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  • facebook app using fbml displays nothing

    - by fusion
    i've made an app in php and html and trying to integrate it with fb. the app on my website is using jquery, but knowing that fbml doesn't support jquery, i've tried to instead use fbjqry. this doesn't work either. i'm not sure where i'm going wrong. /////////////// index.php: <?php // Copyright 2007 Facebook Corp. All Rights Reserved. require_once 'config_fb.php'; //***** Greet the currently logged-in user! echo "<p>Hello, <fb:name uid=\"$user_id\" useyou=\"false\" />!</p>"; include 'quote.html'; ?> ///////////////// quote.html: <!DOCTYPE html> <head> <meta http-equiv="Content-Type" content="text/html; charset=utf-8"> <link rel="stylesheet" type="text/css" href="css/jquote.css" /> <!--<script type="text/javascript" src="scripts/jquery-1.4.2.js"></script>--> <script type="text/javascript" src="fbjqry/utility.js"></script> <script type="text/javascript" src="fbjqry/fjqry.js"></script> <script type="text/javascript"> // On page load, fill the box with content. $(document).ready(function() { $("#quoteContainer").load("quote.php"); }); var auto_refresh = setInterval( function () { $('#quoteContainer').load('quote.php'); }, 5000); // refresh every 10000 milliseconds </script> </head> <div id="wrapper"> <div class="header">&nbsp;Quote of the Day</div> <div id="quoteContainer"> </div> </div> </html> //// from the above file it should take quotes from quote.php and display it, but it doesn't display anything. it seems as though it isn't reading from the quote.php file. is the command of fbjqry different from jquery? if i use iframes instead of fbml, everything loads correctly except that i'd like tab/profile box for this app, which iframes doesn't have.

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  • Performing full screen grab in windows

    - by Steven Lu
    I am working an idea that involves getting a full capture of the screen including windows and apps, analyzing it, and then drawing items back onto the screen, as an overlay. I want to learn image processing techniques and I could get lots of data to work with if I can directly access the Windows screen. I could use this to build automation tools the likes of which have never been seen before. More on that later. I have full screen capture working for the most part. HWND hwind = GetDesktopWindow(); HDC hdc = GetDC(hwind); int resx = GetSystemMetrics(SM_CXSCREEN); int resy = GetSystemMetrics(SM_CYSCREEN); int BitsPerPixel = GetDeviceCaps(hdc,BITSPIXEL); HDC hdc2 = CreateCompatibleDC(hdc); BITMAPINFO info; info.bmiHeader.biSize = sizeof(BITMAPINFOHEADER); info.bmiHeader.biWidth = resx; info.bmiHeader.biHeight = resy; info.bmiHeader.biPlanes = 1; info.bmiHeader.biBitCount = BitsPerPixel; info.bmiHeader.biCompression = BI_RGB; void *data; hbitmap = CreateDIBSection(hdc2,&info,DIB_RGB_COLORS,(void**)&data,0,0); SelectObject(hdc2,hbitmap); Once this is done, I can call this repeatedly: BitBlt(hdc2,0,0,resx,resy,hdc,0,0,SRCCOPY); The cleanup code (I have no idea if this is correct): DeleteObject(hbitmap); ReleaseDC(hwind,hdc); if (hdc2) { DeleteDC(hdc2); } Every time BitBlt is called it grabs the screen and saves it in memory I can access thru data. Performance is somewhat satisfactory. BitBlt executes in 50 milliseconds (sometimes as low as 33ms) at 1920x1200x32. What surprises me is that when I switch display mode to 16 bit, 1920x1200x16, either through my graphics settings beforehand, or by using ChangeDisplaySettings, I get a massively improved screen grab time between 1ms and 2ms, which cannot be explained by the factor of two reduction in bit-depth. Using CreateDIBSection (as above) offers a significant speed up when in 16-bit mode, compared to if I set up with CreateCompatibleBitmap (6-7ms/f). Does anybody know why dropping to 16bit causes such a speed increase? Is there any hope for me to grab 32bit at such speeds? if not for the color depth, but for not forcing a change of screen buffer modes and the awful flickering.

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  • Selecting the contents of an ASP.NET TextBox in an UpdatePanel after a partial page postback

    - by Scott Mitchell
    I am having problems selecting the text within a TextBox in an UpdatePanel. Consider a very simple page that contains a single UpdatePanel. Within that UpdatePanel there are two Web controls: A DropDownList with three statically-defined list items, whose AutoPostBack property is set to True, and A TextBox Web control The DropDownList has a server-side event handler for its SelectedIndexChanged event, and in that event handler there's two lines of code: TextBox1.Text = "Whatever"; ScriptManager.RegisterStartupScript(this, this.GetType(), "Select-" + TextBox1.ClientID, string.Format("document.getElementById('{0}').select();", TextBox1.ClientID), true); The idea is that whenever a user chooses and item from the DropDownList there is a partial page postback, at which point the TextBox's Text property is set and selected (via the injected JavaScript). Unfortunately, this doesn't work as-is. (I have also tried putting the script in the pageLoad function with no luck, as in: ScriptManager.RegisterStartupScript(..., "function pageLoad() { ... my script ... }");) What happens is the code runs, but something else on the page receives focus at the conclusion of the partial page postback, causing the TextBox's text to be unselected. I can "fix" this by using JavaScript's setTimeout to delay the execution of my JavaScript code. For instance, if I update the emitted JavaScript to the following: setTimeout("document.getElementById('{0}').select();", 111); It "works." I put works in quotes because it works for this simple page on my computer. In a more complex page on a slower computer with more markup getting passed between the client and server on the partial page postback, I have to up the timeout to over a second to get it to work. I would hope that there is a more foolproof way to achieve this. Rather than saying, "Delay for X milliseconds," it would be ideal to say, "Run this when you're not going to steal the focus." What's perplexing is that the .Focus() method works beautifully. That is, if I scrap my JavaScript and replace it with a call to TextBox1.Focus(); then the TextBox receives focus (although the text is not selected). I've examined the contents of MicrosoftAjaxWebForms.js and see that the focus is set after the registered scripts run, but I'm my JavaScript skills are not strong enough to decode what all is happening here and why the selected text is unselected between the time it is selected and the end of the partial page postback. I've also tried using Firebug's JavaScript debugger and see that when my script runs the TextBox's text is selected. As I continue to step through it the text remains selected, but then after stepping off the last line of script (apparently) it all of the sudden gets unselected. Any ideas? I am pulling my hair out. Thanks in advance...

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  • CDI SessionScoped Bean results in two instances in same session

    - by Ryan
    I've got two instances of a SessionScoped CDI bean for the same session. I was under the impression that there would be one instance generated for me by CDI, but it generated two. Am I misunderstanding how CDI works, or did I find a bug? Here is the bean code: package org.mycompany.myproject.session; import java.io.Serializable; import javax.enterprise.context.SessionScoped; import javax.faces.context.FacesContext; import javax.inject.Named; import javax.servlet.http.HttpSession; @Named @SessionScoped public class MyBean implements Serializable { private String myField = null; public MyBean() { System.out.println("MyBean constructor called"); FacesContext fc = FacesContext.getCurrentInstance(); HttpSession session = (HttpSession)fc.getExternalContext().getSession(false); String sessionId = session.getId(); System.out.println("Session ID: " + sessionId); } public String getMyField() { return myField; } public void setMyField(String myField) { this.myField = myField; } } Here is the Facelet code: <?xml version='1.0' encoding='UTF-8' ?> <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"> <html xmlns="http://www.w3.org/1999/xhtml" xmlns:h="http://java.sun.com/jsf/html" xmlns:f="http://java.sun.com/jsf/core"> <f:view contentType="text/html" encoding="UTF-8"> <h:head> <title>Test</title> </h:head> <h:body> <h:form id="form"> <h:inputText value="#{myBean.myField}"/> <h:commandButton value="Submit"/> </h:form> </h:body> </f:view> </html> Here is the output from deployment and navigating to page: INFO: Loading application org.mycompany_myproject_war_1.0-SNAPSHOT at /myproject INFO: org.mycompany_myproject_war_1.0-SNAPSHOT was successfully deployed in 8,237 milliseconds. INFO: MyBean constructor called INFO: Session ID: 175355b0e10fe1d0778238bf4634 INFO: MyBean constructor called INFO: Session ID: 175355b0e10fe1d0778238bf4634 Using GlassFish 3.0.1

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  • C++ Undeclared Identifier (but it is declared?)

    - by Joshua
    I'm pretty sure I've included the qanda class, but when I try to declare a vector that contains it or a class of that type I get an error saying that qanda is undefined. Any idea what the problem might be? bot_manager_item.h #pragma once #include "../bot_packet/bot_packet.h" #include <vector> class bot_manager_item; #include "qanda.h" #include "bot_manager.h" class bot_manager_item { public: bot_manager_item(bot_manager* mngr, const char* name, const char* work_dir); ~bot_manager_item(); bool startup(); void cleanup(); void on_push_event(bot_exchange_format f); bool disable; private: void apply_changes(); bot_manager *_mngr; std::string _name; std::string _work_dir; std::string _message; std::string _message_copy; std::vector<qanda> games; qanda test; char _config_full_path[2600]; }; qanda.h #ifndef Q_AND_A #define Q_AND_A #include "users.h" #include "..\bot_packet\bot_packet.h" #include "bot_manager.h" #include <string> #include <algorithm> #include <map> #include <vector> #include <fstream> class qanda { public: qanda(bot_manager * manager, std::string name, std::string directory); ~qanda(){}; void room_message(std::string username, std::string user_message); void timer_tick(); private: // data members std::string question; std::string answer; std::string directory; std::string command_prefix; std::string name; Users users; std::map <std::string, std::string> questions_and_answers; int time_per_question; // seconds int time_between_questions; // seconds int timer; // milliseconds bool is_delayed; bool is_playing; bot_manager * manager; // functions void new_question(); void send_message(std::string msg); void announce_question(); void load_questions(); }; #endif

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  • Thread sleep and thread join.

    - by Dhruv Gairola
    hi guys, if i put a thread to sleep in a loop, netbeans gives me a caution saying Invoking Thread.sleep in loop can cause performance problems. However, if i were to replace the sleep with join, no such caution is given. Both versions compile and work fine tho. My code is below (check the last few lines for "Thread.sleep() vs t.join()"). public class Test{ //Display a message, preceded by the name of the current thread static void threadMessage(String message) { String threadName = Thread.currentThread().getName(); System.out.format("%s: %s%n", threadName, message); } private static class MessageLoop implements Runnable { public void run() { String importantInfo[] = { "Mares eat oats", "Does eat oats", "Little lambs eat ivy", "A kid will eat ivy too" }; try { for (int i = 0; i < importantInfo.length; i++) { //Pause for 4 seconds Thread.sleep(4000); //Print a message threadMessage(importantInfo[i]); } } catch (InterruptedException e) { threadMessage("I wasn't done!"); } } } public static void main(String args[]) throws InterruptedException { //Delay, in milliseconds before we interrupt MessageLoop //thread (default one hour). long patience = 1000 * 60 * 60; //If command line argument present, gives patience in seconds. if (args.length > 0) { try { patience = Long.parseLong(args[0]) * 1000; } catch (NumberFormatException e) { System.err.println("Argument must be an integer."); System.exit(1); } } threadMessage("Starting MessageLoop thread"); long startTime = System.currentTimeMillis(); Thread t = new Thread(new MessageLoop()); t.start(); threadMessage("Waiting for MessageLoop thread to finish"); //loop until MessageLoop thread exits while (t.isAlive()) { threadMessage("Still waiting..."); //Wait maximum of 1 second for MessageLoop thread to //finish. /*******LOOK HERE**********************/ Thread.sleep(1000);//issues caution unlike t.join(1000) /**************************************/ if (((System.currentTimeMillis() - startTime) > patience) && t.isAlive()) { threadMessage("Tired of waiting!"); t.interrupt(); //Shouldn't be long now -- wait indefinitely t.join(); } } threadMessage("Finally!"); } } As i understand it, join waits for the other thread to complete, but in this case, arent both sleep and join doing the same thing? Then why does netbeans throw the caution?

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  • Can MySQL reasonably perform queries on billions of rows?

    - by haxney
    I am planning on storing scans from a mass spectrometer in a MySQL database and would like to know whether storing and analyzing this amount of data is remotely feasible. I know performance varies wildly depending on the environment, but I'm looking for the rough order of magnitude: will queries take 5 days or 5 milliseconds? Input format Each input file contains a single run of the spectrometer; each run is comprised of a set of scans, and each scan has an ordered array of datapoints. There is a bit of metadata, but the majority of the file is comprised of arrays 32- or 64-bit ints or floats. Host system |----------------+-------------------------------| | OS | Windows 2008 64-bit | | MySQL version | 5.5.24 (x86_64) | | CPU | 2x Xeon E5420 (8 cores total) | | RAM | 8GB | | SSD filesystem | 500 GiB | | HDD RAID | 12 TiB | |----------------+-------------------------------| There are some other services running on the server using negligible processor time. File statistics |------------------+--------------| | number of files | ~16,000 | | total size | 1.3 TiB | | min size | 0 bytes | | max size | 12 GiB | | mean | 800 MiB | | median | 500 MiB | | total datapoints | ~200 billion | |------------------+--------------| The total number of datapoints is a very rough estimate. Proposed schema I'm planning on doing things "right" (i.e. normalizing the data like crazy) and so would have a runs table, a spectra table with a foreign key to runs, and a datapoints table with a foreign key to spectra. The 200 Billion datapoint question I am going to be analyzing across multiple spectra and possibly even multiple runs, resulting in queries which could touch millions of rows. Assuming I index everything properly (which is a topic for another question) and am not trying to shuffle hundreds of MiB across the network, is it remotely plausible for MySQL to handle this? UPDATE: additional info The scan data will be coming from files in the XML-based mzML format. The meat of this format is in the <binaryDataArrayList> elements where the data is stored. Each scan produces = 2 <binaryDataArray> elements which, taken together, form a 2-dimensional (or more) array of the form [[123.456, 234.567, ...], ...]. These data are write-once, so update performance and transaction safety are not concerns. My naïve plan for a database schema is: runs table | column name | type | |-------------+-------------| | id | PRIMARY KEY | | start_time | TIMESTAMP | | name | VARCHAR | |-------------+-------------| spectra table | column name | type | |----------------+-------------| | id | PRIMARY KEY | | name | VARCHAR | | index | INT | | spectrum_type | INT | | representation | INT | | run_id | FOREIGN KEY | |----------------+-------------| datapoints table | column name | type | |-------------+-------------| | id | PRIMARY KEY | | spectrum_id | FOREIGN KEY | | mz | DOUBLE | | num_counts | DOUBLE | | index | INT | |-------------+-------------| Is this reasonable?

