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As part of video transmission, video quality assessment plays an important role in improving the user’s visual quality and reducing transmission bitrate. However, the existing assessment schemes for the tile-based transmission mechanism cannot accurately predict the video quality. In this paper, we propose a tile-based panoramic video quality assessment method to optimize the transmission strategy for tile-based streaming. Considering the distortion of panoramic video projection, the inverse of gnomonic projection and bilinear interpolation are utilized to convert the planar tiles to sphere. Video Multimethod Assessment Fusion (VMAF) is applied to obtain the tile quality score. Besides, the user’s head movement and eye movement are combined to generate a tile weight map, and generate the final score of the panoramic video through weighted averaging. In addition, considering the impact of the quality fluctuation of the tiles within field of view, the tile quality loss model is established from multiple dimensions including the encoding parameter, viewport position and video content. Experimental results show that there is a high correlation between the predicted results and actual subjective scores, under two video quality assessment datasets. Furthermore, we integrate the method into the panoramic video transmission system. Compared with the tile-based adaptive rate algorithm, the proposed method degrades the video quality by only 3%, while reducing the bandwidth consumption by 16.99%. [ABSTRACT FROM AUTHOR]

Recently, many researches on convolution neural network (CNN) based in-loop filters have been proposed to improve coding efficiency. However, most existing CNN based filters tend to train and deploy multiple networks for various quantization parameters (QP) and frame types (FT), which drastically increases resources in training these models and the memory burdens for video codec. In this paper, we propose a novel variable CNN (VCNN) based in-loop filter for VVC, which can effectively handle the compressed videos with different QPs and FTs via a single model. Specifically, an efficient and flexible attention module is developed to recalibrate features according to QPs or FTs. Then we embed the module into the residual block so that these informative features can be continuously utilized in the residual learning process. To minimize the information loss in the learning process of the entire network, we utilize a residual feature aggregation module (RFA) for more efficient feature extraction. Based on it, an efficient network architecture VCNN is designed that can not only effectively reduce compression artifacts, but also can be adaptive to various QPs and FTs. To address training data imbalance on various QPs and FTs and improve the robustness of the model, a focal mean square error loss function is employed to train the proposed network. Then we integrate the VCNN into VVC as an additional tool of in-loop filters after the deblocking filter. Extensive experimental results show that our VCNN approach obtains on average 3.63%, 4.36%, 4.23%, 3.56% under all intra, low-delay P, low-delay, and random access configurations, respectively, which is even better than QP-Separate models. [ABSTRACT FROM AUTHOR]

Information hiding based on bitrate switching is a novel steganographic scheme for voice over IP, which can resist traditional steganalysis methods. However, the state-of-the-art method has not achieved its best embedding performance. Thus, in this letter, we present two adaptive bitrate modulating-based steganographic methods introducing Hamming matrix encoding (HME) strategy. The first one can conceal \lfloor \log 2(n+1)\rfloor bits of secret data into n speech frames with at most one bitrate switching by applying the HME strategy to each frame group, while the second one, compared with the state-of-the-art method, can embed (1+\lfloor \log 2(n+1)\rfloor)/2 bits of secret data into n speech frames without increasing the number of bitrate switches by only employing the HME strategy for the frame groups with bitrate switching. The theoretical analysis and experimental evaluation demonstrate that the proposed methods are feasible, and outperform the existing one on both covert bandwidth and embedding efficiency while achieving the comparably good steganographic transparency. [ABSTRACT FROM PUBLISHER]

With a lot of enhanced coding tools introduced, High Efficiency Video Coding (HEVC) achieves significant improvement in coding efficiency at the cost of increased computational complexity. To efficiently reduce the encoding time of HEVC, a wavefront-based high parallel (WHP) solution integrating novel data-level and task-level methods is proposed in this paper. On data level, optimal single-instruction-multiple-data algorithms are designed for the enhanced coding tools, i.e., replacing the multiplication in motion compensation by add and shift operations with reduced instruction cycles, removing the transpose in transform via realignment of coefficients, and minimizing the memory access in sum of absolute difference/sum of squared differences calculation by fully reusing the registers. On task level, a novel inter-frame wavefront (IFW) method is developed by effectively decreasing the dependence of wavefront parallel processing (WPP). In addition, a coding tree block level parallelism analysis method is presented to prove the superior of IFW method compared with other HEVC representative parallel methods. Besides, a three-level thread management scheme is proposed to best exploit the parallelism of IFW method and achieve corresponding encoding speedup. Extensive experimental results show that, the overall WHP solution can bring up to $57.65\times $ , $65.55\times $ , and $88.17\times $ speedup for HEVC encoding of Wide Video Graphics Array, 720p and 1080p standard test sequences, while maintaining the same coding performance as with WPP. The proposed solution is also applied in several leading video companies in China, providing HEVC video service for more than 1.3 million users everyday. [ABSTRACT FROM PUBLISHER]

