Frame Bit Research Articles

The Use of Computer Vision Technology in the Inheritance of Intangible Cultural Heritage: The Case of Regional Cultural Characteristics

Abstract The inheritance of intangible cultural heritage generally faces the situation of having no successors and difficulties in dissemination. The use of digital technology can overcome time and space limitations, expanding the dissemination effect of intangible heritage. A panoramic exhibition model of intangible cultural heritage is constructed using computer vision technology in this paper. In order to ensure that every pixel point in the image has a unique position, the perspective projection algorithm and the chi-square coordinate system are designed according to phase-forming geometry. Secondly, the optical information of each point is reproduced in the form of the image by using spherical projection, and the image alignment and image fusion smooth transition is carried out by improving the minimum deviation matching method of feature line segments based on chromatic aberration chunking. To convert the spherical panoramic image into a view that the user can explore, the inverse projection calculation is carried out. The performance of the model is tested after its construction is completed, and it is used to reproduce the panoramic view of the non-heritage contents of Xijiang Thousand Households Miaozhai in Guizhou. It is found that the matching error of this paper’s algorithm in the 20-frame scene test is close to or slightly higher than that of the SVD algorithm only in the 11th to 16th frame bits, and the rest of the frame errors are lower than that of the SVD algorithm by more than 0.04m. This paper’s algorithm is superior to OpenCV and Xiong methods by at least 0.1 points or more in all five scenes when testing the OG-IQA parameter. It can be concluded that the online exhibition content of non-heritage in Xijiang Miaozhai, Guizhou, produced using the model of this paper, has achieved a better communication effect and meets expectations. This paper provides innovative ideas and effective examples to demonstrate the use of computer vision technology in expanding the dissemination effect of intangible cultural heritage and promoting inheritance.

Strongly Connected Ramanujan Graphs for Highly Symmetric LDPC Codes

A number of studies focus on Low-Density Parity-Check (LDPC) codes to ensure reliable data communications. This study proposes an algebraic algorithm to generate strongly connected Ramanujan graphs able to provide highly symmetric LDPC codes with minimized error floor. Several Ramanujan graphs are created using GAP system software to generate a rank-efficient parity-check matrix with fixed-rate LDPC codes. We find that Ramanujan LDPC codes achieve frame error rate and bit error rate on the order of \({10}^{-5}\) and \({10}^{-6}\), respectively. Furthermore, the codes outperform QC LDPC codes and those Ramanujan LDPC codes in literature.

Inferring End-to-End Latency in Live Videos

This paper provides an end-to-end latency model of real-time generated live videos composed of sub-models of capturing, encoding, network, decoding, rendering, and display refreshing. We build the model by deeply analyzing each sub part’s latency with parameters like video’s Spatial (frame scale <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$s$ </tex-math></inline-formula> ), Temporal (frame rate <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$t$ </tex-math></inline-formula> ), Amplitude (quantization step-size <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$q$ </tex-math></inline-formula> ) Resolutions (STAR), Compressed Frame Bit Numbers ( <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$n_{F}$ </tex-math></inline-formula> ), and other necessary parameters. The latency of 640 combinations of STAR for eight representative video sequences is measured to fit accurate models. We primarily focus on processing procedures, whose latency is becoming the principal affection, as network latency decreases due to the its development. We then introduce a baseline video sequence set to reduce the complexity when fitting the encoding and decoding sub-models by predicting the model parameters on new hardware from parameters of a known one, and improve models’ robustness crossing different scenarios. These sub-models correlate well with the realistic latency data, with the average Pearson Correlation Coefficient (PCC) of 0.99 and the average Spearman’s Rank Correlation Coefficient (SRCC) of 0.98, according to an independent validation test. Furthermore, we validate our end-to-end latency model in a simulated end-to-end live video transmission from generating to displaying, introducing the latency as a strict condition in adaptive bit-rate selection. The simulation shows that our model could significantly increase time during which the latency can satisfy the given tolerance at a low cost.

