- Research Article
- Open Access
Scalable and Media Aware Adaptive Video Streaming over Wireless Networks
© N. Tizon and B. Pesquet-Popescu. 2008
- Received: 29 September 2007
- Accepted: 6 May 2008
- Published: 19 May 2008
This paper proposes an advanced video streaming system based on scalable video coding in order to optimize resource utilization in wireless networks with retransmission mechanisms at radio protocol level. The key component of this system is a packet scheduling algorithm which operates on the different substreams of a main scalable video stream and which is implemented in a so-called media aware network element. The concerned type of transport channel is a dedicated channel subject to parameters (bitrate, loss rate) variations on the long run. Moreover, we propose a combined scalability approach in which common temporal and SNR scalability features can be used jointly with a partitioning of the image into regions of interest. Simulation results show that our approach provides substantial quality gain compared to classical packet transmission methods and they demonstrate how ROI coding combined with SNR scalability allows to improve again the visual quality.
- Scalable Video Code
- Packet Schedule
- Transmission Time Interval
- Syntax Element
- Radio Link Control
Streaming video applications are involved in an increasing number of communication services. The need of interoperability between networks is crucial and media adaptation at the entrance of bottleneck links (e.g., wireless networks) is a key issue. In the last releases of 3G networks , jointly with a high speed transport channel, the high speed downlink packet access (HSDPA) technology provides enhanced channel coding features. On the one hand, packet scheduling functionalities of the shared channel located close to the air interface allow to use radio resources more efficiently. On the other hand, error correction mechanisms like hybrid automatic repeat request (HARQ) or forward error correction (FEC) contribute to build an error resilient system. However, these enhancements are designed to be operational through a large collection of services without considering subsequent optimizations. In the best case, a QoS framework would be implemented with network differentiated operating modes to provide a class of services . To guarantee continuous video playout, streaming services are constrained by strictly delay bounds. Usually, guaranteed bitrates (GBR) are negotiated to maintain required bandwidth in case of congestion. Moreover, to guarantee on-time delivery, the retransmission of lost packets must be limited, leading to an over allocation of resources to face the worst cases. The main drawback of a QoS-oriented network is that it requires a guaranteed bitrate per user and thus it does not allow to take advantage of rate variability of encoded videos. In , a streaming system is proposed with QoS differentiation in order to optimize experienced quality at client side in the case of degraded channel quality. Assuming that the bandwidth allocated to the user is not large enough with respect to negotiated GBR, this study shows that prioritization of packets following the regions of interest (ROI) can achieve a substantial gain on perceived video quality.
In the scope of packetized media streaming over best-effort networks and more precisely channel adaptive video streaming,  proposes a review of recent advances. The closest approach from our works is the well-known rate-distortion optimized packet scheduling method. However, in this technical review, scalable-based solutions are considered as inefficient due to the fact that poor compression performances and wireless networks are not really studied with their most important specificities at radio link layer like radio frame retransmissions. In , Chou and Miao have addressed the problem of rate-distortion optimized packet scheduling conducted as an error-cost optimization problem. In their approach, encoded data partitioned into dependent data units, which can be a scalable stream, are represented as a directed acyclic graph. This representation is used with channel error rate measurements as input parameters of a Lagrangian minimization algorithm. This general framework can be adapted in terms of channel model and transmission protocol between the server and the client. For example in , the error process of a wireless fading channel is approximated by a first-order Markov process. Then, in order to choose the optimal scheduling policy, the server uses this model combined with video frame-based acknowledgment (ACK/NACK) from the client to compute the expected distortion reduction to be maximized. In , a similar approach is proposed considering a measure of congestion instead of the previous distortion. Besides, packet scheduling algorithms can switch between different versions of the streamed video, encoded with different qualities, instead of pruning the previous set of dependent data units. Then, These methods based on rate (congestion)-distortion optimized packet scheduling are in theory likely to provide an optimal solution to media aware scheduling problem. However, without simplification, the Lagrangian optimization is computationally intensive and the channel estimation (delay, capacity) may be more difficult when packets are segmented and retransmitted below application layer (e.g., ARQ at radio link control (RLC) layer). Moreover, in a wireless system, packet scheduling on the shared resource occurs at MAC or RLC layers independently of the application content.
