Real-Time Audio Transmission Using Visible Light Communication

The wide bandwidths of the communication and advanced bilateral robot control technologies have led us to realize multi-sensational teleoperation systems that provide auditory, visual, and haptic information of remote site to its operator simultaneously. Such teleoperation system is expected to be applied to medicine, education, work in hazardous environment, online game or other use. Especially, haptic sensation is extremely required to improve its operability and safety in medical operations. In this chapter, a teleoperation system is assumed to consist of three components, i.e., audio, video, and haptics transmission systems. In addition, each transmission system can be distinguished into local site and remote site. When the distance between the remote site and the local site of a teleoperation system is far enough, the performance of its communication line becomes crucial factor to decide the operability of the teleoperation. In general, since the data rate of video information is higher than the bandwidth of communication line, data compression is indispensable. Moreover, data compression and decompression are considered as time consuming processes. As a result, video transmission system must be delayed in principle. In multi-media context, audio transmission system should be delayed artificially so as to keep its playout delay to be the same as that of video transmission system. For example, lip sync is necessary for its operator to feel the audio and video contents in naturally. From this point of view, the bandwidth of its communication line is the only requirement for realize video and audio transmission systems. Even if these transmission systems are used to make a conversation for its operator with a remote person, it is said that 200ms delay is allowable. This requirement is rather negligible in the Internet in nowadays. On the other hand, a haptics transmission system requires short and stable performance of delay for communication line. This is because the achievable bandwidth of haptic sensation is limited by the round trip time of communication line since haptics transmission controller includes the delay inside of its closed control loop. In fact, a human being can feel the sense of touch at the tip of a finger up to about 400Hz. To recognize this bandwidth of haptic sensation, its sampling period must be higher than 800Hz according to Shannon’s sampling theory. Thus, the round trip time of communication line must be shorter than 1ms. This value is much shorter than the allowable delay of audio and video transmission systems. In short, haptics transmission system requires short delay while video transmission system requires wide bandwidth for communication line. 2

This paper investigates QoE (Quality of Experience) of MVV (Multi-View Video) and audio transmission over IP networks. Unlike previous works that mainly discuss MVV transmission from aspects of video codecs, we study MVV and audio transmission under various IP traffic and delay conditions by experiment. We compare two schemes: a scheme that the user watches a single viewpoint, and the one that he/she can choose one viewpoint from many viewpoints. At the same time, we perform a user study which helps us to understand the user behavior in the MVV system under those conditions. As a result, we show that high QoE can be achieved when the user can change the viewpoint with low load traffic and delay.

The project "Audio and Text Transmission Using Li-Fi Technology" explores the innovative application of Li-Fi (Light Fidelity) for transmitting both audio and text data. Li-Fi employs light signals, typically from LEDs or lasers, for data transmission, and in this project, it is leveraged to achieve two distinct functionalities: audio and text transmission. For audio transmission, a dedicated Li-Fi system is implemented where an audio signal is modulated onto the light signal emitted by an LED or laser. At the receiver end, multiple amplifiers are employed to retrieve and reproduce the audio signal. This process allows for the wireless transmission of audio data using light waves, offering a unique and potentially efficient alternative to traditional audio communication methods. For text transmission, Li-Fi modules connected to Arduino boards are employed for both the transmitter and receiver. The Arduino-based system facilitates the encoding and decoding of text data into light signals, enabling the wireless transmission of textual information using Li-Fi technology. This dual-functional approach expands the versatility of Li-Fi in communication systems, showcasing its potential for transmitting both audio and text data wirelessly through light Keywords: Arduino, Li Fi Transmitter, Li Fi receiver, LCD

Fibre Channel (FC) data transmission protocols are widely used as an interconnecting protocol to cater for the requirement of high-speed data transmission. FC has unique features like high bandwidth, strong anti-interference, flexible topology, distance insensitiveness and low latency. The highspeed, high reliability and low latency of FC communication protocol offers a promising solution for communication protocol in avionics environment. The video / data transmission system performance being critical in aerospace applications, verifying the data flow in the network is vital in order to prevent Hazardous Misleading Information being displayed to the pilot and endangering safety of flight. This paper puts forth a design approach for a receiver module of a Field Programmable Gate Array (FPGA) based high bandwidth Fiber Channel Analyser (FCA). The proposed design is to be implemented using Xilinx Spartan-6 FPGA SP605 evaluation board. The receiver module design addresses the requirements of transmission of soft real time video with resolution (VGA) and link speed with the incorporation of ethernet interface. The concept of developing a high bandwidth FCA for enhanced speed and quality of data, audio and video transmission in avionics system is critical and need of the hour. The outcome of such an effort can be utilized for multiple aerospace applications.

