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Supporting Adaptive Flows in Quality of Service Architecture

Supporting Adaptive Flows in Quality of Service Architecture
Supporting Adaptive Flows in Quality of Service Architecture

Supporting Adaptive Flows in a

Quality of Service Architecture

Andrew Campbell, Geoff Coulson and David Hutchison

Department of Computing,

Lancaster University,

Lancaster LA1 4YR, U.K.

{campbell, geoff, dh}@https://www.doczj.com/doc/984372608.html,

Abstract

Distributed audio and video applications need to adapt to fluctuations in delivered quality of service (QoS). By trading off temporal and spatial quality to available bandwidth, or manipulating the playout time of continuous media in response to variation in delay, audio and video flows can be made to adapt to fluctuating QoS with minimal perceptual distortion. In this paper we extend our previous work on a Quality of Service Architecture (QoS-A) by populating the QoS management planes of our architecture with a framework for the control and management of multi-layer coded flows operating in heterogeneous multimedia networking environments. Two key techniques are proposed: i) an end-to-end rate shaping scheme which adapts the rate of MPEG-coded flows to the available network resources while minimising the distortion observed at the receiver; and ii) an Adaptive Network Service, which offers “hard”

guarantees to the base layer of multi-layer coded flows, and “fairness” guarantees to the enhancement layers based on a bandwidth allocation technique called Weighted Fair Sharing. Keywords

Scalable flows, multimedia transport, dynamic QoS management, end-to-end QoS architecture 1. Introduction

The interplay between user-oriented quality of service (QoS) requirements [Zhang,95] multi-layer coded flows [Shacham,92], and end-to-end communication support [Campbell,95] is an interesting and active area of research [Pasquale,93], [Delgrossi,93], [Tokuda,92] [Hoffman,93]. The hierarchical coding techniques used by coders such as MPEG-1, MPEG-2 and H261 make possible a range of creative adaptation strategies whereby fluctuating bandwidth availability can be accommodated by selectively adding/ removing coding layers.

In this paper we introduce the concept of dynamic QoS management (DQM) which, by exploiting and extending the above principles, controls and manages multi-layer coded flows operating in heterogeneous, multicast, multimedia networking environments. Under the DQM heading, two main techniques are proposed to support scalable1 video over multimedia networks. These are i) end-to-end rate shaping which adapts the rate of multi-layer flows to the available network resources while minimising the distortion observed at the receiver, and ii) an adaptive network service which offers a combination of “hard” guarantees to the base layer of multi-layer coded flows, and “fairness” guarantees to the enhancement layers based on a new bandwidth allocation technique called weighted fair sharing. We also discuss a number of types of scaling object which are used to manage and control QoS. These include QoS filters which manipulate

1Media scaling is a general term, first proposed by [Delgrossi,93], we use to refer to the dynamic manipulation of media flows as they pass through a communications channel.

hierarchically coded flows as they progress through the communications system, QoS adaptors which scale flows at end-systems based on the flow's measured performance and user supplied QoS scaling policy, and QoS groups which provide baseline QoS for multicast flows.

All these components are inter-related in a Quality of Service Architecture (QoS-A) [Campbell,94] which has been developed over the last three years at Lancaster [Campbell,93]. This research has been conducted in co-operation with ATM switch manufacturer GDC (formally Netcomm Ltd) in the UK. The QoS-A is a layered architecture of services and mechanisms for quality of service (QoS) management and control of continuous media flows in multiservice networks. The architecture incorporates the following key notions: flows characterise the production, transmission and eventual consumption of single media streams (both unicast and multicast) with associated QoS; service contracts are binding agreements of QoS levels between users and providers; and flow management provides for the monitoring and maintenance of the contracted QoS levels. The realisation of the flow concept demands active QoS management and tight integration between device management, thread scheduling, communications protocols and networks.

The structure of the paper is as follows. We first present, in section 2, background on the QoS-A. Following this, we describe the salient features of scalable video flows in section 3 before describing, in section 4, the set of scaling objects used in our architecture together with the application programming interface (API) to DQM. Section 5 then presents our scheme for end-to-end DQM. In this section the detailed operation of a number of specific types of scaling object is described. Following this, section 6 introduces our adaptive network service, defines the notion of weighted fair share resource allocation, explains the rate control scheme used in the network, and describes the use of some network oriented QoS filters. Finally in section 7 we present the status of our work and offer some concluding remarks.

2. The Lancaster Quality of Service Architecture

The QoS-A is based on a set of principles that govern the realisation of end-to-end QoS in a distributed systems environment:

i) the integration principle states that QoS must be configurable, predicable and maintainable

over all architectural layers to meet end-to-end QoS [Campbell,93];

ii)the separation principle states that information transfer, control and management are functionally distinct activities in the architecture and operate on different time scales [Lazar,90];

iii)the transparency principle states that applications should be shielded from the complexity of handling QoS management [Campbell,92];

iv)the performance principle guides the division of functionality between architectural modules (viz. end-to-end argument [Saltzer,84], application layer framing and integrated layer processing [Clark,90], and the reduction of layered multiplexing [Tennenhouse,90]).

In functional terms, the QoS-A (see Figure 1) is composed of a number of layers and planes. The upper layer consists of a distributed applications platform augmented with services to provide multimedia communications and QoS specification in an object-based environment [Coulson,93]. Below the platform level is an orchestration layer which provides jitter correction and multimedia synchronisation services across multiple related application flows [Campbell,92]. Supporting this is a transport layer which contains a range of QoS configurable services and mechanisms. Below this, an internetworking layer and lower layers form the basis for end-to-end QoS support. See [Campbell,94] for full details on the QoS-A.

