Application lost the surface

土豆片子 2012-09-24 11:25:24

Application lost the surface
java.io.IOException: invalid preview surface
at android.media.MediaRecorder._prepare(Native Method)
at android.media.MediaRecorder.prepare(MediaRecorder.java:544)
at org.szga.MainActivity.yjbjsendVideo(MainActivity.java:910)
at org.szga.MainActivity.onClick(MainActivity.java:493)
at android.view.View.performClick(View.java:2485)
at android.view.View$PerformClick.run(View.java:9080)
at android.os.Handler.handleCallback(Handler.java:587)
at android.os.Handler.dispatchMessage(Handler.java:92)
at android.os.Looper.loop(Looper.java:123)
at android.app.ActivityThread.main(ActivityThread.java:3695)
at java.lang.reflect.Method.invokeNative(Native Method)
at java.lang.reflect.Method.invoke(Method.java:507)
at com.android.internal.os.ZygoteInit$MethodAndArgsCaller.run(ZygoteInit.java:842)
at com.android.internal.os.ZygoteInit.main(ZygoteInit.java:600)
at dalvik.system.NativeStart.main(Native Method)

用摄像头拍摄视频，在调用recorder.prepare(); 时报错。
源代码如下：
videopreview = (SurfaceView)findViewById(R.id.yjvideo);
videopreview.getHolder().setType(SurfaceHolder.SURFACE_TYPE_PUSH_BUFFERS);

recorder = new MediaRecorder();

recorder.setPreviewDisplay(videopreview.getHolder().getSurface());
recorder.setVideoSource(MediaRecorder.VideoSource.CAMERA);
recorder.setAudioSource(MediaRecorder.AudioSource.MIC);
recorder.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP);
recorder.setVideoSize(800, 480);
recorder.setVideoFrameRate(15);
recorder.setVideoEncoder(MediaRecorder.VideoEncoder.H263);
recorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB);
recorder.setOutputFile(videoFile.getAbsolutePath());
try {
recorder.prepare();
recorder.start();

Thread.sleep(10000);
recorder.stop();
recorder.reset();
recorder.release();
recorder=null;
mCamera.lock();
Tools.getUploadVideoTast(null, videoFile.getPath());
} catch (IllegalStateException e1) {
e1.printStackTrace();
} catch (IOException e1) {
e1.printStackTrace();
}catch (InterruptedException e) {
e.printStackTrace();
}

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顾小林 2013-02-22

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首先 surfaceholder.setcallback surfaceCreated 在这之后调用调用你那些操作另外为什么你的recorder 没有 setCamera呢？