In , media bitrate adaptation problem is set as a tradeoff between the current stream pruning and stream switching among a set of videos with different qualities. In order to provide more flexible schemes, the scalable extension of H.264/AVC, namely, scalable video coding (SVC),  allows to encode in the same bitstream a wide range of spatiotemporal and quality layers. In , a generic wireless multiuser video streaming system uses SVC coding in order to adapt the input stream at the radio link layer as a function of the available bandwidth. Thanks to a media-aware network element (MANE) that assigns priority labels to video packets, in the proposed approach, a drop priority-based (DPB) radio link buffer management strategy  is used to keep a finite queue before the bottleneck link. The main drawback of this method is that the efficiency of source bitrate adaptation depends on buffer dimensioning and with this approach, video packets are transmitted without considering their reception deadlines.
In this paper, our approach is to exploit the SVC coding in order to provide a subset of hierarchically organized substreams at the RLC layer entry point and we propose an algorithm to select scalable substreams to be transmitted to RCL layer depending on the channel transmission conditions. The general idea is to perform a fair scheduling between scalable substreams until the deadline of the oldest unsent data units with higher priorities is approaching. When this deadline is expected to be violated, fairness is no longer maintained and packets with lower priorities are delayed in a first time and later dropped if necessary. In order to do this, we propose an algorithm located in a so-called media aware network element (MANE) which performs a bitstream adaptation between RTP and RLC layers based on an estimation of transport channel conditions. This adaptation is made possible thanks to the splitting of the main scalable stream into different substreams. Each of these substreams conveys a specific combination of SNR and/or temporal layers which corresponds to a specific combination of high-level syntax elements. In addition, SVC coding is tuned, leading to a generalized scalability scheme including regions of interest. ROI coding combined with SNR and temporal scalability provides a wide range of possible bitstream partitions that can be judiciously selected in order to improve psychovisual perception.
The paper is organized as follows: in the next section we describe the scalable video coding context and the related standardized tools. In Section 3, we address the problem of ROI definition and propose an efficient way to transmit partitioning information requiring only a slight modification of the compressed bitstream syntax. Then, in Section 4, we present our developed algorithm to perform bitstream adaptation and packet scheduling at the entrance of RLC layer. Finally, simulation results are presented in Section 5 and we conclude in Section 6.
2.1. SVC Main Concepts
To serve different needs of users with different displays connected through different network links by using a single bitstream, a single coded version of the video should provide spatial, temporal, and quality scalability. As a distinctive feature, SVC allows a generation of an H.264/MPEG-4 AVC compliant, that is, backwards-compatible, base layer and one, or several, enhancement layer(s). Each enhancement layer can be turned into an AVC-compliant standalone (and not anymore scalable) bitstream, using built-in SVC tools. The base-layer bitstream corresponds to a minimum quality, frame rate, and resolution (e.g., QCIF video), and the enhancement-layer bitstreams represent the same video at gradually increased quality and/or increased resolution (e.g., CIF) and/or increased frame rate. A mechanism of prediction between the various enhancement layers allows the reuse of textures and motion-vector fields obtained in preceding layers. This layered approach is able to provide spatial scalability but also a coarse-grain SNR scalability. In a CGS bitstream, all layers have the same spatial resolution but lower layers coefficients are encoded with a coarser quantization steps. In order to achieve a finer granularity of quality, a so-called medium grain scalability (MGS), identical in principle to CGS, allows to partition the transform coefficients of a layer into up to 16 MGS layers. This increases the number of packets and the number of extraction points with different bitrates. Coding efficiency of SVC depends on the application requirements but the goal is to achieve a rate-distortion performance that is comparable to nonscalable H.264/MPEG-4 AVC. The design of the scalable H.264/MPEG4-AVC extension and promising application areas are pointed out in .
