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  • Controlling fan speed on ASUS K43SV

    - by user181677
    ASUS K43SV laptop it very hot. Is it possible to control fan speed with fancontrol? When I run $sudo pwmconfig it displays this message: /usr/sbin/pwmconfig: There are no fan-capable sensor modules installed When I run $sensors, here is the output acpitz-virtual-0 Adapter: Virtual device temp1: +61.0°C (crit = +103.0°C) coretemp-isa-0000 Adapter: ISA adapter Physical id 0: +62.0°C (high = +86.0°C, crit = +100.0°C) Core 0: +62.0°C (high = +86.0°C, crit = +100.0°C) Core 1: +61.0°C (high = +86.0°C, crit = +100.0°C)

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  • Using Durandal to Create Single Page Apps

    - by Stephen.Walther
    A few days ago, I gave a talk on building Single Page Apps on the Microsoft Stack. In that talk, I recommended that people use Knockout, Sammy, and RequireJS to build their presentation layer and use the ASP.NET Web API to expose data from their server. After I gave the talk, several people contacted me and suggested that I investigate a new open-source JavaScript library named Durandal. Durandal stitches together Knockout, Sammy, and RequireJS to make it easier to use these technologies together. In this blog entry, I want to provide a brief walkthrough of using Durandal to create a simple Single Page App. I am going to demonstrate how you can create a simple Movies App which contains (virtual) pages for viewing a list of movies, adding new movies, and viewing movie details. The goal of this blog entry is to give you a sense of what it is like to build apps with Durandal. Installing Durandal First things first. How do you get Durandal? The GitHub project for Durandal is located here: https://github.com/BlueSpire/Durandal The Wiki — located at the GitHub project — contains all of the current documentation for Durandal. Currently, the documentation is a little sparse, but it is enough to get you started. Instead of downloading the Durandal source from GitHub, a better option for getting started with Durandal is to install one of the Durandal NuGet packages. I built the Movies App described in this blog entry by first creating a new ASP.NET MVC 4 Web Application with the Basic Template. Next, I executed the following command from the Package Manager Console: Install-Package Durandal.StarterKit As you can see from the screenshot of the Package Manager Console above, the Durandal Starter Kit package has several dependencies including: · jQuery · Knockout · Sammy · Twitter Bootstrap The Durandal Starter Kit package includes a sample Durandal application. You can get to the Starter Kit app by navigating to the Durandal controller. Unfortunately, when I first tried to run the Starter Kit app, I got an error because the Starter Kit is hard-coded to use a particular version of jQuery which is already out of date. You can fix this issue by modifying the App_Start\DurandalBundleConfig.cs file so it is jQuery version agnostic like this: bundles.Add( new ScriptBundle("~/scripts/vendor") .Include("~/Scripts/jquery-{version}.js") .Include("~/Scripts/knockout-{version}.js") .Include("~/Scripts/sammy-{version}.js") // .Include("~/Scripts/jquery-1.9.0.min.js") // .Include("~/Scripts/knockout-2.2.1.js") // .Include("~/Scripts/sammy-0.7.4.min.js") .Include("~/Scripts/bootstrap.min.js") ); The recommendation is that you create a Durandal app in a folder off your project root named App. The App folder in the Starter Kit contains the following subfolders and files: · durandal – This folder contains the actual durandal JavaScript library. · viewmodels – This folder contains all of your application’s view models. · views – This folder contains all of your application’s views. · main.js — This file contains all of the JavaScript startup code for your app including the client-side routing configuration. · main-built.js – This file contains an optimized version of your application. You need to build this file by using the RequireJS optimizer (unfortunately, before you can run the optimizer, you must first install NodeJS). For the purpose of this blog entry, I wanted to start from scratch when building the Movies app, so I deleted all of these files and folders except for the durandal folder which contains the durandal library. Creating the ASP.NET MVC Controller and View A Durandal app is built using a single server-side ASP.NET MVC controller and ASP.NET MVC view. A Durandal app is a Single Page App. When you navigate between pages, you are not navigating to new pages on the server. Instead, you are loading new virtual pages into the one-and-only-one server-side view. For the Movies app, I created the following ASP.NET MVC Home controller: public class HomeController : Controller { public ActionResult Index() { return View(); } } There is nothing special about the Home controller – it is as basic as it gets. Next, I created the following server-side ASP.NET view. This is the one-and-only server-side view used by the Movies app: @{ Layout = null; } <!DOCTYPE html> <html> <head> <title>Index</title> </head> <body> <div id="applicationHost"> Loading app.... </div> @Scripts.Render("~/scripts/vendor") <script type="text/javascript" src="~/App/durandal/amd/require.js" data-main="/App/main"></script> </body> </html> Notice that I set the Layout property for the view to the value null. If you neglect to do this, then the default ASP.NET MVC layout will be applied to the view and you will get the <!DOCTYPE> and opening and closing <html> tags twice. Next, notice that the view contains a DIV element with the Id applicationHost. This marks the area where virtual pages are loaded. When you navigate from page to page in a Durandal app, HTML page fragments are retrieved from the server and stuck in the applicationHost DIV element. Inside the applicationHost element, you can place any content which you want to display when a Durandal app is starting up. For example, you can create a fancy splash screen. I opted for simply displaying the text “Loading app…”: Next, notice the view above includes a call to the Scripts.Render() helper. This helper renders out all of the JavaScript files required by the Durandal library such as jQuery and Knockout. Remember to fix the App_Start\DurandalBundleConfig.cs as described above or Durandal will attempt to load an old version of jQuery and throw a JavaScript exception and stop working. Your application JavaScript code is not included in the scripts rendered by the Scripts.Render helper. Your application code is loaded dynamically by RequireJS with the help of the following SCRIPT element located at the bottom of the view: <script type="text/javascript" src="~/App/durandal/amd/require.js" data-main="/App/main"></script> The data-main attribute on the SCRIPT element causes RequireJS to load your /app/main.js JavaScript file to kick-off your Durandal app. Creating the Durandal Main.js File The Durandal Main.js JavaScript file, located in your App folder, contains all of the code required to configure the behavior of Durandal. Here’s what the Main.js file looks like in the case of the Movies app: require.config({ paths: { 'text': 'durandal/amd/text' } }); define(function (require) { var app = require('durandal/app'), viewLocator = require('durandal/viewLocator'), system = require('durandal/system'), router = require('durandal/plugins/router'); //>>excludeStart("build", true); system.debug(true); //>>excludeEnd("build"); app.start().then(function () { //Replace 'viewmodels' in the moduleId with 'views' to locate the view. //Look for partial views in a 'views' folder in the root. viewLocator.useConvention(); //configure routing router.useConvention(); router.mapNav("movies/show"); router.mapNav("movies/add"); router.mapNav("movies/details/:id"); app.adaptToDevice(); //Show the app by setting the root view model for our application with a transition. app.setRoot('viewmodels/shell', 'entrance'); }); }); There are three important things to notice about the main.js file above. First, notice that it contains a section which enables debugging which looks like this: //>>excludeStart(“build”, true); system.debug(true); //>>excludeEnd(“build”); This code enables debugging for your Durandal app which is very useful when things go wrong. When you call system.debug(true), Durandal writes out debugging information to your browser JavaScript console. For example, you can use the debugging information to diagnose issues with your client-side routes: (The funny looking //> symbols around the system.debug() call are RequireJS optimizer pragmas). The main.js file is also the place where you configure your client-side routes. In the case of the Movies app, the main.js file is used to configure routes for three page: the movies show, add, and details pages. //configure routing router.useConvention(); router.mapNav("movies/show"); router.mapNav("movies/add"); router.mapNav("movies/details/:id");   The route for movie details includes a route parameter named id. Later, we will use the id parameter to lookup and display the details for the right movie. Finally, the main.js file above contains the following line of code: //Show the app by setting the root view model for our application with a transition. app.setRoot('viewmodels/shell', 'entrance'); This line of code causes Durandal to load up a JavaScript file named shell.js and an HTML fragment named shell.html. I’ll discuss the shell in the next section. Creating the Durandal Shell You can think of the Durandal shell as the layout or master page for a Durandal app. The shell is where you put all of the content which you want to remain constant as a user navigates from virtual page to virtual page. For example, the shell is a great place to put your website logo and navigation links. The Durandal shell is composed from two parts: a JavaScript file and an HTML file. Here’s what the HTML file looks like for the Movies app: <h1>Movies App</h1> <div class="container-fluid page-host"> <!--ko compose: { model: router.activeItem, //wiring the router afterCompose: router.afterCompose, //wiring the router transition:'entrance', //use the 'entrance' transition when switching views cacheViews:true //telling composition to keep views in the dom, and reuse them (only a good idea with singleton view models) }--><!--/ko--> </div> And here is what the JavaScript file looks like: define(function (require) { var router = require('durandal/plugins/router'); return { router: router, activate: function () { return router.activate('movies/show'); } }; }); The JavaScript file contains the view model for the shell. This view model returns the Durandal router so you can access the list of configured routes from your shell. Notice that the JavaScript file includes a function named activate(). This function loads the movies/show page as the first page in the Movies app. If you want to create a different default Durandal page, then pass the name of a different age to the router.activate() method. Creating the Movies Show Page Durandal pages are created out of a view model and a view. The view model contains all of the data and view logic required for the view. The view contains all of the HTML markup for rendering the view model. Let’s start with the movies show page. The movies show page displays a list of movies. The view model for the show page looks like this: define(function (require) { var moviesRepository = require("repositories/moviesRepository"); return { movies: ko.observable(), activate: function() { this.movies(moviesRepository.listMovies()); } }; }); You create a view model by defining a new RequireJS module (see http://requirejs.org). You create a RequireJS module by placing all of your JavaScript code into an anonymous function passed to the RequireJS define() method. A RequireJS module has two parts. You retrieve all of the modules which your module requires at the top of your module. The code above depends on another RequireJS module named repositories/moviesRepository. Next, you return the implementation of your module. The code above returns a JavaScript object which contains a property named movies and a method named activate. The activate() method is a magic method which Durandal calls whenever it activates your view model. Your view model is activated whenever you navigate to a page which uses it. In the code above, the activate() method is used to get the list of movies from the movies repository and assign the list to the view model movies property. The HTML for the movies show page looks like this: <table> <thead> <tr> <th>Title</th><th>Director</th> </tr> </thead> <tbody data-bind="foreach:movies"> <tr> <td data-bind="text:title"></td> <td data-bind="text:director"></td> <td><a data-bind="attr:{href:'#/movies/details/'+id}">Details</a></td> </tr> </tbody> </table> <a href="#/movies/add">Add Movie</a> Notice that this is an HTML fragment. This fragment will be stuffed into the page-host DIV element in the shell.html file which is stuffed, in turn, into the applicationHost DIV element in the server-side MVC view. The HTML markup above contains data-bind attributes used by Knockout to display the list of movies (To learn more about Knockout, visit http://knockoutjs.com). The list of movies from the view model is displayed in an HTML table. Notice that the page includes a link to a page for adding a new movie. The link uses the following URL which starts with a hash: #/movies/add. Because the link starts with a hash, clicking the link does not cause a request back to the server. Instead, you navigate to the movies/add page virtually. Creating the Movies Add Page The movies add page also consists of a view model and view. The add page enables you to add a new movie to the movie database. Here’s the view model for the add page: define(function (require) { var app = require('durandal/app'); var router = require('durandal/plugins/router'); var moviesRepository = require("repositories/moviesRepository"); return { movieToAdd: { title: ko.observable(), director: ko.observable() }, activate: function () { this.movieToAdd.title(""); this.movieToAdd.director(""); this._movieAdded = false; }, canDeactivate: function () { if (this._movieAdded == false) { return app.showMessage('Are you sure you want to leave this page?', 'Navigate', ['Yes', 'No']); } else { return true; } }, addMovie: function () { // Add movie to db moviesRepository.addMovie(ko.toJS(this.movieToAdd)); // flag new movie this._movieAdded = true; // return to list of movies router.navigateTo("#/movies/show"); } }; }); The view model contains one property named movieToAdd which is bound to the add movie form. The view model also has the following three methods: 1. activate() – This method is called by Durandal when you navigate to the add movie page. The activate() method resets the add movie form by clearing out the movie title and director properties. 2. canDeactivate() – This method is called by Durandal when you attempt to navigate away from the add movie page. If you return false then navigation is cancelled. 3. addMovie() – This method executes when the add movie form is submitted. This code adds the new movie to the movie repository. I really like the Durandal canDeactivate() method. In the code above, I use the canDeactivate() method to show a warning to a user if they navigate away from the add movie page – either by clicking the Cancel button or by hitting the browser back button – before submitting the add movie form: The view for the add movie page looks like this: <form data-bind="submit:addMovie"> <fieldset> <legend>Add Movie</legend> <div> <label> Title: <input data-bind="value:movieToAdd.title" required /> </label> </div> <div> <label> Director: <input data-bind="value:movieToAdd.director" required /> </label> </div> <div> <input type="submit" value="Add" /> <a href="#/movies/show">Cancel</a> </div> </fieldset> </form> I am using Knockout to bind the movieToAdd property from the view model to the INPUT elements of the HTML form. Notice that the FORM element includes a data-bind attribute which invokes the addMovie() method from the view model when the HTML form is submitted. Creating the Movies Details Page You navigate to the movies details Page by clicking the Details link which appears next to each movie in the movies show page: The Details links pass the movie ids to the details page: #/movies/details/0 #/movies/details/1 #/movies/details/2 Here’s what the view model for the movies details page looks like: define(function (require) { var router = require('durandal/plugins/router'); var moviesRepository = require("repositories/moviesRepository"); return { movieToShow: { title: ko.observable(), director: ko.observable() }, activate: function (context) { // Grab movie from repository var movie = moviesRepository.getMovie(context.id); // Add to view model this.movieToShow.title(movie.title); this.movieToShow.director(movie.director); } }; }); Notice that the view model activate() method accepts a parameter named context. You can take advantage of the context parameter to retrieve route parameters such as the movie Id. In the code above, the context.id property is used to retrieve the correct movie from the movie repository and the movie is assigned to a property named movieToShow exposed by the view model. The movie details view displays the movieToShow property by taking advantage of Knockout bindings: <div> <h2 data-bind="text:movieToShow.title"></h2> directed by <span data-bind="text:movieToShow.director"></span> </div> Summary The goal of this blog entry was to walkthrough building a simple Single Page App using Durandal and to get a feel for what it is like to use this library. I really like how Durandal stitches together Knockout, Sammy, and RequireJS and establishes patterns for using these libraries to build Single Page Apps. Having a standard pattern which developers on a team can use to build new pages is super valuable. Once you get the hang of it, using Durandal to create new virtual pages is dead simple. Just define a new route, view model, and view and you are done. I also appreciate the fact that Durandal did not attempt to re-invent the wheel and that Durandal leverages existing JavaScript libraries such as Knockout, RequireJS, and Sammy. These existing libraries are powerful libraries and I have already invested a considerable amount of time in learning how to use them. Durandal makes it easier to use these libraries together without losing any of their power. Durandal has some additional interesting features which I have not had a chance to play with yet. For example, you can use the RequireJS optimizer to combine and minify all of a Durandal app’s code. Also, Durandal supports a way to create custom widgets (client-side controls) by composing widgets from a controller and view. You can download the code for the Movies app by clicking the following link (this is a Visual Studio 2012 project): Durandal Movie App