土豆片子 2013-02-22

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貌似是因为sufaceview在使用的时候还没加载完

土豆片子 2013-02-22

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没人回答啊

bear1122ccc 2013-02-04

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怎么回事啊？同问。怎么没答案啊。

土豆片子 2012-10-30

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楼主，请问你解决了吗，我也碰到同样的问题了···

Computer Networking: A Top-Down Approach, 6th Edition Solutions to Review Questions and Problems Version Date: May 2012 This document contains the solutions to review questions and problems for the 5th edition of Computer Networking: A Top-Down Approach by Jim Kurose and Keith Ross. These solutions are being made available to instructors ONLY. Please do NOT copy or distribute this document to others (even other instructors). Please do not post any solutions on a publicly-available Web site. We’ll be happy to provide a copy (up-to-date) of this solution manual ourselves to anyone who asks. Acknowledgments: Over the years, several students and colleagues have helped us prepare this solutions manual. Special thanks goes to HongGang Zhang, Rakesh Kumar, Prithula Dhungel, and Vijay Annapureddy. Also thanks to all the readers who have made suggestions and corrected errors. All material © copyright 1996-2012 by J.F. Kurose and K.W. Ross. All rights reserved Chapter 1 Review Questions There is no difference. Throughout this text, the words “host” and “end system” are used interchangeably. End systems include PCs, workstations, Web servers, mail servers, PDAs, Internet-connected game consoles, etc. From Wikipedia: Diplomatic protocol is commonly described as a set of international courtesy rules. These well-established and time-honored rules have made it easier for nations and people to live and work together. Part of protocol has always been the acknowledgment of the hierarchical standing of all present. Protocol rules are based on the principles of civility. Standards are important for protocols so that people can create networking systems and products that interoperate. 1. Dial-up modem over telephone line: home; 2. DSL over telephone line: home or small office; 3. Cable to HFC: home; 4. 100 Mbps switched Ethernet: enterprise; 5. Wifi (802.11): home and enterprise: 6. 3G and 4G: wide-area wireless. HFC bandwidth is shared among the users. On the downstream channel, all packets emanate from a single source, namely, the head end. Thus, there are no collisions in the downstream channel. In most American cities, the current possibilities include: dial-up; DSL; cable modem; fiber-to-the-home. 7. Ethernet LANs have transmission rates of 10 Mbps, 100 Mbps, 1 Gbps and 10 Gbps. 8. Today, Ethernet most commonly runs over twisted-pair copper wire. It also can run over fibers optic links. 9. Dial up modems: up to 56 Kbps, bandwidth is dedicated; ADSL: up to 24 Mbps downstream and 2.5 Mbps upstream, bandwidth is dedicated; HFC, rates up to 42.8 Mbps and upstream rates of up to 30.7 Mbps, bandwidth is shared. FTTH: 2-10Mbps upload; 10-20 Mbps download; bandwidth is not shared. 10. There are two popular wireless Internet access technologies today: Wifi (802.11) In a wireless LAN, wireless users transmit/receive packets to/from an base station (i.e., wireless access point) within a radius of few tens of meters. The base station is typically connected to the wired Internet and thus serves to connect wireless users to the wired network. 3G and 4G wide-area wireless access networks. In these systems, packets are transmitted over the same wireless infrastructure used for cellular telephony, with the base station thus being managed by a telecommunications provider. This provides wireless access to users within a radius of tens of kilometers of the base station. 11. At time t0 the sending host begins to transmit. At time t1 = L/R1, the sending host completes transmission and the entire packet is received at the router (no propagation delay). Because the router has the entire packet at time t1, it can begin to transmit the packet to the receiving host at time t1. At time t2 = t1 + L/R2, the router completes transmission and the entire packet is received at the receiving host (again, no propagation delay). Thus, the end-to-end delay is L/R1 + L/R2. 12. A circuit-switched network can guarantee a certain amount of end-to-end bandwidth for the duration of a call. Most packet-switched networks today (including the Internet) cannot make any end-to-end guarantees for bandwidth. FDM requires sophisticated analog hardware to shift signal into appropriate frequency bands. 13. a) 2 users can be supported because each user requires half of the link bandwidth. b) Since each user requires 1Mbps when transmitting, if two or fewer users transmit simultaneously, a maximum of 2Mbps will be required. Since the available bandwidth of the shared link is 2Mbps, there will be no queuing delay before the link. Whereas, if three users transmit simultaneously, the bandwidth required will be 3Mbps which is more than the available bandwidth of the shared link. In this case, there will be queuing delay before the link. c) Probability that a given user is transmitting = 0.2 d) Probability that all three users are transmitting simultaneously = = (0.2)3 = 0.008. Since the queue grows when all the users are transmitting, the fraction of time during which the queue grows (which is equal to the probability that all three users are transmitting simultaneously) is 0.008. 14. If the two ISPs do not peer with each other, then when they send traffic to each other they have to send the traffic through a provider ISP (intermediary), to which they have to pay for carrying the traffic. By peering with each other directly, the two ISPs can reduce their payments to their provider ISPs. An Internet Exchange Points (IXP) (typically in a standalone building with its own switches) is a meeting point where multiple ISPs can connect and/or peer together. An ISP earns its money by charging each of the the ISPs that connect to the IXP a relatively small fee, which may depend on the amount of traffic sent to or received from the IXP. 15. Google's private network connects together all its data centers, big and small. Traffic between the Google data centers passes over its private network rather than over the public Internet. Many of these data centers are located in, or close to, lower tier ISPs. Therefore, when Google delivers content to a user, it often can bypass higher tier ISPs. What motivates content providers to create these networks? First, the content provider has more control over the user experience, since it has to use few intermediary ISPs. Second, it can save money by sending less traffic into provider networks. Third, if ISPs decide to charge more money to highly profitable content providers (in countries where net neutrality doesn't apply), the content providers can avoid these extra payments. 16. The delay components are processing delays, transmission delays, propagation delays, and queuing delays. All of these delays are fixed, except for the queuing delays, which are variable. 17. a) 1000 km, 1 Mbps, 100 bytes b) 100 km, 1 Mbps, 100 bytes 18. 10msec; d/s; no; no 19. a) 500 kbps b) 64 seconds c) 100kbps; 320 seconds 20. End system A breaks the large file into chunks. It adds header to each chunk, thereby generating multiple packets from the file. The header in each packet includes the IP address of the destination (end system B). The packet switch uses the destination IP address in the packet to determine the outgoing link. Asking which road to take is analogous to a packet asking which outgoing link it should be forwarded on, given the packet’s destination address. 21. The maximum emission rate is 500 packets/sec and the maximum transmission rate is 350 packets/sec. The corresponding traffic intensity is 500/350 =1.43 > 1. Loss will eventually occur for each experiment; but the time when loss first occurs will be different from one experiment to the next due to the randomness in the emission process. 22. Five generic tasks are error control, flow control, segmentation and reassembly, multiplexing, and connection setup. Yes, these tasks can be duplicated at different layers. For example, error control is often provided at more than one layer. 23. The five layers in the Internet protocol stack are – from top to bottom – the application layer, the transport layer, the network layer, the link layer, and the physical layer. The principal responsibilities are outlined in Section 1.5.1. 24. Application-layer message: data which an application wants to send and passed onto the transport layer; transport-layer segment: generated by the transport layer and encapsulates application-layer message with transport layer header; network-layer datagram: encapsulates transport-layer segment with a network-layer header; link-layer frame: encapsulates network-layer datagram with a link-layer header. 25. Routers process network, link and physical layers (layers 1 through 3). (This is a little bit of a white lie, as modern routers sometimes act as firewalls or caching components, and process Transport layer as well.) Link layer switches process link and physical layers (layers 1 through2). Hosts process all five layers. 26. a) Virus Requires some form of human interaction to spread. Classic example: E-mail viruses. b) Worms No user replication needed. Worm in infected host scans IP addresses and port numbers, looking for vulnerable processes to infect. 27. Creation of a botnet requires an attacker to find vulnerability in some application or system (e.g. exploiting the buffer overflow vulnerability that might exist in an application). After finding the vulnerability, the attacker needs to scan for hosts that are vulnerable. The target is basically to compromise a series of systems by exploiting that particular vulnerability. Any system that is part of the botnet can automatically scan its environment and propagate by exploiting the vulnerability. An important property of such botnets is that the originator of the botnet can remotely control and issue commands to all the nodes in the botnet. Hence, it becomes possible for the attacker to issue a command to all the nodes, that target a single node (for example, all nodes in the botnet might be commanded by the attacker to send a TCP SYN message to the target, which might result in a TCP SYN flood attack at the target). 