Search icon CANCEL
Subscription
0
Cart icon
Your Cart (0 item)
Close icon
You have no products in your basket yet
Arrow left icon
Explore Products
Best Sellers
New Releases
Books
Events
Videos
Audiobooks
Packt Hub
Free Learning
Arrow right icon
timer SALE ENDS IN
0 Days
:
00 Hours
:
00 Minutes
:
00 Seconds

How-To Tutorials

7019 Articles
article-image-using-indexes-manipulate-pandas-objects
Packt
05 Sep 2013
4 min read
Save for later

Using indexes to manipulate pandas objects

Packt
05 Sep 2013
4 min read
(For more resources related to this topic, see here.) Getting ready A good understanding of indexes in pandas is crucial to quickly move the data around. From a business intelligence perspective, they create a distinction similar to that of metrics and dimensions in an OLAP cube. To illustrate this point, this recipe walks through getting stock data out of pandas, combining it, then reindexing it for easy chomping. How to do it... Use the DataReader object to transfer stock price information into a DataFrame and to explore the basic axis of Panel. > from pandas.i git push -u origin master o.data import DataReader > tickers = ['gs', 'ibm', 'f', 'ba', 'axp'] > dfs = {} > for ticker in tickers: dfs[ticker] = DataReader(ticker, "yahoo", '2006-01-01') # a yet undiscussed data structure, in the same way the a # DataFrame is a collection of Series, a Panel is a collection of # DataFrames > pan = pd.Panel(dfs) > pan <class 'pandas.core.panel.Panel'> Dimensions: 5 (items) x 1764 (major_axis) x 6 (minor_axis)Items axis: axp to ibm Major_axis axis: 2006-01-03 00:00:00 to 2013-01-04 00:00:00 Minor_axis axis: Open to Adj Close > pan.items Index([axp, ba, f, gs, ibm], dtype=object) > pan.minor_axis Index([Open, High, Low, Close, Volume, Adj Close], dtype=object) > pan.major_axis <class 'pandas.tseries.index.DatetimeIndex'>[2006-01-03 00:00:00, ..., 2013-01-04 00:00:00] Length: 1764, Freq: None, Timezone: None Use the axis selectors to easily compute different sets of summary statistics. > pan.minor_xs('Open').mean() axp 46.227466 ba 70.746451 f 9.135794 gs 151.655091 ibm 129.570969 # major axis is sliceable as well > day_slice = pan.major_axis[1] > pan.major_xs(day_slice)[['gs', 'ba']] ba gs Open 70.08 127.35 High 71.27 128.91 Low 69.86 126.38 Close 71.17 127.09 Volume 3165000.00 4861600.00 Adj Close 60.43 118.12 Convert the Panel to a DataFrame. > dfs = [] > for df in pan: idx = pan.major_axis idx = pd.MultiIndex.from_tuples(zip([df]*len(idx), idx)) idx.names = ['ticker', 'timestamp'] dfs.append(pd.DataFrame(pan[df].values, index=idx, columns=pan.minor_axis)) > df = pd.concat(dfs) > df Data columns: Open 8820 non-null values High 8820 non-null values Low 8820 non-null values Close 8820 non-null values Volume 8820 non-null values Adj Close 8820 non-null values dtypes: float64(6) Perform the analogous operations as in the preceding examples on the newly created DataFrame. # selecting from a MultiIndex isn't much different than the Panel # (output muted) > df.ix['gs':'ibm'] > df['Open'] How it works... The previous example was certainly contrived, but when indexing and statistical techniques are incorporated, the power of pandas begins to come through. Statistics will be covered in an upcoming recipe. pandas' indexes by themselves can be thought of as descriptors of a certain point in the DataFrame. When ticker and timestamp are the only indexes in a DataFrame, then the point is individualized by the ticker, timestamp, and column name. After the point is individualized, it's more convenient for aggregation and analysis. There's more... Indexes show up all over the place in pandas so it's worthwhile to see some other use cases as well. Advanced header indexes Hierarchical indexing isn't limited to rows. Headers can also be represented by MultiIndex, as shown in the following command line: > header_top = ['Price', 'Price', 'Price', 'Price', 'Volume', 'Price'] > df.columns = pd.MultiIndex.from_tuples(zip(header_top, df.columns) Performing aggregate operations with indexes As a prelude to the following sections, we'll do a single groupby function here since they work with indexes so well. > df.groupby(level=['tickers', 'day'])['Volume'].mean() This answers the question for each ticker and for each day (not date), that is, what was the mean volume over the life of the data. Summary This article talks about the use and importance of indexes in pandas. It also talks about different operations that can be done with indexes. Resources for Article : Further resources on this subject: Installing Panda3D [Article] Setting Up Panda3D and Configuring Development Tools [Article] Collision Detection and Physics in Panda3D Game Development [Article]
Read more
  • 0
  • 0
  • 3109

article-image-introduction-bluestacks
Packt
05 Sep 2013
4 min read
Save for later

Introduction to BlueStacks

Packt
05 Sep 2013
4 min read
(For more resources related to this topic, see here.) So, what is BlueStacks? BlueStacks is a suite of tools designed to allow you to run Android apps easily on a Windows or Mac computer. The following screenshot shows how it looks: At the time of writing, there are two elements to the BlueStacks suite, which are listed as follows: App Player: This is the engine, which runs the Android apps Cloud Connect: This is a synchronization tool As the BlueStacks tools can be freely downloaded, anyone with a PC running on Windows or Mac can download them and start experimenting with their capabilities. This article will walk you through the process of running BlueStacks on a computer and show you some of the ways in which you can make the most out of this emerging technology. There are other ways by which you can run an emulation of Android on your computer. You can, for instance, run a virtual machine or install the Android Software Development Kit (SDK). These assume a degree of technical understanding that isn't necessarily required with BlueStacks, making BlueStacks the quickest and easiest way of running apps on your computer. BlueStacks is particularly interesting for users of Windows 8 tablets, as it opens up a whole library of mature software designed for a touch interface. This is particularly useful for those wanting to use many, free, or cheap Android apps on their laptop or tablet. It is worth noting that, at the time of writing this article, these tools are beta releases, so it is important that you take time to report the bugs that you may find to the developers through their website. The ongoing development and success of the software depends upon this feedback and results in a better product. If you become reliant on a particular feature, it is a good idea to highlight your love to the developers too. This can help influence which features are to be kept and improved upon as the product matures. App Player BlueStacks App Player allows a Windows or Mac user to run Android apps on their desktop or laptop. It does this by running an emulated version of Android within a window that you can interact with using your keyboard and mouse. The App Player can be downloaded and installed for free from the BlueStacks website, http://www.bluestacks.com. Currently, there are two main versions available for different operating systems that are enlisted as follows: Mac OS X Windows XP, Vista, 7, and 8 Once you have installed the software, an Android emulator runs on your machine. This is a light version of Android that can access app stores so that you can download and run free and paid apps and content. Most apps are compatible with App Player; however, there are some which are not (for technical reasons) and some which have been prevented by the App developers from running. If you are running any another operating system on your computer, the more computing power you can make available to the App Player the better. Otherwise, you might experience slow loading apps or worse still ones that do not function properly. To increase your chances of success, first try running App Player without running any other applications (for example, Word). Cloud Connect Cloud Connect provides a means to synchronize the apps running on an existing phone or tablet with the App Player. This means that you do not have to manually install lots of apps. Instead, you install an app on your device and sign up so that your App Player has exactly the same apps as your device. Summary Thus we learned the basics of BlueStacks and saw a brief of App Player and Cloud Connect feature of BlueStacks Resources for Article: Further resources on this subject: So, what is Spring for Android? [Article] Animating Properties and Tweening Pages in Android 3-0 [Article] New Connectivity APIs – Android Beam [Article]
Read more
  • 0
  • 0
  • 19239

article-image-using-linq-query-linqpad
Packt
05 Sep 2013
3 min read
Save for later

Using a LINQ query in LINQPad

Packt
05 Sep 2013
3 min read
(For more resources related to this topic, see here.) The standard version We are going to implement a simple scenario: given a deck of 52 cards, we want to pick a random number of cards, and then take out all of the hearts. From this stack of hearts, we will discard the first two and take the next five cards (if possible), and order them by their face value for display. You can try it in a C# program query in LINQPad: public static Random random = new Random();void Main(){ var deck = CreateDeck(); var randomCount = random.Next(52); var hearts = new Card[randomCount]; var j = 0; // take all hearts out for(var i=0;i<randomCount;i++) { if(deck[i].Suit == "Hearts") { hearts[j++] = deck[i]; } } // resize the array to avoid null references Array.Resize(ref hearts, j); // check that we have at least 2 cards. If not, stop if(hearts.Length <= 2) return; var count = 0; // check how many cards we can take count = hearts.Length - 2; // the most we need to take is 5 if(count > 5) { count = 5; } // take the cards var finalDeck = new Card[count]; Array.Copy(hearts, 2, finalDeck, 0, count); // now order the cards Array.Sort(finalDeck, new CardComparer()); // Display the result finalDeck.Dump();}public class Card{ public string Suit { get; set; } public int Value { get; set; }}// Create the cards' deckpublic Card[] CreateDeck(){ var suits = new [] { "Spades", "Clubs", "Hearts", "Diamonds" }; var deck = new Card[52]; for(var i = 0; i < 52; i++) { deck[i] = new Card { Suit = suits[i / 13], FaceValue = i-(13*(i/13))+1 }; } // randomly shuffle the deck for (var i = deck.Length - 1; i > 0; i--) { var j = random.Next(i + 1); var tmp = deck[j]; deck[j] = deck[i]; deck[i] = tmp; } return deck;}// CardComparer compare 2 cards against their face valuepublic class CardComparer : Comparer<Card>{ public override int Compare(Card x, Card y) { return x.FaceValue.CompareTo(y.FaceValue); }} Even if we didn't consider the CreateDeck() method, we had to do quite a few operations to produce the expected result (your values might be different as we are using random cards). The output is as follows: Depending on the data, LINQPad will add contextual information. For example, in this sample it will add the bottom row with the sum of all the values (here, only FaceValue). Also, if you click on the horizontal graph button, you will get a visual representation of your data, as shown in the following screenshot: This information is not always relevant but it can help you explore your data. Summary In this article we saw how LINQ queries can be used in LINQPad. The powerful query capabilitiesof LINQ has been utilized to the maximum in LINQPad. Resources for Article: Further resources on this subject: Displaying SQL Server Data using a Linq Data Source [Article] Binding MS Chart Control to LINQ Data Source Control [Article] LINQ to Objects [Article]
Read more
  • 0
  • 0
  • 2678