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  • BIOS upgrade lowers CPU temperature

    - by N.N.
    Setup I've got a system with an Asus P8Z68-V PRO motherboard and an Intel Core i7-2600K CPU running at stock speed (no overlocking) which I cool with a Noctua NH-U12P. On the heatsink I've got the two included fans connected via the included Low-Noise Adapters (L.N.A.) 1100 RPM, 16.9 dB(A). In the BIOS settings I've set the CPU and chassis fan profile to silent. Issue Yesterday I upgraded from BIOS version 0501 to 0606. After the upgrade I checked the temperatures in the BIOS monitor and was surprised to see that the CPU temperature was slightly ~30°C. Before the upgrade the CPU temperature was ~50°C with the same BIOS settings (see the following heading for details on temperatures). How can this be? It seems a bit odd that a BIOS upgrade can lower the CPU temperature by 20°C and it also seems odd that the CPU temperature is lower than the chassis temperature. Temperatures When I've checked temperatures the room temperature has been ~23°C. I haven't changed the placement of the computer nor the hardware or cooling setup between BIOS versions. BIOS version 0501 BIOS monitor: CPU: ~50°C Chassis: ~33°C I haven't got any temperature measures from lm-sensors or the like for version 0501 because I only discovered the issue after upgrading to version 0606 and the BIOS updater utility won't let me downgrade to version 0501 (it says "outdated image" when I try to load version 0501). BIOS version 0606 BIOS monitor: CPU: ~30°C Chassis: ~33°C lm-sensors in Ubuntu 11.04 Desktop 64-bit (sudo sensors after an uptime of 4 h 52 min and a load average of 0.22, 0.18, 0.15): coretemp-isa-0000 Adapter: ISA adapter Core 0: +32.0°C (high = +80.0°C, crit = +98.0°C) coretemp-isa-0001 Adapter: ISA adapter Core 1: +35.0°C (high = +80.0°C, crit = +98.0°C) coretemp-isa-0002 Adapter: ISA adapter Core 2: +29.0°C (high = +80.0°C, crit = +98.0°C) coretemp-isa-0003 Adapter: ISA adapter Core 3: +36.0°C (high = +80.0°C, crit = +98.0°C) The BIOS monitor temperatures was checked directly after the lm-sensors temperatures was checked. BIOS version 0706, 0801, 1101 and 3203 I get the same kind of temperatures both in the BIOS monitor and with lm-sensors in BIOS version 0706, 0801, 1101 and 3203 as in 0606. Information from Asus The 0606 changelog mentions nothing explicitly about CPU temperature (but item 3., as indicated by sidran32, might affect temperatures): P8Z68-V PRO 0606 BIOS with IRST 10.6.0.1002 Enable the support of Intel Rapid Storage Technology version 10.6.0.1002 Release Improve DRAM compatibility Improve System stability Improve compatibility with some Raid card model Increase IGD share memory size to 512MB However the following FAQ might give a hint: FAQs I find that the CPU temperature reading in BIOS is about 10~20 degrees centigrade hotter than the reading in OS. Is it normal? Page Tools Solution That is normal as BIOS does not send idle command to the CPU, making most of the power saving features useless. You should be getting similar reading if you disable EIST/C1E/CPU C3 Report/CPU C6 Report in BIOS.

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  • Validation in Silverlight

    - by Timmy Kokke
    Getting started with the basics Validation in Silverlight can get very complex pretty easy. The DataGrid control is the only control that does data validation automatically, but often you want to validate your own entry form. Values a user may enter in this form can be restricted by the customer and have to fit an exact fit to a list of requirements or you just want to prevent problems when saving the data to the database. Showing a message to the user when a value is entered is pretty straight forward as I’ll show you in the following example.     This (default) Silverlight textbox is data-bound to a simple data class. It has to be bound in “Two-way” mode to be sure the source value is updated when the target value changes. The INotifyPropertyChanged interface must be implemented by the data class to get the notification system to work. When the property changes a simple check is performed and when it doesn’t match some criteria an ValidationException is thrown. The ValidatesOnExceptions binding attribute is set to True to tell the textbox it should handle the thrown ValidationException. Let’s have a look at some code now. The xaml should contain something like below. The most important part is inside the binding. In this case the Text property is bound to the “Name” property in TwoWay mode. It is also told to validate on exceptions. This property is false by default.   <StackPanel Orientation="Horizontal"> <TextBox Width="150" x:Name="Name" Text="{Binding Path=Name, Mode=TwoWay, ValidatesOnExceptions=True}"/> <TextBlock Text="Name"/> </StackPanel>   The data class in this first example is a very simplified person class with only one property: string Name. The INotifyPropertyChanged interface is implemented and the PropertyChanged event is fired when the Name property changes. When the property changes a check is performed to see if the new string is null or empty. If this is the case a ValidationException is thrown explaining that the entered value is invalid.   public class PersonData:INotifyPropertyChanged { private string _name; public string Name { get { return _name; } set { if (_name != value) { if(string.IsNullOrEmpty(value)) throw new ValidationException("Name is required"); _name = value; if (PropertyChanged != null) PropertyChanged(this, new PropertyChangedEventArgs("Name")); } } } public event PropertyChangedEventHandler PropertyChanged=delegate { }; } The last thing that has to be done is letting binding an instance of the PersonData class to the DataContext of the control. This is done in the code behind file. public partial class Demo1 : UserControl { public Demo1() { InitializeComponent(); this.DataContext = new PersonData() {Name = "Johnny Walker"}; } }   Error Summary In many cases you would have more than one entry control. A summary of errors would be nice in such case. With a few changes to the xaml an error summary, like below, can be added.           First, add a namespace to the xaml so the control can be used. Add the following line to the header of the .xaml file. xmlns:Controls="clr-namespace:System.Windows.Controls;assembly=System.Windows.Controls.Data.Input"   Next, add the control to the layout. To get the result as in the image showed earlier, add the control right above the StackPanel from the first example. It’s got a small margin to separate it from the textbox a little.   <Controls:ValidationSummary Margin="8"/>   The ValidationSummary control has to be notified that an ValidationException occurred. This can be done with a small change to the xaml too. Add the NotifyOnValidationError to the binding expression. By default this value is set to false, so nothing would be notified. Set the property to true to get it to work.   <TextBox Width="150" x:Name="Name" Text="{Binding Name, Mode=TwoWay, ValidatesOnExceptions=True, NotifyOnValidationError=True}"/>   Data annotation Validating data in the setter is one option, but not my personal favorite. It’s the easiest way if you have a single required value you want to check, but often you want to validate more. Besides, I don’t consider it best practice to write logic in setters. The way used by frameworks like WCF Ria Services is the use of attributes on the properties. Instead of throwing exceptions you have to call the static method ValidateProperty on the Validator class. This call stays always the same for a particular property, not even when you change the attributes on the property. To mark a property “Required” you can use the RequiredAttribute. This is what the Name property is going to look like:   [Required] public string Name { get { return _name; } set { if (_name != value) { Validator.ValidateProperty(value, new ValidationContext(this, null, null){ MemberName = "Name" }); _name = value; if (PropertyChanged != null) PropertyChanged(this, new PropertyChangedEventArgs("Name")); } } }   The ValidateProperty method takes the new value for the property and an instance of ValidationContext. The properties passed to the constructor of the ValidationContextclass are very straight forward. This part is the same every time. The only thing that changes is the MemberName property of the ValidationContext. Property has to hold the name of the property you want to validate. It’s the same value you provide the PropertyChangedEventArgs with. The System.ComponentModel.DataAnnotation contains eight different validation attributes including a base class to create your own. They are: RequiredAttribute Specifies that a value must be provided. RangeAttribute The provide value must fall in the specified range. RegularExpressionAttribute Validates is the value matches the regular expression. StringLengthAttribute Checks if the number of characters in a string falls between a minimum and maximum amount. CustomValidationAttribute Use a custom method to validate the value. DataTypeAttribute Specify a data type using an enum or a custom data type. EnumDataTypeAttribute Makes sure the value is found in a enum. ValidationAttribute A base class for custom validation attributes All of these will ensure that an validation exception is thrown, except the DataTypeAttribute. This attribute is used to provide some additional information about the property. You can use this information in your own code.   [Required] [Range(0,125,ErrorMessage = "Value is not a valid age")] public int Age {   It’s no problem to stack different validation attributes together. For example, when an Age is required and must fall in the range from 0 to 125:   [Required, StringLength(255,MinimumLength = 3)] public string Name {   Or in one row like this, for a required Name with at least 3 characters and a maximum of 255:   Delayed validation Having properties marked as required can be very useful. The only downside to the technique described earlier is that you have to change the value in order to get it validated. What if you start out with empty an empty entry form? All fields are empty and thus won’t be validated. With this small trick you can validate at the moment the user click the submit button.   <TextBox Width="150" x:Name="NameField" Text="{Binding Name, Mode=TwoWay, ValidatesOnExceptions=True, NotifyOnValidationError=True, UpdateSourceTrigger=Explicit}"/>   By default, when a TwoWay bound control looses focus the value is updated. When you added validation like I’ve shown you earlier, the value is validated. To overcome this, you have to tell the binding update explicitly by setting the UpdateSourceTrigger binding property to Explicit:   private void SubmitButtonClick(object sender, RoutedEventArgs e) { NameField.GetBindingExpression(TextBox.TextProperty).UpdateSource(); }   This way, the binding is in two direction but the source is only updated, thus validated, when you tell it to. In the code behind you have to call the UpdateSource method on the binding expression, which you can get from the TextBox.   Conclusion Data validation is something you’ll probably want on almost every entry form. I always thought it was hard to do, but it wasn’t. If you can throw an exception you can do validation. If you want to know anything more in depth about something I talked about in this article let me know. I might write an entire post to that.

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  • Cloud MBaaS : The Next Big Thing in Enterprise Mobility

    - by shiju
    In this blog post, I will take a look at Cloud Mobile Backend as a Service (MBaaS) and how we can leverage Cloud based Mobile Backend as a Service for building enterprise mobile apps. Today, mobile apps are incredibly significant in both consumer and enterprise space and the demand for the mobile apps is unbelievably increasing in day to day business. An enterprise can’t survive in business without a proper mobility strategy. A better mobility strategy and faster delivery of your mobile apps will give you an extra mileage for your business and IT strategy. So organizations and mobile developers are looking for different strategy for meeting this demand and adopting different development strategy for their mobile apps. Some developers are adopting hybrid mobile app development platforms, for delivering their products for multiple platforms, for fast time-to-market. Others are adopting a Mobile enterprise application platform (MEAP) such as Kony for their enterprise mobile apps for fast time-to-market and better business integration. The Challenges of Enterprise Mobility The real challenge of enterprise mobile apps, is not about creating the front-end environment or developing front-end for multiple platforms. The most important thing of enterprise mobile apps is to expose your enterprise data to mobile devices where the real pain is your business data might be residing in lot of different systems including legacy systems, ERP systems etc., and these systems will be deployed with lot of security restrictions. Exposing your data from the on-premises servers, is not a easy thing for most of the business organizations. Many organizations are spending too much time for their front-end development strategy, but they are really lacking for building a strategy on their back-end for exposing the business data to mobile apps. So building a REST services layer and mobile back-end services, on the top of legacy systems and existing middleware systems, is the key part of most of the enterprise mobile apps, where multiple mobile platforms can easily consume these REST services and other mobile back-end services for building mobile apps. For some mobile apps, we can’t predict its user base, especially for products where customers can gradually increase at any time. And for today’s mobile apps, faster time-to-market is very critical so that spending too much time for mobile app’s scalability, will not be worth. The real power of Cloud is the agility and on-demand scalability, where we can scale-up and scale-down our applications very easily. It would be great if we could use the power of Cloud to mobile apps. So using Cloud for mobile apps is a natural fit, where we can use Cloud as the storage for mobile apps and hosting mechanism for mobile back-end services, where we can enjoy the full power of Cloud with greater level of on-demand scalability and operational agility. So Cloud based Mobile Backend as a Service is great choice for building enterprise mobile apps, where enterprises can enjoy the massive scalability power of their mobile apps, provided by public cloud vendors such as Microsoft Windows Azure. Mobile Backend as a Service (MBaaS) We have discussed the key challenges of enterprise mobile apps and how we can leverage Cloud for hosting mobile backend services. MBaaS is a set of cloud-based, server-side mobile services for multiple mobile platforms and HTML5 platform, which can be used as a backend for your mobile apps with the scalability power of Cloud. The information below provides the key features of a typical MBaaS platform: Cloud based storage for your application data. Automatic REST API services on the application data, for CRUD operations. Native push notification services with massive scalability power. User management services for authenticate users. User authentication via Social accounts such as Facebook, Google, Microsoft, and Twitter. Scheduler services for periodically sending data to mobile devices. Native SDKs for multiple mobile platforms such as Windows Phone and Windows Store, Android, Apple iOS, and HTML5, for easily accessing the mobile services from mobile apps, with better security.  Typically, a MBaaS platform will provide native SDKs for multiple mobile platforms so that we can easily consume the server-side mobile services. MBaaS based REST APIs can use for integrating to enterprise backend systems. We can use the same mobile services for multiple platform so hat we can reuse the application logic to multiple mobile platforms. Public cloud vendors are building the mobile services on the top of their PaaS offerings. Windows Azure Mobile Services is a great platform for a MBaaS offering that is leveraging Windows Azure Cloud platform’s PaaS capabilities. Hybrid mobile development platform Titanium provides their own MBaaS services. LoopBack is a new MBaaS service provided by Node.js consulting firm StrongLoop, which can be hosted on multiple cloud platforms and also for on-premises servers. The Challenges of MBaaS Solutions If you are building your mobile apps with a new data storage, it will be very easy, since there is not any integration challenges you have to face. But most of the use cases, you have to extract your application data in which stored in on-premises servers which might be under VPNs and firewalls. So exposing these data to your MBaaS solution with a proper security would be a big challenge. The capability of your MBaaS vendor is very important as you have to interact with your legacy systems for many enterprise mobile apps. So you should be very careful about choosing for MBaaS vendor. At the same time, you should have a proper strategy for mobilizing your application data which stored in on-premises legacy systems, where your solution architecture and strategy is more important than platforms and tools.  Windows Azure Mobile Services Windows Azure Mobile Services is an MBaaS offerings from Windows Azure cloud platform. IMHO, Microsoft Windows Azure is the best PaaS platform in the Cloud space. Windows Azure Mobile Services extends the PaaS capabilities of Windows Azure, to mobile devices, which can be used as a cloud backend for your mobile apps, which will provide global availability and reach for your mobile apps. Windows Azure Mobile Services provides storage services, user management with social network integration, push notification services and scheduler services and provides native SDKs for all major mobile platforms and HTML5. In Windows Azure Mobile Services, you can write server-side scripts in Node.js where you can enjoy the full power of Node.js including the use of NPM modules for your server-side scripts. In the previous section, we had discussed some challenges of MBaaS solutions. You can leverage Windows Azure Cloud platform for solving many challenges regarding with enterprise mobility. The entire Windows Azure platform can play a key role for working as the backend for your mobile apps where you can leverage the entire Windows Azure platform for your mobile apps. With Windows Azure, you can easily connect to your on-premises systems which is a key thing for mobile backend solutions. Another key point is that Windows Azure provides better integration with services like Active Directory, which makes Windows Azure as the de facto platform for enterprise mobility, for enterprises, who have been leveraging Microsoft ecosystem for their application and IT infrastructure. Windows Azure Mobile Services  is going to next evolution where you can expect some exciting features in near future. One area, where Windows Azure Mobile Services should definitely need an improvement, is about the default storage mechanism in which currently it is depends on SQL Server. IMHO, developers should be able to choose multiple default storage option when creating a new mobile service instance. Let’s say, there should be a different storage providers such as SQL Server storage provider and Table storage provider where developers should be able to choose their choice of storage provider when creating a new mobile services project. I have been used Windows Azure and Windows Azure Mobile Services as the backend for production apps for mobile, where it performed very well. MBaaS Over MEAP Recently, many larger enterprises has been adopted Mobile enterprise application platform (MEAP) for their mobile apps. I haven’t worked on any production MEAP solution, but I heard that developers are really struggling with MEAP in different way. The learning curve for a proprietary MEAP platform is very high. I am completely against for using larger proprietary ecosystem for mobile apps. For enterprise mobile apps, I highly recommend to use native iOS/Android/Windows Phone or HTML5  for front-end with a cloud hosted MBaaS solution as the middleware. A MBaaS service can be consumed from multiple mobile apps where REST APIs are using to integrating with enterprise backend systems. Enterprise mobility should start with exposing REST APIs on the enterprise backend systems and these REST APIs can host on Cloud where we can enjoy the power of Cloud for our services. If you are having REST APIs for your enterprise data, then you can easily build mobile frontends for multiple platforms.   You can follow me on Twitter @shijucv