28. Trudy can pretend to be Bob to Alice (and vice-versa) and partially or completely modify the message(s) being sent from Bob to Alice. For example, she can easily change the phrase “Alice, I owe you $1000” to “Alice, I owe you $10,000”. Furthermore, Trudy can even drop the packets that are being sent by Bob to Alice (and vise-versa), even if the packets from Bob to Alice are encrypted. Chapter 1 Problems Problem 1 There is no single right answer to this question. Many protocols would do the trick. Here's a simple answer below: Messages from ATM machine to Server Msg name purpose -------- ------- HELO Let server know that there is a card in the ATM machine ATM card transmits user ID to Server PASSWD User enters PIN, which is sent to server BALANCE User requests balance WITHDRAWL User asks to withdraw money BYE user all done Messages from Server to ATM machine (display) Msg name purpose -------- ------- PASSWD Ask user for PIN (password) OK last requested operation (PASSWD, WITHDRAWL) OK ERR last requested operation (PASSWD, WITHDRAWL) in ERROR AMOUNT sent in response to BALANCE request BYE user done, display welcome screen at ATM Correct operation: client server HELO (userid) --------------> (check if valid userid) <------------- PASSWD PASSWD --------------> (check password) <------------- AMOUNT WITHDRAWL --------------> check if enough $ to cover withdrawl (check if valid userid) <------------- PASSWD PASSWD --------------> (check password) <------------- AMOUNT WITHDRAWL --------------> check if enough $ to cover withdrawl <------------- BYE Problem 2 At time N*(L/R) the first packet has reached the destination, the second packet is stored in the last router, the third packet is stored in the next-to-last router, etc. At time N*(L/R) + L/R, the second packet has reached the destination, the third packet is stored in the last router, etc. Continuing with this logic, we see that at time N*(L/R) + (P-1)*(L/R) = (N+P-1)*(L/R) all packets have reached the destination. Problem 3 a) A circuit-switched network would be well suited to the application, because the application involves long sessions with predictable smooth bandwidth requirements. Since the transmission rate is known and not bursty, bandwidth can be reserved for each application session without significant waste. In addition, the overhead costs of setting up and tearing down connections are amortized over the lengthy duration of a typical application session. b) In the worst case, all the applications simultaneously transmit over one or more network links. However, since each link has sufficient bandwidth to handle the sum of all of the applications' data rates, no congestion (very little queuing) will occur. Given such generous link capacities, the network does not need congestion control mechanisms. Problem 4 Between the switch in the upper left and the switch in the upper right we can have 4 connections. Similarly we can have four connections between each of the 3 other pairs of adjacent switches. Thus, this network can support up to 16 connections. We can 4 connections passing through the switch in the upper-right-hand corner and another 4 connections passing through the switch in the lower-left-hand corner, giving a total of 8 connections. Yes. For the connections between A and C, we route two connections through B and two connections through D. For the connections between B and D, we route two connections through A and two connections through C. In this manner, there are at most 4 connections passing through any link. Problem 5 Tollbooths are 75 km apart, and the cars propagate at 100km/hr. A tollbooth services a car at a rate of one car every 12 seconds. a) There are ten cars. It takes 120 seconds, or 2 minutes, for the first tollbooth to service the 10 cars. Each of these cars has a propagation delay of 45 minutes (travel 75 km) before arriving at the second tollbooth. Thus, all the cars are lined up before the second tollbooth after 47 minutes. The whole process repeats itself for traveling between the second and third tollbooths. It also takes 2 minutes for the third tollbooth to service the 10 cars. Thus the total delay is 96 minutes. b) Delay between tollbooths is 8*12 seconds plus 45 minutes, i.e., 46 minutes and 36 seconds. The total delay is twice this amount plus 8*12 seconds, i.e., 94 minutes and 48 seconds. Problem 6 a) seconds. b) seconds. c) seconds. d) The bit is just leaving Host A. e) The first bit is in the link and has not reached Host B. f) The first bit has reached Host B. g) Want km. Problem 7 Consider the first bit in a packet. Before this bit can be transmitted, all of the bits in the packet must be generated. This requires sec=7msec. The time required to transmit the packet is sec= sec. Propagation delay = 10 msec. The delay until decoding is 7msec + sec + 10msec = 17.224msec A similar analysis shows that all bits experience a delay of 17.224 msec. Problem 8 a) 20 users can be supported. b) . c) . d) . We use the central limit theorem to approximate this probability. Let be independent random variables such that . “21 or more users” when is a standard normal r.v. Thus “21 or more users” . Problem 9 10,000 Problem 10 The first end system requires L/R1 to transmit the packet onto the first link; the packet propagates over the first link in d1/s1; the packet switch adds a processing delay of dproc; after receiving the entire packet, the packet switch connecting the first and the second link requires L/R2 to transmit the packet onto the second link; the packet propagates over the second link in d2/s2. Similarly, we can find the delay caused by the second switch and the third link: L/R3, dproc, and d3/s3. Adding these five delays gives dend-end = L/R1 + L/R2 + L/R3 + d1/s1 + d2/s2 + d3/s3+ dproc+ dproc To answer the second question, we simply plug the values into the equation to get 6 + 6 + 6 + 20+16 + 4 + 3 + 3 = 64 msec. Problem 11 Because bits are immediately transmitted, the packet switch does not introduce any delay; in particular, it does not introduce a transmission delay. Thus, dend-end = L/R + d1/s1 + d2/s2+ d3/s3 For the values in Problem 10, we get 6 + 20 + 16 + 4 = 46 msec. Problem 12 The arriving packet must first wait for the link to transmit 4.5 *1,500 bytes = 6,750 bytes or 54,000 bits. Since these bits are transmitted at 2 Mbps, the queuing delay is 27 msec. Generally, the queuing delay is (nL + (L - x))/R. Problem 13 The queuing delay is 0 for the first transmitted packet, L/R for the second transmitted packet, and generally, (n-1)L/R for the nth transmitted packet. Thus, the average delay for the N packets is: (L/R + 2L/R + ....... + (N-1)L/R)/N = L/(RN) * (1 + 2 + ..... + (N-1)) = L/(RN) * N(N-1)/2 = LN(N-1)/(2RN) = (N-1)L/(2R) Note that here we used the well-known fact: 1 + 2 + ....... + N = N(N+1)/2 It takes seconds to transmit the packets. Thus, the buffer is empty when a each batch of packets arrive. Thus, the average delay of a packet across all batches is the average delay within one batch, i.e., (N-1)L/2R. Problem 14 The transmission delay is . The total delay is Let . Total delay = For x=0, the total delay =0; as we increase x, total delay increases, approaching infinity as x approaches 1/a. Problem 15 Total delay . Problem 16 The total number of packets in the system includes those in the buffer and the packet that is being transmitted. So, N=10+1. Because , so (10+1)=a*(queuing delay + transmission delay). That is, 11=a*(0.01+1/100)=a*(0.01+0.01). Thus, a=550 packets/sec. Problem 17 There are nodes (the source host and the routers). Let denote the processing delay at the th node. Let be the transmission rate of the th link and let . Let be the propagation delay across the th link. Then . Let denote the average queuing delay at node . Then . Problem 18 On linux you can use the command traceroute www.targethost.com and in the Windows command prompt you can use tracert www.targethost.com In either case, you will get three delay measurements. For those three measurements you can calculate the mean and standard deviation. Repeat the experiment at different times of the day and comment on any changes. Here is an example solution: Traceroutes between San Diego Super Computer Center and www.poly.edu The average (mean) of the round-trip delays at each of the three hours is 71.18 ms, 71.38 ms and 71.55 ms, respectively. The standard deviations are 0.075 ms, 0.21 ms, 0.05 ms, respectively. In this example, the traceroutes have 12 routers in the path at each of the three hours. No, the paths didn’t change during any of the hours. Traceroute packets passed through four ISP networks from source to destination. Yes, in this experiment the largest delays occurred at peering interfaces between adjacent ISPs. Traceroutes from www.stella-net.net (France) to www.poly.edu (USA). The average round-trip delays at each of the three hours are 87.09 ms, 86.35 ms and 86.48 ms, respectively. The standard deviations are 0.53 ms, 0.18 ms, 0.23 ms, respectively. In this example, there are 11 routers in the path at each of the three hours. No, the paths didn’t change during any of the hours. Traceroute packets passed three ISP networks from source to destination. Yes, in this experiment the largest delays occurred at peering interfaces between adjacent ISPs. Problem 19 An example solution: Traceroutes from two different cities in France to New York City in United States In these traceroutes from two different cities in France to the same destination host in United States, seven links are in common including the transatlantic link. In this example of traceroutes from one city in France and from another city in Germany to the same host in United States, three links are in common including the transatlantic link. Traceroutes to two different cities in China from same host in United States Five links are common in the two traceroutes. The two traceroutes diverge before reaching China Problem 20 Throughput = min{Rs, Rc, R/M} Problem 21 If only use one path, the max throughput is given by: . If use all paths, the max throughput is given by . Problem 22 Probability of successfully receiving a packet is: ps= (1-p)N. The number of transmissions needed to be performed until the packet is successfully received by the client is a geometric random variable with success probability ps. Thus, the average number of transmissions needed is given by: 1/ps . Then, the average number of re-transmissions needed is given by: 1/ps -1. Problem 23 Let’s call the first packet A and call the second packet B. If the bottleneck link is the first link, then packet B is queued at the first link waiting for the transmission of packet A. So the packet inter-arrival time at the destination is simply L/Rs. If the second link is the bottleneck link and both packets are sent back to back, it must be true that the second packet arrives at the input queue of the second link before the second link finishes the transmission of the first packet. That is, L/Rs + L/Rs + dprop = L/Rs + dprop + L/Rc Thus, the minimum value of T is L/Rc  L/Rs . Problem 24 40 terabytes = 40 * 1012 * 8 bits. So, if using the dedicated link, it will take 40 * 1012 * 8 / (100 *106 ) =3200000 seconds = 37 days. But with FedEx overnight delivery, you can guarantee the data arrives in one day, and it should cost less than $100. Problem 25 160,000 bits 160,000 bits The bandwidth-delay product of a link is the maximum number of bits that can be in the link. the width of a bit = length of link / bandwidth-delay product, so 1 bit is 125 meters long, which is longer than a football field s/R Problem 26 s/R=20000km, then R=s/20000km= 2.5*108/(2*107)= 12.5 bps Problem 27 80,000,000 bits 800,000 bits, this is because that the maximum number of bits that will be in the link at any given time = min(bandwidth delay product, packet size) = 800,000 bits. .25 meters Problem 28 ttrans + tprop = 400 msec + 80 msec = 480 msec. 20 * (ttrans + 2 tprop) = 20*(20 msec + 80 msec) = 2 sec. Breaking up a file takes longer to transmit because each data packet and its corresponding acknowledgement packet add their own propagation delays. Problem 29 Recall geostationary satellite is 36,000 kilometers away from earth surface. 