore synhrony The idea of taking the buffer level as an indiator is often referred to as buffer level ontrol Basi work in this area an be found in [13], [9] and [10] Our model will uses some of their basi ideas and extends them to an appliable solution for the synhronization problem In ontrast to the previous work, we take model 1 and 2 as a basis for synhronization and extend them with a buffer level ontrol We fous mainly on buffer requirements and parameter tuning The next setion examines models 1 and 2 with respet to a buffer level ontrol and presents a buffer model suitable to realize a buffer level ontrol Finally, we disuss the tuning of the model parameters 32 Buffer Level Control 321 System Model The onept of buffer level ontrol is often referred to as a ontrol loop [9] Soures transfer media-units over the network that arrive at the sink where they are buffered before playout The urrent buffer level is periodially measured, and if an ill buffer level is found, the appropriate steps are taken tions may affet either the buffer itself or the server In the former ase, the loop is plaed in the lient, in the latter ase it inludes the lient, the server and the network Koehler et al and Rothermel et al [9], [13] propose a synhronization sheme that does not adapt the playout behavior of the server tions are taken exlusively at the sink whether by hanging the onsumption rate or by skipping/pausing This kind of ontrol loop ompensates for disturbanes to a ertain amount depending on the alloated, available buffer spae but sarifies the realtime stream ontinuity We adopt to a onept where all omponents of the video server arhiteture are inluded in the ontrol loop similar to the approah of Cen et al [2] s shown in figure 2, the arhiteture applies feedbak ations to the soures via ontrol messages in order to maintain synhronization at the sink (a) Feedbak Filter The buffer level for substream k at time t is denoted by q tk This value is periodially passed to a filtering funtion S(q tk ) to filter short-term flutuations aused by jitter and to ompute the smoothed buffer level b tk Examples for filtering funtions are the geometri weighting smoothing funtion (with α as smoothing fator) [13], [2], [11]: b tk = S( q tk ) = α b t 1k + ( 1 α) q tk ( with α [,] 01 ), The main goal of filtering is to distinguish between buffer level hanges aused by jitter and long-term disturbanes If the filter is too sensitive, or no filter is used at all, jitter auses ations for resynhronization although no exeptional situation has ourred On the other hand, a filter that reats to slowly to hanging onditions takes ations too late with the result of a longer period of buffer starvation or overflow Thus, presentation quality suffers 5 The effet of a server drop out is also a gap in the media stream 6

7 Soure System under ontrol media-units Sink virtual buffer range real buffer range Control message o tk Control Funtion C b tk (b) Control Funtion Feedbak Filter b tk S(q tk ) Figure 2 System model for the buffer level ontrol [2] The smoothed buffer level b tk is passed to a ontrol funtion C( b tk ) that takes appropriate ations For eah substream buffer, a lower water mark LW k and an upper water mark UW k are defined When b tk falls below LW k or exeeds UW k, there arises the risk of starvation or overflow, respetively, produing an asynhrony If this happens, a resynhronization or adaptation phase is entered whose purpose is to move b tk bak into between LW k and UW k Depending on the extent of asynhrony, the ontrol funtion sends an offset o tk to the soure The soure either skips the number of media-units speified in the offset or pauses for a duration of o tk media-units We prefer this tehnique over an alteration of sheduling speed, respetively prodution rate, at the soure beause we think the latter is too resoure demanding and the QoS of other lients servied by the server might suffer The sink stays in its resynhronization phase for a time R in order to let the smoothed buffer level reat on the taken measures t the end of the resynhronization phase C( b tk ) ontrols again whether or not the buffer level b tk has moved bak in the normal area into between LW k and UW k If not, a new resynhronization phase is [13] started 322 Buffer Requirements Models 1 and 2 provide the buffer spae b k needed to ompensate jitter ([7]) In the following, we will denote b k as a kernel buffer pplying a buffer level ontrol only to this buffer is not suffiient sine eah buffer level within the range of b k must be regarded as normal due to the jitter effets We fix LW k and UW k to 1 and b k, respetively To realize a buffer level ontrol, we must admit buffer levels