QoS management is realised in three vertical planes in the QoS-A. The protocol plane, which consists of distinct user and control sub-planes,is motivated by the principle of separation. We use

separate protocol profiles for the control and data components of flows because of the essentially different QoS requirements of control and data: control generally requires a low latency full duplex assured service whereas media flows generally require a range of non-assured, high throughput and low latency simplex services. The QoS maintenance plane contains a number of layer specific QoS managers. These are each responsible for the fine grained monitoring and maintenance of their associated protocol entities. For example at the orchestration layer the QoS manager is interested in the tightness of synchronisation between multiple related flows. In contrast, the transport QoS manager is concerned with intra-flow QoS such as bandwidth, loss, jitter and delay. Based on flow monitoring information and a user supplied service contract, QoS managers maintain the level of QoS in the managed flow by means of fine grained resource tuning strategies. The final QoS-A plane pertains to flow management, which is responsible for flow establishment (including end-to-end admission control, QoS based routing and resource reservation), QoS mapping (which translates QoS representations between layers) and QoS Scaling (which constitutes QoS filtering and adaptation for coarse grained QoS maintenance control).

Figure 1: QoS-A

Recent work on the QoS-A has concentrated on realising the architecture in an environment comprising an enhanced Chorus micro-kernel [Coulson,95] and an enhanced multimedia transport service and protocol [Campbell,94] in the local ATM environment.

3. Scalable Video Flows

The fundamental design goal of digital audio-visual information representation schemes in the past several decades has been that of compactness: describe the signal's content with as few bits as possible. The algorithmic foundation of this work has been based on the assumption that information transport occurs over constant bandwidth and constant delay channels: an assumption that does not necessarily hold valid in an environment of packet switched networks working on the premise of multiplexing gain.

Our primary focus in this paper is MPEG-2 coded video. Within this framework, there are two alternative techniques which underpin QoS adaptation: intrinsic and extrinsic techniques. The former are provided by the encoder, and are embedded in the coded bitstream. The latter are provided by QoS filters that operate directly on the compressed bitstream, performing the desired manipulations. Their difference lies in their complexity and performance. Intrinsic techniques can have a very simple implementation, but as we will see offer only a discrete range of possibilities.

Extrinsic techniques are computationally more complex but can operate on a continuum of possibilities. In the following we briefly describe the architecture of MPEG-2, with particular emphasis on the intrinsic adaptation capabilities it provides in the form of scalability profiles. Extrinsic adaptation can be provided through the use of dynamic rate shaping QoS filters [Eleftheriadis,95], which are discussed in section 5.2.1.

3.1 MPEG-2

The algorithmic foundation of MPEG is motion-compensated, block-based transform coding MPEG-2 [H.262,94]. Each block is transformed using the Discrete Cosine Transform (DCT), and is subsequently quantised. Quantisation is the sole source of quality loss in MPEG-2 and, of course, a major source of compression efficiency. The quantised coefficients are converted to a one-dimensional string using a zig-zag pattern and then run-length encoded. There are three types of pictures in a sequence: I, P, and B. I or intra pictures are individually coded, and are fully self-contained. P pictures are predicted from the previous I or P picture, while B (or bi-directional) pictures are interpolated from the closest past and future I or P pictures. Prediction is motion-compensated: the encoder finds the best match of each macroblock in the past or future picture, within a prespecified range. The displacement(s), or motion vector(s), is sent as side information to the decoder.

In order to increase the coding efficiency, MPEG-2 relies heavily on entropy coding. Huffman codes (variable length codewords) are used to represent the various bitstream quantities (run-length codes, motion vectors, etc.). As a result, the output of an MPEG-2 encoder is inherently a variable rate bitstream: the ratio of bits per pixel varies from one block to the next. In order to construct a constant rate bitstream, rate control is used. This is achieved by connecting a buffer to the output of the encoder. The buffer is emptied at a constant rate, and its occupancy is fed back to the encoder. This information is used to control the selection of the quantiser for the current macroblock. High buffer occupancy leads to more coarsely quantised coefficients, and hence less bits per block, and vice versa. Through this self-regulation technique one can achieve a constant output rate.

For transport purposes, video and audio are multiplexed according to the system layer of MPEG-2, which defines the packetisation structure and synchronisation algorithms between the audio and video signals. Two different packetisation structures are defined, namely program streams and transport streams. Both are logically constructed from “packetised elementary stream”(PES) packets; these are the basic units in which individual audio, video, and control information is carried. Program streams are designed for use in relatively error-free environments, use variable length packets, and combine PES packets that have a common time base. Transport streams are designed for noisy channels, utilise fixed-length packets (188 bytes), and can carry programs with independent time bases. The system layer's timing model is based on the assumption that a constant delay transport mechanism is used. Although deviations from this are allowed, the way to address them is not specified. All timing information is based on a common system clock, and timestamps (i.e. an absolute timing method) ensure proper inter-media synchronisation between the audio and video signals upon presentation.

3.2 MPEG-2 Scalability Modes

From the above discussion it is clear that MPEG-2 is especially tuned for transmission over traditional constant bandwidth and delay channels. Some support for flexible and/or robust transmission is provided through the use of scalability modes for channels exhibiting dynamic behaviour. The situation is even more challenging for scalable flows, i.e. in cases where the available bandwidth may vary over time. However, here benefits of multiplexing gain are possible.

MPEG-2 provides for the simultaneous representation of a video signal at various different levels of quality, through the use of multiple independent bitstreams or sub-signals. This is achieved through the use of pyramidal, or hierarchical coding: one first constructs a coarse or base representation of the signal, and then produces successive enhancements. The latter assume that the

base representation is available, and only encode the incremental changes that have to be performed to improve the quality. There are four different scalability modes: spatial, SNR, temporal, and data partitioning. MPEG-2 allows the simultaneous use of up to two different scalability modes (except from data partitioning) in any combination, hence resulting in a 3-level representation of the signal. In spatial scalability, the base and enhancement layers operate at different spatial resolutions (e.g. standard TV and HDTV). In SNR scalability, both layers have the same resolution and the enhancement refines the quantisation process performed in the base layer. In temporal scalability the enhancement layer increases the number of frames per second of the base layer (e.g., from 30 frames per second to 60 fps). Data partitioning is slightly different from the other three scalability modes in the sense that the encoder does not maintain two different prediction loops and hence the base layer is not entirely self-contained. Its benefit is that it can be applied even in single-layer encoders.