150 msec 1,500,000 bits 600,000,000 bits Problem 30 Let’s suppose the passenger and his/her bags correspond to the data unit arriving to the top of the protocol stack. When the passenger checks in, his/her bags are checked, and a tag is attached to the bags and ticket. This is additional information added in the Baggage layer if Figure 1.20 that allows the Baggage layer to implement the service or separating the passengers and baggage on the sending side, and then reuniting them (hopefully!) on the destination side. When a passenger then passes through security and additional stamp is often added to his/her ticket, indicating that the passenger has passed through a security check. This information is used to ensure (e.g., by later checks for the security information) secure transfer of people. Problem 31 Time to send message from source host to first packet switch = With store-and-forward switching, the total time to move message from source host to destination host = Time to send 1st packet from source host to first packet switch = . . Time at which 2nd packet is received at the first switch = time at which 1st packet is received at the second switch = Time at which 1st packet is received at the destination host = . After this, every 5msec one packet will be received; thus time at which last (800th) packet is received = . It can be seen that delay in using message segmentation is significantly less (almost 1/3rd). Without message segmentation, if bit errors are not tolerated, if there is a single bit error, the whole message has to be retransmitted (rather than a single packet). Without message segmentation, huge packets (containing HD videos, for example) are sent into the network. Routers have to accommodate these huge packets. Smaller packets have to queue behind enormous packets and suffer unfair delays. Packets have to be put in sequence at the destination. Message segmentation results in many smaller packets. Since header size is usually the same for all packets regardless of their size, with message segmentation the total amount of header bytes is more. Problem 32 Yes, the delays in the applet correspond to the delays in the Problem 31.The propagation delays affect the overall end-to-end delays both for packet switching and message switching equally. Problem 33 There are F/S packets. Each packet is S=80 bits. Time at which the last packet is received at the first router is sec. At this time, the first F/S-2 packets are at the destination, and the F/S-1 packet is at the second router. The last packet must then be transmitted by the first router and the second router, with each transmission taking sec. Thus delay in sending the whole file is To calculate the value of S which leads to the minimum delay, Problem 34 The circuit-switched telephone networks and the Internet are connected together at "gateways". When a Skype user (connected to the Internet) calls an ordinary telephone, a circuit is established between a gateway and the telephone user over the circuit switched network. The skype user's voice is sent in packets over the Internet to the gateway. At the gateway, the voice signal is reconstructed and then sent over the circuit. In the other direction, the voice signal is sent over the circuit switched network to the gateway. The gateway packetizes the voice signal and sends the voice packets to the Skype user. Chapter 2 Review Questions The Web: HTTP; file transfer: FTP; remote login: Telnet; e-mail: SMTP; BitTorrent file sharing: BitTorrent protocol Network architecture refers to the organization of the communication process into layers (e.g., the five-layer Internet architecture). Application architecture, on the other hand, is designed by an application developer and dictates the broad structure of the application (e.g., client-server or P2P). The process which initiates the communication is the client; the process that waits to be contacted is the server. No. In a P2P file-sharing application, the peer that is receiving a file is typically the client and the peer that is sending the file is typically the server. The IP address of the destination host and the port number of the socket in the destination process. You would use UDP. With UDP, the transaction can be completed in one roundtrip time (RTT) - the client sends the transaction request into a UDP socket, and the server sends the reply back to the client's UDP socket. With TCP, a minimum of two RTTs are needed - one to set-up the TCP connection, and another for the client to send the request, and for the server to send back the reply. One such example is remote word processing, for example, with Google docs. However, because Google docs runs over the Internet (using TCP), timing guarantees are not provided. a) Reliable data transfer TCP provides a reliable byte-stream between client and server but UDP does not. b) A guarantee that a certain value for throughput will be maintained Neither c) A guarantee that data will be delivered within a specified amount of time Neither d) Confidentiality (via encryption) Neither SSL operates at the application layer. The SSL socket takes unencrypted data from the application layer, encrypts it and then passes it to the TCP socket. If the application developer wants TCP to be enhanced with SSL, she has to include the SSL code in the application. A protocol uses handshaking if the two communicating entities first exchange control packets before sending data to each other. SMTP uses handshaking at the application layer whereas HTTP does not. The applications associated with those protocols require that all application data be received in the correct order and without gaps. TCP provides this service whereas UDP does not. When the user first visits the site, the server creates a unique identification number, creates an entry in its back-end database, and returns this identification number as a cookie number. This cookie number is stored on the user’s host and is managed by the browser. During each subsequent visit (and purchase), the browser sends the cookie number back to the site. Thus the site knows when this user (more precisely, this browser) is visiting the site. Web caching can bring the desired content “closer” to the user, possibly to the same LAN to which the user’s host is connected. Web caching can reduce the delay for all objects, even objects that are not cached, since caching reduces the traffic on links. Telnet is not available in Windows 7 by default. to make it available, go to Control Panel, Programs and Features, Turn Windows Features On or Off, Check Telnet client. To start Telnet, in Windows command prompt, issue the following command > telnet webserverver 80 where "webserver" is some webserver. After issuing the command, you have established a TCP connection between your client telnet program and the web server. Then type in an HTTP GET message. An example is given below: Since the index.html page in this web server was not modified since Fri, 18 May 2007 09:23:34 GMT, and the above commands were issued on Sat, 19 May 2007, the server returned "304 Not Modified". Note that the first 4 lines are the GET message and header lines inputed by the user, and the next 4 lines (starting from HTTP/1.1 304 Not Modified) is the response from the web server. FTP uses two parallel TCP connections, one connection for sending control information (such as a request to transfer a file) and another connection for actually transferring the file. Because the control information is not sent over the same connection that the file is sent over, FTP sends control information out of band. The message is first sent from Alice’s host to her mail server over HTTP. Alice’s mail server then sends the message to Bob’s mail server over SMTP. Bob then transfers the message from his mail server to his host over POP3. 17. Received: from 65.54.246.203 (EHLO bay0-omc3-s3.bay0.hotmail.com) (65.54.246.203) by mta419.mail.mud.yahoo.com with SMTP; Sat, 19 May 2007 16:53:51 -0700 Received: from hotmail.com ([65.55.135.106]) by bay0-omc3-s3.