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• Turning a series of raw images into movie frames in Android

  - by Nicholas Killewald

  I've got an Android project I'm working on that, ultimately, will require me to create a movie file out of a series of still images taken with a phone's camera. That is to say, I want to be able to take raw image frames and string them together, one by one, into a movie. Audio is not a concern at this stage. Looking over the Android API, it looks like there are calls in it to create movie files, but it seems those are entirely geared around making a live recording from the camera on an immediate basis. While nice, I can't use that for my purposes, as I need to put annotations and other post-production things on the images as they come in before they get fed into a movie (plus, the images come way too slowly to do a live recording). Worse, looking over the Android source, it looks like a non-trivial task to rewire that to do what I want it to do (at least without touching the NDK). Is there any way I can use the API to do something like this? Or alternatively, what would be the best way to go about this, if it's even feasible on cell phone hardware (which seems to keep getting more and more powerful, strangely...)?

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• Getting level values from PCM raw data using Core Audio

  - by John

  I am trying to extract level data from a PCM audio file using core audio. I have gotten as far as (I believe) getting the raw data into a byte array (UInt8) but it is 16 bit PCM data and I am having trouble reading the data out. The input is from the iPhone microphone, which I have set as: [recordSetting setValue:[NSNumber numberWithInt:kAudioFormatLinearPCM] forKey:AVFormatIDKey]; [recordSetting setValue:[NSNumber numberWithFloat:44100.0] forKey:AVSampleRateKey]; [recordSetting setValue:[NSNumber numberWithInt:1] forKey:AVNumberOfChannelsKey]; [recordSetting setValue:[NSNumber numberWithInt:16] forKey:AVLinearPCMBitDepthKey]; [recordSetting setValue:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsBigEndianKey]; [recordSetting setValue:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsFloatKey]; which is obviously 16 bits. I am then trying to just print out a few values to see if they look reasonable for debug purposes below, and they do not look reasonable (many 0's). ExtAudioFileRef inputFile = NULL; ExtAudioFileOpenURL(track.location, &inputFile); AudioStreamBasicDescription inputFileFormat; UInt32 dataSize = (UInt32)sizeof(inputFileFormat); ExtAudioFileGetProperty(inputFile, kExtAudioFileProperty_FileDataFormat, &dataSize, &inputFileFormat); UInt8 *buffer = malloc(BUFFER_SIZE); AudioBufferList bufferList; bufferList.mNumberBuffers = 1; bufferList.mBuffers[0].mNumberChannels = 1; bufferList.mBuffers[0].mData = buffer; //pointer to buffer of audio data bufferList.mBuffers[0].mDataByteSize = BUFFER_SIZE; //number of bytes in the buffer while(true) { UInt32 frameCount = (bufferList.mBuffers[0].mDataByteSize / inputFileFormat.mBytesPerFrame); // Read a chunk of input OSStatus status = ExtAudioFileRead(inputFile, &frameCount, &bufferList); // If no frames were returned, conversion is finished if(0 == frameCount) break; NSLog(@"---"); int16_t *bufferl = &buffer; for(int i=0;i<100;i++){ //const int16_t *bufferl = bufferl[i]; NSLog(@"%d",bufferl[i]); } } Not sure what I am doing wrong, I think it has to do with reading the byte array. Sorry for the long code post...

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• How to intercept raw soap request/response (data) from WCF client