Measurement and Evaluation of Electric Signal Transmission Through Human Body by Channel Modeling, System Design, and Implementation

Human body communications (HBCs), employing the human body as a signal transmission medium, can provide efficient and intuitive methods to form a network in the body. This article presents a comprehensive study for a highly reliable HBC system, including body channel modeling, transceiver design, and performance evaluation through implementation, in consideration of practical sensor network environments for wearable and implantable devices applicable to healthcare and biosignal acquisition. Body channel characteristics based on capacitive couplings, such as root mean square delay spread and mean path gain (MPG), were explored by measuring the body impulse responses under 12 experimental conditions, determined by the variation in body postures and device locations between the wrist and positions assumed under the scalp through a customized experimental setup with micropig-derived biomembranes (MBMs) used for wrapping the devices to emulate an implantable environment. The proposed transceiver design for digital transmission, including a preamble structure and signal modulation method supporting a maximum data rate of 1 Mb/s, was verified through the performance evaluations conducted for examining the frame detection probability and bit error rate (BER) in the body channel model at multiple operating frequencies of 32, 42, and 64 MHz. The proposed system reliability was demonstrated by the achievement of a BER of below <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">${4.7\,\times \, }{10}^{-8}$ </tex-math></inline-formula> through battery-powered implemented devices with dimensions of <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">${30\times 30 }\,\,{\text {mm}}^{2}$ </tex-math></inline-formula> , comprising a digital signal processing module (DSPM) for signal generation and detection processes, and an analog front-end module (AFEM) for recovering the received signal from signal deterioration by severe path loss and time-dispersive effect in the body channel.

Smart Privacy Protection for Big Video Data Storage Based on Hierarchical Edge Computing.

Recently, the rapid development of the Internet of Things (IoT) has led to an increasing exponential growth of non-scalar data (e.g., images, videos). Local services are far from satisfying storage requirements, and the cloud computing fails to effectively support heterogeneous distributed IoT environments, such as wireless sensor network. To effectively provide smart privacy protection for video data storage, we take full advantage of three patterns (multi-access edge computing, cloudlets and fog computing) of edge computing to design the hierarchical edge computing architecture, and propose a low-complexity and high-secure scheme based on it. The video is divided into three parts and stored in completely different facilities. Specifically, the most significant bits of key frames are directly stored in local sensor devices while the least significant bits of key frames are encrypted and sent to the semi-trusted cloudlets. The non-key frame is compressed with the two-layer parallel compressive sensing and encrypted by the 2D logistic-skew tent map and then transmitted to the cloud. Simulation experiments and theoretical analysis demonstrate that our proposed scheme can not only provide smart privacy protection for big video data storage based on the hierarchical edge computing, but also avoid increasing additional computation burden and storage pressure.

Inter-Frame Polar Coding With Dynamic Frozen Bits

A new inter-frame correlated polar coding scheme is proposed, where two consecutive frames are correlated-encoded and assist each other during decoding. The correlation is achieved by dynamic configuration of the frozen bits. The frozen bits of the second frame partially depend on the unfrozen bits of the first frame in encoding and the number of bits that are viewed as frozen by decoder is alterable in different decoding modes. Using this new encoding/decoding scheme, a failed decoded frame can be decoded again with extra information which corrects the errors in its highly unreliable unfrozen bits. Thus, the probability of successful decoding is improved. Simulation results show that the performance of the proposed polar codes outperforms that of the classical counterpart significantly with negligible memory and complexity increment.

