电视信号——通信类外文文献翻译、中英文翻译
通信类中英文翻译、外文文献翻译

美国科罗拉多州大学关于在噪声环境下对大量连续语音识别系统的改进---------噪声环境下说话声音的识别工作简介在本文中,我们报道美国科罗拉多州大学关于噪声环境下海军研究语音词汇系统方面的最新改进成果。
特别地,我们介绍在有限语音数据的前提下,为了了解不确定观察者和变化的环境的任务(或调查方法),我们必须在提高听觉和语言模式方面努力下工夫。
在大量连续词汇语音识别系统中,我们将展开MAPLR自适应方法研究。
它包括单个或多重最大可能线形回归。
当前噪声环境下语音识别系统使用了大量声音词汇识别的声音识别引擎。
这种引擎在美国科罗拉多州大学目前得到了飞速的发展,本系统在噪声环境下说话声音系统(SPINE-2)评价数据中单词错识率表现为30.5%,比起2001年的SPINE-2来,在相关词汇错识率减少16%。
1.介绍为获得噪声环境下的有活力的连续声音系统的声音,我们试图在艺术的领域做出计算和提出改善,这个工作有几方面的难点:依赖训练的有限数据工作;在训练和测试中各种各样的军事噪声存在;在每次识别适用性阶段中,不可想象的听觉溪流和有限数量的声音。
在2000年11月的SPIN-1和2001年11月SPIN-2中,海军研究词汇通过DARPT在工作上给了很大的帮助。
在2001年参加评估的种类有:SPIIBM,华盛顿大学,美国科罗拉多州大学,AT&T,奥瑞哥研究所,和梅隆卡内基大学。
它们中的许多先前已经报道了SPINE-1和SPLNE-2工作的结果。
在这方面的工作中不乏表现最好的系统.我们在特性和主模式中使用了自适应系统,同时也使用了被用于训练各种参数类型的多重声音平行理论(例如MFCC、PCP等)。
其中每种识别系统的输出通常通过一个假定的熔合的方法来结合。
这种方法能提供一个单独的结果,这个结果的错误率将比任何一个单独的识别系统的结果要低。
美国科罗拉多州大学参加了SPIN-2和SPIN-1的两次评估工作。
我们2001年11月的SPIN-2是美国科罗拉多州大学识别系统基础上第一次被命名为SONIC(大量连续语音识别系统)的。
通信外文翻译外文文献英文文献及译文

通信外文翻译外文文献英文文献及译文通信外文翻译外文文献英文文献及译文Communication SystemA generalized communication system has the following components:(a) Information Source. This produces a message which may be written or spoken words, or some form of data.(b) Transmitter. The transmitter converts the message into a signal, the form of which is suitable for transmission over the communication channel.(c) Communication Channel. The communication channel is the medium used transmit the signal, from the transmitter to the receiver. The channel may be a radio link or a direct wire connection.(d) Receiver. The receiver can be thought of as the inverse of the transmitter. Itchanges the received signal back into a message and passes the message on to its destination which may be a loudspeaker,teleprinter or computer data bank.An unfortunate characteristic of all communication channels is that noise is added to the signal. This unwanted noise may cause distorionsof sound in a telephone, or errors in a telegraph message or data.Frequency Diversion MultiplexingFrequency Diversion Multiplexing(FDM) is a one of analog technologies. A speech signal is 0~3 kHz, single sideband amplitude (SSB) modulation can be used to transfer speech signal to new frequency bands,four similar signals, for example, moved by SSB modulationto share the band from 5 to 20 kHz. The gaps between channels are known as guard spaces and these allow for errors in frequency, inadequate filtering, etc in the engineered system.Once this new baseband signal, a "group" of 4 chEmnels, has been foimed it ismoved around the Lrunk network as a single unit. A hierarchy can be set up withseveral channels fonning a "group". several groups a "supergroup" and several"supergraup" eicher a "nmsrergroup" or "hypergroup".Groups or supergroups are moved around as single units by the communicationsequipment and it is not necessary for the radios to know how many channels are involved. A radio can handle a supergroup provided sufficient bandwidth is available. The size of the groups is a compromise as treating each channel individually involves far more equipment because separate filters, modulators and oscillators are required for every channel rather than for each group. However the failure of one module will lose all of the channels associated with a group.Time Diversion MultiplexingIt is possible, with pulse modulation systems, to use the between samples to transmit signals from other circuits. The technique is knownas time diversion multiplexing (TDM). To do this, it is necessary to employ synchronized switches at eachend of the communication links to enable samples to be transmittedin turn, from each of several circuits. Thus several subscribers appear to use the link simultaneously. Although each user onlyhas periodic short time slots, the original analog signals between samples can be reconstituted at the receiver.Pulse Code ModulationIn analog modulation, the signal was used to modulate the amplitude or frequency of a carrier, directly. However, in digital modulation a stream of pulse, representing the original,is created. This stream is then used to modulate a carrier or alternatively is transmitted directly over a cable. Pulse Code Modulation (PCM) is one of the two techniques commonly used.All pulse systems depend on the analog waveform being sampled at regular intervals. The signal created by sampling our analog speech input is known as pulse amplitude modulation. It is not very useful in practice but is used as an intermediate stage towards forming a PCM signal. It will be seen later that most of the advantages of digital modulation come from the transmitted pulses having two levels only, this being known as a binary system. In PCM the height of each sample is converted into a binary number. There are three step in the process of PCM: sampling, quantizing and coding.Optical Fiber CommunicationsCommunication may be broadly defined as the transfer of information from one point to another. When the information is to be conveyed over any distance acommunication system is usually required. Within a communication system the information transfer is frequently achieved by superimposing or modulating the information on to an electromagnetic wave which acts as a carrier for the informationsignal. This modulated carrier is then transmitted to the required destination where it is received and the originalinformation signal is obtained by demodulation. Sophisticated techniques have been developed for this process by using electromagnetic carrier wavesoperating at radio frequencies as well as microwave and millimeter wave frequencies. However,拻 communication?may also be achieved by using an electromagneticcarrier which is selected from the optical range of frequencies.In this case the information source provides an electrical signal to a transmitter comprising an electrical stage which drives an optical source to give modulation of the light-wave carrier. The optical source which provides the electrical-optical conversion may be either a semiconductor laser or light emitting diode (LED). The transmission medium consists of an optical fiber cable and the receiver consists of an optical detector which drives a further electrical stage and hence provides demodulation optical carrier. Photodiodes (P-N, P-I-N or avalanche) and , in some instances,phototransistor and photoconductors are utilized for the detection of the optical signal and the electrical-optical conversion. Thus there is a requirement for electrical interfacing at either end of the optical link and at present the signal processing is usually performed electrically.The optical carrier may be modulated by using either an analog or digital information signal. Analog modulation involves the variation of the light emitted from the optical source in a continuous manner. With digital modulation, however, discrete changes in the light intensity are obtained (i.e. on-off pulses). Although often simpler to implement, analog modulation with an optical fiber communication system is lessefficient, requiring a far higher signal to noise ratio at the receiver than digital modulation. Also, the linearity needed for analog modulation is not always provided by semiconductor optical source, especially at high modulation frequencies. For thesereasons,analog optical fiber communications link are generally limited to shorter distances and lower bandwidths than digital links.Initially, the input digital signal from the information source is suitably encoded for optical transmission. The laser drive circuit directly modulates the intensity of the semiconductor laser with the encoded digital signal. Hence a digital optical signal is launched into the optical fiber cable. The avalanche photodiode detector (APD) is followed by a fronted-end amplifier and equalizer orfilter to provide gain as well as linear signal processing and noise bandwidth reduction. Finally, the signal obtained is decoded to give the original digital information.Mobile CommunicationCordless Telephone SystemsCordless telephone system are full duplex communication systems that use radio to connect a portable handset to a dedicated base station,which is then connected to a dedicated telephone line with a specific telephone number on the public switched telephone network (PSTN) .In first generation cordless telephone systems5(manufactured in the 1980s), the portable unit communications only to the dedicatedbase unit and only over distances of a few tens of meters.Early cordless telephones operate solely as extensiontelephones to a transceiver connected to a subscriber line on the PSTN and are primarily for in-home use.Second generation cordless telephones have recently been introduced which allowsubscribers to use their handsets at many outdoor locations within urban centers such as London or Hong Kong. Modern cordless telephones are sometimes combined with paging receivers so that a subscriber may first be paged and then respond to the pageusing the cordless telephone. Cordless telephone systems provide the user with limited range and mobility, as it is usually not possible to maintain a call if the user travels outside the range of the base station. Typical second generation base stations provide coverage ranges up to a few hundred meters.Cellular Telephone SystemA cellular telephone system provides a wireless connection to the PSTN for any user location within the radio range of the system.Cellular systems accommodate alarge number of users over a large geographic area, within a limited frequency spectrum. Cellular radio systems provide high quality service that is often comparable to that of the landline telephone systems. High capacity is achieved by limiting the coverage of each base station transmitter to a small geographic area called a cell so that the same radio channels may be reused by another base station located some distance away. A sophisticated switching technique called a handoff enables a call to proceeduninterrupted when the user moves from one cell to another.A basic cellular system consists of mobile station, basestations and a mobile switching center (MSC). The Mobile Switching Center is sometimes called a mobiletelephone switching office (MTSO), since it is responsible for connecting all mobiles to the PSTN in a cellular system. Each mobilecommunicates via radio with one of the base stations and may be handed-off to any number of base stations throughout the duration of a call. The mobile station contains a transceiver, an antenna, and control circuitry,and may be mounted in a vehicle or used as a portable hand-held unit. Thebase stations consists of several transmitters and receivers which simultaneously handlefull duplex communications and generally have towers which support several transmitting and receiving antennas. The base station serves as a bridge between all mobile users in the cell and connects the simultaneous mobile calls via telephone lines or microwave links to the MSC. The MSC coordinates the activities of all the base stations and connects the entire cellular system to the PSTN. A typical MSC handles 100000 cellular subscribers and 5000 simultaneous conversations at a time, and accommodates all billing and system maintenance functions, as well. In large cities, several MSCs are used by a single carrier.Broadband CommunicationAs