  - by JohnIdol

  This question seems to be pretty close to what I am looking for - I was able to setup tracing and I am looking at the log entries for my calls to the service. However I need to see the raw soap request with the data I am sending to the service and I see no way of doing that from the SvcTraceViewer (only log entries are shown but no data sent to the service) - am I just missing configuration? Here's what I got in my web.config: <system.diagnostics> <sources> <source name="System.ServiceModel" switchValue="Verbose" propagateActivity="true"> <listeners> <add name="sdt" type="System.Diagnostics.XmlWriterTraceListener" initializeData="App_Data/Logs/WCFTrace.svclog" /> </listeners> </source> </sources> </system.diagnostics> Any help appreciated! UPDATE: this is all I see in my trace: <E2ETraceEvent xmlns="http://schemas.microsoft.com/2004/06/E2ETraceEvent"> <System xmlns="http://schemas.microsoft.com/2004/06/windows/eventlog/system"> <EventID>262163</EventID> <Type>3</Type> <SubType Name="Information">0</SubType> <Level>8</Level> <TimeCreated SystemTime="2010-05-10T13:10:46.6713553Z" /> <Source Name="System.ServiceModel" /> <Correlation ActivityID="{00000000-0000-0000-1501-0080000000f6}" /> <Execution ProcessName="w3wp" ProcessID="3492" ThreadID="23" /> <Channel /> <Computer>MY_COMPUTER_NAME</Computer> </System> <ApplicationData> <TraceData> <DataItem> <TraceRecord xmlns="http://schemas.microsoft.com/2004/10/E2ETraceEvent/TraceRecord" Severity="Information"> <TraceIdentifier>http://msdn.microsoft.com/en-US/library/System.ServiceModel.Channels.MessageSent.aspx</TraceIdentifier> <Description>Sent a message over a channel.</Description> <AppDomain>MY_DOMAIN</AppDomain> <Source>System.ServiceModel.Channels.HttpOutput+WebRequestHttpOutput/50416815</Source> <ExtendedData xmlns="http://schemas.microsoft.com/2006/08/ServiceModel/MessageTraceRecord"> <MessageProperties> <Encoder>text/xml; charset=utf-8</Encoder> <AllowOutputBatching>False</AllowOutputBatching> <Via>http://xxx.xx.xxx.xxx:9080/MyWebService/myService</Via> </MessageProperties> <MessageHeaders></MessageHeaders> </ExtendedData> </TraceRecord> </DataItem> </TraceData> </ApplicationData>

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• Binary stream 'NN' does not contain a valid BinaryHeader. Possible causes are invalid stream or obje

  - by FinancialRadDeveloper

  I am passing user defined classes over sockets. The SendObject code is below. It works on my local machine, but when I publish to the WebServer which is then communicating with the App Server on my own machine it fails. public bool SendObject(Object obj, ref string sErrMsg) { try { MemoryStream ms = new MemoryStream(); BinaryFormatter bf1 = new BinaryFormatter(); bf1.Serialize(ms, obj); byte[] byArr = ms.ToArray(); int len = byArr.Length; m_socClient.Send(byArr); return true; } catch (Exception e) { sErrMsg = "SendObject Error: " + e.Message; return false; } } I can do this fine if it is one class in my tools project and the other class about UserData just doesn't want to know. Frustrating! Ohh. I think its because the UserData class has a DataSet inside it. Funnily enough I have seen this work, but then after 1 request it goes loopy and I can't get it to work again. Anyone know why this might be? I have looked at comparing the dlls to make sure they are the same on the WebServer and on my local machine and they look to be so as I have turned on versioning in the AssemblyInfo.cs to double check. Edit: Ok it seems that the problem is with size. If I keep it under 1024 byes ( I am guessing here) it works on the web server and doesnt if it has a DataSet inside it.k In fact this is so puzzling I converted the DataSet to a string using ds.GetXml() and this also causes it to blow up. :( So it seems that across the network something with my sockets is wrong and doesn't want to read in the data. JonSkeet where are you. ha ha. I would offer Rep but I don't have any. Grr

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• Why is it assumed that send may return with less than requested data transmitted on a blocking socke

