Resynchronization Marker Research Articles

Application of Error-resilient Transmission of Sleep Apnea Patient Video with Sound over Mobile Network

Mobile video-audio transmission systems have delivered patient video with relevant snoring sound to quantify the severity of the sleep apnea patient over wireless networks, but few have optimized video-audio transmission in combination with transmission protocol over error-prone environments using wireless links. In this paper, the performance of the MPEG (Motion Picture Expert Group)-4 error resilient tools with UDP(User Datagram Protocol) protocol were evaluated over a wireless network to suggest the optimum combination of MPEG-4 error resilient tools and UDP packet size suitable for real-time transmission of video-audio transmission over error-prone mobile networks. Through experimentation, it was found that the packet size should correspond to IP(Internet Protocol) datagram size minus UDP and IP header for optimal video-audio quality. Also, for error resilient tool selection, the combination of resynchronization marker and data partitioning showed the best performance.

The Efficient and Robust Error Resilient Entropy Coding of Compressed Image for Wireless Communications

Many image and video compression algorithms work by splitting the image into blocks and producing variable-length code bits for each block data. If variable-length code data are transmitted consecutively over error-prone channel without any error protection technique, the receiving decoder cannot decode the stream properly. So the standard image and video compression algorithms insert some redundant information into the stream to provide some protection against channel errors. One of such redundancy is resynchronization marker, which enables the decoder to restart the decoding process from a known state in the event of transmission errors, but its frequent use should be restricted not to consume bandwidth too much. The Error Resilient Entropy Code (EREC) is well known method which can regain synchronization without any redundant information. It can work with the overall prefix codes, which many image compression methods use. This paper proposes an improvement to FEREC (Fast Error-Resilient Entropy Coding). It first calculates initial searching position according to bit lengths of consecutive blocks. Second, initial offset is decided using statistical distribution of long and short blocks, and initial offset is adjusted to insure all possible offset value can be examined. The proposed algorithm can speed up the construction of EREC slots, and can preserve compressed image quality in the event of transmission errors. The simulation result shows that the quality of transmitted image is enhanced about 0.3--3.5 dB compared with the existing FEREC when random channel error happens.

A real-time video transport system for the best-effort Internet

It is a challenging problem to design an efficient real-time video transport system for the present best-effort Internet because of the strict bandwidth, loss and time constraints and the lack of quality of service (QoS) guarantees from the present IP networks. This paper presents an end-to-end system architecture based on the MPEG-4 Delivery Multimedia Integration Framework (DMIF) for transporting real-time live video over the Internet. The key features of the system include: (1) MPEG-4 DMIF-based terminal architecture; (2) combination of an end-to-end feedback control mechanism and a rate-adaptive encoding algorithm for the best use of the Internet; (3) a robust and efficient packetization scheme for the MPEG-4 bit-stream by using the resynchronization marker approach specified in the MPEG-4 standard; and (4) efficient error control algorithms adopted at the end systems for visual quality enhancement. Simulation and test results using the actual Internet showed that our system is capable of utilizing the available network resource and achieve good perceptual quality at the application level.

A unified architecture for real-time video-coding systems

This paper presents a unified architecture for a live video over the Internet with emphasis on solving some challenging problems such as network bandwidth adaptation for rate and congestion, loss packet recovery, joint source and channel coding, and packetization. In our architecture, a time-varying bit rate for the source coding and time-varying ratios for the channel coding are simultaneously computed by a new congestion-control protocol. An adaptive rate-control scheme is then proposed to calculate quantization parameters and to determine the number of skipping frames corresponding to the bit rate. An adaptive unequal error-control scheme is also provided to protect the bitstream. Furthermore, a simple and MPEG-4 standard compatible algorithm is designed to packetize generated bitstream at the SyncLayer by using the existing resynchronization marker approach. With the proposed architecture, the coding efficiency and the robustness of the whole system are improved greatly.

Robust H.263+ video transmission using partial backward decodable bit stream (PBDBS)

We propose a novel robust H.263+ video transmission scheme using a partial backward decodable bit stream (PBDBS), which reverses the bit stream of some coded macroblocks (MBs) so that these coded MBs can be decoded in a backward direction. Because of using variable-length coding (VLC), the baseline H.263+ coder will lose synchronization until the next resynchronization marker when errors occur in the coded bit stream. It is wasteful to discard all of the remaining coded MBs in a group of blocks (GOB). Our method is to alleviate this effect by reversing the bitstream that belongs to the coded second half row of MBs in a GOB. When a decoder loses resynchronization, it will search the next resynchronization marker and decode the bitstream backward. It can retrieve half row of MBs that are lost otherwise. This method combined with error concealment can greatly improve the transmitted image quality. Up to 2-dB luminance peak signal-to-noise ratio improvement can be found in our method compared with the baseline H.263+ coder. We also compare our PBDBS method with H.263++ Annex V using data partitioning and reversible VLC. Under lower encoder and decoder complexity, the PBDBS method obtains more pleasant results than Annex V, and it can be utilized in noisy channels such as wireless networks.

Optimal resynchronization marker positioning method using a novel information measure

This paper presents a new resynchronization marker (RM) positioning technique to effectively block the error propagation caused by channel errors. The optimal technique can be defined as the one minimizing the information loss under an error condition. For realization of the optimal technique, the amount of the information lost by errors needs to be measured. Thus, we introduce a novel measure that is successful in reflecting the actual importance of the lost data, which is measured by using error concealment techniques. That is, the amount of information lost by an error at a certain location is numerically specified by the difference between the image recovered by the error concealment and the image normally decoded without any error. Since such a measurement method is appropriate for reflecting the actual damage by errors, it is very useful for determining the RM positions. With the new measure, the optimal RM positioning technique is proposed and verified by experiments.