2.2. Bitstream Adaptation
2.3. Flexible Macroblock Ordering (FMO)
H.264/AVC provides a syntactical tool: FMO, which allows partitioning video frames into slice groups. Seven different modes, corresponding to seven different ordering methods, exist, allowing to group macroblocks inside slice groups. For each frame of a video sequence, it is possible to transmit a set of information called picture parameter set (PPS), in which the parameter specifies the FMO mode of the corresponding frame. According to this parameter, it is also possible to transmit additional information to define the mapping between macroblocks and slice groups. Each slice group corresponds to a network abstraction layer (NAL) unit that will be further used as RTP payload. This mapping will assign each macroblock to a slice group which gives a partitioning (up to eight partitions) of the image. There exist six mapping methods for an H.264 bitstream. In this study, we use the mode 6, called explicit MB, to slice group mapping, where each macroblock is associated to a slice group index in the range . The relation of macroblock to slice group map amounts to finding a relevant partitioning of an image. Evaluation of partitioning relevance strongly depends on the application and often leads to subjective metrics.
3.1. ROI Definition
In image processing, detection of ROIs is often conducted as a segmentation problem if no other assumptions are formulated about the application context and postprocessing operations that will be applied on the signal.
3.2. Mapping Information Coding
The H.264/AVC standard defines a macroblock coding mode applied when no additional motion and residual information need to be transmitted in the bitstream. This mode, called SKIP mode, occurs when the macroblock can be decoded using information from neighbor macroblocks (in the current frame and in the previous frame). In this case, no information concerning the macroblock will be carried by the bitstream. A syntax element, , specifies the number of consecutive skipped macroblocks before reaching a nonskipped macroblock.
In the sequel, we will restrict scalability abilities of SVC to the temporal layering with the well-known hierarchical B pictures structure, and to SNR scalability with MGS slices coding. In fact, we assume that spatial scalability-based adaption has already occurred when reaching the bottleneck link. Thanks to the additional bytes in SVC NAL unit headers, the network is able to select a subset of layers from the main scalable bitstream. Moreover, in the previous section, we described a coding method in order to provide a data differentiation at image content or ROI level. In this section, we propose a packetization method that combines SVC native scalability modes and the underlying scalability provided by ROI partitioning with FMO.
4.1. Packetization and Stream-Based Priority Assignment
4.2. Packet Scheduling for SVC Bitstream
In the remaining of this study, we consider that the MANE sees RLC layer as the bottleneck link and performs packet scheduling from IP layer to RLC layer. In the case of a 3G network, the MANE is most probably between the radio network controller (RNC) and the gateway GPRS support node (GGSN) and we neglect transmission delay variations between the server and the MANE. Then, each RTP packet whose payload is an NAL unit is received by the MANE at where is the sampling instant of the data and the constant delay between the MANE and the server. Next, to simplify this we put knowing that this time only impacts the initial playout delay. Moreover, inside each scalable stream, packets are received in their decoding order which can be different from the sampling order due to the hierarchical B pictures structure. Hence, the head-of-line (HOL) data unit of a stream queue is different from the minimum sampling instant of queued packets: .
Input RTP streams are processed successively. When scheduling RTP packet, the algorithm evaluates the transmission queues of the most important streams and, according to network state, the current packet will be delayed or sent to RLC layer. All streams are next transmitted over the same wireless transport channel and when an RTP packet reaches RLC layer, all necessary time slots are used to send the whole packet. Therefore, the general principle of the algorithm is to allow sending a packet only if packet queues with higher priorities are not congested and if expectable bandwidth is sufficient to transmit the packet before its deadline.
Moreover, packet dependency can occur between packets from the same stream, in the case of a combined scalability-based stream definition, or between packets from different streams. Therefore, in order to provide an efficient transmission of scalable layers, the algorithm delays packet delivering until all packets from lower layers which are necessary to decode the current packet are transmitted.