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  • Unable to start Tomcat6 with HTTPS enabled

    - by ram
    I have the following server.xml settings for my tomcat6 server <!-- COMMENTED <Connector port="8080" maxThreads="150" enableLookups="false" acceptCount="100" scheme="http" redirectPort="8443"/> --> <!-- COMMENTED <Connector port="80" maxThreads="150" enableLookups="false" acceptCount="100" scheme="http" redirectPort="443"/> --> <Connector port="443" maxHttpHeaderSize="8192" maxThreads="150" enableLookups="false" disableUploadTimeout="true" acceptCount="100" scheme="https" secure="true" SSLEnabled="true" SSLCertificateFile="%SSL_CERT%" SSLCertificateKeyFile="%SSL_KEY%" SSLCipherSuite="ALL:!ADH:!kEDH:!SSLv2:!EXPORT40:!EXP:!LOW" compression="on" compressableMimeType="text/html,text/xml,text/plain,application/javascript,application/json,text/javascript"/> Complete server.xml is here but when I try to start the application I get the following error in catalina.*.log file INFO: Initializing Coyote HTTP/1.1 on http-80 Apr 7, 2013 8:38:38 PM org.apache.coyote.http11.Http11AprProtocol init SEVERE: Error initializing endpoint java.lang.Exception: Invalid Server SSL Protocol (error:00000000:lib(0):func(0):reason(0)) at org.apache.tomcat.jni.SSLContext.make(Native Method) at org.apache.tomcat.util.net.AprEndpoint.init(AprEndpoint.java:729) at org.apache.coyote.http11.Http11AprProtocol.init(Http11AprProtocol.java:107) at org.apache.catalina.connector.Connector.initialize(Connector.java:1049) at org.apache.catalina.core.StandardService.initialize(StandardService.java:703) at org.apache.catalina.core.StandardServer.initialize(StandardServer.java:838) at org.apache.catalina.startup.Catalina.load(Catalina.java:538) at org.apache.catalina.startup.Catalina.load(Catalina.java:562) at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) at java.lang.reflect.Method.invoke(Method.java:597) at org.apache.catalina.startup.Bootstrap.load(Bootstrap.java:261) at org.apache.catalina.startup.Bootstrap.main(Bootstrap.java:413) Apr 7, 2013 8:38:38 PM org.apache.catalina.core.StandardService initialize SEVERE: Failed to initialize connector [Connector[HTTP/1.1-443]] LifecycleException: Protocol handler initialization failed: java.lang.Exception: Invalid Server SSL Protocol (error:00000000:lib(0):func(0):reason(0)) at org.apache.catalina.connector.Connector.initialize(Connector.java:1051) at org.apache.catalina.core.StandardService.initialize(StandardService.java:703) at org.apache.catalina.core.StandardServer.initialize(StandardServer.java:838) at org.apache.catalina.startup.Catalina.load(Catalina.java:538) at org.apache.catalina.startup.Catalina.load(Catalina.java:562) at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) at java.lang.reflect.Method.invoke(Method.java:597) at org.apache.catalina.startup.Bootstrap.load(Bootstrap.java:261) at org.apache.catalina.startup.Bootstrap.main(Bootstrap.java:413) I've checked the following things already I have given read permissions for everyone for .crt and .key files I copied server.xml to a different working tomcat6 server and it works there, server.xml from the mentioned working tomcat5 webserver doesn't work here and it fails with the same error Works well with just HTTP enabled explicitly mentioning protocol in the Connector i.e. protocol="org.apache.coyote.http11.Http11AprProtocol" results in the same exception Please help me if I am missing something. Thanks in advance

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  • Use an Ubuntu Live CD to Securely Wipe Your PC’s Hard Drive

    - by Trevor Bekolay
    Deleting files or quickly formatting a drive isn’t enough for sensitive personal information. We’ll show you how to get rid of it for good using a Ubuntu Live CD. When you delete a file in Windows, Ubuntu, or any other operating system, it doesn’t actually destroy the data stored on your hard drive, it just marks that data as “deleted.” If you overwrite it later, then that data is generally unrecoverable, but if the operating system don’t happen to overwrite it, then your data is still stored on your hard drive, recoverable by anyone who has the right software. By securely delete files or entire hard drives, your data will be gone for good. Note: Modern hard drives are extremely sophisticated, as are the experts who recover data for a living. There is no guarantee that the methods covered in this article will make your data completely unrecoverable; however, they will make your data unrecoverable to the majority of recovery methods, and all methods that are readily available to the general public. Shred individual files Most of the data stored on your hard drive is harmless, and doesn’t reveal anything about you. If there are just a few files that you know you don’t want someone else to see, then the easiest way to get rid of them is a built-in Linux utility called shred. Open a terminal window by clicking on Applications at the top-left of the screen, then expanding the Accessories menu and clicking on Terminal. Navigate to the file that you want to delete using cd to change directories and ls to list the files and folders in the current directory. As an example, we’ve got a file called BankInfo.txt on a Windows NTFS-formatted hard drive. We want to delete it securely, so we’ll call shred by entering the following in the terminal window: shred <file> which is, in our example: shred BankInfo.txt Notice that our BankInfo.txt file still exists, even though we’ve shredded it. A quick look at the contents of BankInfo.txt make it obvious that the file has indeed been securely overwritten. We can use some command-line arguments to make shred delete the file from the hard drive as well. We can also be extra-careful about the shredding process by upping the number of times shred overwrites the original file. To do this, in the terminal, type in: shred –remove –iterations=<num> <file> By default, shred overwrites the file 25 times. We’ll double this, giving us the following command: shred –remove –iterations=50 BankInfo.txt BankInfo.txt has now been securely wiped on the physical disk, and also no longer shows up in the directory listing. Repeat this process for any sensitive files on your hard drive! Wipe entire hard drives If you’re disposing of an old hard drive, or giving it to someone else, then you might instead want to wipe your entire hard drive. shred can be invoked on hard drives, but on modern file systems, the shred process may be reversible. We’ll use the program wipe to securely delete all of the data on a hard drive. Unlike shred, wipe is not included in Ubuntu by default, so we have to install it. Open up the Synaptic Package Manager by clicking on System in the top-left corner of the screen, then expanding the Administration folder and clicking on Synaptic Package Manager. wipe is part of the Universe repository, which is not enabled by default. We’ll enable it by clicking on Settings > Repositories in the Synaptic Package Manager window. Check the checkbox next to “Community-maintained Open Source software (universe)”. Click Close. You’ll need to reload Synaptic’s package list. Click on the Reload button in the main Synaptic Package Manager window. Once the package list has been reloaded, the text over the search field will change to “Rebuilding search index”. Wait until it reads “Quick search,” and then type “wipe” into the search field. The wipe package should come up, along with some other packages that perform similar functions. Click on the checkbox to the left of the label “wipe” and select “Mark for Installation”. Click on the Apply button to start the installation process. Click the Apply button on the Summary window that pops up. Once the installation is done, click the Close button and close the Synaptic Package Manager window. Open a terminal window by clicking on Applications in the top-left of the screen, then Accessories > Terminal. You need to figure our the correct hard drive to wipe. If you wipe the wrong hard drive, that data will not be recoverable, so exercise caution! In the terminal window, type in: sudo fdisk -l A list of your hard drives will show up. A few factors will help you identify the right hard drive. One is the file system, found in the System column of  the list – Windows hard drives are usually formatted as NTFS (which shows up as HPFS/NTFS). Another good identifier is the size of the hard drive, which appears after its identifier (highlighted in the following screenshot). In our case, the hard drive we want to wipe is only around 1 GB large, and is formatted as NTFS. We make a note of the label found under the the Device column heading. If you have multiple partitions on this hard drive, then there will be more than one device in this list. The wipe developers recommend wiping each partition separately. To start the wiping process, type the following into the terminal: sudo wipe <device label> In our case, this is: sudo wipe /dev/sda1 Again, exercise caution – this is the point of no return! Your hard drive will be completely wiped. It may take some time to complete, depending on the size of the drive you’re wiping. Conclusion If you have sensitive information on your hard drive – and chances are you probably do – then it’s a good idea to securely delete sensitive files before you give away or dispose of your hard drive. The most secure way to delete your data is with a few swings of a hammer, but shred and wipe from a Ubuntu Live CD is a good alternative! Similar Articles Productive Geek Tips Reset Your Ubuntu Password Easily from the Live CDScan a Windows PC for Viruses from a Ubuntu Live CDRecover Deleted Files on an NTFS Hard Drive from a Ubuntu Live CDCreate a Bootable Ubuntu 9.10 USB Flash DriveCreate a Bootable Ubuntu USB Flash Drive the Easy Way TouchFreeze Alternative in AutoHotkey The Icy Undertow Desktop Windows Home Server – Backup to LAN The Clear & Clean Desktop Use This Bookmarklet to Easily Get Albums Use AutoHotkey to Assign a Hotkey to a Specific Window Latest Software Reviews Tinyhacker Random Tips DVDFab 6 Revo Uninstaller Pro Registry Mechanic 9 for Windows PC Tools Internet Security Suite 2010 Office 2010 Product Guides Google Maps Place marks – Pizza, Guns or Strip Clubs Monitor Applications With Kiwi LocPDF is a Visual PDF Search Tool Download Free iPad Wallpapers at iPad Decor Get Your Delicious Bookmarks In Firefox’s Awesome Bar

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  • Errors trying to run MongoDB