below and above the watermarks Otherwise, it is impossible to get the smoothed buffer level b tk below or above the watermarks We suggest the sheme of a so-alled virtual buffer as indiated in figure 3 by the dashed lines The virtual buffer inludes at least the real buffer omprising the kernel q tk b k q tk LW k lateny b tk buffer b k and an additional buffer The virtual buffer is exlusively used for the alulation of buffer levels below and above the real buffer This allows for a faster reation of the smoothing funtion S(q tk ) The mapping between the real buffer level and the virtual buffer level q tk is performed as follows: If neither buffer starvation nor buffer overflow ours, the real buffer level equals the virtual buffer level If a buffer overflow ours, then the virtual buffer is inreased for eah disarded media-unit while the real buffer level remains unhanged If a buffer starvation ours, then the virtual buffer is dereased eah time when the lient finds an empty buffer while the real buffer level remains unhanged If the normal state of the real buffer is restored by resynhronization measures, the virtual buffer level is reset to the real buffer level The size of b k strongly influenes the graefulness of the resynhronization 6 The smoothened buffer level b tk always has a lateny (see figure 3) ompared with the virtual buffer level q tk, ie q tk might be below LW k while b tk still needs some time to fall below Let b k = 0, for instane Then a buffer starvation ours before it is reognized by the ontrol funtion Hene, presentation quality suffers depending on the value of b We onsider the following three ases for the size of b k k Seleting b k = 0 yields no graefulness at all synhrony immediately affets presentation quality and is soon disovered by a viewer b k an be dimensioned suh that at least the period between the rise of asynhrony and the disovery by the ontrol funtion is overed For full graefulness, b k has to be hosen suh that asynhrony does not affet presentation at all The buffer spae has to over the period between rise, disovery and removal of asynhrony 6 Notie that the start-up lateny is also influened by the size of b k The larger b k is, the longer it takes until the first media unit of a substream is played out beause of the buffer level must exeed LW k before the playout deadline given by model 2 an be applied b k UW k Figure 3 Buffer model with virtual and real buffers b k b k 7

8 323 Parameter Tuning In our model, we have several parameters that must be hosen appropriately in order to trade-off reativeness and overhead (a) Smoothing Parameter α Obviously the lateny of reation to an asynhrony problem depends strongly on the behavior of S(q tk ) The more indolently S(q tk ) reats, the later a resynhronization phase is entered, the more buffer spae b k may be desired to ompensate for asynhrony as muh as possible On the other hand, the more sensitively S(q tk ) reats, the more often resynhronization is done unneessarily (due to the effet of jitter), the less buffer spae b k is needed to provide suffiient graefulness Hene, the tuning of S(q tk ) needs to trade-off between stability and reativity The hoie of S(q tk ), respetively, helps to determine the additional buffer spae b k For further onsideration we examine the filtering funtion given by () with respet to seond ase desribed above, ie the size of b k must over the period between rise and disovery of an asynhrony This ase is most interesting beause it is influened by S(q tk ) The behavior of the filter is determined by the parameter α : large value of α yields strong smoothing, a stronger onsideration of the past, and a more indolent reation small value of α yields weak smoothing, a stronger onsideration of the present, and a more sensitive reation n upper bound for the hoie of α is given by the available memory lower bound should be hosen suh that starvation/overflow events due to jitter an be distinguished from long term disturbanes ordingly, α should be set as high as possible while onsidering the buffer available In our experiments (see [6] for details) we found that a value of 06 or 07 for α is a good ompromise with respet to the buffer requirement and the number of neessary resynhronization ations (b) Degree o tk of Resynhronization Resynhronization is performed by sending an offset to the servers to move the buffer pointer b tk bak into the area between UW k and LW k The size of the offset o tk an be determined by two different strategies: fixed offset or variable offset Employing the fixed offset strategy, o tk is set to a onstant value Resynhronization is done slowly in subsequent resynhronization phases until synhronization