简易数字调制系统性能研究与仿真外文翻译
外文翻译---6 数字数据传输：接口和调制解调器
英文资料及中文翻译6 TRANSMISSIONS OF DIGITAL DATA:INTERFACES AND MODEMS(From Introduction to Data Communications and Net Working,Behrouz Forouzan)Once we have encoder our information into a format that can be transmitted, the next step is to investigate the transmission process itself. Information-processing equipment such as PCs generate encoded signals but ordinarily require assistance to transmit those signals over a communication link. For example, a PC generates a digital signal but needs an additional device to modulate a carrier frequency before it is sent over a telephone line. How do we relay encoded data from the generating device to the next device in the process? The answer is a bundle of wires, a sort of mini communication link, called an interface.Because an interface links two devices not necessarily made by the same manufacturer, its characteristics must be defined and standards must be established. Characteristics of an interface include its mechanical specifications (how many wires are used to transport the signal); its electrical specifications (the frequency, amplitude, and phase of the expected signal); and its functional specifications (if multiple wires are used, what does each one do?). These characteristics are all described by several popular standards and are incorporated in the physical layer of the OSI model.6.1 DIGITAL DATA TRANSMISSIONOf primary concern when considering the transmission of data from one device to another is the wiring. And of primary concern when considering the wiring is the data stream. Do we send one bit at a time, or do we group bits into larger groups and, if so, how? The transmission of binary data across a link can be accomplished either in parallel mode or serial mode. In parallel mode, multiple bits are sent with each clock pulse. In serial mode, one bit is sent with each clock pulse. While there is only one way to send parallel data, there are two subclasses of serial transmission: synchronous and asynchronous (see Figure 6-1).Parallel TransmissionBinary data, consisting of 1s and 0s, may be organized into groups of n bits each. Computers produce and consume data in groups of bits much as we conceive of and use spoken language in the form of words rather than letters. By grouping, we cansend data n bits at a time instead of one. This is called parallel transmission.The mechanism for parallel transmissionis a conceptually simple one: use n wires to send n bits at one time. That way each bit has its own wire, and all n bits of one group can be transmitted with each clock pulse from one device to another. Figure 6-2 shows how parallel transmission works for n=8.Typically the eight wires are bundled in a cable with a connector at each end.Figure 6-2 Parallel transmissionThe advantage of parallel transmission is speed. All else being equal, parallel transmission can increase the transfer speed by a factor of n over serial transmission. But there is a significant disadvantage:cost. Parallel transmission requires n communication lines (wires in the example) just to transmit the data stream. Because this is expensive, parallel transmission is usually limited to short distances, up to a maximum of say 25 feet.Serial TransmissionIn serial transmission one bit follows another, so we need only one communication channel rather than n to transmit data between two communicating devices .The advantage of serial over parallel transmission is that with only one communication channel, serial transmission reduces the cost of transmission over parallel by roughly a factor of n.Since communication within devices is parallel, conversion devices are required at the interface between the sender and the line (parallel-to-parallel).Serial transmission occurs in one of two ways: asynchronous or synchronous. Asynchronous TransmissionAsynchronous transmission is so named because the timing of a signal is unimportant. Instead, information is received and translated by agreed-upon patterns. As long as those patterns are followed, the receiving device can retrieve the information without regard to the rhythm in which it is sent. Patterns are based on grouping the bit stream into bytes. Each group, usually eight bits, is sent along the link as a unit. The sending system handles each group independently, relaying it to the link whenever ready, without regard to a timer.Without a synchronizing pulse, the receiver cannot use timing to predict when the next group will arrive. To alert the receiver to the arrival of a new group, therefore, an extra bit is added to the beginning of each byte. This bit, usually a 0, is called the start bit. To let the receiver know that the byte is finished, one or more additional bits are appended to the end of the byte. These bits, usually 1s, are called stop bits. By this method, each byte is increased in size to at least 10 bits, of which 8 are information and 2 or more are signals to the receiver. In addition, the transmission of each byte may then be followed by a gap of varying duration. This gap can be represented either by an idle channel or by a stream of additional stop bits.In asynchronous transmission we send one start bit (0) at the beginning and one or more stop bits (1s) at the end of each byte. There may be a gap between each byte.The start and stop bits and the gap alert the receiver to the beginning and end of each byte and allow it to synchronize with the data stream. This mechanism is called asynchronous because, at the byte level, sender and receiver do not have to be synchronized. But within each byte, the receiver must still be synchronized with the incoming bit stream. This is, some synchronization is required, but only for the duration of a single byte. The receiving device resynchronizes at the onset of each new byte. When the receiver detects a start bit, it sets a timer and begins counting bits as they come in. after n bits the receiver looks for a stop bit. As soon as it detects the stop bit, it ignores any received pulses until it detects the next start bit.Asynchronou s here means “asynchronous at the byte level,” but the bits are still synchronized; their durations are the same.The addition of stop and start bits and the insertion of gaps into the bit stream make asynchronous transmission slower than forms of transmission that can operate without the addition of control information. But it is cheap and effective, two advantages that make it an attractive choice for situations like low-speed communication. For example, the connection of a terminal to a computer is a natural application for asynchronous transmission. A user types only one character at a time, types extremely slowly in data processing terms, and leaves unpredictable gaps of time between each character.Synchronous TransmissionIn synchronous transmission, the bit stream is combined into longer “frames,” which may contain multiple bytes. Each byte, however, is introduced onto the transmission link without a gap between it and the next one. It is left to the receiver to separate the bit stream into bytes for decoding purposes. In other words, data are transmitted as an unbroken string of 1s and 0s, and the receiver separates that string into the bytes, or characters, it needs to reconstruct the information.In synchronous transmission we send bits one after another without start/stop bits or gaps. It is the responsibility of the receiver to group the bits.Without gaps and start/stop bits, there is no built-in mechanism to help the receiving device adjust its bit synchronization in midstream. Timing becomes very important, therefore, because the accuracy of the received information is completely dependent on the ability of the receiving device to keep an accurate count of the bits as they come in.The advantage of synchronous transmission is speed. With no extra bits or gaps to introduce at the sending end and remove at the receiving end and, by extension, with fewer bits to move across the link, synchronous transmission is faster than asynchronous transmission is faster than asynchronous transmission. For this reason, it is more useful for high-speed applications like the transmission of data from one computer to another. Byte synchronization is accomplished in the data link layer.6.2 DTE-DCE INTERFACAt this point we must clarify two terms important to computer networking: data terminal equipment (DTE). There are usually four basic functional units involved in the communication of data: a DTE and DCE on one end and a DCE and DTE on theother end. The DTE generates the data and passes them, along with any necessary control characters, to a DCE. The DCE does the job of converting the signal to a format appropriate to the transmission medium and introducing it onto the network link. When the signal arrives at the receiving end, this process is reversed.Data Terminal Equipment (DTE)Data terminal equipment (DTE) includes any unit that functions either as a source of or as a destination for binary digital data. At the physical layer, if can be a terminal, microcomputer, computer, printer, fax machine, or any other device that generates or consumes digital data. DTEs do not often communicate directly with one another, they generate and consume information but need an intermediary to be able to communicate. Think of a DTE as operating the way your brain does when you talk. Let’s say you have an idea that you want to communicate to a friend. Your brain creates the idea but cannot transmit that idea to your friend’s brain by itself. Unfortunately or fortunately, we are not a species of mind readers. Instead, your brain passes the idea to your vocal chords and mouth, which convert it to sound waves that can travel through the air or over a telephone line to your friend’s ear and from there to his or her brain, where it is converted back into information. In this model, your brain and your friend’s brain are DTEs. Your vocal chords and mouth are your DCE. His or her ear is also a DCE. The air or telephone wire is your transmission medium.A DTE is any device that is a source of or destination for binary digital data. Data Circuit-Terminating Equipment (DCE)Data circuit-terminating equipment (DCE) includes any functional unit that transmits or receives data in the form of an analog or digital signal through a network. At the physical layer, a DCE takes data generated by a DTE, converts them to an appropriate signal, and then introduces the signal onto the telecommunication link. Commonly used DCEs at this layer include modems . In any network, a DTE generates digital data and passes it to a DCE; the DCE converts the data to a form acceptable to the transmission medium and sends the converted signal to another DCE on the network. The second DCE takes the signal off the line, converts it to a form usable by its DTE, and delivers it. To make this communication possible, both the sending and receiving DCEs must use the same encoding method, much the way that if you want to communicate to someone who understands only Japanese, you must speak Japanese. The two DTEs do not need to be coordinated with each other, but each of them must be coordinated with its own DCE and the DCEs must becoordinated so that data translation occurs without loss of integrity.A DCE is any device that transmits or receives data in the form of an analog or digital signal through a network.6 数字数据传输：接口和调制解调器（选自«数据通信与网络»， Behrouz Forouzan著）我们将信息编码成可以传输的格式，下一步就是探讨传输过程了。
毕业设计外文翻译-基于MATLAB的TD-SCDMA通信系统的调制与解调仿真程序设计
Review of UMTS1.1 UMTS Network ArchitectureThe European/Japanese 3G standard is referred to as UMTS. UMTS is one of a number of standards ratified by the ITU-T under the umbrella of IMT-2000. It is currently the dominant standard, with the US CDMA2000 standard gaining ground, particularly with operators that have deployed cdmaOne as their 2G technology. At time of writing,Japan is the most advanced in terms of 3G network deployment. The three incumbent operators there have implemented three different technologies: J-Phone is using UMTS,KDDI has a CDMA2000 network, and the largest operator NTT DoCoMo is using a system branded as FOMA (Freedom of Multimedia Access). FOMA is based on the original UMTS proposal, prior to its harmonization and standardization.The UMTS standard is specified as a migration from the second generation GSM standard to UMTS via the General Packet Radio System (GPRS) and Enhanced Data for Global Evolution (EDGE), as shown in Figure. This is a sound rationale since as of April 2003, there were over 847 Million GSM subscribers worldwide1, accounting for68% of the global cellular subscriber figures. The emphasis is on keeping as much ofthe GSM network as possible to operate with the new system.We are now well on the road towards Third Generation (3G), where the network will support all traffic types: voice, video and data, and we should see an eventual explosion in the services available on the mobile device. The driving technology for this is the IP protocol. Many cellular operators are now at a position referred to as 2.5G, with the deployment of GPRS, which introduces an IP backbone into the mobile core network.The diagram below, Figure 2, shows an overview of the key components in a GPRS network, and how it fits into the existing GSM infrastructure.The interface between the SGSN and GGSN is known as the Gn interfaceand uses the GPRS tunneling protocol (GTP, discussed later). The primary reason for the introduction of this infrastructure is to offer connections to external packet networks, such as the Internet or a corporate Intranet.This brings the IP protocol into the network as a transport between the SGSN and GGSN. This allows data services such as email or web browsing on the mobile device,with users being charged based on volume of data rather than time connected.The dominant standard for delivery of 3G networks and services is the Universal Mobile Telecommunications System, or UMTS. The first deployment of UMTS is the Release ’99 architecture, shown below in Figure 3.In this network, the major change is in the radio access network (RAN) with the introduction of CDMA technology for the air interface, and ATM as a transport in the transmission part. These changes have been introduced principally to support the transport of voice, video and data services on the same network. The core network remains relatively unchanged, with primarily software upgrades. However, the IP protocol pushes further into the network with the RNC now communicating with the 3G SGSN using IP.The next evolution step is the Release 4 architecture, Figure 4. Here, the GSM core is replaced with an IP network infrastructure based around V oice over IP technology.The MSC evolves into two separate components: a Media Gateway (MGW) and an MSC Server (MSS). This essentially breaks apart the roles of connection and connection control. An MSS can handle multiple MGWs, making the network more scaleable.Since there are now a number of IP clouds in the 3G network, it makes sense to merge these together into one IP or IP/ATM backbone (it is likely both options will be available to operators.) This extends IP right across the whole network, all the way to the BTS.This is referred to as the All-IP network, or the Release 5 architecture, as shown in Figure 5. The HLR/VLR/EIR are generalised and referred to as the HLR Subsystem(HSS).Now the last remnants of traditional telecommunications switching are removed, leaving a network operating completely on the IP protocol, and generalised for the transport of many service types. Real-time services are supported through the introduction of a new network domain, the IP Multimedia Subsystem (IMS).Currently the 3GPP are working on Release 6, which purports to cover all aspects not addressed in frozen releases. Some call UMTS Release 6 4G and it includes such issues as interworking of hot spot radio access technologies such as wireless LAN1.2 UMTS FDD and TDDLike any CDMA system, UMTS needs a wide frequency band in which to operate to effectively spread signals. The defining characteristic of the system is the chip rate, where a chip is the width of one symbol of the CDMA code. UMTS uses a chip rate of 3.84Mchips/s and this converts to a required spectrum carrier of 5MHz wide. Since this is wider than the 1.25MHz needed for the existi ng cdmaOne system, the UMTS air interface is termed ‘wideband’ CDMA.There are actually two radio technologies under the UMTS umbrella: UMTS FDD and TDD. FDD stands for Frequency Division Duplex, and like GSM, separates traffic in the uplink and downlink by placing them at different frequency channels. Therefore an operator must have a pair of frequencies allocated to allow them to run a network, hence the term ‘paired spectrum’. TDD or Time Division Duplex requires only one frequency channel, and uplink and downlink traffic are separated by sending them at different times. The ITU-T spectrum usage, as shown in Figure 6, for FDD is 1920- 980MHz for uplink traffic, and 2110-2170MHz for downlink. The minimum allocation an operator needs is two paired 5MHz channels, one for uplink and one for downlink, at a separation of 190MHz. However, to provide comprehensive coverage and services, it is recommended that an operator be given three channels. Considering the spectrum allocation, there are 12 paired channels available, and many countries have now completed the licencing process for this spectrum, allocating between two and four channels per licence. This has tended to workout a costly process for operators, since the regulatory authorities in some countries, notably in Europe, have auctioned these licences to the highest bidder. This has resulted in spectrum fees as high as tens of billions of dollars in some countries.The Time Division Duplex (TDD) system, which needs only one 5MHz band in which to operate, often referred to as unpaired spectrum. The differences between UMTS FDD and TDD are only evident at the lower layers, particularly on the radio interface. At higher layers, the bulk of the operation of the two systems is the same. As the name suggests, the TDD system separates uplink and downlink traffic by placing them in different time slots. As will be seen later, UMTS uses a 10ms frame structure which is divided into 15 equal timeslots. TDD can allocate these to be either uplink or downlink,with one or more breakpoints between the two in a frame defined. In this way, it is well suited to packet traffic, since this allows great flexibility in dynamically dimensioning for asymmetry in traffic flow.The TDD system should not really be considered as an independent network, but rather as a supplement for an FDD system to provide hotspot coverage at higher data rates. It is rather unsuitable for large scale deployment due to interference between sites, since a BTS may be trying to detect a weak signal from a UE, which is blocked out by a relatively strong signal at the same frequency from a nearby BTS. TDD is ideal for indoor coverage over small areas.Since FDD is the main access technology being developed currently, the explanations presented here will focus purely on this system.1.3 UMTS Bearer ModelThe procedures of a mobile device connecting to a UMTS network can be split into two areas: the access stratum (AS) and the non-access stratum (NAS). The access stratum involves all the layers and subsystems that offer general services to the non-access stratum. In UMTS, the access stratum consists of all of the elements in the radio access network, including the underlying ATMtransport network, and the various mechanisms such as those to provide reliable information exchange. All of the non-access stratum functions are those between the mobile device and the core network, for example, mobility management. Figure 7 shows the architecture model. The AS interacts with the NAS through the use of service access points (SAPs).UMTS radio access network (UTRAN) provides this separation of NAS and AS functions, and allows for AS functions to be fully controlled and implemented within the UTRAN. The two major UTRAN interfaces are the Uu, which is the interface between the mobile device, or User Equipment (UE) and the UTRAN, and the Iu, which is the interface between the UTRAN and the core network. Both of these interfaces can be divided into control and user planes each with appropriate protocol functions.A Bearer Service is a link between two points, which is defined by a certain set of characteristics. In the case of UMTS, the bearer service is delivered using radio access bearers.A Radio access bearer (RAB) is defined as the service that the access stratum (i.e.UTRAN) provides to the non-access stratum for transfer of user data between the User Equipment and Core Network. A RAB can consist of a number of subflows, which are data streams to the core network within the RAB that have different QoS characteristics,such as different reliabilities. A common example of this is different classes of bits with different bit error rates can be realised as different RAB subflows. RAB subflows are established and released at the time the RAB is established and released, and are delivered together over the same transport bearer.A Radio Link is defined as a logical association between a single User Equipment (UE) and a single UTRAN access point, such as an RNC. It is physically comprised of one or more radio bearers and should not be confused with radio access bearer.Looking within the UTRAN, the general architecture model is as shown in Figure 8 below. Now shown are the Node B or Base Station (BTS) and RadioNetwork Controller (RNC) components, and their respective internal interfaces. The UTRAN is subdivided into blocks referred to as Radio Network Subsystems (RNS), where each RNS consists of one controlling RNC (CRNC) and all the BTSs under its control. Unique to UMTS is the interface between RNSs, the Iur interface, which plays a key role in handover procedures. The interface between the BTS and RNC is the Iub interface.All the ‘I’ interfaces: Iu, Iur and Iub, currently3 use ATM as a transport layer. In the context of ATM, the BTS is seen as a host accessing an ATM network, within which the RNC is an ATM switch. Therefore, the Iub is a UNI interface, whereas the Iu and Iur interfaces are considered to be NNI, as illustrated in Figure 9.This distinction is because the BTS to RNC link is a point-to-point connection in that a BTS or RNC will only communicate with the RNC or BTS directly connected to it, and will not require communication beyond that element to another network element.For each user connection to the core network, there is only one RNC, which maintains the link between the UE and core network domain, as highlighted in Figure 10. This RNC is referred to as the serving RNC or SRNC. That SRNC plus the BTSs under its control is then referred to as the SRNS. This is a logical definition with reference to that UE only. In an RNS, the RNC that controls a BTS is known as the controlling RNC or CRNC. This is with reference to the BTS, cells under its control and all the common and shared channels within.As the UE moves, it may perform a soft or hard handover to another cell. In the case of a soft handover, the SRNC will activate the new connection to the new BTS. Should the new BTS be under the control of another RNC, the SRNC will also alert this new RNC to activate a connection along the Iur interface. The UE now has two links, one directly to the SRNC, and the second, through the new RNC along the Iur interface. In this case, this new RNC is logically referred to as a drift RNC or DRNC, see Figure 10. It is not involved in any processing of the call and merely relays it to the SRNC for connection to the core. In summary,SRNC and DRNC are usually associated with the UE and the CRNC is associated with the BTS. Since these are logical functions it is normal practice that a single RNC is capable of dealing with all these functions.A situation may arise where a UE is connected to a BTS for which the SRNC is not the CRNC for that BTS. In that situation, the network may invoke the Serving RNC Relocation procedure to move the core network connection. This process is described inSection 3.通用移动通信系统的回顾1.1 UMTS网络架构欧洲/日本的3G标准，被称为UMTS。
数字调制系统仿真分析liu
基于SystemView的2FSK数字调制系统仿真分析姓名：序号：摘要：数字调制是通信系统中最为重要的环节之一，数字调制技术的改进也是通信系统性能提高的重要途径。
本文在简单介绍SystemView仿真软件应用的基础上，建立了2FSK通信系统的模型。
首先介绍了该系统的基本工作原理。
然后，运用SystemView设计了2FSK数字调制解调方法的仿真模型。
通过仿真，观察了2FSK的2种不同的调制和解调过程中的各环节的波形，并进行对比，判断调制解调的正确性，加深了对2FSK调制解调方法和原理的认识同时也熟悉了SystemView仿真软件的基本操作。
关键词：2FSK 调制解调SystemView 仿真分析引言通信的最终目的是在一定的距离内传递信息。
虽然基带数字信号可以在传输距离相对较近的情况下直接传送，但如果要远距离传输时，特别是在无线或光纤信道上传输时，则必须经过调制将信号频谱搬移到高频处才能在信道中传输。
为了使数字信号在有限带宽的高频信道中传输，必须对数字信号进行载波调制。
而在接受信号时则必须对已调制信号进行解调操作才能接受到发送的信息。
一般数字调制方法分为两种模拟调制法和键控法。
解调方法也可分为两种：相干解调和非相干解调。
本文主要针对2FKS（二进制频移键控）的各种调制解调方法作仿真分析。
System View是美国ELANIX公司推出的，基于Windows环境下运行的用于系统仿真分析的可视化软件工具，它使用功能模块（Token）描述程序。
利用System View，可以构造各种复杂的模拟，数字，模数混合系统和各种多速率系统，因此，它可用于各种线性或非线性控制系统的设计和仿真。
用户在进行系统设计时，只需从System View 配置的图标库中调出有关图标并进行参数设置，完成图标之间的连线，然后运行仿真操作，最终以时域波形、眼图、功率谱等形式给出系统的仿真分析结果。
一、设计的目的本课程设计是实现2FSK的调制和解调仿真分析。
数字调制系统的Monte Carlo仿真和性能分析
数字调制系统的Monte Carlo仿真和性能分析数字调制系统是通过数字信号处理技术实现的一种现代通信系统，普遍应用于广播、移动通信、卫星通信、互联网等领域。
在数字调制系统的设计过程中，通过Monte Carlo仿真和性能分析可以对系统的性能进行评估和优化，下面就数字调制系统的Monte Carlo仿真和性能分析进行介绍。
一、Monte Carlo仿真Monte Carlo方法是通过随机抽样的方式进行试验，通过试验结果的统计分析得出所求问题的数值解。
在数字调制系统中，Monte Carlo方法可以用于评估系统的误码率、功率谱等性能指标。
其步骤如下：1. 确定系统的模型和信道模型2. 定义误码率、功率谱等性能指标3. 确定仿真参数，如信噪比、码率、符号周期等4. 进行多次随机仿真，并统计所求性能指标5. 根据仿真结果对系统进行分析和优化。
二、性能分析性能分析是通过数学解析的方式来分析系统的性能指标。
在数字调制系统中，常用的性能分析方法有极限分析、误差分析和波形分析等。
其主要特点是可以有效地分析系统的性能和优化设计，但需要对系统具有较深的理解和掌握。
1. 极限分析极限分析是通过系统的数学模型和信道模型，使用极限条件来分析系统的性能极限。
例如，在高斯信道中，通过无穷小误差的假设，可以推导出系统的误码率上限，对系统的性能进行分析和优化。
2. 误差分析误差分析是通过对系统中各参数误差的分析，来分析系统的误差传递和影响。
例如，在数字调制系统中，由于声学振荡器（VCO）的频率稳定度存在限制，会对系统的调制误差率产生影响，通过对VCO的误差进行分析和优化，可以提高系统的性能。
3. 波形分析波形分析是通过对传输波形的解析，来分析系统的性能。
例如，在OFDM系统中，通过对多个子载波的功率谱分析，可以优化系统的频带利用率和错误率性能。
总之，数字调制系统的Monte Carlo仿真和性能分析是对系统性能评估和优化的重要手段，在系统设计过程中应该充分运用这些方法，对系统进行全面深入的分析，提高系统的性能和稳定性。
数字无线通信系统中的调制(英文)
AgilentDigital Modulation in Communications Systems—An IntroductionApplication Note 1298This application note introduces the concepts of digital modulation used in many communications systems today. Emphasis is placed on explaining the tradeoffs that are made to optimize efficiencies in system design.Most communications systems fall into one of three categories: bandwidth efficient, power efficient, or cost efficient. Bandwidth efficiency describes the ability of a modulation scheme to accommodate data within a limited bandwidth. Power efficiency describes the ability of the system to reliably send information at the lowest practical power level.In most systems, there is a high priority on band-width efficiency. The parameter to be optimized depends on the demands of the particular system, as can be seen in the following two examples.For designers of digital terrestrial microwave radios, their highest priority is good bandwidth efficiency with low bit-error-rate. They have plenty of power available and are not concerned with power efficiency. They are not especially con-cerned with receiver cost or complexity because they do not have to build large numbers of them. On the other hand, designers of hand-held cellular phones put a high priority on power efficiency because these phones need to run on a battery. Cost is also a high priority because cellular phones must be low-cost to encourage more users. Accord-ingly, these systems sacrifice some bandwidth efficiency to get power and cost efficiency. Every time one of these efficiency parameters (bandwidth, power, or cost) is increased, another one decreases, becomes more complex, or does not perform well in a poor environment. Cost is a dom-inant system priority. Low-cost radios will always be in demand. In the past, it was possible to make a radio low-cost by sacrificing power and band-width efficiency. This is no longer possible. The radio spectrum is very valuable and operators who do not use the spectrum efficiently could lose their existing licenses or lose out in the competition for new ones. These are the tradeoffs that must be considered in digital RF communications design. This application note