This paper defines a Quality of Experience (QoE) parameter for multi-view video streaming service over Wireless Broadband Internet, and proposes cross-layer optimization (CLO) algorithm to maximize the QoE. The proposed CLO algorithm interacts with all layers from PHY layer to video layer. Under the time-varying wireless channel condition, it guarantees service continuity to the user by selecting frames to be transmitted. The CLO technique takes inter-view and the temporal significance of the multi-view video frames into consideration when deciding the Transmission Frame Set (TFS) and its Levels of FEC (LOF). Through simulation of the proposed technique, it is shown that the optimal quality of the multi-view video can be achieved even during hand-over, using the minimum amount of resources. Visual quality of multi-view video streams is improved by about 15.330.2%, especially up to 39.165% during the hand-over. Moreover, continuity of the most important view has been increased by 20.139.1% in comparison with conventional method.

Applying Distributed Video Coding (DVC) to mobile devices that have limited computation and power resources can be a very challenging problem due to its high-complexity decoding. To address this, this paper proposes a DVC bitstream organizer. The proposed DVC bitstream organizer reduces the complexity associated with repetitive channel decoding and SI generation in a flexible manner. It allows users to choose a means of minimizing the computational complexity of the DVC decoder according to their preferences and the device's resource limitations. An experiment shows that the proposed method increases decoding speeds by up to 25 times.

We propose a distributed node selection (DNS) scheme that guarantees quality of service (QoS) of the scalable video broadcasting system over wireless channels. The proposed DNS scheme chooses the destination node based on the SVC layer information, and it selects the best relay from a set of competing candidate nodes by considering two factors: 1) wireless channel conditions between destination and relay candidates and 2) scalable video's layer information. In simulations, the performance of the proposed scheme in terms of quality gains, complexity (overhead) and applicability was examined.

For the panoramic video streaming service, this letter proposes a visual perception-based view navigation trick mode (VP-VNTM) that reduces bandwidth requirements by adjusting the quality of transmitting views in accordance with the view navigation velocity without decreasing the user's visual sensitivity. Experiments show that the proposed VP-VNTM reduces bandwidth requirements by more than 44%.

This letter proposes a method to retrieve the original image X out of multiple sets of SI (Side Information) in distributed video coding (DVC). Using Turbo decoding methods, the most reliable segments from the decoded Yi's were selected for the composition of Y∞, whose conditional entropy H(X| Y∞) became much lower than any individual conditional entropy H(X| Yi). This proposal has improved the peak signal-to-noise ratio (PSNR) by 1.1 to 1.8 dB, compared to the conventional single SI-based approach.

We propose a new channel adaptive distributed video coding algorithm, which is adaptive to time-varying available bitrate and packet loss ratio. The proposed method controls the quantization parameter according to channel condition of especially error-prone mobile channel. Simulation shows that the proposed algorithm outperforms the conventional rate-control-only algorithm.

This letter proposes a new delivery format in order to realize unified transmissions of stereoscopic video contents over a dynamic adaptive streaming scheme. With the proposed delivery format, various forms of stereoscopic video contents regardless of their encoding and composition types can be delivered over the current dynamic adaptive streaming scheme. In addition, the proposed delivery format supports dynamic and efficient switching between 2D and 3D sequences in an interoperable manner for both 2D and 3D digital devices, regardless of their capabilities. This letter describes the designed delivery format and shows dynamic interoperable applications for 2D and 3D mixed contents with the implemented system in order to verify its features and efficiency.

In this paper, we propose a novel layered scalable- multiple description coding (LS-MDC) which offers the benefits of both scalable video coding and multiple description coding for robust video transmission over packet lossy networks. In the proposed LS-MDC method, multiple descriptions including base layer, enhancement layers, and their corresponding FEC parity data are allocated into two network paths of a path diversity system. Unlike the conventional approaches, the source base/enhancement data and their own parities in the proposed method are not transmitted together but are transferred over different paths. Therefore, the effect of burst packet losses can be effectively reduced for the descriptions. Furthermore, in order to minimize the overall distortion for the LS-MDC system and exploit the benefits of path diversity, we also propose an optimal rate allocation scheme that can adaptively control the transmission rate as well as the channel coding rate for media senders. Experiments show that the proposed method provides much better peak signal-to-noise ratio (PSNR) than conventional MDC techniques.

In the past, enhancement techniques for the end-to-end quality of a networked application were studied by looking at each individual layer. Examples of such techniques include the error resilience/concealment methods in the application layer, the FEC/ARQ in the transport layer, and the Quality of Service (QoS) techniques in the network layers. However, an integrated approach that would look across all related layers had yet to be investigated. This paper proposes an approach that combines priority-aware video packetization, adaptively used layered FEC, and QoS controlled networks such as IntServ and DiffServ in order to provide an efficient end-to-end quality in layered streaming video. The combination is more efficient in terms of a simple network price mechanism, that is, in getting the best end-to-end quality under a given total cost constraint. Our proposed approach in DiffServ with video packet (VP) data-splitting and layered FEC guarantees minimal service quality in a scalable and cost effective manner without introducing resource reservation. For video, we also propose performance metrics such as corrupted/frozen frame ratio (CFR, FFR). These provide objective metrics like PSNR as well as a measurement for subjective perceptions. Our approach, which combines related layers such as video coding, transport, and network, has yielded results that have proven to be more cost-effective and practical than the supporting network QoS.

Most digital signal processing (DSP) algorithms for multimedia and communication applications require multiplication and addition operations. Especially matrix-matrix or matrix-vector the multiplications frequently used in DSP implementations needs inner product arithmetic which takes the most processing time. Also multiplications for the DSP algorithms for software defined radio (SDR) applications require different input bitwidths. Therefore, the multiplications for inner product need to be sufficiently flexible in terms of bitwidths to utilize hardware resources efficiently. This paper proposes a novel reconfigurable inner product architecture based on a pipelined adder array, which offers increased flexibility in bitwidths of input arrays. The proposed architecture consists of sixteen 44 multipliers and a pipelined adder array and can compute the inner product of input arrays with any combination of multiples of 4 bitwidths such as 44, 48, 412, ... 1616. Experimental results show that the proposed architecture has latency of maximum 9 clock cycles and throughput of 1 clock cycle for inner product of various bitwidths of input arrays. When TSMC 0.18 µm libraries are used, the chip area and critical path of the proposed architecture are 186,411 gates and 2.79 ns, respectively. The proposed architecture can be applied to a reconfigurable arithmetic engine for real-time SDR system designs.