is restored The value should not be hosen too high beause resynhronization, eg due to lok drift, is in the range of one or several media-units High values ould lead to osillation When applying the variable offset strategy, o tk varies depending on the extent of the ourred asynhrony Notie that when applying the variable offset strategy several resynhronization phases ould be needed as well beause at the time when the offset is alulated (determined by the filtering funtion) the total extent of asynhrony might not yet be reognized Nonetheless, synhronization is generally restored faster with a variable offset We will present some experimental results that ompare both strategies () Duration R of Resynhronization The duration of a resynhronization phase is defined by R fter R seonds the ontrol funtion one more ompares the smoothed buffer level with the watermarks gain, resynhronization ations may be taken R must be hosen suffiiently large that the server an perform the resynhronization, that is, the ation must already have taken effet on the lient Seleting R too small leads to numerous unneessary resynhronization phases where during eah phase the extent of asynhrony is overestimated Low values of R an result in osillation For large values of R several resynhronization phases are needed as well but the total time of resynhronization an beome unaeptably long So, in both ases presentation quality might be strongly influened 33 Experimental Results Based on the prototype implementation of the Video Server rray we have implemented the proposed synhronization sheme for evaluation purposes For implementation details, refer to [1] and [6] The following experiments have been performed on a dediated SUN Spar 10 workstation as a lient We used two videos, eah one distributed aross two servers: Bitburger ommerial, sampled at a rate of 16 fps (frames/se) with a total length of 462 frames 7 sene from the prodution Seaquest, sampled at a rate of 16 fps with a total length of 6710 frames We evaluated the effiieny of the buffer level ontrol mehanism The prototype of the Video Server rray is implemented in an TM-LN environment So we faed the problem that events like gaps or onentrations within a stream are rather unlikely Thus, we simulated these events in the servers The amount of asynhrony an be 7 in the ontext of video streams we use the term frame to denote a media-unit 8

9 speified by the user upon starting a server The server then periodially introdues drop outs in sheduling or sends several frames at one The lient attempts to resynhronize the server by sending bak offsets The following parameters have been used: Smoothing fator for the geometri weighting funtion: α = 07 mount of injeted asynhrony 8 : -8, -4, +4, +8 [frames] Resynhronization strategy: fixed offset and variable offset The variable offset was alulated by taking the differene between q tk and the watermarks The fixed offset was set onstant to 1 We alloated two buffer slots for the substream This orresponds to the kernel buffer b k Furthermore, for the additional buffering b k, we were using three buffer slots eah, above and below b k Consider figure 4, showing the virtual buffer level and the filtered buffer level over time for the resynhronization of a onentration of eight frames The y-axis shows the virtual buffer level while the x-axis denotes the onsumption period The upper bound (watermark) of the real buffer level is denoted by b while the lower bound is not shown in the figure Thus, b k equals b UW and b k is given by UW LW The virtual buffer level ranges from 1 to 108 beause we arbitrarily seleted a number of 50 frames above and below the real buffer to alulate the virtual buffer Figure 4 shows the ourse of resynhronization for the fixed offset strategy Virtual buffer level [frames] Resynhronization of an asynhrony of 8 frames Unfiltered buffer level Filtered buffer level Consumption period [frame] Figure 4 Resynhronization with the fixed offset The first resynhronization phase is entered exatly during onsumption period 142 when the filtered buffer level rosses the upper watermark UW The virtual buffer level rises up to 61, that is, four frames are disarded The lient 8 Negative values denote a drop out while positive values denote a onentration b UW LW then sends an offset of -1 to the server The lient undergoes 7 subsequent resynhronization phases at the whole These phases are indiated by the peaks Synhronization is restored exatly during onsumption period 180 when the filtered buffer level falls below UW We now onsider the same situation with the variable offset strategy The ourse of the filtered and unfiltered buffer level is depited in figure 5 