5.1. Simulation Tools
To evaluate the efficiency of the proposed approach, some experiments have been conducted using a network simulator provided by the 3GPP video ad hoc group .
In addition, the RTP packetization modality is single network abstraction layer (NAL) unit mode (one NAL unit/RTP payload), the division of original stream into many RTP substreams leads to an increase of the number of RTP headers. To limit the multiplications of header information, the interleaved RTP packetization mode allows multitime aggregation packets (NAL units with different time stamps) in the same RTP payload. In our case, we make the assumption that RoHC mechanisms provide RTP/UDP/IP header compression from 40 to 4 bytes in average, which is negligible compared to RTP packet sizes, and we still packetize one NAL unit per RTP payload.
5.2. Simulation Results
5.2.1. Adaptation Capabilities
Performance comparison between H.264 (one RTP stream) and SVC (2 RTP streams: base layer and SNR refinement).
Mother and daughter
5.2.2. Adaptation Capabilities and Bandwidth Allocation
In this section, the simulations are conducted in order to study the combined effects of channel errors and bandwidth decrease. Indeed, the implementation of a dedicated channel with a purely constant bitrate is not really efficient in terms of radio resource utilization between all users. Then, a more advanced resource allocation strategy would decrease the available bandwidth of the user when his conditions become too bad, in order to better serve other users with better experienced conditions. This allocation strategy, which aims at maximizing the overall network throughput or the sum of the data rates that are delivered to all users in the network, corresponds to an ideal functioning mode of the system but it is not really compatible with a QoS-based approach.
5.2.3. Scalability and ROI Combined Approach
In this section, we evaluate the contribution, in terms of psychovisual perception of the ROI-based differentiation combined with SVC intrinsic scalability features. In order to do this, the simulator is configurated like in the previous section with a bandwidth decrease at the 15th second. At the source coding, an ROI partitioning is performed as described in Section 3 and a quality refinement layer is used, leading to a subset of three RTP streams:
the quality base layer of the whole image (high priority),
the refinement layer of the ROI slice group (medium priority),
the refinement layer of the background (low priority).
In addition, our proposed algorithm is designed in order to allow more complex layers combinations with temporal scalability. In our simulations, the utilization of the temporal scalability did not provide a substantial additional perceived quality gain. In theory, it would be possible to perform more sophisticated differentiation between images regions. For example, we can imagine a configuration where the stream with the highest priority contains the following layers:
quality base layer of the ROI with the full temporal resolution,
SNR refinement layer of the ROI with a reduced temporal resolution,
quality base layer of the background with a reduced temporal resolution.
In fact, the bitrate of a quality base layer, and more particularly for the background, is often low. Hence, the bitrate saved by removing from the temporal resolution of the background is not high enough to compensate for the additional SNR refinement layer of the ROI. Therefore, the global bitrate of this RTP stream would be high and it would not be surely transmitted, leading to degraded performances.
This study proposes a complete framework for scalable and media aware adaptive video streaming over wireless networks. At the source coding, we developed an efficient coding method to detect ROIs and transmit ROI mapping information. Next, using the SVC high-level syntax, we proposed to combine ROI partitioning with common scalability features. In order to multiplex scalable layers, we adopted the MANE approach. In our system, the MANE is close to the wireless interface and it manages RTP packets transmission to the RLC layer following priority rules. In order to do this, a bitrate adaptation algorithm performs packet scheduling based on a channel state estimation. This algorithm considers the delay at RLC layer and packet deadlines in order to maximize the video quality avoiding network congestion. Our simulations show that the proposed method outperforms classical nonscalable streaming approaches and the adaptation capabilities can be used to optimize the resource utilization. Finally, the ROI approach combined with SNR scalability allows to improve again the visual quality. Future work will aim at generalizing this study in the case of a shared wireless transport channel.
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