article-image-chef-infrastructure
Packt
05 Sep 2013
10 min read
Save for later

Chef Infrastructure

Packt
05 Sep 2013
10 min read
(For more resources related to this topic, see here.) First, let's talk about the terminology used in the Chef universe. A cookbook is a collection of recipes – codifying the actual resources, which should be installed and configured on your node – and the files and configuration templates needed. Once you've written your cookbooks, you need a way to deploy them to the nodes you want to provision. Chef offers multiple ways for this task. The most widely used way is to use a central Chef Server. You can either run your own or sign up for Opscode's Hosted Chef. The Chef Server is the central registry where each node needs to get registered. The Chef Server distributes the cookbooks to the nodes based on their configuration settings. Knife is Chef's command-line tool called to interact with the Chef Server. You use it for uploading cookbooks and managing other aspects of Chef. On your nodes, you need to install Chef Client – the part that retrieves the cookbooks from the Chef Server and executes them on the node. In this article, we'll see the basic infrastructure components of your Chef setup at work and learn how to use the basic tools. Let's get started with having a look at how to use Git as a version control system for your cookbooks. Using version control Do you manually back up every file before you change it? And do you invent creative filename extensions like _me and _you when you try to collaborate on a file? If you answer yes to any of the preceding questions, it's time to rethink your process. A version control system (VCS) helps you stay sane when dealing with important files and collaborating on them. Using version control is a fundamental part of any infrastructure automation. There are multiple solutions (some free, some paid) for managing source version control including Git, SVN, Mercurial, and Perforce. Due to its popularity among the Chef community, we will be using Git. However, you could easily use any other version control system with Chef. Getting ready You'll need Git installed on your box. Either use your operating system's package manager (such as Apt on Ubuntu or Homebrew on OS X), or simply download the installer from www.git-scm.org. Git is a distributed version control system. This means that you don't necessarily need a central host for storing your repositories. But in practice, using GitHub as your central repository has proven to be very helpful. In this article, I'll assume that you're using GitHub. Therefore, you need to go to github.com and create a (free) account to follow the instructions given in this article. Make sure that you upload your SSH key following the instructions at https://help.github.com/articles/generating-ssh-keys, so that you're able to use the SSH protocol to interact with your GitHub account. As soon as you've created your GitHub account, you should create your repository by visiting https://github.com/new and using chef-repo as the repository name. How to do it... Before you can write any cookbooks, you need to set up your initial Git repository on your development box. Opscode provides an empty Chef repository to get you started. Let's see how you can set up your own Chef repository with Git using Opscode's skeleton. Download Opscode's skeleton Chef repository as a tarball: mma@laptop $ wget http://github.com/opscode/chef-repo/tarball/master...TRUNCATED OUTPUT...2013-07-05 20:54:24 (125 MB/s) - 'master' saved [9302/9302] Extract the downloaded tarball: mma@laptop $ tar zvf master Rename the directory. Replace 2c42c6a with whatever your downloaded tarball contained in its name: mma@laptop $ mv opscode-chef-repo-2c42c6a/ chef-repo Change into your newly created Chef repository: mma@laptop $ cd chef-repo/ Initialize a fresh Git repository: mma@laptop:~/chef-repo $ git init .Initialized empty Git repository in /Users/mma/work/chef-repo/.git/ Connect your local repository to your remote repository on github.com. Make sure to replace mmarschall with your own GitHub username: mma@laptop:~/chef-repo $ git remote add origin git@github.com:mmarschall/chef-repo.git Add and commit Opscode's default directory structure: mma@laptop:~/chef-repo $ git add .mma@laptop:~/chef-repo $ git commit -m "initial commit"[master (root-commit) 6148b20] initial commit10 files changed, 339 insertions(+), 0 deletions(-)create mode 100644 .gitignore...TRUNCATED OUTPUT...create mode 100644 roles/README.md Push your initialized repository to GitHub. This makes it available to all your co-workers to collaborate on it. mma@laptop:~/chef-repo $ git push -u origin master...TRUNCATED OUTPUT...To git@github.com:mmarschall/chef-repo.git* [new branch] master -> master How it works... You've downloaded a tarball containing Opscode's skeleton repository. Then, you've initialized your chef-repo and connected it to your own repository on GitHub. After that, you've added all the files from the tarball to your repository and committed them. This makes Git track your files and the changes you make later. As a last step, you've pushed your repository to GitHub, so that your co-workers can use your code too. There's more... Let's assume you're working on the same chef-repo repository together with your co-workers. They cloned your repository, added a new cookbook called other_cookbook, committed their changes locally, and pushed their changes back to GitHub. Now it's time for you to get the new cookbook down to your own laptop. Pull your co-workers, changes from GitHub. This will merge their changes into your local copy of the repository. mma@laptop:~/chef-repo $ git pull From github.com:mmarschall/chef-repo * branch master -> FETCH_HEAD ...TRUNCATED OUTPUT... create mode 100644 cookbooks/other_cookbook/recipes/default.rb In the case of any conflicting changes, Git will help you merge and resolve them. Installing Chef on your workstation If you want to use Chef, you'll need to install it on your local workstation first. You'll have to develop your configurations locally and use Chef to distribute them to your Chef Server. Opscode provides a fully packaged version, which does not have any external prerequisites. This fully packaged Chef is called the Omnibus Installer. We'll see how to use it in this section. Getting ready Make sure you've curl installed on your box by following the instructions available at http://curl.haxx.se/download.html. How to do it... Let's see how to install Chef on your local workstation using Opscode's Omnibus Chef installer: In your local shell, run the following command: mma@laptop:~/chef-repo $ curl -L https://www.opscode.com/chef/install.sh | sudo bashDownloading Chef......TRUNCATED OUTPUT...Thank you for installing Chef! Add the newly installed Ruby to your path: mma@laptop:~ $ echo 'export PATH="/opt/chef/embedded/bin:$PATH"'>> ~/.bash_profile && source ~/.bash_profile How it works... The Omnibus Installer will download Ruby and all the required Ruby gems into /opt/chef/embedded. By adding the /opt/chef/embedded/bin directory to your .bash_profile, the Chef command-line tools will be available in your shell. There's more... If you already have Ruby installed in your box, you can simply install the Chef Ruby gem by running mma@laptop:~ $ gem install chef. Using the Hosted Chef platform If you want to get started with Chef right away (without the need to install your own Chef Server) or want a third party to give you an Service Level Agreement (SLA) for your Chef Server, you can sign up for Hosted Chef by Opscode. Opscode operates Chef as a cloud service. It's quick to set up and gives you full control, using users and groups to control the access permissions to your Chef setup. We'll configure Knife, Chef's command-line tool to interact with Hosted Chef, so that you can start managing your nodes. Getting ready Before being able to use Hosted Chef, you need to sign up for the service. There is a free account for up to five nodes. Visit http://www.opscode.com/hosted-chef and register for a free trial or the free account. I registered as the user webops with an organization short-name of awo. After registering your account, it is time to prepare your organization to be used with your chef-repo repository. How to do it... Carry out the following steps to interact with the Hosted Chef: Navigate to http://manage.opscode.com/organizations. After logging in, you can start downloading your validation keys and configuration file. Select your organization to be able to see its contents using the web UI. Regenerate the validation key for your organization and save it as <your-organization-short-name>.pem in the .chef directory inside your chef-repo repository. Generate the Knife config and put the downloaded knife.rb into the .chef directory inside your chef-repo directory as well. Make sure you replace webops with the username you chose for Hosted Chef and awo with the short-name you chose for your organization: current_dir = File.dirname(__FILE__)log_level :infolog_location STDOUTnode_name "webops"client_key "#{current_dir}/webops.pem"validation_client_name "awo-validator"validation_key "#{current_dir}/awo-validator.pem"chef_server_url "https://api.opscode.com/organizations/awo"cache_type 'BasicFile'cache_options( :path => "#{ENV['HOME']}/.chef/checksums" )cookbook_path ["#{current_dir}/../cookbooks"] Use Knife to verify that you can connect to your hosted Chef organization. It should only have your validator client so far. Instead of awo, you'll see your organization's short-name: mma@laptop:~/chef-repo $ knife client listawo-validator How it works... Hosted Chef uses two private keys (called validators): one for the organization and the other for every user. You need to tell Knife where it can find these two keys in your knife.rb file. The following two lines of code in your knife.rb file tells Knife about which organization to use and where to find its private key: validation_client_name "awo-validator"validation_key "#{current_dir}/awo-validator.pem" The following line of code in your knife.rb file tells Knife about where to find your users' private key: client_key "#{current_dir}/webops.pem" And the following line of code in your knife.rb file tells Knife that you're using Hosted Chef. You will find your organization name as the last part of the URL: chef_server_url "https://api.opscode.com/organizations/awo" Using the knife.rb file and your two validators Knife can now connect to your organization hosted by Opscode. You do not need your own, self-hosted Chef Server, nor do you need to use Chef Solo in this setup. There's more... This setup is good for you if you do not want to worry about running, scaling, and updating your own Chef Server and if you're happy with saving all your configuration data in the cloud (under Opscode's control). If you need to have all your configuration data within your own network boundaries, you might sign up for Private Chef, which is a fully supported and enterprise-ready version of Chef Server. If you don't need any advanced enterprise features like role-based access control or multi-tenancy, then the open source version of Chef Server might be just right for you. Summary In this article, we learned about key concepts such as cookbooks, roles, and environments and how to use some basic tools such as Git, Knife, Chef Shell, Vagrant, and Berkshelf. Resources for Article: Further resources on this subject: Automating the Audio Parameters – How it Works [Article] Skype automation [Article] Cross-browser-distributed testing [Article]
Read more
  • 0
  • 0
  • 2688

article-image-setting-single-width-column-system-simple
Packt
05 Sep 2013
3 min read
Save for later