can be inferred from the examples of video phone and HDTV, the evolution offuture communications will be via broadband communication centered around video signals. The associated services make up a diverse set of high-speed and broadband services ranging from video services such as video phone,video conferencing,videosurveillance, cable television (CATV) distribution, and HDTV distribution to the high-speed data services such as high-resolution image transmission, high-speed datatransmission, and color facsimile. The means of standardizing these various broadbandcommunication services so that they can be provided in an integrated manner is no other than the broadband integrated services digital network (B-ISDN). Simple put, therefore, the future communications network can be said to be a broadband telecommunicationsystem based on the B-ISDN.For realization of the B-ISDN, the role of several broadband communicationtechnologies is crucial. Fortunately, the remarkable advances in the field of electronics and fiber optics have led to the maturation of broadband communication technologies.As the B-ISDN becomes possible on the optical communication foundation, the relevant manufacturing technologies for light-source and passive devices and for optical fiberhave advanced to considerable levels. Advances in high-speed device and integratedcircuit technologies for broadband signal processing are also worthy of close attention. There has also been notable progress in software, signal processing, and video equipment technologies. Hence, from the technological standpoint, the B-ISDN hasfinally reached a realizable state.On the other, standardization activities associated with broadband communication have been progressing. TheSynchronous Optical Network (SONET) standardization centered around the T1 committee eventually bore fmit in the form of the Synchronous Digital Hierarchy (SDH) standards of the International Consultative Committee in Telegraphy and Telephony (CCITT), paving the way for synchronous digital transmission based on optical communication. The standardization activities of the 5integrated services digital network (ISDN), which commenced in early 1980s with the objective of integrating narrowband services, expanded in scope with the inclusion of broadband services, leading to the standardization of the B-ISDN in late1980抯and establishing the concept of asynchronous transfer mode (ATM)communication in process. In addition, standardization of various video signals is becoming finalized through the cooperation among such organizations as CCITT, the International Radio-communications Consultative Committee (CCIR), and theInternational Standards Organization (ISO), and reference protocols for high-speedpacket communication are being standardized through ISO, CCITT, and the Institute of Electrical and Electronics Engineer (IEEE).Various factors such as these have made broadband communication realizable.5Therefore, the 1990s is the decade in which matured broadband communicationtechnologies will be used in conjunction with broadband standards to realize broadband communication networks. In the broadband communication network, the fiber opticnetwork will represent the physical medium for implementing broadband communication, while synchronous transmission will make possible the transmission of broadband service signals over the optical medium. Also, the B-ISDN will be essentialas the broadband telecommunication network established on the basis of optical medium and synchronous transmission and ATM is the communication means that enables the realization of the B-ISDN. The most important of the broadband services to be providedthrough the B-ISDN are high-speed data communication services and videocommunication services.Image AcquisitionA TV camera is usually used to take instantaneous images and transform them into electrical signals, which will be further translated into binary numbers for the computer to handle. The TV camera scans oneline at a time. Each line is further divided into hundreds of pixels. The whole frame is divided into hundreds (for example, 625) of lines.The brightness of a pixel can be represented by a binary number with certain bits, for example, 8 bits. The value of the binary number varies from 0 to 255, a range great enough to accommodate all possible contrast levels of images taken from real scene.These binary numbers are sorted in an RAM (it must have a great capacity) ready for processing by the computer.Image ProcessingImage processing is for improving the quality of the imagesobtained. First, it is necessary to improve the signal-to-noise ratio. Here noise refers to any interference flaw or aberation that obscure the objects on the image. Second, it is possible to improve contrast, enhance sharpness of edges between images through various computational means.Image AnalysisIt is for outlining all possible objects that are included in the scene. A computer program checks through the binary visual informationin store for it and identifies specific feature and characteristics of those objects. Edges or boundaries are identifiablebecause of the different brightness levels on either side of them. Usingcertain algorithms, the computer program can outline all possible boundaries of the objects in the scene. Image analysis also looks for textures and shadings between lines.Image ComprehensionImage Comprehension means understanding what is in a scene. Matching the prestored binary visual information with certain templates which represent specific objects in a binary form is technique borrowed from artificial intelligence, commonly referred to as "templeite matching"emplate matching? One by one,the templates are checked against the binary information representing the scene. Once a match occurs, an object is identified. The template matching process continues until all possible objects in the scene have been identified, otherwise it fails.通信系统一般的通信系统由下列部分组成:信源。
通信类 英文原文及译文

ABSTRACTIn this paper, we present a system using an Android smartphone that collects, displays sensor data on the screen and streams to the central server simultaneously. Bluetooth and wireless Internet connections are used for data transmissions among the devices. Also, using Near Field Communication (NFC) technology, we have constructed a more efficient and convenient mechanism to achieve an automatic Bluetooth connection and application execution. This system is beneficial on body sensor networks (BSN) developed for medical healthcare applications. For demonstration purposes, an accelerometer, a temperature sensor and electrocardiography (ECG) signal data are used to perform the experiments. Raw sensor data are interpreted to either graphical or text notations to be presented on the smartphone and the central server. Furthermore, a Java-based central server application is used to demonstrate communication with the Android system for data storage and analysis.1INTRODUCTIONMobile communication devices are designed to achieve multiple purposes but mostly are focused on voice and short messaging services. Wireless technology has the benefit of improving data mobility, using different protocols such as Wi- Fi and Bluetooth. In the medical field, many studies introduced body sensor networks (BSN) for healthcare applications. BSN improves the patient’s monitoring system with the help of the modern technology. This can be done by various wearable sensors equipped with wireless capabilities, In addition, as seen in various researches, it is desirable to develop a low power system. Different types of sensors can be used for monitoring movements, temperature changes, heart-beat, blood pressure and more to establish a patient monitoring system. Bluetooth is one of the widely available options for managing wireless networks to simultaneously connect up to 7 ancillary devices.In this paper, we introduce a microcontroller system that communicates via Bluetooth with the smartphone for data collections, and streams data simultaneously to the central server for data storage and analysis via the Internet. This system provides a solution for mobile patients by forming a wireless BSN in Bluetooth and Wi-Fi/cellular Internet connections with a common Android smartphone which can monitor the patient status via wireless data transmission.2SYSTEM DESIGNFigure 1 represents the Mobile Sensor Data Collector that involves Bluetooth, Near Field Communication (NFC) and wireless Internet connections for collecting, streaming, storing and analyzing sensor data in real-time. Three different sensors transfer sampled data to the MSP430BT5190 which communicate with the CC2560 Bluetooth transmitter via UART and sends data to a smartphone using the Android and Bluetooth system. On the phone, it displays received data on the screen and streams to the server for storage and data analysis. The term “real-time” in this paper is used to express that data transfers are achieved without perceivable delays among the devices. Also, since the Android system is capable of running application software in the background mode, the application used in this paper has the ability to transfer data during a phone call.Figure 1Overall Design of Mobile Sensor Data CollectorA Java-based UDP server application is used to collect data sent from the smartphone via the Internet. When receiving data from the smartphone, the server application displays and saves all received data to a text file for later analysis. For experimental purposes, this server was implemented with an ordinary desktop to demonstrate our fundamental idea. Also, UDP protocol was chosen over TCP because UDP usually achieves faster transmission than the TCP protocol by not waiting for an acknowledgment signal back to the origin.3EXPERIMENT RESULTSAs shown in Figure 1, all experiments are initiated using an NFC tagging process to start the Android application and initiate the Bluetooth connection automatically. In this particular smartphone, the NFC tag reader is located on the backside. The user needs to tap on the NFC tag as shown in Figure 2 to run the program. The NFC tag containing theBluetooth MAC address of the CC2560 Bluetooth device is attached to demonstrate where the tag should be located.Figure 2Initiating connection processUp to 7 ancillary sensor nodes can be simultaneously connected to the Android system. However, a single sensor Bluetooth connection was employed for testing purposes.3.1Accelerometer Data CollectionIn this paper, the Android 2.3.3 and 4.0.3 operating systems are tested using Google Nexus S to display collected data and stream data to the server. The design of the new system is achieved first by collecting sensor data from the MSP430BT5190, transferred via the CC2560 Bluetooth transmitter. Then, the Bluetooth transmitter sends data to the smartphone, which displays the collected data in real-time. As an example, Figure 3(a) shows the accelerometer data collected and displayed on the smartphone in text and Figure 3(b) shows the data in the graphical notation.Figure 3Received real-time acceleration data display(a) text notation;(b) graphical notationThese data are being sent to the central server either via Wi-Fi or cellular networks for storage and analysis at the same time. Figure 