MPEG System Streams in Best-Effort Networks
Michael Hemy,Urs Hengartner,Peter Steenkiste,and Thomas Gross
Department of Computer Science Departement Informatik
School of Computer Science ETH Z¨u rich
Carnegie Mellon CH8092Z¨u rich
Abstract
The MPEG format is widely used and therefore an attractive vehicle for the distribution of video and audio material over the Internet.However,the hierarchical structure of MPEG systems complicates
the task of delivering continuous,synchronized streams of video and audio in a best-effort environment
(today’s Internet).If the network throws away packets on encountering congestion,the video and audio
stream may loose synchronization for a number of frames.Therefore,adapting the resource demands of
an MPEG system must be done by an entity that is knowledgeable of the MPEG system structure:an
MPEG system?lter.This paper describes the integration of such an MPEG system?lter into a network.
Our experience to date indicates that mid-range PCs can host such a?lter,and that the?lter succeeds in
adapting the resource requirements of an MPEG system in response to changes in the network load.
1Introduction
Distribution of a movie via a network is a topic of great interest but also poses a number of challenges.Since the movie is played as it is received,the transfer over the network must proceed at a speci?c rate to prevent buffer over?ow or under?ow at the player.If there is competing traf?c in the network,however,there is always the risk of congestion,and consequently,packets may be dropped or delayed.
A number of researchers have suggested the use of reservations of network resources to avoid con-gestion.Reservations,however,are only supported by some networks and often carry prohibitive costs or high overheads.The overhead of setting up a reservation may be tolerable if we play a full-length(i.e.,90 minutes)movie.However,the advent of multi-media indexing and retrieval systems means that we will increasingly see the distribution of many short video clips instead of a small number of large ones.A mul-timedia database like the core of the Informedia system[8]developed at Carnegie Mellon contains a large number of short movie segments–video clips,sound bites from TV news,movie story boards that provide access to individual scenes and shots,commercials,etc.(Today the size of the data collection is about1 TeraByte.)To allow remote access to such a data collection,we investigate transmission of movies over existing best-effort networks that make up the Internet.The key problem is to ensure that the player re-ceives a continuous feed in the presence of variable congestion.Our goal is to maximize the transfer of comprehensible information given the available(varying)bandwidth.
There are many formats to encode movies.We focus on the MPEG format–it is a widely used interna-tional standard[1]that has been adopted by the Informedia project[8].MPEG is supported also by the Java Multimedia Framework(JMF)[9],so we anticipate that MPEG will continue to grow in importance for In-ternet applications(e.g.,JMF makes it easy to include MPEG movies in applets or HTML pages).However,
the characteristics of MPEG systems provide various challenges when streaming MPEG over a best-effort network where random packet losses can result in a signi?cant loss in quality.In this paper,we present a solution based on a?lter that understands the MPEG format and that can adapt the bandwidth requirements of the video stream based on feedback about the network conditions while maintaining the MPEG system format.While we focus on one particular video format,the basic concepts presented here can also be ap-plied to other standards since the key features of MPEG(e.g.,presence of meta data for synchronization, inter-frame encoding)are likely to remain widely used.
In the next two sections,we brie?y review the structure of MPEG systems and the challenges it poses for streaming over best-effort networks,and we provide an overview of related work.Then we present our architecture,describe the implementation,and present a preliminary performance evaluation.
2Background
We give some background on MPEG-1and describe the challenges associated with streaming MPEG-1over best-effort networks.
2.1MPEG-1
MPEG-1was primarily designed for storing video data(moving pictures)and its associated audio data on digital storage media.As such the MPEG standard deals with two aspects of encoding:compression and synchronization.MPEG speci?es an algorithm for compressing video pictures(ISO-11172-2)and audio (ISO-11172-3)and then provides the facility to synchronize multiple audio and multiple video bitstreams (ISO-11172-1)in an MPEG system.MPEG-1is intended for intermediate data rates on the order of1.5 Mbit/sec.
2.1.1MPEG-1Video
An MPEG video stream distinguishes between I-pictures,P-pictures,and B-pictures–these pictures differ in the coding scheme.The three types of coding provide three levels of compression by exploiting similarities within the picture or similarities to neighboring pictures.
2.1.2MPEG-1Audio
An MPEG-1audio stream consists of audio coded using one of three algorithms,which offer different levels of complexity and subjective quality.These algorithms are referred as’layers’in the coding standard. The coding algorithms use psycho-acoustic properties of the human hearing to compress the data(lossy compression).
2.1.3MPEG-1Systems
The MPEG system layer is responsible for combining one or more compressed audio and video bitstreams into a single bitstream.This is done by interleaving data from the video and audio streams,combined with meta data that provides the timing control and synchronization.
2.2Best-effort networks
The infrastructure provided by today’s IP-based networks provides access to a large number of nodes.Unfor-tunately,IPv4(the current standard)provides no framework for resource https://www.doczj.com/doc/b517803893.html,ers are competing for bandwidth,and if a link becomes congested(demand for bandwidth is higher than the link capacity), packets are dropped.Since traf?c conditions change continuously,congestion can start and disappear at
any time.Note that in the current Internet,there is an assumption that it is the source’s responsibility to reduce the data send rate when packet losses are observed to reduce congestion.For most applications,this reduction is done by TCP,the dominant Internet transport protocol,but if an applications takes control of managing the send rate,it should also abide by this rule.
Random packet loss can hurt MPEG-1systems in two ways,besides the obvious fact that the information in the packet is lost.When we analyze random packet losses,we must take into account that network packets may not correspond to MPEG packets and that the latter are a layer completely separate from the video frames.The amount of impact that the loss of a particular packet will have depends on its location in the stream and on the robustness of the player in recovering from errors.In the worst case,we may loose a network packet that contains meta data of the whole MPEG stream(the MPEG system header),and players that rely solely on synchronization information found in the stream will signi?cantly be impacted when such information is lost.In a typical scenario,it is most likely that a packet lost will contain some part of a video frame with meta data(video being the predominant stream).
In the context of the MPEG layers,a network loss translates into a disruption in the system pack layer and may result in concatenating parts from two different MPEG packets.This loss can induce corruption in the lower layers,e.g.,corruption of the video or audio data.If video data has been affected,the frame is decoded incorrectly.An incorrect I-frame or P-frame propagates problems to all dependent frames and corrupts these as well.In the worst case,we may loose a whole group of pictures(GOP),typically equivalent to half a second of video.For various MPEG streams,our experiments have shown that a random loss of 1%of network packets can translate into as high as10%damaged video frames.Similarly,Boyce et al.[4] noticed that packet loss rates as low as3%translated into frame error rates as high as30%.
3Related work
There are several issues that must be dealt with when we adjust multimedia data so that they can be transmit-ted over a best-effort network,and these issues have been addressed in different ways by other researchers. Common to all approaches is the existence of a?lter that removes data as needed.We now review some re-lated work classi?ed according to the following criteria:(i)location of the?lter,(ii)type of?ltering applied, (iii)error recovery scheme,and(iv)adaptation algorithm.
Location of?lter A?lter can be placed either in the network or in the end-system.Bhattacharjee et al.[3], Yeadon et al.[15],and Amir et al.[2]present?lters for video data which are located in the net-work.Since different applications require different?ltering strategies,such network nodes need some knowledge about the type of data being?ltered.Also,strategies that allow a client to?nd out about the location of these nodes have to be developed.RTP[13]proposes“mixers”and“translaters”, which are placed in the network.The former ones mix streams and perform conversion between encoding formats,the latter ones translate across transport protocols(e.g.,tunneling of a multicast stream into several unicast streams).Berkeley’s Continuous Media Player[11],OGI’s distributed video player[5],and the V osaic player[7]use the end-system to?lter video data:Frames can be dropped either at the sender(in case of a shortage of network resources)or at the receiver(in case timely display is impossible).
Type of?ltering There are several ways to?lter a video stream and to reduce its bandwidth:Frame-dropping[3][15][7][11][5],low-pass?ltering[15],color reduction[15],re-quantization[15][6],and transcoding[15][2]Another approach is hierarchical?ltering:The layering coding scheme for MPEG presented by Li et al.[10]multicasts three video streams.Each receiver subscribes to the base stream consisting of the I-frames.Depending on its capabilities,a receiver can additionally subscribe to the stream transmitting the P-frames or even to the stream containing the B-frames.