covers•the reasons for the move to digital modulation;•how information is modulated onto in-phase (I) and quadrature (Q) signals;•different types of digital modulation;•filtering techniques to conserve bandwidth; •ways of looking at digitally modulated signals;•multiplexing techniques used to share the transmission channel;•how a digital transmitter and receiver work;•measurements on digital RF communications systems;•an overview table with key specifications for the major digital communications systems; and •a glossary of terms used in digital RF communi-cations.These concepts form the building blocks of any communications system. If you understand the building blocks, then you will be able to under-stand how any communications system, present or future, works.Introduction25 5 677 7 8 8 9 10 10 1112 12 12 13 14 14 15 15 16 17 18 19 20 21 22 22 23 23 24 25 26 27 28 29 29 30 311. Why Digital Modulation?1.1 Trading off simplicity and bandwidth1.2 Industry trends2. Using I/Q Modulation (Amplitude and Phase Control) to Convey Information2.1 Transmitting information2.2 Signal characteristics that can be modified2.3 Polar display—magnitude and phase representedtogether2.4 Signal changes or modifications in polar form2.5 I/Q formats2.6 I and Q in a radio transmitter2.7 I and Q in a radio receiver2.8 Why use I and Q?3. Digital Modulation Types and Relative Efficiencies3.1 Applications3.1.1 Bit rate and symbol rate3.1.2 Spectrum (bandwidth) requirements3.1.3 Symbol clock3.2 Phase Shift Keying (PSK)3.3 Frequency Shift Keying3.4 Minimum Shift Keying (MSK)3.5 Quadrature Amplitude Modulation (QAM)3.6 Theoretical bandwidth efficiency limits3.7 Spectral efficiency examples in practical radios3.8 I/Q offset modulation3.9 Differential modulation3.10 Constant amplitude modulation4. Filtering4.1 Nyquist or raised cosine filter4.2 Transmitter-receiver matched filters4.3 Gaussian filter4.4 Filter bandwidth parameter alpha4.5 Filter bandwidth effects4.6 Chebyshev equiripple FIR (finite impulse response) filter4.7 Spectral efficiency versus power consumption5. Different Ways of Looking at a Digitally Modulated Signal Time and Frequency Domain View5.1 Power and frequency view5.2 Constellation diagrams5.3 Eye diagrams5.4 Trellis diagramsTable of Contents332 32 32 33 33 34 3435 35 3637 37 37 38 38 39 39 39 40 41 41 42 434344466. Sharing the Channel6.1 Multiplexing—frequency6.2 Multiplexing—time6.3 Multiplexing—code6.4 Multiplexing—geography6.5 Combining multiplexing modes6.6 Penetration versus efficiency7. How Digital Transmitters and Receivers Work7.1 A digital communications transmitter7.2 A digital communications receiver8. Measurements on Digital RF Communications Systems 8.1 Power measurements8.1.1 Adjacent Channel Power8.2 Frequency measurements8.2.1 Occupied bandwidth8.3 Timing measurements8.4 Modulation accuracy8.5 Understanding Error Vector Magnitude (EVM)8.6 Troubleshooting with error vector measurements8.7 Magnitude versus phase error8.8 I/Q phase error versus time8.9 Error Vector Magnitude versus time8.10 Error spectrum (EVM versus frequency)9. Summary10. Overview of Communications Systems11. Glossary of TermsTable of Contents (continued)4The move to digital modulation provides more information capacity, compatibility with digital data services, higher data security, better quality communications, and quicker system availability. Developers of communications systems face these constraints:•available bandwidth•permissible power•inherent noise level of the systemThe RF spectrum must be shared, yet every day there are more users for that spectrum as demand for communications services increases. Digital modulation schemes have greater capacity to con-vey large amounts of information than analog mod-ulation schemes. 1.1 Trading off simplicity and bandwidthThere is a fundamental tradeoff in communication systems. Simple hardware can be used in transmit-ters and receivers to communicate information. However, this uses a lot of spectrum which limits the number of users. Alternatively, more complex transmitters and receivers can be used to transmit the same information over less bandwidth. The transition to more and more spectrally efficient transmission techniques requires more and more complex hardware. Complex hardware is difficult to design, test, and build. This tradeoff exists whether communication is over air or wire, analog or digital.Figure 1. The Fundamental Tradeoff1. Why Digital Modulation?51.2 Industry trendsOver the past few years a major transition has occurred from simple analog Amplitude Mod-ulation (AM) and Frequency/Phase Modulation (FM/PM) to new digital modulation techniques. Examples of digital modulation include•QPSK (Quadrature Phase Shift Keying)•FSK (Frequency Shift Keying)•MSK (Minimum Shift Keying)•QAM (Quadrature Amplitude Modulation) Another layer of complexity in many new systems is multiplexing. Two principal types of multiplex-ing (or “multiple access”) are TDMA (Time Division Multiple Access) and CDMA (Code Division Multiple Access). These are two different ways to add diversity to signals allowing different signals to be separated from one another.Figure 2. Trends in the Industry62.1 Transmitting informationTo transmit a signal over the air, there are three main steps:1.A pure carrier is generated at the transmitter.2.The carrier is modulated with the informationto be transmitted. Any reliably detectablechange in signal characteristics can carryinformation.3.At the receiver the signal modifications orchanges are detected and demodulated.2.2 Signal characteristics that can be modified There are only three characteristics of a signal that can be changed over time: amplitude, phase, or fre-quency. However, phase and frequency are just dif-ferent ways to view or measure the same signal change. In AM, the amplitude of a high-frequency carrier signal is varied in proportion to the instantaneous amplitude of the modulating message signal.Frequency Modulation (FM) is the most popular analog modulation technique used in mobile com-munications systems. In FM, the amplitude of the modulating carrier is kept constant while its fre-quency is varied by the modulating message signal.Amplitude and phase can be modulated simultane-ously and separately, but this is difficult to gener-ate, and especially difficult to detect. Instead, in practical systems the signal is separated into another set of independent components: I(In-phase) and Q(Quadrature). These components are orthogonal and do not interfere with each other.Figure 3. Transmitting Information (Analog or Digital)Figure 4. Signal Characteristics to Modify2. Using I/Q Modulation to Convey Information72.3 Polar display—magnitude and phase repre-sented togetherA simple way to view amplitude and phase is with the polar diagram. The carrier becomes a frequency and phase reference and the signal is interpreted relative to the carrier. The signal can be expressed in polar form as a magnitude and a phase. The phase is relative to a reference signal, the carrier in most communication systems. The magnitude is either an absolute or relative value. Both are used in digital communication systems. Polar diagrams are the basis of many displays used in digital com-munications, although it is common to describe the signal vector by its rectangular coordinates of I (In-phase) and Q(Quadrature).2.4 Signal changes or modifications inpolar formFigure 6 shows different forms of modulation in polar form. Magnitude is represented as the dis-tance from the center and phase is represented as the angle.Amplitude modulation (AM) changes only the magnitude of the signal. Phase modulation (PM) changes only the phase of the signal. Amplitude and phase modulation can be used together. Frequency modulation (FM) looks similar to phase modulation, though frequency is the controlled parameter, rather than relative phase.Figure 6. Signal Changes or Modifications8One example of the difficulties in RF design can be illustrated with simple amplitude modulation. Generating AM with no associated angular modula-tion should result in a straight line on a polar display. This line should run from the origin to some peak radius or amplitude value. In practice, however, the line is not straight. The amplitude modulation itself often can cause a small amount of unwanted phase modulation. The result is a curved line. It could also be a loop if there is any hysteresis in the system transfer function. Some amount of this distortion is inevitable in any sys-tem where modulation causes amplitude changes. Therefore, the degree of effective amplitude modu-lation in a system will affect some distortion parameters.2.5 I/Q formatsIn digital communications, modulation is often expressed in terms of I and Q. This is a rectangular representation of the polar diagram. On a polar diagram, the I axis lies on the zero degree phase reference, and the Q axis is rotated by 90 degrees. The signal vector’s projection onto the I axis is its “I” component and the projection onto the Q axisis its “Q” component.Figure 7. “I-Q” Format92.6 I and Q in a radio transmitterI/Q diagrams are particularly useful because they mirror the way most digital communications sig-nals are created using an I/Q modulator. In the transmitter, I and Q signals are mixed with the same local oscillator (LO). A 90 degree phase shifter is placed in one of the LO paths. Signals that are separated by 90 degrees are also known as being orthogonal to each other or in quadrature. Signals that are in quadrature do not interfere with each other. They are two independent compo-nents of the signal. When recombined, they are summed to a composite output signal. There are two independent signals in I and Q that can be sent and received with simple circuits. This simpli-fies the design of digital radios. The main advan-tage of I/Q modulation is the symmetric ease of combining independent signal components into a single composite signal and later splitting such a composite signal into its independent component parts. 2.7 I and Q in a radio receiverThe composite signal with magnitude and phase (or I and Q) information arrives at the receiver input. The input signal is mixed with the local oscillator signal at the carrier frequency in two forms. One is at an arbitrary zero phase. The other has a 90 degree phase shift. The composite input signal (in terms of magnitude and phase) is thus broken into an in-phase, I, and a quadrature, Q, component. These two components of the signal are independent and orthogonal. One can be changed without affecting the other. Normally, information cannot be plotted in a polar format and reinterpreted as rectangular values without doing a polar-to-rectangular conversion. This con-version is exactly what is done by the in-phase and quadrature mixing processes in a digital radio. A local oscillator, phase shifter, and two mixers can perform the conversion accurately and efficiently.Figure 8. I and Q in a Practical Radio Transmitter Figure 9. I and Q in a Radio Receiver102.8 Why use I and Q?Digital modulation is easy to accomplish with I/Q modulators. Most digital modulation maps the data to a number of discrete points on the I/Q plane. These are known as constellation points. As the sig-nal moves from one point to another, simultaneous amplitude and phase modulation usually results. To accomplish this with an amplitude modulator and a phase modulator is difficult and complex. It is also impossible with a conventional phase modu-lator. The signal may, in principle, circle the origin in one direction forever, necessitating infinite phase shifting capability. Alternatively, simultaneous AM and Phase Modulation is easy with an I/Q modulator. The I and Q control signals are bounded, but infi-nite phase wrap is possible by properly phasing the I and Q signals.This section covers the main digital modulation formats, their main applications, relative spectral efficiencies, and some variations of the main modulation types as used in practical systems. Fortunately, there are a limited number of modula-tion types which form the building blocks of any system.3.1 ApplicationsThe table below covers the applications for differ-ent modulation formats in both wireless communi-cations and video. Although this note focuses on wireless communica-tions, video applications have also been included in the table for completeness and because of their similarity to other wireless communications.3.1.1 Bit rate and symbol rateTo understand and compare different modulation format efficiencies, it is important to first under-stand the difference between bit rate and symbol rate. The signal bandwidth for the communications channel needed depends on the symbol rate, not on the bit rate.Symbol rate =bit ratethe number of bits transmitted with each symbol 3. Digital Modulation Types and Relative EfficienciesBit rate is the frequency of a system bit stream. Take, for example, a radio with an 8 bit sampler, sampling at 10 kHz for voice. The bit rate, the basic bit stream rate in the radio, would be eight bits multiplied by 10K samples per second, or 80 Kbits per second. (For the moment we will ignore the extra bits required for synchronization, error correction, etc.)Figure 10 is an example of a state diagram of a Quadrature Phase Shift Keying (QPSK) signal. The states can be mapped to zeros and ones. This is a common mapping, but it is not the only one. Any mapping can be used.The symbol rate is the bit rate divided by the num-ber of bits that can be transmitted with each sym-bol. If one bit is transmitted per symbol, as with BPSK, then the symbol rate would be the same as the bit rate of 80 Kbits per second. If two bits are transmitted per symbol, as in QPSK, then the sym-bol rate would be half of the bit rate or 40 Kbits per second. Symbol rate is sometimes called baud rate. Note that baud rate is not the same as bit rate. These terms are often confused. If more bits can be sent with each symbol, then the same amount of data can be sent in a narrower spec-trum. This is why modulation formats that are more complex and use a higher number of states can send the