4 shows the received data from the smartphone displayed on the server. The server also saves data to a text file in the designated directory for data analysis.Figure 4Received real-time acceleration data on server An axis value representation depends on the raw sensor data and this raw data could differ from the sensors. There are 3 axes provided from the sensor and each set of data needs to be interpreted. For this particular device used in this paper, x- axis data between -60 and -50 represents LEFT, between +50 and +60 represents RIGHT. This rule applies similarly to the other two axes. This differs from other sensors where the data output of acceleration is normally represented in terms of m/s2. However, a translation algorithm shares the same idea. Figure 5 is the result of translating the accelerometer data based on accelerometer movements.Figure5Accelerometer data translationThis type of the accelerometer translation was extended to the Snake Game sample provided by Android Developers [9]. The original game uses touch screen inputs to control the snake. The touch screen inputs were replaced by accelerometer movements to provide data in LEFT, RIGHT, UP and DOWN. The data analysis was done on the phone itself for test purposes. Figure 6 shows the movement of the snake on the phone that is controlled by accelerometer data from the MSP430 eZ430-RF2560.Figure 6Remote controlling Snake GameThis example emphasizes that accelerometer data can be adapted for the patientmovement detection system. Multiple accelerometers could be implemented to produce more advanced movement analysis.3.2Temperature Sensor Data CollectionA temperature sensor monitoring the real-time room temperature is used to perform the experiment. The procedure of the experiment resembles the previous section but with the different data interpretation. In this particular experiment, a heat gun was used to heat up or cool down the sensor for testing purposes as shown in Figure 7. Similar to the previous accelerometer application, Figure 8(a) shows the text notation of the received data in real-time and Figure 8(b) shows the graphical notation of the received data in real-time. Particularly in the graphical notation output, we provide a warning message if the temperature exceeds more than 35 degrees Celsius. Also, the graphical notation has a range of between 0 degrees Celsius to 50 degrees Celsius for this demonstration.Figure 9 shows the server displaying the received data from the smartphone. It delivers similar outputs compared to the accelerometer demonstration and also saves it to a text file.Figure 7Testing temperature sensor data transmissionFigure 8Received real-time temperature data display(a) text notation;(b) graphical notationFigure 9Received real-time temperature data on server3.3Electrocardiography (ECG) Data CollectionThe ECG signal is an important part of a patient monitoring system. Currently, ECG machines are dependent on wired connections which limit their data mobility. Our system using the Bluetooth protocol for ECG signal collections greatly enhances the mobility. This ECG signal is also sent simultaneously to the server via a wireless Internet connection through the smartphone in real-time. Figure 10 shows the display of received ECG signal on the smartphone and Figure 11 shows the same result transmitted to the server in the text format.Figure 10Received real-time ECG data in graphical notationFigure 11Received real-time ECG data on serverHeart-beat rate (BPM) can be determined after analysis of the data either on the smartphone or the server. In this particular example, it represents a patient’s stablecondition with a normal heart-beat rate at approximately 72 BPM. This type of data can be diagnostically valuable and easily transmitted for consultations with distant experts.3.4Overall Data Transmission Rate (DTR)The Data Transmission Rate (DTR) is another important part of the system considering the data size. In our system, DTR depends on the microcontroller, the Bluetooth transmitter and the wireless Internet connection speed. An UART connection between the sensor and microcontroller is established at the baud rate of 115200 bps which achieves a communication bandwidth up to 15KB/s. This emphasizes that our system is capable of the data transmission by integrating multiple types of sensors for a body sensor network system that can be important for patient monitoring, real-time data analysis and diagnosis.4CONCLUSIONSIn this paper, we introduced a system using the smartphone for collecting real-time sensor data and simultaneously streaming the data to the server using Bluetooth and Internet connections. This design is the advancement over ordinary wired sensor networks which are restricted to a fixed monitoring location. In the proposed system, an accelerometer, a temperature sensor and ECG signals have been selected for data transmission using Bluetooth and wireless Internet connections. Having the Bluetooth transmitter on the smartphone, the Android system receives and displays the data on the screen in the graphical or text format and streams the collected data to the central server for data analysis, diagnosis and archiving. Taking advantage of the Android system, NFC technology was used to reduce the unnecessary Bluetooth connection process. This system is highly scalable to include more sensors to produce an upgraded patient monitoring system that is both more accurate and responsive. Furthermore, storing history of collected sensor data in the central server is extremely critical for reliable patient diagnosis.摘要在本文中,我们提出了一个使用Android智能手机,收集传感器数据显示在屏幕上并同步到中央服务器的数据流同步系统。
数据通信 毕业论文外文文献英文翻译

郑州轻工业学院本科毕业设计(论文)——英文翻译题目差错控制编码解决加性噪声的仿真学生姓名专业班级通信工程05-2 学号 12院(系)计算机与通信工程学院指导教师完成时间 2009年4月26日英文原文:Data communicationsGildas Avoine and Philippe OechslinEPFL, Lausanne, Switzerlandfgildas.avoine, philippe.oechsling@ep.chAbstractData communications are communications and computer technology resulting from the combination of a new means of communication. To transfer information between the two places must have transmission channel, according to the different transmission media, there is wired data communications and wireless data communications division. But they are through the transmission channel data link terminals and computers, different locations of implementation of the data terminal software and hardware and the sharing of information resources.1 The development of data communicationsThe first phase: the main language, through the human, horsepower, war and other means of transmission of original information.Phase II: Letter Post. (An increase means the dissemination of information)The third stage: printing. (Expand the scope of information dissemination)Phase IV: telegraph, telephone, radio. (Electric to enter the time)Fifth stage: the information age, with the exception of language information, there are data, images, text and so on.1.1 The history of modern data communicationsCommunication as a Telecommunications are from the 19th century, the beginning Year 30. Faraday discovered electromagnetic induction in 1831. Morse invented telegraph in 1837. Maxwell's electromagnetic theory in 1833. Bell invented the telephone in 1876. Marconi invented radio in 1895. Telecom has opened up in the new era. Tube invented in 1906 in order to simulate the development of communications.Sampling theorem of Nyquist criteria In 1928. Shannong theorem in 1948. The invention of the 20th century, thesemiconductor 50, thereby the development of digital communications. During the 20th century, the invention of integrated circuits 60. Made during the 20th century, 40 the concept of geostationary satellites, but can not be achieved. During the 20th century, space technology 50. Implementation in 1963 first synchronized satellite communications. The invention of the 20th century, 60 laser, intended to be used for communications, was not successful. 70 The invention of the 20th century, optical fiber, optical fiber communications can be developed.1.2 Key figuresBell (1847-1922), English, job in London in 1868. In 1871 to work in Boston. In 1873, he was appointed professor at Boston University. In 1875, invented many Telegram Rd. In 1876, invented the telephone. Lot of patents have been life. Yes, a deaf wife.Marconi (1874-1937), Italian people, in 1894, the pilot at his father's estate. 1896, to London. In 1897, the company set up the radio reported. In 1899, the first time the British and French wireless communications. 1916, implementation of short-wave radio communications. 1929, set up a global wireless communications network. Kim won the Nobel Prize. Took part in the Fascist Party.1.3 Classification of Communication SystemsAccording to type of information: Telephone communication system, Cable television system ,Data communication systems.Modulation by sub: Baseband transmission,Modulation transfer.Characteristics of transmission signals in accordance with sub: Analog Communication System ,Digital communication system.Transmission means of communication system: Cable Communications,Twisted pair, coaxial cable and so on.And long-distance telephone communication. Modulation: SSB / FDM. Based on the PCM time division multiple coaxial digital base-band transmission technology. Will gradually replace the coaxial fiber.Microwave relay communications:Comparison of coaxial and easy to set up, low investment, short-cycle. Analog phone microwave communications mainly SSB / FM /FDM modulation, communication capacity of 6,000 road / Channel. Digital microwave using BPSK, QPSK and QAM modulation techniques. The use of 64QAM, 256QAM such as multi-level modulation technique enhance the capacity of microwave communications can be transmitted at 40M Channel 1920 ~ 7680 Telephone Rd PCM figure.Optical Fiber Communication: Optical fiber communication is the use of lasers in optical fiber transmission characteristics of long-distance with a large communication capacity, communication, long distance and strong anti-interference characteristics. Currently used for local, long distance, trunk transmission, and progressive development of fiber-optic communications network users. At present, based on the long-wave lasers and single-mode optical fiber, each fiber road approach more than 10,000 calls, optical fiber communication itself is very strong force. Over the past decades, optical fiber communication technology develops very quickly, and there is a variety of applications, access devices, photoelectric conversion equipment, transmission equipment, switching equipment, network equipment and so on. Fiber-optic communications equipment has photoelectric conversion module and digital signal processing unit is composed of two parts.Satellite communications: Distance communications, transmission capacity, coverage, and not subject to geographical constraints and high reliability. At present, the use of sophisticated techniques Analog modulation, frequency division multiplexing and frequency division multiple access. Digital satellite communication using digital modulation, time division multiple road in time division multiple access.Mobile Communications: GSM, CDMA. Number of key technologies for mobile communications: modulation techniques, error correction coding and digital voice encoding. Data Communication Systems.1.4 Five basic types of data communication system:(1)Off-line data transmission is simply the use of a telephone or similar link to transmit data without involving a computer system.The equipment used at both ends of such a link is not part of a computer, or at least does not immediately make the data available for computer process, that is, the data when sent and / or received are 'off-line'.This type of data communication is relatively cheap and simple.