Error recovery Lost data packets can be ignored,retransmitted,or recovered by a Forward Error Correc-tion(FEC)scheme.The?lter by Yeadon et al.[15]and OGI’s player[5]ignore lost packets.The Continuous Media Player[11]pursues the second strategy by employing Cyclic UDP[14],which re-transmits lost high priority data(i.e.,I-frames in the case of MPEG video)to give them a better chance to reach the destination.In the V osaic player[7]and in Columbia’s V oD testbed[6],a client can de-mand retransmission of a lost frame.FEC is applied by Nonnenmacher et al.[12],where requests for retransmissions are handled by FEC transmissions.
Adaptation algorithms The V osaic player[7]continually measures the rate of frames dropped by the re-ceiver due to missing CPU power.If this rate exceeds15%or falls below5%,the server is instructed to lower,respectively to increase,the frame rate.To cope with network congestion,the rate of frames dropped by the network is also measured and fed back to the server every30frames.
Li et al.’s player[10]also uses two thresholds to decide whether a client subscribes to an additional multicast layer or whether it drops one of them.The decision is made after receiving a GOP.In addition to the packet loss ratio,the number of late frames is also taken into account.
In Columbia’s V oD testbed[6],the occupancy of the sender buffer is measured over?ve or ten seconds intervals.In this way,momentary?uctuations due to the varying sizes of the different MPEG frame types can be overcome.The current occupancy is compared to the occupancy from the previous measurement.If necessary,the bitrate of the movie is adapted.Special care is taken to achieve convergence and to avoid oscillations around the desired rate.
In OGI’s player[5],every component(server,network,client)can drop frames in case of missing ressources.Additionally,the(?ltered)display frame rate at the receiver is compared to the sending frame rate.If the difference is large,the sending frame rate is(linearly)decreased.In case of a small difference,the rate is(linearly)increased.
All of the projects described so far deal either only with video data or they transmit video and audio in two separate streams,thus requiring additional synchronization information for their playback at the receiver.Transmitting video and audio data in one stream,as supported by the concept of MPEG systems, has not yet been reported;this approach is at the core of the system that is described in this paper.
4Architecture
In a typical video streaming application,there are two elementary components:the client requesting the video and the server providing it.Typically,the server responds to requests from multiple clients.Our goal is providing the client with the best possible video stream under the current network conditions.
4.1Overview
Since the video server may be too busy to handle the computation required to adapt the MPEG system to network conditions,we place the adaptation(?ltering of data to meet a given resource bound)on other nodes in the network,as depicted in Figure1.
Intermediate nodes labelled’R’designate routers that have no knowledge of the application data and that may randomly throw packets away.The’Filter’nodes implement?ltering by watching a known port, intercepting the client’s requests and adapting the stream sent by the server accordingly.This solution allows us to place the?lter on the nodes that connect to network bottlenecks,and there can be multiple?lters along the path from a server to a client.
The?lter responsibilities can be summarized as follows:
1.receive video stream from server or previous?lter;
Figure1:Filters in the network.
2.send video to client or next?lter;
3.receive requests from client or next?lter;
4.act upon requests or forward them to previous?lter.
4.2Server-client interaction
Requests for changes in the quality of the transmitted movie must originate at the client side.Ultimately the person(s)watching the video determine(s)the quality of the video,and a user(viewer)could be given the capability to interact with the client(player),i.e.,requests to increase or reduce the stream are made by the user.Unfortunately,perception-based models are computationally expensive.Since the client is already busy with decoding and rendering an MPEG stream,we need a non-intrusive method that is also computationally cheap to determine when and how the MPEG stream should be?ltered.We choose to monitor the network traf?c while receiving the network packets and devised an algorithm that adjusts the degree of?ltering to the current packet loss rate.This algorithm is described in Section5.3.4.
4.3Filters
When modifying an MPEG system,an important requirement is that the modi?cations are done on the?y. Encoding MPEG systems requires typically two passes.We need the?lter to be able to remove a selection of video frames,such that the reduced stream is still an MPEG stream,without fully decoding and re-encoding the incoming stream.The algorithms we devised are described in Section5.2.2.The?ltering of each level of the MPEG system is handled by a different algorithm,to decouple the frame-dropping mechanisms as much as possible from the module that selects the policy for reduction or increase of the data rate.The client sends,as mentioned in Section4.2,to the?lter a request to either increase or decrease the bandwidth,and the?lter picks the appropriate level of?ltering based on the current level of?ltering and the direction of the change request(increase or decrease of data).
Since the?lter must analyze the MPEG stream,we make the?lter responsible for pacing the packets according to the required movie bandwidth.
To correctly analyze and?lter the MPEG system,the?lter needs a correct stream at its input.While the ?lter should be able to recover from errors since it can be cascadable to other?lters,having the?rst?lter analyze an uncorrupted MPEG stream improves reliability.Hence the connection between the server and the?rst?lter must be reliable.To satisfy this condition,it may be necessary to place a?lter in proximity to the server.The actual placement,however,will depend on the network typical traf?c.
Figure2:Channels of communication in the network.
There are three main components of our architecture:the client,the server,and at least one instance of a?lter,and we maintain two channels between them,one for control and the other for data,as indicated by Figure2.
The same channels exist between any pair of?lters.The control channel is bidirectional,realized by TCP connections.The requirement that?lters are chainable?ts well into the current WWW browser environment that uses Java.Java applets are restricted to talk only to the host from where the applet originates.Our setup allows the client to communicate with only one host(the last?lter),which will either forward the requests or act upon them.
5Implementation
We now describe the two major parts of our implementation.The?lter reshapes the MPEG system and is described in Section5.2.The overall protocol is described in Section5.3.
5.1Client
The client is based on the Java Media Framework(JMF),a package that supports the replay of audio and video streams in a browser.JMF supports a wide range of video and audio formats,including MPEG-1 systems,and promises to be a widely used package to display multimedia material.
JMF consists of two main components,a player and a data source.The player is responsible for replay-ing the audio and/or video stream and it is typically optimized for a particular platform,e.g.,it may use optimized native methods to deal with speci?c devices.The source is responsible for retrieving the data. Data sources exist that retrieve data from disk,or retrieve data over the network using a variety of proto-cols.The transport protocol described in Section5.3is implemented as a new data source.The system we describe has been used with JMF implementations from Intel and Sun.
5.2Filter
The?lter needs to reduce the bandwidth required by the MPEG stream.Since MPEG is already a com-pressed representation of a movie(video+audio),reducing the data directly impacts the quality.Since most of the data in the MPEG stream is used to encode video,our goal is to develop a?lter that can reduce the video from an MPEG system while maintaining the audio and synchronization information.Additional re-quirements are that the?lter formats the streams as valid MPEG-1systems so we can use a standard player, that the video stream remains smooth,and that multiple levels of reduction are possible so we can better adapt to the available bandwidth.In this section,we describe a software?lter that meets these goals and is ef?cient enough that it can be executed on a PC.
5.2.1Filter operation
The idea behind the?lter is to partially decode the MPEG system so we can identify the different video frame types,the audio information,and the synchronization information.We then drop some of the video
frames,as described in the next section,and we reassemble a new MPEG stream,making sure that the audio and video are appropriately synchronized.Data reduction is done on the?y:the?lter decodes data as it arrives and newly encoded data is immediately forwarded to the receiver.
The partial decoding of the incoming stream is based on a state machine.The states are obtained by identifying the MPEG unique start codes.The state machine maintains state across all the MPEG layers. This fact is important since the video sequence is broken into MPEG packets without considering frame boundaries.This state machine tracks all the audio and video streams simultaneously.When a frame of a particular video stream is detected,the?lter checks whether it should be dropped or forwarded.Then the ?lter searches for the next start code,which is either in a higher layer(GOP,sequence header,packet or pack)or in the same layer(frame).With this method,there can be empty MPEG packets,and even empty GOPs,but keeping these empty packets provides the important bene?t that synchronization is maintained, and the client can continue to decode and render correctly.
5.2.2Selecting the frames to drop
The next question is what frames should be dropped.While each displayed frame gets the same amount of playing time and is thus equally important,there is a big difference in both the size and information contents of each of the encoded frames.These differences are a consequence of the inter-frame encoding.I-pictures are the largest and the“richest”,while B-pictures are the smallest and contain the least information. While the relative sizes suggest that we should drop I-pictures?rst,the information contents and inter-frame dependencies make this of course impractical.The?lter must?rst drop B-pictures.If that action is not suf?cient,the?lter drops P-pictures as well,and in the worst case,it will drop all B-and P-pictures as well as some I-pictures.Note that the?rst levels of reduction do not reduce the data bit rate signi?cantly since B-pictures contain the least amount of physical data.E.g.,if we look at a video sequence that has15frames in a GOP
then removing the B-frames leaves us with sequence
This transformation reduces the frame rate by a factor of three(66%),but typically removes only15%-25% of the data.