same information over a narrower piece of the RF spectrum.3.1.2 Spectrum (bandwidth) requirementsAn example of how symbol rate influences spec-trum requirements can be seen in eight-state Phase Shift Keying (8PSK). It is a variation of PSK. There are eight possible states that the signal can transi-tion to at any time. The phase of the signal can take any of eight values at any symbol time. Since 23= 8, there are three bits per symbol. This means the symbol rate is one third of the bit rate. This is relatively easy to decode.Figure 10. Bit Rate and Symbol Rate Figure 11. Spectrum Requirements3.1.3 Symbol ClockThe symbol clock represents the frequency and exact timing of the transmission of the individual symbols. At the symbol clock transitions, the trans-mitted carrier is at the correct I/Q(or magnitude/ phase) value to represent a specific symbol (a specific point in the constellation).3.2 Phase Shift KeyingOne of the simplest forms of digital modulation is binary or Bi-Phase Shift Keying (BPSK). One appli-cation where this is used is for deep space teleme-try. The phase of a constant amplitude carrier sig-nal moves between zero and 180 degrees. On an I and Q diagram, the I state has two different values. There are two possible locations in the state dia-gram, so a binary one or zero can be sent. The symbol rate is one bit per symbol.A more common type of phase modulation is Quadrature Phase Shift Keying (QPSK). It is used extensively in applications including CDMA (Code Division Multiple Access) cellular service, wireless local loop, Iridium (a voice/data satellite system) and DVB-S (Digital Video Broadcasting — Satellite). Quadrature means that the signal shifts between phase states which are separated by 90 degrees. The signal shifts in increments of 90 degrees from 45 to 135, –45, or –135 degrees. These points are chosen as they can be easily implemented using an I/Q modulator. Only two I values and two Q values are needed and this gives two bits per symbol. There are four states because 22= 4. It is therefore a more bandwidth-efficient type of modulation than BPSK, potentially twice as efficient.Figure 12. Phase Shift Keying3.3 Frequency Shift KeyingFrequency modulation and phase modulation are closely related. A static frequency shift of +1 Hz means that the phase is constantly advancing at the rate of 360 degrees per second (2 πrad/sec), relative to the phase of the unshifted signal.FSK (Frequency Shift Keying) is used in many applications including cordless and paging sys-tems. Some of the cordless systems include DECT (Digital Enhanced Cordless Telephone) and CT2 (Cordless Telephone 2).In FSK, the frequency of the carrier is changed as a function of the modulating signal (data) being transmitted. Amplitude remains unchanged. In binary FSK (BFSK or 2FSK), a “1” is represented by one frequency and a “0” is represented by another frequency.3.4 Minimum Shift KeyingSince a frequency shift produces an advancing or retarding phase, frequency shifts can be detected by sampling phase at each symbol period. Phase shifts of (2N + 1) π/2radians are easily detected with an I/Q demodulator. At even numbered sym-bols, the polarity of the I channel conveys the transmitted data, while at odd numbered symbols the polarity of the Q channel conveys the data. This orthogonality between I and Q simplifies detection algorithms and hence reduces power con-sumption in a mobile receiver. The minimum fre-quency shift which yields orthogonality of I and Q is that which results in a phase shift of ±π/2radi-ans per symbol (90 degrees per symbol). FSK with this deviation is called MSK (Minimum Shift Keying). The deviation must be accurate in order to generate repeatable 90 degree phase shifts. MSK is used in the GSM (Global System for Mobile Communications) cellular standard. A phase shift of +90 degrees represents a data bit equal to “1,”while –90 degrees represents a “0.” The peak-to-peak frequency shift of an MSK signal is equal to one-half of the bit rate.FSK and MSK produce constant envelope carrier signals, which have no amplitude variations. This is a desirable characteristic for improving the power efficiency of transmitters. Amplitude varia-tions can exercise nonlinearities in an amplifier’s amplitude-transfer function, generating spectral regrowth, a component of adjacent channel power. Therefore, more efficient amplifiers (which tend to be less linear) can be used with constant-envelope signals, reducing power consumption.Figure 13. Frequency Shift KeyingMSK has a narrower spectrum than wider devia-tion forms of FSK. The width of the spectrum is also influenced by the waveforms causing the fre-quency shift. If those waveforms have fast transi-tions or a high slew rate, then the spectrumof the transmitter will be broad. In practice, the waveforms are filtered with a Gaussian filter, resulting in a narrow spectrum. In addition, the Gaussian filter has no time-domain overshoot, which would broaden the spectrum by increasing the peak deviation. MSK with a Gaussian filter is termed GMSK (Gaussian MSK).3.5 Quadrature Amplitude ModulationAnother member of the digital modulation family is Quadrature Amplitude Modulation (QAM). QAM is used in applications including microwave digital radio, DVB-C (Digital Video Broadcasting—Cable), and modems.In 16-state Quadrature Amplitude Modulation (16QAM), there are four I values and four Q values. This results in a total of 16 possible states for the signal. It can transition from any state to any other state at every symbol time. Since 16 = 24, four bits per symbol can be sent. This consists of two bits for I and two bits for Q. The symbol rate is one fourth of the bit rate. So this modulation format produces a more spectrally efficient transmission. It is more efficient than BPSK, QPSK, or 8PSK. Note that QPSK is the same as 4QAM.Another variation is 32QAM. In this case there are six I values and six Q values resulting in a total of 36 possible states (6x6=36). This is too many states for a power of two (the closest power of two is 32). So the four corner symbol states, which take the most power to transmit, are omitted. This reduces the amount of peak power the transmitter has to generate. Since 25= 32, there are five bits per sym-bol and the symbol rate is one fifth of the bit rate. The current practical limits are approximately256QAM, though work is underway to extend the limits to 512 or 1024 QAM. A 256QAM system uses 16 I-values and 16 Q-values, giving 256 possible states. Since 28= 256, each symbol can represent eight bits. A 256QAM signal that can send eight bits per symbol is very spectrally efficient. However, the symbols are very close together and are thus more subject to errors due to noise and distortion. Such a signal may have to be transmit-ted with extra power (to effectively spread the symbols out more) and this reduces power efficiency as compared to simpler schemes.Figure 14. Quadrature Amplitude ModulationCompare the bandwidth efficiency when using256QAM versus BPSK modulation in the radio example in section 3.1.1 (which uses an eight-bit sampler sampling at 10 kHz for voice). BPSK uses80 Ksymbols-per-second sending 1 bit per symbol.A system using 256QAM sends eight bits per sym-bol so the symbol rate would be 10 Ksymbols per second. A 256QAM system enables the same amount of information to be sent as BPSK using only one eighth of the bandwidth. It is eight times more bandwidth efficient. However, there is a tradeoff. The radio becomes more complex and is more susceptible to errors caused by noise and dis-tortion. Error rates of higher-order QAM systems such as this degrade more rapidly than QPSK as noise or interference is introduced. A measureof this degradation would be a higher Bit Error Rate (BER).In any digital modulation system, if the input sig-nal is distorted or severely attenuated the receiver will eventually lose symbol lock completely. If the receiver can no longer recover the symbol clock, it cannot demodulate the signal or recover any infor-mation. With less degradation, the symbol clock can be recovered, but it is noisy, and the symbol locations themselves are noisy. In some cases, a symbol will fall far enough away from its intended position that it will cross over to an adjacent posi-tion. The I and Q level detectors used in the demodulator would misinterpret such a symbol as being in the wrong location, causing bit errors. QPSK is not as efficient, but the states are much farther apart and the system can tolerate a lot more noise before suffering symbol errors. QPSK has no intermediate states between the four corner-symbol locations, so there is less opportunity for the demodulator to misinterpret symbols. QPSK requires less transmitter power than QAM to achieve the same bit error rate.3.6 Theoretical bandwidth efficiency limits Bandwidth efficiency describes how efficiently the allocated bandwidth is utilized or the ability of a modulation scheme to accommodate data, within a limited bandwidth. The table below shows the theoretical bandwidth efficiency limits for the main modulation types. Note that these figures cannot actually be achieved in practical radios since they require perfect modulators, demodula-tors, filter, and transmission paths.If the radio had a perfect (rectangular in the fre-quency domain) filter, then the occupied band-width could be made equal to the symbol rate.Techniques for maximizing spectral efficiency include the following:•Relate the data rate to the frequency shift (as in GSM).•Use premodulation filtering to reduce the occupied bandwidth. Raised cosine filters,as used in NADC, PDC, and PHS, give thebest spectral efficiency.•Restrict the types of transitions.Modulation Theoretical bandwidthformat efficiencylimitsMSK 1bit/second/HzBPSK 1bit/second/HzQPSK 2bits/second/Hz8PSK 3bits/second/Hz16 QAM 4 bits/second/Hz32 QAM 5 bits/second/Hz64 QAM 6 bits/second/Hz256 QAM 8 bits/second/HzEffects of going through the originTake, for example, a QPSK signal where the normalized value changes from 1, 1 to –1, –1. When changing simulta-neously from I and Q values of +1 to I and Q values of –1, the signal trajectory goes through the origin (the I/Q value of 0,0). The origin represents 0 carrier magnitude. A value of 0 magnitude indicates that the carrier amplitude is 0 for a moment.Not all transitions in QPSK result in a trajectory that goes through the origin. If I changes value but Q does not (or vice-versa) the carrier amplitude changes a little, but it does not go through zero. Therefore some symbol transi-tions will result in a small amplitude variation, while others will result in a very large amplitude variation. The clock-recovery circuit in the receiver must deal with this ampli-tude variation uncertainty if it uses amplitude variations to align the receiver clock with the transmitter clock. Spectral regrowth does not automatically result from these trajectories that pass through or near the origin. If the amplifier and associated circuits are perfectly linear, the spectrum (spectral occupancy or occupied bandwidth) will be unchanged. The problem lies in nonlinearities in the circuits.A signal which changes amplitude over a very large range will exercise these nonlinearities to the fullest extent. These nonlinearities will cause distortion products. In con-tinuously modulated systems they will cause “spectral regrowth” or wider modulation sidebands (a phenomenon related to intermodulation distortion). Another term which is sometimes used in this context is “spectral splatter.”However this is a term that is more correctly used in asso-ciation with the increase in the bandwidth of a signal caused by pulsing on and off.3.7 Spectral efficiency examples inpractical radiosThe following examples indicate spectral efficien-cies that are achieved in some practical radio systems.The TDMA version of the North American Digital Cellular (NADC) system, achieves a 48 Kbits-per-second data rate over a 30 kHz bandwidth or 1.6 bits per second per Hz. It is a π/4 DQPSK based system and transmits two bits per symbol. The theoretical efficiency would be two bits per second per Hz and in practice it is 1.6 bits per second per Hz.Another example is a microwave digital radio using 16QAM. This kind of signal is more susceptible to noise and distortion than something simpler such as QPSK. This type of signal is usually sent over a direct line-of-sight microwave link or over a wire where there is very little noise and interference. In this microwave-digital-radio example the bit rate is 140 Mbits per second over a very wide bandwidth of 52.5 MHz. The spectral efficiency is 2.7 bits per second per Hz. To implement this, it takes a very clear line-of-sight transmission path and a precise and optimized high-power transceiver.。
数字调制解调技术-外文翻译