    - by SomeKittens
    I'm running Ubuntu Server 12.04 (32 bit) on an old (1998) computer. Everything's working fine until I try and start MongoDB. somekittens@DLserver01:~$ mongo MongoDB shell version: 2.2.2 connecting to: test Sun Dec 16 22:47:50 Error: couldn't connect to server 127.0.0.1:27017 src/mongo/shell/mongo.js:91 exception: connect failed Googling the error lead me to all sorts of "repair" options, none of which fixed anything. I've also removed MongoDB and installed it again (using apt-get, have not built from source). Mongo's log shows the following error: Thu Dec 13 18:36:32 warning: 32-bit servers don't have journaling enabled by default. Please use --journal if you want durability. Thu Dec 13 18:36:32 Thu Dec 13 18:36:32 [initandlisten] MongoDB starting : pid=758 port=27017 dbpath=/var/lib/mongodb 32-bit host=DLserver01 Thu Dec 13 18:36:32 [initandlisten] Thu Dec 13 18:36:32 [initandlisten] ** NOTE: when using MongoDB 32 bit, you are limited to about 2 gigabytes of data Thu Dec 13 18:36:32 [initandlisten] ** see http://blog.mongodb.org/post/137788967/32-bit-limitations Thu Dec 13 18:36:32 [initandlisten] ** with --journal, the limit is lower Thu Dec 13 18:36:32 [initandlisten] Thu Dec 13 18:36:32 [initandlisten] db version v2.2.2, pdfile version 4.5 Thu Dec 13 18:36:32 [initandlisten] git version: d1b43b61a5308c4ad0679d34b262c5af9d664267 Thu Dec 13 18:36:32 [initandlisten] build info: Linux domU-12-31-39-01-70-B4 2.6.21.7-2.fc8xen #1 SMP Fri Feb 15 12:39:36 EST 2008 i686 BOOST_LIB_VERSION=1_49 Thu Dec 13 18:36:32 [initandlisten] options: { config: "/etc/mongodb.conf", dbpath: "/var/lib/mongodb", logappend: "true", logpath: "/var/log/mongodb/mongodb.log" } Thu Dec 13 18:36:32 [initandlisten] Unable to check for journal files due to: boost::filesystem::basic_directory_iterator constructor: No such file or directory: "/var/lib/mongodb/journal" ************** Unclean shutdown detected. Please visit http://dochub.mongodb.org/core/repair for recovery instructions. ************* Thu Dec 13 18:36:32 [initandlisten] exception in initAndListen: 12596 old lock file, terminating Thu Dec 13 18:36:32 dbexit: Thu Dec 13 18:36:32 [initandlisten] shutdown: going to close listening sockets... Thu Dec 13 18:36:32 [initandlisten] shutdown: going to flush diaglog... Thu Dec 13 18:36:32 [initandlisten] shutdown: going to close sockets... Thu Dec 13 18:36:32 [initandlisten] shutdown: waiting for fs preallocator... Thu Dec 13 18:36:32 [initandlisten] shutdown: closing all files... Thu Dec 13 18:36:32 [initandlisten] closeAllFiles() finished Thu Dec 13 18:36:32 dbexit: really exiting now Running through the recovery instructions lead to the following adventure: somekittens@DLserver01:/var/log/mongodb$ mongod --repair Sun Dec 16 22:42:54 Sun Dec 16 22:42:54 warning: 32-bit servers don't have journaling enabled by default. Please use --journal if you want durability. Sun Dec 16 22:42:54 Sun Dec 16 22:42:54 [initandlisten] MongoDB starting : pid=1887 port=27017 dbpath=/data/db/ 32-bit host=DLserver01 Sun Dec 16 22:42:54 [initandlisten] Sun Dec 16 22:42:54 [initandlisten] ** NOTE: when using MongoDB 32 bit, you are limited to about 2 gigabytes of data Sun Dec 16 22:42:54 [initandlisten] ** see http://blog.mongodb.org/post/137788967/32-bit-limitations Sun Dec 16 22:42:54 [initandlisten] ** with --journal, the limit is lower Sun Dec 16 22:42:54 [initandlisten] Sun Dec 16 22:42:54 [initandlisten] db version v2.2.2, pdfile version 4.5 Sun Dec 16 22:42:54 [initandlisten] git version: d1b43b61a5308c4ad0679d34b262c5af9d664267 Sun Dec 16 22:42:54 [initandlisten] build info: Linux domU-12-31-39-01-70-B4 2.6.21.7-2.fc8xen #1 SMP Fri Feb 15 12:39:36 EST 2008 i686 BOOST_LIB_VERSION=1_49 Sun Dec 16 22:42:54 [initandlisten] options: { repair: true } Sun Dec 16 22:42:54 [initandlisten] exception in initAndListen: 10296 ********************************************************************* ERROR: dbpath (/data/db/) does not exist. Create this directory or give existing directory in --dbpath. See http://dochub.mongodb.org/core/startingandstoppingmongo ********************************************************************* , terminating Sun Dec 16 22:42:54 dbexit: Sun Dec 16 22:42:54 [initandlisten] shutdown: going to close listening sockets... Sun Dec 16 22:42:54 [initandlisten] shutdown: going to flush diaglog... Sun Dec 16 22:42:54 [initandlisten] shutdown: going to close sockets... Sun Dec 16 22:42:54 [initandlisten] shutdown: waiting for fs preallocator... Sun Dec 16 22:42:54 [initandlisten] shutdown: closing all files... Sun Dec 16 22:42:54 [initandlisten] closeAllFiles() finished Sun Dec 16 22:42:54 dbexit: really exiting now somekittens@DLserver01:/var/log/mongodb$ sudo mkdir /data somekittens@DLserver01:/var/log/mongodb$ sudo mkdir /data/db somekittens@DLserver01:/var/log/mongodb$ mongod --repair Sun Dec 16 22:43:51 Sun Dec 16 22:43:51 warning: 32-bit servers don't have journaling enabled by default. Please use --journal if you want durability. Sun Dec 16 22:43:51 Sun Dec 16 22:43:51 [initandlisten] MongoDB starting : pid=1909 port=27017 dbpath=/data/db/ 32-bit host=DLserver01 Sun Dec 16 22:43:51 [initandlisten] Sun Dec 16 22:43:51 [initandlisten] ** NOTE: when using MongoDB 32 bit, you are limited to about 2 gigabytes of data Sun Dec 16 22:43:51 [initandlisten] ** see http://blog.mongodb.org/post/137788967/32-bit-limitations Sun Dec 16 22:43:51 [initandlisten] ** with --journal, the limit is lower Sun Dec 16 22:43:51 [initandlisten] Sun Dec 16 22:43:51 [initandlisten] db version v2.2.2, pdfile version 4.5 Sun Dec 16 22:43:51 [initandlisten] git version: d1b43b61a5308c4ad0679d34b262c5af9d664267 Sun Dec 16 22:43:51 [initandlisten] build info: Linux domU-12-31-39-01-70-B4 2.6.21.7-2.fc8xen #1 SMP Fri Feb 15 12:39:36 EST 2008 i686 BOOST_LIB_VERSION=1_49 Sun Dec 16 22:43:51 [initandlisten] options: { repair: true } Sun Dec 16 22:43:51 [initandlisten] exception in initAndListen: 10309 Unable to create/open lock file: /data/db/mongod.lock errno:13 Permission denied Is a mongod instance already running?, terminating Sun Dec 16 22:43:51 dbexit: Sun Dec 16 22:43:51 [initandlisten] shutdown: going to close listening sockets... Sun Dec 16 22:43:51 [initandlisten] shutdown: going to flush diaglog... Sun Dec 16 22:43:51 [initandlisten] shutdown: going to close sockets... Sun Dec 16 22:43:51 [initandlisten] shutdown: waiting for fs preallocator... Sun Dec 16 22:43:51 [initandlisten] shutdown: closing all files... Sun Dec 16 22:43:51 [initandlisten] closeAllFiles() finished Sun Dec 16 22:43:51 [initandlisten] shutdown: removing fs lock... Sun Dec 16 22:43:51 [initandlisten] couldn't remove fs lock errno:9 Bad file descriptor Sun Dec 16 22:43:51 dbexit: really exiting now somekittens@DLserver01:/var/log/mongodb$ service mongodb stop stop: Unknown instance: somekittens@DLserver01:/var/log/mongodb$ sudo mongod --repair Sun Dec 16 22:45:04 Sun Dec 16 22:45:04 warning: 32-bit servers don't have journaling enabled by default. Please use --journal if you want durability. Sun Dec 16 22:45:04 Sun Dec 16 22:45:04 [initandlisten] MongoDB starting : pid=1921 port=27017 dbpath=/data/db/ 32-bit host=DLserver01 Sun Dec 16 22:45:04 [initandlisten] Sun Dec 16 22:45:04 [initandlisten] ** NOTE: when using MongoDB 32 bit, you are limited to about 2 gigabytes of data Sun Dec 16 22:45:04 [initandlisten] ** see http://blog.mongodb.org/post/137788967/32-bit-limitations Sun Dec 16 22:45:04 [initandlisten] ** with --journal, the limit is lower Sun Dec 16 22:45:04 [initandlisten] Sun Dec 16 22:45:04 [initandlisten] db version v2.2.2, pdfile version 4.5 Sun Dec 16 22:45:04 [initandlisten] git version: d1b43b61a5308c4ad0679d34b262c5af9d664267 Sun Dec 16 22:45:04 [initandlisten] build info: Linux domU-12-31-39-01-70-B4 2.6.21.7-2.fc8xen #1 SMP Fri Feb 15 12:39:36 EST 2008 i686 BOOST_LIB_VERSION=1_49 Sun Dec 16 22:45:04 [initandlisten] options: { repair: true } Sun Dec 16 22:45:04 [initandlisten] Unable to check for journal files due to: boost::filesystem::basic_directory_iterator constructor: No such file or directory: "/data/db/journal" Sun Dec 16 22:45:04 [initandlisten] finished checking dbs Sun Dec 16 22:45:04 dbexit: Sun Dec 16 22:45:04 [initandlisten] shutdown: going to close listening sockets... Sun Dec 16 22:45:04 [initandlisten] shutdown: going to flush diaglog... Sun Dec 16 22:45:04 [initandlisten] shutdown: going to close sockets... Sun Dec 16 22:45:04 [initandlisten] shutdown: waiting for fs preallocator... Sun Dec 16 22:45:04 [initandlisten] shutdown: closing all files... Sun Dec 16 22:45:04 [initandlisten] closeAllFiles() finished Sun Dec 16 22:45:04 [initandlisten] shutdown: removing fs lock... Sun Dec 16 22:45:04 dbexit: really exiting now Which didn't change anything. What can I do to resolve this? It's an old computer (640MB RAM, single-core P2). Could that be causing it?

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  • SQL SERVER – Extending SQL Azure with Azure worker role – Guest Post by Paras Doshi

    - by pinaldave
    This is guest post by Paras Doshi. Paras Doshi is a research Intern at SolidQ.com and a Microsoft student partner. He is currently working in the domain of SQL Azure. SQL Azure is nothing but a SQL server in the cloud. SQL Azure provides benefits such as on demand rapid provisioning, cost-effective scalability, high availability and reduced management overhead. To see an introduction on SQL Azure, check out the post by Pinal here In this article, we are going to discuss how to extend SQL Azure with the Azure worker role. In other words, we will attempt to write a custom code and host it in the Azure worker role; the aim is to add some features that are not available with SQL Azure currently or features that need to be customized for flexibility. This way we extend the SQL Azure capability by building some solutions that run on Azure as worker roles. To understand Azure worker role, think of it as a windows service in cloud. Azure worker role can perform background processes, and to handle processes such as synchronization and backup, it becomes our ideal tool. First, we will focus on writing a worker role code that synchronizes SQL Azure databases. Before we do so, let’s see some scenarios in which synchronization between SQL Azure databases is beneficial: scaling out access over multiple databases enables us to handle workload efficiently As of now, SQL Azure database can be hosted in one of any six datacenters. By synchronizing databases located in different data centers, one can extend the data by enabling access to geographically distributed data Let us see some scenarios in which SQL server to SQL Azure database synchronization is beneficial To backup SQL Azure database on local infrastructure Rather than investing in local infrastructure for increased workloads, such workloads could be handled by cloud Ability to extend data to different datacenters located across the world to enable efficient data access from remote locations Now, let us develop cloud-based app that synchronizes SQL Azure databases. For an Introduction to developing cloud based apps, click here Now, in this article, I aim to provide a bird’s eye view of how a code that synchronizes SQL Azure databases look like and then list resources that can help you develop the solution from scratch. Now, if you newly add a worker role to the cloud-based project, this is how the code will look like. (Note: I have added comments to the skeleton code to point out the modifications that will be required in the code to carry out the SQL Azure synchronization. Note the placement of Setup() and Sync() function.) Click here (http://parasdoshi1989.files.wordpress.com/2011/06/code-snippet-1-for-extending-sql-azure-with-azure-worker-role1.pdf ) Enabling SQL Azure databases synchronization through sync framework is a two-step process. In the first step, the database is provisioned and sync framework creates tracking tables, stored procedures, triggers, and tables to store metadata to enable synchronization. This is one time step. The code for the same is put in the setup() function which is called once when the worker role starts. Now, the second step is continuous (or on demand) synchronization of SQL Azure databases by propagating changes between databases. This is done on a continuous basis by calling the sync() function in the while loop. The code logic to synchronize changes between SQL Azure databases should be put in the sync() function. Discussing the coding part step by step is out of the scope of this article. Therefore, let me suggest you a resource, which is given here. Also, note that before you start developing the code, you will need to install SYNC framework 2.1 SDK (download here). Further, you will reference some libraries before you start coding. Details regarding the same are available in the article that I just pointed to. You will be charged for data transfers if the databases are not in the same datacenter. For pricing information, go here Currently, a tool named DATA SYNC, which is built on top of sync framework, is available in CTP that allows SQL Azure <-> SQL server and SQL Azure <-> SQL Azure synchronization (without writing single line of code); however, in some cases, the custom code shown in this blogpost provides flexibility that is not available with Data SYNC. For instance, filtering is not supported in the SQL Azure DATA SYNC CTP2; if you wish to have such a functionality now, then you have the option of developing a custom code using SYNC Framework. Now, this code can be easily extended to synchronize at some schedule. Let us say we want the databases to get synchronized every day at 10:00 pm. This is what the code will look like now: (http://parasdoshi1989.files.wordpress.com/2011/06/code-snippet-2-for-extending-sql-azure-with-azure-worker-role.pdf) Don’t you think that by writing such a code, we are imitating the functionality provided by the SQL server agent for a SQL server? Think about it. We are scheduling our administrative task by writing custom code – in other words, we have developed a “Light weight SQL server agent for SQL Azure!” Since the SQL server agent is not currently available in cloud, we have developed a solution that enables us to schedule tasks, and thus we have extended SQL Azure with the Azure worker role! Now if you wish to track jobs, you can do so by storing this data in SQL Azure (or Azure tables). The reason is that Windows Azure is a stateless platform, and we will need to store the state of the job ourselves and the choice that you have is SQL Azure or Azure tables. Note that this solution requires custom code and also it is not UI driven; however, for now, it can act as a temporary solution until SQL server agent is made available in the cloud. Moreover, this solution does not encompass functionalities that a SQL server agent provides, but it does open up an interesting avenue to schedule some of the tasks such as backup and synchronization of SQL Azure databases by writing some custom code in the Azure worker role. Now, let us see one more possibility – i.e., running BCP through a worker role in Azure-hosted services and then uploading the backup files either locally or on blobs. If you upload it locally, then consider the data transfer cost. If you upload it to blobs residing in the same datacenter, then no transfer cost applies but the cost on blob size applies. So, before choosing the option, you need to evaluate your preferences keeping the cost associated with each option in mind. In this article, I have shown that Azure worker role solution could be developed to synchronize SQL Azure databases. Moreover, a light-weight SQL server agent for SQL Azure can be developed. Also we discussed the possibility of running BCP through a worker role in Azure-hosted services for backing up our precious SQL Azure data. Thus, we can extend SQL Azure with the Azure worker role. But remember: you will be charged for running Azure worker roles. So at the end of the day, you need to ask – am I willing to build a custom code and pay money to achieve this functionality? I hope you found this blog post interesting. If you have any questions/feedback, you can comment below or you can mail me at Paras[at]student-partners[dot]com Reference: Pinal Dave (http://blog.SQLAuthority.com) Filed under: Pinal Dave, PostADay, SQL, SQL Authority, SQL Azure, SQL Query, SQL Server, SQL Tips and Tricks, T SQL, Technology

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  • SQL SERVER – Working with FileTables in SQL Server 2012 – Part 1 – Setting Up Environment

    - by pinaldave
    Filestream is a very interesting feature, and an enhancement of FileTable with Filestream is equally exciting. Today in this post, we will learn how to set up the FileTable Environment in SQL Server. The major advantage of FileTable is it has Windows API compatibility for file data stored within an SQL Server database. In simpler words, FileTables remove a barrier so that SQL Server can be used for the storage and management of unstructured data that are currently residing as files on file servers. Another advantage is that the Windows Application Compatibility for their existing Windows applications enables to see these data as files in the file system. This way, you can use SQL Server to access the data using T-SQL enhancements, and Windows can access the file using its applications. So for the first step, you will need to enable the Filestream feature at the database level in order to use the FileTable. -- Enable Filestream EXEC sp_configure filestream_access_level, 2 RECONFIGURE GO -- Create Database CREATE DATABASE FileTableDB ON PRIMARY (Name = FileTableDB, FILENAME = 'D:\FileTable\FTDB.mdf'), FILEGROUP FTFG CONTAINS FILESTREAM (NAME = FileTableFS, FILENAME='D:\FileTable\FS') LOG ON (Name = FileTableDBLog, FILENAME = 'D:\FileTable\FTDBLog.ldf') WITH FILESTREAM (NON_TRANSACTED_ACCESS = FULL, DIRECTORY_NAME = N'FileTableDB'); GO Now, you can run the following code and figure out if FileStream options are enabled at the database level. -- Check the Filestream Options SELECT DB_NAME(database_id), non_transacted_access, non_transacted_access_desc FROM sys.database_filestream_options; GO You can see the resultset of the above query which returns resultset as the following image shows. As you can see , the file level access is set to 2 (filestream enabled). Now let us create the filetable in the newly created database. -- Create FileTable Table USE FileTableDB GO CREATE TABLE FileTableTb AS FileTable WITH (FileTable_Directory = 'FileTableTb_Dir'); GO Now you can select data using a regular select table. SELECT * FROM FileTableTb GO It will return all the important columns which are related to the file. It will provide details like filesize, archived, file types etc. You can also see the FileTable in SQL Server Management Studio. Go to Databases >> Newly Created Database (FileTableDB) >> Expand Tables Here, you will see a new folder which says “FileTables”. When expanded, it gives the name of the newly created FileTableTb. You can right click on the newly created table and click on “Explore FileTable Directory”. This will open up the folder where the FileTable data will be stored. When you click on the option, it will open up the following folder in my local machine where the FileTable data will be stored: \\127.0.0.1\mssqlserver\FileTableDB\FileTableTb_Dir In tomorrow’s blog post as Part 2, we will go over two methods of inserting the data into this FileTable. 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: Filestream

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  • Need to increase nginx throughput to an upstream unix socket -- linux kernel tuning?