In addition to the frame rate,the?lter must consider the smoothness of the video stream,i.e.,we want to prevent“stop and go”jerkiness.This goal can be achieved by distributing the dropped frames as evenly as possible,although the precise placement of the different frame types restricts what frames can be dropped at each reduction level.E.g.,if the current reduced video sequence is IPPPP,the only additional frame that can be dropped without affecting others is the last P-frame.
The resulting algorithms are as follows:Based on the information in the beginning of the MPEG stream, the?lter can deduce the pattern of I,P,and B frames.The drop patterns for different levels are evaluated to conform with the dependencies and to provide smoothness.Since B-frames do not have dependencies,the ?rst algorithm removes the middle B-frame from each group of contiguous B-frames.The second algorithm removes two B-frames by selecting equally spaced frames.The third algorithm removes all the B-frames.
E.g.,in a GOP consisting of the following sequence:
the frames in parenthesis represent the ones dropped for the?rst level.The same GOP in the second level will result in the following frames dropped:
When all B-frames are being dropped,the bandwidth is further reduced by dropping P-frames.The next P-frame to be dropped is always the last one before the next I-frame.When all B-and P-frames are gone,we start dropping I-frames.At this point,the quality of the video typically degrades signi?cantly.If we remove one of two I-frames(above all B-and P-frames),we remain with a video rate of1frame per second.The algorithms can remove further I-frames,distancing them apart as much as needed to achieve the required bandwidth.The resulting stream looks like a slideshow at this point.The maximum reduction will leave an audio-only stream.(In practice,we found out that most players are not able to play an MPEG system stream that is de?ned to have both audio and video when the video stream is competely removed.) After removing frames,the network packets are built in a way to minimize the impact of network packet loss for the client.We require that each MPEG packet starts at the beginning of a network packet.
5.3Transport Protocol
We developed a datagram protocol to distribute MPEG systems over an unreliable network.The?lter receives feedback from the client and adapts its frame drop rate to the network condition between the?lter and the client.
5.3.1Transmission mechanism
We maintain two open connections between the client and the?lter,as well as between the?lter and server. The control connection is used exchange control information and it is always based on TCP.The data con-nection is used to transfer the MPEG data.Data is transferred between the?lter and the client using UDP, as we described above.For the server-?lter connection,we use TCP.The reason is that we expect the?lter to be placed before the bottleneck,so there should be suf?cient bandwidth available between the server and?lter.Packet loss and timeouts should therefore not be a concern,and occasional packet loss can be handled without a problem by having enough buffered data.Moreover,this setup allows us to focus on the ?lter-client protocol.
5.3.2Data streaming
At the?lter,large MPEG systems packets generated by the?ltering process are broken into smaller packets that?t into a network packet.Each packet gets a sequence number.Additionally,the current drop rate of the?lter is also included in the packet header.This information is required by the adaptation algorithm.
The?lter sends each packet at exactly the same time as the corresponding data packet from the non-?ltered movie would have been sent.This rate is determined by the MPEG bitrate of the movie.In our experiments,only movies with constant bitrates have been considered.
5.3.3Control protocol
The control channels are used for a variety of purposes.First,they are used by the client to request video clips and other information.As explained above,the browser-based client can have only a single open con-nection so all requests are sent over the same channel.Requests for the server are forwarded by intermediate nodes.Second,the control channel is used to carry the feedback that is used in the adaptation algorithm from the client to the?lter.Finally,the JMF player does not play movies that have losses in the?rst few Kbytes.For this reason,the?lter transmits the?rst couple of GOPs(typically24or30frames)to the client over the control channel.
The control packets contain an opcode that identi?es the request or information type,plus parameters. Note that there are clearly other ways of managing the control channel.Some of the control information could for example be sent using RTP[13].
5.3.4Adaptation algorithm
The receiver can change the bandwidth of the MPEG stream by sending requests to the ?lter to increase or decrease the frame drop rate.To decide whether the ?lter should increase or decrease the bandwidth,the receiver continuously measures the current packet loss rate using a sliding window of length S packets.S is typically set to 500in our experiments.If the packet loss rate is higher than a threshold ,the client will ask the ?lter to increase the frame drop rate.The value of should be such that a packet loss rate of still
results in acceptable video quality.A second threshold (
)is used to determine when the frame drop rate should be reduced:if the packet drop rate is less than ,the receiver will ask the ?lter to lower the frame drop rate.The reason for using two thresholds is to allow the BW recovery to be less aggresive.This way the protocol id more friendly to competing traf?c.Note that the waiting time before the client ?nally issues a request to reduce the frame drop rate is longer than the waiting before issuing a request to increase it.This behavior is somewhat similar to TCP’s congestion control,that is,it reduces bandwidth more aggressively than it increases it.In our experiments,we typically set to 5%and to 1%.
After every request to increase or decrease the frame drop rate,the client temporarily suspends measur-ing the packet loss rate until it is noti?ed by the ?lter that the drop rate change took place.This depends on the number of packets ’en-route’and the frame being processed by the ?lter when it receives the request to change the drop rate.To inform the client about the ?lter response,the header of a data packet includes the active drop rate.
6Status and evaluation
At this point in time,we have experimented only with setups that include a single ?lter (however,the design and protocols support the chained operation described earlier).The ?lter and server are connected by a non-congested network.
Figure 3depicts how the ?lter adapts to network congestion for setup in a testbed.The server is a is 400MHz Pentium II machine (with 264MBbyte),the client a 200MHz Pentium Pro (with 64MByte);the ?lter executes on the server machine.These two systems are connected via a link that can be transparently “loaded”by a traf?c generator to cause congestion for the server-client connection.Until receiving packet #9084,no packets have been lost.During the receipt of packets 9084..9214,4.94%of the bytes have been lost.As can be seen by the overlay ?gure,which depicts the response of the ?lter (in dropping packets),the ?lter responds by successively removing more frames.Eventually the removal of frames is too aggressive and the ?lter responds by removing fewer frames,until it is possible to return to the “no-removal”mode.)r r r r r r r r r r )x x x x x x x x x x x
x x 02
4
6
8
10
12
14
16
1820
03366894094409940104401094011440L o s s R a t e (% B y t e s )D r o p R a t e (% F r a m e s )Packet #
r Loss Rate
x Drop Rate
x x x x x x Figure 3:Adaptation in response to network losses.
Figure 4shows the effectiveness of the ?lter in a real-life setup.The server and client have the same
properties as described above.The two machines are connected via a network with three segments:a local area network connecting to a T1line,the T1line to the of?ce of the local telephone company,and a synchronous DSL line from the phone company to a residence.The DSL line is the critical link,its maximum UDP throughput was measured to be 649Mbit/sec.About 95%of this maximum bandwidth was available for the connection between ?lter (server)and client.For this experiment,several times an MPEG-1movie with a bandwidth requirement of 1.07Mbit/sec is transmitted from server to client.Figure 4depicts the average transmission rate (measured at sender)and the receive rate (measured at the client).With adaptation,the sender stays close to the maximum bandwidth of the critical link.Without adaptation,the sender rate is unconstrained (the sender is connected to a local area network),but a large portion of the data is dropped along the way to the client.The randomly dropped packets without adaptation damage
R a t e [K b i t s /s e c ]Figure 4:Effect of adaptation on network load.
more frames than are suppressed by adaptation resulting in a very low quality movie.Figure 4shows that the receive rate is almost similar in both cases.However the ?ltered data received with adaptation contains much more usable information,where usable information is measured in terms of the amount of good frames that can be decoded at the client.
7Concluding remarks
The MPEG system format presents a number of challenges to a system that attempts to deliver MPEG streams over a best effort network.However,the complexity of the MPEG system exists for good reasons (to meet the requirements of many users of digital video)and given the amount of data available in this format,supporting the delivery of movies or newsclips in the MPEG system format is attractive.