Although the emerging video coding standard High Efficiency Video Coding (HEVC) successfully doubles the compression efficiency of H.264/AVC, its growing computational complexity makes real-time decoding of high-definition HEVC videos a very challenging issue for the existing personal computers and mobile devices. In this paper, a systematical, efficient HEVC decoding solution on general processors is provided, consisting of structure-level, data-level, and task-level approaches. First, a redesigned overall structure of a HEVC decoder with data redundancy reduction mechanism is introduced, which cuts down basic data operation cost and achieves an average decoding speedup of 2.37 \times compared to the HM 10.0 decoder. On this basis, novel single-instruction multiple-data (SIMD) algorithms such as low-complexity motion compensation, transpose-free transform, symmetric deblocking filter, and parallel-index sample adaptive offset are developed, which further parallelize the data operations of each decoding task and bring another 2.67 \times decoding speedup. Finally, a frame-based task-level parallel framework is employed with a flexible entry scheme to efficiently support the simultaneous processing of multiple decoding tasks for different HEVC parallel strategies. The overall solution achieves decoding fps of 40-75 for 4k HEVC videos on the Intel i7-2600 3.4 GHz quad-core processor (4-thread decoding) and 35-55 for 720p videos on the ARM Cortex-A9 1.2 GHz duo-core processor (2-thread decoding). This proposal is the recommended cross-platform HEVC decoding solution of Intel, AMD, and Cisco, and has provided HEVC service to over 1500 million people in China via the Xunlei Kankan video client. [ABSTRACT FROM PUBLISHER]

Natural type selection was originally proposed by Zamir and Rose for universal single description coding. In this paper, we generalize this principle to universal multiple description coding (MDC). Two schemes based on random codebooks are proposed: one is of fixed distortion and the other is of fixed weight. Their operational sum-rate-distortion functions are derived, which coincide with the EGC (El Gamal-Cover) sum-rate bound if the parameters of the schemes are optimized. It is also shown that in both schemes the joint type of reconstruction codewords can be used to improve the rate-distortion (R-D) performance. Based on our theoretical results, a practical universal scheme is proposed by leveraging the MDC methods based on low-density generator matrix (LDGM) codes. The performance of this scheme is compared experimentally with the EGC bound, which shows its effectiveness. [ABSTRACT FROM AUTHOR]

In the first part of this paper, we derive a source model describing the relationship between the rate, distortion, and quantization steps of the dead-zone plus uniform threshold scalar quantizers with nearly uniform reconstruction quantizers for generalized Gaussian distribution. This source model consists of rate-quantization, distortion-quantization (D-Q), and distortion-rate (D-R) models. In this part, we first rigorously confirm the accuracy of the proposed source model by comparing the calculated results with the coding data of JM 16.0. Efficient parameter estimation strategies are then developed to better employ this source model in our two-pass rate control method for H.264 variable bit rate coding. Based on our D-Q and D-R models, the proposed method is of high stability, low complexity and is easy to implement. Extensive experiments demonstrate that the proposed method achieves: 1) average peak signal-to-noise ratio variance of only 0.0658 dB, compared to 1.8758 dB of JM 16.0's method, with an average rate control error of 1.95% and 2) significant improvement in smoothing the video quality compared with the latest two-pass rate control method. [ABSTRACT FROM AUTHOR]

This paper provides a systematic rate-distortion (R-D) analysis of the dead-zone plus uniform threshold scalar quantization (DZ+UTSQ) with nearly uniform reconstruction quantization (NURQ) for generalized Gaussian distribution (GGD), which consists of two aspects: R-D performance analysis and R-D modeling. In R-D performance analysis, we first derive the preliminary constraint of optimum entropy-constrained DZ+UTSQ/NURQ for GGD, under which the property of the GGD distortion-rate (D-R) function is elucidated. Then for the GGD source of actual transform coefficients, the refined constraint and precise conditions of optimum DZ+UTSQ/NURQ are rigorously deduced in the real coding bit rate range, and efficient DZ+UTSQ/NURQ design criteria are proposed to reasonably simplify the utilization of effective quantizers in practice. In R-D modeling, inspired by R-D performance analysis, the D-R function is first developed, followed by the novel rate-quantization (R-Q) and distortion-quantization (D-Q) models derived using analytical and heuristic methods. The D-R, R-Q, and D-Q models form the source model describing the relationship between the rate, distortion, and quantization steps. One application of the proposed source model is the effective two-pass VBR coding algorithm design on an encoder of H.264/AVC reference software, which achieves constant video quality and desirable rate control accuracy. [ABSTRACT FROM AUTHOR]

In this paper, we are to solve the drift problem of multiple description video coding on packet loss channels by using state-of-the-art distributed techniques. We first present an asymptotically optimal code design of multiple descriptions in the Wyner–Ziv (MDWZ) setting. Then we propose a distributed multiple description video coding (DMDVC) scheme, which performs MDWZ coding on each nonintra coded frame. Instead of the prediction loops used in traditional multiple description video coding, Slepian–Wolf based coding is used to exploit interframe correlations. A bitplane extraction scheme is proposed to improve the balance between two descriptions, so that side informations can be interchanged between the side decoders of DMDVC with negligible quality degradation, which is crucial to robust transmission over packet loss channels. Experiment results demonstrate the robustness of our scheme, especially at high packet loss rates. [ABSTRACT FROM AUTHOR]