    - by Ben Lee
    I am running an nginx server that acts as a proxy to an upstream unix socket, like this: upstream app_server { server unix:/tmp/app.sock fail_timeout=0; } server { listen ###.###.###.###; server_name whatever.server; root /web/root; try_files $uri @app; location @app { proxy_set_header X-Forwarded-For $proxy_add_x_forwarded_for; proxy_set_header X-Forwarded-Proto $scheme; proxy_set_header Host $http_host; proxy_redirect off; proxy_pass http://app_server; } } Some app server processes, in turn, pull requests off /tmp/app.sock as they become available. The particular app server in use here is Unicorn, but I don't think that's relevant to this question. The issue is, it just seems that past a certain amount of load, nginx can't get requests through the socket at a fast enough rate. It doesn't matter how many app server processes I set up, it doesn't even matter what the app is (tried it with a dummy app with just a single endpoint that returned an empty page with status 404). The bottleneck seems to be the socket, not the app. I'm getting a flood of these messages in the nginx error log: connect() to unix:/tmp/app.sock failed (11: Resource temporarily unavailable) while connecting to upstream Many requests result in status code 502, and those that don't take a long time to complete. The nginx write queue stat hovers around 1000. Anyway, I feel like I'm missing something obvious here, because this particular configuration of nginx and app server is pretty common, especially with Unicorn (it's the recommended method in fact). Are there any linux kernel options that needs to be set, or something in nginx? Any ideas about how to increase the throughput to the upstream socket? Something that I'm clearly doing wrong? Additional information on the environment: $ uname -a Linux app1 3.2.0-24-generic #39-Ubuntu SMP Mon May 21 16:52:17 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux $ ruby -v ruby 1.9.3p194 (2012-04-20 revision 35410) [x86_64-linux] $ unicorn -v unicorn v4.3.1 $ nginx -V nginx version: nginx/1.2.1 built by gcc 4.6.3 (Ubuntu/Linaro 4.6.3-1ubuntu5) TLS SNI support enabled Current kernel tweaks: net.core.rmem_default = 65536 net.core.wmem_default = 65536 net.core.rmem_max = 16777216 net.core.wmem_max = 16777216 net.ipv4.tcp_rmem = 4096 87380 16777216 net.ipv4.tcp_wmem = 4096 65536 16777216 net.ipv4.tcp_mem = 16777216 16777216 16777216 net.ipv4.tcp_window_scaling = 1 net.ipv4.route.flush = 1 net.ipv4.tcp_no_metrics_save = 1 net.ipv4.tcp_moderate_rcvbuf = 1 net.core.somaxconn = 8192 net.netfilter.nf_conntrack_max = 131072

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  • SQL SERVER – A Puzzle – Fun with NULL – Fix Error 8117

    - by pinaldave
    During my 8 years of career, I have been involved in many interviews. Quite often, I act as the  interview. If I am the interviewer, I ask many questions – from easy questions to difficult ones. When I am the interviewee, I frequently get an opportunity to ask the interviewer some questions back. Regardless of the my capacity in attending the interview, I always make it a point to ask the interviewer at least one question. What is NULL? It’s always fun to ask this question during interviews, because in every interview, I get a different answer. NULL is often confused with false, absence of value or infinite value. Honestly, NULL is a very interesting subject as it bases its behavior in server settings. There are a few properties of NULL that are universal, but the knowledge about these properties is not known in a universal sense. Let us run this simple puzzle. Run the following T-SQL script: SELECT SUM(data) FROM (SELECT NULL AS data) t It will return the following error: Msg 8117, Level 16, State 1, Line 1 Operand data type NULL is invalid for sum operator. Now the error makes it very clear that NULL is invalid for sum Operator. Frequently enough, I have showed this simple query to many folks whom I came across. I asked them if they could modify the subquery and return the result as NULL. Here is what I expected: Even though this is a very simple looking query, so far I’ve got the correct answer from only 10% of the people to whom I have asked this question. It was common for me to receive this kind of answer – convert the NULL to some data type. However, doing so usually returns the value as 0 or the integer they passed. SELECT SUM(data) FROM (SELECT ISNULL(NULL,0) AS data) t I usually see many people modifying the outer query to get desired NULL result, but that is not allowed in this simple puzzle. This small puzzle made me wonder how many people have a clear understanding about NULL. Well, here is the answer to my simple puzzle. Just CAST NULL AS INT and it will return the final result as NULL: SELECT SUM(data) FROM (SELECT CAST(NULL AS INT) AS data) t Now that you know the answer, don’t you think it was very simple indeed? This blog post is especially dedicated to my friend Madhivanan who has written an excellent blog post about NULL. I am confident that after reading the blog post from Madhivanan, you will have no confusion regarding NULL in the future. Read: NULL, NULL, NULL and nothing but NULL. Reference: Pinal Dave (http://blog.sqlauthority.com) Filed under: PostADay, SQL, SQL Authority, SQL Puzzle, SQL Query, SQL Scripts, SQL Server, SQL Tips and Tricks, T SQL, Technology

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  • Book Review (Book 10) - The Information: A History, a Theory, a Flood

    - by BuckWoody
    This is a continuation of the books I challenged myself to read to help my career - one a month, for year. You can read my first book review here, and the entire list is here. The book I chose for March 2012 was: The Information: A History, a Theory, a Flood by James Gleick. I was traveling at the end of last month so I’m a bit late posting this review here. Why I chose this book: My personal belief about computing is this: All computing technology is simply re-arranging data. We take data in, we manipulate it, and we send it back out. That’s computing. I had heard from some folks about this book and it’s treatment of data. I heard that it dealt with the basics of data - and the semantics of data, information and so on. It also deals with the earliest forms of history of information, which fascinates me. It’s similar I was told, to GEB which a favorite book of mine as well, so that was a bonus. Some folks I talked to liked it, some didn’t - so I thought I would check it out. What I learned: I liked the book. It was longer than I thought - took quite a while to read, even though I tend to read quickly. This is the kind of book you take your time with. It does in fact deal with the earliest forms of human interaction and the basics of data. I learned, for instance, that the genesis of the binary communication system is based in the invention of telegraph (far-writing) codes, and that the earliest forms of communication were expensive. In fact, many ciphers were invented not to hide military secrets, but to compress information. A sort of early “lol-speak” to keep the cost of transmitting data low! I think the comparison with GEB is a bit over-reaching. GEB is far more specific, fanciful and so on. In fact, this book felt more like something fro Richard Dawkins, and tended to wander around the subject quite a bit. I imagine the author doing his research and writing each chapter as a book that followed on from the last one. This is what possibly bothered those who tended not to like it, I think. Towards the middle of the book, I think the author tended to be a bit too fragmented even for me. He began to delve into memes, biology and more - I think he might have been better off breaking that off into another work. The existentialism just seemed jarring. All in all, I liked the book. I recommend it to any technical professional, specifically ones involved with data technology in specific. And isn’t that all of us? :)

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  • Performance problems loading XML with SSIS, an alternative way!

    - by AtulThakor
    I recently needed to load several thousand XML files into a SQL database, I created an SSIS package which was created as followed: Using a foreach container to loop through a directory and load each file path into a variable, the “Import XML” dataflow would then load each XML file into a SQL table.       Running this, it took approximately 1 second to load each file which seemed a massive amount of time to parse the XML and load the data, speaking to my colleague Martin Croft, he suggested the use of T-SQL Bulk Insert and OpenRowset, so we adjusted the package as followed:     The same foreach container was used but instead the following SQL command was executed (this is an expression):     "INSERT INTO MyTable(FileDate) SELECT   CAST(bulkcolumn AS XML)     FROM OPENROWSET(         BULK         '" + @[User::CurrentFile]  + "',         SINGLE_BLOB ) AS x"     Using this method we managed to load approximately 20 records per second, much faster…for data loading! For what we wanted to achieve this was perfect but I’ll leave you with the following points when making your own decision on which solution you decide to choose!      Openrowset Method Much faster to get the data into SQL You’ll need to parse or create a view over the XML data to allow the data to be more usable(another post on this!) Not able to apply validation/transformation against the data when loading it The SQL Server service account will need permission to the file No schema validation when loading files SSIS Slower (in our case) Schema validation Allows you to apply transformations/joins to the data Permissions should be less of a problem Data can be loaded into the final form through the package When using a schema validation errors can fail the package (I’ll do another post on this)

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  • SQL SERVER – SSMS: Disk Usage Report

    - by Pinal Dave
    Let us start with humor!  I think we the series on various reports, we come to a logical point. We covered all the reports at server level. This means the reports we saw were targeted towards activities that are related to instance level operations. These are mostly like how a doctor diagnoses a patient. At this point I am reminded of a dialog which I read somewhere: Patient: Doc, It hurts when I touch my head. Doc: Ok, go on. What else have you experienced? Patient: It hurts even when I touch my eye, it hurts when I touch my arms, it even hurts when I touch my feet, etc. Doc: Hmmm … Patient: I feel it hurts when I touch anywhere in my body. Doc: Ahh … now I get it. You need a plaster to your finger John. Sometimes the server level gives an indicator to what is happening in the system, but we need to get to the root cause for a specific database. So, this is the first blog in series where we would start discussing about database level reports. To launch database level reports, expand selected server in Object Explorer, expand the Databases folder, and then right-click any database for which we want to look at reports. From the menu, select Reports, then Standard Reports, and then any of database level reports. In this blog, we would talk about four “disk” reports because they are similar: Disk Usage Disk Usage by Top Tables Disk Usage by Table Disk Usage by Partition Disk Usage This report shows multiple information about the database. Let us discuss them one by one.  We have divided the output into 5 different sections. Section 1 shows the high level summary of the database. It shows the space used by database files (mdf and ldf). Under the hood, the report uses, various DMVs and DBCC Commands, it is using sys.data_spaces and DBCC SHOWFILESTATS. Section 2 and 3 are pie charts. One for data file allocation and another for the transaction log file. Pie chart for “Data Files Space Usage (%)” shows space consumed data, indexes, allocated to the SQL Server database, and unallocated space which is allocated to the SQL Server database but not yet filled with anything. “Transaction Log Space Usage (%)” used DBCC SQLPERF (LOGSPACE) and shows how much empty space we have in the physical transaction log file. Section 4 shows the data from Default Trace and looks at Event IDs 92, 93, 94, 95 which are for “Data File Auto Grow”, “Log File Auto Grow”, “Data File Auto Shrink” and “Log File Auto Shrink” respectively. Here is an expanded view for that section. If default trace is not enabled, then this section would be replaced by the message “Trace Log is disabled” as highlighted below. Section 5 of the report uses DBCC SHOWFILESTATS to get information. Here is the enhanced version of that section. This shows the physical layout of the file. In case you have In-Memory Objects in the database (from SQL Server 2014), then report would show information about those as well. Here is the screenshot taken for a different database, which has In-Memory table. I have highlighted new things which are only shown for in-memory database. The new sections which are highlighted above are using sys.dm_db_xtp_checkpoint_files, sys.database_files and sys.data_spaces. The new type for in-memory OLTP is ‘FX’ in sys.data_space. The next set of reports is targeted to get information about a table and its storage. These reports can answer questions like: Which is the biggest table in the database? How many rows we have in table? Is there any table which has a lot of reserved space but its unused? Which partition of the table is having more data? Disk Usage by Top Tables This report provides detailed data on the utilization of disk space by top 1000 tables within the Database. The report does not provide data for memory optimized tables. Disk Usage by Table This report is same as earlier report with few difference. First Report shows only 1000 rows First Report does order by values in DMV sys.dm_db_partition_stats whereas second one does it based on name of the table. Both of the reports have interactive sort facility. We can click on any column header and change the sorting order of data. Disk Usage by Partition This report shows the distribution of the data in table based on partition in the table. This is so similar to previous output with the partition details now. Here is the query taken from profiler. SELECT row_number() OVER (ORDER BY a1.used_page_count DESC, a1.index_id) AS row_number ,      (dense_rank() OVER (ORDER BY a5.name, a2.name))%2 AS l1 ,      a1.OBJECT_ID ,      a5.name AS [schema] ,       a2.name ,       a1.index_id ,       a3.name AS index_name ,       a3.type_desc ,       a1.partition_number ,       a1.used_page_count * 8 AS total_used_pages ,       a1.reserved_page_count * 8 AS total_reserved_pages ,       a1.row_count FROM sys.dm_db_partition_stats a1 INNER JOIN sys.all_objects a2  ON ( a1.OBJECT_ID = a2.OBJECT_ID) AND a1.OBJECT_ID NOT IN (SELECT OBJECT_ID FROM sys.tables WHERE is_memory_optimized = 1) INNER JOIN sys.schemas a5 ON (a5.schema_id = a2.schema_id) LEFT OUTER JOIN  sys.indexes a3  ON ( (a1.OBJECT_ID = a3.OBJECT_ID) AND (a1.index_id = a3.index_id) ) WHERE (SELECT MAX(DISTINCT partition_number) FROM sys.dm_db_partition_stats a4 WHERE (a4.OBJECT_ID = a1.OBJECT_ID)) >= 1 AND a2.TYPE <> N'S' AND  a2.TYPE <> N'IT' ORDER BY a5.name ASC, a2.name ASC, a1.index_id, a1.used_page_count DESC, a1.partition_number Using all of the above reports, you should be able to get the usage of database files and also space used by tables. I think this is too much disk information for a single blog and I hope you have used them in the past to get data. Do let me know if you found anything interesting using these reports in your environments. Reference: Pinal Dave (http://blog.sqlauthority.com)Filed under: PostADay, SQL, SQL Authority, SQL Query, SQL Server, SQL Server Management Studio, SQL Tips and Tricks, T SQL Tagged: SQL Reports

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  • IBM System x3850 X5 TPC-H Benchmark

    - by jchang
    IBM just published a TPC-H SF 1000 result for their x3850 X5 , 4-way Xeon 7560 system featuring a special MAX5 memory expansion board to support 1.5TB memory. In Dec 2010, IBM also published a TPC-H SF1000 for their Power 780 system, 8-way, quad-core, (4 logical processors per physical core). In Feb 2011, Ingres published a TPC-H SF 100 on a 2-way Xeon 5680 for their VectorWise column-store engine (plus enhancements for memory architecture, SIMD and compression). The figure table below shows TPC-H...(read more)

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  • Oracle’s Sun Server X4-8 with Built-in Elastic Computing