  - by Ernelli

  The standard method to send data on a stream socket has always been to call send with a chunk of data to write, check the return value to see if all data was sent and then keep calling send again until the whole message has been accepted. For example this is a simple example of a common scheme: int send_all(int sock, unsigned char *buffer, int len) { int nsent; while(len 0) { nsent = send(sock, buffer, len, 0); if(nsent == -1) // error return -1; buffer += nsent; len -= nsent; } return 0; // ok, all data sent } Even the BSD manpage mentions that ...If no messages space is available at the socket to hold the message to be transmitted, then send() normally blocks... Which indicates that we should assume that send may return without sending all data. Now I find this rather broken but even W. Richard Stevens assumes this in his standard reference book about network programming, not in the beginning chapters, but the more advanced examples uses his own writen (write all data) function instead of calling write. Now I consider this still to be more or less broken, since if send is not able to transmit all data or accept the data in the underlying buffer and the socket is blocking, then send should block and return when the whole send request has been accepted. I mean, in the code example above, what will happen if send returns with less data sent is that it will be called right again with a new request. What has changed since last call? At max a few hundred CPU cycles have passed so the buffer is still full. If send now accepts the data why could'nt it accept it before? Otherwise we will end upp with an inefficient loop where we are trying to send data on a socket that cannot accept data and keep trying, or else? So it seems like the workaround, if needed, results in heavily inefficient code and in those circumstances blocking sockets should be avoided at all an non blocking sockets together with select should be used instead.

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• Qt Socket blocking functions required to run in QThread where created. Any way past this?

  - by Alexander Kondratskiy

  The title is very cryptic, so here goes! I am writing a client that behaves in a very synchronous manner. Due to the design of the protocol and the server, everything has to happen sequentially (send request, wait for reply, service reply etc.), so I am using blocking sockets. Here is where Qt comes in. In my application I have a GUI thread, a command processing thread and a scripting engine thread. I create the QTcpSocket in the command processing thread, as part of my Client class. The Client class has various methods that boil down to writing to the socket, reading back a specific number of bytes, and returning a result. The problem comes when I try to directly call Client methods from the scripting engine thread. The Qt sockets randomly time out and when using a debug build of Qt, I get these warnings: QSocketNotifier: socket notifiers cannot be enabled from another thread QSocketNotifier: socket notifiers cannot be disabled from another thread Anytime I call these methods from the command processing thread (where Client was created), I do not get these problems. To simply phrase the situation: Calling blocking functions of QAbstractSocket, like waitForReadyRead(), from a thread other than the one where the socket was created (dynamically allocated), causes random behaviour and debug asserts/warnings. Anyone else experienced this? Ways around it? Thanks in advance.

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• Best way to do interprocess communication on Mac OS X

  - by jbrennan

  I'm looking at building a Cocoa application on the Mac with a back-end daemon process (really just a mostly-headless Cocoa app, probably), along with 0 or more "client" applications running locally (although if possible I'd like to support remote clients as well; the remote clients would only ever be other Macs or iPhone OS devices). The data being communicated will be fairly trivial, mostly just text and commands (which I guess can be represented as text anyway), and maybe the occasional small file (an image possibly). I've looked at a few methods for doing this but I'm not sure which is "best" for the task at hand. Things I've considered: Reading and writing to a file (…yes), very basic but not very scalable. Pure sockets (I have no experience with sockets but I seem to think I can use them to send data locally and over a network. Though it seems cumbersome if doing everything in Cocoa Distributed Objects: seems rather inelegant for a task like this NSConnection: I can't really figure out what this class even does, but I've read of it in some IPC search results I'm sure there are things I'm missing, but I was surprised to find a lack of resources on this topic.

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• How to host WCF service and TCP server on same socket?

  - by Ole Jak

  Tooday I use ServiceHost for self hosting WCF cervices. I want to host near to my WCF services my own TCP programm for direct sockets operations (like lien to some sort of broadcasting TCP stream) I need control over namespaces (so I would be able to let my clients to send TCP streams directly into my service using some nice URLs like example.com:port/myserver/stream?id=1 or example.com:port/myserver/stream?id=anything and so that I will not be bothered with Idea of 1 client for 1 socket at one time moment, I realy want to keep my WCF services on the same port as my own server or what it is so to be able to call www.example.com:port/myWCF/stream?id=222...) Can any body please help me with this? I am using just WCF now. And I do not enjoy how it works. That is one of many resons why I want to start migration to clear TCP=) I can not use net-tcp binding or any sort of other cool WS-* binding (tooday I use the simpliest one so that my clients like Flash, AJAX, etc connect to me with ease). I needed Fast and easy in implemrnting connection protocol like one I created fore use with Sockets for real time hi ammount of data transfering. So.. Any Ideas? Please - I need help.