One of the serious problems that must be addressed in a best-effort network is how to deal with conges-tion.The network layer is unable to understand the intricacies of the hierarchical MPEG system format,so we decided to implement an MPEG ?lter that transforms the MPEG streams to meet resource constraints.Such a ?lter processes a set of synchronized video and audio streams and removes frames as is necessary to deal with the network conditions.A careful design of the ?lter avoids expensive operations in the ?lter host and a current mid-range PC suf?ces as a host.Such a ?lter is well able to adjust the resource demands in a timely manner,and the overall success of this architecture has encouraged us to begin a limited wide-area deployment.
Although we have an operational system,many more tasks remain to be done.We are currently investi-gating the behavior in a inter-continental setting since past studies indicated that the trans-atlantic links are more congested that the intra-continental links.Furthermore,a detailed analysis of the response behavior of
the?lter is needed.
References
[1]ISO/IEC JTC1/SC29/N071.Coding of moving pictures and associated audio-for digital storage media at
up to about1.5Mbits/s-Part1:Systems,Part2:video,1992.CD11172.
[2]E.Amir,S.McCanne,and R.Katz.An Active Service Framework and its Application to Real-time Multimedia
Transcoding.In Proceedings of ACM SIGCOMM’98,pages178–189,V ancouver,Canada,September1998. [3]S.Bhattacharjee,K.L.Calvert,and E.W.Zegura.On Active Networking and Congestion.Technical Report
GIT-CC-96/02,Georgia Institute of Technology,1996.
[4]J.M.Boyce and R.D.Gaglianello.Packet loss effects on MPEG video sent over the public Internet.In Proceed-
ings of ACM MULTIMEDIA’98,pages181–190,Bristol,England,Sept1998.
[5]S.Cen,C.Pu,R.Staehli,C.Cowan,and J.Walpole.A Distributed Real-Time MPEG Video Audio Player.In
Proceedings of NOSSDAV’95,pages18–21,Durham,New Hampshire,April1995.
[6]S.-F.Chang,D.Eleftheriadis,D.Anastassiou,S.Jacobs,H.Kalva,and J.Zamora.Columbia’s VOD and Multi-
media Research Testbed With Heterogeneous Network Support.Journal on Multimedia Tools and Applications, 5(2):171–184,Sept1997.
[7]Z.Chen,S.-M.Tan,R.H.Campbell,and Y.Li.Real Time Video and Audio in the World Wide Web.In
Proceedings of Fourth International World Wide Web Conference,Boston,Massachusetts,Dec1995.
[8]M.Christel,T.Kanade,M.Mauldin,R.Reddy,M.Sirbu,S.Stevens,and https://www.doczj.com/doc/b517803893.html,rmedia digital video
https://www.doczj.com/doc/b517803893.html,m.ACM,38(4):57–58,April1995.
[9]Java Multimedia Framework.
[10]X.Li,S.Paul,P.Pancha,and https://www.doczj.com/doc/b517803893.html,yered Video Multicast with Retransmission(LVMR):Evaluation of
Error Recovery Schemes.In Proceedings of NOSSDAV’97,St.Louis,Missouri,May1997.
[11]K.Mayer-Patel and L.A.Rowe.Design and Performance of the Berkeley Continuous Media Toolkit.In SPIE
Proceedings V ol.3020,pages194–2006,San Jose,California,Feb1997.
[12]J.Nonnenmacher,E.Biersack,and D.Towsley.Parity-Based Loss Recovery for Reliable Multicast Transmission.
In Proceedings of ACM SIGCOMM’97,pages298–300,Cannes,France,September1997.
[13]H.Schulzrinne,S.Casner,R.Frederick,and V.Jacobson.RFC1889:RTP:A Transport Protocol for Real-Time
Applications,January1996.
[14]B.C.Smith.Implementation Techniques for Continuous Media Systems and Applications.PhD thesis,University
of California at Berkeley,1994.
[15]N.Yeadon,F.Garcia,D.Hutchison,and D.Shepherd.Filters:QoS Support Mechanisms for Multipeer Commu-
nications.IEEE Journal on Selected Areas in Communications,14(7):1245–1262,Sept1996.
应用ADS设计微带线带通滤波器 1、微带带通微带线的基本知识 微波带通滤波器是应用广泛、结构类型繁多的微波滤波器，但适合微带结构的带通滤波器结构就不是那么多了，这是由于微带线本身的局限性，因为微带结构是个平面电路，中心导带必须制作在一个平面基片上，这样所有的具有串联短截线的滤波器都不能用微带结构来实现；其次在微带结构中短路端不易实现和精确控制，因而所有具有短路短截线和谐振器的滤波器也不太适合于微带结构。 微带线带通滤波器的电路结构的主要形式有5种： 1、电容间隙耦合滤波器 带宽较窄，在微波低端上显得太长，不够紧凑，在2GHz以上有辐射损耗。 2、平行耦合微带线带通滤波器 窄带滤波器，有5％到25％的相对带宽，能够精确设计，常为人们所乐用。但其在微波低端显得过长，结构不够紧凑；在频带较宽时耦合间隙较小，实现比较困难。 3、发夹线带通滤波器 把耦合微带线谐振器折迭成发夹形式而成。这种滤波器由于容易激起表面波，性能不够理想，故常把它与耦合谐振器混合来用，以防止表面波的直接耦合。这种滤波器的精确设计较难。
4、1/4波长短路短截线滤波器 5、半波长开路短截线滤波器 下面主要介绍平行耦合微带线带通滤波器的设计，这里只对其整个设计过程和方法进行简单的介绍。 2、平行耦合线微带带通滤波器 平行耦合线微带带通滤波器是由几节半波长谐振器组合而成的，它不要求对地连接，结构简单，易于实现，是一种应用广泛的滤波器。整个电路可以印制在很薄的介质基片上(可以簿到1mm以下)，故其横截面尺寸比波导、同轴线结构的小得多；其纵向尺寸虽和工作波长可以比拟，但采用高介电常数的介质基片，使线上的波长比自由空间小了几倍，同样可以减小；此外，整个微带电路元件共用接地板，只需由导体带条构成电路图形，结构大为紧凑，从而大大减小了体积和重量。 关于平行耦合线微带带通滤波器的设计方法，已有不少资料予以介绍。但是，在设计过程中发现，到目前为止所查阅到的各种文献，还没有一种能够做到准确设计。在经典的工程设计中，为避免繁杂的运算，一般只采用简化公式并查阅图表，这就造成较大的误差。而使用电子计算机进行辅助设计时，则可以力求数学模型精确，而不追求过分的简化。基于实际设计的需要，我对于平行耦合线微带