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  • How to handle very frequent updates to a Lucene index

    - by fsm
    I am trying to prototype an indexing/search application which uses very volatile indexing data sources (forums, social networks etc), here are some of the performance requirements, Very fast turn-around time (by this I mean that any new data (such as a new message on a forum) should be available in the search results very soon (less than a minute)) I need to discard old documents on a fairly regular basis to ensure that the search results are not dated. Last but not least, the search application needs to be responsive. (latency on the order of 100 milliseconds, and should support at least 10 qps) All of the requirements I have currently can be met w/o using Lucene (and that would let me satisfy all 1,2 and 3), but I am anticipating other requirements in the future (like search relevance etc) which Lucene makes easier to implement. However, since Lucene is designed for use cases far more complex than the one I'm currently working on, I'm having a hard time satisfying my performance requirements. Here are some questions, a. I read that the optimize() method in the IndexWriter class is expensive, and should not be used by applications that do frequent updates, what are the alternatives? b. In order to do incremental updates, I need to keep committing new data, and also keep refreshing the index reader to make sure it has the new data available. These are going to affect 1 and 3 above. Should I try duplicate indices? What are some common approaches to solving this problem? c. I know that Lucene provides a delete method, which lets you delete all documents that match a certain query, in my case, I need to delete all documents which are older than a certain age, now one option is to add a date field to every document and use that to delete documents later. Is it possible to do range queries on document ids (I can create my own id field since I think that the one created by lucene keeps changing) to delete documents? Is it any faster than comparing dates represented as strings? I know these are very open questions, so I am not looking for a detailed answer, I will try to treat all of your answers as suggestions and use them to inform my design. Thanks! Please let me know if you need any other information.

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  • Thread class closing from other Class (Activity) with protected void onStop() Android

    - by user1761337
    I have a Problem with Closing the Thread. I will Closing the Thread with onStop,onPause and onDestroy. This is my Source in the Activity Class: @Override protected void onStop(){ super.onStop(); finish(); } @Override protected void onPause() { super.onPause(); finish(); } @Override public void onDestroy() { this.mWakeLock.release(); super.onDestroy(); } And the Thread Class: public class GameThread extends Thread { private SurfaceHolder mSurfaceHolder; private Handler mHandler; private Context mContext; private Paint mLinePaint; private Paint blackPaint; //for consistent rendering private long sleepTime; //amount of time to sleep for (in milliseconds) private long delay=1000/30; //state of game (Running or Paused). int state = 1; public final static int RUNNING = 1; public final static int PAUSED = 2; public final static int STOPED = 3; GameSurface gEngine; public GameThread(SurfaceHolder surfaceHolder, Context context, Handler handler,GameSurface gEngineS){ //data about the screen mSurfaceHolder = surfaceHolder; mHandler = handler; mContext = context; gEngine=gEngineS; } //This is the most important part of the code. It is invoked when the call to start() is //made from the SurfaceView class. It loops continuously until the game is finished or //the application is suspended. private long beforeTime; @Override public void run() { //UPDATE while (state==RUNNING) { Log.d("State","Thread is runnig"); //time before update beforeTime = System.nanoTime(); //This is where we update the game engine gEngine.Update(); //DRAW Canvas c = null; try { //lock canvas so nothing else can use it c = mSurfaceHolder.lockCanvas(null); synchronized (mSurfaceHolder) { //clear the screen with the black painter. //reset the canvas c.drawColor(Color.BLACK); //This is where we draw the game engine. gEngine.doDraw(c); } } finally { // do this in a finally so that if an exception is thrown // during the above, we don't leave the Surface in an // inconsistent state if (c != null) { mSurfaceHolder.unlockCanvasAndPost(c); } } this.sleepTime = delay-((System.nanoTime()-beforeTime)/1000000L); try { //actual sleep code if(sleepTime>0){ this.sleep(sleepTime); } } catch (InterruptedException ex) { Logger.getLogger(GameThread.class.getName()).log(Level.SEVERE, null, ex); } while (state==PAUSED){ Log.d("State","Thread is pausing"); try { this.sleep(1000); } catch (InterruptedException e) { // TODO Auto-generated catch block e.printStackTrace(); } } } }} How i can close the Thread from Activity Class??

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  • Active Directory Time Synchronisation - Time-Service Event ID 50

    - by George
    I have an Active Directory domain with two DCs. The first DC in the forest/domain is Server 2012, the second is 2008 R2. The first DC holds the PDC Emulator role. I sporadically receive a warning from the Time-Service source, event ID 50: The time service detected a time difference of greater than %1 milliseconds for %2 seconds. The time difference might be caused by synchronization with low-accuracy time sources or by suboptimal network conditions. The time service is no longer synchronized and cannot provide the time to other clients or update the system clock. When a valid time stamp is received from a time service provider, the time service will correct itself. Time sync in the domain is configured with the second DC to synchronise using the /syncfromflags:DOMHIER flag. The first DC is configured to sync time using a /syncfromflags:MANUAL /reliable:YES, from a peerlist consisting of a number of UK based stratum 2 servers, such as ntp2d.mcc.ac.uk. I'm confused why I receive this event warning. It implies that my PDC emulator cannot synchronise time with a supposedly reliable external time source, and it quotes a time difference of 5 seconds for 900 seconds. It's worth also mentioning that I used to use a UK pool from ntp.org but I would receive the warning much more often. Since updating to a number of UK based academic time servers, it seems to be more reliable. Can someone with more experience shed some light on this - perhaps it is purely transient? Should I disregard the warning? Is my configuration sound? EDIT: I should add that the DCs are virtual, and installed on two separate VMware ESXi/vSphere physical hosts. I can also confirm that as per MDMarra's comment and best practice, VMware timesync is disabled, since: c:\Program Files\VMware\VMware Tools\VMwareToolboxCmd.exe timesync status returns Disabled. EDIT 2 Some strange new issue has cropped up. I've noticed a pattern. Originally, the event ID 50 warnings would occur at about 1230pm each day. This is interesting since our veeam backup happens at 12 midday. Since I made the changes discussed here, I now receive an event ID 51 instead of 50. The new warning says that: The time sample received from peer server.ac.uk differs from the local time by -40 seconds (Or approximately 40 seconds). This has happened two days in a row. Now I'm even more confused. Obviously the time never updates until I manually intervene. The issue seems to be related to virtualisation and veeam. Something may be occuring when veeam is backing up the PDCe. Any suggestions? UPDATE & SUMMARY msemack's excellent list of resources below (the accepted answer) provided enough information to correctly configure the time service in the domain. This should be the first port of call for any future people looking to verify their configuration. The final "40 second jump" issue I have resolved (there are no more warnings) through adjusting the VMware time sync settings as noted in the veeam knowledge base article here: http://www.veeam.com/kb1202 In any case, should any future reader use ESXi, veeam or not, the resources here are an excellent source of information on the time sync topic and msemack's answer is particularly invaluable.

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  • WebSocket and Java EE 7 - Getting Ready for JSR 356 (TOTD #181)

    - by arungupta
    WebSocket is developed as part of HTML 5 specification and provides a bi-directional, full-duplex communication channel over a single TCP socket. It provides dramatic improvement over the traditional approaches of Polling, Long-Polling, and Streaming for two-way communication. There is no latency from establishing new TCP connections for each HTTP message. There is a WebSocket API and the WebSocket Protocol. The Protocol defines "handshake" and "framing". The handshake defines how a normal HTTP connection can be upgraded to a WebSocket connection. The framing defines wire format of the message. The design philosophy is to keep the framing minimum to avoid the overhead. Both text and binary data can be sent using the API. WebSocket may look like a competing technology to Server-Sent Events (SSE), but they are not. Here are the key differences: WebSocket can send and receive data from a client. A typical example of WebSocket is a two-player game or a chat application. Server-Sent Events can only push data data to the client. A typical example of SSE is stock ticker or news feed. With SSE, XMLHttpRequest can be used to send data to the server. For server-only updates, WebSockets has an extra overhead and programming can be unecessarily complex. SSE provides a simple and easy-to-use model that is much better suited. SSEs are sent over traditional HTTP and so no modification is required on the server-side. WebSocket require servers that understand the protocol. SSE have several features that are missing from WebSocket such as automatic reconnection, event IDs, and the ability to send arbitrary events. The client automatically tries to reconnect if the connection is closed. The default wait before trying to reconnect is 3 seconds and can be configured by including "retry: XXXX\n" header where XXXX is the milliseconds to wait before trying to reconnect. Event stream can include a unique event identifier. This allows the server to determine which events need to be fired to each client in case the connection is dropped in between. The data can span multiple lines and can be of any text format as long as EventSource message handler can process it. WebSockets provide true real-time updates, SSE can be configured to provide close to real-time by setting appropriate timeouts. OK, so all excited about WebSocket ? Want to convert your POJOs into WebSockets endpoint ? websocket-sdk and GlassFish 4.0 is here to help! The complete source code shown in this project can be downloaded here. On the server-side, the WebSocket SDK converts a POJO into a WebSocket endpoint using simple annotations. Here is how a WebSocket endpoint will look like: @WebSocket(path="/echo")public class EchoBean { @WebSocketMessage public String echo(String message) { return message + " (from your server)"; }} In this code "@WebSocket" is a class-level annotation that declares a POJO to accept WebSocket messages. The path at which the messages are accepted is specified in this annotation. "@WebSocketMessage" indicates the Java method that is invoked when the endpoint receives a message. This method implementation echoes the received message concatenated with an additional string. The client-side HTML page looks like <div style="text-align: center;"> <form action=""> <input onclick="send_echo()" value="Press me" type="button"> <input id="textID" name="message" value="Hello WebSocket!" type="text"><br> </form></div><div id="output"></div> WebSocket allows a full-duplex communication. So the client, a browser in this case, can send a message to a server, a WebSocket endpoint in this case. And the server can send a message to the client at the same time. This is unlike HTTP which follows a "request" followed by a "response". In this code, the "send_echo" method in the JavaScript is invoked on the button click. There is also a <div> placeholder to display the response from the WebSocket endpoint. The JavaScript looks like: <script language="javascript" type="text/javascript"> var wsUri = "ws://localhost:8080/websockets/echo"; var websocket = new WebSocket(wsUri); websocket.onopen = function(evt) { onOpen(evt) }; websocket.onmessage = function(evt) { onMessage(evt) }; websocket.onerror = function(evt) { onError(evt) }; function init() { output = document.getElementById("output"); } function send_echo() { websocket.send(textID.value); writeToScreen("SENT: " + textID.value); } function onOpen(evt) { writeToScreen("CONNECTED"); } function onMessage(evt) { writeToScreen("RECEIVED: " + evt.data); } function onError(evt) { writeToScreen('<span style="color: red;">ERROR:</span> ' + evt.data); } function writeToScreen(message) { var pre = document.createElement("p"); pre.style.wordWrap = "break-word"; pre.innerHTML = message; output.appendChild(pre); } window.addEventListener("load", init, false);</script> In this code The URI to connect to on the server side is of the format ws://<HOST>:<PORT>/websockets/<PATH> "ws" is a new URI scheme introduced by the WebSocket protocol. <PATH> is the path on the endpoint where the WebSocket messages are accepted. In our case, it is ws://localhost:8080/websockets/echo WEBSOCKET_SDK-1 will ensure that context root is included in the URI as well. WebSocket is created as a global object so that the connection is created only once. This object establishes a connection with the given host, port and the path at which the endpoint is listening. The WebSocket API defines several callbacks that can be registered on specific events. The "onopen", "onmessage", and "onerror" callbacks are registered in this case. The callbacks print a message on the browser indicating which one is called and additionally also prints the data sent/received. On the button click, the WebSocket object is used to transmit text data to the endpoint. Binary data can be sent as one blob or using buffering. The HTTP request headers sent for the WebSocket call are: GET ws://localhost:8080/websockets/echo HTTP/1.1Origin: http://localhost:8080Connection: UpgradeSec-WebSocket-Extensions: x-webkit-deflate-frameHost: localhost:8080Sec-WebSocket-Key: mDbnYkAUi0b5Rnal9/cMvQ==Upgrade: websocketSec-WebSocket-Version: 13 And the response headers received are Connection:UpgradeSec-WebSocket-Accept:q4nmgFl/lEtU2ocyKZ64dtQvx10=Upgrade:websocket(Challenge Response):00:00:00:00:00:00:00:00:00:00:00:00:00:00:00:00 The headers are shown in Chrome as shown below: The complete source code shown in this project can be downloaded here. The builds from websocket-sdk are integrated in GlassFish 4.0 builds. Would you like to live on the bleeding edge ? Then follow the instructions below to check out the workspace and install the latest SDK: Check out the source code svn checkout https://svn.java.net/svn/websocket-sdk~source-code-repository Build and install the trunk in your local repository as: mvn install Copy "./bundles/websocket-osgi/target/websocket-osgi-0.3-SNAPSHOT.jar" to "glassfish3/glassfish/modules/websocket-osgi.jar" in your GlassFish 4 latest promoted build. Notice, you need to overwrite the JAR file. Anybody interested in building a cool application using WebSocket and get it running on GlassFish ? :-) This work will also feed into JSR 356 - Java API for WebSocket. On a lighter side, there seems to be less agreement on the name. Here are some of the options that are prevalent: WebSocket (W3C API, the URL is www.w3.org/TR/websockets though) Web Socket (HTML5 Demos - html5demos.com/web-socket) Websocket (Jenkins Plugin - wiki.jenkins-ci.org/display/JENKINS/Websocket%2BPlugin) WebSockets (Used by Mozilla - developer.mozilla.org/en/WebSockets, but use WebSocket as well) Web sockets (HTML5 Working Group - www.whatwg.org/specs/web-apps/current-work/multipage/network.html) Web Sockets (Chrome Blog - blog.chromium.org/2009/12/web-sockets-now-available-in-google.html) I prefer "WebSocket" as that seems to be most common usage and used by the W3C API as well. What do you use ?