数字调制解调技术英文文献Technology of digital modulation and demodulation plays a important role in digital communication system, the combination of digital communication technology and FPGA is a certainly trend . With the development of software radio, the requirement for technology of modulation and demodulation is higher and higher. This paper starts with studying digital modulation and demodulation theory at first, and analyses basic principle of three kinds of important modulation and demodulation way ( FSK, MSK, GMSK ).The Rohde &Schwarz SME03, Signal Generator, provides AM modulation and External FSK digital modulation required for the development and testing of digital mobile radio receivers.The application of PWM in digital modulation and demodulation for analog communication signals in several modulation modes Research results prove that the design of digital IF modulator and demodulator of Software Radio appeases the capability and requirement of Software Radio.A transfusion speed monitor system is designed based on infrared technology with modulation and demodulation.It's the combination of modulator and demodulator.Time synchronization that is key technology of digital demodulation is cc allied by software.The paper provides the design of hardware of digital IF modulator and demodulator of Software Radio which includes Digital Signal Processor、Micro Control Unit and AD/DA convertor etc.Digital Down/Up Converter(DDC/DUC), modulation and demodulation are discussed in the dissertation as some essencial parts of SDR platform. Two Way Automatic Communication System(TWACS) is a new valuable communication technology for distribution networks,which has special of modulation and demodulation. In this paper, we study the OFDM technology based on 802.16a, realize the baseband modulation and demodulation by using TMS320C6201, and optimize the software module. The paper introduces the principle of QPSK modulation and demodulation, the circuit are also be realized based on FPGA.With the improvement of the technology, especially in the fields such as computer technology , data coding and compress , digital modulation and VLSI, the world electronic information industry enter into the digital era. First, the features of fax communication and the mode of modulation and demodulation are described.In automatic classification of digital modulation signal,computing envelope variance after difference has important meaning to distinguish PSK and FSK signal.The science and technology of space flight.The effect on modulation and demodulation of QPSK via carrier phase noise can not be ignored, and it is difficult to analyze.Digital modulation error parameters, such as error vector magnitude EVM, is important in test and measurement of information system. This paper introduces the technology research progress in the metrology of digital modulation error parameters. First, we point out the basic problems existing in the field, which is about traceability and parameter range of calibration, and describe the relevant research, such as the thinking and technology of the `RF waveform metrology'. Then, we highlight the research progress of our team: 1). The metrology method and system for digital demodulation error parameter based on CW combination, which fits BPSK, QPSK, 8PSK, 16QAM, 64QAM modulation: this method can achieve traceability and error setting ability in a wide range, when standard EvmRms is 1.585%, the expanded uncertainty (k=2) is 0.009%. 2). The metrology method and system for digitaldemodulation error parameter based on analog AM or PM. 3). The metrology method and system for digital demodulation error parameter based on IQ gain imbalance and phase imbalance. 4). The metrology method and system for digital demodulation error parameter based on analog PM in the aspect of GMSK and FSK modulation. 5). The metrology method and system for digital demodulation error parameter based on Baseband waveform design. Based on these methods, our proposal are given as follows: first, establish public metrology standard for digital modulation error parameters; second, develop a new type of instrument "vector signal analyzer calibrator".In this paper, we propose a novel method of chaotic modulation based on the combination of Chaotic Pulse Position Modulation (CPPM) and Chaotic Pulse WidthModulation (CPWM). This combination looks very promising for the improvement of information privacy in chaos-based digital communications. In the CPPM+CPWM method, each pulse is a chaotic symbol which carries binary information of two bits corresponding to its position and width, where the position is determined by the interval between rising edge of the current pulse compared to the previous one and the width is determined by the duration between the rising edge and the falling edge of the same pulse. This offers the increase of bit rate, bandwidth efficiency and privacy in comparison with the method of CPPM. The schemes of Modulation andDemodulation (MoDem) of CPPM+CPWM are proposed, designed and analyzed that based on the conventional schemes of CPPM. The numerical simulation in time domain of the system of CPPM+CPWM MoDem is implemented in Matlab/Simulink. It gives a summary of theoretical and practical studies on the properties of pulse-phase modulation, developed mainly in 1943. The properties of pulse-phasemodulation are studied by means of Fourier transformations. Although some approximations are introduced, the calculations lead to the following definite conclusions: (1) Pulse-phase modulation introduces no amplitude distortion except at sub-multiples of the recurrent frequency. (2) The harmonic distortion, if any, is negligible and this method of modulation can be used for high-quality broadcasting.(3) Pulse-phase modulation is subject to a special type of distortion called ?cross-distortion,? produced by side bands of the recurrent frequency appearing in the signal bandwidth. Curves of the approximate amount of this type of distortion are given, and it is shown that, in practical multi-channel systems, this distortion isnegligible, provided that the recurrent pulse frequency is at least double the highest signal frequency to be transmitted, and preferably equal to, or greater than, three times this frequency. This study is followed by considerations on the signal/noise ratio in pulse-phase modulation. Pulse-phasemodulation is compared with amplitude modulation and a formula, giving the improvement in the signal/noise ratio due to pulse-phase modulation, is established by very simple considerations. It is shown that this ratio improves as the frequency bandwidth used in pulse-phase modulation. It is shown how an improvement of 3 db in signal/noise ratio can be obtained by suppressing the noise on the synchronizing pulse, and a practical circuit developed and applied in 1943 by the author is described. Finally, a typical example of pulse technique is given. In practical circuits the modulator and demodulator pulses are not perfectly shaped, because of the departure from linearity due to finite time-constants. This introduces harmonic distortion. It is shown how this distortion can be practically elimi- nated by designing circuits so that the time constant is equal at modulationand demodulation.It present a novel technique for digital data modulation and demodulationcalled triangular modulation (TM). The modulation technique was developed primarily to maximize the amount of data sent over a limited bandwidth channel while still maintaining very good noise rejection and signal distortion performance. Themodulation technique involves breaking digital data into a series of parallel words. Each word is then represented by one half period of a triangular waveform whose slope is proportional to the value of the parallel word it represents. Thedemodulation technique for this uniquely defined waveform involves first digitizing the waveform at a higher constant sampling rate. A linear regression algorithm using the method of least squares is then used to compute the slope of the digitized waveform to a very high precision. This process is repeated for each rising and falling edge of the triangular modulated waveform. All encoded data is extracted by precise slope computation since each slope uniquely defines the encoded data word it represents. The ability of the demodulation algorithm to compute the exact slope of the modulated waveform determines how many bits can be represented by the modulated waveform. Transmission channel bandwidth limitations determine the allowable range of slopes used. Several simulations are performed to provide a sample of how the modulation method will perform in various real world environments. The paper also discusses several application areas where themodulation technique will provide superior results over other modulation methods.The theory of constant envelope orthogonal frequency division multiplexing (CE-OFDM) is analyzed in this paper, along with the introduction of the implementation method of CE-OFDM technique. Besides, the modulation and demodulation process is simulated and analyzed. And the results indicate that CE-OFDM conducts phasemodulation on the basis of OFDM modulation. Thus, FFT/IFFT is implemented in the transmitting and receiving terminals. Furthermore, the method of equalization applied in the demodulation process can optimize system performance. And also, CE-OFDM solves the problem of high peak-to-average power ratio (PAPR) in OFDM, reducing PAPR to 0Db.High efficient modulation technology is a hot research topic. UNB modulation, for its good performance, is paid to more attention. First, the article introduces EBPSK modulation scheme as UNB modulation method, gives its time and frequency domain characteristics and presents its optimized form in the same time, which can lower the sideband power level, while keeping the modulation information un-lost. Then, filter design is discussed about two zero and two pole digital filter, which shows narrower bandwidths and a fast response speed to the EBPSK based UNB modulated signals, although the filter bandwidth is much narrower, the modulationinformation still can be seen after the modulated signals filtered using it. Last, simulation is done about EBPSK based UNB modulation and demodulation, and experimental results show that EBPSK based UNB modulation has high bandwidth efficiency and a good, even better BER performance using the filters.中文译文数字调制解调技术在数字通信中占有非常重要的地位,数字通信技术与FPGA 的结合是现代通信系统发展的一个必然趋势。
数字调制系统仿真实验
实验二 数字调制系统仿真实验一、实验目的1、掌握ASK ，PSK （DPSK ）和多进制数字键控等数字调制技术的原理2、掌握数字调制系统仿真的方法 二、实验内容1、设计一个数字调制系统2、编写一个带有拓扑排序功能的有向无环图 三、基本原理当调制信号位二进制数字信号时，这种调制称为二进制数字调制。
在二进制数字调制中，载波的幅度、频率或相位只有两种变化状态，常用的二进制数字调制方式有以下几种：二进制振幅键控调制（2ASK ）、二进制频移键控（2FSK ）、二进制移相键控（2PSK ）和二进制相对（或差分）相位键控（2DPSK ）。
1、 二进制振幅键控（2ASK ）1) 调制方法2ASK 信号可表示为：式中，g(t)是持续时间为Ts 的矩形脉冲，即：产生2ASK 的方法有两种，如图所示。
. s (t )2ASK 调制原理框图cosωc t乘法器 载波 cosωc te 0(t )s (t )e 0(t )开关电路相应的调制输出如下图所示：⎩⎨⎧≤=tT t t g s 其它02/1)(⎩⎨⎧-=出现以概率出现以概率P P a n 110t nT t g a t t s t e cn s n c ωωcos ])([cos )()(0∑-==1) 2ASK 信号的解调相干解调法：.cos ωc t相干解调法相乘器 定时脉冲输入输出 带通 滤波器低通 滤波器抽样 判决器包络检波法.包络检波法带通 滤波器半波或 全波整流定时脉冲低通 滤波器 抽样 判决器 输入输出2、 二进制频移键控（2FSK ）1) 调制方法2FSK 信号可表示为：)cos(])([)cos(])([)cos()()cos()()(2112110n ns n n ns n n n t nT t g a t nT t g a t t s t t s t e θωϕωθωϕω+-++-=+++=∑∑式中，g(t)是持续时间为Ts 的矩形脉冲，即：⎩⎨⎧≤=tT t t g s 其它02/1)(产生2FSK 的方法有两种，如图所示。
基于matlab仿真的数字调制与解调设计本科毕业设计(论文)
基于matlab仿真的数字调制与解调设计本科毕业设计（论文）摘要数字调制是通信系统中最为重要的环节之一，数字调制技术的改进也是通信系统性能提高的重要途径。
本文首先分析了数字调制系统的几种基本调制解调方法，然后，运用Matlab设计了这几种数字调制解调方法的仿真程序，主要包括PSK，DPSK和16QAM。
通过仿真，分析了这三种调制解调过程中各环节时域和频域的波形，并考虑了信道噪声的影响。
通过仿真更深刻地理解了数字调制解调系统基本原理。
最后，对三种调制解调系统的性能进行了比较。
关键词：数字调制；分析与仿真；Matlab。
AbstractDigital modulation is one of the most important part in communication system, and the improvement of digital modulation technology is an important way for the improvement of communication system capability. In this paper, some usual methods of digital modulation are introduced firstly. Then their simulation programs are built by using MATLAB, they mainly includePSK,DPSK,16QAM. Through simulation, we analyzed the time and frequency waveform for every part of these three modulations, and also consider theeffect of the channel noise. Through the simulation, we understand the basic theory of modulation and demodulation more clearly. At last, the capability of these digital modulations have been compared.Keywords: Digital modulation; analysis; simulation; MATLAB.目录第一章引言……………………………………..………………………1 1.1 研究背景……………………………………..……………………1 1.2 通信的发展现状和趋势………………………………………1 1.3 研究目的与意义…………………………………………………2 1.4 本文内容安排……………………………………………………2 第二章数字调制解调相关原理……………………………………3 2.1 二进制相移键控(2PSK) ………………………………………3 2.2 二进制差分相移键控(2DPSK)…………………………………5 2.3 正交振幅调制(QAM) ................................................8 第三章数字调制解调仿真...................................................10 3.1 2PSK调制和解调仿真.........................................................10 3.2 2DPSK调制和解调仿真.........................................................14 3.3 16QAM调制和解调仿真.........................................................18 3.4 各种调制比较............................................................24 第四章结束语..................................................................25 参考文献..............................................................................26 致谢....................................................................................27 附录 (28)第一章引言1.1 研究背景随着通信系统复杂性的增加，传统的手工分析与电路板试验等分析设计方法已经不能适应发展的需要，通信系统计算机模拟仿真技术日益显示出其巨大的优越性。
通信系统中数字调制技术的研究与仿真本科生毕业论文
通信系统中数字调制技术的研究与仿真摘要在日常的生活中，通信是人们用来传递信息的方式。
随着数字系统的飞速发展，对数字系统的性能和调制解调技术要求也越来越高。
同时，由于计算技术的发展，通信系统的仿真已日益普遍，已逐渐成为今天设计和分析通信系统的主要工具。
本次设计将使用MATLAB软件设计函数和Simulink建模对数字调相技术进行仿真和研究。
本文在第一章中介绍了通信系统的组成、MATLAB的使用以及Simulink模块的组建。
第二章深入分析了2ASK、2PSK、2FSK的调制解调原理理论知识，熟悉了原理后，在第三章中用MATLAB编程和Simulink对它们进行仿真和研究。
本设计主要实现2ASK、2PSK、2FSK调制解调过程的仿真，并分析它们的性能差异。
最后一章对数字调制与解调作了一个总结。
关键词：MATLAB调制解调2ASK 2PSK 2FSKResearch and Simulation of Digital Modulation Technology in Communication SystemMajor: communication engineeringStudent: Qin Kai Supervisor: Tang QuanAbstractIn day-to-day life，communication is used to convey information. With the rapid development of digital systems，digital system for modem performance and the technical requirements of increasingly high.At the same time，the development of computing technology，simulation of communication systems have become increasingly common，have gradually become the design and analysis of today's main tool for communication systems.In chapter 1, this paper introduces the composition of the communication system, the use of MATLAB and Simulink module is established. The second chapter in-depth analysis of the 2 ASK, 2 PSK, 2 FSK of demodulation principle theory knowledge, be familiar with the theory, in the third chapter using MATLAB programming and Simulink simulation and research on them. This design mainly realizes 2 ASK, 2 PSK, 2 FSK demodulation process Simulink, and analyzes the performance of their differences. The last chapter of digital modulation and demodulation made a summary.Key words：MATLAB modem 2ASK 2PSK 2FSK毕业设计（论文）原创性声明和使用授权说明原创性声明本人郑重承诺：所呈交的毕业设计（论文），是我个人在指导教师的指导下进行的研究工作及取得的成果。
MSK调制解调系统设计的外文翻译
一：中文译文调频宽频传输MX589 GMSK调制移键控(msk)0.3 GMSK 1200作者Fred Kostedt, Engineer for MX-COM.James C. Kemerling, Engineer for MX-COM.引言随着计算机的普及，数据传输在当今社会的需求也在不断的增加，进而出现了传输数据的无线链路。
二进制数据组成的“一零”和“零一”的过渡，产生了丰富的谐波频谱内容，这并不适合射频传输。
因此，数字调制领域已得到了蓬勃的发展。
从最近的标准可以看到，如蜂窝数字分组数据（CDPD）和Mobitex*指定高斯滤波最小频移键控（GMSK）的调制方法就是数字调制领域的先进技术。