Setting up a single-width column system (Simple)

Packt
05 Sep 2013
3 min read
(For more resources related to this topic, see here.) Getting ready To perform the steps listed in this article, we will need a text editor, a browser, and a copy of the Masonry plugin. Any text editor will do, but my browser of choice is Google Chrome, as the V8 JavaScript engine that ships with it generally performs better and supports CSS3 transitions, and as a result we see smoother animations when resizing the browser window. We need to make sure we have a copy of the most recent version of Masonry, which was Version 2.1.08 at the time of writing this article. This version is compatible with the most recent version of jQuery, which is Version 1.9.1. A production copy of Masonry can be found on the GitHub repository at the following address: https://github.com/desandro/masonry/blob/master/jquery.masonry.min.js For jQuery, we will be using a content delivery network (CDN) for ease of development. Open the basic single-column HTML file to follow along. You can download this file from the following location: http://www.packtpub.com/sites/default/files/downloads/1-single-column.zip How to do it... Set up the styling for the masonry-item class with the proper width, padding, and margins. We want our items to have a total width of 200 pixels, including the padding and margins. <style> .masonry-item { background: #FFA500; float: left; margin: 5px; padding: 5px; width: 180px; }</style> Set up the HTML structure on which you are going to use Masonry. At a minimum, we need a tagged Masonry container with the elements inside tagged as Masonry items. <div id='masonry-container'> <div class='masonry-item '> Maecenas faucibus mollis interdum. </div> <div class='masonry-item '> Maecenas faucibus mollis interdum. Donec sed odio dui. Nullamquis risus eget urna mollis ornare vel eu leo. Vestibulum idligula porta felis euismod semper. </div> <div class='masonry-item '> Nullam quis risus eget urna mollis ornare vel eu leo. Crasjusto odio, dapibus ac facilisis in, egestas eget quam. Aeneaneu leo quam. Pellentesque ornare sem lacinia quam venenatisvestibulum. </div></div> All Masonry options need not be included, but it is recommended (by David DeSandro, the creator of Masonry) to set itemSelector for single-column usage. We will be setting this every time we use Masonry. <script> $(function() { $('#masonry-container').masonry({ // options itemSelector : '.masonry-item', }); });</script> How it works... Using jQuery, we select our Masonry container and use the itemSelector option to select the elements that will be affected by Masonry. The size of the columns will be determined by the CSS code. Using the box model, we set our Masonry items to a width of 90 px (80-px wide, with a 5-px padding all around the item). The margin is our gutter between elements, which is also 5-px wide. With this setup, we can con firm that we have built the basic single-column grid system, with each column being 100-px wide. The end result should look like the following screenshot: Summary This article showed you how to set up the very basic Masonry single-width column system around which Masonry revolves. Resources for Article : Further resources on this subject: Designing Site Layouts in Inkscape [Article] New features in Domino Designer 8.5 [Article] Using jQuery and jQueryUI Widget Factory plugins with RequireJS [Article]
Read more
  • 0
  • 0
  • 1852

article-image-cocos2d-x-installation
Packt
05 Sep 2013
10 min read
Save for later

Cocos2d-x: Installation

Packt
05 Sep 2013
10 min read
(For more resources related to this topic, see here.) Download and installation All the examples in this article were developed on a Mac using Xcode. Although you can use Cocos2d-x to develop your games for other platforms, using different systems, the examples will focus on iOS and Mac. Xcode is free and can be downloaded from the Mac App store (https://developer.apple.com/xcode/index.php), but in order to test your code on an iOS device and publish your games, you will need a developer account with Apple, which will cost you USD 99 a year. You can find more information on their website: https://developer.apple.com/ So, assuming you have an internet connection, and that Xcode is ready to rock, let's begin! Time for action – downloading and installing Cocos2d-x We start by downloading the framework: Go to http://download.cocos2d-x.org/ and download the latest stable version of Cocos2d-x. For this article I'll be using version Cocos2d-2.0-x-2.0.4, which means the 2.0.4 C++ port of version 2.0 of Cocos2d. Uncompress the files somewhere on your machine. Open Terminal and type cd (that is cd and a space). Drag the uncompressed folder you just downloaded to the Terminal window. You should see the path to the folder added to the command line. Hit returnto go to that folder in Terminal. Now type: sudo ./install-templates-xcode.sh -u Hit return again and you're done. What just happened? You have successfully installed the Cocos2d-x templates in your machine. With these in place, you can select the type of Cocos2d-x application you wish to build inside Xcode, and the templates will take care of copying all the necessary files into your application. Next, open Xcode and select Create a new Xcode Project.You should see something like this: So let's build our first application. Hello-x World-x Let's create that old chestnut in computer programming: the hello world example. Time for action – creating an application Open Xcode and select File | New | Project... and follow these steps: In the dialogue box select cocos2d-x under the iOS menu and choose the cocos2dx template. Hit Next . Give the application a name, but not HelloWorld. I'll show you why in a second. You will be then asked to select a place to save the project and you are done. Once your application is ready, click Run to build it. After that, this is what you should see in the simulator: When you run a cocos2d-x application in Xcode it is quite common for the program to post some warnings regarding your code, or most likely the frameworks. These will mostly reference deprecated methods, or statements that do not precisely follow more recent, and stricter rules of the current SDK. But that's okay. These warnings, though certainly annoying, can be ignored. What just happened? You created your first Cocos2d-x application using the cocos2dx template, sometimes referred to as the basic template. The other template options include one with Box2D, one with Chipmunk (both related to physics simulation), one with JavaScript, and one with Lua. The last two options allow you to code some or all of your game using those script languages instead of the native C++; and they work just as you would expect a scripting language to work, meaning the commands written in either Javascript or Lua are actually replaced and interpreted as C++ commands by the compiler. Now if you look at the files created by the basic template you will see a HelloWorldScene class file. That's the reason I didn't want you to call your application HelloWorld, because I didn't want you to have the impression that the file name was based on your project name. It isn't. You will always get a HelloWorldScene file unless you change the template itself. Now let's go over the sample application and its files: The folder structure First you have the Resources folder, where you find the images used by the application. The ios folder has the necessary underlying connections between your app and iOS. For other platforms, you will have their necessary linkage files in separate folders targeting their respective platform (like an android folder the Android platform, for instance.) In the libs folder you have all the cocos2dx files, plus CocosDenshion files (for sound support) and a bunch of other extensions. Using a different template for your projects will result in a different folder structure here, based on what needs to be added to your project. So you will see a Box2D folder, for example, if you choose the Box2D template. In the Classes folder you have your application. In here, everything is written in C++ and this is the home for the part of your code that will hopefully not need to change, however many platforms you target with your application. Now let us go over the main classes of the basic application. The iOS linkage classes AppController and RootViewController are responsible for setting up OpenGL in iOS as well as telling the underlying operating system that your application is about to say Hello... To the World. These classes are written with a mix of Objective-C and C++, as all the nice brackets and the .mm extensions show. You will change very little if anything in these classes; and again that will reflect in changes to the way iOS handles your application. So other targets would require the same instructions or none at all depending on the target. In AppController for instance, I could add support for multitouch. And in RootViewController, I could limit the screen orientations supported by my application. The AppDelegate class This class marks the first time your C++ app will talk to the underlying OS. It attempts to map the main events that mobile devices wants to dispatch and listen to. From here on, all your application will be written in C++ (unless you need something else). In AppDelegate you should setup CCDirector (the cocos2d-x all powerful singleton manager object) to run your application just the way you want. You can: Get rid of the application status information Change the frame rate of your application Tell CCDirector where your high definition images are, and where your standard definition images are, as well as which to use You can change the overall scale of your application to suit different screens The AppDelegate class is also the best place to start any preloading process And, most importantly, it is here you tell the CCDirector object what CCScene to begin your application with Here too you will handle what happens to your application if the OS decides to kill it, push it aside, or hang it upside down to dry. All you need to do is place your logic inside the correct event handler: applicationDidEnterBackground or applicationWillEnterForeground. The HelloWorldScene class When you run the application you get a screen with the words Hello World and a bunch of numbers in one corner. These are the display stats you decided you wanted around in the AppDelegate class. The actual screen is created by the oddly named HelloWorldScene class. It is a Layer class that creates its own scene (don't worry if you don't know what a Layer class is, or a Scene class, you will soon enough). When it initializes, HelloWorldScene puts a button on screen that you can press to exit the application. The button is actually a Menu item, part of a Menu group consisting of one button, two image states for that button, and one callback event, triggered when the said button is pressed. The Menu group automatically handles touch events targeting its members, so you don't get to see any of that code floating about. There is also the necessary Label object to show the Hello World message and the background image. Who begets whom If you never worked with either Cocos2d or Cocos2d-x before, the way the initial scene() method is instantiated may lead to dizziness. To recap, in AppDelegate you have: CCScene *pScene = HelloWorld::scene(); pDirector->runWithScene(pScene); CCDirector needs a CCScene object to run, which you can think of as being your application, basically. CCScene needs something to show, which in this case is a CCLayer class. CCScene is then said to contain a CCLayer class. Here a CCScene object is created through a static method scene inside a CCLayer derived class. So the layer creates the scene, and the scene immediately adds the layer to itself. Huh? Relax. This incestuous-like instantiation will most likely only happen once, and you have nothing to do with it when it happens. So you can easily ignore all these funny goings-on and look the other way. I promise instantiations will be much easier after this first one. Further information Follow these steps to access one of the best sources for reference material on Cocos2d-x: its Test project. Time for action – running the test samples You open the test project just like you would do for any other Xcode project: Go inside the folder you downloaded for the framework, and navigate to samples/TestCpp/proj.ios/TestCpp.xcodeproj. Open that project in Xcode. When you run the project, you will see inside the simulator a long list of tests, all nicely organized by topic. Pick any one to review. Better yet, navigate to samples/TestCpp/Classes and if you have a program like TextWrangler or some equivalent, you can open that entire directory inside a Disk Browser window and have all that information ready for referencing right at your desktop. What just happened? With the test samples you can visualize most features in Cocos2d-x and see what they do, as well as some of the ways you can initialize and customize them. I will refer to the code found in the tests quite often. As usual with programming, there is always a different way to accomplish a given task, so sometimes after showing you one way, I'll refer to another one that you can find (and by then easily understand) inside the Test classes. The other tools Now comes the part where you may need to spend a bit more money to get some extremely helpful tools. In this articles examples I use four of them: A tool to help build sprite sheets: I'll use Texture Packer (http://www.codeandweb.com/texturepacker). There are other alternatives, like Zwoptex (http://zwopple.com/zwoptex/). And they usually offer some features for free. A tool to help build particle effects: I'll use Particle Designer (http://www.71squared.com/en/particledesigner). Depending on your operating system you may find free tools online for this. Cocos2d-x comes bundled with some common particle effects that you can customize. But to do it blindly is a process I do not recommend. A tool to help build bitmap fonts: I'll use Glyph Designer (http://www.71squared.com/en/glyphdesigner). But there are others: bmGlyph (which is not as expensive), FontBuilder (which is free). It is not extremely hard to build a Bitmap font by hand, not nearly as hard as building a particle effect from scratch, but doing it once is enough to convince you to get one of these tools fast. A tool to produce sound effects: No contest. cfxr for Mac or the original sfxr for Windows. Both are free (http://www.drpetter.se/project_sfxr.html and http://thirdcog.eu/apps/cfxr respectively). Summary You just learned how to install Cocos2d-x templates and create a basic application. You also learned enough of the structure of a basic Cocos2d-x application to get started to build your first game. Resources for Article: Further resources on this subject: Getting Started With Cocos2d [Article] Cocos2d: Working with Sprites [Article] Cocos2d for iPhone: Surfing Through Scenes [Article]
Read more
  • 0
  • 0
  • 8926
Unlock access to the largest independent learning library in Tech for FREE!
Get unlimited access to 7500+ expert-authored eBooks and video courses covering every tech area you can think of.
Renews at $19.99/month. Cancel anytime
article-image-so-what-powershell-30-wmi
Packt
05 Sep 2013
5 min read
Save for later