    - by kgee
    We are excited to announce the release of Oracle's new 8-socket server, Sun Server X4-8. It’s the most flexible 8-socket x86 server Oracle has ever designed, and also the most powerful. Not only does it use the fastest Intel® Xeon® E7 v2 processors, but also its memory, I/O and storage subsystems are all designed for maximum performance and throughput. Like its predecessor, the Sun Server X4-8 uses a “glueless” design that allows for maximum performance for Oracle Database, while also reducing power consumption and improving reliability. The specs are pretty impressive. Sun Server X4-8 supports 120 cores (or 240 threads), 6 TB memory, 9.6 TB HDD capacity or 3.2 TB SSD capacity, contains 16 PCIe Gen 3 I/O expansion slots, and allows for up to 6.4 TB Sun Flash Accelerator F80 PCIe Cards. The Sun Server X4-8 is also the most dense x86 server with its 5U chassis, allowing 60% higher rack-level core and DIMM slot density than the competition.  There has been a lot of innovation in Oracle’s x86 product line, but the latest and most significant is a capability called elastic computing. This new capability is built into each Sun Server X4-8.   Elastic computing starts with the Intel processor. While Intel provides a wide range of processors each with a fixed combination of core count, operational frequency, and power consumption, customers have been forced to make tradeoffs when they select a particular processor. They have had to make educated guesses on which particular processor (core count/frequency/cache size) will be best suited for the workload they intend to execute on the server.Oracle and Intel worked jointly to define a new processor, the Intel Xeon E7-8895 v2 for the Sun Server X4-8, that has unique characteristics and effectively combines the capabilities of three different Xeon processors into a single processor. Oracle system design engineers worked closely with Oracle’s operating system development teams to achieve the ability to vary the core count and operating frequency of the Xeon E7-8895 v2 processor with time without the need for a system level reboot.  Along with the new processor, enhancements have been made to the system BIOS, Oracle Solaris, and Oracle Linux, which allow the processors in the system to dynamically clock up to faster speeds as cores are disabled and to reach higher maximum turbo frequencies for the remaining active cores. One customer, a stock market trading company, will take advantage of the elastic computing capability of Sun Server X4-8 by repurposing servers between daytime stock trading activity and nighttime stock portfolio processing, daily, to achieve maximum performance of each workload.To learn more about Sun Server X4-8, you can find more details including the data sheet and white papers here.Josh Rosen is a Principal Product Manager for Oracle’s x86 servers, focusing on Oracle’s operating systems and software. He previously spent more than a decade as a developer and architect of system management software. Josh has worked on system management for many of Oracle's hardware products ranging from the earliest blade systems to the latest Oracle x86 servers.

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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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  • Thread placement policies on NUMA systems - update

    - by Dave
    In a prior blog entry I noted that Solaris used a "maximum dispersal" placement policy to assign nascent threads to their initial processors. The general idea is that threads should be placed as far away from each other as possible in the resource topology in order to reduce resource contention between concurrently running threads. This policy assumes that resource contention -- pipelines, memory channel contention, destructive interference in the shared caches, etc -- will likely outweigh (a) any potential communication benefits we might achieve by packing our threads more densely onto a subset of the NUMA nodes, and (b) benefits of NUMA affinity between memory allocated by one thread and accessed by other threads. We want our threads spread widely over the system and not packed together. Conceptually, when placing a new thread, the kernel picks the least loaded node NUMA node (the node with lowest aggregate load average), and then the least loaded core on that node, etc. Furthermore, the kernel places threads onto resources -- sockets, cores, pipelines, etc -- without regard to the thread's process membership. That is, initial placement is process-agnostic. Keep reading, though. This description is incorrect. On Solaris 10 on a SPARC T5440 with 4 x T2+ NUMA nodes, if the system is otherwise unloaded and we launch a process that creates 20 compute-bound concurrent threads, then typically we'll see a perfect balance with 5 threads on each node. We see similar behavior on an 8-node x86 x4800 system, where each node has 8 cores and each core is 2-way hyperthreaded. So far so good; this behavior seems in agreement with the policy I described in the 1st paragraph. I recently tried the same experiment on a 4-node T4-4 running Solaris 11. Both the T5440 and T4-4 are 4-node systems that expose 256 logical thread contexts. To my surprise, all 20 threads were placed onto just one NUMA node while the other 3 nodes remained completely idle. I checked the usual suspects such as processor sets inadvertently left around by colleagues, processors left offline, and power management policies, but the system was configured normally. I then launched multiple concurrent instances of the process, and, interestingly, all the threads from the 1st process landed on one node, all the threads from the 2nd process landed on another node, and so on. This happened even if I interleaved thread creating between the processes, so I was relatively sure the effect didn't related to thread creation time, but rather that placement was a function of process membership. I this point I consulted the Solaris sources and talked with folks in the Solaris group. The new Solaris 11 behavior is intentional. The kernel is no longer using a simple maximum dispersal policy, and thread placement is process membership-aware. Now, even if other nodes are completely unloaded, the kernel will still try to pack new threads onto the home lgroup (socket) of the primordial thread until the load average of that node reaches 50%, after which it will pick the next least loaded node as the process's new favorite node for placement. On the T4-4 we have 64 logical thread contexts (strands) per socket (lgroup), so if we launch 48 concurrent threads we will find 32 placed on one node and 16 on some other node. If we launch 64 threads we'll find 32 and 32. That means we can end up with our threads clustered on a small subset of the nodes in a way that's quite different that what we've seen on Solaris 10. So we have a policy that allows process-aware packing but reverts to spreading threads onto other nodes if a node becomes too saturated. It turns out this policy was enabled in Solaris 10, but certain bugs suppressed the mixed packing/spreading behavior. There are configuration variables in /etc/system that allow us to dial the affinity between nascent threads and their primordial thread up and down: see lgrp_expand_proc_thresh, specifically. In the OpenSolaris source code the key routine is mpo_update_tunables(). This method reads the /etc/system variables and sets up some global variables that will subsequently be used by the dispatcher, which calls lgrp_choose() in lgrp.c to place nascent threads. Lgrp_expand_proc_thresh controls how loaded an lgroup must be before we'll consider homing a process's threads to another lgroup. Tune this value lower to have it spread your process's threads out more. To recap, the 'new' policy is as follows. Threads from the same process are packed onto a subset of the strands of a socket (50% for T-series). Once that socket reaches the 50% threshold the kernel then picks another preferred socket for that process. Threads from unrelated processes are spread across sockets. More precisely, different processes may have different preferred sockets (lgroups). Beware that I've simplified and elided details for the purposes of explication. The truth is in the code. Remarks: It's worth noting that initial thread placement is just that. If there's a gross imbalance between the load on different nodes then the kernel will migrate threads to achieve a better and more even distribution over the set of available nodes. Once a thread runs and gains some affinity for a node, however, it becomes "stickier" under the assumption that the thread has residual cache residency on that node, and that memory allocated by that thread resides on that node given the default "first-touch" page-level NUMA allocation policy. Exactly how the various policies interact and which have precedence under what circumstances could the topic of a future blog entry. The scheduler is work-conserving. The x4800 mentioned above is an interesting system. Each of the 8 sockets houses an Intel 7500-series processor. Each processor has 3 coherent QPI links and the system is arranged as a glueless 8-socket twisted ladder "mobius" topology. Nodes are either 1 or 2 hops distant over the QPI links. As an aside the mapping of logical CPUIDs to physical resources is rather interesting on Solaris/x4800. On SPARC/Solaris the CPUID layout is strictly geographic, with the highest order bits identifying the socket, the next lower bits identifying the core within that socket, following by the pipeline (if present) and finally the logical thread context ("strand") on the core. But on Solaris on the x4800 the CPUID layout is as follows. [6:6] identifies the hyperthread on a core; bits [5:3] identify the socket, or package in Intel terminology; bits [2:0] identify the core within a socket. Such low-level details should be of interest only if you're binding threads -- a bad idea, the kernel typically handles placement best -- or if you're writing NUMA-aware code that's aware of the ambient placement and makes decisions accordingly. Solaris introduced the so-called critical-threads mechanism, which is expressed by putting a thread into the FX scheduling class at priority 60. The critical-threads mechanism applies to placement on cores, not on sockets, however. That is, it's an intra-socket policy, not an inter-socket policy. Solaris 11 introduces the Power Aware Dispatcher (PAD) which packs threads instead of spreading them out in an attempt to be able to keep sockets or cores at lower power levels. Maximum dispersal may be good for performance but is anathema to power management. PAD is off by default, but power management polices constitute yet another confounding factor with respect to scheduling and dispatching. If your threads communicate heavily -- one thread reads cache lines last written by some other thread -- then the new dense packing policy may improve performance by reducing traffic on the coherent interconnect. On the other hand if your threads in your process communicate rarely, then it's possible the new packing policy might result on contention on shared computing resources. Unfortunately there's no simple litmus test that says whether packing or spreading is optimal in a given situation. The answer varies by system load, application, number of threads, and platform hardware characteristics. Currently we don't have the necessary tools and sensoria to decide at runtime, so we're reduced to an empirical approach where we run trials and try to decide on a placement policy. The situation is quite frustrating. Relatedly, it's often hard to determine just the right level of concurrency to optimize throughput. (Understanding constructive vs destructive interference in the shared caches would be a good start. We could augment the lines with a small tag field indicating which strand last installed or accessed a line. Given that, we could augment the CPU with performance counters for misses where a thread evicts a line it installed vs misses where a thread displaces a line installed by some other thread.)

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  • Much Ado About Nothing: Stub Objects

