数字调制解调技术中英文资料外文翻译文献

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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。

单片机外文文献翻译--基于MSP430的FSK调制解调

中文翻译材料英文题目FSK Modulation and Demodulation With the MSP430 Microcotroller中文题目基于MSP430的FSK调制解调学院：计算机科学与技术学院专业：通信工程学生姓名：指导教师：二O一三年六月IMPORTANT NOTICETexas Instruments and its subsidiaries (TI) reserve the right to make changes to their products or to discontinueany product or service without notice, and advise customers to obtain the latest version of relevant informationto verify, before placing orders, that information being relied on is current and complete. All products are sold subject to the terms and conditions of sale supplied at the time of order acknowledgement, including those pertaining to warranty, patent infringement, and limitation of liability.TI warrants performance of its semiconductor products to the specifications applicable at the time of sale in accordance with TI’s standard warranty. Testing and other quality control techniques are utilized to the extentTI deems necessary to support this warranty. Specific testing of all parameters of each device is not necessarily performed, except those mandated by government requirements.CERTAIN APPLICATIONS USING SEMICONDUCTOR PRODUCTS MAY INVOLVE POTENTIAL RISKS OF DEATH, PERSONAL INJURY, OR SEVERE PR OPERTY OR ENVIRONMENTAL DAMAGE (“CRITICAL APPLICATIONS”). TI SEMICONDUCTOR PRODUCTS ARE NOT DESIGNED, AUTHORIZED, OR WARRANTED TO BE SUITABLE FOR USE IN LIFE-SUPPORT DEVICES OR SYSTEMS OR OTHER CRITICAL APPLICATIONS. INCLUSION OF TI PRODUCTS IN SUCH APPLICATIONS IS UNDERSTOOD TO BE FULLY AT THE CUSTOMER’S RISK.In order to minimize risks associated with the customer’s applications, adequate design and operatingsafeguards must be provided by the customer to minimize inherent or procedural hazards.TI assumes no liability for applications assistance or customer product design. TI does not warrant or representthat any license, either express or implied, is granted under any patent right, copyright, mask work right, or other intellectual property right of TI covering or relating to any combination, machine, or process in which such semiconductor products or services might be or are used. TI’s publication of information regarding any thirdparty’s products or services does not constitute TI’s approval, warranty or endorsement thereof.Copyright 1998, Texas Instruments IncorporatedContents1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (1)2 Demodulation Theory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.1 Choosing the Sampling Rate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.2 Front End Processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.3 FSK Demodulation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.4 Bit Synchronization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3 Modulation Theory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.1 Choosing the Sampling Rate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.2 Constructing the Look Up Table . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.3 FSK Modulation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2 2 2 2 3 4 4 4 44 Data Conversion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (5)4.1 A/D Conversion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (5)4.2 D/A Conversion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (5)5 Power Consumption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (6)6 Exercising the Software . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (7)6.1 FSK Receiver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (7)6.2 FSK Transmitter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (7)7 Example Circuits . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (8)7.1 Using the MSP430C325 as Main Processor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (8)7.2 Example Telephone Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (8)8 Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (10)9 References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (11)FSK Modulation and Demodulation With the MSP430 MicrocontrolleriFSK Modulation and Demodulation With the MSP430MicrocontrollerABSTRACTThis application report describes a software program for performing V.23 FSK modemtransceiver functions using an MSP430 microcontroller. It makes use of novel filterarchitecture to perform DSP functions on a processor with only shift and add capabilities.1 IntroductionMany measurement applications (for example, electric and gas meters) requirea way to communicate electronically with a central office so that measured datacan be reported back to the central office and new tariffs can be set in the remotesite. Telephony provides a convenient means of data communication.Frequency shift keying (FSK) and dual tone multi frequency (DTMF) are twopopular methods of representing binary data over telephone circuits. Thisapplication report describes a V.23-compliant FSK transceiver software module.Integrating the measurement and communication functions onto the same chipyields cost as well as power-saving benefits. Using the MSP430, a high MIPs ultralow power microprocessor, allows power to be drawn from the telephone line insome cases.This report describes the mathematical formulas for FSK signal transmission anddetection. A list of the software modules is included with a reference schematicfor telephone interface and low cost A/D converter. The schematic is only areference, since the precise implementation can vary from country to country.1Demodulation Theory2 Demodulation TheoryA quadrature demodulator provides the FSK demodulation. In this type ofdemodulation, the signal and its delayed version are multiplied together and then low-pass filtered. If the delay, T, is set such that Wcarrier ⋅ T = /2, then thelow-pass filter result is proportional to the frequency deviation from the carrier and therefore represents the bit value sent.If wWcarrierwhere w = 2π ⋅ f : + " Wdelta and T+ Wcarrier +p 2 ³2.1 cos[wt].cos[w(t –T)]coswT coswTsin[" Wdelta] + ) cos(2wt –wT) ³ Low Pass Filter +" sin[Wdelta]_ Choosing the Sampling RateThe sampling is chosen to be Fcarrier4 for the purpose of obtaining thedelayed sample without computational overhead. For V.23, the F carrierfrequency is 1700 Hz and therefore the sampling rate becomes 6800 Hz. Using a 32768-Hz crystal yields 6793.3 Hz, which is 0.1% out. The sampling frequency is set by the 8-bit interval timer. Because this timer is limited to 256 counts, the interrupt rated is set to twice the sampling rate and the processing is divided into two halves with signal sampling performed every other interrupt.2.2 Front End ProcessingMost A/D converters, including the successive approximation A/D converter in the MSP430C325, need a dc bias; this yields an unsigned integer sample with an offset. Before this sample can be processed further, it needs to go through an unbias filter to take out the dc bias and turn the sample into a signed integer value. This unbias filtering also gives 30 dB or so of rejection for main frequencies.2.3 FSK DemodulationThe signed integer sample and its delayed version are multiplied together; in this application, an 8×8 signed multiplication loop is used.The product, made up of two frequency elements, is low-pass filtered to remove the double frequency element. The remainder is a signed integer valuerepresenting the original bit value transmitted.The low-pass filter uses the digital wave filtering technique. This technique gives stable characteristics with very good coefficient tolerance. All multiplication is done through shifts and adds with the number of shift/add operations minimized through rounding off the coefficients. Because the filter has good coefficienttolerance, this rounding off does not affect the filter performance. The Butterworth filter used here gives approximately 40-dB attenuation in the stop band with 1-dB pass and ripple.2SLAA037Demodulation Theory2.4Bit SynchronizationThe bit values coming out from demodulation need to be determined andsynchronized to produce the incoming data bit stream. This process is alsoknown as bit slicing and clock recovery. Because the sampling rate at 6800 is notan integer multiple of the data rate (baud rate) at 1200, an additional step isneeded to consolidate between the two rates. This is done through a count-downcounter with a sequence