So, what is PowerShell 3.0 WMI?

Packt
05 Sep 2013
5 min read
(For more resources related to this topic, see here.) Microsoft created Windows Management Instrumentation ( WMI ) as a management layer to the Windows operating system. This management layer allows you to retrieve information pertaining to the operating system or physical hardware on a system. It also allows you to manipulate components within the operating system. A good example for this section is a hard drive. WMI provides the ability to view the physical hard drive as well as the logical components of the hard drive. Using a call to the Win32_diskdrive class, you have the ability to view the physical aspects of the disk drive, such as the tracks, sectors, manufacturer, and even the serial number. The win32_logicaldisk class provides you with the ability to see the logical aspects of a drive, such as partitions, free space, and volume names. Not limited to just Windows operating system management, WMI also has the extensibility to allow third-party developers to create WMI providers for use within their projects. This means that you can create the management hooks within your code that will allow remote management through a common standardized framework. Many companies have adopted the use of WMI providers for items such as storage subsystems, application virtualization, hardware virtualization, and enhanced management of the hardware connected to a workstation or server system. Microsoft chose to follow the Common Information Model ( CIM ) industry standard for WMI. The preceding diagram takes a simplified look at Microsoft's implementation of WMI through the use of the CIM standard. There are three layers to their model, which are as follows: WMI consumers: The WMI consumers are exactly what their name states. They consume the available APIs to access the managed component. The WMI consumers are the real users of the C/C++ and .NET clients, and they use scripting languages, such as PowerShell 3.0, to access management data and interact with the managed components. In the hard drive example, the WMI consumer is the PowerShell code that calls information about the hard drive. This would look as follows: get-wmiobject –class win32_logicaldisk WMI infrastructure: The WMI infrastructure includes the CIM Object Manager ( CIMOM ), which stores a repository of the available WMI providers. If a third-party WMI provider doesn't register with the CIM Object Manager, the Windows operating system will not be able to manage the component through WMI. In the hard drive example, the CIMOM will import the Managed Object Format ( MOF) file of the hard drive into the WMI repository. This will register the hard drive's available WMI properties and methods into use by a WMI consumer. WMI providers: The last components, WMI providers, are made up of a driver (DLL) and a MOF file. These two components are responsible for returning the management data to the WMI consumer, through the WMI infrastructure. This allows the WMI consumer to interact with the managed components. In the hard drive example, the WMI consumer will access the hard drive through the WMI infrastructure utilizing the hard drive driver (DLL). The WMI consumer will have the ability to retrieve the information pertaining to the physical components on that hard drive. WMI integration with PowerShell 3.0 PowerShell has the ability to interact with WMI through the use of the built-in cmdlets. These cmdlets act as the WMI consumers and interact with the WMI. As WMI evolved with the release of new operating systems, PowerShell also needed to evolve in parallel to manage those systems. With the release of Windows 8 and Windows Server 2012, Microsoft created a new iteration of the Microsoft Windows Management Framework ( WMF ), Version 3.0. The new release of Windows Management Framework updates WMI to Version 3.0, PowerShell to Version 3.0, and installs the new Windows Remote Management ( WinRM ), OData IIS Extensions, and Server Manager CIM Provider. PowerShell 3.0 includes new WMI management cmdlets, displayed in the following table, which leverage the use of the new functionality within Windows Management Framework 3.0. The new CIM cmdlets provide a richer WMI experience leveraging stateful communications to the remote systems. They also provide the ability to create CIM calls through PowerShell 3.0 to non-Windows-based WMI systems that support Web Services-Management ( WSman ). This provides engineers the ability to tap into a variety of systems for management purposes. Get-CimAssociatedInstance New-CimSession Get-CimClass New-CimSessionOption Get-CimInstance Register-CimIndicationEvent Get-CimSession Remove-CimInstance Invoke-CimMethod Remove-CimSession New-CimInstance Set-CimInstance Using PowerShell in your environment PowerShell is quickly becoming the de-facto standard for managing the Windows-based systems in small and large organizations. It is being used for tasks, such as automated software deployments, dynamic system provisioning, and system maintenance. It is also being heavily used in Microsoft products, such as System Center 2012 - Service Manager, that allow for business process automation. PowerShell 3.0 has introduced a variety of new cmdlets that further simplify the administration of systems through the use of WMI. Whatever the administrative task, the PowerShell community has several examples of the ways to manage systems through the use of the WMI consumers. The examples provided in this article are just the tip of the iceberg compared to what you can accomplish utilizing PowerShell 3.0 and Windows Management Instrumentation. You may be surprised by the number of manual administration steps that can be automated by creating PowerShell scripts. The sky is the limit when it comes to scripting with PowerShell! Summary In this article, we learned what Windows Management Instrumentation (WMI) is, how PowerShell 3.0 utilizes it, and why it's applicable to systems engineers. Resources for Article : Further resources on this subject: Using the Windows Azure Platform PowerShell Cmdlets [Article] Accessing Oracle [Article] Inventorying Servers with PowerShell [Article]
Read more
  • 0
  • 0
  • 9339