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  • Tricks and Optimizations for you Sitecore website

    - by amaniar
    When working with Sitecore there are some optimizations/configurations I usually repeat in order to make my app production ready. Following is a small list I have compiled from experience, Sitecore documentation, communicating with Sitecore Engineers etc. This is not supposed to be technically complete and might not be fit for all environments.   Simple configurations that can make a difference: 1) Configure Sitecore Caches. This is the most straight forward and sure way of increasing the performance of your website. Data and item cache sizes (/databases/database/ [id=web] ) should be configured as needed. You may start with a smaller number and tune them as needed. <cacheSizes hint="setting"> <data>300MB</data> <items>300MB</items> <paths>5MB</paths> <standardValues>5MB</standardValues> </cacheSizes> Tune the html, registry etc cache sizes for your website.   <cacheSizes> <sites> <website> <html>300MB</html> <registry>1MB</registry> <viewState>10MB</viewState> <xsl>5MB</xsl> </website> </sites> </cacheSizes> Tune the prefetch cache settings under the App_Config/Prefetch/ folder. Sample /App_Config/Prefetch/Web.Config: <configuration> <cacheSize>300MB</cacheSize> <!--preload items that use this template--> <template desc="mytemplate">{XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXXXX}</template> <!--preload this item--> <item desc="myitem">{XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXXXX }</item> <!--preload children of this item--> <children desc="childitems">{XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXXXX}</children> </configuration> Break your page into sublayouts so you may cache most of them. Read the caching configuration reference: http://sdn.sitecore.net/upload/sitecore6/sc62keywords/cache_configuration_reference_a4.pdf   2) Disable Analytics for the Shell Site <site name="shell" virtualFolder="/sitecore/shell" physicalFolder="/sitecore/shell" rootPath="/sitecore/content" startItem="/home" language="en" database="core" domain="sitecore" loginPage="/sitecore/login" content="master" contentStartItem="/Home" enableWorkflow="true" enableAnalytics="false" xmlControlPage="/sitecore/shell/default.aspx" browserTitle="Sitecore" htmlCacheSize="2MB" registryCacheSize="3MB" viewStateCacheSize="200KB" xslCacheSize="5MB" />   3) Increase the Check Interval for the MemoryMonitorHook so it doesn’t run every 5 seconds (default). <hook type="Sitecore.Diagnostics.MemoryMonitorHook, Sitecore.Kernel"> <param desc="Threshold">800MB</param> <param desc="Check interval">00:05:00</param> <param desc="Minimum time between log entries">00:01:00</param> <ClearCaches>false</ClearCaches> <GarbageCollect>false</GarbageCollect> <AdjustLoadFactor>false</AdjustLoadFactor> </hook>   4) Set Analytics.PeformLookup (Sitecore.Analytics.config) to false if your environment doesn’t have access to the internet or you don’t intend to use reverse DNS lookup. <setting name="Analytics.PerformLookup" value="false" />   5) Set the value of the “Media.MediaLinkPrefix” setting to “-/media”: <setting name="Media.MediaLinkPrefix" value="-/media" /> Add the following line to the customHandlers section: <customHandlers> <handler trigger="-/media/" handler="sitecore_media.ashx" /> <handler trigger="~/media/" handler="sitecore_media.ashx" /> <handler trigger="~/api/" handler="sitecore_api.ashx" /> <handler trigger="~/xaml/" handler="sitecore_xaml.ashx" /> <handler trigger="~/icon/" handler="sitecore_icon.ashx" /> <handler trigger="~/feed/" handler="sitecore_feed.ashx" /> </customHandlers> Link: http://squad.jpkeisala.com/2011/10/sitecore-media-library-performance-optimization-checklist/   6) Performance counters should be disabled in production if not being monitored <setting name="Counters.Enabled" value="false" />   7) Disable Item/Memory/Timing threshold warnings. Due to the nature of this component, it brings no value in production. <!--<processor type="Sitecore.Pipelines.HttpRequest.StartMeasurements, Sitecore.Kernel" />--> <!--<processor type="Sitecore.Pipelines.HttpRequest.StopMeasurements, Sitecore.Kernel"> <TimingThreshold desc="Milliseconds">1000</TimingThreshold> <ItemThreshold desc="Item count">1000</ItemThreshold> <MemoryThreshold desc="KB">10000</MemoryThreshold> </processor>—>   8) The ContentEditor.RenderCollapsedSections setting is a hidden setting in the web.config file, which by default is true. Setting it to false will improve client performance for authoring environments. <setting name="ContentEditor.RenderCollapsedSections" value="false" />   9) Add a machineKey section to your Web.Config file when using a web farm. Link: http://msdn.microsoft.com/en-us/library/ff649308.aspx   10) If you get errors in the log files similar to: WARN Could not create an instance of the counter 'XXX.XXX' (category: 'Sitecore.System') Exception: System.UnauthorizedAccessException Message: Access to the registry key 'Global' is denied. Make sure the ApplicationPool user is a member of the system “Performance Monitor Users” group on the server.   11) Disable WebDAV configurations on the CD Server if not being used. More: http://sitecoreblog.alexshyba.com/2011/04/disable-webdav-in-sitecore.html   12) Change Log4Net settings to only log Errors on content delivery environments to avoid unnecessary logging. <root> <priority value="ERROR" /> <appender-ref ref="LogFileAppender" /> </root>   13) Disable Analytics for any content item that doesn’t add value. For example a page that redirects to another page.   14) When using Web User Controls avoid registering them on the page the asp.net way: <%@ Register Src="~/layouts/UserControls/MyControl.ascx" TagName="MyControl" TagPrefix="uc2" %> Use Sublayout web control instead – This way Sitecore caching could be leveraged <sc:Sublayout ID="ID" Path="/layouts/UserControls/MyControl.ascx" Cacheable="true" runat="server" />   15) Avoid querying for all children recursively when all items are direct children. Sitecore.Context.Database.SelectItems("/sitecore/content/Home//*"); //Use: Sitecore.Context.Database.GetItem("/sitecore/content/Home");   16) On IIS — you enable static & dynamic content compression on CM and CD More: http://technet.microsoft.com/en-us/library/cc754668%28WS.10%29.aspx   17) Enable HTTP Keep-alive and content expiration in IIS.   18) Use GUID’s when accessing items and fields instead of names or paths. Its faster and wont break your code when things get moved or renamed. Context.Database.GetItem("{324DFD16-BD4F-4853-8FF1-D663F6422DFF}") Context.Item.Fields["{89D38A8F-394E-45B0-826B-1A826CF4046D}"]; //is better than Context.Database.GetItem("/Home/MyItem") Context.Item.Fields["FieldName"]   Hope this helps.

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  • The Proper Use of the VM Role in Windows Azure

    - by BuckWoody
    At the Professional Developer’s Conference (PDC) in 2010 we announced an addition to the Computational Roles in Windows Azure, called the VM Role. This new feature allows a great deal of control over the applications you write, but some have confused it with our full infrastructure offering in Windows Hyper-V. There is a proper architecture pattern for both of them. Virtualization Virtualization is the process of taking all of the hardware of a physical computer and replicating it in software alone. This means that a single computer can “host” or run several “virtual” computers. These virtual computers can run anywhere - including at a vendor’s location. Some companies refer to this as Cloud Computing since the hardware is operated and maintained elsewhere. IaaS The more detailed definition of this type of computing is called Infrastructure as a Service (Iaas) since it removes the need for you to maintain hardware at your organization. The operating system, drivers, and all the other software required to run an application are still under your control and your responsibility to license, patch, and scale. Microsoft has an offering in this space called Hyper-V, that runs on the Windows operating system. Combined with a hardware hosting vendor and the System Center software to create and deploy Virtual Machines (a process referred to as provisioning), you can create a Cloud environment with full control over all aspects of the machine, including multiple operating systems if you like. Hosting machines and provisioning them at your own buildings is sometimes called a Private Cloud, and hosting them somewhere else is often called a Public Cloud. State-ful and Stateless Programming This paradigm does not create a new, scalable way of computing. It simply moves the hardware away. The reason is that when you limit the Cloud efforts to a Virtual Machine, you are in effect limiting the computing resources to what that single system can provide. This is because much of the software developed in this environment maintains “state” - and that requires a little explanation. “State-ful programming” means that all parts of the computing environment stay connected to each other throughout a compute cycle. The system expects the memory, CPU, storage and network to remain in the same state from the beginning of the process to the end. You can think of this as a telephone conversation - you expect that the other person picks up the phone, listens to you, and talks back all in a single unit of time. In “Stateless” computing the system is designed to allow the different parts of the code to run independently of each other. You can think of this like an e-mail exchange. You compose an e-mail from your system (it has the state when you’re doing that) and then you walk away for a bit to make some coffee. A few minutes later you click the “send” button (the network has the state) and you go to a meeting. The server receives the message and stores it on a mail program’s database (the mail server has the state now) and continues working on other mail. Finally, the other party logs on to their mail client and reads the mail (the other user has the state) and responds to it and so on. These events might be separated by milliseconds or even days, but the system continues to operate. The entire process doesn’t maintain the state, each component does. This is the exact concept behind coding for Windows Azure. The stateless programming model allows amazing rates of scale, since the message (think of the e-mail) can be broken apart by multiple programs and worked on in parallel (like when the e-mail goes to hundreds of users), and only the order of re-assembling the work is important to consider. For the exact same reason, if the system makes copies of those running programs as Windows Azure does, you have built-in redundancy and recovery. It’s just built into the design. The Difference Between Infrastructure Designs and Platform Designs When you simply take a physical server running software and virtualize it either privately or publicly, you haven’t done anything to allow the code to scale or have recovery. That all has to be handled by adding more code and more Virtual Machines that have a slight lag in maintaining the running state of the system. Add more machines and you get more lag, so the scale is limited. This is the primary limitation with IaaS. It’s also not as easy to deploy these VM’s, and more importantly, you’re often charged on a longer basis to remove them. your agility in IaaS is more limited. Windows Azure is a Platform - meaning that you get objects you can code against. The code you write runs on multiple nodes with multiple copies, and it all works because of the magic of Stateless programming. you don’t worry, or even care, about what is running underneath. It could be Windows (and it is in fact a type of Windows Server), Linux, or anything else - but that' isn’t what you want to manage, monitor, maintain or license. You don’t want to deploy an operating system - you want to deploy an application. You want your code to run, and you don’t care how it does that. Another benefit to PaaS is that you can ask for hundreds or thousands of new nodes of computing power - there’s no provisioning, it just happens. And you can stop using them quicker - and the base code for your application does not have to change to make this happen. Windows Azure Roles and Their Use If you need your code to have a user interface, in Visual Studio you add a Web Role to your project, and if the code needs to do work that doesn’t involve a user interface you can add a Worker Role. They are just containers that act a certain way. I’ll provide more detail on those later. Note: That’s a general description, so it’s not entirely accurate, but it’s accurate enough for this discussion. So now we’re back to that VM Role. Because of the name, some have mistakenly thought that you can take a Virtual Machine running, say Linux, and deploy it to Windows Azure using this Role. But you can’t. That’s not what it is designed for at all. If you do need that kind of deployment, you should look into Hyper-V and System Center to create the Private or Public Infrastructure as a Service. What the VM Role is actually designed to do is to allow you to have a great deal of control over the system where your code will run. Let’s take an example. You’ve heard about Windows Azure, and Platform programming. You’re convinced it’s the right way to code. But you have a lot of things you’ve written in another way at your company. Re-writing all of your code to take advantage of Windows Azure will take a long time. Or perhaps you have a certain version of Apache Web Server that you need for your code to work. In both cases, you think you can (or already have) code the the software to be “Stateless”, you just need more control over the place where the code runs. That’s the place where a VM Role makes sense. Recap Virtualizing servers alone has limitations of scale, availability and recovery. Microsoft’s offering in this area is Hyper-V and System Center, not the VM Role. The VM Role is still used for running Stateless code, just like the Web and Worker Roles, with the exception that it allows you more control over the environment of where that code runs.

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  • API Message Localization

    - by Jesse Taber
    In my post, “Keep Localizable Strings Close To Your Users” I talked about the internationalization and localization difficulties that can arise when you sprinkle static localizable strings throughout the different logical layers of an application. The main point of that post is that you should have your localizable strings reside as close to the user-facing modules of your application as possible. For example, if you’re developing an ASP .NET web forms application all of the localizable strings should be kept in .resx files that are associated with the .aspx views of the application. In this post I want to talk about how this same concept can be applied when designing and developing APIs. An API Facilitates Machine-to-Machine Interaction You can typically think about a web, desktop, or mobile application as a collection “views” or “screens” through which users interact with the underlying logic and data. The application can be designed based on the assumption that there will be a human being on the other end of the screen working the controls. You are designing a machine-to-person interaction and the application should be built in a way that facilitates the user’s clear understanding of what is going on. Dates should be be formatted in a way that the user will be familiar with, messages should be presented in the user’s preferred language, etc. When building an API, however, there are no screens and you can’t make assumptions about who or what is on the other end of each call. An API is, by definition, a machine-to-machine interaction. A machine-to-machine interaction should be built in a way that facilitates a clear and unambiguous understanding of what is going on. Dates and numbers should be formatted in predictable and standard ways (e.g. ISO 8601 dates) and messages should be presented in machine-parseable formats. For example, consider an API for a time tracking system that exposes a resource for creating a new time entry. The JSON for creating a new time entry for a user might look like: 1: { 2: "userId": 4532, 3: "startDateUtc": "2012-10-22T14:01:54.98432Z", 4: "endDateUtc": "2012-10-22T11:34:45.29321Z" 5: }   Note how the parameters for start and end date are both expressed as ISO 8601 compliant dates in UTC. Using a date format like this in our API leaves little room for ambiguity. It’s also important to note that using ISO 8601 dates is a much, much saner thing than the \/Date(<milliseconds since epoch>)\/ nonsense that is sometimes used in JSON serialization. Probably the most important thing to note about the JSON snippet above is the fact that the end date comes before the start date! The API should recognize that and disallow the time entry from being created, returning an error to the caller. You might inclined to send a response that looks something like this: 1: { 2: "errors": [ {"message" : "The end date must come after the start date"}] 3: }   While this may seem like an appropriate thing to do there are a few problems with this approach: What if there is a user somewhere on the other end of the API call that doesn’t speak English?  What if the message provided here won’t fit properly within the UI of the application that made the API call? What if the verbiage of the message isn’t consistent with the rest of the application that made the API call? What if there is no user directly on the other end of the API call (e.g. this is a batch job uploading time entries once per night unattended)? The API knows nothing about the context from which the call was made. There are steps you could take to given the API some context (e.g.allow the caller to send along a language code indicating the language that the end user speaks), but that will only get you so far. As the designer of the API you could make some assumptions about how the API will be called, but if we start making assumptions we could very easily make the wrong assumptions. In this situation it’s best to make no assumptions and simply design the API in such a way that the caller has the responsibility to convey error messages in a manner that is appropriate for the context in which the error was raised. You would work around some of these problems by allowing callers to add metadata to each request describing the context from which the call is being made (e.g. accepting a ‘locale’ parameter denoting the desired language), but that will add needless clutter and complexity. It’s better to keep the API simple and push those context-specific concerns down to the caller whenever possible. For our very simple time entry example, this can be done by simply changing our error message response to look like this: 1: { 2: "errors": [ {"code": 100}] 3: }   By changing our error error from exposing a string to a numeric code that is easily parseable by another application, we’ve placed all of the responsibility for conveying the actual meaning of the error message on the caller. It’s best to have the caller be responsible for conveying this meaning because the caller understands the context much better than the API does. Now the caller can see error code 100, know that it means that the end date submitted falls before the start date and take appropriate action. Now all of the problems listed out above are non-issues because the caller can simply translate the error code of ‘100’ into the proper action and message for the current context. The numeric code representation of the error is a much better way to facilitate the machine-to-machine interaction that the API is meant to facilitate. An API Does Have Human Users While APIs should be built for machine-to-machine interaction, people still need to wire these interactions together. As a programmer building a client application that will consume the time entry API I would find it frustrating to have to go dig through the API documentation every time I encounter a new error code (assuming the documentation exists and is accurate). The numeric error code approach hurts the discoverability of the API and makes it painful to integrate with. We can help ease this pain by merging our two approaches: 1: { 2: "errors": [ {"code": 100, "message" : "The end date must come after the start date"}] 3: }   Now we have an easily parseable numeric error code for the machine-to-machine interaction that the API is meant to facilitate and a human-readable message for programmers working with the API. The human-readable message here is not intended to be viewed by end-users of the API and as such is not really a “localizable string” in my opinion. We could opt to expose a locale parameter for all API methods and store translations for all error messages, but that’s a lot of extra effort and overhead that doesn’t add a lot real value to the API. I might be a bit of an “ugly American”, but I think it’s probably fine to have the API return English messages when the target for those messages is a programmer. When resources are limited (which they always are), I’d argue that you’re better off hard-coding these messages in English and putting more effort into building more useful features, improving security, tweaking performance, etc.