As an example, the spatial scalability mode can be used to transmit digital TV in both standard and HDTV formats. Although two sub-signals are actually generated, this is not identical to simulcast: the total bandwidth required is much smaller, since the HDTV layer uses the base, standard TV layer as a reference point. Scalability can be very useful in transmission of video over adaptive channels.

3.3 Discrete and Continuous QoS Adaptation

Although MPEG-2's scalability features are useful in resolving heterogeneity problems described above [Delgrossi,93], and are useful in numerous applications, their use in continually QoS-varying channels is problematic. This is because they only allow the representation of the signal at a fixed number of discrete quality points (temporal or spatial resolution, or spatial quality). These points are typically significantly apart, and transitions between the two are perceptually significant. Table 1 [Paek,95] shows an example of hybrid scalability with spatial (E1) and SNR (E2) enhancement layers.

layer name profile symbol frame size bit rate subjective QoS

base layer main BL 304x112 0.32 Mbps VHS

enhancement 1 layer spatial E1 608x224 0.83 Mbps super VHS

enhancement 2 layer SNR E2 608x224 1.85 Mbps laser disc Table 1: MPEG-2 hybrid scalable bitstream using spatial and SNR scalability (24 fps)

Consider, for example, a channel that temporarily undergoes rate variability for a period of a few seconds. Switching to a lower quality point (by discarding the enhancement layer(s)) for such a brief interval will create a perceptual “flash” that is very annoying to users. An additional issue is that, as soon as compression parameters are established, it is impossible to modify them later on (after compression is completed). Hence scalability modes can only really be used for well-defined, simple channels that vary slowly. Since the wide range of different access mechanisms to multimedia information makes it very difficult to select a priori a set of universally interoperable coding parameters, it is necessary to provide extrinsic mechanisms that allow the representation of the “signal at a continuum of qualities and rates” [Delgrossi,93], so that scalable flows can be accommodated.

This is possible through the use of a class of dynamic rate shaping QoS filters [Eleftheriadis,95b] and the provision of adaptive network services - providing a QoS continuum for fully scalable flows. The adaptive service introduced in this paper uses explicit feedback from network resource management to dynamically shape the video source based on available network resources. Some benefits of an adaptive scheme are non-reliance on video modelling techniques and statistical QoS specification and specific support for the semantics of scalable video flows e.g.

MPEG-2 scalable profiles. Dynamic rate shaping filters manipulate the rate of MPEG-coded video, matching it to the available bandwidth (indicated by the adaptive service) while minimising the distortion observed by the receiver.

4. Scaling Objects and API Extensions

In this section, we introduce a set of scaling objects used to manipulate hierarchically coded flows as they progress through the communications system. These comprise QoS adaptors, QoS filters and QoS groups as defined in section 1. We also introduce an extension to the QoS-A application programmer’s interface which gives access to the scaling object types.

4.1 Scaling Objects

4.1.1 QoS Adaptors

QoS adaptors are used in conjunction with the QoS-A flow monitoring function to ensure that the user and provider QoS specified in the service contract are actually maintained. In this role QoS adaptors are seen as quality of service arbiters between the user and network. QoS adaptors scale flows at the end-systems based on a user supplied QoS scaling policy (see section 4.2) and the measured performance of on-going flows. QoS adaptation is discussed in detail in section 5.

4.1.2 QoS Filters

QoS filters manipulate multi-layer coded flows [Shacham,92], [Hoffman,93] [Zhang,95] at the end-systems and as they progress through the network. We describe three distinct styles of QoS filters:

i) shaping filters, which manipulate coded video and audio by exploiting the structural

composition of flows to match network, end-system or application QoS capability; shaping filters are generally situated at the edge of the network at the source; they require non-trivial computational power; examples are the dynamic rate shaping (DRS) filter and the source bit rate (SBR) filter - see section 5.2;

ii) selection filters, which are used for sub-signal selection and media dropping (e.g. video frame dropping) are of low complexity and low computational intensity - selection filters are designed to operate in the network and are located at switches; they require only minimal computational power; examples are sub-signal filter, hierarch filter, hybrid filter -see section 6.3;

iii)temporal filters, which manipulate the timing characteristics of media to meet delay bound QoS are also low in complexity and trivial computationally - temporal filters are generally placed at receivers or sinks of continuous media where jitter compensation or orchestration of multiple related media is required; examples are sync filter, orch filter - see section 5.3).

4.1.3 QoS Groups

Before potential senders and receivers can communicate they must first join a QoS group [Aurrecoechea,95]. The concept of a QoS group is used to associate a baseline QoS capability to a particular flow. All sub-signals of a multi-layer stream can be mapped into a single flow and multicast to multiple receivers [Shacham,92]. Then, each receiver can select to take either the complete signal advertised by the QoS group or a partial signal based on resource availability. Alternatively each sub-signal can be associated with a distinct QoS group. In this case, receivers “tune” into different QoS groups (using signal selection) to build up the overall signal. Both methods are supported in DQM. Receivers and senders interact with QoS groups to determine what the baseline service is, and tailor their capability to consume the signal by selecting filter styles and specifying the degree of adaptability sustainable (viz. discrete, continuous; see section 4.2).