article-image-soa-application-design
Packt
04 Sep 2013
9 min read
Save for later

SOA Application Design

Packt
04 Sep 2013
9 min read
(For more resources related to this topic, see here.) In this article, we'll focus on recipes for designing high performance SOA Suite 11g applications. These recipes look at how you can design your applications for high performance and scalability, where high performance is defined as providing low response times even under load, and scalability is defined as the ability to expand to cope with large numbers of requests. While many of the recipes in other articles can be applied after the application has been designed and written, those in this article need to be applied while the application is being written, and may require that your application is implemented in a certain way. Designing an application with performance as a requirement from the start is much easier than trying to add performance to an application that is already live. So, the recipes in this article provide some of the best value for money in terms of getting the most performance out of your SOA Suite infrastructure. However, while this book focuses on decisions that should be made during the design stages of a development process, this article is not a list of general SOA Suite design patterns. As for many of the recipes in other articles, a lot of the focus in this article is on reducing the amount of time your application spends waiting on external services and the SOA Suite database tables. There are many aspects to the performance of a SOA Suite application, and the design guidelines depend very much on the particular business problems that your application is designed to solve. Factors such as payload size, number of external systems being orchestrated, data transformation complexity, and persistence requirements, all have an impact on the performance of your application. Performance is a relative term, with each application and use-case having its own requirements, but there are a number of basic principles that can help ensure that your application will have a good chance of meeting its goals. Design for peak loads, not average loads. Average loads can be very misleading; there are many situations in which the average load of a system is not a good indicator of the expected load. A good example of this would be a tax return system, where the usage for most of the year is very low, building into a peak in 30 or so days before people's tax returns are due. Smaller payloads are faster. When designing your application, try and limit the amount of payload data that goes through your composites and processes. It is often better to store the data in a database and send the key and metadata through the processes, only going to retrieve data when required. Understand your transaction boundaries. Many applications suffer performance problems because their transactions boundaries are in the wrong places, causing work to be redone unnecessarily when failures happen, or leaving data in an inconsistent state. Understand what causes your application to access the database, and why. Much of the performance overhead of Oracle SOA Suite applications is in repeated trips to the database. These trips add value by persisting state between steps or within processes, but the overuse of steps that cause database persistence is a common cause of performance problems. Follow standard web service design patterns, such as using asynchronous callbacks and stateless invocations, where you are using web services. Using BPEL process parallelization By having your BPEL process execute steps in parallel when there are no dependencies, you can increase the performance by spending less time waiting for external systems to complete. Getting ready You will need JDeveloper installed, and have an open BPEL project. How to do it... Follow these steps to use BPEL process parallelization: Expand the BPEL Constructs section in the component palette. Drag Flow from the palette onto the process. Click on the + icon next to the flow to expand it. Populate the flow with the process steps. How it works... If you have a number of tasks that do not have dependencies on each other, you can improve performance by executing the preceding tasks in parallel. This is most effective with partner links, where you know you are waiting on an external system to produce a response. The default behaviors of these flows is still to use a single thread to execute the branches if external systems are invoked. See the Using non-blocking service invocations in BPEL recipe to learn how to execute flows that contain partner links in parallel. There's more… It is possible to include a limited amount of synchronization between branches of a flow, so that tasks on one branch will wait for tasks on another branch to complete before proceeding. This is best used with caution, but it can provide benefits, and allow tasks that would not otherwise easily lend themselves to parallelization to be run in parallel. Using non-blocking service invocations in BPEL flows We can reduce the latency of forked external service invocations in a BPEL process to the longest flow's execution time if we assign a thread to each flow, making it multi-threaded. Getting ready You'll need a composite loaded in JDeveloper to execute this recipe. This composite will need a flow that makes calls to a partner link external service. How to do it... Follow these steps to use non-blocking service invocations: Right-click on each partner link that is being executed in your BPEL process flow, and select Edit. In the Property tab, select the green + icon and add nonBlockingInvoke as a property name. In the Value box at the bottom, enter true. How it works... This recipe causes flow branches to be executed in parallel, with a new thread to be used for each branch flow. For multiple service invocations that each have a high latency, this can greatly improve the total BPEL execution time. For example, assume we have a BPEL process that calls two web services, one that takes four seconds to execute, and one that takes six seconds to execute. Applying this change will prevent the BPEL process making the calls serially, which would take 10 seconds in total, and enforce parallel service calls in separate threads, reducing the execution time to just over six seconds, or the latency of the longest call plus time to collate the results in the main BPEL process execution thread. While it may sound like a silver bullet performance improvement, this recipe is actually not necessarily going to improve the execution time of our BPEL process! Consider that we may now be at the mercy of greater thread context switching in the CPU; for every invocation of our process, we now have a larger number of threads that will be spawned. If each service invocation has a low latency, the overhead of creating threads and collating callbacks might actually be greater than the cost of invoking the services in a single thread. Our example in this explanation is contrived, so ensure to test the response time of your composite and the profile of your application, when placed under operational load (which may result in lots of threads spawning), as these may well be different with the configuration applied. There's more… This recipe used an alternative way of setting property values to that which we've used elsewhere in the book. Previously, we've edited composite files directly; here, we used the JDeveloper BPEL graphical editor to achieve the same end result. If you check the composite.xml source, you'll see a property added with a name, such as partnerLink.[your service name].nonBlockingInvoke for each service added. Turning off payload validation and composite state monitoring Payload validation checks all inbound and outbound message data thus adding an overhead, especially for large message types. Composite state monitoring allows for administrators to view the results of all instance invocations. We can disable these to improve performance. Getting ready You will need to know the administration credentials for your Oracle SOA Suite WebLogic domain, and have access to the Oracle Enterprise Manager console. How to do it... By following these steps, we can turn off payload validation: Log in to Enterprise Manager. Open the SOA tab, and right-click on soa_infra , select SOA Administration and Common Properties . Un-tick the checkbox for Payload Validation to disable this feature. Un-tick the checkbox for Capture Composite Instance State. How it works... In this recipe, we globally disabled payload validation. This instructs SOA Suite to not check the inbound and outbound message payloads against the schemas associated with our services. This can be particularly useful, not only if the payload is coming from a trusted source, but even if the source is untrusted. A common alternative to payload validation is to add steps to manually validate the payloads at the point that we first receive the request, while not validating those that have come from internal or trusted sources. There are a number of levels of granularity for payload validation; it can be applied at the SOA Engine (BPEL) and composite levels to allow for fine-grained application of this property. You can access these properties via the enterprise manager console right-click menu on the SOA engines and deployed composites. For performance, I would recommend disabling this in all environments above development. Composite state management is responsible for tracking and representing the health of our running composites. This is a powerful administration feature, but costs a lot in terms of performance. Anecdotal testing shows that this can be responsible for up to 30 percent of processing time. As such, for high throughput applications, the value of this feature should be considered. There's more… See the recipes on audit logging to further control composite recording activities at runtime. Ensure that you check the payload validation at the Engine and Composite levels to ensure that they meet your performance requirements.
Read more
  • 0
  • 0
  • 5384