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  • XNA 2D Collision with specific tiles

    - by zenzero
    I am new to game programming and to these sites for help. I am making a 2D game but I can't seem to get the collision between my character and certain tiles. I have a map filled with grass tiles and water tiles and I want to keep my character from walking on the water tiles. I have a Tiles class that I use so that the tiles are objects and also has the collision method in it, a TileEngine class used create the map and it also holds a list of Tiles, and the class James which is for my character. I also have a Camera class that centers the camera on my character if that has anything to do with the problem. The character's movement is intended to be restricted to 4 directions(up, down, left, right). As an extra note, the bottom right water tile does have collision, but the collision does not occur for any of the other water tiles. Here is my TileEngine class using System; using System.Collections.Generic; using System.Linq; using Microsoft.Xna.Framework; using Microsoft.Xna.Framework.Audio; using Microsoft.Xna.Framework.Content; using Microsoft.Xna.Framework.GamerServices; using Microsoft.Xna.Framework.Graphics; using Microsoft.Xna.Framework.Input; using Microsoft.Xna.Framework.Media; namespace Test2DGame2 { class TileEngine : Microsoft.Xna.Framework.Game { //makes a list of Tiles objects public List<Tiles> tilesList = new List<Tiles>(); public TileEngine() {} public static int tileWidth = 64; public static int tileHeight = 64; public int[,] map = { {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, }, {0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, }, {0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 0, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 1, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,}, {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,}, }; public void drawMap(SpriteBatch spriteBatch) { for (int y = 0; y < map.GetLength(0); y++) { for (int x = 0; x < map.GetLength(1); x++) { //make a Rectangle tilesList[map[y, x]].rectangle = new Rectangle(x * tileWidth, y * tileHeight, tileWidth, tileHeight); //draw the Tiles objects spriteBatch.Draw(tilesList[map[y, x]].texture, tilesList[map[y, x]].rectangle, Color.White); } } } } } Here is my Tiles class using System; using System.Collections.Generic; using System.Linq; using Microsoft.Xna.Framework; using Microsoft.Xna.Framework.Audio; using Microsoft.Xna.Framework.Content; using Microsoft.Xna.Framework.GamerServices; using Microsoft.Xna.Framework.Graphics; using Microsoft.Xna.Framework.Input; using Microsoft.Xna.Framework.Media; namespace Test2DGame2 { class Tiles { public Texture2D texture; public Rectangle rectangle; public Tiles(Texture2D texture) { this.texture = texture; } //check to see if james collides with the tile from the right side public void rightCollision(James james) { if (james.GetBounds().Intersects(rectangle)) { james.position.X = rectangle.Left - james.front.Width; } } } } I have a method for rightCollision because I could only figure out how to get the collisions from specifying directions. and here is the James class for my character using System; using System.Collections.Generic; using System.Linq; using Microsoft.Xna.Framework; using Microsoft.Xna.Framework.Audio; using Microsoft.Xna.Framework.Content; using Microsoft.Xna.Framework.GamerServices; using Microsoft.Xna.Framework.Graphics; using Microsoft.Xna.Framework.Input; using Microsoft.Xna.Framework.Media; namespace Test2DGame2 { class James { public Texture2D front; public Texture2D back; public Texture2D left; public Texture2D right; public Vector2 center; public Vector2 position; public James(Texture2D front) { position = new Vector2(0, 0); this.front = front; center = new Vector2(front.Width / 2, front.Height / 2); } public James(Texture2D front, Vector2 newPosition) { this.front = front; position = newPosition; center = new Vector2(front.Width / 2, front.Height / 2); } public void move(GameTime gameTime) { KeyboardState keyboard = Keyboard.GetState(); float SCALE = 20.0f; float speed = gameTime.ElapsedGameTime.Milliseconds / 100.0f; if (keyboard.IsKeyDown(Keys.Up)) { position.Y -=speed * SCALE; } else if (keyboard.IsKeyDown(Keys.Down)) { position.Y += speed * SCALE; } else if (keyboard.IsKeyDown(Keys.Left)) { position.X -= speed * SCALE; } else if (keyboard.IsKeyDown(Keys.Right)) { position.X += speed * SCALE; } } public void draw(SpriteBatch spriteBatch) { spriteBatch.Draw(front, position, null, Color.White, 0, center, 1.0f, SpriteEffects.None, 0.0f); } //get the boundingbox for James public Rectangle GetBounds() { return new Rectangle( (int)position.X, (int)position.Y, front.Width, front.Height); } } }

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  • Optimizing AES modes on Solaris for Intel Westmere

    - by danx
    Optimizing AES modes on Solaris for Intel Westmere Review AES is a strong method of symmetric (secret-key) encryption. It is a U.S. FIPS-approved cryptographic algorithm (FIPS 197) that operates on 16-byte blocks. AES has been available since 2001 and is widely used. However, AES by itself has a weakness. AES encryption isn't usually used by itself because identical blocks of plaintext are always encrypted into identical blocks of ciphertext. This encryption can be easily attacked with "dictionaries" of common blocks of text and allows one to more-easily discern the content of the unknown cryptotext. This mode of encryption is called "Electronic Code Book" (ECB), because one in theory can keep a "code book" of all known cryptotext and plaintext results to cipher and decipher AES. In practice, a complete "code book" is not practical, even in electronic form, but large dictionaries of common plaintext blocks is still possible. Here's a diagram of encrypting input data using AES ECB mode: Block 1 Block 2 PlainTextInput PlainTextInput | | | | \/ \/ AESKey-->(AES Encryption) AESKey-->(AES Encryption) | | | | \/ \/ CipherTextOutput CipherTextOutput Block 1 Block 2 What's the solution to the same cleartext input producing the same ciphertext output? The solution is to further process the encrypted or decrypted text in such a way that the same text produces different output. This usually involves an Initialization Vector (IV) and XORing the decrypted or encrypted text. As an example, I'll illustrate CBC mode encryption: Block 1 Block 2 PlainTextInput PlainTextInput | | | | \/ \/ IV >----->(XOR) +------------->(XOR) +---> . . . . | | | | | | | | \/ | \/ | AESKey-->(AES Encryption) | AESKey-->(AES Encryption) | | | | | | | | | \/ | \/ | CipherTextOutput ------+ CipherTextOutput -------+ Block 1 Block 2 The steps for CBC encryption are: Start with a 16-byte Initialization Vector (IV), choosen randomly. XOR the IV with the first block of input plaintext Encrypt the result with AES using a user-provided key. The result is the first 16-bytes of output cryptotext. Use the cryptotext (instead of the IV) of the previous block to XOR with the next input block of plaintext Another mode besides CBC is Counter Mode (CTR). As with CBC mode, it also starts with a 16-byte IV. However, for subsequent blocks, the IV is just incremented by one. Also, the IV ix XORed with the AES encryption result (not the plain text input). Here's an illustration: Block 1 Block 2 PlainTextInput PlainTextInput | | | | \/ \/ AESKey-->(AES Encryption) AESKey-->(AES Encryption) | | | | \/ \/ IV >----->(XOR) IV + 1 >---->(XOR) IV + 2 ---> . . . . | | | | \/ \/ CipherTextOutput CipherTextOutput Block 1 Block 2 Optimization Which of these modes can be parallelized? ECB encryption/decryption can be parallelized because it does more than plain AES encryption and decryption, as mentioned above. CBC encryption can't be parallelized because it depends on the output of the previous block. However, CBC decryption can be parallelized because all the encrypted blocks are known at the beginning. CTR encryption and decryption can be parallelized because the input to each block is known--it's just the IV incremented by one for each subsequent block. So, in summary, for ECB, CBC, and CTR modes, encryption and decryption can be parallelized with the exception of CBC encryption. How do we parallelize encryption? By interleaving. Usually when reading and writing data there are pipeline "stalls" (idle processor cycles) that result from waiting for memory to be loaded or stored to or from CPU registers. Since the software is written to encrypt/decrypt the next data block where pipeline stalls usually occurs, we can avoid stalls and crypt with fewer cycles. This software processes 4 blocks at a time, which ensures virtually no waiting ("stalling") for reading or writing data in memory. Other Optimizations Besides interleaving, other optimizations performed are Loading the entire key schedule into the 128-bit %xmm registers. This is done once for per 4-block of data (since 4 blocks of data is processed, when present). The following is loaded: the entire "key schedule" (user input key preprocessed for encryption and decryption). This takes 11, 13, or 15 registers, for AES-128, AES-192, and AES-256, respectively The input data is loaded into another %xmm register The same register contains the output result after encrypting/decrypting Using SSSE 4 instructions (AESNI). Besides the aesenc, aesenclast, aesdec, aesdeclast, aeskeygenassist, and aesimc AESNI instructions, Intel has several other instructions that operate on the 128-bit %xmm registers. Some common instructions for encryption are: pxor exclusive or (very useful), movdqu load/store a %xmm register from/to memory, pshufb shuffle bytes for byte swapping, pclmulqdq carry-less multiply for GCM mode Combining AES encryption/decryption with CBC or CTR modes processing. Instead of loading input data twice (once for AES encryption/decryption, and again for modes (CTR or CBC, for example) processing, the input data is loaded once as both AES and modes operations occur at in the same function Performance Everyone likes pretty color charts, so here they are. I ran these on Solaris 11 running on a Piketon Platform system with a 4-core Intel Clarkdale processor @3.20GHz. Clarkdale which is part of the Westmere processor architecture family. The "before" case is Solaris 11, unmodified. Keep in mind that the "before" case already has been optimized with hand-coded Intel AESNI assembly. The "after" case has combined AES-NI and mode instructions, interleaved 4 blocks at-a-time. « For the first table, lower is better (milliseconds). The first table shows the performance improvement using the Solaris encrypt(1) and decrypt(1) CLI commands. I encrypted and decrypted a 1/2 GByte file on /tmp (swap tmpfs). Encryption improved by about 40% and decryption improved by about 80%. AES-128 is slighty faster than AES-256, as expected. The second table shows more detail timings for CBC, CTR, and ECB modes for the 3 AES key sizes and different data lengths. » The results shown are the percentage improvement as shown by an internal PKCS#11 microbenchmark. And keep in mind the previous baseline code already had optimized AESNI assembly! The keysize (AES-128, 192, or 256) makes little difference in relative percentage improvement (although, of course, AES-128 is faster than AES-256). Larger data sizes show better improvement than 128-byte data. Availability This software is in Solaris 11 FCS. It is available in the 64-bit libcrypto library and the "aes" Solaris kernel module. You must be running hardware that supports AESNI (for example, Intel Westmere and Sandy Bridge, microprocessor architectures). The easiest way to determine if AES-NI is available is with the isainfo(1) command. For example, $ isainfo -v 64-bit amd64 applications pclmulqdq aes sse4.2 sse4.1 ssse3 popcnt tscp ahf cx16 sse3 sse2 sse fxsr mmx cmov amd_sysc cx8 tsc fpu 32-bit i386 applications pclmulqdq aes sse4.2 sse4.1 ssse3 popcnt tscp ahf cx16 sse3 sse2 sse fxsr mmx cmov sep cx8 tsc fpu No special configuration or setup is needed to take advantage of this software. Solaris libraries and kernel automatically determine if it's running on AESNI-capable machines and execute the correctly-tuned software for the current microprocessor. Summary Maximum throughput of AES cipher modes can be achieved by combining AES encryption with modes processing, interleaving encryption of 4 blocks at a time, and using Intel's wide 128-bit %xmm registers and instructions. References "Block cipher modes of operation", Wikipedia Good overview of AES modes (ECB, CBC, CTR, etc.) "Advanced Encryption Standard", Wikipedia "Current Modes" describes NIST-approved block cipher modes (ECB,CBC, CFB, OFB, CCM, GCM)

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  • SQL SERVER – Core Concepts – Elasticity, Scalability and ACID Properties – Exploring NuoDB an Elastically Scalable Database System