(2)Remote batch is the term used for the way in which data communication technology is used geographically to separate the input and / or output of data from the computer on which they are processed in batch mode.(3)On-line data collection is the method of using communications technology to provide input data to a computer as such input arises-the data are then stored in the computer (say on a magnetic disk) and processed either at predetermined intervals or as required.(4)Enquiry-response systems provide, as the term suggests, the facility for a user to extract information from a computer.The enquiry facility is passive, that is, does not modify the information stored.The interrogation may be simple, for example, 'RETRIEVE THE RECORD FOR EMPLOYEE NUMBER 1234 'or complex.Such systems may use terminals producing hard copy and / or visual displays.(5)Real-time systems are those in which information is made available to and processed by a computer system in a dynamic manner so that either the computer may cause action to be taken to influence events as they occur (for example as in a process control application) or human operators may be influenced by the accurate and up-to-date information stored in the computer, for example as in reservation systems.2 Signal spectrum with bandwidthElectromagnetic data signals are encoded, the signal to be included in the data transmission. Signal in time for the general argument to show the message (or data) as a parameter (amplitude, frequency or phase) as the dependent variable. Signal of their value since the time variables are or not continuous, can be divided into continuous signals and discrete signals; according to whether the values of the dependent variable continuous, can be divided into analog signals and digital Signal.Signals with time-domain and frequency domain performance of the two most basic forms and features. Time-domain signal over time to reflect changing circumstances. Frequency domain characteristics of signals not only contain the same information domain, and the spectrum of signal analysis, can also be a clear understanding of the distribution ofthe signal spectrum and share the bandwidth. In order to receive the signal transmission and receiving equipment on the request channel, Only know the time-domain characteristics of the signal is not enough, it is also necessary to know the distribution of the signal spectrum. Time-domain characteristics of signals to show the letter .It’s changes over time. Because most of the signal energy is concentrated in a relatively narrow band, so most of our energy focused on the signal that Paragraph referred to as the effective band Bandwidth, or bandwidth. Have any signal bandwidth. In general, the greater the bandwidth of the signal using this signal to send data Rate on the higher bandwidth requirements of transmission medium greater. We will introduce the following simple common signal and bandwidth of the spectrum.More or less the voice signal spectrum at 20 Hz ~ 2000 kHz range (below 20 Hz infrasound signals for higher than 2000 KHz. For the ultrasonic signal), but with a much narrower bandwidth of the voice can produce an acceptable return, and the standard voice-frequency signal gnal 0 ~ 4 MHz, so the bandwidth of 4 MHz.As a special example of the monostable pulse infinite bandwidth. As for the binary signal, the bandwidth depends on the generalThe exact shape of the signal waveform, as well as the order of 0,1. The greater the bandwidth of the signal, it more faithfully express the number of sequences.3 The cut-off frequency channel with bandwidthAccording to Fourier series we know that if a signal for all frequency components can be completely the same through the transmission channel to the receiving end, then at the receiving frequency components of these formed by stacking up the signal and send the signal side are exactly the same, That is fully recovered from the receiving end of the send-side signals. But on the real world, there is no channel to no wear and tear through all the Frequency components. If all the Fourier components are equivalent attenuation, then the signal reception while Receive termination at an amplitude up Attenuation, but the distortion did not happen. However, all the transmission channel and equipment for different frequency components of the degree of attenuation is differentSome frequency components almost no attenuation, and attenuation of some frequency components by anumber, that is to say, channel also has a certain amount of vibrationIncrease the frequency characteristics, resulting in output signal distortion. Usually are frequency of 0 Hz to fc-wide channel at Chuan harmonic lost during the attenuation does not occur (or are a very small attenuation constant), whereas in the fc frequency harmonics at all above the transmission cross Decay process a lot, we put the signal in the transmission channel of the amplitude attenuation of a component to the original 0.707(that is, the output signal Reduce by half the power) when the frequency of the corresponding channel known as the cut-off frequency (cut - off frequency).Cut-off frequency transmission medium reflects the inherent physical properties. Other cases, it is because people interested in Line filter is installed to limit the bandwidth used by each user. In some cases, because of the add channel Two-pass filter, which corresponds to two-channel cut-off frequency f1 and f2, they were called up under the cut-off frequency and the cut-off frequency.This difference between the two cut-off frequency f2-f1 is called the channel bandwidth. If the input signal bandwidth is less than the bandwidth of channel, then the entire input signal Frequency components can be adopted by the Department of channels, which the letter Road to be the output of the output waveform will be true yet. However, if the input signal bandwidth greater than the channel bandwidth, the signal of a Frequency components can not be more on the channel, so that the signal output will be sent with the sending end of the signal is somewhat different, that is produced Distortion. In order to ensure the accuracy of data transmission, we must limit the signal bandwidth.4 Data transfer rateChannel maximum data transfer rate Unit time to be able to transfer binary data transfer rate as the median. Improve data transfer rate means that the space occupied by each Reduce the time that the sequence of binary digital pulse will reduce the cycle time, of course, will also reduce the pulse width.The previous section we already know, even if the binary digital pulse signal through a limited bandwidth channel will also be the ideal generated wave Shape distortion, and when must the input signal bandwidth, the smaller channel bandwidth, output waveformdistortion will be greater. Another angle Degree that when a certain channel bandwidth, the greater the bandwidth of the input signal, the output signal the greater the distortion, so when the data transmissionRate to a certain degree (signal bandwidth increases to a certain extent), in the on-channel output signal from the receiver could not have been Distortion of the output signal sent to recover a number of sequences. That is to say, even for an ideal channel, the limited bandwidth limit System of channel data transfer rate.At early 1924, H. Nyquist (Nyquist) to recognize the basic limitations of this existence, and deduced that the noise-free Limited bandwidth channel maximum data transfer rate formula. In 1948, C. Shannon (Shannon) put into the work of Nyquist 1 Step-by-step expansion of the channel by the random noise interference. Here we do not add on to prove to those now seen as the result of a classic.Nyquist proved that any continuous signal f (t) through a noise-free bandwidth for channel B, its output signal as a Time bandwidth of B continuous signal g (t). If you want to output digital signal, it must be the rate of g (t) for interval Sample. 2B samples per second times faster than are meaningless, because the signal bandwidth B is higher than the high-frequency component other than a letter has been Road decay away. If g (t) by V of discrete levels, namely, the likely outcome of each sample for the V level of a discrete one, The biggest channel data rate Rm ax as follows:Rmax = 2Blog 2 V (bit / s)For example, a 3000 Hz noise bandwidth of the channel should not transmit rate of more than 6,000 bits / second binary digital signal.In front of us considered only the ideal noise-free channel. There is noise in the channel, the situation will rapidly deteriorate. Channel Thermal noise with signal power and noise power ratio to measure the signal power and noise power as the signal-to-noise ratio (S ignal - to -- Noise Ratio). If we express the signal power S, and N express the noise power, while signal to noise ratio should be expressed as S / N. However, people Usually do not use the absolute value of signal to noise ratio, but the use of 10 lo g1 0S / N to indicate the units are decibels (d B). For the S / N equal 10 Channel, said its signal to noise ratio for the 1 0 d B; the same token, if the channel S / N equal to one hundred, then the signal to noiseratio for the 2 0 d B; And so on. S hannon noise channel has about the maximum data rate of the conclusions are: The bandwidth for the BH z, signal to noise ratio for the S / N Channel, the maximum data rate Rm ax as follows:Rmax = Blog 2 (1 + S / N) (bits / second)For example, for a bandwidth of 3 kHz, signal to noise ratio of 30 dB for the channel, regardless of their use to quantify the number of levels, nor Fast sampling rate control, the data transfer rate can not be greater than 30,000 bits / second. S h a n n o n the conclusions are derived based on information theory Out for a very wide scope, in order to go beyond this conclusion, like you want to invent perpetual motion machine, as it is almost impossible.It is worth noting that, S hannon conclusions give only a theoretical limit, and in fact, we should be pretty near the limit Difficult.SUMMARYMessage signals are (or data) of a magnetic encoder, the signal contains the message to be transmitted. Signal according to the dependent variable Whether or not a row of values, can be classified into analog signals and digital signals, the corresponding communication can be divided into analog communication and digital communication.Fourier has proven: any signal (either analog or digital signal) are different types of harmonic frequencies Composed of any signal has a corresponding bandwidth. And any transmission channel signal attenuation signals will, therefore, Channel transmission of any signal at all, there is a data transfer rate limitations, and this is Chengkui N yquist (Nyquist) theorem and S hannon (Shannon) theorem tells us to conclusions.Transmission medium of computer networks and communication are the most basic part of it at the cost of the entire computer network in a very Large proportion. In order to improve the utilization of transmission medium, we can use multiplexing. Frequency division multiplexing technology has many Road multiplexing, wave division multiplexing and TDM three that they use on different occasions.Data exchange technologies such as circuit switching, packet switching and packetswitching three have their respective advantages and disadvantages. M odem are at Analog phone line for the computer's binary data transmission equipment. Modem AM modulation methods have, FM, phase modulation and quadrature amplitude modulation, and M odem also supports data compression and error control. The concept of data communications Data communication is based on "data" for business communications systems, data are pre-agreed with a good meaning of numbers, letters or symbols and their combinations.