of preload value (5,6,5). Every 17 samples, the samplingrate and the data baud rate are resynchronized. Bit synchronization or clockrecovery is done by monitoring bit value transitions. Lead or lag information isthen obtained and the count-down counter is adjusted accordingly. Because ofthe difference between the sampling clock and the data clock, the data bit is neversampled at the middle of the baud period; instead a –5% to 13% variation isintroduced. However, this should not have any adverse effect on the accuracy ofthe system, as it has been verified experimentally.3FSK Modulation and Demodulation With the MSP430 MicrocontrollerModulation Theory3 Modulation TheoryFSK modulation involves alternating the value of a delta frequency from a carrierfrequency according to the value of the bit to be represented. For V.23, a bit valueof 0 = 400 Hz and a bit value of 1 = –400 Hz.FSK signal + Amplitudecos[t| 2p(Fcarrier" Fdelta)]The sinusoidal signal is generated through a lookup table which contains cosine values from 0 to 2π. A parameter called PHASER (16 bit) represents the current angle: 0=0 degree, 8000 hex = 180 degree 10000 hex = 360 degree. With each sample, this angle is advanced by another parameter DELTA (16 bit) which determines the frequency of the signal (larger DELTA value = higher frequency). Frequency modulation is realized by changing the DELTA value according to the bit value to be transmitted at each baud period, according to the following formula:DELTA + Fdesired Fsampling65536.The advantage of this method over a digital oscillator method is that this methodpreserves the phase relationship even when the frequency is shifted from sampleto sample.3.1Choosing the Sampling RateThe 8-bit interval timer sets the sampling rate to 19200 samples/s. This rate issubdividable into the data baud rate of 1200. Also, it is sufficiently high to makethe D/A process simpler.3.2Constructing the Look Up TableTo save ROM space, only the first quadrant (0 to 127 degrees) in Q7 format iscoded. This is done by dividing the first quadrant (90 degrees) into 128 steps ofapproximately 0.7 degrees each. The remaining three quadrants can be workedout from this first quadrant table using additional computation.3.3FSK ModulationThe parameter PHASER is advanced by the amount DELTA at every interrupt.The first 9 bits of the PHASER is used to look up the cosine value. For the cosinefunction, the third and fourth quadrant are the same as the second and firstquadrant, and so only the absolute value of the first 9 bits of PHASER is used.Next, all second quadrant values are derived from the first quadrant ROM table.The 8-bit result value is stored onto P0.OUT.Every 16 interrupts, the parameter DELTA is updated with the next frequency bylooking at the next bit to be transmitted.4SLAA037Data Conversion 4 Data ConversionThis section describes the required digital-to-analog (D/A) and analog-to-digital(A/D) data conversions.4.1A/D ConversionThe most straightforward way to digitize the incoming FSK signal is to use the12-bit mode of the internal 14-bit A/D converter of the MSP430C325. However,not all of the 12 bits are needed to achieve good dynamic range for the FSKdemodulation. Simulation results indicate that an 8-bit A/D stage gives gooddynamic range up to 25 dB using internal AGC software. With an additionalexternal AGC stage, the dynamic range can be further widened. As economicalmeans of building 8-bit single slope A/D exists, this extends the application of thismodule to the rest of the MSP430 family. The application software included hereuses a single slope A/D (universal timer with external comparator) for thedemodulator. This makes the software universally applicable for the whole family.4.2D/A ConversionA 6-bit external R–2R ladder is used to construct the D/A converter. Because thecarrier frequency of 19200 Hz is nine times the highest frequency of the FSK of2100 Hz, the post filtering stage should be relatively simple. In the applicationcircuit, a single capacitor forms a single pole low pass filter but more poles canbe realized using additional passive networks.5FSK Modulation and Demodulation With the MSP430 MicrocontrollerPower Consumption5 Power ConsumptionThe