实验八 有源滤波器的设计 一．实验目的 1． 学习有源滤波器的设计方法。 2． 掌握有源滤波器的安装与调试方法。 3． 了解电阻、电容和Q 值对滤波器性能的影响。 二．预习要求 1． 根据滤波器的技术指标要求，选用滤波器电路，计算电路中各元件的数值。设计出 满足技术指标要求的滤波器。 2. 根据设计与计算的结果，写出设计报告。 3. 制定出实验方案，选择实验用的仪器设备。 三．设计方法 有源滤波器的形式有好几种，下面只介绍具有巴特沃斯响应的二阶滤波器的设计。 巴特沃斯低通滤波器的幅频特性为： n c uo u A j A 21)(??? ? ??+= ωωω ， n=1，2，3，. . . （1） 写成： n c uo u A j A 211) (??? ? ??+=ωωω （2） )(ωj A u 其中A uo 为通带内的电压放大倍数，ωC A uo 为截止角频率，n 称为滤波器的阶。从（2） 式中可知，当ω=0时，（2）式有最大值1； 0.707A uo ω=ωC 时，（2）式等于0.707，即A u 衰减了3dB ；n 取得越大，随着ω的增加，滤波器的输出电压衰减越快，滤波器的幅频特性越接近于理想特性。如图1所示。ω 当 ω>>ωC 时， n c uo u A j A ??? ? ??≈ωωω1 )( （3） 图1低通滤波器的幅频特性曲线
两边取对数，得： lg 20c uo u n A j A ωω ωlg 20)(-≈ （4） 此时阻带衰减速率为： -20ndB/十倍频或-6ndB/倍频，该式称为衰减估算式。 表1列出了归一化的、n 为1 ~ 8阶的巴特沃斯低通滤波器传递函数的分母多项式。 在表1的归一化巴特沃斯低通滤波器传递函数的分母多项式中，S L = c s ω，ωC 是低通 滤波器的截止频率。 对于一阶低通滤波器，其传递函数： c c uo u s A s A ωω+= )( （5） 归一化的传递函数： 1 )(+= L uo L u s A s A （6） 对于二阶低通滤波器，其传递函数：2 22)(c c c uo u s Q s A s A ωωω++ = （7） 归一化后的传递函数： 1 1)(2 ++= L L uo L u s Q s A s A （8） 由表1可以看出，任何高阶滤波器都可由一阶和二阶滤波器级联而成。对于n 为偶数的高阶滤波器，可以由2n 节二阶滤波器级联而成；而n 为奇数的高阶滤波器可以由2 1-n 节二
滤波器是一种只传输指定频段信号，抑制其它频段信号的电路。 滤波器分为无源滤波器与有源滤波器两种： ①无源滤波器: 由电感L、电容C及电阻R等无源元件组成 ②有源滤波器: 一般由集成运放与RC网络构成，它具有体积小、性能稳定等优点，同时，由于集成运放的增益和输入阻抗都很高，输出阻抗很低，故有源滤波器还兼有放大与缓冲作用。 利用有源滤波器可以突出有用频率的信号，衰减无用频率的信号，抑制干扰和噪声，以达到提高信噪比或选频的目的，因而有源滤波器被广泛应用于通信、测量及控制技术中的小信号处理。 从功能来上有源滤波器分为： 低通滤波器（LPF）、高通滤波器（HPF）、 带通滤波器（BPF）、带阻滤波器（BEF）、 全通滤波器（APF）。 其中前四种滤波器间互有联系，LPF与HPF间互为对偶关系。当LPF的通带截止频率高于HPF的通带截止频率时，将LPF与HPF相串联，就构成了BPF，而LPF与HPF并联，就构成BEF。在实用电子电路中，还可能同时采用几种不同型式的滤波电路。滤波电路的主要性能指标有通带电压放大倍数AVP、通带截止频率fP及阻尼系数Q等。 带通滤波器（BPF） （a）电路图(b)幅频特性 图1 压控电压源二阶带通滤波器 工作原理：这种滤波器的作用是只允许在某一个通频带范围内的信号通过，而比通频带下限频率低和比上限频率高的信号均加以衰减或抑制。典型的带通滤波器可以从二阶低通滤波器中将其中一级改成高通而成。如图1（a）所示。 电路性能参数 通带增益 中心频率 通带宽度 选择性 此电路的优点是改变Rf和R4的比例就可改变频宽而不影响中心频率。 例．要求设计一个有源二阶带通滤波器，指标要求为： 通带中心频率 通带中心频率处的电压放大倍数： 带宽： 设计步骤： 1）选用图2电路。 2）该电路的传输函数： 品质因数： 通带的中心角频率： 通带中心角频率处的电压放大倍数： 取，则：
带通滤波器设计步骤 1、根据需求选择合适的低通滤波器原型 2、把带通滤波器带宽作为低通滤波器的截止频率，根据抑制点的频率距离带通滤波器中心频点距离的两倍作为需要抑制的频率，换算抑制频率与截止频率的比值，得出m 的值，然后根据m 值选择低通滤波器的原型参数值。 滤波器的时域特性 任何信号通过滤波器都会产生时延。Bessel filter 是特殊的滤波器在于对于通带内的所有频率而言，引入的时延都是恒定的。这就意味着相对于输入，输出信号的相位变化与工作的频率是成比例的。而其他类型的滤波器（如Butterworth, Chebyshev,inverse Chebyshev,and Causer ）在输出信号中引入的相位变化与频率不成比例。相位随频率变化的速率称之为群延迟（group delay ）。群延迟随滤波器级数的增加而增加。 模拟滤波器的归一化 归一化的滤波器是通带截止频率为w=1radian/s, 也就是1/2πHz 或约0.159Hz 。这主要是因为电抗元件在1弧度的时候，描述比较简单，XL=L, XC=1/C ，计算也可以大大简化。归一化的无源滤波器的特征阻抗为1欧姆。归一化的理由就是简化计算。 Bessel filter 特征：通带平坦，阻带具有微小的起伏。阻带的衰减相对缓慢，直到原理截止频率高次谐波点的地方。原理截止频率点的衰减具有的经验公式为n*6dB/octave ，其中，n 表示滤波器的阶数，octave 表示是频率的加倍。例如，3阶滤波器，将有18dB/octave 的衰减变化。正是由于在截止频率的缓慢变化，使得它有较好的时域响应。 Bessel 响应的本质截止频率是在与能够给出1s 延迟的点，这个点依赖于滤波器的阶数。 逆切比雪夫LPF 原型参数计算公式（Inverse Chebyshev filter parameters calculate equiations ） ) (cosh )(cosh 11Ω=--Cn n 其中 1101.0-=A Cn , A 为抑制频率点的衰减值，以dB 为单位；Ω为抑制频率与截止频率的比值 例：假设LPF 的3dB 截止频率为10Hz,在15Hz 的频点需要抑制20dB,则有： 95.91020*1.0==Cn ；Ω=15/10=1.5 1.39624.0988.2) 5.1(cosh )95.9(cosh 11===--n ,因此，滤波器的阶数至少应该为4
RC 有源带通滤波器的设计 滤波器的功能是让一定频率范围内的信号通过，而将此频率范围之外的信号加以抑制或使其急剧衰 减。当干 扰信号与有用信号不在同一频率范围之内，可使用滤波器有效的抑制干扰。 用LC 网络组成的无源滤波器在低频范围内有体积重量大，价格昂贵和衰减大等缺点，而用集成运放 和RC 网络组成的有源滤波器则比较适用于低频，此外，它还具有一定的增益，且因输入与输出之间有良 好的隔离而便于级联。由于大多数反映生理信息的光电信号具有频率低、幅度小、易受干扰等特点，因而 RC 有源滤波器普遍应用于光电弱信号检测电路中。 一.技术指标 总增益为1 ; 通带频率范围为 300Hz —3000Hz ,通带内允许的最大波动为 -1db —+1db ; 阻带边缘频率范围为 225Hz 和4000Hz 、阻带内最小衰减为 20db ; 二?设计过程 1 .采用低通-高通级联实现带通滤波器； 将带通滤波器的技术指标分成低通滤波器和高通滤波器两个独立的技术指标，分别设计出低通滤波器 和高通 滤波器，再级联即得带通滤波器。 低通滤波器的技术指标为： f PH = 3000Hz A max - 1d B G =1 f SH = 4000Hz A min = 20dB 高通滤波器的技术指标为： f pL = 300Hz A max = 1d B G = 1 f si_ - 225Hz A min - 20dB 2. 选用切比雪夫逼近方式计算阶数 (1).低通滤波器阶数 N >ch 4[J(10 0.1Amin -1)/(10 0.1Ami N 1 _ ■ 1 Ch ( f SH / f PH ) (2).高通滤波器阶数 N 2 ch'[ *. (10 0.1Amin -1)/(100.1Amax -1)] Ch^(f pL /f SL ) 3. 求滤波器的传递函数 1) .根据Ni 查表求出归一化低通滤波器传递函数 H LP (S)二 H LP (S)| S S' 2= --- 2冗PH 2) .根据Na 查表求出归一化高通滤波器传递函数 N 2 H_P (S '，去归一化得 H^s '，去归一化得
范道中学七年级数学导学提纲课题：幂的乘方 出卷人：施培新审核人：陈益锋 2012-2-22 姓名 _____ 课前参与 （一）预习内容：课本P43—44 （二）知识整理： 1.探索： （1）（2）是幂2的_____次方,其意义是_____个2的连乘积， 可写成：（2）=2×2=2= 2=2。 （2）（a）是幂a的_____次方,其意义是____个a的连乘积， 可写成：（a）=（）×（）×（）= a= a= a； 由此得：（a）是幂a的______次方,其意义是______个a的连乘积， 可写成：（a）=（）=a=a。 2.归纳：幂的乘方的法则：__________________________________________； 即写成公式： （a）=a（m、n为正整数）。 3.尝试练习： （1）（10）= （5）（－5）= （2）（10）= （6）（－5）= （3）（b）= (7 [（n－m）] 5 （4）（b）= (8 a·（a）2+ a·（a）3