    - by user9154181
    The Solaris 11 link-editor (ld) contains support for a new type of object that we call a stub object. A stub object is a shared object, built entirely from mapfiles, that supplies the same linking interface as the real object, while containing no code or data. Stub objects cannot be executed — the runtime linker will kill any process that attempts to load one. However, you can link to a stub object as a dependency, allowing the stub to act as a proxy for the real version of the object. You may well wonder if there is a point to producing an object that contains nothing but linking interface. As it turns out, stub objects are very useful for building large bodies of code such as Solaris. In the last year, we've had considerable success in applying them to one of our oldest and thorniest build problems. In this discussion, I will describe how we came to invent these objects, and how we apply them to building Solaris. This posting explains where the idea for stub objects came from, and details our long and twisty journey from hallway idea to standard link-editor feature. I expect that these details are mainly of interest to those who work on Solaris and its makefiles, those who have done so in the past, and those who work with other similar bodies of code. A subsequent posting will omit the history and background details, and instead discuss how to build and use stub objects. If you are mainly interested in what stub objects are, and don't care about the underlying software war stories, I encourage you to skip ahead. The Long Road To Stubs This all started for me with an email discussion in May of 2008, regarding a change request that was filed in 2002, entitled: 4631488 lib/Makefile is too patient: .WAITs should be reduced This CR encapsulates a number of cronic issues with Solaris builds: We build Solaris with a parallel make (dmake) that tries to build as much of the code base in parallel as possible. There is a lot of code to build, and we've long made use of parallelized builds to get the job done quicker. This is even more important in today's world of massively multicore hardware. Solaris contains a large number of executables and shared objects. Executables depend on shared objects, and shared objects can depend on each other. Before you can build an object, you need to ensure that the objects it needs have been built. This implies a need for serialization, which is in direct opposition to the desire to build everying in parallel. To accurately build objects in the right order requires an accurate set of make rules defining the things that depend on each other. This sounds simple, but the reality is quite complex. In practice, having programmers explicitly specify these dependencies is a losing strategy: It's really hard to get right. It's really easy to get it wrong and never know it because things build anyway. Even if you get it right, it won't stay that way, because dependencies between objects can change over time, and make cannot help you detect such drifing. You won't know that you got it wrong until the builds break. That can be a long time after the change that triggered the breakage happened, making it hard to connect the cause and the effect. Usually this happens just before a release, when the pressure is on, its hard to think calmly, and there is no time for deep fixes. As a poor compromise, the libraries in core Solaris were built using a set of grossly incomplete hand written rules, supplemented with a number of dmake .WAIT directives used to group the libraries into sets of non-interacting groups that can be built in parallel because we think they don't depend on each other. From time to time, someone will suggest that we could analyze the built objects themselves to determine their dependencies and then generate make rules based on those relationships. This is possible, but but there are complications that limit the usefulness of that approach: To analyze an object, you have to build it first. This is a classic chicken and egg scenario. You could analyze the results of a previous build, but then you're not necessarily going to get accurate rules for the current code. It should be possible to build the code without having a built workspace available. The analysis will take time, and remember that we're constantly trying to make builds faster, not slower. By definition, such an approach will always be approximate, and therefore only incremantally more accurate than the hand written rules described above. The hand written rules are fast and cheap, while this idea is slow and complex, so we stayed with the hand written approach. Solaris was built that way, essentially forever, because these are genuinely difficult problems that had no easy answer. The makefiles were full of build races in which the right outcomes happened reliably for years until a new machine or a change in build server workload upset the accidental balance of things. After figuring out what had happened, you'd mutter "How did that ever work?", add another incomplete and soon to be inaccurate make dependency rule to the system, and move on. This was not a satisfying solution, as we tend to be perfectionists in the Solaris group, but we didn't have a better answer. It worked well enough, approximately. And so it went for years. We needed a different approach — a new idea to cut the Gordian Knot. In that discussion from May 2008, my fellow linker-alien Rod Evans had the initial spark that lead us to a game changing series of realizations: The link-editor is used to link objects together, but it only uses the ELF metadata in the object, consisting of symbol tables, ELF versioning sections, and similar data. Notably, it does not look at, or understand, the machine code that makes an object useful at runtime. If you had an object that only contained the ELF metadata for a dependency, but not the code or data, the link-editor would find it equally useful for linking, and would never know the difference. Call it a stub object. In the core Solaris OS, we require all objects to be built with a link-editor mapfile that describes all of its publically available functions and data. Could we build a stub object using the mapfile for the real object? It ought to be very fast to build stub objects, as there are no input objects to process. Unlike the real object, stub objects would not actually require any dependencies, and so, all of the stubs for the entire system could be built in parallel. When building the real objects, one could link against the stub objects instead of the real dependencies. This means that all the real objects can be built built in parallel too, without any serialization. We could replace a system that requires perfect makefile rules with a system that requires no ordering rules whatsoever. The results would be considerably more robust. We immediately realized that this idea had potential, but also that there were many details to sort out, lots of work to do, and that perhaps it wouldn't really pan out. As is often the case, it would be necessary to do the work and see how it turned out. Following that conversation, I set about trying to build a stub object. We determined that a faithful stub has to do the following: Present the same set of global symbols, with the same ELF versioning, as the real object. Functions are simple — it suffices to have a symbol of the right type, possibly, but not necessarily, referencing a null function in its text segment. Copy relocations make data more complicated to stub. The possibility of a copy relocation means that when you create a stub, the data symbols must have the actual size of the real data. Any error in this will go uncaught at link time, and will cause tragic failures at runtime that are very hard to diagnose. For reasons too obscure to go into here, involving tentative symbols, it is also important that the data reside in bss, or not, matching its placement in the real object. If the real object has more than one symbol pointing at the same data item, we call these aliased symbols. All data symbols in the stub object must exhibit the same aliasing as the real object. We imagined the stub library feature working as follows: A command line option to ld tells it to produce a stub rather than a real object. In this mode, only mapfiles are examined, and any object or shared libraries on the command line are are ignored. The extra information needed (function or data, size, and bss details) would be added to the mapfile. When building the real object instead of the stub, the extra information for building stubs would be validated against the resulting object to ensure that they match. In exploring these ideas, I immediately run headfirst into the reality of the original mapfile syntax, a subject that I would later write about as The Problem(s) With Solaris SVR4 Link-Editor Mapfiles. The idea of extending that poor language was a non-starter. Until a better mapfile syntax became available, which seemed unlikely in 2008, the solution could not involve extentions to the mapfile syntax. Instead, we cooked up the idea (hack) of augmenting mapfiles with stylized comments that would carry the necessary information. A typical definition might look like: # DATA(i386) __iob 0x3c0 # DATA(amd64,sparcv9) __iob 0xa00 # DATA(sparc) __iob 0x140 iob; A further problem then became clear: If we can't extend the mapfile syntax, then there's no good way to extend ld with an option to produce stub objects, and to validate them against the real objects. The idea of having ld read comments in a mapfile and parse them for content is an unacceptable hack. The entire point of comments is that they are strictly for the human reader, and explicitly ignored by the tool. Taking all of these speed bumps into account, I made a new plan: A perl script reads the mapfiles, generates some small C glue code to produce empty functions and data definitions, compiles and links the stub object from the generated glue code, and then deletes the generated glue code. Another perl script used after both objects have been built, to compare the real and stub objects, using data from elfdump, and validate that they present the same linking interface. By June 2008, I had written the above, and generated a stub object for libc. It was a useful prototype process to go through, and it allowed me to explore the ideas at a deep level. Ultimately though, the result was unsatisfactory as a basis for real product. There were so many issues: The use of stylized comments were fine for a prototype, but not close to professional enough for shipping product. The idea of having to document and support it was a large concern. The ideal solution for stub objects really does involve having the link-editor accept the same arguments used to build the real object, augmented with a single extra command line option. Any other solution, such as our prototype script, will require makefiles to be modified in deeper ways to support building stubs, and so, will raise barriers to converting existing code. A validation script that rederives what the linker knew when it built an object will always be at a disadvantage relative to the actual linker that did the work. A stub object should be identifyable as such. In the prototype, there was no tag or other metadata that would let you know that they weren't real objects. Being able to identify a stub object in this way means that the file command can tell you what it is, and that the runtime linker can refuse to try and run a program that loads one. At that point, we needed to apply this prototype to building Solaris. As you might imagine, the task of modifying all the makefiles in the core Solaris code base in order to do this is a massive task, and not something you'd enter into lightly. The quality of the prototype just wasn't good enough to justify that sort of time commitment, so I tabled the project, putting it on my list of long term things to think about, and moved on to other work. It would sit there for a couple of years. Semi-coincidentally, one of the projects I tacked after that was to create a new mapfile syntax for the Solaris link-editor. We had wanted to do something about the old mapfile syntax for many years. Others before me had done some paper designs, and a great deal of thought had already gone into the features it should, and should not have, but for various reasons things had never moved beyond the idea stage. When I joined Sun in late 2005, I got involved in reviewing those things and thinking about the problem. Now in 2008, fresh from relearning for the Nth time why the old mapfile syntax was a huge impediment to linker progress, it seemed like the right time to tackle the mapfile issue. Paving the way for proper stub object support was not the driving force behind that effort, but I certainly had them in mind as I moved forward. The new mapfile syntax, which we call version 2, integrated into Nevada build snv_135 in in February 2010: 6916788 ld version 2 mapfile syntax PSARC/2009/688 Human readable and extensible ld mapfile syntax In order to prove that the new mapfile syntax was adequate for general purpose use, I had also done an overhaul of the ON consolidation to convert all mapfiles to use the new syntax, and put checks in place that would ensure that no use of the old syntax would creep back in. That work went back into snv_144 in June 2010: 6916796 OSnet mapfiles should use version 2 link-editor syntax That was a big putback, modifying 517 files, adding 18 new files, and removing 110 old ones. I would have done this putback anyway, as the work was already done, and the benefits of human readable syntax are obvious. However, among the justifications listed in CR 6916796 was this We anticipate adding additional features to the new mapfile language that will be applicable to ON, and which will require all sharable object mapfiles to use the new syntax. I never explained what those additional features were, and no one asked. It was premature to say so, but this was a reference to stub objects. By that point, I had already put together a working prototype link-editor with the necessary support for stub objects. I was pleased to find that building stubs was indeed very fast. On my desktop system (Ultra 24), an amd64 stub for libc can can be built in a fraction of a second: % ptime ld -64 -z stub -o stubs/libc.so.1 -G -hlibc.so.1 \ -ztext -zdefs -Bdirect ... real 0.019708910 user 0.010101680 sys 0.008528431 In order to go from prototype to integrated link-editor feature, I knew that I would need to prove that stub objects were valuable. And to do that, I knew that I'd have to switch the Solaris ON consolidation to use stub objects and evaluate the outcome. And in order to do that experiment, ON would first need to be converted to version 2 mapfiles. Sub-mission accomplished. Normally when you design a new feature, you can devise reasonably small tests to show it works, and then deploy it incrementally, letting it prove its value as it goes. The entire point of stub objects however was to demonstrate that they could be successfully applied to an extremely large and complex code base, and specifically to solve the Solaris build issues detailed above. There was no way to finesse the matter — in order to move ahead, I would have to successfully use stub objects to build the entire ON consolidation and demonstrate their value. In software, the need to boil the ocean can often be a warning sign that things are trending in the wrong direction. Conversely, sometimes progress demands that you build something large and new all at once. A big win, or a big loss — sometimes all you can do is try it and see what happens. And so, I spent some time staring at ON makefiles trying to get a handle on how things work, and how they'd have to change. It's a big and messy world, full of complex interactions, unspecified dependencies, special cases, and knowledge of arcane makefile features... ...and so, I backed away, put it down for a few months and did other work... ...until the fall, when I felt like it was time to stop thinking and pondering (some would say stalling) and get on with it. Without stubs, the following gives a simplified high level view of how Solaris is built: An initially empty directory known as the proto, and referenced via the ROOT makefile macro is established to receive the files that make up the Solaris distribution. A top level setup rule creates the proto area, and performs operations needed to initialize the workspace so that the main build operations can be launched, such as copying needed header files into the proto area. Parallel builds are launched to build the kernel (usr/src/uts), libraries (usr/src/lib), and commands. The install makefile target builds each item and delivers a copy to the proto area. All libraries and executables link against the objects previously installed in the proto, implying the need to synchronize the order in which things are built. Subsequent passes run lint, and do packaging. Given this structure, the additions to use stub objects are: A new second proto area is established, known as the stub proto and referenced via the STUBROOT makefile macro. The stub proto has the same structure as the real proto, but is used to hold stub objects. All files in the real proto are delivered as part of the Solaris product. In contrast, the stub proto is used to build the product, and then thrown away. A new target is added to library Makefiles called stub. This rule builds the stub objects. The ld command is designed so that you can build a stub object using the same ld command line you'd use to build the real object, with the addition of a single -z stub option. This means that the makefile rules for building the stub objects are very similar to those used to build the real objects, and many existing makefile definitions can be shared between them. A new target is added to the Makefiles called stubinstall which delivers the stub objects built by the stub rule into the stub proto. These rules reuse much of existing plumbing used by the existing install rule. The setup rule runs stubinstall over the entire lib subtree as part of its initialization. All libraries and executables link against the objects in the stub proto rather than the main proto, and can therefore be built in parallel without any synchronization. There was no small way to try this that would yield meaningful results. I would have to take a leap of faith and edit approximately 1850 makefiles and 300 mapfiles first, trusting that it would all work out. Once the editing was done, I'd type make and see what happened. This took about 6 weeks to do, and there were many dark days when I'd question the entire project, or struggle to understand some of the many twisted and complex situations I'd uncover in the makefiles. I even found a couple of new issues that required changes to the new stub object related code I'd added to ld. With a substantial amount of encouragement and help from some key people in the Solaris group, I eventually got the editing done and stub objects for the entire workspace built. I found that my desktop system could build all the stub objects in the workspace in roughly a minute. This was great news, as it meant that use of the feature is effectively free — no one was likely to notice or care about the cost of building them. After another week of typing make, fixing whatever failed, and doing it again, I succeeded in getting a complete build! The next step was to remove all of the make rules and .WAIT statements dedicated to controlling the order in which libraries under usr/src/lib are built. This came together pretty quickly, and after a few more speed bumps, I had a workspace that built cleanly and looked like something you might actually be able to integrate someday. This was a significant milestone, but there was still much left to do. I turned to doing full nightly builds. Every type of build (open, closed, OpenSolaris, export, domestic) had to be tried. Each type failed in a new and unique way, requiring some thinking and rework. As things came together, I became aware of things that could have been done better, simpler, or cleaner, and those things also required some rethinking, the seeking of wisdom from others, and some rework. After another couple of weeks, it was in close to final form. My focus turned towards the end game and integration. This was a huge workspace, and needed to go back soon, before changes in the gate would made merging increasingly difficult. At this point, I knew that the stub objects had greatly simplified the makefile logic and uncovered a number of race conditions, some of which had been there for years. I assumed that the builds were faster too, so I did some builds intended to quantify the speedup in build time that resulted from this approach. It had never occurred to me that there might not be one. And so, I was very surprised to find that the wall clock build times for a stock ON workspace were essentially identical to the times for my stub library enabled version! This is why it is important to always measure, and not just to assume. One can tell from first principles, based on all those removed dependency rules in the library makefile, that the stub object version of ON gives dmake considerably more opportunities to overlap library construction. Some hypothesis were proposed, and shot down: Could we have disabled dmakes parallel feature? No, a quick check showed things being build in parallel. It was suggested that we might be I/O bound, and so, the threads would be mostly idle. That's a plausible explanation, but system stats didn't really support it. Plus, the timing between the stub and non-stub cases were just too suspiciously identical. Are our machines already handling as much parallelism as they are capable of, and unable to exploit these additional opportunities? Once again, we didn't see the evidence to back this up. Eventually, a more plausible and obvious reason emerged: We build the libraries and commands (usr/src/lib, usr/src/cmd) in parallel with the kernel (usr/src/uts). The kernel is the long leg in that race, and so, wall clock measurements of build time are essentially showing how long it takes to build uts. Although it would have been nice to post a huge speedup immediately, we can take solace in knowing that stub objects simplify the makefiles and reduce the possibility of race conditions. The next step in reducing build time should be to find ways to reduce or overlap the uts part of the builds. When that leg of the build becomes shorter, then the increased parallelism in the libs and commands will pay additional dividends. Until then, we'll just have to settle for simpler and more robust. And so, I integrated the link-editor support for creating stub objects into snv_153 (November 2010) with 6993877 ld should produce stub objects PSARC/2010/397 ELF Stub Objects followed by the work to convert the ON consolidation in snv_161 (February 2011) with 7009826 OSnet should use stub objects 4631488 lib/Makefile is too patient: .WAITs should be reduced This was a huge putback, with 2108 modified files, 8 new files, and 2 removed files. Due to the size, I was allowed a window after snv_160 closed in which to do the putback. It went pretty smoothly for something this big, a few more preexisting race conditions would be discovered and addressed over the next few weeks, and things have been quiet since then. Conclusions and Looking Forward Solaris has been built with stub objects since February. The fact that developers no longer specify the order in which libraries are built has been a big success, and we've eliminated an entire class of build error. That's not to say that there are no build races left in the ON makefiles, but we've taken a substantial bite out of the problem while generally simplifying and improving things. The introduction of a stub proto area has also opened some interesting new possibilities for other build improvements. As this article has become quite long, and as those uses do not involve stub objects, I will defer that discussion to a future article.

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  • Customer Support Spotlight: Clemson University

    - by cwarticki
    I've begun a Customer Support Spotlight series that highlights our wonderful customers and Oracle loyalists.  A week ago I visited Clemson University.  As I travel to visit and educate our customers, I provide many useful tips/tricks and support best practices (as found on my blog and twitter). Most of all, I always discover an Oracle gem who deserves recognition for their hard work and advocacy. Meet George Manley.  George is a Storage Engineer who has worked in Clemson's Data Center all through college, partially in the Hardware Architecture group and partially in the Storage group. George and the rest of the Storage Team work with most all of the storage technologies that they have here at Clemson. This includes a wide array of different vendors' disk arrays, with the most of them being Oracle/Sun 2540's.  He also works with SAM/QFS, ACSLS, and our SL8500 Tape Libraries (all three Oracle/Sun products). (pictured L to R, Matt Schoger (Oracle), Mark Flores (Oracle) and George Manley) George was kind enough to take us for a data center tour.  It was amazing.  I rarely get to see the inside of data centers, and this one was massive. Clemson Computing and Information Technology’s physical resources include the main data center located in the Information Technology Center at the Innovation Campus and Technology Park. The core of Clemson’s computing infrastructure, the data center has 21,000 sq ft of raised floor and is powered by a 14MW substation. The ITC power capacity is 4.5MW.  The data center is the home of both enterprise and HPC systems, and is staffed by CCIT staff on a 24 hour basis from a state of the art network operations center within the ITC. A smaller business continuance data center is located on the main campus.  The data center serves a wide variety of purposes including HPC (supercomputing) resources which are shared with other Universities throughout the state, the state's medicaid processing system, and nearly all other needs for Clemson University. Yes, that's no typo (14,256 cores and 37TB of memory!!! Thanks for the tour George and thank you very much for your time.  The tour was fantastic. I enjoyed getting to know your team and I look forward to many successes from Clemson using Oracle products. -Chris WartickiGlobal Customer Management

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  • A New Threat To Web Applications: Connection String Parameter Pollution (CSPP)