FSK concept is designed with low power in mind. The FSK demodulatortakes less than 2 MIPs. With a low power op-amp as a front-end, total powerconsumption of less that 1.5 mA should be achievable. Thus, it is possible thatthe power can be derived entirely from the telephone line. A schematic is includedfor a suggested telephone line interface. The precise configuration may vary fromcountry to country.SLAA0376Exercising the Software 6 Exercising the SoftwareThis section describes operation of the software.6.1FSK ReceiverThe FSK signal is derived from the telecom interface circuit. This signal shouldhave a dc bias of 1.2 V and a peak-to-peak level of 400 mV. The software decodesthis FSK signal and produces three outputs which lets the user monitor thedemodulated data.TP.3. This is the clock signal recovered from the input FSK.TP.5. This is the data recovered from the input FSK; data is latched out everyrising edge of TP.3.P0.2–P0.7. These six bits output the low pass filtered result. With an externalR–2R ladder this becomes very useful in monitoring the analogue FSKdemodulator output level. It is hard limited to 8 bits with the MSB 6 bits loadedto port P06.2FSK TransmitterThe transmitter software outputs an FSK signal according to the BIT MAP datadefined in TX_DATA_TABLE. The bitmap pattern starts with a preamble followedby a long MARK period. Then the actual data is transmitted. This table uses a zeroword as an end marker, and the software restarts the whole data sequence uponreaching a zero value in the bit map data.7FSK Modulation and Demodulation With the MSP430 MicrocontrollerExample Circuits7 Example CircuitsThis section shows and describes example circuits.7.1Using the MSP430C325 as Main ProcessorFigure 1 shows an example circuit using the MSP430C325 as the mainprocessor. The circuit is tested with 400 mV peak-to-peak FSK input. To obtainthe same results, Rx needs to be biased at 1.2 V with a 400 mV peak-to-peak FSKsignal superimposed.VSSR1 R2PO.2PO.3PO.4PO.5PO.6PO.7MSP430E325TP.5TP.1 TP.4 CIN TP.3RX_CLKLine InterfaceRXTXHook 14066AC13VCC1N414833 kΩVoltage RampPNPSample_HoldNPN 1 nF 6_ 5+B 2RX_DATA7Figure 1. Main Processor and A/D Converter7.2Example Telephone InterfaceFigure 2 shows an example telephone interface, and Table 1 lists FSK transceiverperformance data.8SLAA037Example Circuits 20 kΩ1 ∝Φ+ 1 kΩ1 kΩVREF(1.5 V)TLC22796_5+33 kΩ20 kΩ20 kΩ20 kΩ10 kΩ9_10+Telephone LineAB33 nF500 &6–8 V ZenersTuning ForMinimum Side Tone6–8 V ZenersTX7DC TelephoneIsolationTransformer8150 kΩ400 mV pk–pkRX 1 ∝Φ+–+131233 kΩ33 kΩ680Hook150 kΩ14This is a reference circuit only and may not be applicable under some circumstances.Figure 2. Telephone InterfaceTable 1. FSK Transceiver PerformanceRAM (BYTES)FSK Receiver FSK Transmitter1812ROM (BYTES)512400MIPS (APPROX.)21.4FSK Modulation and Demodulation With the MSP430 Microcontroller9Summary8 SummaryFSK transceivers are normally realized by either analog means or by the use ofDSPs with hardware MAC units. Using an MSP430 RISC processor without ahardware MAC to achieve the transceiver function is a very unusual approach.The ability to create filters using digital wave filtering techniques, together with theorthogonal instruction set and the 16 bit architecture of the MSP430, makes thecode very ROM and MIPs efficient. Moreover, the ultra low power capability of theMSP430 means that power can readily be derived from the phone line. This leadsto component-efficient designs. The author has conducted other tests toconclude that, with some enhancements, the FSK receiver can work with an 8-bitA/D converter with enough sensitivity. Therefore the FSK transceiver can beimplemented economically across the whole MSP430 family.SLAA03710References9 References1. Texas Instruments: MSP430 Family, Architecture User’s Guide and ModuleLibrary.2. Texas Instruments Digital Signal Processing Application with the TMS320Family Volume 2.3. Gaszi, L: Explicit Formulas for Lattice Wave Digital Filters; IEEE Trans. OnCircuits and Systems VOL. CAS-32, NO. 1, January 198511FSK Modulation and Demodulation With the MSP430 Microcontroller基于MSP430的FSK调制解调——应用报告声明德州仪器（TI）及其附属公司（TI）保留改进产品或停止任何服务的权力，并且不再另行通知，建议客户获核实最新版本或相关信息，在下订单前，该信息是当前最有效和完整的。