article-image-learning-musescore
Packt
04 Sep 2013
15 min read
Save for later

Learning MuseScore

Packt
04 Sep 2013
15 min read
(For more resources related to this topic, see here.) Entering notes In order to enter notes into our score, we need to enter Note Entry mode. MuseScore has various modes that we can use to accomplish special tasks. You can enter Note Entry mode by clicking on the N button in the toolbar. You can tell whether you are in Note Entry mode at any given time by checking whether the N button is depressed. You may also enter/exit Note Entry mode by pressing the N key. After you enter Note Entry mode, the quarter note should be selected by default. If you hover over the staff, you should see a light blue outline of a note appear. Clicking here will cause a quarter note of that pitch to be inserted. In the toolbar, you will see several notes of different lengths, such as half notes, eighth notes, and whole notes. This area is called the Note Entry toolbar, and indicates which note will be inserted when you click on the staff. Right now, the quarter note should be selected. Click on the half note, and then click an area of the staff on top of the rest that is immediately after the quarter note we just inserted. A half note of the pitch you chose will be added. In MuseScore, whenever we add notes, we must overwrite other notes. First, we overwrote a whole rest with a quarter note, which caused three beats of rest to be added after the quarter note. Then, we overwrote a quarter rest with a half note. Since the half note was longer than the quarter rest, it also overwrote one beat from the half rest following it, and changed the rest to a quarter rest to accommodate the size of the half note. To add an accidental, simply insert the note without the accidental, and then press the appropriate accidental button in the toolbar. For example, let's insert an F eighth note. We click on the eighth note button, then on the F line of the staff, and finally on the sharp button in the toolbar. We can insert dotted notes in a similar fashion by using the dot button on the Note Entry toolbar. In the next measure, let's add a G dotted quarter note by clicking on the quarter note in the Note Entry toolbar, then clicking on the dot button, and then clicking on a G in the staff. The dot will stay selected after you insert the note. If you would like to deselect the dot, you can click on it again. It is also automatically deselected when you change the note duration. Thus, you should always select the dot after you select the value of the note you would like to be dotted. It is possible to notate more quickly using keyboard shortcuts. The number keys 1 through 9 will select different durations, and the letters A through G will insert the designated note. The 0 key inserts a rest. Inserting notes this way will always insert the closest note with the desired pitch. If you hold Ctrl (or on Mac) while pressing the up or down arrow keys, MuseScore will move the last note you inserted up or down an octave. So, inserting a C half note and moving it up an octave can be accomplished by pressing the sequence 6 C Ctrl + ↑. Notes can be adjusted by a half step by pressing the up or down arrows without holding the Ctrl key. Hitting the up arrow will always create sharps, and the down arrow creates flats. This allows us to insert an F eighth note with the keystroke sequence 4 f ↑. While at first the keyboard shortcuts may seem complicated, as you get the hang of MuseScore, it is worthwhile to learn them. They will allow you to notate music extremely quickly and make your overall experience with MuseScore much more pleasurable. Making chords is also very straightforward. We just click on top of our previously inserted note after selecting a note of the same value. Be careful! If a different note length is selected, it will overwrite the previous note. Chords can also be inserted rapidly with keyboard shortcuts. Just start by inserting the first note of the chord normally. If you would like to insert a note of the chord above the previous note, hold Alt and press the interval above the previous note you would like to insert. To insert it below, hold Shift and do the same. Notes are always inserted in the present key signature. So to insert a C first inversion chord, press the sequence E Alt + 3 Alt + 4, or to insert a C second inversion chord, press the sequence G Alt + 4 Alt + 3. Alternatively, after inserting the first note, you can hold Shift and type the letter names of the notes to add to the chord. So pressing the sequence G Shift+C Shift+E would insert the same C second inversion chord. If you ever make a mistake, you can always undo your latest changes by going to the Edit menu and selecting Undo. You can also use the keyboard shortcut Ctrl + Z (or + Z on Mac). Let's put some notes and chords in some measures for both the trombone and piano parts so that we have something to work with. Inserting triplets To insert a triplet, first enter Note Entry mode. Then, from the Note Entry toolbar, choose the total duration that you would like all three triplets to sum to. Next, insert the first note of the triplet in the position you would like the triplet to occupy. After this, exit Note Entry mode, and from the Notes menu, under the Tuplets submenu, click on the Triplet option. A triplet will be created with the selected note as the first note. MuseScore will automatically enter Note Entry mode for you again, and select the correct duration of note needed to complete the triplet. From here, you can replace the two rests with notes by inserting the correct notes on top of them, as we did when we entered notes previously. Also, there is a keyboard shortcut to make this process easier. While in Note Entry mode, select the proper duration you would like the entire triplet to be, as before, but then hit Ctrl + 3 (or + 3 on Mac). The triplet will be inserted, and the proper note duration to fill in the triplet will be selected. You can now enter the notes of the triplet as you would enter normal notes. For instance, to insert a triplet arpeggio of an F major triad totaling one beat, we would press the sequence 5 Ctrl + 3 F A C. For a B major triad totaling two beats, we would similarly press 6 Ctrl + 3 B D ↑ F ↑. Inserting ties Ties are very easy to create in MuseScore. The simplest way to insert a tie is to insert both of the notes that you want to be tied together, exit Note Entry mode, click on the first note, and then click on the tie button in the toolbar, or press the + key. Make sure the two notes you are trying to tie together have the same pitch, or no tie will be inserted. This method works for individual notes, and also for chords. In order to have flexibility when tying chords, you must tie each note of the chord individually if you want the full chord to be tied. An easy way to do this is to ensure that you are not in Note Entry mode, hold Shift, click on the first note of the first chord so that the whole chord is selected, and press the + key. Again, for this to work, you must have two chords with identical pitches next to each other. If you are working with keyboard shortcuts, then there is also a faster way to enter ties that does not require the use of the mouse. After you enter a note in Note Entry mode, the note you just entered will be selected, and the cursor will be located on the right-hand side of this note, as shown in the following screenshot: Then, using the appropriate keyboard shortcut, select the duration of note you would like this note to be tied to. Finally, press the + key. MuseScore will insert a note of the selected duration tied to the previous note. So, pressing the sequence 5 C 4 + will insert a quarter note C tied to an eighth note. While this method is extremely convenient for single notes, it does not work for chords. Often, it is necessary to flip the tie for visual appeal, especially when tying chords. This can be accomplished by ensuring that you are not in Note Entry mode, clicking on a tie, and then pressing the X key. Even though ties look very similar to slurs in many situations, they are created differently. Slurs will be discussed later. Copying and pasting Suppose that we would like to repeat a measure in the bass line, or that the next measure in the melody is very similar to the previous measure. As in a word processor, we can copy and paste measures and fragments of music. First, let's copy and paste a measure. Exit Note Entry mode by ensuring the N button in the toolbar is not selected. Then, click on a portion of the measure where no notes are present. The measure should be selected, as indicated by the blue box around it. Now, either go to the Edit menu and click on Copy, or press Ctrl + C ( + C on Mac). The measure will be copied to the clipboard. Now, click on a portion of the target measure without any notes, and either click on Paste from the Edit menu, or press Ctrl + V ( + V on Mac). The notes will be inserted, and the target measure will be overwritten. It is also possible to copy any portion of your score, even if it spans partial measures or multiple staves. First, click on the note at the top-left of the region you want to copy. In the following example, this would be the E♭ in the right hand. Then, press and hold the Shift key, and click on the note at the bottom right corner of the region you would like to copy. Here, that would be the D in the left hand. MuseScore will select all of the notes in between. Once you have selected the region, you can copy it in the same way you copied the measure before. To paste the region, click on the first note or rest in the uppermost stave where you would like to paste it, and paste as we did with a single measure using either Ctrl + V or Paste from the Edit menu. If your selection has different measure breaks or is in a different meter than the destination, the selection will be reflowed to fit the destination, and ties will be added as necessary. Inserting and deleting measures Often, it is helpful to insert or delete a measure in your score. Luckily, MuseScore makes this extremely easy. To insert a measure, select the measure (as we did when we copied a measure) immediately after the location where you would like to insert the measure. Then, go to the Create menu, and under the Measures submenu, select Insert Measure. A measure will be inserted. To insert multiple measures, select Insert Measures. A dialog box will prompt you for how many measures to insert. If you would like to add measures to the end of the score, you can select Append Measures from under the Measures submenu within the Create menu. There is no need to select any measures to perform this operation. To delete measures, simply select the measure by clicking any blank area within the measure, and then go to the Edit menu, and click on Delete Selected Measures. Doing so will delete this measure position within all staves, not just the selected staff. You can also select multiple measures (as we did earlier when we were copying by selecting one measure, holding the Shift key, and selecting additional measures), and use the same menu button to delete all of the measures that you have selected. Chord symbols In jazz and popular music, it is very common to give musicians chord symbols to read from. To create a chord symbol, make sure you are not in Note Entry mode, and click on a note that you would like to add a chord symbol to. Then, either go to the Create menu, go to the Text submenu, and select Chord Name, or press Ctrl + K ( + K on Mac). A text box should appear that looks exactly like the ones we saw before. Now, you can type the name of the chord in the same way you would write it on paper. (For example, D minor would be Dm, and a G7 chord would just be G7.) All lowercase b characters will be converted into flat signs, and all # characters will be converted into sharps. To move to the next location in the measure, press the space bar. If you press the space bar repeatedly, you will move forward without inserting any chords. Now that our chords are inserted, we can optionally make them look stylized. To do this, go to the Style menu and click on Edit General Style. Then, click on the Chordnames option on the left-hand side. You should see a textbox appear on the right-hand side containing the text stdchords.xml. Change this to jazzchords.xml, and then press OK. The chords you entered should be appropriately stylized. Many styles of notation, especially within jazz music, use chord symbols and slashes to indicate improvisation. To create these slashes in MuseScore, insert four quarter notes on the middle line of the staff. Then, after exiting Note Entry mode, right-click on each note and select Note Properties. Check the box that says Stemless. Also, find the option labeled velocity type and choose user, and then change the value of the box velocity (0-127) to 0. Now press OK. Then, locate the section of the palette labeled Note Heads, and drag the parallelogram slash shape on top of each note. This will create the slash notation. Beaming The proper beaming of notes is a key feature of quality engraved scores that often goes unappreciated. It is extremely easy to change the beaming patterns to enhance the readability of your score. There are several utilities in the palette that allow for this. To start, go to the section of the palette labeled Beam Properties. Hovering over each icon will tell you what it does. These properties can be applied to different notes. The Start beam option is for notes in the middle of an existing beam. It breaks the existing beam at the specified note, and starts a new beam on that note. The Middle of the beam option will ensure that the selected note is beamed to the notes on both sides of it, and the No beam option will break any beams going to the selected note. Let's learn how to use these with a simple use case scenario. Suppose you enter three eighth notes followed by an eighth rest. MuseScore will automatically choose the following beaming: However, to a musician who is sight-reading, it may be easy to confuse this with a triplet. To correct this, simply drag the No beam icon on top of the third eighth note in the passage. The note should highlight red as you hover over it, before you drop it. Once you let go of the mouse button, MuseScore will automatically adjust the beam according to what you specified. Similarly, choosing the beaming wisely can make difficult passages easier to read. Let's consider the case of two sixteenth notes followed by two eighth notes and two more sixteenth notes. Especially with the sharps and flats in this example, it would not be easy to sight-read such a passage. However, dragging the Start beam option on top of the B♮ makes this passage much cleaner and easier to read. To undo any of these changes, ensure that you are not in Note Entry mode, and click on the note that you have changed. Then, in the Beam Properties section of the palette, double-click the A icon to reset it back to default. Though MuseScore uses standard conventions for whether to put the beam above or below the notes, if you would like to change this, simply ensure that you are not in Note Entry mode, click on the beam, and press the X key. The beam will flip to the other side of the staff. Summary In this article, we learned the basics of creating notes including ties and triplets, copying and pasting measures, creating chord symbols, and also changing the beaming patterns to enhance the readability of our score. Resources for Article: Further resources on this subject: Importing and Adding Background Music with Audacity 1.3 [Article] New iPad Features in iOS 6 [Article] Quick start – media files and XBMC [Article]
Read more
  • 0
  • 0
  • 4553

article-image-introduction-xenconvert
Packt
04 Sep 2013
3 min read
Save for later