4.2 QoS Specification API

In the original QoS-A we designed a service contract based API which formalised the end-to-end QoS requirements of the user and the potential degree of service commitment of the provider. In this section, we detail extensions to the flow specification, QoS commitment and QoS scaling clauses of the service contract required to accommodate adaptive multi-layer flows. The API presented here is not complete in that there are no primitives given for establishing and renegotiating connections or for manipulating QoS groups. Full details of these aspects are given in [Campbell,94].

typedef enum {MPEG1, MPEG2, H261, JPEG} mediaType;

typedef enum {besteffort, adaptive, deterministic} commit;

typedef enum {continuous, discrete} adaptMode;

typedef enum {

DRS, SBR, sub_signal, hierarch, hybrid, sync, orch

} filterStyle;

typedef struct {

adaptMode adaptation;

filterStyle filtering;

events adaptEvents;

actions newQoS;

signal bandwidth;

signal loss;

signal delay;

signal jitter

} QoSscalingPolicy;

typedef struct {

gid flow_id;

mediaType media;

commit commitment;

subFlow BL;

subFlow E1;

subFlow E2;

int delay;

int loss;

int jitter;

QoSscalingPolicy qospolicy

} flowSpec;

Multi-layered flows are characterised by three sub-signals in the flowSpec flow specification: a base layer (BL) and up to two enhancement layers (E1 and E2). Each layer is represented by a frame size and subjective or perceptive QoS as illustrated in Table 1. Based on these characteristics, the MPEG-2 coder [Paek,95], [Eleftheriadis,95] determines approximate bit rate for each sub-layer. In the case of MPEG-2's hybrid scalability, BL would represent the main profile bit rate requirement (e.g. 0.32 Mbps) for basic quality, E1 would represent the spatial scalability mode bit rate requirement (e.g. 0.83 Mbps) for enhancement, and E2 would represent the SNR scalability mode bit rate requirement (e.g. 1.85 Mbps) for further enhancement. The remaining flow specification performance parameters for jitter, delay and loss are assumed to be common across the all sub-signals (i.e. a single layer of a multi-layer video flow). The QoS commitment field has been extended to offer an adaptive network service that specifically caters for the needs of scalable audio and video flows in heterogeneous networking environments (see section 6).

The QoSscalingPolicy field of the flowSpec characterises the degree of adaptation that a flow can tolerate and still achieve meaningful QoS. The scaling policy consists of clauses that cover adaptation modes, QoS filter styles, and event/ action pairings for QoS management purposes. Two types of adaptation mode are supported: continuous mode, for applications that can exploit any availability of bandwidth above the base layer; and discrete mode for applications which can only accept discrete improvement in bandwidth based on a full enhancement (viz. E1, E2).

The QoS scaling policy provides user-selectable QoS adaptation and QoS filtering. While receivers select filter styles to match their capability to consume media at the receiver (from the set of temporal filters), senders select filter styles to shape flows in response to the availability of network resources such as bandwidth and delay (from the set of shaping filters). Network oriented filters (i.e. selection filters) can be chosen by either senders or receivers.

In addition, senders and receivers can both select periodic performance notifications including available bandwidth, measured delay, jitter and losses for on-going flows. The signal fields in the scaling policy allow the user to specify the interval over which a QoS parameter is to be monitored and the user informed. Multiple signals can be selected depending on application needs.

5. Dynamic QoS Management

5.1 Architectural Components

Dynamic QoS management spans the QoS-A QoS maintenance and flow management planes. In the QoS maintenance plane, the most important aspect of DQM is QoS adaptation in the end-systems. Based on the receiver supplied QoS scaling policy, QoS adaptors take remedial action to scale flows, inform the user of a QoS indication and degradation, fine tune resources and initiate complete end-to-end QoS renegotiation based on a new flowSpec [Campbell,94]. In the flow management plane, DQM consists of two sub-components: QoS group management maintains and advertises QoS groups created by senders for the benefit of potential receivers; filter management [Yeadon,94] instantiates and reconfigures filters in a flow at optimal points in the media path at flow establishment time, when new receivers join QoS groups or when a new flowSpec is given on a QoS renegotiation.

In implementation, each of these architectural modules has well defined interfaces and methods defined in CORBA IDL. CORBA [OMG,93] runs on the end-systems and in the ATM switches, providing a seamless object oriented environment throughout the communication system base (see [Aurrecoechea,94] for full details).

5.1.1 Illustrative Scenario

DQM can be viewed as operating in three distinct domains:

i) sender-oriented DQM, where senders select source filters and adaptation modes, and

establish flow specifications. The sender-side transport protocol provides periodic bandwidth and delay assessments to the source filters (i.e. DRS or SBR filters) which regulate the source flow. Senders create QoS groups which announce the QoS of the flow to receivers via QoS group management;

ii) receiver-oriented DQM, where receivers join QoS groups and select the portion of the signal which matches their QoS capability. Receiver selected network based filters propagate through the network and perform source and signal selection. In addition, receiver-based QoS filters (i.e. sync-filter and orch-filter) are instantiated by default unless otherwise directed. These filters are used to smooth and synchronise multiple media. The receiver-side transport protocol provides bandwidth management and produces adaptation signals according to the QoS scaling policy;

iii) network-oriented DQM, which provides an adaptive network service (see section 6) to

receivers and senders. Network level QoS filters (i.e. sub-signal, hierarch and hybrid-filters) are instantiated based on user selection, and propagated in the network under the control of filter management.

In Figure 1 a sender at end-system A creates a flow by instantiating a QoS group which announces the characteristics of the flow (viz. layer, frame size, subjective quality) and its adaptation mode. Receivers at end-systems B, C and D join the QoS group. In the example scenario shown the receivers each “tune” into different parts of the multi-layer signal: C takes BL, the main profile (which constitutes a bandwidth of 0.32 Mbps for VHS perceptual QoS), B takes BL and E1 (which constitutes an aggregate bandwidth of 1.15 Mbps for super VHS perceptual QoS), and D takes the complete signal BL+E1+E2 (which constitutes an aggregate bandwidth of 3 Mbps for laser disc perceptual QoS). In this example the complete signal is multiplexed onto a single flow, therefore, sub-signal selection filters are propagated by filter management. Receivers, senders, or any third party or filter management can select, instantiate and modify source, network and receiver-based QoS filters.