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• Winsock tcp/ip Socket listening but connection refused, race condition?

  - by Wayne

  Hello folks. This involves two automated unit tests which each start up a tcp/ip server that creates a non-blocking socket then bind()s and listen()s in a loop on select() for a client that connects and downloads some data. The catch is that they work perfectly when run separately but when run as a test suite, the second test client will fail to connect with WSACONNREFUSED... UNLESS there is a Thread.Sleep() of several seconds between them??!!! Interestingly, there is retry loop every 1 second for connecting after any failure. So the second test loops for a while until timeout after 10 minutes. During that time, netstat -na shows the correct port number is in the LISTEN state for the server socket. So if it is in the listen state? Why won't it accept the connection? In the code, there are log messages that show the select NEVER even gets a socket ready to read (which means ready to accept a connection when it applies to a listening socket). Obviously the problem must be related to some race condition between finishing one test which means close() and shutdown() on each end of the socket, and the start up of the next. This wouldn't be so bad if the retry logic allowed it to connect eventually after a couple of seconds. However it seems to get "gummed up" and won't even retry. However, for some strange reason the listening socket SAYS it's in the LISTEN state even through keeps refusing connections. So that means it's the Windoze O/S which is actually catching the SYN packet and returning a RST packet (which means "Connection Refused"). The only other time I ever saw this error was when the code had a problem that caused hundreds of sockets to get stuck in TIME_WAIT state. But that's not the case here. netstat shows only about a dozen sockets with only 1 or 2 in TIME_WAIT at any given moment. Please help.

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• How to hand-over a TCP listening socket with minimal downtime?

  - by Shtééf

  While this question is tagged EventMachine, generic BSD-socket solutions in any language are much appreciated too. Some background: I have an application listening on a TCP socket. It is started and shut down with a regular System V style init script. My problem is that it needs some time to start up before it is ready to service the TCP socket. It's not too long, perhaps only 5 seconds, but that's 5 seconds too long when a restart needs to be performed during a workday. It's also crucial that existing connections remain open and are finished normally. Reasons for a restart of the application are patches, upgrades, and the like. I unfortunately find myself in the position that, every once in a while, I need to do this kind of thing in production. The question: I'm looking for a way to do a neat hand-over of the TCP listening socket, from one process to another, and as a result get only a split second of downtime. I'd like existing connections / sockets to remain open and finish processing in the old process, while the new process starts servicing new connectinos. Is there some proven method of doing this using BSD-sockets? (Bonus points for an EventMachine solution.) Are there perhaps open-source libraries out there implementing this, that I can use as is, or use as a reference? (Again, non-Ruby and non-EventMachine solutions are appreciated too!)

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• With the advent of HTML 5, is there a point in using COMET anymore?

  - by h2g2java

  I am very tempted to use long wait http or periodic polling by the client to set up pseudo-sockets on the browser side, for an application that would be used publicly. But then on the 2nd thought, I am thinking HTML 5 is here. But on the 3rd thought, what is the percentage of browsers out there that remain non-HTML5 within 12 months, 24 months, 36 months? If there are at least 20% of browsers still incapable of HTML5, then I cannot depend on HTML5 because 20% of users not being able to access an application is a significant amount. What do you think, how would your advice be (to me and to developers in general)? Q1. Is there any point in rigging in COMET into an application anymore? I am thinking of gwt comet - http://code.google.com/p/gwt-comet/. Q2. Should we release a new public application within the next 2 months that is dependent on HTML5 sockets and tell non-HTML5 browser users "sorry, your browser version cannot access this application"? Or should we architect the apps to use communication like GWT RPC? Q3. I am also very distrustful of long wait http request. I have never used it before but I have a horrible feeling about it. I have been using 10 to 20 second client-side polling. Is long wait http request risky (risk of hanging a browser session)? Does long wait request present any additional security risk?