GMSK是一种简单而有效的数字调制的无线数据传输办法。
为了使我们更好的理解GMSK数字调制的方法，我们会分析MSK和GMSK的基本理论知识，以及如何运用GMSK数字调制方法来实现CDPD和Mobitex系统。
GMSK调制解调器降低了系统的复杂性，从而降低系统的成本。
但是，也有一些重要的实施细则需要加以考虑。
本文将涵盖其中的一些细节，把重点放在“典型”调频收音机拓扑接口的单芯片中的基带调制解调器的中频/射频部分。
背景如果我们看一下傅里叶级数展开的一个数据信号，我们可以看到谐波延伸到无穷远。
当这些谐波被总结，他们给它的数据信号急剧转变。
因此，一个过滤了的NRZ数据流用来调制射频载波将产生相当大的RF频谱的带宽。
当然，有严格的FCC的法规频谱和这种制度，这种使用情况通常被认为是不切实际的。
但是，如果我们在开始就移除高次谐波的傅立叶级数（即让数据信号通过一个低通滤器），其过程中的数据将逐步急剧减少。
这表明，premodulation过滤是一种在无线数据传输过程中减少被占领的频谱的很有效的方法。
除了紧凑的频谱，无线数据调制方案必须在有噪音的情况下能获得良好的误码率（BER）性能。
其性能也应该是线性的独立的功率放大设备从而允许使用C类功率放大器。
数字调制技术的仿真实现及性能研究
课程设计说明书题目数字调制技术的仿真实现及性能研究系(部) 电子与通信工程系专业(班级)姓名学号指导教师起止日期2012.12.10--2012.12.14长沙学院课程设计鉴定表姓名学号专业班级指导设计题目数字调制技术的仿真实现及性能研究教师指导教师意见：评定等级：教师签名：日期：答辩小组意见：评定等级：答辩小组长签名：日期：教研室意见：教研室主任签名：日期：系（部）意见：系主任签名：日期：说明课程设计成绩分“优秀”、“良好”、“及格”、“不及格”四类；目录1.2 课程设计原理 (2)1.2 数字调制仿真 (2)1.2.1 、仿真流程图 (3)1.2.2、2ASK仿真及结果分析 (4)源程序 (4)仿真结果 (5)1.2.3 、2FSK仿真及结果分析 (6)源程序 (6)仿真结果 (8)1.2.4、2PSK仿真及结果分析 (9)源程序 (9)仿真结果 (12)1.3误码率的分析 (13)1.3.1、源程序 (13)1.3.2、仿真结果 (14)1.4 课程设计总结及心得体会 (14)1.1 课程设计任务1.根据题目，查阅有关资料，掌握数字带通调制技术。
2.设计数字带通调制系统，画出系统仿真流程图。
3.学习MATLAB软件，掌握MATLAB各种函数的使用。
4.根据数字带通调制原理，运用MATLAB进行编程，仿真调制过程，记录并分析仿真结果。
结果图中必须包含：基带信号图、已调信号波形图、已调信号频谱图、解调恢复的基带信号图、误码率信噪比分析图。
其中误码率统计必须根据实际信号进行统计，不能使用误码率公式直接绘图。
6.形成设计报告。
1.2 课程设计原理二进制数字调制技术原理数字信号的传输方式分为基带传输和带通传输，在实际应用中，大多数信道具有带通特性而不能直接传输基带信号。
为了使数字信号在带通信道中传输，必须使用数字基带信号对载波进行调制，以使信号与信道的特性相匹配。
这种用数字基带信号控制载波，把数字基带信号变换为数字带通信号的过程称为数字调制。
基于maltlabam调制与解调
基于maltlab的调制与解调1. 简介Maltlab是一种非常常用的数学软件，广泛应用于科学和工程领域。
Maltlab在通信系统中也有着重要的作用，尤其在调制与解调的模拟和仿真方面。
调制与解调是数字通信中的重要环节，它涉及到信号的编码和解码，对于数字通信系统的设计和优化有着重要的意义。
2. 调制调制是指将数字信号转换成模拟信号的过程。
在数字通信系统中，常用的调制方式有振幅调制（AM）、频率调制（FM）和相位调制（PM）等。
在Maltlab中，可以通过信号处理工具箱来进行调制过程的模拟和仿真。
需要生成一个数字信号，然后通过调制器将其转换成模拟信号。
可以通过Maltlab提供的函数来实现不同调制方式的模拟。
3. 解调解调是指将模拟信号转换成数字信号的过程。
在数字通信系统中，常用的解调方式和调制方式对应。
用于解调AM信号的方法就是通过包络检波器。
Maltlab同样可以通过信号处理工具箱来进行解调过程的模拟和仿真。
可以通过使用不同的解调算法来实现解调过程，并通过Maltlab的仿真工具来观察解调后的数字信号。
4. 应用调制与解调是数字通信系统中的核心环节，对于系统的性能和稳定性有着重要的影响。
通过Maltlab可以方便地对调制与解调过程进行模拟和仿真，帮助工程师更好地理解系统的工作原理，并对系统进行优化和改进。
Maltlab在通信系统中的应用已经非常广泛，通过Maltlab可以方便地对调制与解调的过程进行数学建模和仿真，帮助工程师进行系统设计和性能评估。
总结Maltlab作为一种强大的数学软件，在通信系统中有着广泛的应用。
调制与解调作为数字通信系统中的重要环节，对系统的性能和稳定性有着重要的影响。
通过Maltlab可以方便地进行调制与解调过程的模拟和仿真，帮助工程师更好地理解系统的工作原理，并对系统进行优化和改进。
通过Maltlab的支持，调制与解调的研究和应用将会得到更好的推动，为数字通信系统的发展和改进提供更好的技术支持。
数字调制系统的设计与仿真
摘要数字调制是通信系统中最为重要的环节之一，数字调制技术的改进也是通信系统性能提高的重要途径。
本文首先分析了数字调制系统的2PSK和2DPSK的调制解调方法，然后，运用MATLAB设计了这两种数字调制解调方法的仿真程序。
通过仿真，分析了在这两种调制解调过程中各环节时域和频域的波形，并考虑了信道噪声的影响。
关键词：2PSK 2DPSK MATLAB 设计与仿真前言本次课程设计首先分析了数字调制系统的几种基本调制解调方法，然后，运用MATLAB 设计了两种数字调制解调方法的仿真程序，主要包括2PSK，2DPSK。
通过仿真，分析了这两种调制解调过程中各环节时域和频域的波形，并考虑了信道噪声的影响。
通过仿真更深刻地理解了数字调制解调系统基本原理。
最后，对这两种调制解调系统的性能进行了比较。
数字信号的传输可分为基带传输和带通传输，实际中的大多数的信道（如无线信道）因具有带通特性而不能直接传送基带信号，这是因为基带信号往往具有丰富的低频分量，为了使数字信号能在带通信道中传输，必须用数字基带信号对载波进行调制，以使信号与信道相匹配，这种用基带信号控制载波，把数字基带信号变换成数字带通信号的过程称为数字调制。
由于传输失真、传输损耗以及保证带内特性的原因，基带信号不适合在各种信道上进行长距离传输。
为了进行长途传输，必须对数字信号进行载波调制，将信号频谱搬移到高频处才能在信道中传输。
因此，大部分现代通信系统都使用数字调制技术。
在仿真中用到的MATLAB软件是集科学计算(computation) 、可视化(visualization)、编程(programming)于一身，并提供了丰富的Windows图形界面一种设计方法软件。
目录第一章设计目标 (1)1.1设计目标 (1)1.2设计任务及要求 (1)1.3设计内容 (1)1.4设计思路及步骤 (2)1.4.1设计思路 (2)1.4.2设计步骤 (2)1.5设计意义 (3)第二章二进制相移键控信号 (3)2.1二进制相移键控(2PSK) (3)2.2二进制差分相移键控(2DPSK) (7)第三章二进制相移仿真及结果分析 (8)3.1MATLAB简介 (8)3.2数字调制与解调原理 (8)3.3 程序流图设计 (9)3.4仿真中用到的MATLAB函数 (10)3.5仿真参数设计 (10)3.6仿真结果分析 (11)3.7 误码率与信噪比的关系分析 (12)总结 (14)参考文献 (15)附录 (16)第一章设计目标1.1设计目标：本次通信系统综合训练的目的是让学生在掌握通信系统基本原理和仿真软件的基础上，对通信系统中的数字调制系统进行设计和仿真。
fsk调制实验报告
fsk调制实验报告FSK调制实验报告引言：FSK调制（Frequency Shift Keying）是一种常见的数字调制技术，它通过改变载波频率的方式来传输数字信号。
本实验旨在通过搭建一个简单的FSK调制系统，探索其原理和应用。
一、实验目的本实验的目的是通过搭建FSK调制系统，了解FSK调制的原理和实现过程，并观察调制后的信号特性。
二、实验材料和方法1. 实验材料：- 信号发生器- 两个可调频率的载波信号源- 一台示波器- 连接线等实验器材2. 实验步骤：1) 将信号发生器连接到示波器，设置信号发生器的输出频率为f1和f2，分别代表二进制数字0和1。
2) 将两个可调频率的载波信号源连接到信号发生器的输出端，分别作为0和1的载波信号。
3) 将两个载波信号输入到一个开关电路中，以控制两个载波信号的切换。
4) 将开关电路的输出信号连接到示波器，观察调制后的信号波形。
三、实验结果与分析在实验中，我们设置了信号发生器的输出频率为f1和f2，分别代表二进制数字0和1。
通过控制开关电路，我们可以将这两个频率的载波信号进行切换，从而实现FSK调制。
在示波器上观察到的波形图显示，当切换到f1时，信号呈现一种特定的频率和振幅；而当切换到f2时，信号的频率和振幅发生变化。
这种频率和振幅的变化正是FSK调制的特征，通过这种方式，我们可以将数字信号转换为调制后的模拟信号进行传输。
通过实验观察，我们可以发现FSK调制的一个重要特点是频谱分离。
由于不同数字对应的频率不同，调制后的信号在频域上呈现出不同的频谱分布。
这种频谱分离的特性使得FSK调制在抗干扰性方面表现出色，能够有效地抵抗信道噪声和其他干扰。
四、实验应用FSK调制在通信领域有广泛的应用。
其中，最常见的应用之一是在调制解调器中，用于数字通信系统中的调制和解调过程。
通过使用FSK调制技术，数字信号可以被转换为模拟信号进行传输，从而实现了数字通信的可靠传输。
此外，FSK调制还被广泛应用于无线传感器网络、远程遥控和遥感等领域。
数字滤波器的仿真与实现_中英文翻译
英文原文The simulation and the realization of the digital filterWith the information age and the advent of the digital world, digital signal processing has become one of today's most important disciplines and door technology. Digital signal processing in communications, voice, images, automatic control, radar, military, aerospace, medical and household appliances, and many other fields widely applied. In the digital signal processing applications, the digital filter is important and has been widely applied.1、figures Unit on :Analog and digital filtersIn signal processing, the function of a filter is to remove unwanted parts of the signal, such as random noise, or to extract useful parts of the signal, such as the components lying within a certain frequency range.The following block diagram illustrates the basic idea.There are two main kinds of filter, analog and digital. They are quite different in their physical makeup and in how they work. An analog filter uses analog electronic circuits made up from components such as resistors, capacitors and op amps to produce the required filtering effect. Such filter circuits are widely used in such applications as noise reduction, video signal enhancement, graphic equalisers in hi-fi systems, and many other areas. There are well-established standard techniques for designing an analog filter circuit for a given requirement. At all stages, the signal being filtered is an electrical voltage or current which is the direct analogue of the physical quantity (e.g. a sound or video signal or transducer output) involved. A digital filter uses a digital processor to perform numerical calculations on sampled values of the signal. The processor may be a general-purpose computer such as a PC, or a specialised DSP (Digital Signal Processor) chip. The analog input signal must first be sampled and digitised using an ADC (analog to digital converter). The resulting binary numbers, representing successive sampled values of the input signal, are transferred to the processor,which carries out numerical calculations on them. These calculations typically involve multiplying the input values by constants and adding the products together. If necessary, the results of these calculations, which now represent sampled values of the filtered signal, are output through a DAC (digital to analog converter) to convert the signal back to analog form.Note that in a digital filter, the signal is represented by a sequence of numbers, rather than a voltage or current.The following diagram shows the basic setup of such a system.Unit refers to the input signals used to filter hardware or software. If the filter input, output signals are separated, they are bound to respond to the impact of the Unit is separated, such as digital filters filter definition. Digital filter function, which was to import sequences X transformation into export operations through a series Y.According to figures filter function 24-hour live response characteristics, digital filters can be divided into two, namely, unlimited long live long live the corresponding IIR filter and the limited response to FIR filters. IIR filters have the advantage of the digital filter design can use simulation results, and simulation filter design of a large number of tables may facilitate simple. It is the shortcomings of the nonlinear phase; Linear phase if required, will use the entire network phase-correction. Image processing and transmission of data collection is required with linear phase filters identity. And FIR linear phase digital filter to achieve, but an arbitrary margin characteristics. Impact from the digital filter response of the units can be divided into two broad categories : the impact of the limited response (FIR) filters, and unlimited number of shocks to (IIR) digital filters.FIR filters can be strictly linear phase, but because the system FIR filter function extremity fixed at the original point, it can only use the higher number of bands to achieve their high selectivity for the same filter design indicators FIR filter called band than a few high-IIR 5-10 times, the cost is higher, Signal delay is also larger. But if the same linear phase, IIR filters must be network-wide calibration phase, the same section also increase the number of filters and network complexity. FIR filters can be used to achieve non-Digui way, not in a limited precision of a shock, and into the homes and quantitative factors of uncertainty arising from the impact of errors than IIR filter small number, and FIR filter can be used FFT algorithms, the computational speed. But unlike IIR filter can filter through the simulation results, there is no ready-made formula FIR filter must use computer-aided design software (such as MATLAB) to calculate. So, a broader application of FIR filters, and IIR filters are not very strict requirements on occasions.Unit from sub-functions can be divided into the following four categories :(1) Low-filter (LPF);(2) high-filter (HPF);(3) belt-filter (BPF);(4) to prevent filter (BSF).The following chart dotted line for the ideals of the filter frequency characteristics :A1(f) A2(f)10 f2cf 0 f2cf(a) (b)A3(f) A4(f)0 f1c f2cf 0 f1cf2cf(c) (d)(a)LPF (b)HPF (c)BPF (d)BSF2、MATLAB introducedMATLAB is a matrix laboratory (Matrix Laboratory) is intended. In addition to an excellent value calculation capability, it also provides professional symbols terms, word processing, visualization modeling, simulation and real-time control functions. MATLAB as the world's top mathematical software applications, with a strong engineering computing, algorithms research, engineering drawings, applications development, data analysis and dynamic simulation, and other functions, in aerospace, mechanical manufacturing and construction fields playing an increasingly important role. And the C language function rich, the use of flexibility, high-efficiency goals procedures. High language both advantages as well as low level language features. Therefore, C language is the most widely used programming language. Although MATLAB is a complete, fully functional programming environment, but in some cases, data and procedures with the external environment of the world is very necessary and useful. Filter design using Matlab, could be adjusted with the design requirements and filter characteristics of the parameters, visual simple, greatly reducing the workload for the filter design optimization.In the electricity system protection and secondary computer control, many signal processing and analysis are based on are certain types Yeroskipou and the second harmonics of the system voltage and current signals (especially at D process), are mixed with a variety of complex components, the filter has been installed power system during the critical components. Current computer protection and the introduction of two digital signal processing software main filter. Digital filter design using traditional cumbersome formula, the need to change the parameters after recalculation, especially in high filters, filter design workload. Uses MATLAB signal processing boxes can achieve rapid and effective digital filter design and simulation.MATLAB is the basic unit of data matrix, with its directives Biaodashi mathematics, engineering, commonly used form is very similar, it is used to solve a problem than in MATLAB C, Fortran and other languages End precision much the same thing. The popular MATLAB 5.3/Simulink3.0 including hundreds of internal function with the main pack and 30types of tool kits (Toolbox). kits can be divided into functional tool kits and disciplines toolkit. MATLAB tool kit used to expand the functional symbols terms, visualization simulation modelling, word processing and real-time control functions. professional disciplines toolkit is a stronger tool kits, tool kits control, signal processing tool kit, tool kits, etc. belonging to such communicationsMATLAB users to open widely welcomed. In addition to the internal function, all the packages MATLAB tool kits are readable document and the document could be amended, modified or users through Yuanchengxu the construction of new procedures to prepare themselves for kits.3、Digital filter designDigital filter design of the basic requirementsDigital filter design must go through three steps :(1) Identification of indicators : In the design of a filter, there must be some indicators. These indicators should be determined on the basis of the application. In many practical applications, digital filters are often used to achieve the frequency operation. Therefore, indicators in the form of general jurisdiction given frequency range and phase response. Margins key indicators given in two ways. The first is absolute indicators. It provides a function to respond to the demands of the general application of FIR filter design. The second indicator is the relative indicators. Its value in the form of answers to decibels. In engineering practice, the most popular of such indicators. For phase response indicators forms, usually in the hope that the system with a linear phase frequency bands human. Using linear phase filter design with the following response to the indicators strengths：①it only contains a few algorithms, no plural operations；②there is delay distortion, only a fixed amount of delay; ③the filter length N (number of bands for N-1), the volume calculation for N/2 magnitude.(2) Model approach : Once identified indicators can use a previous study of the basic principles and relationships, a filter model to be closer to the target system.