Figure 1: Dynamic QoS management of scalable flows

5.2 Sender-Oriented DQM

Figure 2 shows the functions of the sender-side transport protocol supporting dynamic QoS management, and the interface to a dynamic rate shaping filter. Currently, senders can select from two types of shaping filter at the source: dynamic rate shaping (DRS) and source bite rate (SBR) QoS filters. Both of these QoS filters manipulate the signal to meet the available bandwidth by keeping the signal meaningful at the receiver. The sender-side transport mechanisms includes a QoS adaptor, flow monitor and media scheduler. Bandwidth updates are synchronously received by the flow monitor mechanism from the network as part of the adaptive service (described in section 6). The QoS adaptor is responsible for synchronously informing the source filter of the current bandwidth availability (B flow) and measured delay (D flow), and calculating new schedules and deadlines for transport service data units [Coulson,95]. Media progresses from the source filter to the TSAP, and is scheduled by the media scheduler to the network at the NSAP based on the calculated deadlines.

The QoS adaptor is also responsible for informing the sending application of the on-going QoS based on options selected in the QoS scaling policy. Informing the application of the current state of the resources associated with a specific flow is key in implementing adaptive applications in end-systems. In this case the application manages the flow by receiving updates and interacting with the QoS adaptor to adjust the flow, e.g. change adaptation mode from continuous to discrete,

request more bandwidth for BL, E1 and E2, or change the characteristics of the source filter, etc.

5.2.1 The DRS Filter

We define rate shaping as an operation which, given an input video bitstream and a set of rate constraints, produces a video bitstream that complies with these constraints. For our purposes, both bitstreams are assumed to meet the same syntax specification, and we also assume that a motion compensated block-based transform coding scheme is used. This includes both MPEG-1 and MPEG-2, as well as H.261 and so-called “motion” JPEG.

Although a number of techniques have been developed for the rate shaping of live sources [Kanakia,93] these cannot be used for the transmission of pre-compressed material (e.g. in VoD systems). The dynamic rate shaping filter is interposed between the encoder and the network and ensures that the encoder's output can be perfectly matched to the network's quality of service characteristics. The filter does not require interaction with the encoder and hence is fully applicable to both live and stored video applications.

Figure 2: Sender-side transport QoS mechanisms

Because the encoder and the network are decoupled, universal interoperability can be achieved both between codecs and networks, and also among codecs with different specifications. An attractive aspect is the existence of low-complexity algorithms which allow software-based implementation in high-end computers. In order for rate shaping to be viable it has to be implementable with reasonable ease while yielding acceptable visual quality. With respect to complexity, the straightforward approach of decoding the video bitstream and recoding it at the target rate would be obviously unacceptable; the delay incurred would also be an important deterrent. Hence only algorithms of complexity less than that of a cascaded decoder and encoder are of practical interest. These algorithms operate directly in the compressed domain of the video signal, manipulating the bitstream so that rate reduction can be effected. In terms of quality, it should be noted that recoding does not necessarily yield optimal conversion; in fact, since an optimal encoder (in an operational rate-distortion sense) is impractical due to its complexity, recoding can only serve as an indicator of an acceptable quality range. In fact, regular recoding can be quite lacking in terms of quality, with dynamic rate shaping providing significantly superior results.

The rate shaping operation is depicted in Figure 3. Of particular interest is the source of the rate constraints B flow(t). In the simplest of cases, B flow(t) may be just a constant and known a priori (e.g. the bandwidth of a circuit-switched connection). It is also possible that B flow(t) has a well known statistical characterisation (e.g. a policing function). In our approach B flow(t) is generated by the adaptive network service.

Figure 3: Dynamic rate shaping scheme

The objective of a rate shaping algorithm is to minimise the conversion distortion, i.e.: min

()()

()()

B t B T t y t y t

The attainable rate variation (B ^B) is in practice limited, and depends primarily on the number

of B pictures of the bitstream; no assumption is made on the rate properties of the input bitstream,which can indeed be arbitrary. There are two fundamental ways to reduce the rate:

i) by modifying the quantised transform coefficients by employing coarser quantisation, and ii) by eliminating transform coefficients.

In general, both schemes could be used to perform rate shaping; requantisation, however, leads to recoding-like algorithms which are not amenable to fast implementation and do not perform as well as selective-transmission ones. A selective transmission approach gives rise to a family of different algorithms, that perform optimally under different constraints; for full details see [Eleftheriadis,95b].

5.3 Receiver-Oriented DQM

QoS adaptors, which are resident in the transport protocol at both senders and receivers,arbitrate between the receiver specified QoS and the monitored QoS of the on-going flow. In essence the transport protocol “controls” the progress of the media while the receiver “monitors and adapts” to the flow based on the flow specification and the scaling policy. When the transport protocol is in monitoring mode [Campbell,94] the flow monitor uses an absolute timing method to determine frame receptions times based on timestamps/sample-stamps [Jeffay,92], [Jacobson,93],[Shenker,93]. The flow monitor, as shown in Figure 4, updates the flow state to include these measured reception times statistics. Based on these flow statistics, the sync-filter (see section 5.3.3) derives new playout times used by the media scheduler to adjust the playout point of the flows to the decoding delivery device.