    - by pinaldave
    I have been recently exploring Elasticity and Scalability attributes of databases. You can see that in my earlier blog posts about NuoDB where I wanted to look at Elasticity and Scalability concepts. The concepts are very interesting, and intriguing as well. I have discussed these concepts with my friend Joyti M and together we have come up with this interesting read. The goal of this article is to answer following simple questions What is Elasticity? What is Scalability? How ACID properties vary from NOSQL Concepts? What are the prevailing problems in the current database system architectures? Why is NuoDB  an innovative and welcome change in database paradigm? Elasticity This word’s original form is used in many different ways and honestly it does do a decent job in holding things together over the years as a person grows and contracts. Within the tech world, and specifically related to software systems (database, application servers), it has come to mean a few things - allow stretching of resources without reaching the breaking point (on demand). What are resources in this context? Resources are the usual suspects – RAM/CPU/IO/Bandwidth in the form of a container (a process or bunch of processes combined as modules). When it is about increasing resources the simplest idea which comes to mind is the addition of another container. Another container means adding a brand new physical node. When it is about adding a new node there are two questions which comes to mind. 1) Can we add another node to our software system? 2) If yes, does adding new node cause downtime for the system? Let us assume we have added new node, let us see what the new needs of the system are when a new node is added. Balancing incoming requests to multiple nodes Synchronization of a shared state across multiple nodes Identification of “downstate” and resolution action to bring it to “upstate” Well, adding a new node has its advantages as well. Here are few of the positive points Throughput can increase nearly horizontally across the node throughout the system Response times of application will increase as in-between layer interactions will be improved Now, Let us put the above concepts in the perspective of a Database. When we mention the term “running out of resources” or “application is bound to resources” the resources can be CPU, Memory or Bandwidth. The regular approach to “gain scalability” in the database is to look around for bottlenecks and increase the bottlenecked resource. When we have memory as a bottleneck we look at the data buffers, locks, query plans or indexes. After a point even this is not enough as there needs to be an efficient way of managing such large workload on a “single machine” across memory and CPU bound (right kind of scheduling)  workload. We next move on to either read/write separation of the workload or functionality-based sharing so that we still have control of the individual. But this requires lots of planning and change in client systems in terms of knowing where to go/update/read and for reporting applications to “aggregate the data” in an intelligent way. What we ideally need is an intelligent layer which allows us to do these things without us getting into managing, monitoring and distributing the workload. Scalability In the context of database/applications, scalability means three main things Ability to handle normal loads without pressure E.g. X users at the Y utilization of resources (CPU, Memory, Bandwidth) on the Z kind of hardware (4 processor, 32 GB machine with 15000 RPM SATA drives and 1 GHz Network switch) with T throughput Ability to scale up to expected peak load which is greater than normal load with acceptable response times Ability to provide acceptable response times across the system E.g. Response time in S milliseconds (or agreed upon unit of measure) – 90% of the time The Issue – Need of Scale In normal cases one can plan for the load testing to test out normal, peak, and stress scenarios to ensure specific hardware meets the needs. With help from Hardware and Software partners and best practices, bottlenecks can be identified and requisite resources added to the system. Unfortunately this vertical scale is expensive and difficult to achieve and most of the operational people need the ability to scale horizontally. This helps in getting better throughput as there are physical limits in terms of adding resources (Memory, CPU, Bandwidth and Storage) indefinitely. Today we have different options to achieve scalability: Read & Write Separation The idea here is to do actual writes to one store and configure slaves receiving the latest data with acceptable delays. Slaves can be used for balancing out reads. We can also explore functional separation or sharing as well. We can separate data operations by a specific identifier (e.g. region, year, month) and consolidate it for reporting purposes. For functional separation the major disadvantage is when schema changes or workload pattern changes. As the requirement grows one still needs to deal with scale need in manual ways by providing an abstraction in the middle tier code. Using NOSQL solutions The idea is to flatten out the structures in general to keep all values which are retrieved together at the same store and provide flexible schema. The issue with the stores is that they are compromising on mostly consistency (no ACID guarantees) and one has to use NON-SQL dialect to work with the store. The other major issue is about education with NOSQL solutions. Would one really want to make these compromises on the ability to connect and retrieve in simple SQL manner and learn other skill sets? Or for that matter give up on ACID guarantee and start dealing with consistency issues? Hybrid Deployment – Mac, Linux, Cloud, and Windows One of the challenges today that we see across On-premise vs Cloud infrastructure is a difference in abilities. Take for example SQL Azure – it is wonderful in its concepts of throttling (as it is shared deployment) of resources and ability to scale using federation. However, the same abilities are not available on premise. This is not a mistake, mind you – but a compromise of the sweet spot of workloads, customer requirements and operational SLAs which can be supported by the team. In today’s world it is imperative that databases are available across operating systems – which are a commodity and used by developers of all hues. An Ideal Database Ability List A system which allows a linear scale of the system (increase in throughput with reasonable response time) with the addition of resources A system which does not compromise on the ACID guarantees and require developers to learn new paradigms A system which does not force fit a new way interacting with database by learning Non-SQL dialect A system which does not force fit its mechanisms for providing availability across its various modules. Well NuoDB is the first database which has all of the above abilities and much more. In future articles I will cover my hands-on experience with it. Reference: Pinal Dave (http://blog.SQLAuthority.com) Filed under: PostADay, SQL, SQL Authority, SQL Query, SQL Server, SQL Tips and Tricks, T SQL, Technology Tagged: NuoDB

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  • Is RTD Stateless or Stateful?

    - by [email protected]
    Yes.   A stateless service is one where each request is an independent transaction that can be processed by any of the servers in a cluster.  A stateful service is one where state is kept in a server's memory from transaction to transaction, thus necessitating the proper routing of requests to the right server. The main advantage of stateless systems is simplicity of design. The main advantage of stateful systems is performance. I'm often asked whether RTD is a stateless or stateful service, so I wanted to clarify this issue in depth so that RTD's architecture will be properly understood. The short answer is: "RTD can be configured as a stateless or stateful service." The performance difference between stateless and stateful systems can be very significant, and while in a call center implementation it may be reasonable to use a pure stateless configuration, a web implementation that produces thousands of requests per second is practically impossible with a stateless configuration. RTD's performance is orders of magnitude better than most competing systems. RTD was architected from the ground up to achieve this performance. Features like automatic and dynamic compression of prediction models, automatic translation of metadata to machine code, lack of interpreted languages, and separation of model building from decisioning contribute to achieving this performance level. Because  of this focus on performance we decided to have RTD's default configuration work in a stateful manner. By being stateful RTD requests are typically handled in a few milliseconds when repeated requests come to the same session. Now, those readers that have participated in implementations of RTD know that RTD's architecture is also focused on reducing Total Cost of Ownership (TCO) with features like automatic model building, automatic time windows, automatic maintenance of database tables, automatic evaluation of data mining models, automatic management of models partitioned by channel, geography, etcetera, and hot swapping of configurations. How do you reconcile the need for a low TCO and the need for performance? How do you get the performance of a stateful system with the simplicity of a stateless system? The answer is that you make the system behave like a stateless system to the exterior, but you let it automatically take advantage of situations where being stateful is better. For example, one of the advantages of stateless systems is that you can route a message to any server in a cluster, without worrying about sending it to the same server that was handling the session in previous messages. With an RTD stateful configuration you can still route the message to any server in the cluster, so from the point of view of the configuration of other systems, it is the same as a stateless service. The difference though comes in performance, because if the message arrives to the right server, RTD can serve it without any external access to the session's state, thus tremendously reducing processing time. In typical implementations it is not rare to have high percentages of messages routed directly to the right server, while those that are not, are easily handled by forwarding the messages to the right server. This architecture usually provides the best of both worlds with performance and simplicity of configuration.   Configuring RTD as a pure stateless service A pure stateless configuration requires session data to be persisted at the end of handling each and every message and reloading that data at the beginning of handling any new message. This is of course, the root of the inefficiency of these configurations. This is also the reason why many "stateless" implementations actually do keep state to take advantage of a request coming back to the same server. Nevertheless, if the implementation requires a pure stateless decision service, this is easy to configure in RTD. The way to do it is: Mark every Integration Point to Close the session at the end of processing the message In the Session entity persist the session data on closing the session In the session entity check if a persisted version exists and load it An excellent solution for persisting the session data is Oracle Coherence, which provides a high performance, distributed cache that minimizes the performance impact of persisting and reloading the session. Alternatively, the session can be persisted to a local database. An interesting feature of the RTD stateless configuration is that it can cope with serializing concurrent requests for the same session. For example, if a web page produces two requests to the decision service, these requests could come concurrently to the decision services and be handled by different servers. Most stateless implementation would have the two requests step onto each other when saving the state, or fail one of the messages. When properly configured, RTD will make one message wait for the other before processing.   A Word on Context Using the context of a customer interaction typically significantly increases lift. For example, offer success in a call center could double if the context of the call is taken into account. For this reason, it is important to utilize the contextual information in decision making. To make the contextual information available throughout a session it needs to be persisted. When there is a well defined owner for the information then there is no problem because in case of a session restart, the information can be easily retrieved. If there is no official owner of the information, then RTD can be configured to persist this information.   Once again, RTD provides flexibility to ensure high performance when it is adequate to allow for some loss of state in the rare cases of server failure. For example, in a heavy use web site that serves 1000 pages per second the navigation history may be stored in the in memory session. In such sites it is typical that there is no OLTP that stores all the navigation events, therefore if an RTD server were to fail, it would be possible for the navigation to that point to be lost (note that a new session would be immediately established in one of the other servers). In most cases the loss of this navigation information would be acceptable as it would happen rarely. If it is desired to save this information, RTD would persist it every time the visitor navigates to a new page. Note that this practice is preferred whether RTD is configured in a stateless or stateful manner.  

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  • More Fun With Math

    - by PointsToShare
    More Fun with Math   The runaway student – three different ways of solving one problem Here is a problem I read in a Russian site: A student is running away. He is moving at 1 mph. Pursuing him are a lion, a tiger and his math teacher. The lion is 40 miles behind and moving at 6 mph. The tiger is 28 miles behind and moving at 4 mph. His math teacher is 30 miles behind and moving at 5 mph. Who will catch him first? Analysis Obviously we have a set of three problems. They are all basically the same, but the details are different. The problems are of the same class. Here is a little excursion into computer science. One of the things we strive to do is to create solutions for classes of problems rather than individual problems. In your daily routine, you call it re-usability. Not all classes of problems have such solutions. If a class has a general (re-usable) solution, it is called computable. Otherwise it is unsolvable. Within unsolvable classes, we may still solve individual (some but not all) problems, albeit with different approaches to each. Luckily the vast majority of our daily problems are computable, and the 3 problems of our runaway student belong to a computable class. So, let’s solve for the catch-up time by the math teacher, after all she is the most frightening. She might even make the poor runaway solve this very problem – perish the thought! Method 1 – numerical analysis. At 30 miles and 5 mph, it’ll take her 6 hours to come to where the student was to begin with. But by then the student has advanced by 6 miles. 6 miles require 6/5 hours, but by then the student advanced by another 6/5 of a mile as well. And so on and so forth. So what are we to do? One way is to write code and iterate it until we have solved it. But this is an infinite process so we’ll end up with an infinite loop. So what to do? We’ll use the principles of numerical analysis. Any calculator – your computer included – has a limited number of digits. A double floating point number is good for about 14 digits. Nothing can be computed at a greater accuracy than that. This means that we will not iterate ad infinidum, but rather to the point where 2 consecutive iterations yield the same result. When we do financial computations, we don’t even have to go that far. We stop at the 10th of a penny.  It behooves us here to stop at a 10th of a second (100 milliseconds) and this will how we will avoid an infinite loop. Interestingly this alludes to the Zeno paradoxes of motion – in particular “Achilles and the Tortoise”. Zeno says exactly the same. To catch the tortoise, Achilles must always first come to where the tortoise was, but the tortoise keeps moving – hence Achilles will never catch the tortoise and our math teacher (or lion, or tiger) will never catch the student, or the policeman the thief. Here is my resolution to the paradox. The distance and time in each step are smaller and smaller, so the student will be caught. The only thing that is infinite is the iterative solution. The race is a convergent geometric process so the steps are diminishing, but each step in the solution takes the same amount of effort and time so with an infinite number of steps, we’ll spend an eternity solving it.  This BTW is an original thought that I have never seen before. But I digress. Let’s simply write the code to solve the problem. To make sure that it runs everywhere, I’ll do it in JavaScript. function LongCatchUpTime(D, PV, FV) // D is Distance; PV is Pursuers Velocity; FV is Fugitive’ Velocity {     var t = 0;     var T = 0;     var d = parseFloat(D);     var pv = parseFloat (PV);     var fv = parseFloat (FV);     t = d / pv;     while (t > 0.000001) //a 10th of a second is 1/36,000 of an hour, I used 1/100,000     {         T = T + t;         d = t * fv;         t = d / pv;     }     return T;     } By and large, the higher the Pursuer’s velocity relative to the fugitive, the faster the calculation. Solving this with the 10th of a second limit yields: 7.499999232000001 Method 2 – Geometric Series. Each step in the iteration above is smaller than the next. As you saw, we stopped iterating when the last step was small enough, small enough not to really matter.  When we have a sequence of numbers in which the ratio of each number to its predecessor is fixed we call the sequence geometric. When we are looking at the sum of sequence, we call the sequence of sums series.  Now let’s look at our student and teacher. The teacher runs 5 times faster than the student, so with each iteration the distance between them shrinks to a fifth of what it was before. This is a fixed ratio so we deal with a geometric series.  We normally designate this ratio as q and when q is less than 1 (0 < q < 1) the sum of  + … +  is  – 1) / (q – 1). When q is less than 1, it is easier to use ) / (1 - q). Now, the steps are 6 hours then 6/5 hours then 6/5*5 and so on, so q = 1/5. And the whole series is multiplied by 6. Also because q is less than 1 , 1/  diminishes to 0. So the sum is just  / (1 - q). or 1/ (1 – 1/5) = 1 / (4/5) = 5/4. This times 6 yields 7.5 hours. We can now continue with some algebra and take it back to a simpler formula. This is arduous and I am not going to do it here. Instead let’s do some simpler algebra. Method 3 – Simple Algebra. If the time to capture the fugitive is T and the fugitive travels at 1 mph, then by the time the pursuer catches him he travelled additional T miles. Time is distance divided by speed, so…. (D + T)/V = T  thus D + T = VT  and D = VT – T = (V – 1)T  and T = D/(V – 1) This “strangely” coincides with the solution we just got from the geometric sequence. This is simpler ad faster. Here is the corresponding code. function ShortCatchUpTime(D, PV, FV) {     var d = parseFloat(D);     var pv = parseFloat (PV);     var fv = parseFloat (FV);     return d / (pv - fv); } The code above, for both the iterative solution and the algebraic solution are actually for a larger class of problems.  In our original problem the student’s velocity (speed) is 1 mph. In the code it may be anything as long as it is less than the pursuer’s velocity. As long as PV > FV, the pursuer will catch up. Here is the really general formula: T = D / (PV – FV) Finally, let’s run the program for each of the pursuers.  It could not be worse. I know he’d rather be eaten alive than suffering through yet another math lesson. See the code run? Select  “Catch Up Time” in www.mgsltns.com/games.htm The host is running on Unix, so the link is case sensitive. That’s All Folks