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2668); Sat, 19 May 2007 16:52:42 -0700 Received: from mail pickup service by hotmail.com with Microsoft SMTPSVC; Sat, 19 May 2007 16:52:41 -0700 Message-ID: Received: from 65.55.135.123 by by130fd.bay130.hotmail.msn.com with HTTP; Sat, 19 May 2007 23:52:36 GMT From: "prithula dhungel" To: prithula@yahoo.com Bcc: Subject: Test mail Date: Sat, 19 May 2007 23:52:36 +0000 Mime-Version: 1.0 Content-Type: Text/html; format=flowed Return-Path: prithuladhungel@hotmail.com Figure: A sample mail message header Received: This header field indicates the sequence in which the SMTP servers send and receive the mail message including the respective timestamps. In this example there are 4 “Received:” header lines. This means the mail message passed through 5 different SMTP servers before being delivered to the receiver’s mail box. The last (forth) “Received:” header indicates the mail message flow from the SMTP server of the sender to the second SMTP server in the chain of servers. The sender’s SMTP server is at address 65.55.135.123 and the second SMTP server in the chain is by130fd.bay130.hotmail.msn.com. The third “Received:” header indicates the mail message flow from the second SMTP server in the chain to the third server, and so on. Finally, the first “Received:” header indicates the flow of the mail messages from the forth SMTP server to the last SMTP server (i.e. the receiver’s mail server) in the chain. Message-id: The message has been given this number BAY130-F26D9E35BF59E0D18A819AFB9310@phx.gbl (by bay0-omc3-s3.bay0.hotmail.com. Message-id is a unique string assigned by the mail system when the message is first created. From: This indicates the email address of the sender of the mail. In the given example, the sender is “prithuladhungel@hotmail.com” To: This field indicates the email address of the receiver of the mail. In the example, the receiver is “prithula@yahoo.com” Subject: This gives the subject of the mail (if any specified by the sender). In the example, the subject specified by the sender is “Test mail” Date: The date and time when the mail was sent by the sender. In the example, the sender sent the mail on 19th May 2007, at time 23:52:36 GMT. Mime-version: MIME version used for the mail. In the example, it is 1.0. Content-type: The type of content in the body of the mail message. In the example, it is “text/html”. Return-Path: This specifies the email address to which the mail will be sent if the receiver of this mail wants to reply to the sender. This is also used by the sender’s mail server for bouncing back undeliverable mail messages of mailer-daemon error messages. In the example, the return path is “prithuladhungel@hotmail.com”. With download and delete, after a user retrieves its messages from a POP server, the messages are deleted. This poses a problem for the nomadic user, who may want to access the messages from many different machines (office PC, home PC, etc.). In the download and keep configuration, messages are not deleted after the user retrieves the messages. This can also be inconvenient, as each time the user retrieves the stored messages from a new machine, all of non-deleted messages will be transferred to the new machine (including very old messages). Yes an organization’s mail server and Web server can have the same alias for a host name. The MX record is used to map the mail server’s host name to its IP address. You should be able to see the sender's IP address for a user with an .edu email address. But you will not be able to see the sender's IP address if the user uses a gmail account. It is not necessary that Bob will also provide chunks to Alice. Alice has to be in the top 4 neighbors of Bob for Bob to send out chunks to her; this might not occur even if Alice provides chunks to Bob throughout a 30-second interval. Recall that in BitTorrent, a peer picks a random peer and optimistically unchokes the peer for a short period of time. Therefore, Alice will eventually be optimistically unchoked by one of her neighbors, during which time she will receive chunks from that neighbor. The overlay network in a P2P file sharing system consists of the nodes participating in the file sharing system and the logical links between the nodes. There is a logical link (an “edge” in graph theory terms) from node A to node B if there is a semi-permanent TCP connection between A and B. An overlay network does not include routers. Mesh DHT: The advantage is in order to a route a message to the peer (with ID) that is closest to the key, only one hop is required; the disadvantage is that each peer must track all other peers in the DHT. Circular DHT: the advantage is that each peer needs to track only a few other peers; the disadvantage is that O(N) hops are needed to route a message to the peer that is closest to the key. 25. File Distribution Instant Messaging Video Streaming Distributed Computing With the UDP server, there is no welcoming socket, and all data from different clients enters the server through this one socket. With the TCP server, there is a welcoming socket, and each time a client initiates a connection to the server, a new socket is created. Thus, to support n simultaneous connections, the server would need n+1 sockets. For the TCP application, as soon as the client is executed, it attempts to initiate a TCP connection with the server. If the TCP server is not running, then the client will fail to make a connection. For the UDP application, the client does not initiate connections (or attempt to communicate with the UDP server) immediately upon execution Chapter 2 Problems Problem 1 a) F b) T c) F d) F e) F Problem 2 Access control commands: USER, PASS, ACT, CWD, CDUP, SMNT, REIN, QUIT. Transfer parameter commands: PORT, PASV, TYPE STRU, MODE. Service commands: RETR, STOR, STOU, APPE, ALLO, REST, RNFR, RNTO, ABOR, DELE, RMD, MRD, PWD, LIST, NLST, SITE, SYST, STAT, HELP, NOOP. Problem 3 Application layer protocols: DNS and HTTP Transport layer protocols: UDP for DNS; TCP for HTTP Problem 4 The document request was http://gaia.cs.umass.edu/cs453/index.html. The Host : field indicates the server's name and /cs453/index.html indicates the file name. The browser is running HTTP version 1.1, as indicated just before the first pair. The browser is requesting a persistent connection, as indicated by the Connection: keep-alive. This is a trick question. This information is not contained in an HTTP message anywhere. So there is no way to tell this from looking at the exchange of HTTP messages alone. One would need information from the IP datagrams (that carried the TCP segment that carried the HTTP GET request) to answer this question. Mozilla/5.0. The browser type information is needed by the server to send different versions of the same object to different types of browsers. Problem 5 The status code of 200 and the phrase OK indicate that the server was able to locate the document successfully. The reply was provided on Tuesday, 07 Mar 2008 12:39:45 Greenwich Mean Time. The document index.html was last modified on Saturday 10 Dec 2005 18:27:46 GMT. There are 3874 bytes in the document being returned. The first five bytes of the returned document are : Applications by Actively Querying DNS Caches”, in IMC'03, October 2729, 2003, Miami Beach, Florida, USA Problem 21 Yes, we can use dig to query that Web site in the local DNS server. For example, “dig cnn.com” will return the query time for finding cnn.com. If cnn.com was just accessed a couple of seconds ago, an entry for cnn.com is cached in the local DNS cache, so the query time is 0 msec. Otherwise, the query time is large. Problem 22 For calculating the minimum distribution time for client-server distribution, we use the following formula: Dcs = max {NF/us, F/dmin} Similarly, for calculating the minimum distribution time for P2P distribution, we use the following formula: Where, F = 15 Gbits = 15 * 1024 Mbits us = 30 Mbps dmin = di = 2 Mbps Note, 300Kbps = 300/1024 Mbps. Client Server N 10 100 1000 u 300 Kbps 7680 51200 512000 700 Kbps 7680 51200 512000 2 Mbps 7680 51200 512000 Peer to Peer N 10 100 1000 u 300 Kbps 7680 25904 47559 700 Kbps 7680 15616 21525 2 Mbps 7680 7680 7680 Problem 23 Consider a distribution scheme in which the server sends the file to each client, in parallel, at a rate of a rate of us/N. Note that this rate is less than each of the client’s download rate, since by assumption us/N ≤ dmin. Thus each client can also receive at rate us/N. Since each client receives at rate us/N, the time for each client to receive the entire file is F/( us/N) = NF/ us. Since all the clients receive the file in NF/ us, the overall distribution time is also NF/ us. Consider a distribution scheme in which the server sends the file to each client, in parallel, at a rate of dmin. Note that the aggregate rate, N dmin, is less than the server’s link rate us, since by assumption us/N ≥ dmin. Since each client receives at rate dmin, the time for each client to receive the entire file is F/ dmin. Since all the clients receive the file in this time, the overall distribution time is also F/ dmin. From Section 2.6 we know that DCS ≥ max {NF/us, F/dmin} (Equation 1) Suppose that us/N ≤ dmin. Then from Equation 1 we have DCS ≥ NF/us . But from (a) we have DCS ≤ NF/us . Combining these two gives: DCS = NF/us when us/N ≤ dmin. (Equation 2) We can similarly show that: DCS =F/dmin when us/N ≥ dmin (Equation 3). Combining Equation 2 and Equation 3 gives the desired result. Problem 24 Define u = u1 + u2 + ….. + uN. By assumption us <= (us + u)/N Equation 1 Divide the file into N parts, with the ith part having size (ui/u)F. The server transmits the ith part to peer i at rate ri = (ui/u)us. Note that r1 + r2 + ….. + rN = us, so that the aggregate server rate does not exceed the link rate of the server. Also have each peer i forward the bits it receives to each of the N-1 peers at rate ri. The aggregate forwarding rate by peer i is (N-1)ri. We have (N-1)ri = (N-1)(usui)/u = (us + u)/N Equation 2 Let ri = ui/(N-1) and rN+1 = (us – u/(N-1))/N In this distribution scheme, the file is broken into N+1 parts. The server sends bits from the ith part to the ith peer (i = 1, …., N) at rate ri. Each peer i forwards the bits arriving at rate ri to each of the other N-1 peers. Additionally, the server sends bits from the (N+1) st part at rate rN+1 to each of the N peers. The peers do not forward the bits from the (N+1)st part. The aggregate send rate of the server is r1+ …. + rN + N rN+1 = u/(N-1) + us – u/(N-1) = us Thus, the server’s send rate does not exceed its link rate. The aggregate send rate of peer i is (N-1)ri = ui Thus, each peer’s send rate does not exceed its link rate. In this distribution scheme, peer i receives bits at an aggregate rate of Thus each peer receives the file in NF/(us+u). (For simplicity, we neglected to specify the size of the file part for i = 1, …., N+1. We now provide that here. Let Δ = (us+u)/N be the distribution time. For i = 1, …, N, the ith file part is Fi = ri Δ bits. The (N+1)st file part is FN+1 = rN+1 Δ bits. It is straightforward to show that F1+ ….. + FN+1 = F.) The solution to this part is similar to that of 17 (c). We know from section 2.6 that Combining this with a) and b) gives the desired result. Problem 25 There are N nodes in the overlay network. There are N(N-1)/2 edges. Problem 26 Yes. His first