参考文献[1]C.Y.Huang and A.Polydoros,“Two small SNR classification rules for CPM,”inProc.IEEE Milcom,vol.3,San Diego,CA,USA,Oct.1992,pp.1236–1240.[2]“Envelope-based classification schemes for continuous-phase binary Frequency-shift-keyed modulations,”in Pr oc.IEEE Milcom,vol.3,Fort Monmouth,NJ,USA,Oct.1994,pp. 796–800.[3]A.E.El-Mahdy and N.M.Namazi,“Classification of multiple M-ary frequency-shift keying over a rayleigh fading channel,”IEEE m.,vol.50,no.6,pp.967–974,June 2002.[4]Consulative Committee for Space Data Systems(CCSDS),Radio Frequency and Modulation SDS,2001,no.401.[5]E.E.Azzouz and A.K.Nandi,“Procedure for automatic recognition of analogue and digital modulations,”IEE mun,vol.143,no.5,pp.259–266,Oct.1996.[6]A.Puengn im,T.Robert,N.Thomas,and J.Vidal,“Hidden Markov models for digital modulation classification in unknown ISI channels,”in Eusipco2007,Poznan,Poland, September 2007,pp.1882–1885.[7]E.Vassalo and M.Visintin,“Carrier phase synchronization for GMSK signals,”I nt.J.Satell. Commun.,vol.20,no.6,pp.391–415,Nov.2002.[8]J.G.Proakis,Digital Communications.Mc Graw Hill,2001.[9]L.Rabiner,“A tutorial on hidden Markov models and selected applications in speechrecognition,”Proc.IEEE,vol.77,no.2,pp.257–286,1989.英文译文:数据通信Gildas Avoine and Philippe OechslinEPFL, Lausanne, Switzerlandfgildas.avoine, philippe.oechsling@ep.ch摘要数据通信是通信技术和计算机技术相结合而产生的一种新的通信方式。
广播电视新闻学中英文对照外文翻译文献

中英文对照外文翻译(文档含英文原文和中文翻译)外文:Communicating uncertainty - how Australian television reported H1N1 risk in 2009:a content analysis Abstract1.Background: Health officials face particular challenges in communicating with the public about emerging infectious diseases of unknown severity such as the 2009 H1N1(swine …flu) pandemic (pH1N1). Statements intended to create awareness and convey the seriousness of infectious disease threats can draw accusations of scaremongering, while officials can be accused of complacency if such statements are not made. In these communication contexts, news journalists, often reliant on official sources to understand issues are pivotal in selecting and emphasising aspects of official discourse deemed sufficiently newsworthy to present to the public. This paper presents a case-study of news communication regarding the emergence of pH1N1.2.Methods: We conducted a content analysis of all television news items aboutpH1N1. We examined news and current affairs items broadcast on 5 free-to-air Sydney television channels between April 25 2009 (the first report) and October 9 (prior to the vaccine release) for statements about the seriousness of the disease how the public could minimise contagion government responses to emerging information.3.Results: pH1N1 was the leading health story for eight of 24 weeks and was in the top 5 for 20 weeks. 353 news items were identified, yielding 3086 statements for analysis, with 63.4% related to the seriousness of the situation, 12.9% providing advice for viewers and 23.6% involving assurances from government. Coverage focused on infection/mortality rates, the spread of the virus, the need for public calm, the vulnerability of particular groups, direct and indirect advice for viewers, and government reassurances about effective management.4.Conclusions: Overall, the reporting of 2009 pH1N1 in Sydney, Australia was generally non-alarmist, while conveying that pH1N1 was potentially serious. Daily infection rate tallies and commentary on changes in the pandemic alert level were seldom contextualised to assist viewers in understanding personal relevance. Suggestions are made about how future reporting of emerging infectious diseases could be enhanced.BackgroundIn recent years , Australians have been exposed to a range of large – scale news coverage and health promotion campaigns about communicable disease. These have included seasonal influenza advisories; campaigns promoting immunisation for vaccine-preventable diseases; traveller vaccination messages; sexually transmitted disease prevention campaigns, including human papilloma virus vaccine to prevent cervical cancer ; HIV/AIDS and hepatitis B and C prevention. With the exception of HIV/AIDS and sexually-transmitted diseases, each of these has a vaccine and clear directives about how to avoid infection, forming the central communicative focus of such campaigns.The WHO-declared global pH1N1 (swine …flu) pandemic of 2009 has attractedresearch attention from virologists and infectious disease specialists, but less from communication scholars. From the first reports of Mexican cases in late April 2009, what would become sustained Australian reportage rapidly turned to the likelihood of Australian cases involving perhaps epidemic and high mortality numbers. Australians were exposed to daily news featuring the country‟s senior health officials and an array of infectious disease experts, who unavoidably, had to deal with the uncertain and complex trajectories and virulence of the disease in the context of news production cultures characterized by seven second sound-bites and an appetite for unambiguous , easily understood information.MethodsSince May 2005, the Australian Health News Research Collaboration has recorded and categorised all news, current af fairs and …infotainment‟ programmes related to health and medicine on Sydney free-to-air television stations . We searched the AHNRC database and included all items tagged with …H1N1‟ or …swine flu‟ in the period April 25, 2009 (the first mention) until October 9 2009. All stories were video clips which were used for the content analysis reported here. Using a list of content categories that emerged progressively from the content as the pH1N1 story evolved, two authors (AF and MI) compared coding on a set of 15 random clips that each watched and coded individually. After resolving any coding differences and agreeing upon how particular items should be handled, they coded the remainder of the items. These categories related to statements made regarding [1] the seriousness of H1N1, [2] recommended actions viewers were advised to take about avoiding contracting or spreading pH1N1, and [3] reassurances that the government was handling the situation. A statement was any direct (X said “Y”) or attributed ("X said that...”) quote by the journalists or news actors featured in each item. A test of inter-coder reliability produced a Kappa statistic of 0.63, indicating a good level of agreement.The role of the Internet as a platform for delivering public health interventions to specific patient groups and to the general public is constantly increasing, due in particular to its disseminating potential: the worldwide penetration of the Internet isincreasing and the use of this medium for seeking health information is frequent . Moreover, the Internet potential for individual tailoring and interactivity is superior to that of other high reach-delivery channels .ResultsA total of 353 news stories were identified, containing 3,086 statements related to the three key areas of inquiry. During the 24 weeks reported here, pH1N1 was the leading health story for eight weeks and for 20weeks remained in the top five most frequently reported health stories. We also note that the virus was rarely referred to by the name pH1N1 during the coverage and instead, was routinely termed swine …flu . When reporting exact quotes we have therefore retained the term swine …flu.(1)Seriousness Of pH1N1Of all statements , 63. 4% (n = 1, 958 /3 ,0 86 ) related to the seriousness of pH1N1 . This was communicated via four recurring stories : (i) daily tallies of infection and mortality ; ( ii ) des- criptions of spread of the virus; (iii) the need for calm responses; and (iv) the vulnerability of particular groups. We briefly summarise other statements which did not comprise significant proportions of the coverage, but may have been important to those who incidentally saw some news stories. This included similarities between pH1N1 and other viruses, government management plans, and the need for systems covering diagnosis and the anticipated vaccine roll out.(2)Advice And Recommended Actions For ViewersIn more than one third of stories (n = 131/353 - 37%) direct or indirect advice was given on what viewers could do to prevent spreading infection . However, these statements accounted for just 12.9% (n =399/3,086) of all statements. Just over a quarter (27.8%,n = 111/399) focused on basic personal hygiene, another quarter related to preventing infection by being mindful of issues of proximity (27.8%, n = 111/399) and a fifth advised seeing a doctor and seeking further information (20.6%, n = 82/399).(3)Reassurance That Government Was Handling The SituationOf all statements recorded , 23.6 % ( n = 7 2 9 / 3 , 0 8 6 )assured viewers that the government was handling the situation by elaborating on its current and proposed actions.About a third of these statements ( 2 9 . 8 % , n = 2 1 7 /7 2 9 ) referred to the immediate need for the Government to develop, test and then distribute a vaccine starting with priority groups.A quarter of these statements ( 2 5 . 9 % , n = 1 8 9 / 7 2 9 )reassured the public that the government was putting significant effort into border control measures designed to prevent pH1N1 entering Australia, and following up and containing detected infection. These statements generally concerned quarantine measures , the use of thermal imaging at airports or statements about new measures and ongoing monitoring of the situation。
通信类英文文献及翻译.doc