Introduction to XenConvert

Packt
04 Sep 2013
3 min read
(For more resources related to this topic, see here.) System requirements Since XenConvert can only convert Windows-based hosts and installs on the same host, the requirements are pretty much the same, as follows: Operating system: Windows XP, Windows Vista, Windows 7, Windows Server 2003 (SP1 or later), Windows Server 2008 (R2) .Net Framework 4.0 Disk Space: 40 MB free disk space XenServer version 6.0 or 6.1 Converting a physical machine to a virtual machine Let's take a quick look at how to convert a physical machine to a virtual machine. First we need to install XenConvert on the source physical machine. We can download XenConvert from this link: http://www.citrix.com/downloads/xenserver/tools/conversion.html. Once the standard Windows installation process is complete, launch the XenConvert tool; but before that we need to prepare the host machine for the conversion. To know more about XenConvert, refer to the XenConvert guide at http://support.citrix.com/article/CTX135017. Preparing the host machine For best results, prepare the host machine as follows: Enable Windows Automount on Windows Server operating systems. Disable Windows Autoplay. Remove any virtualization software before performing a conversion. Ensure that adequate free space exists at the destination, which is approximately 101 percent of used space of all source volumes. Remove any network interface teams; they are not applicable to a virtual machine. We need to run the XenConvert tool on the host machine to start the physical-to-virtual conversion. We can convert the physical machine directly to our XenServer if this host machine is accessible. The other options are to convert to VHD, OVF, or vDisk, which can be imported later on to XenServer using some methods. These options are more useful if we don't have enough disk space or connectivity with XenServer. I chose XenServer and clicked on Next . We can select multiple partitions to be included in the conversion, or select none from the drop-down menu in Source Volume and those disks won't be included in the conversion. We can also increase or decrease the size of the new virtual partition to be allocated for this virtual machine. Click on Next . We'll be asked to provide the details of the XenServer host. The hostname needs either an IP address or a FQDN of the XenServer; a username and password are standard login requirements. In the Workspace field, enter the path to the folder to store the intermediate OVF package that XenConvert will use during the conversion process. XenConvert will store the OVF package in the path we give. Click on Next to select the storage repositories found with XenServer and continue to the last step, in which we'll be provided with the summary of the conversion. Soon after the conversion is completed, we'll be able to have this new machine in our XenCenter. We'll need to have XenServer Tools installed on this new virtual machine. Summary In this article we covered an advanced topic that explained the process of converting a physical Windows server to a virtual machine using XenConvert. Resources for Article : Further resources on this subject: Citrix XenApp Performance Essentials [Article] Defining alerts [Article] Publishing applications [Article]
Read more
  • 0
  • 0
  • 8933
article-image-understanding-point-time-recovery
Packt
04 Sep 2013
28 min read
Save for later