Audio watermarking based on synergy between Lucas regular sequence and Fast Fourier Transform

The growth of digital technology over the recent years has led to increased sending and saving of electronic media. Taking advantage of digital works without observing copyright is always considered as an important threat for publishers. One of the techniques proposed for overcoming this threat is watermarking. Audio watermarking is one of the most challenging fields of watermarking, because detection of watermark in audio is much easier than in image. In this paper, a new approach is proposed for audio watermarking based on the values of energy levels in FFT domain. Despite preservation of imperceptibility and robustness, this method has also a very high capacity. The main idea of the proposed method is employing a regular embedding procedure on the basis of closing the energy of the sub-frames to Lucas math sequence, which is capable of embedding two bits in each audio frame. In addition, appropriate parameters values of the proposed approach such as frame-size and frequency-band are calculated by an intelligent recursive tuning process. Extraction process is done in a blind manner and without any need to the host original signal. In order to evaluate the proposed method, five types of audio files including Electronic, Classic, Jazz, Blues, and Rock audio files have been used in wave format. The results show that despite having a high watermarking capacity, the proposed method has a high imperceptibility and an acceptable robustness against the common attacks such as Noise Addition, 64 kbps MPEG Compression (MP3), FFT-Invert, Re-sampling etc., compared with the previous methods.

Buffer-Driven Rate Control and Packet Distribution for Real-Time Videos in Heterogeneous Wireless Networks

Video transmission bandwidth can be substantially increased by concurrent transmission through multiple radio interfaces introduced by heterogeneous wireless networks. However, delivering high-quality real-time videos with stringent end-to-end delay constraints is still challenging due to the time-varying multipath channel status. Thus, to integrate available multipath bandwidth and guarantee low end-to-end delay, this paper investigates a real-time concurrent multipath video transmission scheme in heterogeneous access networks. First, we present a multipath buffer analysis model (MBAM) to derive the lower and upper bounds of the target video frame bits. Second, given the low-latency and received video quality smoothness, we propose a buffer-driven rate control and packet distribution (BRCPD) algorithm aided by frame-level rate control and multipath transmission buffer awareness based on MBAM. Finally, we implement the proposed application-layer solution BRCPD algorithm in a real-world testbed. The experiment results verify that different from existing multipath video transmission schemes, the proposed algorithm is able to leverage the MBAM-based frame-level rate control to guarantee the target low end-to-end delay. In particular, the BRCPD algorithm is able to guarantee an overdue loss rate of less than 5% under strictly limited delay constraints (250 ms) in the real-world testbed.

Optimize x265 Rate Control: An Exploration of Lookahead in Frame Bit Allocation and Slice Type Decision.

To improve the Rate-Distortion (R-D) quality, x265 rate-control made a variety vital decisions, such as scene cut detection, slice type decision, and coding-unit quantization parameter (QP) offsets, leveraging on lookahead to evaluate the information propagation through the current and the near future consecutive frames. However, as the frame base QP that dominates the bit amount allocated to one frame was only determined by the long-term complexity history in the original algorithm, the frame bit allocation became insensitive to the recent scene changes with the growth of coding time. In addition, the specified threshold in slice type decision, which was compared with the estimated frame coding costs to detect the B-type slice, did not consider the impacts of quantization. The aforementioned irrational elements degraded the rate accuracy and the R-D performance of x265. In this paper, the frame base QP is determined with not only the coding complexity history, but also the complexity changes and the data dependencies between the current and the near future pictures by exploring lookahead. Moreover, the quantization scale is introduced to the threshold specification in slice type decision, which identifies more pictures as B-type properly when increasing QP. The proposed algorithms were conducted in x265 version 2.4. Experiments revealed that, under the default preset (-preset medium), 0.617dB and up to 1.705dB BDPSNR quality gains were achieved, while saving the encoding time by 1.13% and improving the rate accuracy by 4.2% on average.

Efficient decoding algorithm for 3D video over wireless channels

In wireless communication networks, additive channel noise and time varying multipath fading are considered amongst the main challenges and these effects will bring significant quality degradation to the transmitted 2D/3D video streams. This paper presents an efficient decoding algorithm for Distributed Video Coding (DVC) for enhanced performance of colour and depth based 3D video over error prone wireless channels. The maximum a-posteriori (MAP) algorithm for turbo decoding is modified considering the effects of the channel errors on both Wyner-Ziv and key frame bit streams of colour and depth videos. The proposed codec is simulated using a W-CDMA wireless channel model and the results are analyzed to determine the effect of the modifications. The performance of the state-of-the-art H.264/AVC video codec is also presented for comparison under similar conditions. The results show that the proposed modifications provide a significant improvement in the DVC codec performance under unfavourable channel conditions for colour plus depth based 3D video.