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.。

中央空调控制系统中英文对照外文翻译文献

中英文对照外文翻译(文档含英文原文和中文翻译)基于PLC的中央空调控制系统1引言在PLC被开发出来的三十年里，它经过不断地发展，已经能结合模拟I/O，网络通信以及采用新的编程标准如IEC 61131-3。

然而，工程师们只需利用数字I/O和少量的模拟I/O数以及简单的编程技巧就可开发出80%的工业应用。

PLC已经广泛的应用在所有的工业部门。

据“美国市场信息”的世界PLC以及软件市场报告称，1995年全球PLC及其软件的市场经济规模约50亿美元[5]。

随着电子技术和计算机技术的发展，PLC的功能得到大大的增强。

由于采用传统的工具可以解决80%的工业应用，这样就强烈地需要有低成本简单的PLC；从而促进了低成本微型PLC的增长，它带有用梯形逻辑编程的数字I/O。

然而，这也在控制技术上造成了不连续性，一方面80%的应用需要使用简单的低成本控制器，而另一方面其它的20%应用则超出了传统控制系统所能提供的功能。

工程师在开发这些20%的应用需要有更高的循环速率，高级控制算法，更多模拟功能以及能更好地和企业网络集成。

在八十和九十年代，那些要开发“20%应用”的工程师们已考虑在工业控制中使用PC。

PC所提供的软件功能可以执行高级任务，提供丰富的图形化编程和用户环境，并且PC的COTS部件使控制工程师能把不断发展的技术用于其它应用。

这些技术包括浮点处理器；高速I/O总线，如PCI和以太网；固定数据存储器；图形化软件开发工具。

而且PC还能提供无比的灵活性，高效的软件以及高级的低成本硬件。

冰蓄冷中央空调是将电网夜间谷荷多余电力以冰的冷量形式储存起来，在白天用电高峰时将冰融化提供空调服务。

由于我国大部分地区夜间电价比白天低得多，所以采用冰储冷中央空调能大大减少用户的运行费用。

冰蓄冷中央空调系统配置的设备比常规空调系统要增加一些，自动化程度要求较高，但它能自动实现在满足建筑物全天空调要求的条件下将每天所蓄的能量全部用完，最大限度地节省运行费用。

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。

数字信号处理的简单介绍文献翻译及英文原文

单位代码01学号070110105分类号密级文献翻译数字信号处理的简单论述院（系）名称信息工程学院专业名称通信工程学生姓名徐治明指导教师赵春雨2011年 4 月 5 日英文译文数字信号处理的简单论述冈萨雷斯（美国）一、数字信号处理的概述数字信号处理是将信号以数字方式表示并处理的理论和技术。

数字信号处理与模拟信号处理是信号处理的子集。

数字信号处理的目的是对真实世界的连续模拟信号进行测量或滤波。

因此在进行数字信号处理之前需要将信号从模拟域转换到数字域，这通常通过模数转换器实现。

而数字信号处理的输出经常也要变换到模拟域，这是通过数模转换器实现的。

数字信号处理的算法需要利用计算机或专用处理设备如数字信号处理器（DSP）和专用集成电路（ASIC）等。

数字信号处理技术及设备具有灵活、精确、抗干扰强、设备尺寸小、造价低、速度快等突出优点，这些都是模拟信号处理技术与设备所无法比拟的。

数字信号处理的核心算法是离散傅立叶变换(DFT)，是DFT使信号在数字域和频域都实现了离散化，从而可以用通用计算机处理离散信号。

而使数字信号处理从理论走向实用的是快速傅立叶变换(FFT)，FFT的出现大大减少了DFT的运算量，使实时的数字信号处理成为可能、极大促进了该学科的发展。

世界上三大DSP芯片生产商:1.德克萨斯仪器公司(TI) 2.模拟器件公司(ADI) 3.摩托罗拉公司(Motorola).这三家公司几乎垄断了通用DSP芯片市场。

数字信号处理的经典书籍是麻省理工学院奥本海姆编著的《Discrete Time Signal Processing》，有中译本《离散时间信号处理》由西安交通大学出版。