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  • How to create a virtual network with Azure Connect

    - by Herve Roggero
    If you are trying to establish a virtual network between machines located in disparate networks, you can either use VPN, Virtual Network or Azure Connect. If you want to establish a connection between machines located in Windows Azure, you should consider using the Virtual Network service. If you want to establish a connection between local machines and Virtual Machines in Windows Azure, you may be able to use your existing VPN device (assuming you have one), as long as the device is supported by Microsoft. If the VPN device you are using isn’t supported, or if you are trying to create a virtual network between machines from disparate networks (such as machines located in another cloud provider), you can use Azure Connect. This blog post explains how Azure Connect can help you create virtual networks between multiple servers in the cloud, various servers in different cloud environments, and on-premise. Note: Azure Connect is currently in Technical Preview. About Azure Connect Let’s do a quick review of Azure Connect. This technology implements an IPSec tunnel from machines to to a relay service located in the Microsoft cloud (Azure). So in essence, Azure Connect doesn’t provide a point-to-point connection between machines; the network communication is tunneled through the relay service. The relay service in turn offers a mechanism to enforce basic communication rules that you define through Groups. We will review this later. You could network two or more VMs in the Azure cloud (although you should consider using a Virtual Network if you go this route), or servers in the Azure cloud and other machines in the Amazon cloud for example, or even two or more on-premise servers located in different locations for which a direct network connection is not an option. You can place any number of machines in your topology. Azure Connect gives you great flexibility on how you want to build your virtual network across various environments. So Azure Connect makes sense when you want to: Connect machines located in different cloud providers Connect on-premise machines running in different locations Connect Azure VMs with on-premise (if you do not have a VPN device, or if your device is not supported) Connect Azure Roles (Worker Roles, Web Roles) with on-premise servers or in other cloud providers The diagram below shows you a high level network topology that involves machines in the Windows Azure cloud, other cloud providers and on-premise. You should note that the only required component in this diagram is the Relay itself. The other machines are optional (although your network is useful only if you have two or more machines involved). Relay agents are currently available in three geographic areas: US, Europe and Asia. You can change which region you want to use in the Windows Azure management portal. High Level Network Topology With Azure Connect Azure Connect Agent Azure Connect establishes a virtual network and creates virtual adapters on your machines; these virtual adapters communicate through the Relay using IPSec. This is achieved by installing an agent (the Azure Connect Agent) on all the machines you want in your network topology. However, you do not need to install the agent on Worker Roles and Web Roles; that’s because the agent is already installed for you. Any other machine, including Virtual Machines in Windows Azure, needs the agent installed.  To install the agent, simply go to your Windows Azure portal (http://windows.azure.com) and click on Networks on the bottom left panel. You will see a list of subscriptions under Connect. If you select a subscription, you will be able to click on the Install Local Endpoint icon on top. Clicking on this icon will begin the download and installation process for the agent. Activating Roles for Azure Connect As previously mentioned, you do not need to install the Azure Connect Agent on Worker Roles and Web Roles because it is already loaded. However, you do need to activate them if you want the roles to participate in your network topology. To do this, you will need to click on the Get Activation Token icon. The activation token must then be copied and placed in the configuration file of your roles. For more information on how to perform this step, visit MSDN at http://msdn.microsoft.com/en-us/library/windowsazure/gg432964.aspx. Firewall Rules Note that specific firewall rules must exist to allow the agent to communicate through the Relay. You will need to allow TCP 443 and ICMPv6. For additional information, please visit MSDN at http://msdn.microsoft.com/en-us/library/windowsazure/gg433061.aspx. CA Certificates You can optionally require agents to sign their activation request with the Relay using a trusted certificate issued by a Certificate Authority (CA). Click on Activation Options to learn more. Groups To create your network topology you must first create a group. A group represents a logical container of endpoints (or machines) that can communicate through the Relay. You can create multiple groups allowing you to manage network communication differently. For example you could create a DEVELOPMENT group and a PRODUCTION group. To add an endpoint you must first install an agent that will create a virtual adapter on the machine on which it is installed (as discussed in the previous section). Once you have created a group and installed the agents, the machines will appear in the Windows Azure management portal and you can start assigning machines to groups. The next figure shows you that I created a group called LocalGroup and assigned two machines (both on-premise) to that group. Groups and Computers in Azure Connect As I mentioned previously you can allow these machines to establish a network connection. To do this, you must enable the Interconnected option in the group. The following diagram shows you the definition of the group. In this topology I chose to include local machines only, but I could also add worker roles and web roles in the Azure Roles section (you must first activate your roles, as discussed previously). You could also add other Groups, allowing you to manage inter-group communication. Defining a Group in Azure Connect Testing the Connection Now that my agents have been installed on my two machines, the group defined and the Interconnected option checked, I can test the connection between my machines. The next screenshot shows you that I sent a PING request to DEVLAP02 from DEVDSK02. The PING request was successful. Note however that the time is in the hundreds of milliseconds on average. That is to be expected because the machines are connecting through the Relay located in the cloud. Going through the Relay introduces an extra hop in the communication chain, so if your systems rely on high performance, you may want to conduct some basic performance tests. Sending a PING Request Through The Relay Conclusion As you can see, creating a network topology between machines using the Azure Connect service is simple. It took me less than five minutes to create the above configuration, including the time it took to install the Azure Connect agents on the two machines. The flexibility of Azure Connect allows you to create a virtual network between disparate environments, as long as your operating systems are supported by the agent. For more information on Azure Connect, visit the MSDN website at http://msdn.microsoft.com/en-us/library/windowsazure/gg432997.aspx. About Herve Roggero Herve Roggero, Windows Azure MVP, is the founder of Blue Syntax Consulting, a company specialized in cloud computing products and services. Herve's experience includes software development, architecture, database administration and senior management with both global corporations and startup companies. Herve holds multiple certifications, including an MCDBA, MCSE, MCSD. He also holds a Master's degree in Business Administration from Indiana University. Herve is the co-author of "PRO SQL Azure" from Apress and runs the Azure Florida Association (on LinkedIn: http://www.linkedin.com/groups?gid=4177626). For more information on Blue Syntax Consulting, visit www.bluesyntax.net. Special Thanks I would like thank those that helped me figure out how Azure Connect works: Marcel Meijer - http://blogs.msmvps.com/marcelmeijer/ Michael Wood - Http://www.mvwood.com Glenn Block - http://www.codebetter.com/glennblock Yves Goeleven - http://cloudshaper.wordpress.com/ Sandrino Di Mattia - http://fabriccontroller.net/ Mike Martin - http://techmike2kx.wordpress.com

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  • Looking into the JQuery Cycle Plugin

    - by nikolaosk
    I have been using JQuery for a couple of years now and it has helped me to solve many problems on the client. You can find all my posts about JQuery in this link. In this post I will be providing you with a hands-on example on the JQuery Cycle Plugin.I have been using extensively this plugin in my websites.You can rotate a series of images using various transitions with this plugin.It is a slideshow type of experience. I will be writing more posts regarding the most commonly used JQuery Plugins.  In this hands-on example I will be using Expression Web 4.0.This application is not a free application. You can use any HTML editor you like.You can use Visual Studio 2012 Express edition. You can download it here.  You can download this plugin from this link I launch Expression Web 4.0 and then I type the following HTML markup (I am using HTML 5) <!DOCTYPE html><html lang="en">  <head>    <title>Liverpool Legends</title>        <meta http-equiv="Content-Type" content="text/html;charset=utf-8" >            <script type="text/javascript" src="jquery-1.8.3.min.js"> </script>     <script type="text/javascript" src="jquery.cycle.all.js"></script>              <script type="text/javascript">        $(function() {            $('#main').cycle({ fx: 'fade'});        });    </script>       </head>  <body>    <header>        <h1>Liverpool Legends</h1>    </header>        <div id="main">                   <img src="championsofeurope.jpg" alt="Champions of Europe">                        <img src="steven_gerrard.jpg" alt="Steven Gerrard">                        <img src="ynwa.jpg" alt="You will never walk alone">                       </div>            <footer>        <p>All Rights Reserved</p>      </footer>     </body>  </html> This is a very simple markup. I have added three photos (make sure you use your own when trying this example)I have added references to the JQuery library (current version is 1.8.3) and the JQuery Cycle Plugin. Then I have added 3 images in the main div element.The Javascript code that makes it all happen follows.  <script type="text/javascript">        $(function() {            $('#main').cycle({ fx: 'fade'});        });    </script>  It couldn't be any simpler than that. I view my simple in Internet Explorer 10 and it works as expected. I have this series of images transitioning one after the other using the "fade" effect. I have tested this simple solution in all major browsers and it works fine.We can have a different transition effect by changing the JS code. Have a look at the code below       <script type="text/javascript">        $(function() {            $('#main').cycle({                     fx: 'cover',        speed: 500,        timeout: 2000                        });        });    </script>   We set the speed to 500 milliseconds, that is the speed we want to have for the ‘cover’ transition.The timeout is set to two seconds which is the time the photo will show until the next transition will take place.We can customise this plugin further but this is a short introduction to the plugin.Hope it helps!!!

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  • Strange Flash AS3 xml Socket behavior

    - by Rnd_d
    I have a problem which I can't understand. To understand it I wrote a socket client on AS3 and a server on python/twisted, you can see the code of both applications below. Let's launch two clients at the same time, arrange them so that you can see both windows and press connection button in both windows. Then press and hold any button. What I'm expecting: Client with pressed button sends a message "some data" to the server, then the server sends this message to all the clients(including the original sender) . Then each client moves right the button 'connectButton' and prints a message to the log with time in the following format: "min:secs:milliseconds". What is going wrong: The motion is smooth in the client that sends the message, but in all other clients the motion is jerky. This happens because messages to those clients arrive later than to the original sending client. And if we have three clients (let's name them A,B,C) and we send a message from A, the sending time log of B and C will be the same. Why other clients recieve this messages later than the original sender? By the way, on ubuntu 10.04/chrome all the motion is smooth. Two clients are launched in separated chromes. windows screenshot Can't post linux screenshot, need more than 10 reputation to post more hyperlinks. Listing of log, four clients simultaneously: [16:29:33.280858] 62.140.224.1 >> some data [16:29:33.280912] 87.249.9.98 << some data [16:29:33.280970] 87.249.9.98 << some data [16:29:33.281025] 87.249.9.98 << some data [16:29:33.281079] 62.140.224.1 << some data [16:29:33.323267] 62.140.224.1 >> some data [16:29:33.323326] 87.249.9.98 << some data [16:29:33.323386] 87.249.9.98 << some data [16:29:33.323440] 87.249.9.98 << some data [16:29:33.323493] 62.140.224.1 << some data [16:29:34.123435] 62.140.224.1 >> some data [16:29:34.123525] 87.249.9.98 << some data [16:29:34.123593] 87.249.9.98 << some data [16:29:34.123648] 87.249.9.98 << some data [16:29:34.123702] 62.140.224.1 << some data AS3 client code package { import adobe.utils.CustomActions; import flash.display.Sprite; import flash.events.DataEvent; import flash.events.Event; import flash.events.IOErrorEvent; import flash.events.KeyboardEvent; import flash.events.MouseEvent; import flash.events.SecurityErrorEvent; import flash.net.XMLSocket; import flash.system.Security; import flash.text.TextField; public class Main extends Sprite { private var socket :XMLSocket; private var textField :TextField = new TextField; private var connectButton :TextField = new TextField; public function Main():void { if (stage) init(); else addEventListener(Event.ADDED_TO_STAGE, init); } private function init(event:Event = null):void { socket = new XMLSocket(); socket.addEventListener(Event.CONNECT, connectHandler); socket.addEventListener(DataEvent.DATA, dataHandler); stage.addEventListener(KeyboardEvent.KEY_DOWN, keyDownHandler); addChild(textField); textField.y = 50; textField.width = 780; textField.height = 500; textField.border = true; connectButton.selectable = false; connectButton.border = true; connectButton.addEventListener(MouseEvent.MOUSE_DOWN, connectMouseDownHandler); connectButton.width = 105; connectButton.height = 20; connectButton.text = "click here to connect"; addChild(connectButton); } private function connectHandler(event:Event):void { textField.appendText("Connect\n"); textField.appendText("Press and hold any key\n"); } private function dataHandler(event:DataEvent):void { var now:Date = new Date(); textField.appendText(event.data + " time = " + now.getMinutes() + ":" + now.getSeconds() + ":" + now.getMilliseconds() + "\n"); connectButton.x += 2; } private function keyDownHandler(event:KeyboardEvent):void { socket.send("some data"); } private function connectMouseDownHandler(event:MouseEvent):void { var connectAddress:String = "ep1c.org"; var connectPort:Number = 13250; Security.loadPolicyFile("xmlsocket://" + connectAddress + ":" + String(connectPort)); socket.connect(connectAddress, connectPort); } } } Python server code from twisted.internet import reactor from twisted.internet.protocol import ServerFactory from twisted.protocols.basic import LineOnlyReceiver import datetime class EchoProtocol(LineOnlyReceiver): ##### name = "" id = 0 delimiter = chr(0) ##### def getName(self): return self.transport.getPeer().host def connectionMade(self): self.id = self.factory.getNextId() print "New connection from %s - id:%s" % (self.getName(), self.id) self.factory.clientProtocols[self.id] = self def connectionLost(self, reason): print "Lost connection from "+ self.getName() del self.factory.clientProtocols[self.id] self.factory.sendMessageToAllClients(self.getName() + " has disconnected.") def lineReceived(self, line): print "[%s] %s >> %s" % (datetime.datetime.now().time(), self, line) if line=="<policy-file-request/>": data = """<?xml version="1.0"?> <!DOCTYPE cross-domain-policy SYSTEM "http://www.adobe.com/xml/dtds/cross-domain-policy.dtd"> <!-- Policy file for xmlsocket://ep1c.org --> <cross-domain-policy> <allow-access-from domain="*" to-ports="%s" /> </cross-domain-policy>""" % PORT self.send(data) else: self.factory.sendMessageToAllClients( line ) def send(self, line): print "[%s] %s << %s" % (datetime.datetime.now().time(), self, line) if line: self.transport.write( str(line) + chr(0)) else: print "Nothing to send" def __str__(self): return self.getName() class ChatProtocolFactory(ServerFactory): protocol = EchoProtocol def __init__(self): self.clientProtocols = {} self.nextId = 0 def getNextId(self): id = self.nextId self.nextId += 1 return id def sendMessageToAllClients(self, msg): for client in self.clientProtocols: self.clientProtocols[client].send(msg) def sendMessageToClient(self, id, msg): self.clientProtocols[id].send(msg) PORT = 13250 print "Starting Server" factory = ChatProtocolFactory() reactor.listenTCP(PORT, factory) reactor.run()