In order to achieve two-way real-time audio and video transmission in low bandwidth satellite communication channel between the helicopter and the ground station, we propose a real-time audio and video transmission design and implementation to control transmission delay and bit error rate in a reasonable range. This particular method really ensures the full use of the satellite channel and transmission quality of audio and video, at the same time, reduces the transmission delay and smooth video shows, etc. By simulation experiments, some satisfactory results in transmitting have been achieved.

Objectives: To analyze the performance of wireless VOIP system using wavelets. Method: In this paper a Wavelet based Data (Image, Audio and Video) Transmission System for wireless VOIP is proposed. The performance of each wavelet in this proposed system is analyzed on the basis of SNR, PSNR, RMSE, PRSE and Compression ratio. Packet Loss and Delay is also evaluated to analyze transmission capabilities. All these parameters are evaluated for the proposed system without using wavelets and with using different wavelets. Findings: The simulation results shows that the proposed system gives enhanced performance by using different wavelets when compared by without using wavelets. Finally, the Daubechies family has shown better results when compared with its counterpart’s. Applications: This wavelet based proposed system can used for high speed 4G / 5G IP based systems.

Visible light communication (VLC) has recently drawn more attention due to the existing light sources having a bandwidth that can deliver data transmission at higher bit rates. To maintain desired brightness levels along with reliable data transmission, VLC requires to control dimming levels. In this paper, a dimming control scheme based on pulse position modulation (PPM) is proposed, which constructs a dimming encoded frame per symbol of input data. The scheme utilizes multiple headers, which is selected depending on the dimming levels and the number of bits in a symbol of input data. The proposed scheme for the VLC system supports various dimming levels. The multi header PPM frame generation for input data K = 4 $K=4$ and dimming values α = 0.25, 0.375, and 0.5 is simulated using NI Labview. The proposed dimming scheme has an excellent short-run length property which is well under the maximum flickering time period and is thus capable of flicker mitigation. The slot error performance of the proposed dimming control scheme in a VLC system for varying dimming levels is simulated in Matlab. Simulation results show that the proposed scheme has excellent error performance for a low dimming value of α = 0.25. The experimental results are also carried out in the optisystem considering the practical measured parameters of the indoor environment for free space channel and simulating the VLC system for the proposed scheme. The simulation is carried for different data rates over varying link ranges and the experimental results show that the proposed scheme performs best when dimming value α = 0.25 and data rate 2 Gbps.

The popularity of 3G networks makes revolutionary changes occurred in M-learning, which enables us to do real-time video and audio learning and interactive video and audio learning. The article put emphasis on how to protect video and audio data transmission and the quality of M-learning.

Views Icon Views Article contents Figures & tables Video Audio Supplementary Data Peer Review Share Icon Share Twitter Facebook Reddit LinkedIn Tools Icon Tools Reprints and Permissions Cite Icon Cite Search Site Citation William E. Glenn, Henry Helmken; Satellite distribution of digital compressed video audio and data. AIP Conf. Proc. 25 January 1995; 325 (1): 83–89. https://doi.org/10.1063/1.47256 Download citation file: Ris (Zotero) Reference Manager EasyBib Bookends Mendeley Papers EndNote RefWorks BibTex toolbar search Search Dropdown Menu toolbar search search input Search input auto suggest filter your search All ContentAIP Publishing PortfolioAIP Conference Proceedings Search Advanced Search |Citation Search

Audio data processing and transmission in Wireless Sensor Networks (WSNs) require systems which satisfy the equilibrium between heavy data traffic and limited resources. In the lossy nature of the network, transmission techniques matter to sustain a certain content validity to be preserved. Besides, data handling is a worth-stressing issue to sustain a decreased computation and traffic overhead. An admissible quality of the voice signals being gathered at the sink node is to be directly maintained with network transmission assessments. To this respect, basic characteristics of a voice signal must be compromised with network properties. In this study, we investigate the quality of voice signals sent through the homogeneously constructed multi-hop network in which recorded signal segments in the source node are touched with the counterparts expected to be collected in each hop with a transmission rating factor. The basic characteristics of the voice samples used in the testbed are essayed with different in-network qualifications that can directly affect the quality; so that a reasonable trade-off between voice quality and sensor network capabilities is tried to be proposed.