QoS mechanisms that intrinsically support such adaptive approaches were first recognised in the late 70's by Cohen [Cohen,77] as part of research in carrying voice over packet-switched networks. More recently, adaptive QoS mechanisms have been introduced at part of the Internet suite of application-level multimedia tools (e.g., vat [Jacobson,93], ivs [Turletti,93], and vic [McCanne,94]). Vat which is used for voice conferencing, recreates the timing characteristics of voice flows by having the sender timestamp on-going voice samples. The receiver then uses these timestamps as a basis to reconstruct initial flow, removing any network induced jitter prior to playout. These multimedia tools are widely used in the Internet today and have proved moderately successful - given the nature of best effort delivery systems, i.e., no resource reservation is made. In the near future, however, an integrated services Internet [Braden,94] will offer support for flow reservation (e.g., RSVP [Zhang,95]) and new QoS commitments (e.g., predictive QoS [Shenker,95]) which are more suitable for continuous media delivery.

Receiver-oriented adaptation can be broken down into a number of receiver-side transport functions, i.e. bandwidth management, late frame management and delay jitter management, which are described in the following sub-sections (please refer to figure 4). We argue in this paper that these adaptive QoS mechanisms are inherently part of the transport potocol and not, as in the case of vat, ivs and vic, part of the application domain itself.

Figure 4: Receiver-side transport QoS mechanisms

5.3.1 Bandwidth Management

Bandwidth management receives bandwidth indications in the control message portion of the TSDU (or in separate control messages) and adapts the receiver appropriately. The adaptive service, built on the notion of weighted fair share resource allocation, (see section 6.1) periodically informs the receiver that more bandwidth is available or announces that the flow is being throttled back. Bandwidth management only covers the enhancement signals of multi-resolution flows. The base layer is not included since resources are guaranteed to the base layer. The announcement of available bandwidth on a flow allows the receiver to take either a full or partial enhancement layer. The choice depends on whether the flow is in continuous or discrete adaptation mode.

5.3.2 Late Frame Management

Late frame management monitors late arrivals in relation to the loss metric and the current playout times and takes appropriate action to trade off timeliness and loss. Packets that arrive after their expected playout points are discarded by the media scheduler and the late-packet metrics in the playout statistics are updated. The media scheduler is based on a split-level scheduler architecture [Coulson,95] which provides hard deadline guarantees to base layer flows via admission control, and best effort deadlines to enhancements layers. Some remedial action may be taken by the QoS adaptor should the loss metric exceed the loss parameter in the flow specification. If the QoS adaptor determines that too many packet losses have occurred over an era, it pushes out the playout time to counteract the late state of packets from the network. Similarly, if loss remains well within the prescribed ranges then the QoS adaptor will automatically and incrementally “pull in” the playout time until loss is detected.

5.3.3 Delay Jitter Management

Our transport protocol utilises sync-filters for delay-jitter management by calculating the playout times of flows based on the user supplied jitter parameter in the flow specification1. Sync filters calculate the mean and variation in the end-to-end delay based on reception times measured

1Temporal filters can also operate on multiple related audio and video flows to provide low-level orchestration management (in conjunction with the orch-filter). These filter types, however, are not discussed further in this paper.

by the flow monitor. Sync filters take the absolute, mean and variation in delay into account when calculating the playout estimate. A smoothing factor based on a linear recursive filtering mechanism characterised by a smoothing constant is used to dampen the movement of the playout adjustment. Intuitively, the playout time needs to be set “far enough” beyond the delay estimate so that only a small fraction of the arriving packets are lost due to late packets. The QoS adaptor trades off late packets versus timeliness based on the delay and loss parameters in the flow specification. The objective of delay jitter management is to pull in the playout offset, while the objective of late-packet management is not to exceed the loss characterised in the service contract. The QoS adaptation manager moderates between timeliness and loss. Based on these metrics the adaptation policy can adjust the damping factor, and acceptable ranges over which the playout point can operate.

time

Figure 5: Sync-filter: timeliness and packet loss regulation Figure 5 [Zhang,94] shows packets arriving at the receiving end-system. Each packet includes a timestamp used in calculating the flow statistics for delay-jitter management. Selection of the playout point is important: an aggressive playout time which favours timeliness (such as t1) will results in a large number of late-packets. In contrast, a conservative playout point (such as t3) will be less responsive and timely but will result in no identifiable packet loss. In the DQM scheme late packets are the same as lost packets, and therefore the loss parameter in the flow specification moderates. An optimum playout schedule is represented by t2 in the diagram; here, continuous media delivery benefits from timely delivery with the exception of some packet loss - which may be deemed acceptable to the receiver in media perception terms.

6. Adaptive Network Service

The adaptive network service provides “hard” guarantees to the base layer (BL) of a multi-layer flow and “weighted fair share” (WFS) guarantees to each of the enhancement layers (E1 and E2). To achieve this, the base layer undergoes a full end-to-end admission control test [Coulson,95]. On the other hand, enhancement layers are admitted without any such test but must compete for residual bandwidth among all other adaptive flows. Enhancement layers are rate controlled based on explicit feed back about the current state of the on-going flow and the availability of residual bandwidth.

6.1 Weighted Fair Share Resource Partitioning

Both end-system and network communication resources are partitioned between the deterministic and adaptive service commitment classes. This is achieved by creating and maintaining “firewall” capacity regions for each class. Resources reserved for each class, but not currently in use can be “borrowed” by the best effort service class on condition of pre-emption [Coulson,95]. The adaptive service capacity region (called the available capacity region and denoted by B avail) is further sub-divided into two regions: i) guaranteed capacity region (B guar),

which is used to guarantee all base rate layer flow requirements; and ii) residual capacity region (B resid ), which is used to accommodate all enhancement rates where competing flows share the residual bandwidth.