4.推广：[（a m ）n ]p =____________ (m 、n 、p 为正整数。 5.幂的乘方法则的逆用为___________________________。 （三）思考： 通过预习，你认为本节内容主要研究了什么？你还有什么问题需和大家一起探讨？你有没有新的发现和大家一起分享！ 课中参与 例题1、计算：（1）（55）3 （2）（53）5 （3）(3x 5 （4）(35 x 例题2、计算：（1）[（a －b ）] （2）[（x －y ）] 例题3、计算：（1）－（y 4）3 （2）[（－y ）4]3 （3）（－y 4）3 例题4、计算：（1）（a ）·a （2）（b ）·（b ） （3）a ·（a ）－a ·（a ）2 拓展：1、（1）[（2）] （2）[（－3）] 2、已知3=2，3y =3，求（1）33x ，3 2y 的值。 （2）求3的值. 3、已知：3=a ，3=b ，用含a 、b 的代数式表示3 。 课后参与 课题：幂的乘方 姓名_____ 一、填空： （1）（7）5=_________； （2）[(－22]3=_________； （3） （a ） =________； （4）（－a 5）3=_________； （5）[（a －2）]=________； （6）[（x －y ）]=______；
电子系统设计实验报告 姓名 指导教师 专业班级 学院 提交日期2011年11月1日
目录 第一章设计题目 (1) 1.1 设计任务 (1) 1.2 设计要求 (1) 第二章原理分析及参数计算 (1) 2.1 总方案设计 (1) 2.1.1 方案框图 (1) 2.1.2 原理图设计 (1) 2.2 单元电路的设计及参数计算 (2) 2.2.1 二阶低通滤波器 (2) 2.2.2 二阶高通滤波器 (3) 2.3 元器件选择 (4) 第三章电路的组装与调试 (5) 3.1 MultiSim电路图 (5) 3.2 MultiSim仿真分析 (5) 3.1.1 四阶低通滤波器 (5) 3.1.2 四阶高通滤波器 (5) 3.1.3 总电路图 (6) 3.3 实际测试结果 (6) 第四章设计总结 (6) 附录………………………………………………………………………………… 附录Ⅰ元件清单………………………………………………………………… 附录Ⅱ Protel原理图…………………………………………………………… 附录Ⅲ PCB图（正面）………………………………………………………… 附录Ⅳ PCB图（反面）………………………………………………………… 参考文献…………………………………………………………………………
第一章 设计题目 1.1 设计任务 采用无限增益多重反馈滤波器，设计一四阶带通滤波器，通带增益01A =， 1L f kHz =，2H f kHz =，设计方案如图1.1所示。 图1.1 四阶带通滤波器方案图 1.2 设计要求 1.用Protel99 画出原理图，计算各元件参数，各元件参数选用标称值； 2.用Mutisum 对电路进行仿真，给出幅频特性的仿真结果； 3.在面包板上搭接实际电路，并测试滤波器的幅频特性； 4.撰写设计报告。 第二章 设计方案 2.1 方案设计 2.1.1方案框图(如图2.1.1) 图2.1.1 四阶带通滤波器总框图 2.1.2原理图设计 本原理图根据结构框图组成了4个二阶滤波器，上面两个分别为c f =2kHz ，Q=0.541，A=1的低通滤波器和c f =2kHz ，Q=1.306，A=1的低通滤波器；下面两个分别为c f =1kHz ，Q=0.541，A=1的高通滤波器和c f =1kHz ，Q=1.306，A=1的高通滤波器，其中P1、P2、P3作为接线座用来接线，原理图如图2.1.2,具体参数计算见2.2节。 V i V o 二阶低通滤波器 二阶低通滤波器 二阶高通滤波器 二阶高通滤波器
实验四微带线带通滤波器 设计 Prepared on 24 November 2020
实验四：基于ADS软件的平行耦合微带线带通滤波器的设计与仿真一、实验原理 滤波器是用来分离不同频率信号的一种器件，在微波电路系统中，滤波器的性能对电路的性能指标有很大的影响，微带电路具有体积小，重量轻、频带宽等诸多优点，在微波电路系统应用广泛，其中用微带做滤波器是其主要应用之一。平行耦合微带线带通滤波器在微波集成电路中是被广为应用的带通滤波器。 1、滤波器的介绍 滤波波器可以分为四种：低通滤波器和高通滤波器、带通滤波器和带阻滤波器。射频滤波器又可以分为以下波导滤波器、同轴线滤波器、带状线滤波器、微带滤波器。 滤波的性能指标： 频率范围：滤波器通过或截断信号的频率界限 通带衰减：滤波器残存的反射以及滤波器元件的损耗引起 阻带衰减：取通带外与截止频率为一定比值的某频率的衰减值 寄生通带：有分布参数的频率周期性引起，在通带外又产生新的通带 2、平行耦合微带线滤波器的理论 当频率达到或接近GHz时，滤波器通常由分布参数元件构成，平行耦合微带传输线由两个无屏蔽的平行微带传输线紧靠在一起构成，由于两个传输线之间电磁场的相互作用，在两个传输线之间会有功率耦合，这种传输线也因此称为耦合传输线。 平行耦合微带线可以构成带通滤波器，这种滤波器是由四分之一波长耦合线段构成，她是一种常用的分布参数带通滤波器。 当两个无屏蔽的传输线紧靠一起时，由于传输线之间电磁场的相互作用，在传输线之间会有功率耦合，这种传输线称之为耦合传输线。根据传输线理论，每条单独的微带线都等价为小段串联电感和小段并联电容。每条微带线的特性阻抗为Z0，相互耦合的部分长度为L，微带线的宽度为W，微带线之间的距离为S，偶模特性阻抗为Z e，奇模特性阻抗为Z0。单个微带线单元虽然具有滤波特性，但其不能提供陡峭的通带到阻带的过渡。 如果将多个单元级联，级联后的网络可以具有良好的滤波特性。 二、耦合微带线滤波器的设计的流程
课程设计 课程名称：数字信号处理 题目编号： 0801 题目名称：切比雪夫I型带通IIR数字滤波器 专业名称：电子信息工程 班级： 1203班 学号： 学生姓名：段超 任课教师：陈忠泽 2015年08月30日
目 录 1. 数字滤波器的设计任务及要求（编号0801） ................... 2 2. 数字滤波器的设计及仿真 .. (2) 2.1数字滤波器（编号0801）的设计 ................................... 2 2.2数字滤波器（编号0801）的性能分析 . (4) 3. 数字滤波器的实现结构对其性能影响的分析 (6) 3.1数字滤波器的实现结构一（0801）及其幅频响应 ...................... 7 3.2数字滤波器的实现结构二（0801）及其幅频响应 ...................... 9 3.3 数字滤波器的实现结构对其性能影响的小结 (12) 4. 数字滤波器的参数字长对其性能影响的分析 (12) 4.1数字滤波器的实现结构一（0801）参数字长及幅频响应特性变化 ...... 14 4.2数字滤波器的实现结构二（0801）参数字长及幅频响应特性变化 ....... 17 4.3 数字滤波器的参数字长对其性能影响的小结 (20) 5. 结论及体会 (20) 5.1 滤波器设计、分析结论 .......................................... 21 5.2 我的体会 ...................................................... 21 5.3 展望 . (21) 1. 数字滤波器的设计任务及要求 （1）切比雪夫I 型带通IIR 数字滤波器各项指标如下： 阻带下截止频率1s ω=rad i d π32 ) ln(; 通带下截止频率1p ω=rad i i d d π???? ?? +20 )(32)ln(log 10 ; 通带上截止频率2p ω=rad i i d d π??? ? ? ?-20 )(32 )ln(-1log 10 ;
NANHUA University
课程设计(论文)
题
目
用切比雪夫Ⅱ型 IIR 设计带通（数字频带变换）滤波器
学院名称 指导教师 班 学 级 号
电 气 工 程 学 院 陈 忠 泽
电子 091 20094470121
学生姓名
潘星
2012 年
12
月
1
1．课程设计的内容和要求（包括原始数据、技术要求、工作要求等） ：
1.设计内容： 根据自己在班里的学号20（最后两位）查表一得到一个四位数，0702，由该四位数 索引表二确定待设计数字滤波器的类型：等波纹FIR数字带通滤波器 2.滤波器的设计指标： ⑴阻带下截止频率 ? sl ? e ? 0.2?rad ⑵通带下截止频率 ? pl ? e 50 ? 0.3?rad ⑶通带上截止频率 ? pu ? e
id 50
id 50
id
? 0.7?rad
⑷阻带上截止频率 ?su ? e ? 0.8?rad ⑸通带最大衰减错误！未找到引用源。 ⑹阻带最小衰减错误！未找到引用源。 其中，错误！未找到引用源。—你的学号的最后两位 3. 滤波器的初始设计通过手工计算完成； 4. 在计算机辅助计算基础上分析滤波器结构对其性能指标的影响 （至少选择两种以上合 适的滤波器结构进行分析）； 5. 在计算机辅助计算基础上分析滤波器参数的字长对其性能指标的影响； 6. 以上各项要有理论分析和推导、原程序以及表示计算结果的图表； 7. 课程设计结束时提交设计说明书。
id 50
2