现在是第二版。

二、特征和分类信号(signal)是一种物理体现，或是传递信息的函数。

而信息是信号的具体内容。

模拟信号(analog signal)：指时间连续、幅度连续的信号。

数字信号(digital signal)：时间和幅度上都是离散(量化)的信号。

数字信号处理中英文对照外文翻译文献

中英文对照外文翻译(文档含英文原文和中文翻译)数字信号处理一、导论数字信号处理（DSP）是由一系列的数字或符号来表示这些信号的处理的过程的。

数字信号处理与模拟信号处理属于信号处理领域。

DSP包括子域的音频和语音信号处理，雷达和声纳信号处理，传感器阵列处理，谱估计，统计信号处理，数字图像处理，通信信号处理，生物医学信号处理，地震数据处理等。

由于DSP的目标通常是对连续的真实世界的模拟信号进行测量或滤波，第一步通常是通过使用一个模拟到数字的转换器将信号从模拟信号转化到数字信号。

通常，所需的输出信号却是一个模拟输出信号，因此这就需要一个数字到模拟的转换器。

即使这个过程比模拟处理更复杂的和而且具有离散值，由于数字信号处理的错误检测和校正不易受噪声影响，它的稳定性使得它优于许多模拟信号处理的应用（虽然不是全部）。

DSP算法一直是运行在标准的计算机，被称为数字信号处理器（DSP）的专用处理器或在专用硬件如特殊应用集成电路（ASIC）。

目前有用于数字信号处理的附加技术包括更强大的通用微处理器，现场可编程门阵列（FPGA），数字信号控制器（大多为工业应用，如电机控制）和流处理器和其他相关技术。

在数字信号处理过程中，工程师通常研究数字信号的以下领域：时间域（一维信号），空间域（多维信号），频率域，域和小波域的自相关。

他们选择在哪个领域过程中的一个信号，做一个明智的猜测（或通过尝试不同的可能性）作为该域的最佳代表的信号的本质特征。

从测量装置对样品序列产生一个时间或空间域表示，而离散傅立叶变换产生的频谱的频率域信息。

自相关的定义是互相关的信号本身在不同时间间隔的时间或空间的相关情况。

二、信号采样随着计算机的应用越来越多地使用，数字信号处理的需要也增加了。

为了在计算机上使用一个模拟信号的计算机，它上面必须使用模拟到数字的转换器（ADC）使其数字化。

采样通常分两阶段进行，离散化和量化。

在离散化阶段，信号的空间被划分成等价类和量化是通过一组有限的具有代表性的信号值来代替信号近似值。

5G无线通信网络中英文对照外文翻译文献

5G无线通信网络中英文对照外文翻译文献(文档含英文原文和中文翻译)翻译：5G无线通信网络的蜂窝结构和关键技术摘要第四代无线通信系统已经或者即将在许多国家部署。

然而，随着无线移动设备和服务的激增，仍然有一些挑战尤其是4G所不能容纳的，例如像频谱危机和高能量消耗。

无线系统设计师们面临着满足新型无线应用对高数据速率和机动性要求的持续性增长的需求，因此他们已经开始研究被期望于2020年后就能部署的第五代无线系统。

在这篇文章里面，我们提出一个有内门和外门情景之分的潜在的蜂窝结构，并且讨论了多种可行性关于5G无线通信系统的技术，比如大量的MIMO技术，节能通信，认知的广播网络和可见光通信。

面临潜在技术的未知挑战也被讨论了。

介绍信息通信技术（ICT）创新合理的使用对世界经济的提高变得越来越重要。

无线通信网络在全球ICT战略中也许是最挑剔的元素，并且支撑着很多其他的行业，它是世界上成长最快最有活力的行业之一。

欧洲移动天文台（EMO）报道2010年移动通信业总计税收1740亿欧元,从而超过了航空航天业和制药业。

无线技术的发展大大提高了人们在商业运作和社交功能方面通信和生活的能力无线移动通信的显著成就表现在技术创新的快速步伐。

从1991年二代移动通信系统(2G)的初次登场到2001年三代系统（3G）的首次起飞，无线移动网络已经实现了从一个纯粹的技术系统到一个能承载大量多媒体内容网络的转变。

4G无线系统被设计出来用来满足IMT-A技术使用IP面向所有服务的需求。

在4G系统中，先进的无线接口被用于正交频分复用技术（OFDM），多输入多输出系统（MIMO）和链路自适应技术。

4G无线网络可支持数据速率可达1Gb/s的低流度，比如流动局域无线访问，还有速率高达100M/s的高流速，例如像移动访问。

LTE系统和它的延伸系统LTE-A，作为实用的4G系统已经在全球于最近期或不久的将来部署。

然而，每年仍然有戏剧性增长数量的用户支持移动宽频带系统。

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数字调制解调技术中英文资料外文翻译文献英文文献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'sthe 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, isimportant 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 theimprovement 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 is negligible,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 parallelword 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。