Data acquisition techniques for computer-based systems are discussed for making continuous pressure, temperature and flow measurements. To ensure that the data faithfully represent the essential signal content over the time interval of interest, the important considerations for each system are measurement resolution, signal bandwidth, sampling, filtering and aliasing. Key data transmission techniques are discussed, starting with downhole recorders that must be hauled out of the hole for data retrieval, to telemetry systems for sending real-time or bulk data transmissions to the surface. The important issues for downhole recorders are storage media and capacity. For signal transmission systems, the issues are information modulation, data and error rate, and signal-to-noise ratio.

LMDS system is a two-way digital cellular system operating at 20–50 GHz. It offers the services of the audio, video, multimedia and data transmission, so LMDS can meet all kinds of users’ demands. However, rain attenuation and inter cell interference (ICI) influence the factor of Quality of Service (QoS) in LMDS system. In order to combat the impacts of rain fade and of inter cell interference on LMDS systems, we propose a new criterion for the choice of the puncturing patterns, leading systematic rate-compatible codes with respect to the rain attenuation to improve the performance of LMDS systems. In accordance with the system situation, a parameter with regard to rain attenuation will be used to adjust the code rate to reduce the redundancy bit and to improve the bandwidth efficiency. Simulation results show that the proposed rate compatible punctured Turbo Code for LMDS systems compared with the one without punctured Turbo Code scheme lead to an improvement of bandwidth efficiency. Moreover, the BER of two schemes are very close.

Real-time communication (RTC) is a new standard and industry-wide effort that expand the web browsing model, allowing access to information in areas like social media, chat, video conferencing, and television over the internet, and unified communication. These systems users can view, record, remark, or edit video and audio content flows using time-critical cloud infrastructures that enforce the quality of services. However, there are many proprietary protocols and codecs available that are not easily interoperable and scalable to implement multipoint video-conference systems. WebRTC (Web Real-Time Communication) is a State-of-the-Art open technology that makes real-time communication capabilities in audio, video, and data transmission possible in real-time communication through web browsers using JavaScript APIs (Application Programming Interfaces) without plug-ins. This paper aims to introduce the P2P video conferencing system based on Web Real-Time Communication (WebRTC). In this paper, we have proposed a web-based peer-to-peer real-time communication system using the Mozilla Firefox together with the ScaleDrone service that enables users to communicate with high-speed data transmission over the communication channel using WebRTC technology, HTML5 and use Node.js server address. Our experiments show that WebRTC is a capable building block for scalable live video conferencing within a web browser.We also study a thermal camera for the videoconference system to identify the temperature body results regarding the COVID 19 crisis.

In the modern world, communication between people is being more important than ever, people are seeking for a more vivid, real-time way of communication. WebRTC (Web Real-Time Communication) is an upcoming technology that enables real-time communications capabilities in audio, video and data transmission via web browsers through Javascript APIs (Application Programming Interfaces). In this paper, we implemented a new design of a cross platform multimedia conferencing system based on WebRTC technology and HTML5. The system works well under low bandwidth 10Mbps network support with 800*600 resolution ratio. Test results indicate that the system is stable and fully functional, which offers new possibilities for individuals or small businesses to build their own multimedia conferencing system.

The proposed system outlines a novel camouflage-enabled mobile robot designed for various roles that require secure and rapid audio, video, and navigation data transmission and the capabilities to camouflage with any background to aid in remote monitoring. The robot is equipped with a self-sustaining power management and obstacle detection system. The design addresses the challenges of converting operations by integrating state-of-the-art Visible Light Communication Technology for secure and rapid communication. The system’s performance is analyzed using statistical methods and Little’s theorem, providing insights into operational efficiency and transmission reliability.