MATLAB设计模拟带通滤波器 参数自己改一下就可以了 cheb1 % wp1=0.45*pi;wp2=0.65*pi;ws1=0.3*pi;ws2=0.75*pi;Rp=1;Rs=40 % =============双线型变换法========================================= wp1=0.45*pi; wp2=0.65*pi; ws1=0.3*pi; ws2=0.75*pi; Rp=1; Rs=40; Wp1=tan(wp1/2); Wp2=tan(wp2/2); Ws1=tan(ws1/2); Ws2=tan(ws2/2); BW=Wp2-Wp1; W0=Wp1*Wp2; W00=sqrt(W0); WP=1; WS=WP*(W0^2-Ws1^2)/(Ws1*BW); [N,Wn]=cheb1ord(WP,WS,Rp,Rs,'s'); [B,A]=cheby1(N,Rp,Wn,'s'); [BT,AT]=lp2bp(B,A,W00,BW); [num,den]=bilinear(BT,AT,0.5); [h,omega]=freqz(num,den,64); subplot(2,2,1);stem(omega/pi,abs(h)); xlabel('\omega/\pi');ylabel('|H(z)|'); subplot(2,2,2);stem(omega/pi,20*log10(abs(h))); xlabel('\omega/\pi');ylabel('增益.dB'); % =============直接法================================= wp1=0.45*pi; wp2=0.65*pi; ws1=0.3*pi; ws2=0.75*pi; Rp=1; Rs=40; Wp=[wp1/pi,wp2/pi]; Ws=[ws1/pi,ws2/pi]; [N,Wn]=cheb1ord(Wp,Ws,Rp,Rs); [B,A]=cheby1(N,Rp,Wn); [h,omega]=freqz(B,A,64); subplot(2,2,3);stem(omega/pi,abs(h)); xlabel('\omega/\pi');ylabel('|H(z)|'); subplot(2,2,4);stem(omega/pi,20*log10(abs(h))); xlabel('\omega/\pi');ylabel('增益.dB'); %cheby2% % wp1=0.45*pi;wp2=0.65*pi;ws1=0.3*pi;ws2=0.75*pi;Rp=1;Rs=40 % =============双线型变换法========================================= wp1=0.45*pi; wp2=0.65*pi; ws1=0.3*pi; ws2=0.75*pi;
￥ 实验八 有源滤波器的设计 一．实验目的 1． 学习有源滤波器的设计方法。 2． 掌握有源滤波器的安装与调试方法。 3． 了解电阻、电容和Q 值对滤波器性能的影响。 ; 二．预习要求 1． 根据滤波器的技术指标要求，选用滤波器电路，计算电路中各元件的数值。设计出 满足技术指标要求的滤波器。 2. 根据设计与计算的结果，写出设计报告。 3. 制定出实验方案，选择实验用的仪器设备。 三．设计方法 ， 有源滤波器的形式有好几种，下面只介绍具有巴特沃斯响应的二阶滤波器的设计。 巴特沃斯低通滤波器的幅频特性为： n c uo u A j A 21)(??? ? ??+= ωωω ， n=1，2，3，. . . （1） 写成： n c uo u A j A 211) (??? ? ??+=ωωω （2） )(ωj A u 其中A uo 为通带内的电压放大倍数， C A uo 为截止角频率，n 称为滤波器的阶。从（2） 式中可知，当=0时，（2）式有最大值1； 。 =C 时，（2）式等于，即A u 衰减了 n=2 3dB ；n 取得越大，随着的增加，滤波器 n=8 的输出电压衰减越快，滤波器的幅频特性 越接近于理想特性。如图1所示。 0 C
当 >> C 时， n c uo u A j A ??? ? ??≈ωωω1 )( （3） 图1低通滤波器的幅频特性曲线 两边取对数，得： lg 20c uo u n A j A ωω ωlg 20)(-≈ （4） 此时阻带衰减速率为： 20ndB/十倍频或6ndB/倍频，该式称为衰减估算式。 [ 表1列出了归一化的、n 为1 ~ 8阶的巴特沃斯低通滤波器传递函数的分母多项式。 n 归一化的巴特沃斯低通滤波器传递函数的分母多项式 1 1+L s 2 122++L L s s 》 3 )1()1(2+?++L L L s s s 4 )184776.1()176537.0(2 2++?++L L L L s s s s 5 )1()161803.1()161807.0(22+?++?++L L L L L s s s s s 6 )193185.1()12()151764.0(222++?++?++L L L L L L s s s s s s [ 7 )1()180194.1()124698.1()144504.0(2 22+?++?++?++L L L L L L L s s s s s s s 8 )196157.1()166294.1()111114.1()139018.0(2222++?++?++?++L L L L L L L L s s s s s s s s 在表1的归一化巴特沃斯低通滤波器传递函数的分母多项式中，S L = c s ω， C 是低通 滤波器的截止频率。 对于一阶低通滤波器，其传递函数： c c uo u s A s A ωω+= )( （5） 归一化的传递函数： 1 )(+= L uo L u s A s A （6） 对于二阶低通滤波器，其传递函数：2 22)(c c c uo u s Q s A s A ωωω++ = （7） >
一、低通滤波器的设计及参数值在线计算 图1所示是一个低通通滤波器，它的截止频率如下公式所示： 公式1 图1 图2是实用的低通滤波器电路，它使用通用运算放大器（运放）接成单电源供电模式，简单易行。图中C2为足够大的电容器，所谓足够大是指C2和R2的时间常数要远小于R1和C1的时间常数，图中为10U。该电路通带内的电压放大倍数为R1/R2，若R1=R2则放大倍数为1。该电路截止频率有R1,C1的时间常数决定，满足公式1。
图2 下图是当R1=R2=15915Ω（不是标准电阻值，可参考这里找出最接近的电阻），C1=10nF（算得频率是1k）的pspice仿真结果。这时增益=1，输出二分之一根 号二即0.707V就是截至频率点，图上可以看出是1kHz
图3 输入C1，R1的值计算频率F: 输入C1，频率F的值计算电阻R1: 低通滤波器的设计及参数值在线计算:https://www.doczj.com/doc/b517803893.html,/lowpass.htm
二、有源带通滤波器的设计及参数值在线计算 图1所示是一个多路负反馈二阶有源带通滤波器，它使用单个通用运算放大器（通用运放）接成单电源供电模式，易于实现。它的上限截止频率和下限截止频率可以非常近，具有非常很强的频率选择性。令C1=C2=C，Req是R1和R2并联的值。品质因数Q等于中心频率除以带宽，Q = fC/BW。由式可以看出可以通过让R3的值远大于Req来获得大的Q值 Q值越大，频率选择性越好，带宽越小。反之则反。令中心频率为fc，则计算公式如下： 其中
关于本有源带通滤波器电路的详细论述及PSPICE仿真结果请访问： 有源带通滤波器 借助本工具软件，您可以： 输入增益GAIN，带宽BW，中心频率F，电容值C，计算有源带通滤波器电阻值R1,R2,R3：
数字信号处理探究性学习 数字滤波器的设计
一、课题目的 1．通过课下的自我的学习，加深对书本理论知识的理解，提升自身的实际应用能力； 2．巩固所学的数字信号处理理论知识，使自身对信号的采集、处理、传输、显示和存储等有一个系统的掌握和理解； 3．培养自我学习的能力和对相关课程的兴趣； 二、课题介绍 传统的数字滤波器的设计过程复杂，计算工作量大，滤波特性调整困难，影响了它的应用。在此介绍一种利用MATLAB信号处理工具箱（Signal Proces程序设计和利用信号处理工具箱的FDATool工具进行界面设计的详细步骤。利用MATLAB设计滤波器，可以随时对比设计sing Toolbox）快速有效的设计由软件组成的常规数字滤波器的设计方法。给出了使用MATLAB语言进行要求和滤波器特性调整参数，直观简便，极大的减轻了工作量，有利于滤波器设计的最优化。此外还介绍了如何利用MATLAB环境下的仿真软件Simulink 对所设计的滤波器进行模拟仿真。 在电力系统微机保护和二次控制中，很多信号的处理与分析都是基于对正弦基波和某些整次谐波的分析，而系统电压电流信号（尤其是故障瞬变过程）中混有各种复杂成分，所以滤波器一直是电力系统二次装置的关键部件。目前微机保护和二次信号处理软件主要采用数字滤波器。传统的数字滤波器设计使用繁琐的公式计算，改变参数后需要重新计算，在设计滤波器尤其是高阶滤波器时工作量很大。利用MATLAB 信号处理工具箱（Signal Processing Toolbox）可以快速有效的实现数字滤波器的设计与仿真。 三、课题内容 用两种方法实现数字滤波器的设计(传统设计方法与Matlab设计)并进行Simulink仿真 四、具体实现 1 数字滤波器及传统设计方法 数字滤波器可以理解为是一个计算程序或算法，将代表输入信号的数字时间序列转化为代表输出信号的数字时间序列，并在转化过程中，使信号按预定的形式变化。数字滤波器有多种分类，根据数字滤波器冲激响应的时域特征，可将数字滤波器分为两种，即无限长冲激响应（IIR）滤波器和有限长冲激响应（FI R）滤波器。 IIR数字滤波器具有无限宽的冲激响应，与模拟滤波器相匹配。所以IIR滤波器的设计可以采取在模拟滤波器设计的基础上进一步变换的方法。FIR数字滤波器的单位脉冲响应是有限长序列。它的设计问题实质上是确定能满足所要求的转移序列或脉冲响应的常数问题，设计方法主要有窗函数法、频率采样法和等波纹最佳逼近法等。
电子与电气工程学院 课程设计报告 课程名称模拟电子技术课程设计设计题目二阶带通滤波器的设计专业名称自动化 班级自动化143班 学号201 学生姓名 指导教师 2016年5月30日
电气学院电子技术课程设计 任务书 设计名称：二阶带通滤波器的设计 学生姓名：指导教师： 起止时间：自2016 年 5 月16 日起至2016 年 5 月30 日止 一、课程设计目的 1.制作一个二阶带通滤波器。 2.用桥式整流电容滤波集成稳压块电路设计电路所需的正负直流电源（±12V）。 二、课程设计任务和基本要求 设计任务： 1.分别用压控电压源和无限增益多路反馈二种方法设计电路； 2.中心频率f O=1KHz； 3.增益A V＝1---2； 4.品质因数Q＝1~2； 5.用桥式整流电容滤波集成稳压块电路设计电路所需的正负直流电源（±12V）。 基本要求： 1.具有放大信号源的作用，能输出相应的波形； 2.能够通过一定频率范围内的信号源。 三、设计目标 设计的二阶带通滤波器能通过一定频率范围内的信号源。当输入幅度为1V、频率小于100Hz或大于8000Hz的正弦信号时，基本不能输出正弦波形，而是幅度很小且不规则的曲线。当输入频率为中心频率周围的正弦信号时，能输出完整且稳定的波形。即二阶带通滤波器有滤波功能。