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• sendto is returning ENOBUF

  - by user338159

  Hi, I am currently running an old system on Tru64 which involves lots of UDP sockets using the sendto() function. The sockets are used in our code to send messages to/from various processes and then eventually on to a thick client app that is connected remotely. Occasionally the socket to the thick client gets stuck, this can cause some of these messages to get built up. My question is how can I determine the current buffer size, and how do I determine the maximum message buffer. The code below gives a snippet of how I set up the port and use the sendto function. /* need to adjust the maximum size we can send on this / / as it needs to be able to cope with the biggest / / messages we send / lenlen = sizeof(len) ; / allow double for when the system is under load */ len = 2 * C_MAX_MESSAGE_DATA_SIZE ; lpos_setsockopt(FATAL, msg_socket,SOL_SOCKET, SO_SNDBUF, &len, lenlen, &error_no) ; result = sendto( msg_socket, (char *)message, (int)message_len, flags, dest_addr, addrlen); Note. We have ported this application to Linux and the problem does not seem to appear there. Any help would be greatly appreciated. Regards

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• How to detect a Socket disconnection?

  - by AngryHacker

  I've implemented a task using the async Sockets pattern in Silverlight 3. I started with Michael Schwarz's implementation and built on top of that. So basically, my Silverlight app establishes a persistent socket connection to a device and then data flows both ways as necessary between the device and the Silverlight app. One thing I am struggling with is how to detect disconnection. I could think of 2 approaches: Keep-Alive. I know this can be done at the Sockets level, but I am not sure how to do this in an async model. How would the Socket class let me know there has been a disconnection. Manual keep alive. Basically, I am having the Silverlight app send a dummy packet every 20 seconds or so. If it fails, I'd assume disconnection. However, incredibly, SocketAsyncEventArgs.SocketError always reports success, even if I simply unplug the device that the Silverlight app is connected to. I am not sure whether this is a bug or what or perhaps I need to upgrade to SL4. Any ideas, direction or implementation would be appreciated.

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• Rails 3 - raw/html_safe not working in some cases?

  - by Frexuz

  I'm having difficulties with output not being encoded even though I'm using raw or html_safe. This one is writing out the &nbsp in my final HTLM page. def build_tag_cloud(tag_cloud, style_list) tag_cloud.sort!{ |x,y| x.permalink <=> y.permalink } max, min = 0, 0 tag_cloud.each do |tag| max = tag.followers.to_i if tag.followers.to_i > max min = tag.followers.to_i if tag.followers.to_i < min end divisor = ((max - min) / style_list.size) + 1 html = "" tag_cloud.each do |tag| name = raw(tag.name.gsub('&','&amp;').gsub(' ','&nbsp;')) link = raw(link_to "#{name}", {:controller => "/shows", :action => "show", :permalink => tag.permalink}, :class => "#{style_list[(tag.followers.to_i - min) / divisor]}") html += raw("<li>#{link}</li> ") end return raw(html.to_s) end What is allowed in using raw and html_safe? And how should my example above be fixed?

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• Peer did not return a certificate