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  • Query optimization using composite indexes

    - by xmarch
    Many times, during the process of creating a new Coherence application, developers do not pay attention to the way cache queries are constructed; they only check that these queries comply with functional specs. Later, performance testing shows that these perform poorly and it is then when developers start working on improvements until the non-functional performance requirements are met. This post describes the optimization process of a real-life scenario, where using a composite attribute index has brought a radical improvement in query execution times.  The execution times went down from 4 seconds to 2 milliseconds! E-commerce solution based on Oracle ATG – Endeca In the context of a new e-commerce solution based on Oracle ATG – Endeca, Oracle Coherence has been used to calculate and store SKU prices. In this architecture, a Coherence cache stores the final SKU prices used for Endeca baseline indexing. Each SKU price is calculated from a base SKU price and a series of calculations based on information from corporate global discounts. Corporate global discounts information is stored in an auxiliary Coherence cache with over 800.000 entries. In particular, to obtain each price the process needs to execute six queries over the global discount cache. After the implementation was finished, we discovered that the most expensive steps in the price calculation discount process were the global discounts cache query. This query has 10 parameters and is executed 6 times for each SKU price calculation. The steps taken to optimise this query are described below; Starting point Initial query was: String filter = "levelId = :iLevelId AND  salesCompanyId = :iSalesCompanyId AND salesChannelId = :iSalesChannelId "+ "AND departmentId = :iDepartmentId AND familyId = :iFamilyId AND brand = :iBrand AND manufacturer = :iManufacturer "+ "AND areaId = :iAreaId AND endDate >=  :iEndDate AND startDate <= :iStartDate"; Map<String, Object> params = new HashMap<String, Object>(10); // Fill all parameters. params.put("iLevelId", xxxx); // Executing filter. Filter globalDiscountsFilter = QueryHelper.createFilter(filter, params); NamedCache globalDiscountsCache = CacheFactory.getCache(CacheConstants.GLOBAL_DISCOUNTS_CACHE_NAME); Set applicableDiscounts = globalDiscountsCache.entrySet(globalDiscountsFilter); With the small dataset used for development the cache queries performed very well. However, when carrying out performance testing with a real-world sample size of 800,000 entries, each query execution was taking more than 4 seconds. First round of optimizations The first optimisation step was the creation of separate Coherence index for each of the 10 attributes used by the filter. This avoided object deserialization while executing the query. Each index was created as follows: globalDiscountsCache.addIndex(new ReflectionExtractor("getXXX" ) , false, null); After adding these indexes the query execution time was reduced to between 450 ms and 1s. However, these execution times were still not good enough.  Second round of optimizations In this optimisation phase a Coherence query explain plan was used to identify how many entires each index reduced the results set by, along with the cost in ms of executing that part of the query. Though the explain plan showed that all the indexes for the query were being used, it also showed that the ordering of the query parameters was "sub-optimal".  Parameters associated to object attributes with high-cardinality should appear at the beginning of the filter, or more specifically, the attributes that filters out the highest of number records should be placed at the beginning. But examining corporate global discount data we realized that depending on the values of the parameters used in the query the “good” order for the attributes was different. In particular, if the attributes brand and family had specific values it was more optimal to have a different query changing the order of the attributes. Ultimately, we ended up with three different optimal variants of the query that were used in its relevant cases: String filter = "brand = :iBrand AND familyId = :iFamilyId AND departmentId = :iDepartmentId AND levelId = :iLevelId "+ "AND manufacturer = :iManufacturer AND endDate >= :iEndDate AND salesCompanyId = :iSalesCompanyId "+ "AND areaId = :iAreaId AND salesChannelId = :iSalesChannelId AND startDate <= :iStartDate"; String filter = "familyId = :iFamilyId AND departmentId = :iDepartmentId AND levelId = :iLevelId AND brand = :iBrand "+ "AND manufacturer = :iManufacturer AND endDate >=  :iEndDate AND salesCompanyId = :iSalesCompanyId "+ "AND areaId = :iAreaId  AND salesChannelId = :iSalesChannelId AND startDate <= :iStartDate"; String filter = "brand = :iBrand AND departmentId = :iDepartmentId AND familyId = :iFamilyId AND levelId = :iLevelId "+ "AND manufacturer = :iManufacturer AND endDate >= :iEndDate AND salesCompanyId = :iSalesCompanyId "+ "AND areaId = :iAreaId AND salesChannelId = :iSalesChannelId AND startDate <= :iStartDate"; Using the appropriate query depending on the value of brand and family parameters the query execution time dropped to between 100 ms and 150 ms. But these these execution times were still not good enough and the solution was cumbersome. Third and last round of optimizations The third and final optimization was to introduce a composite index. However, this did mean that it was not possible to use the Coherence Query Language (CohQL), as composite indexes are not currently supporte in CohQL. As the original query had 8 parameters using EqualsFilter, 1 using GreaterEqualsFilter and 1 using LessEqualsFilter, the composite index was built for the 8 attributes using EqualsFilter. The final query had an EqualsFilter for the multiple extractor, a GreaterEqualsFilter and a LessEqualsFilter for the 2 remaining attributes.  All individual indexes were dropped except the ones being used for LessEqualsFilter and GreaterEqualsFilter. We were now running in an scenario with an 8-attributes composite filter and 2 single attribute filters. The composite index created was as follows: ValueExtractor[] ve = { new ReflectionExtractor("getSalesChannelId" ), new ReflectionExtractor("getLevelId" ),    new ReflectionExtractor("getAreaId" ), new ReflectionExtractor("getDepartmentId" ),    new ReflectionExtractor("getFamilyId" ), new ReflectionExtractor("getManufacturer" ),    new ReflectionExtractor("getBrand" ), new ReflectionExtractor("getSalesCompanyId" )}; MultiExtractor me = new MultiExtractor(ve); NamedCache globalDiscountsCache = CacheFactory.getCache(CacheConstants.GLOBAL_DISCOUNTS_CACHE_NAME); globalDiscountsCache.addIndex(me, false, null); And the final query was: ValueExtractor[] ve = { new ReflectionExtractor("getSalesChannelId" ), new ReflectionExtractor("getLevelId" ),    new ReflectionExtractor("getAreaId" ), new ReflectionExtractor("getDepartmentId" ),    new ReflectionExtractor("getFamilyId" ), new ReflectionExtractor("getManufacturer" ),    new ReflectionExtractor("getBrand" ), new ReflectionExtractor("getSalesCompanyId" )}; MultiExtractor me = new MultiExtractor(ve); // Fill composite parameters.String SalesCompanyId = xxxx;...AndFilter composite = new AndFilter(new EqualsFilter(me,                   Arrays.asList(iSalesChannelId, iLevelId, iAreaId, iDepartmentId, iFamilyId, iManufacturer, iBrand, SalesCompanyId)),                                     new GreaterEqualsFilter(new ReflectionExtractor("getEndDate" ), iEndDate)); AndFilter finalFilter = new AndFilter(composite, new LessEqualsFilter(new ReflectionExtractor("getStartDate" ), iStartDate)); NamedCache globalDiscountsCache = CacheFactory.getCache(CacheConstants.GLOBAL_DISCOUNTS_CACHE_NAME); Set applicableDiscounts = globalDiscountsCache.entrySet(finalFilter);      Using this composite index the query improved dramatically and the execution time dropped to between 2 ms and  4 ms.  These execution times completely met the non-functional performance requirements . It should be noticed than when using the composite index the order of the attributes inside the ValueExtractor was not relevant.

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  • Revisiting ANTS Performance Profiler 7.4

    - by James Michael Hare
    Last year, I did a small review on the ANTS Performance Profiler 6.3, now that it’s a year later and a major version number higher, I thought I’d revisit the review and revise my last post. This post will take the same examples as the original post and update them to show what’s new in version 7.4 of the profiler. Background A performance profiler’s main job is to keep track of how much time is typically spent in each unit of code. This helps when we have a program that is not running at the performance we expect, and we want to know where the program is experiencing issues. There are many profilers out there of varying capabilities. Red Gate’s typically seem to be the very easy to “jump in” and get started with very little training required. So let’s dig into the Performance Profiler. I’ve constructed a very crude program with some obvious inefficiencies. It’s a simple program that generates random order numbers (or really could be any unique identifier), adds it to a list, sorts the list, then finds the max and min number in the list. Ignore the fact it’s very contrived and obviously inefficient, we just want to use it as an example to show off the tool: 1: // our test program 2: public static class Program 3: { 4: // the number of iterations to perform 5: private static int _iterations = 1000000; 6: 7: // The main method that controls it all 8: public static void Main() 9: { 10: var list = new List<string>(); 11: 12: for (int i = 0; i < _iterations; i++) 13: { 14: var x = GetNextId(); 15: 16: AddToList(list, x); 17: 18: var highLow = GetHighLow(list); 19: 20: if ((i % 1000) == 0) 21: { 22: Console.WriteLine("{0} - High: {1}, Low: {2}", i, highLow.Item1, highLow.Item2); 23: Console.Out.Flush(); 24: } 25: } 26: } 27: 28: // gets the next order id to process (random for us) 29: public static string GetNextId() 30: { 31: var random = new Random(); 32: var num = random.Next(1000000, 9999999); 33: return num.ToString(); 34: } 35: 36: // add it to our list - very inefficiently! 37: public static void AddToList(List<string> list, string item) 38: { 39: list.Add(item); 40: list.Sort(); 41: } 42: 43: // get high and low of order id range - very inefficiently! 44: public static Tuple<int,int> GetHighLow(List<string> list) 45: { 46: return Tuple.Create(list.Max(s => Convert.ToInt32(s)), list.Min(s => Convert.ToInt32(s))); 47: } 48: } So let’s run it through the profiler and see what happens! Visual Studio Integration First, let’s look at how the ANTS profilers integrate with Visual Studio’s menu system. Once you install the ANTS profilers, you will get an ANTS menu item with several options: Notice that you can either Profile Performance or Launch ANTS Performance Profiler. These sound similar but achieve two slightly different actions: Profile Performance: this immediately launches the profiler with all defaults selected to profile the active project in Visual Studio. Launch ANTS Performance Profiler: this launches the profiler much the same way as starting it from the Start Menu. The profiler will pre-populate the application and path information, but allow you to change the settings before beginning the profile run. So really, the main difference is that Profile Performance immediately begins profiling with the default selections, where Launch ANTS Performance Profiler allows you to change the defaults and attach to an already-running application. Let’s Fire it Up! So when you fire up ANTS either via Start Menu or Launch ANTS Performance Profiler menu in Visual Studio, you are presented with a very simple dialog to get you started: Notice you can choose from many different options for application type. You can profile executables, services, web applications, or just attach to a running process. In fact, in version 7.4 we see two new options added: ASP.NET Web Application (IIS Express) SharePoint web application (IIS) So this gives us an additional way to profile ASP.NET applications and the ability to profile SharePoint applications as well. You can also choose your level of detail in the Profiling Mode drop down. If you choose Line-Level and method-level timings detail, you will get a lot more detail on the method durations, but this will also slow down profiling somewhat. If you really need the profiler to be as unintrusive as possible, you can change it to Sample method-level timings. This is performing very light profiling, where basically the profiler collects timings of a method by examining the call-stack at given intervals. Which method you choose depends a lot on how much detail you need to find the issue and how sensitive your program issues are to timing. So for our example, let’s just go with the line and method timing detail. So, we check that all the options are correct (if you launch from VS2010, the executable and path are filled in already), and fire it up by clicking the [Start Profiling] button. Profiling the Application Once you start profiling the application, you will see a real-time graph of CPU usage that will indicate how much your application is using the CPU(s) on your system. During this time, you can select segments of the graph and bookmark them, giving them mnemonic names. This can be useful if you want to compare performance in one part of the run to another part of the run. Notice that once you select a block, it will give you the call tree breakdown for that selection only, and the relative performance of those calls. Once you feel you have collected enough information, you can click [Stop Profiling] to stop the application run and information collection and begin a more thorough analysis. Analyzing Method Timings So now that we’ve halted the run, we can look around the GUI and see what we can see. By default, the times are shown in terms of percentage of time of the total run of the application, though you can change it in the View menu item to milliseconds, ticks, or seconds as well. This won’t affect the percentages of methods, it only affects what units the times are shown. Notice also that the major hotspot seems to be in a method without source, ANTS Profiler will filter these out by default, but you can right-click on the line and remove the filter to see more detail. This proves especially handy when a bottleneck is due to a method in the BCL. So now that we’ve removed the filter, we see a bit more detail: In addition, ANTS Performance Profiler gives you the ability to decompile the methods without source so that you can dive even deeper, though typically this isn’t necessary for our purposes. When looking at timings, there are generally two types of timings for each method call: Time: This is the time spent ONLY in this method, not including calls this method makes to other methods. Time With Children: This is the total of time spent in both this method AND including calls this method makes to other methods. In other words, the Time tells you how much work is being done exclusively in this method, and the Time With Children tells you how much work is being done inclusively in this method and everything it calls. You can also choose to display the methods in a tree or in a grid. The tree view is the default and it shows the method calls arranged in terms of the tree representing all method calls and the parent method that called them, etc. This is useful for when you find a hot-spot method, you can see who is calling it to determine if the problem is the method itself, or if it is being called too many times. The grid method represents each method only once with its totals and is useful for quickly seeing what method is the trouble spot. In addition, you can choose to display Methods with source which are generally the methods you wrote (as opposed to native or BCL code), or Any Method which shows not only your methods, but also native calls, JIT overhead, synchronization waits, etc. So these are just two ways of viewing the same data, and you’re free to choose the organization that best suits what information you are after. Analyzing Method Source If we look at the timings above, we see that our AddToList() method (and in particular, it’s call to the List<T>.Sort() method in the BCL) is the hot-spot in this analysis. If ANTS sees a method that is consuming the most time, it will flag it as a hot-spot to help call out potential areas of concern. This doesn’t mean the other statistics aren’t meaningful, but that the hot-spot is most likely going to be your biggest bang-for-the-buck to concentrate on. So let’s select the AddToList() method, and see what it shows in the source window below: Notice the source breakout in the bottom pane when you select a method (from either tree or grid view). This shows you the timings in this method per line of code. This gives you a major indicator of where the trouble-spot in this method is. So in this case, we see that performing a Sort() on the List<T> after every Add() is killing our performance! Of course, this was a very contrived, duh moment, but you’d be surprised how many performance issues become duh moments. Note that this one line is taking up 86% of the execution time of this application! If we eliminate this bottleneck, we should see drastic improvement in the performance. So to fix this, if we still wanted to maintain the List<T> we’d have many options, including: delay Sort() until after all Add() methods, using a SortedSet, SortedList, or SortedDictionary depending on which is most appropriate, or forgoing the sorting all together and using a Dictionary. Rinse, Repeat! So let’s just change all instances of List<string> to SortedSet<string> and run this again through the profiler: Now we see the AddToList() method is no longer our hot-spot, but now the Max() and Min() calls are! This is good because we’ve eliminated one hot-spot and now we can try to correct this one as well. As before, we can then optimize this part of the code (possibly by taking advantage of the fact the list is now sorted and returning the first and last elements). We can then rinse and repeat this process until we have eliminated as many bottlenecks as possible. Calls by Web Request Another feature that was added recently is the ability to view .NET methods grouped by the HTTP requests that caused them to run. This can be helpful in determining which pages, web services, etc. are causing hot spots in your web applications. Summary If you like the other ANTS tools, you’ll like the ANTS Performance Profiler as well. It is extremely easy to use with very little product knowledge required to get up and running. There are profilers built into the higher product lines of Visual Studio, of course, which are also powerful and easy to use. But for quickly jumping in and finding hot spots rapidly, Red Gate’s Performance Profiler 7.4 is an excellent choice. Technorati Tags: Influencers,ANTS,Performance Profiler,Profiler