    - by eric.maurice
    Hi, this is Shaomin Wang. I am a security analyst in Oracle's Security Alerts Group. My primary responsibility is to evaluate the security vulnerabilities reported externally by security researchers on Oracle Fusion Middleware and to ensure timely resolution through the Critical Patch Update. Today, I am going to talk about a serious type of attack: Connection String Parameter Pollution (CSPP). Earlier this year, at the Black Hat DC 2010 Conference, two Spanish security researchers, Jose Palazon and Chema Alonso, unveiled a new class of security vulnerabilities, which target insecure dynamic connections between web applications and databases. The attack called Connection String Parameter Pollution (CSPP) exploits specifically the semicolon delimited database connection strings that are constructed dynamically based on the user inputs from web applications. CSPP, if carried out successfully, can be used to steal user identities and hijack web credentials. CSPP is a high risk attack because of the relative ease with which it can be carried out (low access complexity) and the potential results it can have (high impact). In today's blog, we are going to first look at what connection strings are and then review the different ways connection string injections can be leveraged by malicious hackers. We will then discuss how CSPP differs from traditional connection string injection, and the measures organizations can take to prevent this kind of attacks. In web applications, a connection string is a set of values that specifies information to connect to backend data repositories, in most cases, databases. The connection string is passed to a provider or driver to initiate a connection. Vendors or manufacturers write their own providers for different databases. Since there are many different providers and each provider has multiple ways to make a connection, there are many different ways to write a connection string. Here are some examples of connection strings from Oracle Data Provider for .Net/ODP.Net: Oracle Data Provider for .Net / ODP.Net; Manufacturer: Oracle; Type: .NET Framework Class Library: - Using TNS Data Source = orcl; User ID = myUsername; Password = myPassword; - Using integrated security Data Source = orcl; Integrated Security = SSPI; - Using the Easy Connect Naming Method Data Source = username/password@//myserver:1521/my.server.com - Specifying Pooling parameters Data Source=myOracleDB; User Id=myUsername; Password=myPassword; Min Pool Size=10; Connection Lifetime=120; Connection Timeout=60; Incr Pool Size=5; Decr Pool Size=2; There are many variations of the connection strings, but the majority of connection strings are key value pairs delimited by semicolons. Attacks on connection strings are not new (see for example, this SANS White Paper on Securing SQL Connection String). Connection strings are vulnerable to injection attacks when dynamic string concatenation is used to build connection strings based on user input. When the user input is not validated or filtered, and malicious text or characters are not properly escaped, an attacker can potentially access sensitive data or resources. For a number of years now, vendors, including Oracle, have created connection string builder class tools to help developers generate valid connection strings and potentially prevent this kind of vulnerability. Unfortunately, not all application developers use these utilities because they are not aware of the danger posed by this kind of attacks. So how are Connection String parameter Pollution (CSPP) attacks different from traditional Connection String Injection attacks? First, let's look at what parameter pollution attacks are. Parameter pollution is a technique, which typically involves appending repeating parameters to the request strings to attack the receiving end. Much of the public attention around parameter pollution was initiated as a result of a presentation on HTTP Parameter Pollution attacks by Stefano Di Paola and Luca Carettoni delivered at the 2009 Appsec OWASP Conference in Poland. In HTTP Parameter Pollution attacks, an attacker submits additional parameters in HTTP GET/POST to a web application, and if these parameters have the same name as an existing parameter, the web application may react in different ways depends on how the web application and web server deal with multiple parameters with the same name. When applied to connections strings, the rule for the majority of database providers is the "last one wins" algorithm. If a KEYWORD=VALUE pair occurs more than once in the connection string, the value associated with the LAST occurrence is used. This opens the door to some serious attacks. By way of example, in a web application, a user enters username and password; a subsequent connection string is generated to connect to the back end database. Data Source = myDataSource; Initial Catalog = db; Integrated Security = no; User ID = myUsername; Password = XXX; In the password field, if the attacker enters "xxx; Integrated Security = true", the connection string becomes, Data Source = myDataSource; Initial Catalog = db; Integrated Security = no; User ID = myUsername; Password = XXX; Intergrated Security = true; Under the "last one wins" principle, the web application will then try to connect to the database using the operating system account under which the application is running to bypass normal authentication. CSPP poses serious risks for unprepared organizations. It can be particularly dangerous if an Enterprise Systems Management web front-end is compromised, because attackers can then gain access to control panels to configure databases, systems accounts, etc. Fortunately, organizations can take steps to prevent this kind of attacks. CSPP falls into the Injection category of attacks like Cross Site Scripting or SQL Injection, which are made possible when inputs from users are not properly escaped or sanitized. Escaping is a technique used to ensure that characters (mostly from user inputs) are treated as data, not as characters, that is relevant to the interpreter's parser. Software developers need to become aware of the danger of these attacks and learn about the defenses mechanism they need to introduce in their code. As well, software vendors need to provide templates or classes to facilitate coding and eliminate developers' guesswork for protecting against such vulnerabilities. Oracle has introduced the OracleConnectionStringBuilder class in Oracle Data Provider for .NET. Using this class, developers can employ a configuration file to provide the connection string and/or dynamically set the values through key/value pairs. It makes creating connection strings less error-prone and easier to manager, and ultimately using the OracleConnectionStringBuilder class provides better security against injection into connection strings. For More Information: - The OracleConnectionStringBuilder is located at http://download.oracle.com/docs/cd/B28359_01/win.111/b28375/OracleConnectionStringBuilderClass.htm - Oracle has developed a publicly available course on preventing SQL Injections. The Server Technologies Curriculum course "Defending Against SQL Injection Attacks!" is located at http://st-curriculum.oracle.com/tutorial/SQLInjection/index.htm - The OWASP web site also provides a number of useful resources. It is located at http://www.owasp.org/index.php/Main_Page

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  • C# Extension Methods - To Extend or Not To Extend...

    - by James Michael Hare
    I've been thinking a lot about extension methods lately, and I must admit I both love them and hate them. They are a lot like sugar, they taste so nice and sweet, but they'll rot your teeth if you eat them too much.   I can't deny that they aren't useful and very handy. One of the major components of the Shared Component library where I work is a set of useful extension methods. But, I also can't deny that they tend to be overused and abused to willy-nilly extend every living type.   So what constitutes a good extension method? Obviously, you can write an extension method for nearly anything whether it is a good idea or not. Many times, in fact, an idea seems like a good extension method but in retrospect really doesn't fit.   So what's the litmus test? To me, an extension method should be like in the movies when a person runs into their twin, separated at birth. You just know you're related. Obviously, that's hard to quantify, so let's try to put a few rules-of-thumb around them.   A good extension method should:     Apply to any possible instance of the type it extends.     Simplify logic and improve readability/maintainability.     Apply to the most specific type or interface applicable.     Be isolated in a namespace so that it does not pollute IntelliSense.     So let's look at a few examples in relation to these rules.   The first rule, to me, is the most important of all. Once again, it bears repeating, a good extension method should apply to all possible instances of the type it extends. It should feel like the long lost relative that should have been included in the original class but somehow was missing from the family tree.    Take this nifty little int extension, I saw this once in a blog and at first I really thought it was pretty cool, but then I started noticing a code smell I couldn't quite put my finger on. So let's look:       public static class IntExtensinos     {         public static int Seconds(int num)         {             return num * 1000;         }           public static int Minutes(int num)         {             return num * 60000;         }     }     This is so you could do things like:       ...     Thread.Sleep(5.Seconds());     ...     proxy.Timeout = 1.Minutes();     ...     Awww, you say, that's cute! Well, that's the problem, it's kitschy and it doesn't always apply (and incidentally you could achieve the same thing with TimeStamp.FromSeconds(5)). It's syntactical candy that looks cool, but tends to rot and pollute the code. It would allow things like:       total += numberOfTodaysOrders.Seconds();     which makes no sense and should never be allowed. The problem is you're applying an extension method to a logical domain, not a type domain. That is, the extension method Seconds() doesn't really apply to ALL ints, it applies to ints that are representative of time that you want to convert to milliseconds.    Do you see what I mean? The two problems, in a nutshell, are that a) Seconds() called off a non-time value makes no sense and b) calling Seconds() off something to pass to something that does not take milliseconds will be off by a factor of 1000 or worse.   Thus, in my mind, you should only ever have an extension method that applies to the whole domain of that type.   For example, this is one of my personal favorites:       public static bool IsBetween<T>(this T value, T low, T high)         where T : IComparable<T>     {         return value.CompareTo(low) >= 0 && value.CompareTo(high) <= 0;     }   This allows you to check if any IComparable<T> is within an upper and lower bound. Think of how many times you type something like:       if (response.Employee.Address.YearsAt >= 2         && response.Employee.Address.YearsAt <= 10)     {     ...     }     Now, you can instead type:       if(response.Employee.Address.YearsAt.IsBetween(2, 10))     {     ...     }     Note that this applies to all IComparable<T> -- that's ints, chars, strings, DateTime, etc -- and does not depend on any logical domain. In addition, it satisfies the second point and actually makes the code more readable and maintainable.   Let's look at the third point. In it we said that an extension method should fit the most specific interface or type possible. Now, I'm not saying if you have something that applies to enumerables, you create an extension for List, Array, Dictionary, etc (though you may have reasons for doing so), but that you should beware of making things TOO general.   For example, let's say we had an extension method like this:       public static T ConvertTo<T>(this object value)     {         return (T)Convert.ChangeType(value, typeof(T));     }         This lets you do more fluent conversions like:       double d = "5.0".ConvertTo<double>();     However, if you dig into Reflector (LOVE that tool) you will see that if the type you are calling on does not implement IConvertible, what you convert to MUST be the exact type or it will throw an InvalidCastException. Now this may or may not be what you want in this situation, and I leave that up to you. Things like this would fail:       object value = new Employee();     ...     // class cast exception because typeof(IEmployee) != typeof(Employee)     IEmployee emp = value.ConvertTo<IEmployee>();       Yes, that's a downfall of working with Convertible in general, but if you wanted your fluent interface to be more type-safe so that ConvertTo were only callable on IConvertibles (and let casting be a manual task), you could easily make it:         public static T ConvertTo<T>(this IConvertible value)     {         return (T)Convert.ChangeType(value, typeof(T));     }         This is what I mean by choosing the best type to extend. Consider that if we used the previous (object) version, every time we typed a dot ('.') on an instance we'd pull up ConvertTo() whether it was applicable or not. By filtering our extension method down to only valid types (those that implement IConvertible) we greatly reduce our IntelliSense pollution and apply a good level of compile-time correctness.   Now my fourth rule is just my general rule-of-thumb. Obviously, you can make extension methods as in-your-face as you want. I included all mine in my work libraries in its own sub-namespace, something akin to:       namespace Shared.Core.Extensions { ... }     This is in a library called Shared.Core, so just referencing the Core library doesn't pollute your IntelliSense, you have to actually do a using on Shared.Core.Extensions to bring the methods in. This is very similar to the way Microsoft puts its extension methods in System.Linq. This way, if you want 'em, you use the appropriate namespace. If you don't want 'em, they won't pollute your namespace.   To really make this work, however, that namespace should only include extension methods and subordinate types those extensions themselves may use. If you plant other useful classes in those namespaces, once a user includes it, they get all the extensions too.   Also, just as a personal preference, extension methods that aren't simply syntactical shortcuts, I like to put in a static utility class and then have extension methods for syntactical candy. For instance, I think it imaginable that any object could be converted to XML:       namespace Shared.Core     {         // A collection of XML Utility classes         public static class XmlUtility         {             ...             // Serialize an object into an xml string             public static string ToXml(object input)             {                 var xs = new XmlSerializer(input.GetType());                   // use new UTF8Encoding here, not Encoding.UTF8. The later includes                 // the BOM which screws up subsequent reads, the former does not.                 using (var memoryStream = new MemoryStream())                 using (var xmlTextWriter = new XmlTextWriter(memoryStream, new UTF8Encoding()))                 {                     xs.Serialize(xmlTextWriter, input);                     return Encoding.UTF8.GetString(memoryStream.ToArray());                 }             }             ...         }     }   I also wanted to be able to call this from an object like:       value.ToXml();     But here's the problem, if i made this an extension method from the start with that one little keyword "this", it would pop into IntelliSense for all objects which could be very polluting. Instead, I put the logic into a utility class so that users have the choice of whether or not they want to use it as just a class and not pollute IntelliSense, then in my extensions namespace, I add the syntactical candy:       namespace Shared.Core.Extensions     {         public static class XmlExtensions         {             public static string ToXml(this object value)             {                 return XmlUtility.ToXml(value);             }         }     }   So now it's the best of both worlds. On one hand, they can use the utility class if they don't want to pollute IntelliSense, and on the other hand they can include the Extensions namespace and use as an extension if they want. The neat thing is it also adheres to the Single Responsibility Principle. The XmlUtility is responsible for converting objects to XML, and the XmlExtensions is responsible for extending object's interface for ToXml().

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  • Configuring Multi-Tap on Synaptics Touchpad

    - by nunos
    I am having a hard time configuring my notebook's touchpad. The touchpad already works. It successfully responds to one-finger tap, two-finger tap and two-finger vertical scrolling. What I want to accomplish: change two-finger tap action from right-mouse click to middle-mouse click add three-finger tap functionality to yield right-mouse click action (i have checked that the three-finger tap is supported by my laptop's touchpad since it works on Windows) I read on a forum to use this as a guide. I have successfully accomplished point 1 with synclient TapButton2=2. However, I have to do it everytime I log in. I have tried to put that command on /etc/rc.local but the computer always boots and logins with the default configuration. Regarding point 2, I have tried synclient TapButton3=3 but it doesn't do anything when I three-finger tap the touchpad. I am running Ubuntu 11.10 on an Asus N82JV. /etc/X11/xorg.conf: nuno@mozart:~$ cat /etc/X11/xorg.conf Section "InputClass" Identifier "touchpad catchall" Driver "synaptics" MatchIsTouchpad "on" MatchDevicePath "/dev/input/event*" Option "TapButton1" "1" Option "TapButton2" "2" Option "TapButton3" "3" EndSection /usr/share/X11/xorg.conf.d/50-synaptics.conf: nuno@mozart:~$ cat /usr/share/X11/xorg.conf.d/50-synaptics.conf # Example xorg.conf.d snippet that assigns the touchpad driver # to all touchpads. See xorg.conf.d(5) for more information on # InputClass. # DO NOT EDIT THIS FILE, your distribution will likely overwrite # it when updating. Copy (and rename) this file into # /etc/X11/xorg.conf.d first. # Additional options may be added in the form of # Option "OptionName" "value" # Section "InputClass" Identifier "touchpad catchall" Driver "synaptics" MatchIsTouchpad "on" MatchDevicePath "/dev/input/event*" Option "TapButton1" "1" Option "TapButton2" "2" Option "TapButton3" "3" EndSection xinput list: nuno@mozart:~$ xinput list ? Virtual core pointer id=2 [master pointer (3)] ? ? Virtual core XTEST pointer id=4 [slave pointer (2)] ? ? Microsoft Microsoft® Nano Transceiver v2.0 id=12 [slave pointer (2)] ? ? Microsoft Microsoft® Nano Transceiver v2.0 id=13 [slave pointer (2)] ? ? ETPS/2 Elantech Touchpad id=16 [slave pointer (2)] ? Virtual core keyboard id=3 [master keyboard (2)] ? Virtual core XTEST keyboard id=5 [slave keyboard (3)] ? Power Button id=6 [slave keyboard (3)] ? Video Bus id=7 [slave keyboard (3)] ? Video Bus id=8 [slave keyboard (3)] ? Sleep Button id=9 [slave keyboard (3)] ? USB2.0 2.0M UVC WebCam id=10 [slave keyboard (3)] ? Microsoft Microsoft® Nano Transceiver v2.0 id=11 [slave keyboard (3)] ? Asus Laptop extra buttons id=14 [slave keyboard (3)] ? AT Translated Set 2 keyboard id=15 [slave keyboard (3)]

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  • I have an apache process that takes 98% CPU. How can I find what apache call it runs?

    - by Nir
    As you can see below, a single Apache process hangs and takes large amount of CPU resources. How can I find what http call this apache process runs? PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 12554 www-data 20 0 776m 285m 199m R 97 3.7 67:15.84 apache2 14580 www-data 20 0 748m 372m 314m S 4 4.8 0:13.60 apache2 12561 www-data 20 0 784m 416m 322m S 3 5.4 0:58.10 apache2 12592 www-data 20 0 785m 427m 332m S 2 5.6 0:57.06 apache2

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