Gerchberg–Saxton-type (GS-type) algorithms have been widely applied in photonics to reconstruct the object structures.
However, using random guesses as the initial inputs, the reconstruction quality of GS-type algorithms is unpredictable.
And, it always leads to a large number of iterations to reach convergence. In this paper, a singular value decomposition
(SVD) based method is proposed to generate an effective phase guess for GS-type algorithms using a low rank
approximation. Experimental results demonstrate that under the same reconstruction error, the proposed SVD based
guesses reduce the iteration times by more than 50% on average compared with that of random guesses. Furthermore,
they can outperform random guesses both in terms of steady state error and iteration times. Compared with the average
performance of random guesses, the proposed approach reduces the steady state error of recovered images by 70.7% on
average and reduces the iteration times by 56.1% on average.
In wireless communications, channel coding and error control are essential to protect the video data from wireless
interference. The power it consumed, which is determined by the protection method it used, will directly affect the
system performance especially on the decoding side. In this paper, a channel coding and error control system, called joint
channel protection (JCP) system here, is proposed as an improvement of the hybrid automatic repeat request (HARQ)
system to integrate the complexity controllability. The complexity models of the encoder and decoder are established
based on theoretical analysis and statistical data retrieval using the time complexity concept, and the relative variation in
the computational complexity is carefully studied to provide a proportional variation reference for complexity control.
Based on the models, strategies are designed to control the system complexity by adjusting the packet length, iterative
decoding times and retransmission ratio according to the decoding quality and complexity level.
The discrete wavelet transform (DWT) has been widely used in scalable video coding for its advantages in multi-resolution analysis and subband decomposition. In this paper, a spatially scalable video coding system based on H.264 coding method and in-band overcomplete discrete wavelet transform (ODWT) technique is proposed, which integrates the good compression performance of H.264 in low frequency domain with the efficient motion estimation of in- band ODWT in wavelet domain. Intra prediction, coefficients scan manner and inter prediction are improved to overcome the inefficiency of H.264 coding in high frequency subbands caused by different pixels distribution properties. Through series of subband analysis and statistical data retrieval for the three high frequency decompositions, intra prediction directions are optimized and three subsets of prediction mode are presented for the three high subbands
respectively. They save over 30% bits for intra mode with similar performance. Moreover, novel zigzag scan tables are proposed to improve the coding efficiency by utilizing the oriented frequency characteristics of each high band. To inter prediction, an adaptive motion estimation method is proposed in which the motion information of low band is adaptively and effectively utilized to achieve much more accurate motion vector and more efficient motion compensation in high bands coding. Experimental results show that, all of the proposed methods endue the spatially scalable video coding system with over 0.4 dB gain in PSNR and 10.4% in rate reduction.
In block-based video compression technology, blocking artifacts are obvious because of the luminance and chrominance discontinuities which are caused by block-based discrete cosine transform (DCT) and motion compensation. As a kind of solution, an in-loop filter has been successfully used in H.264 adapting to quantization parameter and video content. In this paper, blocking artifacts distribution properties are analyzed carefully to reflect the blocking effect more
accurately in the low bit rate applications. Two important parameters, named blocking severity and pixel variation, are defined to describe the boundary strength and the gradient of the samples across the edge respectively. Through series of statistical data retrieval and analysis for these parameters using multiple representative video sequences, a novel blocking artifacts distribution model is concluded. Based on this distribution model, an improved filter is proposed to H.264 with novel strength determination rule and different alpha model. Comparing with H.264 anchor results, the proposed de-blocking filter shows better performance especially in subjective aspect, which could be widely used in low bit rate applications.
In this paper, a novel VLC method based on 2nd-run-level coding and dynamic truncation is proposed to compress the DCT coefficients efficiently. In the proposed VLC, 2nd-run-level is first employed, following the traditional run-level coding, to further reduce the considerable redundancy existing in the original level and run sequences. In order to achieve a higher degree of context adaptability and coding efficiency, dynamic truncation is introduced and employed in the sequential coding of 2D symbols without large amounts of 2D-VLC tables required, though adding extra coding complexity to some extent. Adaptive EG/GR selection is also presented and recommended since it brings extra improvement in compression efficiency without increasing any computational complexity. Experimental results show that when compared with context-based 2D-VLC, the proposed VLC method gains 0.25 dB ~0.79 dB in PSNR and achieves 5.30% ~ 11.58% improvement in bit rate reduction.
With the popularity of wireless communication, much attention has been drawn to the reliable transmission for the fading and shadowing channels. In this paper, after a careful analysis of several existing schemes of hybrid ARQ with rate compatible punctured turbo (RCPT) codes, a novel hybrid ARQ scheme is proposed to fit for the unideal feedback channel in the worse wireless communication environment, which results in many advantages and improves performance of data transmission in practical wireless network.
In this paper, a novel interactive voice response (IVR) system is proposed, which is apparently different from the traditional. Using software operation and network control, the IVR system is presented which only depends on software in the server in which the system lies and the hardware in network terminals on user side, such as gateway (GW), personal gateway (PG), PC and so on. The system transmits the audio using real time protocol (RTP) protocol via internet to the network terminals and controls flow using finite state machine (FSM) stimulated by H.245 massages sent from user side and the system control factors. Being compared with other existing schemes, this IVR system results in several advantages, such as greatly saving the system cost, fully utilizing the existing network resources and enhancing the flexibility. The system is capable to be put in any service server anywhere in the Internet and even fits for the wireless applications based on packet switched communication. The IVR system has been put into reality and passed the system test.