Resynhronization starts during onsumption period Virtual buffer level [frames] gain, a number of four frames is disarded The lient first sends an offset of -3 frames to the server lready after this resynhronization ation, the buffer level falls below UW for a short period of time Now, two additional resynhronization phases are needed until synhrony is restored In eah phase an offset of -2 is sent to the server Synhronization is exatly restored during onsumption period 149 In ontrast to the fixed offset strategy, only 19 frame periods are needed to regain synhrony The results also show learly that resynhronization with a variable offset beomes even more effiient for larger asynhronies beause the adoption is performed faster 4 Conlusion Resynhronization of an asynhrony of 8 frames Unfiltered buffer level Filtered buffer level Consumption period [frame] Figure 5 Resynhronization with variable offset We have presented a sheme for intra- and inter-stream synhronization of distributed stored multimedia streams Our sheme omprises three models that assure synhronization in an environment with different delays, jitter, server drop-outs, lok drift, and an alteration of the average delay The mehanisms desribed do not rely on synhronized loks within the network In ontrast to existing synhronization solutions, the sheme is suitable for streams that are striped aross multiple server nodes as well as for a single server approah b UW LW 9

10 The sheme presented has been suessfully implemented in our video server prototype [1] where eah video is distributed (striped) over n server nodes knowledgment The work desribed in this paper was supported by the Siemens Nixdorf G, Munih 5 Referenes [1] C Bernhardt and E W Biersak The server array: salable video server arhiteture In W Effelsberg, Danthine, D Ferrari, and O Spaniol, editors, High- Speed Networks for Multimedia ppliations Kluwer Publishers, msterdam, The Netherlands, 1996 [2] S Cen, C Pu, R Staehli, C Cowan, and J Walpole distributed real-time MPEG video audio player In T D C Little and R Gusella, editors, Proeedings of the 5th International Workshop on Network and Operating System Support for Digital udio and Video (NOSSDV 95), volume 1018 of LNCS, pages , Durham, NH, pril 1995 Springer Verlag, Heidelberg, Germany [3] J S Corma Synhronisation Servies for Digital Continuous Media PhD thesis, University of Cambridge, Cambridge, England, Otober 1992 [4] W Effelsberg, T Meyer, and R Steinmetz Taxonomy on Multimedia-Synhronization In Proeedings of the Fourth Workshop on Future Trends of Distributed Computing Systems, Lisbon, Portugal, Sep 1993, pages Eyrolles, 1993 [5] J Esobar, C Patridge, and D Deutsh Flow synhronization protool CM Transations on Networking, 2(2): , pril 1994 [6] W Geyer Stream synhronisation in a salable video server array Master s thesis, Institut Eureom, Sophia ntipolis, Frane, September 1995 [7] W Geyer, C Bernhardt, and E Biersak synhronization sheme for stored multimedia streams In B Butsher, E Moeller, and H Push, editors, Interative Distributed Multimedia Systems and Servies (European Workshop IDMS 96, Berlin, Germany), volume 1045 of LNCS, pages Springer Verlag, Heidelberg, Germany, Mar 1996 [8] Y Ishibashi and S Tasaka synhronization mehanism for ontinuous media in multimedia ommuniations In IEEE Infoom 95, volume 3, pages , Boston, Massahusetts, pril 1995 [9] D Koehler and H Mueller Multimedia playout synhronization using buffer level ontrol In 2nd International Workshop on dvaned Teleservies and High-Speed Communiation rhitetures, pages , Heidelberg, Germany, September 1994 [10] T D C Little and F Kao n intermediate skew ontrol system for multimedia data presentation In Proeedings of the 3rd International Workshop on Network and Operating System Support for Digital udio and Video, pages , San Diego, C, November 1992 [11] H Massalin and C Pu Fine-grain adaptive sheduling using feedbak Computing System, 3(1): , 1990 [12] P V Rangan, H M Vin, and S Ramanathan Designing an on-demand multimedia servie IEEE Communiations Magazine, 30(7):56 65, July 1992 [13] K Rothermel and T Helbig n adaptive stream synhronization protool In T D C Little and R Gusella, editors, 5th International Workshop on Network and Operating System Support for Digital udio and Video, volume 1018 of LNCS, Durham, New Hampshire, US, pril 1995 Springer Verlag, Heidelberg, Germany [14] H Santoso, L Dairaine, S Fdida, and E Horlait Preserving temporal signature: way to onvey time onstrained flows In IEEE Globeom, pages , Deember 1993 [15] R Steinmetz and K Nahrstedt Multimedia: Computing, Communiations and ppliations Innovative Tehnology Series Prentie Hall, Englewood Cliffs, NJ,