Three goals motivate our adaptive service design. The first goal is to admit as many base layer (BL) sub-signals as possible. As more base layers are admitted the guaranteed capacity region B guar grows to meet the hard guarantees for all base signals. In contrast, the residual capacity region B resid shrinks as enhancement layers compete for diminishing residual bandwidth resources.The following invariants must be maintained at each end system and switch:

B avail = B guar +B resid , and

BL (i )i =1

N

∑≤B avail

Our second goal is to share [Steenstrup,94], [Tokuda,92] the residual capacity B resid among

competing enhancement sub-signals based on a flow specific weighting factor, W , which allocates residual bandwidth in proportion to the range of bandwidth requested that in turn is related to the range of perceptual QoS acceptable to the user. In DQM, residual resources are allocated based on the range of bandwidth requirements specified by the users (i.e. BL.. BL+E1+E2 is the range of bandwidth required, e.g. from 0.32 Mbps to 3 Mbps for the hybrid scalable MPEG-2 flow in Table 1). As a result, as resources become available each flow experiences the same “percentage increase” in the perceptible QoS, we call this weighted fair share (WFS). W is calculated for each flow as the ratio of a flow's perceptual QoS range to the sum of all perceptual QoS ranges.

W (i )=(BL i

+E 1i +E 2i )

/

(BL j +E 1j j =1

N

∑+E 2j )

All residual resources B resid are allocated in proportion to the W metric. Using this factor we

calculate the proportion of residual bandwidth allocated to a flow to be B wfs (i)=W (i).B resid and the proportion of the available bandwidth allocated to be B flow (i)= B wfs (i)+ BL(i).

Our third and final goal is to adapt flows both discretely and continuously, based on the adaptation mode. In the discrete mode no residual bandwidth is allocated by the WFS mechanism unless a complete enhancement can be accommodated (i.e., B wfs (i) = E1(i) | E1(i)+E2(i), e.g. 0.83Mbps or 2.68 Mbps from Table 1). In continuous mode any increment of residual bandwidth B wfs (i) can be utilised (i.e. 0 < B wfs (i) ≤ E1(i) + E2(i), e.g. from 0 to 2.68 Mbps from Table 1).6.2 Rate Control Scheme

We build on the rate-based scheme described in [Campbell,94] where the QoS-A transport protocol at the receiver measures the bandwidth, delay, jitter and loss over an interval which we call an “era”. An era is simply defined as the reciprocal of the frame rate in the flow specification (e.g. for a frame rate of 24 frames per second as shown in Table 1 the interval era is approximately 42 ms). The receiver-side transport protocol periodically informs the sender-side about the currently available bandwidth, and the measured delay, loss and jitter. This information is used by the source or virtual source 1 to calculate the rate to use over the next interval. The reported rate is temporally correlated with the on-going flow. An important result in [Kanakia,93] shows that variable rate encoders can track QoS variations as long as feedback is available within four frame times or less. This feedback is used by the dynamic rate shaping filter and network based filters to control the data generation rate of the video or the selection of the signal respectively. In the case of dynamic rate shaping, the rate is adjusted while keeping the perceptual quality of the video flow meaningful to the user.

Based on the concept of eras, control messages are forwarded from the receiver-side transport protocol to either virtual source or the source-side transport protocol using reverse path

1

We use the term virtual source to represent a network switch that modifies the source flow via filtering.

forwarding. A core-switch [Ballardie,93] where flows are filtered is always considered to be a virtual source for one or more receivers; for full details see [Aurrecoechea,94]. The WFS mechanism updates the advertised rate as the control messages traverse the switches on the reserve path to the source or virtual source. Therefore any switch can adjust the flow's advertised rate before the source or virtual source receives the rate based control message. The source-side transport protocol hands the measured delay and aggregate bandwidth off (B flow ) to the dynamic rate shaping filter.

DQM maintains flow state at each end-system and switch that a flow traverses. Flow state is updated by the WFS algorithm and the rate-based flow control mechanism and comprises:

i) capacity (viz. B avail , B guar , B resid );ii) policy (viz. filterStyle, adaptMode);iii)flowSpec (viz. BL, E1, E2) ;iv)WFS share (viz., B flow , B wfs , W ).

The end-systems hold an expanded share tuple for measured delay, loss and jitter metrics. An admission control test is conducted at each end-system and switch on route to the core for the base layer signal. This test simply determines whether there is sufficient bandwidth available to guarantee the base layer BL given the current network load:

BL (j )j =1

N

∑≤B avail

If the admission control test is successful, WFS determines the additional percentage of the

residual bandwidth made available (B wfs ) to meet any enhancement requirements in the flowSpec:

B wfs (i )=W fact (i ).(B avail ?

BL (j )j =1

N

∑)The WFS rate computation mechanism causes new B wfs rates to be computed for all adaptive

enhancement signals that traverse the output link of a switch; switches are typically non-blocking which means the critical resources are the output links, however, our scheme can be generalised to other switch architectures [Coulson,95].6.3 Network Filtering

Currently our scheme supports two types of selection filters in the network. These are low complexity and computationally simple filters for selecting sub-signals. Selection filters do not transform the structure of the internal stream, i.e. they have no knowledge of the format of the encoded flow above differentiating between BL, E1 and E2 sub-signals. The two basic types of section filter used are:

i) sub-signal filters : these manipulate base and enhancement layers of multi-layer video multiplexed on a single flow. The definition of sub-signals is kept general here; a flow may be comprised of an anchor and scalable extensions or the I and P pictures of MPEG-2's simple profile, or the individual hybrid scalable profile. Sub-signal filters are installed in switches when a receiver joins an on-going flow;ii) hierarchical filters : these manipulate base and enhancement layers which are transmitted and received on independent flows in a non multiplexed fashion. In functional terms sub-signal and hierarchical filters can be considered to be equivalent in some cases. In sub-signal filtering one flow characterises the complete signal and in hierarchical-filtering a set of flows characterise the complete signal.