附录一、英文原文:Detecting Anomaly Traf?c using Flow Data in the realVoIP networkI. INTRODUCTIONRecently, many SIP[3]/RTP[4]-based VoIP applications and services haveappeared and their penetration ratio is gradually increasing due to the freeor cheap call charge and the easy subscription method. Thus, some of the subscribers to the PSTN service tend to change their home telephone servicesto VoIP products. For example, companies in Korea such as LGDacom, SamsungNet- works, and KT have begun to deploy SIP/RTP-based VoIP services. It is reportedthat more than ?ve million users have subscribed the commercial VoIP servicesand 50% of all the users are joined in 2009 in Korea [1]. According to IDC, itis expected that the number of VoIP users in US will increase to 27 millionsin 2009 [2]. Hence, as the VoIP service becomes popular, it is not surprisingthat a lot of VoIP anomaly traf ?c has been already known [5]. So, Most commercial service such as VoIP services should provide essential security functions regarding privacy, authentication, integrity and non-repudiation for preventing malicious traf ?c. Particu- larly, most of current SIP/RTP-based VoIP servicessupply the minimal security function related with authentication. Though secure transport-layer protocols such as Transport Layer Security (TLS) [6] or Secure RTP(SRTP) [7] have been standardized, they have not been fully implemented anddeployed in current VoIP applications because of the overheads of implementation and performance. Thus, un-encrypted VoIP packets could be easily sniffed andforged, especially in wireless LANs. In spite of authentication,the authentication keys such as MD5in the SIP header could be maliciously exploited, because SIP is a text-based protocol and unencrypted SIP packets are easilydecoded. Therefore, VoIP services are very vulnerable to attacks exploiting SIP and RTP. We aim at proposing a VoIP anomaly traf ?c detection method using the?ow-based traf ?c measurement archi-tecture. We consider three representativeVoIP anomalies called CANCEL,BYEDenial of Service (DoS) and RTP?ooding attacks in this paper, because we found that malicious users in wireless LANcould easily perform these attacks in the real VoIP network. For monitoring VoIP packets,we employ the IETF IP Flow Information eXport (IPFIX) [9] standard that is based on NetFlow v9. This traf ?c measurement method provides a ?exible and extensible template structure for various protocols, which is useful for observing SIP/RTP ?ows [10]. In order to capture and export VoIP packets into IPFIX ?ows, we de?ne two additional IPFIX templates for SIP and RTP ?ows. Furthermore, we add four IPFIX ?elds to observe packets which are necessary to detect VoIP source spoo?ng attacks in WLANs.II. RELATED WORK[8] proposed a ?ooding detection method by the Hellinger Distance (HD) concept. In [8], they have pre- sented INVITE, SYN and RTP?ooding detection meth-ods.The HD is the difference value between a training data set and a testing dataset. The training data set collected traf?c over n sampling period of duration testing data set collected traf?c next the training data set in the sameperiod. If the HD is close to ‘1’, this testing data set is regarded as anomaly traf ?c. For using this method, they assumed that initial training data set didnot have any anomaly traf ?c. Since this method was based on packet counts, itmight not easily extended to detect other anomaly traf ?c except ?ooding. On the other hand, [11] has proposed a VoIP anomaly traf ?c detection method using Extended Finite State Machine (EFSM). [11] has suggested INVITE ?ooding, BYEDoS anomaly traf ?c and media spamming detection methods. However, the statemachine required more memory because it had to maintain each ?ow. [13] has presented NetFlow-based VoIP anomaly detection methods for INVITE, REGIS-TER,RTP?ooding, and REGISTER/INVITEscan. How-ever, the VoIP DoSattacks consideredin this paper were not considered. In [14], an IDS approach to detect SIPanomalies was developed, but only simulation results are presented. For monitoring VoIP traf ?c, SIPFIX [10] has been proposed as an IPFIX extension.The key ideas of the SIPFIX are application-layer inspection and SDP analysisfor carrying media session information. Yet, this paper presents only the possibility of applying SIPFIX to DoS anomaly traf ?c detection and prevention. Wedescribed the preliminary idea of detecting VoIP anomaly traf ?c in [15]. This paper elaborates BYEDoSanomaly traf ?c and RTP?ooding anomaly traf ?c detec-tion method based on IPFIX. Based on [15], we have considered SIP and RTP anomalytraf ?c generated in wireless LAN. In this case, it is possible to generate thesimiliar anomaly traf ?c with normal VoIP traf ?c, because attackers can easilyextract normal user information from unencrypted VoIP packets. In this paper,we have extended the idea with additional SIP detection methods using informationof wireless LAN packets. Furthermore, we have shown the real experiment resultsat the commercial VoIP network.III. THE VOIP ANOMALY TRAFFIC DETECTION METHOD A. CANCEL DoS Anomaly Traf ?c DetectionAs the SIP INVITE message is not usually encrypted, attackers could extract ?elds necessary to reproduce the forged SIP CANCELmessage by snif ?ng SIP INVITE packets, especially in wireless LANs. Thus, wecannot tell the difference between the normal SIP CANCEL message and the replicated one, because the faked CANCEL packet includes the normal ?elds inferred from the SIP INVITE message. Theattacker will perform the SIP CANCELDoS attack at the samewireless LAN, because the purpose of the SIP CANCELattack is to prevent the normal call estab-lishment when a victim is waiting for calls. Therefore, as soon as the attacker catchesa call invitation message for a victim, it will send a SIP CANCELmessage, which makes the call establishment failed. Wehave generated faked SIP CANCELmessage using sniffed a SIP INVITE in SIP header of this CANCEL message is the sameas normal SIP CANCEL message, because the attacker can obtain the SIP header?eld from unencrypted normal SIP message in wireless LANenvironment. Therefore it is impossible to detect the CANCEL DoS anomaly traf ?c using SIP headers, we use the different values of the wireless LANframe. That is, the sequence number in the frame will tell the difference between a victim host and an attacker.Welook into source MACaddress and sequence number in the MAC frame including a SIP CANCEL messageas shown in Algorithm 1. We compare the source MAC address of SIP CANCEL packets with that of the previously saved SIP INVITE ?ow. If the source MAC address of a SIP CANCEL ?ow is changed, it will be highly probablethat the CANCEL packet is generated by a unknown user. However, the source MAC address could be spoofed. Regarding source spoo ?ng detection, we employ the method in [12] that uses sequence numbers of frames. We calculate the gapbetween n-th and (n-1)-th frames. As the sequence number ?eld in a MAC header uses 12 bits, it varies from 0 to 4095. When we ?nd that the sequence number gap between a single SIP ?ow is greater than the threshold value of N that willbe set from the experiments, we determine that the SIP host address as beenspoofed for the anomaly traf ?c.B. BYE DoS Anomaly Traf ?c DetectionIn commercial VoIP applications, SIP BYE messages use the same authentication ?eld is included in the SIP IN-VITE message for security andaccounting purposes. How-ever, attackers can reproduce BYEDoS packets through snif ?ng normal SIP INVITE packets in wireless faked SIP BYE message is samewith the normal SIP BYE. Therefore, it is dif ?cult to detect the BYEDoS anomaly traf ?c using only SIP header snif ?ng SIP INVITE message, the attacker at the same or different subnets could terminate the normal in- progress call, because it could succeed in generating a BYE message to the SIP proxy server. In theSIP BYE attack, it is dif ?cult to distinguish from the normal call termination procedure. That is, we apply the timestamp of RTP traf ?c for detecting the SIP BYE attack. Generally, after normal call termination, the bi-directional RTP?ow is terminated in a bref space of time. However, if the call terminationprocedure is anomaly, we can observe that a directional RTP media ?ow is still ongoing, whereas an attacked directional RTP?ow is broken. Therefore, in order to detect the SIP BYE attack, we decide that we watch a directional RTP ?ow for a long time threshold of N sec after SIP BYEmessage. The threshold of N is also set from the 2 explains the procedure to detect BYE DoS anomal traf ?c using captured timestamp of the RTPpacket. Wemaintain SIP session information between clients with INVITE and OK messages including the same Call-ID and 4-tuple(source/destination IP Address and port number) of the BYEpacket. Weset a time threshold value by adding Nsec to the timestamp value of the BYE message. Thereason why we use the captured timestamp is that a few RTP packets are observed under second. If RTP traf ?c is observed after the time threshold, this willbe considered as a BYE DoS attack, because the VoIP session will be terminatedwith normal BYEmessages. C. RTPAnomaly Traf ?c Detection Algorithm 3 describes an RTP ?ooding detection method that uses SSRC and sequence numbers of the RTP header. During a single RTPsession, typically, the sameSSRCvalue is maintained. If SSRCis changed, it is highly probable that anomaly has occurred. In addition,if there is a big sequence number gap between RTP packets, we determine thatanomaly RTPtraf ?c has happened. As inspecting every sequence number for a packet is dif ?cult, we calculate the sequence number gap using the ?rst, last, maximum and minimum sequence numbers. In the RTP header, the sequence number ?eld uses 16 bits from 0 to 65535. When we observe a wide sequence number gap in our algorithm, we consider it as an RTP ?ooding attack.IV. PERFORMANCE EVALUATIONA. Experiment EnvironmentIn order to detect VoIP anomaly traf ?c, we established an experimental environment as ?gure 1. In this envi-ronment, we employed two VoIP phones with wireless LANs, one attacker, a wireless access router and an IPFIX ?ow collector.For the realistic performance evaluation, we directly used one of the workingVoIP networks deployed in Korea where an 11-digit telephone number (070-XXXX-XXXX) has been assigned to a SIP wireless SIP phones supporting ,we could make calls to/from the PSTNor cellular phones. In the wireless access router, we used two wireless LAN cards- one is to support the AP service, andthe other is to monitor packets. Moreover, in order to observe VoIP packetsin the wireless access router, we modi ?ed nProbe [16], that is an open IPFIX?ow generator, to create and export IPFIX ?ows related with SIP, RTP, and information. As the IPFIX collector, we have modi ?ed libip ?x so that it could provide the IPFIX ?ow decoding function for SIP, RTP, and templates. We used MySQL for the ?ow DB.B. Experimental ResultsIn order to evaluate our proposed algorithms, we gen-erated 1,946 VoIP callswith two commercial SIP phones and a VoIP anomaly traf ?c generator. Table I showsour experimental results with precision, recall, and F-score that is the harmonic mean of precision and recall. In CANCEL DoS anomaly traf ?c detection, our algorithm represented a few false negative cases, which was related with thegap threshold of the sequence number in MAC header. The average of the F-score value for detecting the SIP CANCEL anomaly is %.For BYE anomaly tests, wegenerated 755 BYEmes-sages including 118 BYEDoSanomalies in the exper-iment. The proposed BYE DoS anomaly traf ?c detec-tion algorithm found 112 anomalieswith the F-score of %. If an RTP?ow is terminated before the threshold, we regard the anomaly ?ow as a normal one. In this algorithm, we extract RTP sessioninformation from INVITE and OK or session description messages using the sameCall-ID of BYE message. It is possible not to capture those packet, resultingin a few false-negative cases. The RTP ?ooding anomaly traf ?c detection experiment for 810 RTP sessions resulted in the F score of 98%.The reason offalse-positive cases was related with the sequence number in RTP header. If the sequence number of anomaly traf ?c is overlapped with the range of the normaltraf ?c, our algorithm will consider it as normal traf ?c.V. CONCLUSIONSWe have proposed a ?ow-based anomaly traf ?c detec-tion method against SIP and RTP-based anomaly traf ?c in this paper. We presented VoIP anomaly traf ?c detection methods with ?ow data on the wireless access router. Weused the IETF IPFIX standard to monitor SIP/RTP ?ows passing through wireless access routers, because its template architecture is easily extensible to several protocols.For this purpose, we de ?ned two new IPFIX templates for SIP and RTP traf ?c and four new IPFIX ?elds for traf ?c. Using these IPFIX ?ow templates,we proposed CANCEL/BYE DoS and RTP?ooding traf ?c detection algorithms. From experimental results on the working VoIP network in Korea, we showed that our method is able to detect three representative VoIP attacks on SIP phones. In CANCEL/BYE DoS anomaly traf ?cdetection method, we employed threshold values about time and sequence numbergap for class ?cation of normal and abnormal VoIP packets. This paper has notbeen mentioned the test result about suitable threshold values. For the futurework, we will show the experimental result about evaluation of thethreshold values for our detection method.二、英文翻译:交通流数据检测异常在真实的世界中使用的VoIP 网络一.介绍最近 , 许多 SIP[3],[4]基于服务器的VoIP应用和服务出现了,并逐渐增加他们的穿透比及由于自由和廉价的通话费且极易订阅的方法。