(3) Achieved : the results of the above two filters, usually by differential equations, system function or pulse response to describe. According to this description of hardware or software used to achieve it.4、Introduced FPGAProgrammable logic device is a generic logic can use a variety of chips, which is to achieve ASIC ASIC (Application Specific Integrated Circuit) semi-customized device, Its emergence and development of electronic systems designers use CAD tools to design their own laboratory in the ASIC device. Especially FPGA (Field Programmable Gate Array) generated and development, as a microprocessor, memory, the figures for electronic system design and set a new industry standard (that is based on standard product sales catalogue in the market to buy). Is a digital system for microprocessors, memories, FPGA or three standard building blocks constitute their integration direction.Digital circuit design using FPGA devices, can not only simplify the design process and can reduce the size and cost of the entire system, increasing system reliability. They do not need to spend the traditional sense a lot of time and effort required to create integrated circuits, to avoid the investment risk and become the fastest-growing industries of electronic devices group. Digital circuit design system FPGA devices using the following main advantages(1)Design flexibleUse FPGA devices may not in the standard series device logic functional limitations. And changes in system design and the use of logic in any one stage of the process, and only through the use of re-programming the FPGA device can be completed, the system design provides for great flexibility.(2) Increased functional densityFunctional density in a given space refers to the number of functional integration logic. Programmable logic chip components doors several high, a FPGA can replace several films, film scores or even hundreds of small-scale digital IC chip illustrated in the film. FPGA devices using the chip to use digital systems in small numbers, thus reducing the number of chips used to reduce the number of printed size and printed, and will ultimately lead to a reduction in the overall size of the system.(3) Improve reliabilityPrinting plates and reduce the number of chips, not only can reduce system size, but it greatly enhanced system reliability. A higher degree of integration than systems in many low-standard integration components for the design of the same system, with much higher reliability. FPGA device used to reduce the number of chips required to achieve the system in the number printed on the cord and joints are reduced, the reliability of the system can beimproved.(4) Shortening the design cycleAs FPGA devices and the programmable flexibility, use it to design a system for longer than traditional methods greatly shortened. FPGA device master degrees high, use printed circuit layout wiring simple. At the same time, success in the prototype design, the development of advanced tools, a high degree of automation, their logic is very simple changes quickly. Therefore, the use of FPGA devices can significantly shorten the design cycle system, and speed up the pace of product into the market, improving product competitiveness.(5) Work fastFPGA/CPLD devices work fast, generally can reach several original Hertz, far larger than the DSP device. At the same time, the use of FPGA devices, the system needed to achieve circuitclasses and small, and thus the pace of work of the entire system will be improved.(6) Increased system performance confidentialityMany FPGA devices have encryption functions in the system widely used FPGA devices can effectively prevent illegal copying products were others(7) To reduce costsFPGA device used to achieve digital system design, if only device itself into the price, sometimes you would not know it advantages, but there are many factors affecting the cost of the system, taken together, the cost advantages of using FPGA is obvious. First, the use of FPGA devices designed to facilitate change, shorten design cycles, reduce development costs for system development; Secondly, the size and FPGA devices allow automation needs plug-ins, reducing the manufacturing system to lower costs; Again, the use of FPGA devices can enhance system reliability, reduced maintenance workload, thereby lowering the cost of maintenance services for the system. In short, the use of FPGA devices for system design to save costs.FPGA design principles :FPGA design an important guiding principles : the balance and size and speed of exchange, the principles behind the design of the filter expression of a large number of certification.Here, "area" means a design exertion FPGA/CPLD logic resources of the FPGA can be used to the typical consumption (FF) and the search table (IUT) to measure more general measure can be used to design logic equivalence occupied by the door is measured. "pace"means stability operations in the chip design can achieve the highest frequency, the frequency of the time series design situation, and design to meet the clock cycle -- PADto pad, Clock Setup Time, Clock Hold Beijing, Clock-to-Output Delay, and other characteristics of many time series closely related. Area (area) and speed (speed) runs through the two targets FPGA design always is the ultimate design quality evaluation criteria. On the size and speed of the two basic concepts : balance of size and speed and size and speed of swap.One pair of size and speed is the unity of opposites contradictions body. Requirements for the design of a design while the smallest, highest frequency of operation is unrealistic. More scientific goal should be to meet the design requirements of the design time series (includes requirements for the design frequency) premise, the smallest chip area occupied. Or in the specified area, the design time series cushion greater frequency run higher. This fully embodies the goals of both size and speed balanced thinking. On the size and speed requirements should not be simply interpreted as raising the level and design engineers perfect sexual pursuit, and should recognize that they are products and the quality and cost of direct relevance. If time series cushion larger design, running relatively high frequency, that the design Jianzhuangxing stronger, more quality assurance system as a whole; On the other hand, the smaller size of consumption design is meant to achieve in chip unit more functional modules, the chip needs fewer, the entire system has been significantly reduced cost. As a contradiction of the two components, the size and speed is not the same status. In contrast, meet the timetables and work is more important for some frequency when both conflicts, the use of priority guidelines.Area and the exchange rate is an important FPGA design ideas. Theoretically, if a design time series cushion larger, can run much higher than the frequency design requirements, then we can through the use of functional modules to reduce the consumption of the entire chip design area, which is used for space savings advantages of speed; Conversely, if the design of a time series demanding, less than ordinary methods of design frequency then generally flow through the string and data conversion, parallel reproduction of operational module, designed to take on the whole "string and conversion" and operate in the export module to chip in the data "and string conversion" from the macro point of view the whole chip meets the requirements of processing speed, which is equivalent to the area of reproduction - rate increase.For example. Assuming that the digital signal processing system is 350Mb/s input data flow rate, and in FPGA design, data processing modules for maximum processing speed of150Mb/s, because the data throughput processing module failed to meet requirements, it is impossible to achieve directly in the FPGA. Such circumstances, they should use "area-velocity" thinking, at least three processing modules from the first data sets will be imported and converted, and then use these three modules parallel processing of data distribution, then the results "and string conversion," we have complete data rate requirements. We look at both ends of the processing modules, data rate is 350Mb/s, and in view of the internal FPGA, each sub-module handles the data rate is 150Mb/s, in fact, all the data throughput is dependent on three security modules parallel processing subsidiary completed, that is used by more chip area achieve high-speed processing through "the area of reproduction for processing speed enhancement" and achieved design.FPGA is the English abbreviation Field of Programmable Gate Array for the site programmable gate array, which is in Pal, Gal, Epld, programmable device basis to further develop the product. It is as ASIC (ASIC) in the field of a semi-customized circuit and the emergence of both a customized solution to the shortage circuit, but overcome the original programmable devices doors circuit few limited shortcomings.FPGA logic module array adopted home (Logic Cell Array), a new concept of internal logic modules may include CLB (Configurable Logic Block), export import module IOB (Input Output Block) and internal links (Interconnect) 3. FPGA basic features are :(1) Using FPGA ASIC design ASIC using FPGA circuits, the chip can be used,while users do not need to vote films production.(2) FPGA do other customized or semi-customized ASIC circuits throughout the Chinese specimen films.3) FPGA internal capability and rich I/O Yinjue.4) FPGA is the ASIC design cycle, the shortest circuit, the lowest development costs, risks among the smallest device5) FPGA using high-speed Chmos crafts, low consumption, with CMOS, TTL low-power compatibleIt can be said that the FPGA chip is for small-scale systems to improve system integration, reliability one of the bestCurrently FPGA many varieties, the Revenue software series, TI companies TPC series, the fiex ALTERA company seriesFPGA is stored in films from the internal RAM procedures for the establishment of the state of its work, therefore, need to programmed the internal Ram. Depending on the different configuration, users can use a different programming methodsPlus electricity, FPGA, EPROM chips will be read into the film, programming RAM中data, configuration is completed, FPGA into working order. Diaodian, FPGA resume into white films, the internal logic of relations disappear, FPGA to repeated use. FPGA's programming is dedicated FPGA programming tool, using generic EPROM, prom programming device can. When the need to modify functional FPGA, EPROM can only change is. Thus, with a FPGA, different programming data to produce different circuit functions. Therefore, the use of FPGA very flexible.There are a variety of FPGA model : the main model for a parallel FPGA plus a EPROM manner; From the model can support a number of films FPGA; serial prom programming model could be used serial prom FPGA programming FPGA; The external model can be engineered as microprocessors from its programming microprocessors.Verilog HDL is a hardware description language for the algorithm level, doors at the level of abstract level to switch-level digital system design modelling. Modelling of the target figure by the complexity of the system can be something simple doors and integrity of electronic digital systems. Digital system to the levels described, and in the same manner described in Hin-time series modelling.Verilog HDL language with the following description of capacity : design behaviour characteristics, design data flow characteristics, composition and structure designed to control and contain the transmission and waveform design a certification mechanism. All this with the use of a modelling language. In addition, Verilog HDL language programming language interface provided by the interface in simulation, design certification from the external design of the visit, including specific simulation control and operation.Verilog HDL language grammar is not only a definition, but the definition of each grammar structure are clear simulation, simulation exercises. Therefore, the use of such language to use Verilog simulation models prepared by a certification. From the C programming language, the language inherited multiple operating sites and structures. Verilog HDL provides modelling capacity expansion, many of the initial expansion would be difficult to understand. However, the core subsets of Verilog HDL language very easy to learn and use, which is sufficient formost modelling applications. Of course, the integrity of the hardware description language is the most complex chips from the integrity of the electronic systems described.historyVerilog HDL language initially in 1983 by Gateway Design Automation companies for product development simulator hardware modelling language. Then it is only a dedicated language. Since their simulation, simulation devices widely used products, Verilog HDL as a user-friendly and practical language for many designers gradually accepted. In an effort to increase the popularity of the language activities, Verilog HDL language in 1990 was a public area. Open Verilog International (OVI) is to promote the development of Verilog international organizations. 1992, decided to promote OVI OVI standards as IEEE Verilog standards. The effort will ultimately succeed, a IEEE1995 Verilog language standard, known as IEEE Std 1364-1995. Integrity standards in Verilog hardware description language reference manual contains a detailed description.Main capacity：Listed below are the main Verilog hardware description language ability*Basic logic gate, and, for example, or have embedded in the language and nand* Users of the original definition of the term (UDP), the flexibility. Users can be defined in the original language combinations logic original language, the original language of logic could also be time series* Switches class infrastructure models, such as the nmos and pmos also be embedded in the language* Hin-language structure designated for the cost of printing the design and trails Shi Shi and design time series checks.* Available three different ways to design or mixed mode modelling. These methods include : acts described ways - use process of structural modelling; Data flow approach - use of a modelling approach Fuzhi expression; Structured way - using examples of words to describe modular doors and modelling.