Code Division of the Logical Channels for Transmission of Transactions in the Intelligent Mechatronic Systems

As a rule, modern engineering and manufacturing projects are saturated with electronic equipment for automated monitoring and control, as well as cable communication lines and radio frequency channels for transmission of transactions. During functioning of the intelligent mechatronic systems the number of the communication lines of the frequency ranges is restricted due to application of the CDMA principles. A drawback of this method of channelizing is connected with a high level of the interfering signals in case of an increased quantity of the simultaneously functioning subscribers, whereof signals at the receiving side are highlighted with the hardware and software correlators. Taking this into account, a numeric method was proposed for the accelerated data streams of the bits extraction from the frame bits during the code division multiple access to the transmission of the transactions on the basis of the use of the Walsh function and the intelligent mechatronic systems. The scheme of the code division of the bit streams was analyzed due to introduction of the information source bit streams to the mechatronic system of the intelligent multiplexers at the transmitting channel side and the intelligent bit streams separators at the receiving side. A peculiarity of this code division scheme for the logical channels is that it is fully implemented due to the use of the software as a part of the intelligent mechatronic system. Construction of the software models of the intelligent multiplexer and the intelligent separator proved their high efficiency and stability of their functioning. When up to 256 logical channels are formed in one physical channel, a reliable separation of the information bits is ensured, as well as their diversion by the sources with the allowed delays in the transactions' fulfillment. An essential extra time advantage occurs, because all the separator transactions in the real processor of the bit frames processing take place with the help of short commands of the integer arithmetic.

Evaluating Intelligibility and Battery Drain of Mobile Sign Language Video Transmitted at Low Frame Rates and Bit Rates

Mobile sign language video conversations can become unintelligible if high video transmission rates cause network congestion and delayed video. In an effort to understand the perceived lower limits of intelligible sign language video intended for mobile communication, we evaluated sign language video transmitted at four low frame rates (1, 5, 10, and 15 frames per second [fps]) and four low fixed bit rates (15, 30, 60, and 120 kilobits per second [kbps]) at a constant spatial resolution of 320 × 240 pixels. We discovered an “intelligibility ceiling effect,” in which increasing the frame rate above 10fps did not improve perceived intelligibility, and increasing the bit rate above 60kbps produced diminishing returns. Given the study parameters, our findings suggest that relaxing the recommended frame rate and bit rate to 10fps at 60kbps will provide intelligible video conversations while reducing total bandwidth consumption to 25% of the ITU-T standard (at least 25fps and 100kbps). As part of this work, we developed the Human Signal Intelligibility Model , a new conceptual model useful for informing evaluations of video intelligibility and our methodology for creating linguistically accessible web surveys for deaf people. We also conducted a battery-savings experiment quantifying battery drain when sign language video is transmitted at the lower frame rates and bit rates. Results confirmed that increasing the transmission rates monotonically decreased the battery life.

An Ultra-Low-Power Energy-Efficient Dual-Mode Wake-Up Receiver

An ultra-low-power wake-up receiver is proposed to reduce the energy consumption through adoption of two wake-up modes of operation with different power dissipations and detection bandwidths. During the Monitoring (MO) mode, the start frame bits are received at a data rate of 10 kbps, while during the Identifier (ID) mode the remaining wake-up pattern is transmitted at 200 kbps. By lowering the gain of the front-end amplifier in the MO mode the power dissipation is reduced, which in turn causes an increase in the overall noise figure of the receiver. However, adequate sensitivity is maintained by intentionally lowering the data rate as well as the detection bandwidth of the receiver. The proposed wake-up receiver is designed and fabricated in IBM 130 nm technology with a core size of about 0.2 ${\rm mm}^{2}$ for the target frequency range of 902–928 MHz. The measured results show that the proposed dual-mode receiver achieves a sensitivity of $-$ 78.5 dBm and $-$ 75 dBm while dissipating an average power of 16.4 $\mu{\rm W}$ and 22.9 $\mu{\rm W}$ during MO and ID modes, respectively.