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  • Queued Loadtest to remove Concurrency issues using Shared Data Service in OpenScript

    - by stefan.thieme(at)oracle.com
    Queued Processing to remove Concurrency issues in Loadtest ScriptsSome scripts act on information returned by the server, e.g. act on first item in the returned list of pending tasks/actions. This may lead to concurrency issues if the virtual users simulated in a load test scenario are not synchronized in some way.As the load test cases should be carried out in a comparable and straight forward manner simply cancel a transaction in case a collision occurs is clearly not an option. In case you increase the number of virtual users this approach would lead to a high number of requests for the early steps in your transaction (e.g. login, retrieve list of action points, assign an action point to the virtual user) but later steps would be rarely visited successfully or at all, depending on the application logic.A way to tackle this problem is to enqueue the virtual users in a Shared Data Service queue. Only the first virtual user in this queue will be allowed to carry out the critical steps (retrieve list of action points, assign an action point to the virtual user) in your transaction at any one time.Once a virtual user has passed the critical path it will dequeue himself from the head of the queue and continue with his actions. This does theoretically allow virtual users to run in parallel all steps of the transaction which are not part of the critical path.In practice it has been seen this is rarely the case, though it does not allow adding more than N users to perform a transaction without causing delays due to virtual users waiting in the queue. N being the time of the total transaction divided by the sum of the time of all critical steps in this transaction.While this problem can be circumvented by allowing multiple queues to act on individual segments of the list of actions, e.g. per country filter, ends with 0..9 filter, etc.This would require additional handling of these additional queues of slots for the virtual users at the head of the queue in order to maintain the mutually exclusive access to the first element in the list returned by the server at any one time of the load test. Such an improved handling of multiple queues and/or multiple slots is above the subject of this paper.Shared Data Services Pre-RequisitesStart WebLogic Server to host Shared Data ServicesYou will have to make sure that your WebLogic server is installed and started. Shared Data Services may not work if you installed only the minimal installation package for OpenScript. If however you installed the default package including OLT and OTM, you may follow the instructions below to start and verify WebLogic installation.To start the WebLogic Server deployed underneath of Oracle Load Testing and/or Oracle Test Manager you can go to your Start menu, Oracle Application Testing Suite and select the Restart Oracle Application Testing Suite Application Service entry from the Tools submenu.To verify the service has been started you can run the Microsoft Management Console for Services by Selecting Run from the Start Menu and entering services.msc. Look for the entry that reads Oracle Application Testing Suite Application Service, once it has changed it status from Starting to Started you can proceed to verify the login. Please note that this may take several minutes, I would say up to 10 minutes depending on the strength of your CPU horse-power.Verify WebLogic Server user credentialsYou will have to make sure that your WebLogic Server is installed and started. Next open the Oracle WebLogic Server Adminstration Console on http://localhost:8088/console.It may take a while until the application is deployed and started. It may display the following until the Administration Console has been deployed on the fly.Afterwards you can login using the username oats and the password that you selected during install time for your Application Testing Suite administrative purposes.This will bring up the Home page of you WebLogic Server. You have actually verified that you are able to login with these credentials already. However if you want to check the details, navigate to Security Realms, myrealm, Users and Groups tab.Here you could add users to your WebLogic Server which could be used in the later steps. Details on the Groups required for such a custom user to work are exceeding this quick overview and have to be selected with the WebLogic Server Adminstration Guide in mind.Shared Data Services pre-requisites for Load testingOpenScript Preferences have to be set to enable Encryption and provide a default Shared Data Service Connection for Playback.These are pre-requisites you want to use for load testing with Shared Data Services.Please note that the usage of the Connection Parameters (individual directive in the script) for Shared Data Services did not playback reliably in the current version 9.20.0370 of Oracle Load Testing (OLT) and encryption of credentials still seemed to be mandatory as well.General Encryption settingsSelect OpenScript Preferences from the View menu and navigate to the General, Encryption entry in the tree on the left. Select the Encrypt script data option from the list and enter the same password that you used for securing your WebLogic Server Administration Console.Enable global shared data access credentialsSelect OpenScript Preferences from the View menu and navigate to the Playback, Shared Data entry in the tree on the left. Enable the global shared data access credentials and enter the Address, User name and Password determined for your WebLogic Server to host Shared Data Services.Please note, that you may want to replace the localhost in Address with the hosts realname in case you plan to run load tests with Loadtest Agents running on remote systems.Queued Processing of TransactionsEnable Shared Data Services Module in Script PropertiesThe Shared Data Services Module has to be enabled for each Script that wants to employ the Shared Data Service Queue functionality in OpenScript. It can be enabled under the Script menu selecting Script Properties. On the Script Properties Dialog select the Modules section and check Shared Data to enable Shared Data Service Module for your script. Checking the Shared Data Services option will effectively add a line to your script code that adds the sharedData ScriptService to your script class of IteratingVUserScript.@ScriptService oracle.oats.scripting.modules.sharedData.api.SharedDataService sharedData;Record your scriptRecord your script as usual and then add the following things for Queue handling in the Initialize code block, before the first step and after the last step of your critical path and in the Finalize code block.The java code to be added at individual locations is explained in the following sections in full detail.Create a Shared Data Queue in InitializeTo create a Shared Data Queue go to the Java view of your script and enter the following statements to the initialize() code block.info("Create queueA with life time of 120 minutes");sharedData.createQueue("queueA", 120);This will create an instantiation of the Shared Data Queue object named queueA which is maintained for upto 120 minutes.If you want to use the code for multiple scripts, make sure to use a different queue name for each one here and in the subsequent steps. You may even consider to use a dynamic queueName based on filters of your result list being concurrently accessed.Prepare a unique id for each IterationIn order to keep track of individual virtual users in our queue we need to create a unique identifier from the virtual user id and the used username right after retrieving the next record from our databank file.getDatabank("Usernames").getNextDatabankRecord();getVariables().set("usernameValue1","VU_{{@vuid}}_{{@iterationnum}}_{{db.Usernames.Username}}_{{@timestamp}}_{{@random(10000)}}");String usernameValue = getVariables().get("usernameValue1");info("Now running virtual user " + usernameValue);As you can see from the above code block, we have set the OpenScript variable usernameValue1 to VU_{{@vuid}}_{{@iterationnum}}_{{db.Usernames.Username}}_{{@timestamp}}_{{@random(10000)}} which is a concatenation of the virtual user id and the iterationnumber for general uniqueness; as well as the username from our databank, the timestamp and a random number for making it further unique and ease spotting of errors.Not all of these fields are actually required to make it really unique, but adding the queue name may also be considered to help troubleshoot multiple queues.The value is then retrieved with the getVariables.get() method call and assigned to the usernameValue String used throughout the script.Please note that moving the getDatabank("Usernames").getNextDatabankRecord(); call to the initialize block was later considered to remove concurrency of multiple virtual users running with the same userid and therefor accessing the same "My Inbox" in step 6. This will effectively give each virtual user a userid from the databank file. Make sure you have enough userids to remove this second hurdle.Enqueue and attend Queue before Critical PathTo maintain the right order of virtual users being allowed into the critical path of the transaction the following pseudo step has to be added in front of the first critical step. In the case of this example this is right in front of the step where we retrieve the list of actions from which we select the first to be assigned to us.beginStep("[0] Waiting in the Queue", 0);{info("Enqueued virtual user " + usernameValue + " at the end of queueA");sharedData.offerLast("queueA", usernameValue);info("Wait until the user is the first in queueA");String queueValue1 = null;do {// we wait for at least 0.7 seconds before we check the head of the// queue. This is the time it takes one user to move through the// critical path, i.e. pass steps [5] Enter country and [6] Assign// to meThread.sleep(700);queueValue1 = (String) sharedData.peekFirst("queueA");info("The first user in queueA is currently: '" + queueValue1 + "' " + queueValue1.getClass() + " length " + queueValue1.length() );info("The current user is '"+ usernameValue + "' " + usernameValue.getClass() + " length " + usernameValue.length() + ": indexOf " + usernameValue.indexOf(queueValue1) + " equals " + usernameValue.equals(queueValue1) );} while ( queueValue1.indexOf(usernameValue) < 0 );info("Now the user is the first in queueA");}endStep();This will enqueue the username to the tail of our Queue. It will will wait for at least 700 milliseconds, the time it takes for one user to exit the critical path and then compare the head of our queue with it's username. This last step will be repeated while the two are not equal (indexOf less than zero). If they are equal the indexOf will yield a value of zero or larger and we will perform the critical steps.Dequeue after Critical PathAfter the virtual user has left the critical path and complete its last step the following code block needs to dequeue the virtual user. In the case of our example this is right after the action has been actually assigned to the virtual user. This will allow the next virtual user to retrieve the list of actions still available and in turn let him make his selection/assignment.info("Get and remove the current user from the head of queueA");String pollValue1 = (String) sharedData.pollFirst("queueA");The current user is removed from the head of the queue. The next one will now be able to match his username against the head of the queue.Clear and Destroy Queue for FinishWhen the script has completed, it should clear and destroy the queue. This code block can be put in the finish block of your script and/or in a separate script in order to clear and remove the queue in case you have spotted an error or want to reset the queue for some reason.info("Clear queueA");sharedData.clearQueue("queueA");info("Destroy queueA");sharedData.destroyQueue("queueA");The users waiting in queueA are cleared and the queue is destroyed. If you have scripts still executing they will be caught in a loop.I found it better to maintain a separate Reset Queue script which contained only the following code in the initialize() block. I use to call this script to make sure the queue is cleared in between multiple Loadtest runs. This script could also even be added as the first in a larger scenario, which would execute it only once at very start of the Loadtest and make sure the queues do not contain any stale entries.info("Create queueA with life time of 120 minutes");sharedData.createQueue("queueA", 120);info("Clear queueA");sharedData.clearQueue("queueA");This will create a Shared Data Queue instance of queueA and clear all entries from this queue.Monitoring QueueWhile creating the scripts it was useful to monitor the contents, i.e. the current first user in the Queue. The following code block will make sure the Shared Data Queue is accessible in the initialize() block.info("Create queueA with life time of 120 minutes");sharedData.createQueue("queueA", 120);In the run() block the following code will continuously monitor the first element of the Queue and write an informational message with the current username Value to the Result window.info("Monitor the first users in queueA");String queueValue1 = null;do {queueValue1 = (String) sharedData.peekFirst("queueA");if (queueValue1 != null)info("The first user in queueA is currently: '" + queueValue1 + "' " + queueValue1.getClass() + " length " + queueValue1.length() );} while ( true );This script can be run from OpenScript parallel to a loadtest performed by the Oracle Load Test.However it is not recommend to run this in a production loadtest as the performance impact is unknown. Accessing the Queue's head with the peekFirst() method has been reported with about 2 seconds response time by both OpenScript and OTL. It is advised to log a Service Request to see if this could be lowered in future releases of Application Testing Suite, as the pollFirst() and even offerLast() writing to the tail of the Queue usually returned after an average 0.1 seconds.Debugging QueueWhile debugging the scripts the following was useful to remove single entries from its head, i.e. the current first user in the Queue. The following code block will make sure the Shared Data Queue is accessible in the initialize() block.info("Create queueA with life time of 120 minutes");sharedData.createQueue("queueA", 120);In the run() block the following code will remove the first element of the Queue and write an informational message with the current username Value to the Result window.info("Get and remove the current user from the head of queueA");String pollValue1 = (String) sharedData.pollFirst("queueA");info("The first user in queueA was currently: '" + pollValue1 + "' " + pollValue1.getClass() + " length " + pollValue1.length() );ReferencesOracle Functional Testing OpenScript User's Guide Version 9.20 [E15488-05]Chapter 17 Using the Shared Data Modulehttp://download.oracle.com/otn/nt/apptesting/oats-docs-9.21.0030.zipOracle Fusion Middleware Oracle WebLogic Server Administration Console Online Help 11g Release 1 (10.3.4) [E13952-04]Administration Console Online Help - Manage users and groupshttp://download.oracle.com/docs/cd/E17904_01/apirefs.1111/e13952/taskhelp/security/ManageUsersAndGroups.htm

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