claim is possible, as long as there are enough peers staying in the swarm for a long enough time. Bob can always receive data through optimistic unchoking by other peers. His second claim is also true. He can run a client on each host, let each client “free-ride,” and combine the collected chunks from the different hosts into a single file. He can even write a small scheduling program to make the different hosts ask for different chunks of the file. This is actually a kind of Sybil attack in P2P networks. Problem 27 Peer 3 learns that peer 5 has just left the system, so Peer 3 asks its first successor (Peer 4) for the identifier of its immediate successor (peer 8). Peer 3 will then make peer 8 its second successor. Problem 28 Peer 6 would first send peer 15 a message, saying “what will be peer 6’s predecessor and successor?” This message gets forwarded through the DHT until it reaches peer 5, who realizes that it will be 6’s predecessor and that its current successor, peer 8, will become 6’s successor. Next, peer 5 sends this predecessor and successor information back to 6. Peer 6 can now join the DHT by making peer 8 its successor and by notifying peer 5 that it should change its immediate successor to 6. Problem 29 For each key, we first calculate the distances (using d(k,p)) between itself and all peers, and then store the key in the peer that is closest to the key (that is, with smallest distance value). Problem 30 Yes, randomly assigning keys to peers does not consider the underlying network at all, so it very likely causes mismatches. Such mismatches may degrade the search performance. For example, consider a logical path p1 (consisting of only two logical links): ABC, where A and B are neighboring peers, and B and C are neighboring peers. Suppose that there is another logical path p2 from A to C (consisting of 3 logical links): ADEC. It might be the case that A and B are very far away physically (and separated by many routers), and B and C are very far away physically (and separated by many routers). But it may be the case that A, D, E, and C are all very close physically (and all separated by few routers). In other words, a shorter logical path may correspond to a much longer physical path. Problem 31 If you run TCPClient first, then the client will attempt to make a TCP connection with a non-existent server process. A TCP connection will not be made. UDPClient doesn't establish a TCP connection with the server. Thus, everything should work fine if you first run UDPClient, then run UDPServer, and then type some input into the keyboard. If you use different port numbers, then the client will attempt to establish a TCP connection with the wrong process or a non-existent process. Errors will occur. Problem 32 In the original program, UDPClient does not specify a port number when it creates the socket. In this case, the code lets the underlying operating system choose a port number. With the additional line, when UDPClient is executed, a UDP socket is created with port number 5432 . UDPServer needs to know the client port number so that it can send packets back to the correct client socket. Glancing at UDPServer, we see that the client port number is not “hard-wired” into the server code; instead, UDPServer determines the client port number by unraveling the datagram it receives from the client. Thus UDP server will work with any client port number, including 5432. UDPServer therefore does not need to be modified. Before: Client socket = x (chosen by OS) Server socket = 9876 After: Client socket = 5432 Problem 33 Yes, you can configure many browsers to open multiple simultaneous connections to a Web site. The advantage is that you will you potentially download the file faster. The disadvantage is that you may be hogging the bandwidth, thereby significantly slowing down the downloads of other users who are sharing the same physical links. Problem 34 For an application such as remote login (telnet and ssh), a byte-stream oriented protocol is very natural since there is no notion of message boundaries in the application. When a user types a character, we simply drop the character into the TCP connection. In other applications, we may be sending a series of messages that have inherent boundaries between them. For example, when one SMTP mail server sends another SMTP mail server several email messages back to back. Since TCP does not have a mechanism to indicate the boundaries, the application must add the indications itself, so that receiving side of the application can distinguish one message from the next. If each message were instead put into a distinct UDP segment, the receiving end would be able to distinguish the various messages without any indications added by the sending side of the application. Problem 35 To create a web server, we need to run web server software on a host. Many vendors sell web server software. However, the most popular web server software today is Apache, which is open source and free. Over the years it has been highly optimized by the open-source community. Problem 36 The key is the infohash, the value is an IP address that currently has the file designated by the infohash. Chapter 3 Review Questions Call this protocol Simple Transport Protocol (STP). At the sender side, STP accepts from the sending process a chunk of data not exceeding 1196 bytes, a destination host address, and a destination port number. STP adds a four-byte header to each chunk and puts the port number of the destination process in this header. STP then gives the destination host address and the resulting segment to the network layer. The network layer delivers the segment to STP at the destination host. STP then examines the port number in the segment, extracts the data from the segment, and passes the data to the process identified by the port number. The segment now has two header fields: a source port field and destination port field. At the sender side, STP accepts a chunk of data not exceeding 1192 bytes, a destination host address, a source port number, and a destination port number. STP creates a segment which contains the application data, source port number, and destination port number. It then gives the segment and the destination host address to the network layer. After receiving the segment, STP at the receiving host gives the application process the application data and the source port number. No, the transport layer does not have to do anything in the core; the transport layer “lives” in the end systems. For sending a letter, the family member is required to give the delegate the letter itself, the address of the destination house, and the name of the recipient. The delegate clearly writes the recipient’s name on the top of the letter. The delegate then puts the letter in an envelope and writes the address of the destination house on the envelope. The delegate then gives the letter to the planet’s mail service. At the receiving side, the delegate receives the letter from the mail service, takes the letter out of the envelope, and takes note of the recipient name written at the top of the letter. The delegate then gives the letter to the family member with this name. No, the mail service does not have to open the envelope; it only examines the address on the envelope. Source port number y and destination port number x. An application developer may not want its application to use TCP’s congestion control, which can throttle the application’s sending rate at times of congestion. Often, designers of IP telephony and IP videoconference applications choose to run their applications over UDP because they want to avoid TCP’s congestion control. Also, some applications do not need the reliable data transfer provided by TCP. Since most firewalls are configured to block UDP traffic, using TCP for video and voice traffic lets the traffic though the firewalls. Yes. The application developer can put reliable data transfer into the application layer protocol. This would require a significant amount of work and debugging, however. Yes, both segments will be directed to the same socket. For each received segment, at the socket interface, the operating system will provide the process with the IP addresses to determine the origins of the individual segments. For each persistent connection, the Web server creates a separate “connection socket”. Each connection socket is identified with a four-tuple: (source IP address, source port number, destination IP address, destination port number). When host C receives and IP datagram, it examines these four fields in the datagram/segment to determine to which socket it should pass the payload of the TCP segment. Thus, the requests from A and B pass through different sockets. The identifier for both of these sockets has 80 for the destination port; however, the identifiers for these sockets have different values for source IP addresses. Unlike UDP, when the transport layer passes a TCP segment’s payload to the application process, it does not specify the source IP address, as this is implicitly specified by the socket identifier. Sequence numbers are required for a receiver to find out whether an arriving packet contains new data or is a retransmission. To handle losses in the channel. If the ACK for a transmitted packet is not received within the duration of the timer for the packet, the packet (or its ACK or NACK) is assumed to have been lost. Hence, the packet is retransmitted. A timer would still be necessary in the protocol rdt 3.0. If the round trip time is known then the only advantage will be that, the sender knows for sure that either the packet or the ACK (or NACK) for the packet has been lost, as compared to the real scenario, where the ACK (or NACK) might still be on the way to the sender, after the timer expires. However, to detect the loss, for each packet, a timer of constant duration will still be necessary at the sender. The packet loss caused a time out after which all the five packets were retransmitted. Loss of an ACK didn’t trigger any retransmission as Go-Back-N uses cumulative acknowledgements. The sender was unable to send sixth packet as the send window size is fixed to 5. When the packet was lost, the received four packets were buffered the receiver. After the timeout, sender retransmitted the lost packet and receiver delivered the buffered packets to application in correct order. Duplicate ACK was sent by the receiver for the lost ACK. The sender was unable to send sixth packet as the send win