A Mechanism to Prevent Stuff Bits in CAN for Achieving Jitterless Communication

The bit stuffing mechanism adopted in controller area networks leads to unwanted jitter on frame reception times, which worsens timing accuracy, even if countermeasures are adopted to avoid contentions on the bus. Several solutions have been proposed so far for dealing with stuff bits in the payload of messages, but they are not effective for the cyclic redundancy check. In this paper, a mechanism is presented that prevents the occurrence of stuff bits in the whole frame completely. It makes the duration of frame transmissions fixed and, hence, it achieves very accurate reception times. An optimized codec has been implemented to demonstrate that this approach is feasible and can be profitably adopted in low-cost networked embedded systems with demanding timing constraints.

Cross-Layer prioritized H.264 video packetization and error protection over noisy channels

Video transmission over wireless channels is affected by channel-induced packet losses. Distortion due to channel errors can be alleviated by applying forward error correction. Aggregating H.264/AVC slices to form video packets with sizes adapted to their importance can also improve transmission reliability. Larger packets are more likely to be in error but smaller packets require more overhead. We present a cross-layer dynamic programming (DP) approach to minimize the expected received video distortion by jointly addressing the priority-adaptive packet formation at the application layer and rate compatible punctured convolutional (RCPC) code rate allocation at the physical layer for prioritized slices of each group of pictures (GOP). Some low priority slices are also discarded to improve protection to more important slices and meet the channel bit-rate limitations. We propose two schemes. Our first scheme carries out joint optimization for all slices of a GOP at a time. The second scheme extends our cross-layer DP-based approach to slices of each frame by predicting the expected channel bit budget per frame for live streaming. The prediction uses a generalized linear model developed over the cumulative mean squared error per frame, channel SNR, and normalized compressed frame bit budget. The parameters are determined over a video dataset that spans high, medium and low motion complexity. The predicted frame bit budget is used to derive the packet sizes and corresponding RCPC code rates for live transmission using our DP-based approach. Simulation results show that both proposed schemes significantly improve the received video quality over contemporary error protection schemes.

Reconfigurable wrapper architecture for UART CORE testing using FSM approach

The UART is a device used for communication of data either in asynchronous or synchronous mode. Nowadays, more number of designers introduces UART as a system on chip for various purposes. These UART SoC cores are to be checked on the basis of its compatibility and performance and a complete study is advisable. For this purpose, a reusable and reconfigurable wrapper architecture is designed, which is based on a finite machine called the wrapper finite state machine WFSM in Verilog and its baud rate as well as frame bit rate is computed. This WFSM monitors and verifies bit rate, packet transmission with single stop bit and two stop bits, with and without parity bits, distance coverage, frame transmission time, junk handling, utilisation of CUP, utilisation of power by UART with wrapper in FPGA are studied at different baud rates. It is designed and implemented in Quarters II in CyclonII DEI FPGA with the help of Zero-Plus logic analyzer.

Motion-compensated coding and frame rate up-conversion: models and analysis.

Block-based motion estimation (ME) and motion compensation (MC) techniques are widely used in modern video processing algorithms and compression systems. The great variety of video applications and devices results in diverse compression specifications, such as frame rates and bit rates. In this paper, we study the effect of frame rate and compression bit rate on block-based ME and MC as commonly utilized in inter-frame coding and frame rate up-conversion (FRUC). This joint examination yields a theoretical foundation for comparing MC procedures in coding and FRUC. First, the video signal is locally modeled as a noisy translational motion of an image. Then, we theoretically model the motion-compensated prediction of available and absent frames as in coding and FRUC applications, respectively. The theoretic MC-prediction error is studied further and its autocorrelation function is calculated, yielding useful separable-simplifications for the coding application. We argue that a linear relation exists between the variance of the MC-prediction error and temporal distance. While the relevant distance in MC coding is between the predicted and reference frames, MC-FRUC is affected by the distance between the frames available for interpolation. We compare our estimates with experimental results and show that the theory explains qualitatively the empirical behavior. Then, we use the models proposed to analyze a system for improving of video coding at low bit rates, using a spatio-temporal scaling. Although this concept is practically employed in various forms, so far it lacked a theoretical justification. We here harness the proposed MC models and present a comprehensive analysis of the system, to qualitatively predict the experimental results.