  - by pfista

  I am trying to get two way SSL authentication working between a Python server and an Android client application. I have access to both the server and client, and would like to implement client authentication using my own certificate. So far I have been able to verify the server certificate and connect without client authentication. What sort of certificate does the client need and how do I get it to automatically send it to the server during the handshake process? Here is the client and server side code that I have so far. Is my approach wrong? Server Code while True: # Keep listening for clients c, fromaddr = sock.accept() ssl_sock = ssl.wrap_socket(c, keyfile = "serverPrivateKey.pem", certfile = "servercert.pem", server_side = True, # Require the client to provide a certificate cert_reqs = ssl.CERT_REQUIRED, ssl_version = ssl.PROTOCOL_TLSv1, ca_certs = "clientcert.pem", #TODO must point to a file of CA certificates?? do_handshake_on_connect = True, ciphers="!NULL:!EXPORT:AES256-SHA") print ssl_sock.cipher() thrd = sock_thread(ssl_sock) thrd.daemon = True thrd.start() I suspect I may be using the wrong file for ca_certs...? Client Code private boolean connect() { try { KeyStore keystore = KeyStore.getInstance("BKS"); // Stores the client certificate, to be sent to server KeyStore truststore = KeyStore.getInstance("BKS"); // Stores the server certificate we want to trust // TODO: change hard coded password... THIS IS REAL BAD MKAY truststore.load(mSocketService.getResources().openRawResource(R.raw.truststore), "test".toCharArray()); keystore.load(mSocketService.getResources().openRawResource(R.raw.keystore), "test".toCharArray()); // Use the key manager for client authentication. Keys in the key manager will be sent to the host KeyManagerFactory keyFManager = KeyManagerFactory.getInstance(KeyManagerFactory.getDefaultAlgorithm()); keyFManager.init(keystore, "test".toCharArray()); // Use the trust manager to determine if the host I am connecting to is a trusted host TrustManagerFactory trustMFactory = TrustManagerFactory.getInstance(TrustManagerFactory .getDefaultAlgorithm()); trustMFactory.init(truststore); // Create the socket factory and add both the trust manager and key manager SSLCertificateSocketFactory socketFactory = (SSLCertificateSocketFactory) SSLCertificateSocketFactory .getDefault(5000, new SSLSessionCache(mSocketService)); socketFactory.setTrustManagers(trustMFactory.getTrustManagers()); socketFactory.setKeyManagers(keyFManager.getKeyManagers()); // Open SSL socket directly to host, host name verification is NOT performed here due to // SSLCertificateFactory implementation mSSLSocket = (SSLSocket) socketFactory.createSocket(mHostname, mPort); mSSLSocket.setSoTimeout(TIMEOUT); // Most SSLSocketFactory implementations do not verify the server's identity, allowing man-in-the-middle // attacks. This implementation (SSLCertificateSocketFactory) does check the server's certificate hostname, // but only for createSocket variants that specify a hostname. When using methods that use InetAddress or // which return an unconnected socket, you MUST verify the server's identity yourself to ensure a secure // connection. verifyHostname(); // Safe to proceed with socket now ... I have generated a client private key, a client certificate, a server private key, and a server certificate using openssl. I then added the client certificate to keystore.bks (which I store in /res/raw/keystore.bks) I then added the server certificate to the truststore.bks So now when the client tries to connect I am getting this error server side: ssl.SSLError: [Errno 1] _ssl.c:504: error:140890C7:SSL routines:SSL3_GET_CLIENT_CERTIFICATE:peer did not return a certificate And when I try to do this in the android client SSLSession s = mSSLSocket.getSession(); s.getPeerCertificates(); I get this error: javax.net.ssl.SSLPeerUnverifiedException: No peer certificate So obviously the keystore I am using doesn't appear to have a correct peer certificate in it and thus isn't sending one to the server. What should I put in the keystore to prevent this exception? Furthermore, is this method of two way SSL authentication safe and effective?

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• Architecture strategies for a complex competition scoring system