首先接到这一个项目，说是要用mediastreamer2做一个网络电话。之前也是从来没有接触过。于是首先开始在百度中搜索一下需要哪些东西，以及那些步骤。最后大致了解了一下，做这个项目最终要的就是需要移植好多的库，每一个库都需要配置，最后在交叉编译好动态库，然后在执行mediastreamer2的时候去调用这些动态库和头文件就OK了。 1、首先meidastream2是基于ortp库的，那么首先就是下载源码，交叉编译。交叉编译ortp 下载源码：http://savannah.c3sl.ufpr.br/linphone/ortp/sources/?C=S;O=A 我使用0.18.0版本 ortp-0.18.0.tar.gz 然后通过winshare（Windows和Linux的通信）把下载好的库文件拷贝到Linux下，然后解压 tar zxvf ortp-0.18.0.tar.gz 注意这个时候可能会发生错误，是没有权限的问题，那么就在命令行前边加上sudo 然后配置把下边这三行写成一个脚本 vim **.sh ./configure CC=arm-linux-gcc --host=arm-linux --target=arm-linux --prefix=/home/protocol_stack/install/ make make install 然后用chmod 777 **.sh 执行脚本./**.sh 这样子就完成了配置，编译，安装。（安装目录为/home/protocol_stack/install/,也就是最后生成的头文件，可执行文件，库文件都会在这个目录下） 2、因为项目是要用到SIP协议的，所以我们还需要移植sip的库 osip2和eXosip2协议，这两个协议对应两个库，osip是简单的osip协议，但是因为API少等一系列原因，增加了eXosip2对osip2的补充。交叉编译osip2 下载源码：http://ftp.gnu.org/gnu/osip/ 我使用的版本是3.6.0 libosip2-3.6.0.tar.gz 然后通过winshare（Windows和Linux的通信）把下载好的库文件拷贝到Linux下，然后解压 tar zxvf libosip2-3.6.0.tar.gz 注意这个时候可能会发生错误，是没有权限的问题，那么就在命令行前边加上sudo 然后配置把下边这三行写成一个脚本 vim **.sh ./configure --host=arm-linux --target=arm-linux --prefix=/home/protocol_stack/install/ make make install 然后用chmod 777 **.sh 执行脚本./**.sh 交叉编译eXosip2 下载源码：http://ftp.gnu.org/gnu/osip/ 我使用的版本是3.6.0 libeXosip2-3.6.0.tar.gz 然后通过winshare（Windows和Linux的通信）把下载好的库文件拷贝到Linux下，然后解压 tar zxvf libeXosip2-3.6.0.tar.gz 注意这个时候可能会发生错误，是没有权限的问题，那么就在命令行前边加上sudo 然后配置把下边这三行写成一个脚本 vim **.sh ./configure --host=arm-linux --target=arm-linux --prefix=/home/protocol_stack/install/ PKG_CONFIG_PATH=/home/protocol_stack/install/lib/pkgconfig make make install 然后用chmod 777 **.sh 执行脚本./**.sh 接下来可以编译mediastreamer2了，不过ms2，依赖好多库：ogg、speex、pulseaudio。而pulseaudio又依赖许多库：alsa、json、libtool。 3、交叉编译ogg 下载源码：http://xiph.org/downloads/ 我使用1.3.1版本 libogg-1.3.3.tar.gz 然后通过winshare（Windows和Linux的通信）把下载好的库文件拷贝到Linux下，然后解压 tar zxvf libogg-1.3.3.tar.gz 注意这个时候可能会发生错误，是没有权限的问题，那么就在命令行前边加上sudo 然后配置把下边这三行写成一个脚本 vim **.sh ./configure CC=arm-linux-gcc --prefix=/home/protocol_stack/install/ --host=arm-linux make make install 然后用chmod 777 **.sh 执行脚本./**.sh 4、交叉编译speex 下载源码：http://www.speex.org/downloads/ 我使用1.2rc1版本 speex-1.2rc1.tar.gz 然后通过winshare（Windows和Linux的通信）把下载好的库文件拷贝到Linux下，然后解压 tar zxvf speex-1.2rc1.tar.gz 注意这个时候可能会发生错误，是没有权限的问题，那么就在命令行前边加上sudo 然后配置把下边这三行写成一个脚本 vim **.sh ./configure CC=arm-linux-gcc --prefix=/home/protocol_stack/install/ --with-ogg=/home/protocol_stack/install/ --enable-fixed-point --disable-float-api \ --host=arm-linux make make install 然后用chmod 777 **.sh 执行脚本./**.sh 5、交叉编译pulseaudio 下载源码：http://freedesktop.org/software/pulseaudio/releases/ 我使用1.0版本 pulseaudio-1.0.tar.gz 然后通过winshare（Windows和Linux的通信）把下载好的库文件拷贝到Linux下，然后解压 tar zxvf pulseaudio-1.0.tar.gz 注意这个时候可能会发生错误，是没有权限的问题，那么就在命令行前边加上sudo 然后配置把下边这三行写成一个脚本 vim **.sh ./configure CC=arm-linux-gcc CXX=arm-linux-g++ --prefix=/home/protocol_stack/install --host=arm-linux --disable-rpath --disable-nls --disable-dbus --disable-bluez --disable-samplerate --disable-solaris --disable-gconf --disable-avahi --disable-jack --disable-lirc --disable-glib2 --disable-gtk2 --disable-openssl --disable-ipv6 --disable-asyncns --disable-per-user-esound-socket --disable-oss-output --disable-oss-wrapper --disable-x11 --enable-neon-opt=no --with-database=simple PKG_CONFIG_PATH=/home/protocol_stack/install/lib/pkgconfig CPPFLAGS=-I/home/protocol_stack/install/include LDFLAGS=-L/home/protocol_stack/install/lib CFLAGS=-I/home/protocol_stack/install/include make make install 然后用chmod 777 **.sh 执行脚本./**.sh 错误1： checking for ltdl.h... no configure: error: Unable to find libltdl version 2. Makes sure you have libtool 2.4 or later installed. make: *** No targets specified and no makefile found. Stop. 分析；找不到libltdl。确保你有libtool 2.4及以上的版本。下载libtool 2.4.2版本这时需要交叉编译libtool 下载源码：ftp://ftp.gnu.org/gnu/libtool/ 我使用2.4.2版本 libtool-2.4.2.tar.gz 然后通过winshare（Windows和Linux的通信）把下载好的库文件拷贝到Linux下，然后解压 tar zxvf libtool-2.4.2.tar.gz 注意这个时候可能会发生错误，是没有权限的问题，那么就在命令行前边加上sudo 然后配置把下边这三行写成一个脚本 vim **.sh ./configure --host=arm-linux --prefix =/home/protocol_stack/install/ make make install 然后用chmod 777 **.sh 执行脚本./**.sh 交叉编译alsa： http://www.alsa-project.org/main/index.php/Main_Page 这个库的版本需要根据你嵌入式Linux内核中alsa的版本而定，可以使用命令查看内核中alsa的版本： # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.24. 可以到内核中alsa驱动版本是1.0.24，所以我选1.0.24版本 alsa-lib-1.0.24.1.tar.gz 然后通过winshare（Windows和Linux的通信）把下载好的库文件拷贝到Linux下，然后解压 tar zxvf speex-1.2rc1.tar.gz 注意这个时候可能会发生错误，是没有权限的问题，那么就在命令行前边加上sudo 然后配置把下边这三行写成一个脚本 vim **.sh ./configure --host=arm-linux --prefix =/home/protocol_stack/install/ make make install 然后用chmod 777 **.sh 执行脚本./**.sh 错误：configure: error: Package requirements ( sndfile >= 1.0.20 ) were not met: No package 'sndfile' found 分析：缺少库 libsndfile库，那么接下来再进行交叉编译libsndfile libsndfile-1.0.25.tar.gz http://www.linuxfromscratch.org/blfs/view/svn/multimedia/libsndfile.html 然后通过winshare（Windows和Linux的通信）把下载好的库文件拷贝到Linux下，然后解压 tar zxvf libsndfile-1.0.25.tar.gz 注意这个时候可能会发生错误，是没有权限的问题，那么就在命令行前边加上sudo 然后配置把下边这三行写成一个脚本 vim **.sh ./configure --host=arm-linux --prefix =/home/protocol_stack/install/ make make install 然后用chmod 777 **.sh 执行脚本./**.sh 7、最后编译mediastreamer2 下载源码：http://ftp.twaren.net/Unix/NonGNU//linphone/mediastreamer/ 我使用2.8版本 mediastreamer-2.8.0.tar.gz 然后通过winshare（Windows和Linux的通信）把下载好的库文件拷贝到Linux下，然后解压 tar zxvf mediastreamer-2.8.0.tar.gz 注意这个时候可能会发生错误，是没有权限的问题，那么就在命令行前边加上sudo 然后配置把下边这三行写成一个脚本 vim **.sh ./configure CC=arm-linux-gcc --prefix=/home/protocol_stack/install/ PKG_CONFIG_PATH=/home/protocol_stack/install/lib/pkgconfig --disable-gsm --enable-video=no --enable-macsnd=no --disable-static --disable-sdl --disable-x11 --disable-ffmpeg --host=arm-linux --target=arm-linux make make install 然后用chmod 777 **.sh 执行脚本./**.sh 上面的configure选项没有屏蔽v4l1和v4l2，所以还得交叉编译v4l 编译v4l libv4l-0.6.4.tar.gz 下载源码：http://pkgs.fedoraproject.org/repo/pkgs/libv4l/ 然后通过winshare（Windows和Linux的通信）把下载好的库文件拷贝到Linux下，然后解压 tar zxvf libv4l-0.6.4.tar.gz 注意这个时候可能会发生错误，是没有权限的问题，那么就在命令行前边加上sudo 然后配置我使用0.6.4版本 libv4l-0.6.4.tar.gz make clean make CC=arm-linux-gcc make install PREFIX=/home/protocol_stack/install 编译mediastreamer2出错：(1)checking for LIBCHECK... no checking for LIBJSON... no configure: error: Package requirements ( json >= 0.9 ) were not met: No package 'json' found 解决方法就是交叉编译json 下载源码：http://ftp.debian.org/debian/pool/main/j/json-c/ 分析：缺少json库，那么我们继续交叉编译json库 json-c_0.12.1.orig.tar.gz 然后通过winshare（Windows和Linux的通信）把下载好的库文件拷贝到Linux下，然后解压 tar zxvf mediastreamer-2.8.0.tar.gz 注意这个时候可能会发生错误，是没有权限的问题，那么就在命令行前边加上sudo 然后配置 ./configure --host=linux-arm \ --prefix =/home/protocol_stack/install/ make && make install 好了，json库已经编译完成了。