Closed-Form Error Analysis of AF-CARQ with CSI-Assisted Relay over Nakagami- $$m$$ m Fading Channels

This paper presents a new Amplify-and-Forward Cooperative Automatic Repeat reQuest protocol with channel state information-assisted relay which is more suitable for the only-read access networks. The channels of any pair of terminations are quasi static flat Nakagami- $$m$$ m fading channels, which are mutually independent and non-identically distributed. Assuming that the coherent equal gain combining is adopted to combine the retransmitted signals from the same link at the destination and selective combining is adopted to the signals from different links. Based on the approximation of product of Nakagami- $$m$$ m variables, we obtain the end-to-end signal-to-noise rate of any link. The closed-form expression of the average bit error rate for several modulation schemes is obtained by analyzing cumulative distribution function (CDF) and Gaussian Q-function. Then we analyze the amount of fading of the fading channels by the $$n$$ n order moment which is obtained by CDF. Numerical simulation results show that the relay node can resist the fading of system effectively comparing with the system without relay node. And with the increasing of the number of transmission, the performance advantage of relay link is more and more obviously. It is better to let the maximum transmission time $$F=6$$ F = 6 , which is very useful for improving the transmission efficiency of the truncated ARQ system. The maximum reduction of amount of fading can be reached when $$t=3$$ t = 3 , if the total number of transmission is $$f=6$$ f = 6 . The number of bits in the frame should not have too big.

Defense Against Primary User Emulation Attacks in Cognitive Radio Networks Using Advanced Encryption Standard

This paper considers primary user emulation attacks in cognitive radio networks operating in the white spaces of the digital TV (DTV) band. We propose a reliable AES-assisted DTV scheme, in which an AES-encrypted reference signal is generated at the TV transmitter and used as the sync bits of the DTV data frames. By allowing a shared secret between the transmitter and the receiver, the reference signal can be regenerated at the receiver and used to achieve accurate identification of the authorized primary users. In addition, when combined with the analysis on the autocorrelation of the received signal, the presence of the malicious user can be detected accurately whether or not the primary user is present. We analyze the effectiveness of the proposed approach through both theoretical analysis and simulation examples. It is shown that with the AES-assisted DTV scheme, the primary user, as well as malicious user, can be detected with high accuracy under primary user emulation attacks. It should be emphasized that the proposed scheme requires no changes in hardware or system structure except for a plug-in AES chip. Potentially, it can be applied directly to today's DTV system under primary user emulation attacks for more efficient spectrum sharing.

IVC의 저지연 부호화 모드를 위한 비참조 P 프레임의 부호화 기법

Non-reference P frame coding is used to enhance coding efficiency in low-delay encoding configuration of Internet Video Coding (IVC), which is being standardized as a royalty-free video codec in MPEG. The existing method of non-reference P frame coding which was adopted in the reference Test Model of IVC (ITM) 4.0 adaptively applies a non-reference P frame with a fixed coding structure based on the magnitude of motion vectors (MVs), however, which unexpectedly degrades the coding efficiency for some sequences. In this paper, the existing non-reference P frame coding is improved by changing non-reference P frame coding structure and applying a new adaptive method using the ratio of the amount of generated bits of non-reference frames to that of reference frames as well as MVs. Experimental results show that the proposed non-reference P frame coding gives 6.6% BD-rate bit saving in average over ITM 7.0.