* Verilog HDL has two types of data : data types and sequence data line network types. Line network types that the physical links between components and sequence types that abstract data storage components.* To describe the level design, the structure can be used to describe any level module example* Design size can be arbitrary; Language is design size (size) impose any restrictions* Verilog HDL is no longer the exclusive language of certain companies but IEEE standards.* And the machine can read Verilog language, it may as EDA tools and languages of the world between the designers* Verilog HDL language to describe capacity through the use of programming language interface (PLI) mechanism further expansion. PLI is to allow external functions of the visit Verilog module information, allowing designers and simulator world Licheng assembly* Design to be described at a number of levels, from the switch level, doors level, register transfer level (RTL) to the algorithm level, including the level of process and content* To use embedded switching level of the original language in class switch design integrity modelling* Same language can be used to generate simulated incentive and certification by the designated testing conditions, such as the value of imports of the designated*Verilog HDL simulation to monitor the implementation of certification, the certification process of implementing the simulation can be designed to monitor and demonstrate value. These values can be used to compare with the expectations that are not matched in the case of print news reports.* Acts described in the class, not only in the RTL level Verilog HDL design description, and to describe their level architecture design algorithm level behavioural description* Examples can use doors and modular structure of language in a class structure described* Verilog HDL mixed mode modelling capabilities in the design of a different design in each module can level modelling* Verilog HDL has built-in logic function, such as*Structure of high-level programming languages, such as conditions of expression, and the cycle of expression language, language can be used* To it and can display regular modelling* Provide a powerful document literacy* Language in the specific circumstances of non-certainty that in the simulator, different models can produce different results; For example, describing events in the standard sequence of events is not defined.5、In troduction of DSPToday, DSP is w idely used in the modern techno logy and it has been the key part of many p roducts and p layed more and mo re impo rtant ro le in our daily life.Recent ly, Northw estern Po lytechnica lUniversity Aviation Microelect ronic Center has comp leted the design of digital signal signal p rocesso r co re NDSP25, w h ich is aim ing at TM S320C25 digital signal p rocesso r of Texas Inst rument TM S320 series. By using top 2dow n design flow , NDSP25 is compat ible w ith inst ruct ion and interface t im ing of TM S320C25.Digital signal processors (DSP) is a fit for real-time digital signal processing for high-speed dedicated processors, the main variety used for real-time digital signal processing to achieve rapid algorithms. In today's digital age background, the DSP has become the communications, computer, and consumer electronics products, and other fields based device.Digital signal processors and digital signal processing is inseparably, we usually say "DSP" can also mean the digital signal processing (Digital Signal Processing), is that in this digital signal processors Lane. Digital signal processing is a cover many disciplines applied to many areas and disciplines, refers to the use of computers or specialized processing equipment, the signals in digital form for the collection, conversion, recovery, valuation, enhancement, compression, identification, processing, the signals are compliant form. Digital signal processors for digital signal processing devices, it is accompanied by a digital signal processing to produce. DSP development process is broadly divided into three phases : the 20th century to the 1970s theory that the 1980s and 1990s for the development of products. Before the emergence of the digital signal processing in the DSP can only rely on microprocessors (MPU) to complete. However, the advantage of lower high-speed real-time processing can not meet the requirements. Therefore, until the 1970s, a talent made based DSP theory and algorithms. With LSI technology development in 1982 was the first recipient of the world gave birth to the DSP chip. Years later, the second generation based on CMOS工艺DSP chips have emerged. The late 1980s, the advent of the third generation of DSP chips. DSP is the fastest-growing 1990s, there have been four successive five-generation and the generation DSP devices. After 20 years of development, the application of DSP products has been extended to people's learning, work and all aspects of life and gradually become electronics products determinants.。
外文翻译--中文
多无线电发射极地和笛卡尔的相关架构的性能分析高效率开关模式功率放大器摘要：多无线电无线本文在频段的发射机架构为800 MHz - 6 GHz。
作为一个常量的后果在通信系统的演进过程中，移动发射TERS必须能够工作在不同频段和模式，根据现有的标准规格。
一个独特的多模架构的概念是一个EVO，多标准收发器lution特点是每个标准的电路并行。
多无线电概念就是优化表面和功耗。
发射机架构，使用的是抽样技术和基带，或PWM信号的编码之前，他们放大出现很好的候选人为多模。
发射机的几个原因——他们允许使用开关模式功率大、放大器的功率大，这是由于高度灵活易于集成他们的数字性质。
但是，当传输效率大，许多元素都必须考虑：信号的编码效率，射频滤波器，PA的效率。
本文探讨这些架构的利益——多无线发射器，能够支持现有的无线800兆赫和6 GHz 之间的通信标准。
它的计算和比较不同的可能architec、iMAX和LTE标准tures信号质量和发射功率的效率。
关键词：多模发射器，高效率的射频发射器，极性发射架构，笛卡尔变送器架构，E类功率放大器，PWM编码，WiMAX和LTE。
1引进和语境包括蜂窝无线通信系统通信，个人局域网（PAN），本地区域网络（LAN）和城域网（MAN）中已经提出了长足的发展。
近年来持续不断的改进。
共存同一个设备上的不同的无线标准是必要的以满足用户期望流动性，无处不在的CON-在同一时间连接和高速率的数据。
此coexis-唐塞不应增加设备的大小或减少其电池寿命。
这种演变的目标是减少外部元件数目和增加集成在低成本CMOS 技术的运移。
另一个阶段它奉行对认知无线电的演变并意味着每个阶段的沟通的灵活性链是认知无线电多。
而不是包括每个标准，通用的独立架构能够生成所有的发射器架构不同的标准波形，似乎是最好的解决办法（尤其是在消耗功率和表面占用pation）。
这个概念被称为多无线电发射机。
认知多无线电有能力multistan-dard的概念，而且是能够执行一个高效的环境频谱扫描和反应环境条件选择的拨款通信标准。
(完整word版)2ASK、2FSK、2PSK数字调制系统的Matlab实现及性能分析
2ASK、2FSK、2PSK数字调制系统的Matlab实现及性能分析比较引言:数字信号有两种传输方式，分别是基带传输方式和调制传输方式,即带通，在实际应用中,因基带信号含有大量低频分量不利于传送，所以必须经过载波和调制形成带通信号，通过数字基带信号对载波某些参量进行控制，使之随机带信号的变化而变化，这这一过程即为数字调制。
数字调制为信号长距离高效传输提供保障，现已广泛应用于生活和生产中.另外根据控制载波参量方式的不同，数字调制主要有调幅（ASK ），调频(FSK ），调相(PSK) 三种基本形式。
本次课题针对于二进制的2ASK 、2FSK 、2PSK 进行讨论，应用Matlab 矩阵实验室进行仿真,分析和修改，通过仿真系统生成一个人机交互界面，以利于仿真系统的操作。
通过对系统的仿真，更加直观的了解数字调制系统的性能及影响其性能的各种因素,以便于比较，评论和改进。
关键词： 数字，载波，调制,2ASK,2FSK ，2PSK ，Matlab ，仿真，性能，比较，分析正文：一 。
数字调制与解调原理1.1 2ASK（1）2ASK2ASK 就是把频率、相位作为常量，而把振幅作为变量，信息比特是通过载波的幅度来传递的。
由于调制信号只有0或1两个电平，相乘的结果相当于将载频或者关断，或者接通，它的实际意义是当调制的数字信号"1时，传输载波;当调制的数字信号为"0"时,不传输载波。
表达式为：⎩⎨⎧===001,cos )(2k k c ASK a a t A t s 当，当ω1。
2 2FSK2FSK 可以看做是2个不同频率的2ASK 的叠加，其调制与解调方法与2ASK 差不多，主要频率F1和F2，不同的组合产生所要求的2FSK 调制信号. 公式如下：1。
3 2PSK2PSK 以载波的相位变化为基准，载波的相位随数字基带序列信号的1或者0而改变，通常用已经调制完的载波的0或者π表示数据1或者0，每种相位与之一一对应。
实验2：AM调制与解调仿真
实验2：AM调制与解调仿真一、实验目的1、掌握AM的调制原理和MATLAB Simulink仿真方法2、掌握AM的解调原理和MATLAB Simulink仿真方法二、实验原理1、AM调制原理所谓调制，就是在传送信号的一方将所要传送的信号附加在高频振荡上，再由天线发射出去。
这里高频振荡波就是携带信号的运载工具，也叫载波。
振幅调制，就是由调制信号去控制高频载波的振幅，直至随调制信号做线性变化。
在线性调回系列中，最先应用的一种幅度调制是全条幅或常规调幅，简称为调制（AM）。
在频域中已调波频谱是基带调制信号频谱的线性位移；在时域中，已调波包络与调制信号波形呈线性关系。
m（t）为取值连续的调制信号，c（t）为正弦载波。
下图为AM调制原理图：2、AM解调原理从高频已调信号中恢复出调制信号的过程为解调，又称为检波。
对于振幅调制信号，解调就是从它的幅度变化上提取调制信号的过程，解调是调制的逆过程。
下图为AM解调原理图：三、实验步骤1、AM调制方式的MATLAB Simulink仿真（1）原理图（2）仿真图（3）仿真分析①调制器Constant和Add 以及低通滤波器，sine wave2和product1是对已调信号频谱进行线性搬移，低通滤波器是滤除高频部分，得到原始信号②调制后调制后信号加上了2v的偏置，频率变大了，幅度随时间在不断的呈现周期性变化，在1~2.5之间，大于调制前的幅度。
③模拟信号的调制是将要发送的模拟信号附加到高频振荡上，再由天线发射出去，这里的高频振荡就是载波。
振幅调制就是由调制信号去控制高频振荡的振幅，直至随调制信号做线性变化。
2、AM解调方式的MATLAB Simulink仿真（1）原理图（2）仿真图（3）仿真分析①调制器Sine wave2和product1是低通滤波器，Sine wave2 和 product1是对已调信号的频谱进行线性搬移，低通滤波器是滤除信号的高频部分以得到原始信号。
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本科生毕业设计外文资料翻译题目简易数字调制系统性能研究与仿真专业班级姓名指导教师所在学院附件1.外文资料翻译译文;2.外文原文附件1：外文资料翻译译文独立移动通信核心网的接入技术一、引言无线通信产业正着手探究第五代无线通信系统（5G）的接入需求和架构。
该工作的重点多集中于通过技术手段来提高空中接口的性能，如采用更快和更灵活的处理器，高级的多分支天线加工技术，先进的传输波形，以及使用毫米波长的频率。
这些手段将提高网络覆盖率，增强网络访问速度和能力。
5G技术还能提供一个提升移动通信核心网的契机，以减少网络运营商成本并提高用户体验。
对于该通信网络，我们所关注的其一个特定的方面为，多路访问技术和核心网之间的交互式通信过程。
WiFi和4G间的相互通信正变得越发普遍，推动了设备间通过无线多址技术来进行通信，并且促进网络的拓扑结构，如“受信任的非3GPP访问”支持无缝访问的选项，身份验证，无记名平台的交互式操作，以及在某些情况下的无缝移动应用。
我们预计设备间通过多个无线空中接口类型的通信机制将会得到持续发展。
然而，由于用来对不同空中接口类型提供支持的网络架构是由不同的标准制定厂商独立定义的,目前在网络功能和执行过程上几乎没有共同点。
这就导致了一个基于将移动网络重定向到另一个接入点上的交互式连接的过程，并在该过程中独立地执行网络连接步骤。
例如，对于每个接入链路，不同的接入链路中的信号传输会在很大程度上独立地形成在移动性，身份验证，通信策略上的重复。
在5G时代，我们期望设备可以同时通过包括诸如WiFi，4G，5G，亚6GHz以及高频段5G或毫米波通信等各种技术的多类型接口来进行连接。
利用这些技术，确保所部署的核心网络节点可以达到成本的最小化是十分重要的。
同时，在硬件传输据率和延迟允许的时候,服务和政策条件应统一适用于所有的接入技术。
在本文中，我们描述了独立的通信核心网接入技术的概念并达到了这些目标。
本文的结构组织如下：在第二节中，我们给出了一个对4G多制式网络融合技术的概述，并重点研究3GPP和WiFi多制式网络融合技术，指出了其特殊功能和其作用领域及其所产生的无效性。
随后我们定义一个对独立核心网络的访问，并在第三节中给出位于网络架构第二层的多制式网络融合技术和位于第三层的访问独立功能上的区别。
最后我们在第四节中给出一个访问独立通信网的设计并在第五节中作出总结。
二、对4G中WiFi互联技术的概述移动网络运营商(MNOs)所经历的在数据流量上的巨大增长已经过去一段时间，这是由信息传播设备（尤其是智能手机）所推动的。
运营商通过部署4G网络，目前已适应了这样的增长，并且通过部署额外的收发系统以及小型基站来进一步提升数据处理能力。
同时，在家里和公司中，对无线网络的访问正变得无处不在，并且很多服务商在数据流量交换密集的区域部署了大量公共接入点。
移动通信行业和3GPP（第3代移动通信合作计划）技术对此积极回应，发挥了一个对无线网络的普及作用，特别是补充了当设备不是很方便携带时的可接入性。
为了达到WiFi和3GPP之间实现互联的需求，人们定义了横跨控制，管理和承载层面上的相关技术及标准。
由于4G 和3GPP核心通信网并未被设计成独立访问的系统，因此我们对互联方案所涉及的内容做一个简单的描述，以强调其用于WiFi互联技术上的特殊功能和作用。
附件2：外文原文ACCESS INDEPENDENT MOBILE CORE NETWORKS一、AbstractOne of the goals of 5G is to simplify the 4G core network and enable interworking across multiple Radio Access Technologies (RA Ts) without the addition of specialized network elements, protocols and interfaces that perform often redundant functions for each access technology. Achieving this reduces operator costs, and can improve the subscriber experience when devices can access multiple RATs. In this paper we propose the notion of Access Independent Core Network (AICN) where devices can connect to the core through any technology and consume a set of services. We show that it is possible to achieve access independence through a core network architecture that includes access independent functions, access independent bearer plane protocols, and access independent signaling between the network controller and the device and access switch. We outline benefits of the proposed architecture through examples involving WiFi integration. The wireless industry is beginning to explore requirements and architectures for the evolution of wireless access to a fifth generation (5G). Much of this effort is focused on improving air-interface performance by exploiting technology enablers such as faster and more flexible processors, advanced, multi-branch antenna processing, advanced waveforms, and use of millimeter wavelength frequencies. These enablers will improve coverage and increase access speeds and capacity. 5G also offers an opportunity to enhance the mobile core network to reduce network operator costs and improve user experience. One particular aspect of the networking that we focus on is the inter- working between multiple access technologies and the core network. WiFi and 4G inter-working is becoming increasingly prevalent resulting in devices communicating over multiple radio access technologies, and network topologies such as “trusted non-3GPP access”that support seamless access selection, authentication, bearer plane inter -operation, and in some cases seamless mobility. We expect this trend to continue with devices communicating over multiple wireless air-interface types. However, because the supporting network architectures for the different air-interface types were defined independently by different standards bodies, today there is little commonality in network functions and procedures. This leads to a course level of interworking based on redirecting the mobile to an alternative access where it exercises these separate network procedures. For example, signaling on different accesslinks is largely separate resulting in duplication of mobility, authentication and policy signaling for each access.At the time of 5G, devices are expected to simultaneously connect through multiple interfaces involving a variety of technologies such as WiFi, 4G, 5G sub 6 GHz and 5G high band or mmwave. It is important to ensure that the number of core network nodes deployed to exploit this variety of technologies is minimized to reduce costs. At the same time, when the physical data rates and latencies permit, services and policies should be uniformly applicable to all access technologies. In this paper, we describe the concept of an access independent core network that achieves these goals.This paper is organized as follows. In Section 2 we give an overview of 4G multi -RAT interworking, focusing on 3GPP and Wifi RA Ts, and indicate where special functions and procedures are required resulting in inefficiencies. We then define an access independent core, distinguishing between RAT specific layer 2 functions and access independent functions at layer 3 in Section 3. Lastly we present a design for an access independent network in Section 4 and conclude in Section 5.二、OVERVIEW OF WIFI INTERWORKING IN 4GFor some time now Mobile Network Operators (MNOs) have experienced an enormous growth in data traffic, driven by proliferation of data capable devices, particularly smart phones. Network operators have accommodated this growth by deploying 4G, and further increasing capacity through additional carriers and by deploying small cells. At the same time Wifi access has become ubiquitous in homes and enterprises, and a variety of service providers have deployed public hot spots in areas where traffic is dense. Both the mobile industry and 3GPP have responded positively, embracing the expanding role of Wifi in providing connectivity, particularly to supplement access capacity when devices are not highly mobile.To meet the needs of interworking between WiFi and 3GPP, several techniques spanning the control plane, management plane and the bearer plane have been defined and standardized. We describe briefly the various aspects of the interworking solution to highlight the fact that special functions and procedures are required for WiFi interworking,because the 4G 3GPP core networkand signaling have not been designed to be access independent. 附件3：外文资料翻译译文通信系统中数字调制技术的研究与仿真近几年来，移动通信中调制解调技术优势在发展速度和新应用数目方面，给人非常深刻的印象。