接下来我们继续编译mediastreamer2 。。。。。但是还是有问题，怎么办呢？还是哪个问题还是找不到json库。分析：在json的论坛中，找到了解决方案：把编译生成的/lib/pkgconfig/这个目录下生成了一个json-c.pc。最后mediastreamer2在调用的时候找的是json.pc。那么我们就把这个文件名改为json.pc #mv json-c.pc json.pc OK，这次这个是可以编译的过去了。接下来继续编译。。。又出现问题了 /home/protocol_stack/install/lib/libjson.so: undefined reference to `rpl_malloc' /home/protocol_stack/install/lib/libjson.so: undefined reference to `rpl_realloc' 问题分析：这个错误的原因是因为没有定义 rpl_malloc 和 rpl_realloc 这两个变量。那么我们应该怎么办么？那么就在这个目录下进行查这两个变量是在哪里定义的？于是：#grep "rpl_malloc" -nR * ....... 找到了，原来这两个变量是在这个config的文件中的。是一个宏开关那么就好办了，我们就直接把这两个宏进行注释。嗯嗯，继续。。。我们重新编译json库。。。嗯嗯编译好了，接下来继续来编译mediastreamer2 。。。。又出错了，还是这个原因 /home/protocol_stack/install/lib/libjson.so: undefined reference to `rpl_malloc' /home/protocol_stack/install/lib/libjson.so: undefined reference to `rpl_realloc 嗯嗯，还是这个原因？究竟是为什么呢。再次来到json的目录下，再次看有没有把那两个宏开关给关闭？嗯哼？竟然没有关闭？分析？明白了。原来是我把配置和编译同时执行了。这个宏开关是./configure ...生成的。那么就只好，这样。把./configure。。。生成的config文件，再进行关闭宏开关。最后直接make && make install -j8 直接编译，安装，是不能再次进行配置的。因为以配置config文件就会再次生成，那么宏开关就又开了。 OK，安装好了，下来继续进行编译mediastreamer2.。。。。。。。。。。。。。。。。。。。。。。。。。又出现了问题？ error: /user/include/python2.7/pyconfig.h:15:52: fatal error: arm-linux-gnueabi/python2.7/pyconfig.h: No such file or directory compilation terminated. 分析：：找不到arm-linux-gnueabi/python2.7/pyconfig.h这文件。那就继续交叉编译python 好吧，继续下载python,然后再进行交叉编译，但是编译Python的时候出来一系列的问题。根本没有办法解决。那么该怎么办呢？时候一个小时又一个小时的过去？最后有一个大胆的想法，既然python都编译不下去。那就不要了。于是，在mediastreamer2的./configure 中添上一项 --without-python 。。。。再次配置编译。。。。。。。。。。。 error: /user/include/python2.7/pyconfig.h:15:52: fatal error: arm-linux-gnueabi/python2.7/pyconfig.h: No such file or directory compilation termiated. 嗯哼？还是一样的错误。怎么办呢？于是乎就又在论坛上进行找灵感。。。。。还是找不到。。。又一结合前边几个库的配置编译，发现不使能一个模块还可以用另外一个--disable-python 。。。于是乎就把--without-python改为了--disable-python 继续编译。。。。。。。。。。。。。。。。。。。。。。。。。到了这个节骨眼上了，编译每跳一下，我的心就跟着逗一下。。。。心酸。。。。。。。。。。。。。。。。。。竟然编译成功了。。。。哈哈。。。。。。。。。于是，马上就把编译好的库，拷贝到了开发板。。。嗯嗯，本来还想把编译好的库目录树拷贝下的，但是太多了，放不下。。。算了吧。。。。找到编译好的库在库中的/bin中找到arm-linux-mediastream 然后执行./arm-linux-mediastream 。。。。报错了问题： error : while loading shared libraries: libmediastreamer.so.1: cannot open shared object file: No such file 答案：分析：遇到这个问题就是，libmediastreamer.so.1这个动态库，在可执行文件armlinuxmediastreamer执行的时候，会调用这个动态库，但是环境变量中找不到这个动态库。那么我们就是要把我们编译好的动态链接库的目录加到环境变量中 LD_LIBRARY_PATH=$LD_LIBRARY_PATH://arm/lib/这个目录下就是放着我们编译好的所有的动态链接库（包括libmediastreamer.so.1）执行步骤：LD_LIBRARY_PATH=$LD_LIBRARY_PATH://arm/lib export LD_LIBRARY_PATH ./arm-linux-mediastream mediastream --local --remote --payload [ --fmtp ] [ --jitter ] [ --width ] [ --height ] [ --bitrate ] [ --ec (enable echo canceller) ] [ --ec-tail ] [ --ec-delay ] [ --ec-framesize ] [ --agc (enable automatic gain control) ] [ --ng (enable noise gate)] [ --ng-threshold (noise gate threshold) ] [ --ng-floorgain (gain applied to the signal when its energy is below the threshold.) ] [ --capture-card ] [ --playback-card ] [ --infile <input wav file> specify a wav file to be used for input, instead of soundcard ] [ --outfile specify a wav file to write audio into, instead of soundcard ] [ --camera ] [ --el (enable echo limiter) ] [ --el-speed (gain changes are smoothed with a coefficent) ] [ --el-thres (Threshold above which the system becomes active) ] [ --el-force (The proportional coefficient controlling the mic attenuation) ] [ --el-sustain (Time in milliseconds for which the attenuation is kept unchanged after) ] [ --el-transmit-thres (TO BE DOCUMENTED) ] [ --rc (enable adaptive rate control) ] [ --zrtp (enable zrtp) ] [ --verbose (most verbose messages) ] [ --video-windows-id <video surface:preview surface>] [ --srtp (enable srtp, master key is generated if absent from comand line) [ --netsim-bandwidth (simulates a network download bandwidth limit) 于是按照第一种方式进行参数添加 ./arm-linux-mediastream --local 8888 --remote 127.0.0.1:88 88 OK运行正常了下面是运行信息。。。 ortp-message-audio_stream_process_rtcp: interarrival jitter=119 , lost packets percentage since last report=0.000000, round trip time=0.000000 seconds ortp-message-oRTP-stats: RTP stats : ortp-message- number of rtp packet sent=150 ortp-message- number of rtp bytes sent=25800 bytes ortp-message- number of rtp packet received=150 ortp-message- number of rtp bytes received=25800 bytes ortp-message- number of incoming rtp bytes successfully delivered to the application=25284 ortp-message- number of rtp packet lost=0 ortp-message- number of rtp packets received too late=0 ortp-message- number of bad formatted rtp packets=0 ortp-message- number of packet discarded because of queue overflow=0 ortp-message-Bandwidth usage: download=81.290281 kbits/sec, upload=81.288664 kbits/sec ortp-message-Receiving RTCP SR ortp-message-Receiving RTCP SDES ortp-message-Found CNAME=unknown@unknown ortp-message-Found TOOL=oRTP-0.18.0 ortp-message-Found NOTE=This is free sofware (LGPL) ! ortp-message-Quality indicator : 4.888437 运行正常了。。。。。。

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android的多媒体系统多媒体系统的结构和业务多媒体系统的宏鸡观结构应用层，java框架层,c语言层,硬件抽像层,其中输入输出由HAL层,处理环节由packetView的OpenCore实现, 多媒体业备有以下几种: musicPlayer(音频播放器) viderPlayer(视频播放器) Camera(照相机) soundRecord(录音机) videoCa...

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