GSM移动通信系统综述——通信类外文文献翻译、中英文翻译

GSM移动通信系统综述GSM的历史在十九世纪八十年代,蜂窝电话系统在欧洲迅速发展起来,特别是在斯堪的纳维亚和联合国,还有法国和德国。
每个国家发展自己的系统,在设备和运营方面和别的其他国家不相同。
这是一个不受欢迎的情况,因为移动设备不仅受国界的限制,(这在统一的欧洲变的越来越不重要),而且还受每种设备类型的市场限制,以至于如此的经济规模和储蓄不能被实现。
欧洲首先认识到这种情况,在1982年欧洲邮电行政大会成立了一个欧洲移动特别小组,简称GSM,形成这个小组为了研究和发展欧洲的移动陆地通信系统,所提出的这个系统必须遵循以下几个标准;●好的话音质量。
●低的终端服务成本。
●支持国际漫游。
●支持手持终端。
●支持新的服务和设备。
●高效的光谱。
●ISDN兼容性。
在1989年,GSM的责任是被欧洲电讯学会标准所接受。
GSM规范的第一阶段于1990年被公布,商业服务在1991年被推行,到1993年,在22个国家有36个GSM网络。
虽然标准定型在欧洲,但GSM不只是欧洲的标准,超过200个GSM 网络(包括DCS1800和PCS1900)在世界上110个国家运营。
在1994年初,世界上有1.3百万个用户,到1997年10月已经超过了55百万个用户。
北美洲进入GSM领域比较晚,而且随之有一个GSM派生物叫PCS1900,GSM在每个大陆存在,而缩写词GSM代表了全球移动通信系统。
GSM 的发展选择了一个(在时间上)被分割的数字系统,相反的是,像美洲的AMPS和联合国TACS 一样标准的模拟的细胞系统。
他们相信那个处于压缩状态的算法和数字信号处理器的进展,允许实现原来的标准和在连续不断改进的系统方面的质量和费用。
超过八千页的GSM系统介绍尽量允许给中间供给者以灵活性和竞争性,但是足够的标准化保证在系统组成部分之间互相交织。
这个被通过为每个在系统中的定义的功能实体提供功能和交织描述。
GSM所提供的服务从开始,GSM的计划者想在提供的服务和信号使用的控制方面考虑ISDN 的兼容性。
通信类英文文献及翻译

附录一、英文原文:Detecting Anomaly Traffic using Flow Data in the realVoIP networkI. INTRODUCTIONRecently, many SIP[3]/RTP[4]-based VoIP applications and services have appeared and their penetration ratio is gradually increasing due to the free or cheap call charge and the easy subscription method. Thus, some of the subscribers to the PSTN service tend to change their home telephone services to VoIP products. For example, companies in Korea such as LG Dacom, Samsung Net- works, and KT have begun to deploy SIP/RTP-based VoIP services. It is reported that more than five million users have subscribed the commercial VoIP services and 50% of all the users are joined in 2009 in Korea [1]. According to IDC, it is expected that the number of VoIP users in US will increase to 27 millions in 2009 [2]. Hence, as the VoIP service becomes popular, it is not surprising that a lot of VoIP anomaly traffic has been already known [5]. So, Most commercial service such as VoIP services should provide essential security functions regarding privacy, authentication, integrity and non-repudiation for preventing malicious traffic. Particu- larly, most of current SIP/RTP-based VoIP services supply the minimal security function related with authentication. Though secure transport-layer protocols such as Transport Layer Security (TLS) [6] or Secure RTP (SRTP) [7] have been standardized, they have not been fully implemented anddeployed in current VoIP applications because of the overheads of implementation and performance. Thus, un-encrypted VoIP packets could be easily sniffed and forged, especially in wireless LANs. In spite of authentication,the authentication keys such as MD5 in the SIP header could be maliciously exploited, because SIP is a text-based protocol and unencrypted SIP packets are easily decoded. Therefore, VoIP services are very vulnerable to attacks exploiting SIP and RTP. We aim at proposing a VoIP anomaly traffic detection method using the flow-based traffic measurement archi-tecture. We consider three representative VoIP anomalies called CANCEL, BYE Denial of Service (DoS) and RTP flooding attacks in this paper, because we found that malicious users in wireless LAN could easily perform these attacks in the real VoIP network. For monitoring VoIP packets, we employ the IETF IP Flow Information eXport (IPFIX) [9] standard that is based on NetFlow v9. This traffic measurement method provides a flexible and extensible template structure for various protocols, which is useful for observing SIP/RTP flows [10]. In order to capture and export VoIP packets into IPFIX flows, we define two additional IPFIX templates for SIP and RTP flows. Furthermore, we add four IPFIX fields to observe packets which are necessary to detect VoIP source spoofing attacks in WLANs.II. RELATED WORK[8] proposed a flooding detection method by the Hellinger Distance (HD) concept. In [8], they have pre- sented INVITE, SYN and RTP flooding detection meth-ods. The HD is the difference value between a training data set and a testing data set. The training data set collected traffic over n sampling period of duration Δ testing data set collected traffic next the training data set in the same period. If the HD is close to ‘1’, this testing data set is regarded as anomaly traffic. For using this method, they assumed that initial training data set didnot have any anomaly traffic. Since this method was based on packet counts, it might not easily extended to detect other anomaly traffic except flooding. On the other hand, [11] has proposed a VoIP anomaly traffic detection method using Extended Finite State Machine (EFSM). [11] has suggested INVITE flooding, BYE DoS anomaly traffic and media spamming detection methods. However, the state machine required more memory because it had to maintain each flow. [13] has presented NetFlow-based VoIP anomaly detection methods for INVITE, REGIS-TER, RTP flooding, and REGISTER/INVITE scan. How-ever, the VoIP DoS attacks considered in this paper were not considered. In [14], an IDS approach to detect SIP anomalies was developed, but only simulation results are presented. For monitoring VoIP traffic, SIPFIX [10] has been proposed as an IPFIX extension. The key ideas of the SIPFIX are application-layer inspection and SDP analysis for carrying media session information. Yet, this paper presents only the possibility of applying SIPFIX to DoS anomaly traffic detection and prevention. We described the preliminary idea of detecting VoIP anomaly traffic in [15]. This paper elaborates BYE DoS anomaly traffic and RTP flooding anomaly traffic detec-tion method based on IPFIX. Based on [15], we have considered SIP and RTP anomaly traffic generated in wireless LAN. In this case, it is possible to generate the similiar anomaly traffic with normal VoIP traffic, because attackers can easily extract normal user information from unencrypted VoIP packets. In this paper, we have extended the idea with additional SIP detection methods using information of wireless LAN packets. Furthermore, we have shown the real experiment results at the commercial VoIP network.III. THE VOIP ANOMALY TRAFFIC DETECTION METHOD A. CANCEL DoS Anomaly Traffic DetectionAs the SIP INVITE message is not usually encrypted, attackers could extract fields necessary to reproduce the forged SIP CANCEL message by sniffing SIP INVITE packets, especially in wireless LANs. Thus, we cannot tell the difference between the normal SIP CANCEL message and the replicated one, because the faked CANCEL packet includes the normal fields inferred from the SIP INVITE message. The attacker will perform the SIP CANCEL DoS attack at the same wireless LAN, because the purpose of the SIP CANCEL attack is to prevent the normal call estab-lishment when a victim is waiting for calls. Therefore, as soon as the attacker catches a call invitation message for a victim, it will send a SIP CANCEL message, which makes the call establishment failed. We have generated faked SIP CANCEL message using sniffed a SIP INVITE in SIP header of this CANCEL message is the same as normal SIP CANCEL message, because the attacker can obtain the SIP header field from unencrypted normal SIP message in wireless LAN environment. Therefore it is impossible to detect the CANCEL DoS anomaly traffic using SIP headers, we use the different values of the wireless LAN frame. That is, the sequence number in the frame will tell the difference between a victim host and an attacker. We look into source MAC address and sequence number in the MAC frame including a SIP CANCEL message as shown in Algorithm 1. We compare the source MAC address of SIP CANCEL packets with that of the previously saved SIP INVITE flow. If the source MAC address of a SIP CANCEL flow is changed, it will be highly probable that the CANCEL packet is generated by a unknown user. However, the source MAC address could be spoofed. Regarding source spoofing detection, we employ the method in [12] that uses sequence numbers of frames. We calculate the gap between n-th and (n-1)-th frames. As the sequence number field in a MAC header uses 12 bits, it varies from 0 to 4095. When we find that the sequence number gap between a single SIP flow is greater than the threshold value of N that willbe set from the experiments, we determine that the SIP host address as been spoofed for the anomaly traffic.B. BYE DoS Anomaly Traffic DetectionIn commercial VoIP applications, SIP BYE messages use the same authentication field is included in the SIP IN-VITE message for security and accounting purposes. How-ever, attackers can reproduce BYE DoS packets through sniffing normal SIP INVITE packets in wireless faked SIP BYE message is same with the normal SIP BYE. Therefore, it is difficult to detect the BYE DoS anomaly traffic using only SIP header sniffing SIP INVITE message, the attacker at the same or different subnets could terminate the normal in- progress call, because it could succeed in generating a BYE message to the SIP proxy server. In the SIP BYE attack, it is difficult to distinguish from the normal call termination procedure. That is, we apply the timestamp of RTP traffic for detecting the SIP BYE attack. Generally, after normal call termination, the bi-directional RTP flow is terminated in a bref space of time. However, if the call termination procedure is anomaly, we can observe that a directional RTP media flow is still ongoing, whereas an attacked directional RTP flow is broken. Therefore, in order to detect the SIP BYE attack, we decide that we watch a directional RTP flow for a long time threshold of N sec after SIP BYE message. The threshold of N is also set from the 2 explains the procedure to detect BYE DoS anomal traffic using captured timestamp of the RTP packet. We maintain SIP session information between clients