  - by mikewassmer

  Competition description: There are about 10 teams competing against each other over a 6-week period. Each team's total score (out of a 1000 total available points) is based on the total of its scores in about 25,000 different scoring elements. Most scoring elements are worth a small fraction of a point and there will about 10 X 25,000 = 250,000 total raw input data points. The points for some scoring elements are awarded at frequent regular time intervals during the competition. The points for other scoring elements are awarded at either irregular time intervals or at just one moment in time. There are about 20 different types of scoring elements. Each of the 20 types of scoring elements has a different set of inputs, a different algorithm for calculating the earned score from the raw inputs, and a different number of total available points. The simplest algorithms require one input and one simple calculation. The most complex algorithms consist of hundreds or thousands of raw inputs and a more complicated calculation. Some types of raw inputs are automatically generated. Other types of raw inputs are manually entered. All raw inputs are subject to possible manual retroactive adjustments by competition officials. Primary requirements: The scoring system UI for competitors and other competition followers will show current and historical total team scores, team standings, team scores by scoring element, raw input data (at several levels of aggregation, e.g. daily, weekly, etc.), and other metrics. There will be charts, tables, and other widgets for displaying historical raw data inputs and scores. There will be a quasi-real-time dashboard that will show current scores and raw data inputs. Aggregate scores should be updated/refreshed whenever new raw data inputs arrive or existing raw data inputs are adjusted. There will be a "scorekeeper UI" for manually entering new inputs, manually adjusting existing inputs, and manually adjusting calculated scores. Decisions: Should the scoring calculations be performed on the database layer (T-SQL/SQL Server, in my case) or on the application layer (C#/ASP.NET MVC, in my case)? What are some recommended approaches for calculating updated total team scores whenever new raw inputs arrives? Calculating each of the teams' total scores from scratch every time a new input arrives will probably slow the system to a crawl. I've considered some kind of "diff" approach, but that approach may pose problems for ad-hoc queries and some aggegates. I'm trying draw some sports analogies, but it's tough because most games consist of no more than 20 or 30 scoring elements per game (I'm thinking of a high-scoring baseball game; football and soccer have fewer scoring events per game). Perhaps a financial balance sheet analogy makes more sense because financial "bottom line" calcs may be calculated from 250,000 or more transactions. Should I be making heavy use of caching for this application? Are there any obvious approaches or similar case studies that I may be overlooking?

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• Windows Phone 7 : Socket Programming

  - by davidhayes

  Is there any support for TCP/IP Sockets in Windows Phone 7? I'm not having any luck finding it so far. I want to connect to a port on a server and issue commands/receive responses

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• Using socket API on IPhone

  - by dl3nar

  Hi, for a little project I have to do the following task on my IPhone: open a TCP socket send a command to the server shutdown the WRITE part of the connection read the response from the server close the connection I'm not experienced with socket programming - I've just started with network programming and I've already used the CFStream interface. But obviously streams are not adequate for this task. Who can point me in the right direction? I tried to find a tutorial on Apples website about sockets, but there is nothing. Regards, Thomas

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• Debugging in XCode as root

  - by Anton

  In my program I need to create sockets and bind them to listen HTTP port (80). The program works fine when I launch it from command line with sudo, escalating permissions to root. Running under XCode gives a 'permission denied' error on the call to binding function (asio::ip::tcp::acceptor::bind()). How can I do debugging under XCode? All done in C++ and boost.asio on Mac OS X 10.5 with XCode 3.1.2.

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• ruby socket dgram example

  - by Bub Bradlee

  I'm trying to use unix sockets and SOCK_DGRAM in ruby, but am having a really hard time figuring out how to do it. So far, I've been trying things like this: sock_path = 'test.socket' s1 = Socket.new(Socket::AF_UNIX, Socket::SOCK_DGRAM, 0) s1.bind(Socket.pack_sockaddr_un(sock_path)) s2 = Socket.new(Socket::AF_UNIX, Socket::SOCK_DGRAM, 0) s2.bind(Socket.pack_sockaddr_un(sock_path)) s1.send("HELLO") s2.recv(5) # should equal "HELLO" Does anybody have experience with this?

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• What is SocketOptionName.ReuseAddress used for?

  - by Hemant

  I used to think that using SocketOptionName.ReuseAddress, I can reuse a port that is in TIME_WAIT state. But I tried to experiment with it and it seems it has no effect. If I check sockets using netstat, and it shows the socket is in TIME_WAIT state and I immediately run the client again, I get the exception: Only one usage of each socket address (protocol/network address/port) is normally permitted 172.16.16.16:12345 I cannot make anything out of it. Please can you elaborate what SocketOptionName.ReuseAddress is good for?