电气学院电子技术课程设计 指导老师评价表
目录 摘要与关键字........................................................................................................................................ - 1 - 一、二阶带通滤波器的设计要求 .......................................................................................... - 2 - 1.1 设计任务及要求.................................................................................................................. - 2 - 1.1.1基本要求 ........................................................................................................................... - 2 - 1.1.2设计任务 ........................................................................................................................... - 2 - 1.1.3设计目标 ........................................................................................................................... - 2 - 二、电路设计原理及方案 ........................................................................................................... - 2 - 2.1二阶带通滤波器的特点 ................................................................................................... - 2 - 2.2设计原理 ................................................................................................................................... - 2 - 2.3方案设计与论证 ................................................................................................................... - 2 - 三、单元电路设计与参数计算................................................................................................ - 3 - 3.1压控电压源二阶带通滤波电路 ................................................................................... - 3 - 3.2无限增益多路反馈二阶带通电路 .............................................................................. - 5 - 3.3用桥式整流电容滤波集成稳压块电路设计电路所需的正负直流电源 （±12V）.......................................................................................................................................... - 6 - 四、总原理图........................................................................................................................................ - 8 - 4.1总原理图 ................................................................................................................................... - 8 - 4.2元件清单 ................................................................................................................................... - 9 - 五、性能测试与分析.................................................................................................................. - 10 - 5.1直流稳压电源性能测试与分析 ................................................................................. - 10 - 5.2压控电压源二阶带通滤波电路性能测试与分析 ............................................ - 11 - 5.3无限增益多路反馈二阶带通电路性能测试与分析 ....................................... - 14 - 六、结论 .............................................................................................................................................. - 16 -
滤波器电路设计实验报告 院系：物理科学与技术学院 专业班级： 学号： 学生： 指导教师：鸣 2013年12月20日
目录 0、设计要求 (1) 1、电路基本模型的选择以及参数的计算。 (1) 2、电路元件参数的计算 (4) 3、Multisim仿真 (5) 4、器件的选择 (8) 5、Protel制板 (9) 6、体会 (9)
一、电路基本模型的选择以及参数的计算。 （1）选择有源滤波器 有源滤波器：由有源器件构成的滤波器。 一般由集成运放与RC 网络构成，它具有体积小、性能稳定等优点，同时，由于集成运放的增益和输入阻抗都很高，输出阻抗很低，故有源滤波器还兼有放大与缓冲作用。 利用有源滤波器可以突出有用频率的信号，衰减无用频率的信号，抑制干扰和噪声，以达到提高信噪比或选频的目的，因而有源滤波器被广泛应用于通信、测量及控制技术中的小信号处理。 （2）滤波电路传递函数 分为：低通（LPF ）、高通（HPF ）、带通（BPF ）、带阻（BEF ）、全通（APF ） 理想滤波电路的频响在通带具有一定幅值和线性相移，而在阻带其幅值为0。实际电路往往难以达到理想要求。 根据不同要求，常用低通有三种： 巴特沃斯滤波器通带最平坦，阻带下降慢。 切比雪夫滤波器通带有纹波，阻带下降较快。 贝塞尔滤波器通带有纹波，阻带下降慢，且群时延恒定，失真小。 我们选择通带平坦的巴特沃思滤波器 n 阶巴特沃思传递函数。 ()A j ω= n: 阶数 ωC ：3dB 截止角频率 A0：通带电压增益 0|()|1()()10 n n c A j A ωωω≈= 026lg 220 A n A ?=-=≈ 因此本电路采用二阶巴特沃思低通滤波器与二阶巴特沃思高通滤波器级联而成。 基本框图如下
二阶有源带通滤波器设计及参数计算 标准化管理处编码[BBX968T-XBB8968-NNJ668-MM9N]
滤波器是一种只传输指定频段信号，抑制其它频段信号的电路。滤波器分为无源滤波器与有源滤波器两种：①无源滤波器: 由电感L、电容C及电阻R等无源元件组成②有源滤波器: 一般由集成运放与RC网络构成，它具有体积小、性能稳定等优点，同时，由于集成运放的增益和输入阻抗都很高，输出阻抗很低，故有源滤波器还兼有放大与缓冲作用。利用有源滤波器可以突出有用频率的信号，衰减无用频率的信号，抑制干扰和噪声，以达到提高信噪比或选频的目的，因而有源滤波器被广泛应用于通信、测量及控制技术中的小信号处理。从功能来上有源滤波器分为：低通滤波器（L P F）、高通滤波器（H P F）、带通滤波器（B P F）、带阻滤波器（B E F）、全通滤波器（APF）。其中前四种滤波器间互有联系，LPF与HPF间互为对偶关系。当LPF的通带截止频率高于HPF的通带截止频率时，将LPF与HPF相串联，就构成了BPF，而LPF与HPF并联，就构成BEF。在实用电子电路中，还可能同时采用几种不同型式的滤
波电路。滤波电路的主要性能指标有通带电压放大倍数AVP、通带截止频率fP及阻尼系数Q等。带通滤波器（BPF） （a）电路图(b)幅频特性 图1 压控电压源二阶带通滤波器 工作原理：这种滤波器的作用是只允许在某一个通频带范围内的信号通过，而比通频带下限频率低和比上限频率高的信号均加以衰减或抑制。典型的带通滤波器可以从二阶低通滤波器中将其中一级改成高通而成。如图1（a）所示。 电路性能参数 通带增益 中心频率 通带宽度 选择性 此是改变Rf和R4的比例就可改变频宽而不影响中心频率。 例．要求设计一个有源二阶带通滤波器，指标要求为：