with INVITE and OK messages including the same Call-ID and 4-tuple (source/destination IP Address and port number) of the BYE packet. We set a time threshold value by adding Nsec to the timestamp value of the BYE message. The reason why we use the captured timestamp is that a few RTP packets are observed under second. If RTP traffic is observed after the time threshold, this willbe considered as a BYE DoS attack, because the VoIP session will be terminated with normal BYE messages. C. RTP Anomaly Traffic Detection Algorithm 3 describes an RTP flooding detection method that uses SSRC and sequence numbers of the RTP header. During a single RTP session, typically, the same SSRC value is maintained. If SSRC is changed, it is highly probable that anomaly has occurred. In addition, if there is a big sequence number gap between RTP packets, we determine that anomaly RTP traffic has happened. As inspecting every sequence number for a packet is difficult, we calculate the sequence number gap using the first, last, maximum and minimum sequence numbers. In the RTP header, the sequence number field uses 16 bits from 0 to 65535. When we observe a wide sequence number gap in our algorithm, we consider it as an RTP flooding attack.IV. PERFORMANCE EVALUATIONA. Experiment EnvironmentIn order to detect VoIP anomaly traffic, we established an experimental environment as figure 1. In this envi-ronment, we employed two VoIP phones with wireless LANs, one attacker, a wireless access router and an IPFIX flow collector. For the realistic performance evaluation, we directly used one of the working VoIP networks deployed in Korea where an 11-digit telephone number (070-XXXX-XXXX) has been assigned to a SIP wireless SIP phones supporting , we could make calls to/from the PSTN or cellular phones. In the wireless access router, we used two wireless LAN cards- one is to support the AP service, and the other is to monitor packets. Moreover, in order to observe VoIP packets in the wireless access router, we modified nProbe [16], that is an open IPFIX flow generator, to create and export IPFIX flows related with SIP, RTP, and information. As the IPFIX collector, we have modified libipfix so that it could provide the IPFIX flow decoding function for SIP, RTP, and templates. We used MySQL for the flow DB.B. Experimental ResultsIn order to evaluate our proposed algorithms, we gen-erated 1,946 VoIP calls with two commercial SIP phones and a VoIP anomaly traffic generator. Table I showsour experimental results with precision, recall, and F-score that is the harmonic mean of precision and recall. In CANCEL DoS anomaly traffic detection, our algorithm represented a few false negative cases, which was related with the gap threshold of the sequence number in MAC header. The average of the F-score value for detecting the SIP CANCEL anomaly is %.For BYE anomaly tests, we generated 755 BYE mes-sages including 118 BYE DoS anomalies in the exper-iment. The proposed BYE DoS anomaly traffic detec-tion algorithm found 112 anomalies with the F-score of %. If an RTP flow is terminated before the threshold, we regard the anomaly flow as a normal one. In this algorithm, we extract RTP session information from INVITE and OK or session description messages using the same Call-ID of BYE message. It is possible not to capture those packet, resulting in a few false-negative cases. The RTP flooding anomaly traffic detection experiment for 810 RTP sessions resulted in the F score of 98%.The reason of false-positive cases was related with the sequence number in RTP header. If the sequence number of anomaly traffic is overlapped with the range of the normal traffic, our algorithm will consider it as normal traffic.V. CONCLUSIONSWe have proposed a flow-based anomaly traffic detec-tion method against SIP and RTP-based anomaly traffic in this paper. We presented VoIP anomaly traffic detection methods with flow data on the wireless access router. We used the IETF IPFIX standard to monitor SIP/RTP flows passing through wireless access routers, because its template architecture is easily extensible to several protocols. For this purpose, we defined two new IPFIX templates for SIP and RTP traffic and four new IPFIX fields for traffic. Using these IPFIX flow templates,we proposed CANCEL/BYE DoS and RTP flooding traffic detection algorithms. From experimental results on the working VoIP network in Korea, we showed that our method is able to detect three representative VoIP attacks on SIP phones. In CANCEL/BYE DoS anomaly trafficdetection method, we employed threshold values about time and sequence number gap for classfication of normal and abnormal VoIP packets. This paper has not been mentioned the test result about suitable threshold values. For the future work, we will show the experimental result about evaluation of the threshold values for our detection method.二、英文翻译:交通流数据检测异常在真实的世界中使用的VoIP网络一 .介绍最近,许多SIP[3],[4]基于服务器的VoIP应用和服务出现了,并逐渐增加他们的穿透比及由于自由和廉价的通话费且极易订阅的方法。
- 1、下载文档前请自行甄别文档内容的完整性,平台不提供额外的编辑、内容补充、找答案等附加服务。
- 2、"仅部分预览"的文档,不可在线预览部分如存在完整性等问题,可反馈申请退款(可完整预览的文档不适用该条件!)。
- 3、如文档侵犯您的权益,请联系客服反馈,我们会尽快为您处理(人工客服工作时间:9:00-18:30)。
原文:Television Video SignalsAlthough over 50 years old , the standard television signal is still one of the most common way to transmit an image. Figure 8.3 shows how the television signal appears on an oscilloscope. This is called composite video, meaning that there are vertical and horizontal synchronization (sync) pulses mixed with the actual picture information.These pulses are used in the television receiver to synchronize the vertical and horizontal deflection circuits to match the video being displayed. Each second of standard video contains 30 complete images, commonly called frames , A video engineer would say that each frame contains 525 lines, the television jargon for what programmers call rows. This number is a little deceptive because only 480 to 486 of these lines contain video information; the remaining 39to 45 lines are reserved for sync pulses to keep the television’s circuits synchronized with the video signal.Standard television uses an interlaced format to reduce flicker in the displayed image. This means that all the odd lines of each frame are transmitted first, followed by the even lines. The group of odd lines is called the odd field, and the group of even lines is called the even field. Since each frame consists of two fields, the video signal transmits 60 fields per second. Each field starts with a complex series of vertical sync pulses lasting 1.3 milliseconds. This is followed by either the even or odd lines of video. Each line lasts for 63.5 microseconds, including a 10.2 microsecond horizontal sync pulse, separating one line from the next. Within each line, the analog voltage corresponds to the gray scale of the image, with brighter values being in the direction away from the sync pulses. This place the sync beyond the black range. In video jargon, the sync pulses are said to be blacker than black..The hardware used for analog-to-digital conversion of video signals is called a frame grabber. This is usually in the form of an electronics card that plugs into a computer, and connects to a camera through a coaxial cable. Upon command from software, the frame grabber waits for the beginning of the next frame, as indicated by the vertical sync pulses. During the following two fields,each line of video is sampled many times, typically 512,640 or 720 samples per line, at 8bits per sample. These samples are stored in memory as one row of the digital image.This way of acquiring a digital image results in an important difference between the vertical and horizontal directions. Each row in the digital image corresponds to one line in the video signal, and therefore to one row of wells in the CCD. Unfortunately,the columns are not so straightforward. In the CCD, each row contains between about 400 and 800 wells (columns), depending on the particular device used. When a row of wells is read from the CCD, the resulting line of video is filtered into a smooth analog signal, such as in Figure 8.3. In other words, the video signal does not depend on how many columns are present in the CCD. The resolution in the horizontal direction is limited by how rapidly the analog signal is allowed to change. This is usually set at 3.2 MHz for color television, resulting in a rise time of about 100 nanoseconds, i.e, about1/500th of the 53.2 microsecond video line.When the video signal is digitized in the frame grabber, it is converted back into columns, However, these columns in the digitized image have no relation to the columns in the CCD. The number of columns in the digital image depends solely on how many times the frame grabber samples each line of video. For example, a CCD might have 800 wells per row, while the digitized image might only have 512 pixels (i.e , columns) per row.The number of columns in the digitized image is also important for another reason. The standard television image has an aspect ratio of 4 to 3, i.e. , it is slightly wider than it is high. Motion pictures have the wider aspect ratio of 25 to 9. CCDs used for scientific applications often have an aspect ratio of 1 to 1, i.e , a perfect square. In any event, the aspect ratio of a CCD is fixed by the placement of the electrodes, and cannot be altered. However, the aspect ratio of the digitized image depends on the number of samples per line. This becomes a problem when the image is displayed, either on a video monitor or in a hardcopy. If the aspect ratio isn’t properly reproduced, the image looks squashed horizontally or vertically.The 525 line video signal described here is called NTSC (National Television Systems Committee), a standard defined way back in 1954. This is the system used in the United States and Japan. In Europe there are two similar standards called PAL (Phase Alternation by Line) and SECAM (Sequential Chrominance And Memory). The basic concepts are the same , just the numbers are different. Both PAL and SECAM operate with 25 interlaced frames per second, with 625 lines per frame. Just as with NTSC, some of these lines occur during the vertical sync, resulting in about 576 lines that carry picture information. Other more subtle differences relate to how color and sound are added to the signal.The most straightforward way of transmitting color television would be to have three separate analog signals, one for each of the three colors the human eye can detect: red, green and blue. Unfortunately, the historical development of television did not allow such a simple scheme. The color television signal was developed to allow existing blackand white television sets to remain in use without modification. This was done by retaining the same signal for brightness information , but adding a separate signal for color information. In video jargon, the brightness is called the luminance signal, while the color is the chrominance signal. The chrominance signal is contained on a 3.58 MHz carrier wave added to the black and white video signal. Sound is added in this same way, on a 4.5 MHz carrier wave. The television receiver separates these three signals, processes them individually, and recombines them in the final diplay.译文:电视信号尽管已经拥有50年的历史了,电视信号依然是常用的传递信息的途径之一。