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• silverlight Socket: Unhandled Error in Silverlight Application An attempt was made to access a sock

  - by Yang

  I basically try to reproduce the Socket example from here: http://www.silverlightshow.net/items/Sockets-and-their-implementation-in-SL2-Beta-1-including-a-chat-like-example.aspx I only made a small change in the client side, i.e., String safeHost = "127.0.0.1"; int port = 4509; Then I got this permission error? Any idea why? Unhandled Error in Silverlight Application An attempt was made to access a socket in a way forbidden by its access permissions.

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• FTP monitoring and downloading of new files

  - by Jojo

  Hi Guys, I have an FTP monitoring/downloading application using C# sockets. I got this error message: 421 Disconnecting you since you were inactive for 300 seconds. Can someone have an explanation for this? I did a search on this one but still I can't seem to find a good explanation. Thanks.

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• How to play MP3's from socket connections in Flash AS3?

  - by Ole Jak

  How to play mp3's from socket connections in Flash AS3? I have some simple programm which is listening to some port (TCP). I connect to it using as3 sockets and it starts sending me mp3 data How to play that data?

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• PHP miniwebsever file download

  - by snikolov

  $httpsock = @socket_create_listen("9090"); if (!$httpsock) { print "Socket creation failed!\n"; exit; } while (1) { $client = socket_accept($httpsock); $input = trim(socket_read ($client, 4096)); $input = explode(" ", $input); $input = $input[1]; $fileinfo = pathinfo($input); switch ($fileinfo['extension']) { default: $mime = "text/html"; } if ($input == "/") { $input = "index.html"; } $input = ".$input"; if (file_exists($input) && is_readable($input)) { echo "Serving $input\n"; $contents = file_get_contents($input); $output = "HTTP/1.0 200 OK\r\nServer: APatchyServer\r\nConnection: close\r\nContent-Type: $mime\r\n\r\n$contents"; } else { //$contents = "The file you requested doesn't exist. Sorry!"; //$output = "HTTP/1.0 404 OBJECT NOT FOUND\r\nServer: BabyHTTP\r\nConnection: close\r\nContent-Type: text/html\r\n\r\n$contents"; function openfile() { $filename = "a.pl"; $file = fopen($filename, 'r'); $filesize = filesize($filename); $buffer = fread($file, $filesize); $array = array("Output"=$buffer,"filesize"=$filesize,"filename"=$filename); return $array; } $send = openfile(); $file = $send['filename']; $filesize = $send['filesize']; $output = 'HTTP/1.0 200 OK\r\n'; $output .= "Content-type: application/octet-stream\r\n"; $output .= 'Content-Disposition: attachment; filename="'.$file.'"\r\n'; $output .= "Content-Length:$filesize\r\n"; $output .= "Accept-Ranges: bytes\r\n"; $output .= "Cache-Control: private\n\n"; $output .= $send['Output']; $output .= "Content-Transfer-Encoding: binary"; $output .= "Connection: Keep-Alive\r\n"; } socket_write($client, $output); socket_close ($client); } socket_close ($httpsock); Hello, I am snikolov i am creating a miniwebserver with php and i would like to know how i can send the client a file to download with his browser such as firefox or internet explore i am sending a file to the user to download via sockets, but the cleint is not getting the filename and the information to download can you please help me here,if i declare the file again i get this error in my server Fatal error: Cannot redeclare openfile() (previously declared in C:\User s\fsfdsf\sfdsfsdf\httpd.php:31) in C:\Users\hfghfgh\hfghg\httpd.php on li ne 29, if its possible, i would like to know if the webserver can show much banwdidth the user request via sockets, perl has the same option as php but its more hardcore than php i dont understand much about perl, i even saw that a miniwebserver can show much the client user pulls from the server would it be possible that you can assist me with this coding, i much aprreciate it thank you guys.

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