In addition, hybrid filters combine the characteristics of sub-signal and hierarchical filtering techniques to meet the needs of complex sub-signal selection. For example hierarchical filters allow the BL, E1 and E2 to be carried over distinct flows, and the user can accordingly tune into each sub-signal as required. As an example, the base and enhancement layers of the hybrid scalable MPEG-2 flow are each in turn made up of I, P and B pictures at each layer i.e. BL (I,P,B), E1 (I,P,B) and E2 (I,P,B). Using hybrid filters, the receiver can join the BL QoS group for the main profile and the E1 QoS group for the spatial enhancement and then select sub-signals within each profile as required (e.g. the I and P pictures of the BL).

7. Conclusion

At Lancaster University we are investigating heterogeneity issues present in applications, communications systems and networks. Resolving heterogeneous QoS demands in networked multimedia systems is a particularly acute problem that we are addressing within the framework of our QoS-A. As part of that work we have described a scheme for the dynamic management of multi-layer flows in heterogeneous multimedia and multicast networking environments. Dynamic QoS management manipulates and adapts hierarchically coded flows at the end-systems and in the network using a set of scaling objects. The approach is based on three basic concepts: the scalable profiles of the MPEG-2 standard that can provide discrete adaptation, dynamic rate shaping algorithms for compressed digital video that provide continuous adaptation, and the weighted fair share service for adaptive flows. At the present time DQM is being implemented at Lancaster University. The experimental infrastructure at Lancaster is based on 80486 machines running a multimedia enhanced Chorus micro-kernel [Coulson,95] and connected by programmable Olivetti Research Limited 4x4 ATM switches. This work is being carried out collaboratively with Columbia University in the US. At Columbia we are currently using CORBA [Lazar,94] to propagate selection filters in the network using ASX200 switches. In addition to the implementation we are conducting an extensive simulation study into the feasibility of the adaptive network service for large scale use, and investigating the feasibility of extending the adaptive service concept into our enhance Chorus micro-kernel itself [Coulson,95]. In other QoS related research at Lancaster we are developing a networked scalable multimedia storage system [Pegler,95] based on the dynamic allocation of compressed file components within a hierarchy of heterogeneous servers, and specialised filter servers [Yeadon,94] which implement transcoder QoS filters.

8. Acknowledgement

The authors would like to thank members of the Multimedia Projects Group at Lancaster and the ISO QoS Working Group for the many enlightening discussions on end-to-end QoS. In particular. The authors are very grateful to the Alexandros Eleftheriadis and Cristina Aurrecoechea for their thoughtful comments and suggestions on this work. The research on Dynamic Rate Shaping was the subject of Alexandros Eleftheriadis' Thesis at Columbia University.

The QoS-A project is funded as part of the UK SERC Specially Promoted Programme in Integrated Multiservice Communication Networks (GR/H77194) in co-operation with GDC (formally Netcomm Ltd). Andrew Campbell wishes to thank the EPSRC for their kind support while he was visiting the Center for Telecommunications Research, Columbia University.

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地理位置 位于墨西哥大学城以南的库库尔坎(Kukulcan),是玛雅文化前古典期晚期(公元前800年——公元前200年)中部高原文化的重要文化遗址之一,“库库尔坎”的原意是“舞蹈唱歌的地方”,或表示“带有羽毛的蛇神”。 建筑结构 “库库尔坎”金字塔是我们现在看到的早期墨西哥金字塔是一座用土筑成的 九层圆形祭坛,高29米 ,周边各宽55米多,周长250米左右,最高一层建有一座6米高的方形坛庙,库库尔坎金字塔高约30 米,四周环绕91 级台阶,加起来一共364级台阶,再加上塔顶的羽蛇神庙,共有365阶,象征了一年中的365 天。台阶的两侧宽达1米的边墙,北边墙下端,有一个带羽行的大蛇头石刻,蛇头高1.43米,长达1.80米,宽1.07米,蛇嘴里吐出一条大舌头,颇为独特。 这座古老的建筑,在建造之前,经过了精心的几何设计,它所表达出的精确度和玄妙而充满戏剧性的效果,令后人叹为观止:每年春分和秋分两天的日落时分,北面一组台阶的边墙会在阳光照射下形成弯弯曲曲的七段等腰三角形,连同底部雕刻的蛇头,宛若一条巨蛇从塔顶向大地游动,象征着羽蛇神在春分时苏醒,爬出庙宇。每一次,这个幻像持续整整3 小时22 分,分秒不差。这个神秘景观被称为“光影蛇形”。 每当“库库尔坎”金字塔出同蛇影奇观的时候,古代玛雅人就欢聚在一起,高歌起舞,庆祝这位羽毛蛇神的降临。 库库尔坎金字塔,是玛雅人对其掌握的建筑几何知识的绝妙展示,而金字塔旁边的天文台,更是把这种高超的几何和天文知识表现得淋漓尽致。 建造工艺 在没有金属工具、没有大牲畜和轮车的情况下,古代玛雅人却能够开采大量重达数十吨的石头,跋山涉水、一路艰辛地运到目的地,建成一个个雄伟的金字塔。金字塔最高的可达70米,其规模之巨大、施工难度之高,令人吃惊。 古代玛雅的金字塔和古埃及的金字塔在建筑形式上有着明显的不同。埃及的金字塔的塔顶是尖的,而玛雅金字塔却是平顶,塔体呈方形,底大顶小,层层叠叠,塔顶的台上还建有庙宇;在用途上也不一样:埃及金字塔是法老的陵墓,而玛雅的金字塔除个别外,一般是用来祭祀或观察天象的。除金字塔外,玛雅人还兴建了不少功能性强,技艺高的民用建筑,主要有:房屋(包括庙宇、府邸、民居等)、公共设施(球场、广场、集市等)、基础设施(桥梁、大道、码头、堤坝、护墙等)和水利工程(水渠、水库、水井、梯田)等。

库库尔坎金字塔位于墨西哥

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