Understanding Point-In-Time-Recovery

Packt
04 Sep 2013
28 min read
(For more resources related to this topic, see here.) Understanding the purpose of PITR PostgreSQL offers a tool called pg_dump to backup a database. Basically, pg_dump will connect to the database, read all the data in transaction isolation level "serializable" and return the data as text. As we are using "serializable", the dump is always consistent. So, if your pg_dump starts at midnight and finishes at 6 A.M, you will have created a backup, which contains all the data as of midnight but no further data. This kind of snapshot creation is highly convenient and perfectly feasible for small to medium amounts of data. A dump is always consistent. This means that all foreign keys are intact; new data added after starting the dump will be missing. It is most likely the most common way to perform standard backups. But, what if your data is so valuable and maybe so large in size that you want to backup it incrementally? Taking a snapshot from time to time might be enough for some applications; for highly critical data, it is clearly not. In addition to that, replaying 20 TB of data in textual form is not efficient either. Point-In-Time-Recovery has been designed to address this problem. How does it work? Based on a snapshot of the database, the XLOG will be replayed later on. This can happen indefinitely or up to a point chosen by you. This way, you can reach any point in time. This method opens the door to many different approaches and features: Restoring a database instance up to a given point in time Creating a standby database, which holds a copy of the original data Creating a history of all changes In this article, we will specifically feature on the incremental backup functionality and describe how you can make your data more secure by incrementally archiving changes to a medium of choice. Moving to the bigger picture The following picture provides an overview of the general architecture in use for Point-In-Time-Recovery: PostgreSQL produces 16 MB segments of transaction log. Every time one of those segments is filled up and ready, PostgreSQL will call the so called archive_command. The goal of archive_command is to transport the XLOG file from the database instance to an archive. In our image, the archive is represented as the pot on the bottom-right side of the image. The beauty of the design is that you can basically use an arbitrary shell script to archive the transaction log. Here are some ideas: Use some simple copy to transport data to an NFS share Run rsync to move a file Use a custom made script to checksum the XLOG file and move it to an FTP server Copy the XLOG file to a tape The possible options to manage XLOG are only limited by imagination. The restore_command is the exact counterpart of the archive_command. Its purpose is to fetch data from the archive and provide it to the instance, which is supposed to replay it (in our image, this is labeled as Restored Backup). As you have seen, replay might be used for replication or simply to restore a database to a given point in time as outlined in this article. Again, the restore_command is simply a shell script doing whatever you wish, file by file. It is important to mention that you, the almighty administrator, are in charge of the archive. You have to decide how much XLOG to keep and when to delete it. The importance of this task cannot be underestimated. Keep in mind, when then archive_command fails for some reason, PostgreSQL will keep the XLOG file and retry after a couple of seconds. If archiving fails constantly from a certain point on, it might happen that the master fills up. The sequence of XLOG files must not be interrupted; if a single file is missing, you cannot continue to replay XLOG. All XLOG files must be present because PostgreSQL needs an uninterrupted sequence of XLOG files; if a single file is missing, the recovery process will stop there at the very latest. Archiving the transaction log After taking a look at the big picture, we can take a look and see how things can be put to work. The first thing you have to do when it comes to Point-In-Time-Recovery is to archive the XLOG. PostgreSQL offers all the configuration options related to archiving through postgresql.conf. Let us see step by step what has to be done in postgresql.conf to start archiving: First of all, you should turn archive_mode on. In the second step, you should configure your archive command. The archive command is a simple shell command taking two parameters: %p: This is a placeholder representing the XLOG file that should be archived, including its full path (source). %f: This variable holds the name of the XLOG without the path pointing to it. Let us set up archiving now. To do so, we should create a place to put the XLOG. Ideally, the XLOG is not stored on the same hardware as the database instance you want to archive. For the sake of this example, we assume that we want to apply an archive to /archive. The following changes have to be made to postgresql.conf: wal_level = archive # minimal, archive, or hot_standby # (change requires restart) archive_mode = on # allows archiving to be done # (change requires restart) archive_command = 'cp %p /archive/%f' # command to use to archive a logfile segment # placeholders: %p = path of file to archive # %f = file name only Once those changes have been made, archiving is ready for action and you can simply restart the database to activate things. Before we restart the database instance, we want to focus your attention on wal_level. Currently three different wal_level settings are available: minimal archive hot_standby The amount of transaction log produced in the case of just a single node is by far not enough to synchronize an entire second instance. There are some optimizations in PostgreSQL, which allow XLOG-writing to be skipped in the case of single-node mode. The following instructions can benefit from wal_level being set to minimal: CREATE TABLE AS, CREATE INDEX, CLUSTER, and COPY (if the table was created or truncated within the same transaction). To replay the transaction log, at least archive is needed. The difference between archive and hot_standby is that archive does not have to know about currently running transactions. For streaming replication, however, this information is vital. Restarting can either be done through pg_ctl –D /data_directory –m fast restart directly or through a standard init script. The easiest way to check if our archiving works is to create some useless data inside the database. The following code snippets shows a million rows can be made easily: test=# CREATE TABLE t_test AS SELECT * FROM generate_series(1,1000000);SELECT 1000000test=# SELECT * FROM t_test LIMIT 3;generate_series----------------- 1 2 3(3 rows) We have simply created a list of numbers. The important thing is that 1 million rows will trigger a fair amount of XLOG traffic. You will see that a handful of files have made it to the archive: iMac:archivehs$ ls -ltotal 131072-rw------- 1 hs wheel 16777216 Mar 5 22:31000000010000000000000001-rw------- 1 hs wheel 16777216 Mar 5 22:31000000010000000000000002-rw------- 1 hs wheel 16777216 Mar 5 22:31000000010000000000000003-rw------- 1 hs wheel 16777216 Mar 5 22:31000000010000000000000004 Those files can be easily used for future replay operations. If you want to save storage, you can also compress those XLOG files. Just add gzip to your archive_command. Taking base backups In the previous section, you have seen that enabling archiving takes just a handful of lines and offers a great deal of flexibility. In this section, we will see how to create a so called base backup, which can be used to apply XLOG later on. A base backup is an initial copy of the data. Keep in mind that the XLOG itself is more or less worthless. It is only useful in combination with the initial base backup. In PostgreSQL, there are two main options to create an initial base backup: Using pg_basebackup Traditional copy/rsync based methods The following two sections will explain in detail how a base backup can be created: Using pg_basebackup The first and most common method to create a backup of an existing server is to run a command called pg_basebackup, which was introduced in PostgreSQL 9.1.0. Basically pg_basebackup is able to fetch a database base backup directly over a database connection. When executed on the slave, pg_basebackup will connect to the database server of your choice and copy all the data files in the data directory over to your machine. There is no need to log into the box anymore, and all it takes is one line of code to run it; pg_basebackup will do all the rest for you. In this example, we will assume that we want to take a base backup of a host called sample.postgresql-support.de. The following steps must be performed: Modify pg_hba.conf to allow replication Signal the master to take pg_hba.conf changes into account Call pg_basebackup Modifying pg_hba.conf To allow remote boxes to log into a PostgreSQL server and to stream XLOG, you have to explicitly allow replication. In PostgreSQL, there is a file called pg_hba.conf, which tells the server which boxes are allowed to connect using which type of credentials. Entire IP ranges can be allowed or simply discarded through pg_hba.conf. To enable replication, we have to add one line for each IP range we want to allow. The following listing contains an example of a valid configuration: # TYPE DATABASE USER ADDRESS METHODhost replication all 192.168.0.34/32 md5 In this case we allow replication connections from 192.168.0.34. The IP range is identified by 32 (which simply represents a single server in our case). We have decided to use MD5 as our authentication method. It means that the pg_basebackup has to supply a password to the server. If you are doing this in a non-security critical environment, using trust as authentication method might also be an option. What happens if you actually have a database called replication in your system? Basically, setting the database to replication will just configure your streaming behavior, if you want to put in rules dealing with the database called replication, you have to quote the database name as follows: "replication". However, we strongly advise not to do this kind of trickery to avoid confusion. Signaling the master server Once pg_hba.conf has been changed, we can tell PostgreSQL to reload the configuration. There is no need to restart the database completely. We have three options to make PostgreSQL reload pg_hba.conf: By running an SQL command: SELECT pg_reload_conf(); By sending a signal to the master: kill –HUP 4711 (with 4711 being the process ID of the master) By calling pg_ctl: pg_ctl –D $PGDATA reload (with $PGDATA being the home directory of your database instance) Once we have told the server acting as data source to accept streaming connections, we can move forward and run pg_basebackup. pg_basebackup – basic features pg_basebackup is a very simple-to-use command-line tool for PostgreSQL. It has to be called on the target system and will provide you with a ready-to-use base backup, which is ready to consume the transaction log for Point-In-Time-Recovery. The syntax of pg_basebackup is as follows: iMac:dbhs$ pg_basebackup --help pg_basebackup takes a base backup of a running PostgreSQL server. Usage: pg_basebackup [OPTION]... Options controlling the output: -D, --pgdata=DIRECTORY receive base backup into directory -F, --format=p|t output format (plain (default), tar) -x, --xlog include required WAL files in backup (fetch mode) -X, --xlog-method=fetch|stream include required WAL files with specified method -z, --gzip compress tar output -Z, --compress=0-9 compress tar output with given compression level General options: -c, --checkpoint=fast|spread set fast or spread checkpointing -l, --label=LABEL set backup label -P, --progress show progress information -v, --verbose output verbose messages -V, --version output version information, then exit -?, --help show this help, then exit Connection options: -h, --host=HOSTNAME database server host or socket directory -p, --port=PORT database server port number -s, --status-interval=INTERVAL time between status packets sent to server (in seconds) -U, --username=NAME connect as specified database user -w, --no-password never prompt for password -W, --password force password prompt (should happen automatically) A basic call to pg_basebackup would look like that: iMac:dbhs$ pg_basebackup -D /target_directory -h sample.postgresql-support.de In this example, we will fetch the base backup from sample.postgresql-support.de and put it into our local directory called /target_directory. It just takes this simple line to copy an entire database instance to the target system. When we create a base backup as shown in this section, pg_basebackup will connect to the server and wait for a checkpoint to happen before the actual copy process is started. In this mode, this is necessary because the replay process will start exactly at this point in the XLOG. The problem is that it might take a while until a checkpoint kicks in; pg_basebackup does not enforce a checkpoint on the source server straight away to make sure that normal operations are not disturbed. If you don't want to wait on a checkpoint, consider using --checkpoint=fast. It will enforce an instant checkpoint and pg_basebackup will start copying instantly. By default, a plain base backup will be created. It will consist of all the files in directories found on the source server. If the base backup should be stored on tape, we suggest to give –-format=t a try. It will automatically create a TAR archive (maybe on a tape). If you want to move data to a tape, you can save an intermediate step easily this way. When using TAR, it is usually quite beneficial to use it in combination with --gzip to reduce the size of the base backup on disk. There is also a way to see a progress bar while doing the base backup but we don't recommend to use this option (--progress) because it requires pg_basebackup to determine the size of the source instance first, which can be costly. pg_basebackup – self-sufficient backups Usually a base backup without XLOG is pretty worthless. This is because the base backup is taken while the master is fully operational. While the backup is taken, those storage files in the database instance might have been modified heavily. The purpose of the XLOG is to fix those potential issues in the data files reliably. But, what if we want to create a base backup, which can live without (explicitly archived) XLOG? In this case, we can use the --xlog-method=stream option. If this option has been chosen, pg_basebackup will not just copy the data as it is but it will also stream the XLOG being created during the base backup to our destination server. This will provide us with just enough XLOG to allow us to start a base backup made that way directly. It is self-sufficient and does not need additional XLOG files. This is not Point-In-Time-Recovery but it can come in handy in case of trouble. Having a base backup, which can be started right away, is usually a good thing and it comes at fairly low cost. Please note that --xlog-method=stream will require two database connections to the source server, not just one. You have to keep that in mind when adjusting max_wal_senders on the source server. If you are planning to use Point-In-Time-Recovery and if there is absolutely no need to start the backup as it is, you can safely skip the XLOG and save some space this way (default mode). Making use of traditional methods to create base backups These days pg_basebackup is the most common way to get an initial copy of a database server. This has not always been the case. Traditionally, a different method has been used which works as follows: Call SELECT pg_start_backup('some label'); Copy all data files to the remote box through rsync or any other means. Run SELECT pg_stop_backup(); The main advantage of this old method is that there is no need to open a database connection and no need to configure XLOG-streaming infrastructure on the source server. A main advantage is also that you can make use of features such as ZFS-snapshots or similar means, which can help to dramatically reduce the amount of I/O needed to create an initial backup. Once you have started pg_start_backup, there is no need to hurry. It is not necessary and not even especially desirable to leave the backup mode soon. Nothing will happen if you are in backup mode for days. PostgreSQL will archive the transaction log as usual and the user won't face any kind of downside. Of course, it is bad habit not to close backups soon and properly. However, the way PostgreSQL works internally does not change when a base backup is running. There is nothing filling up, no disk I/O delayed, or anything of this sort. Tablespace issues If you happen to use more than one tablespace, pg_basebackup will handle this just fine if the filesystem layout on the target box is identical to the filesystem layout on the master. However, if your target system does not use the same filesystem layout there is a bit more work to do. Using the traditional way of doing the base backup might be beneficial in this case. In case you are using --format=t (for TAR), you will be provided with one TAR file per tablespace. Keeping an eye on network bandwidth Let us imagine a simple scenario involving two servers. Each server might have just one disk (no SSDs). Our two boxes might be interconnected through a 1 Gbit link. What will happen to your applications if the second server starts to run a pg_basebackup? The second box will connect, start to stream data at full speed and easily kill your hard drive by using the full bandwidth of your network. An application running on the master might instantly face disk wait and offer bad response times. Therefore it is highly recommended to control the bandwidth used up by rsync to make sure that your business applications have enough spare capacity (mainly disk, CPU is usually not an issue). If you want to limit rsync to, say, 20 MB/sec, you can simply use rsync --bwlimit=20000. This will definitely make the creation of the base backup take longer but it will make sure that your client apps will not face problems. In general we recommend a dedicated network interconnect between master and slave to make sure that a base backup does not affect normal operations. Limiting bandwidth cannot be done with pg_basebackup onboard functionality.Of course, you can use any other tool to copy data and achieve similar results. If you are using gzip compression with –-gzip, it can work as an implicit speed brake. However, this is mainly a workaround. Replaying the transaction log Once we have created ourselves a shiny initial base backup, we can collect the XLOG files created by the database. When the time has come, we can take all those XLOG files and perform our desired recovery process. This works as described in this section. Performing a basic recovery In PostgreSQL, the entire recovery process is governed by a file named recovery.conf, which has to reside in the main directory of the base backup. It is read during startup and tells the database server where to find the XLOG archive, when to end replay, and so forth. To get you started, we have decided to include a simple recovery.conf sample file for performing a basic recovery process: restore_command = 'cp /archive/%f %p'recovery_target_time = '2013-10-10 13:43:12' The restore_command is essentially the exact counterpart of the archive_command you have seen before. While the archive_command is supposed to put data into the archive, the restore_command is supposed to provide the recovering instance with the data file by file. Again, it is a simple shell command or a simple shell script providing one chunk of XLOG after the other. The options you have here are only limited by imagination; all PostgreSQL does is to check for the return code of the code you have written, and replay the data provided by your script. Just like in postgresql.conf, we have used %p and %f as placeholders; the meaning of those two placeholders is exactly the same. To tell the system when to stop recovery, we can set the recovery_target_time. The variable is actually optional. If it has not been specified, PostgreSQL will recover until it runs out of XLOG. In many cases, simply consuming the entire XLOG is a highly desirable process; if something crashes, you want to restore as much data as possible. But, it is not always so. If you want to make PostgreSQL stop recovery at a specific point in time, you simply have to put the proper date in. The crucial part here is actually to know how far you want to replay XLOG; in a real work scenario this has proven to be the trickiest question to answer. If you happen to a recovery_target_time, which is in the future, don't worry, PostgreSQL will start at the very last transaction available in your XLOG and simply stop recovery. The database instance will still be consistent and ready for action. You cannot break PostgreSQL, but, you might break your applications in case data is lost because of missing XLOG. Before starting PostgreSQL, you have to run chmod 700 on the directory containing the base backup, otherwise, PostgreSQL will error out: iMac:target_directoryhs$ pg_ctl -D /target_directorystartserver startingFATAL: data directory "/target_directory" has group or world accessDETAIL: Permissions should be u=rwx (0700). This additional security check is supposed to make sure that your data directory cannot be read by some user accidentally. Therefore an explicit permission change is definitely an advantage from a security point of view (better safe than sorry). Now that we have all the pieces in place, we can start the replay process by starting PostgreSQL: iMac:target_directoryhs$ pg_ctl –D /target_directory startserver startingLOG: database system was interrupted; last known up at 2013-03-1018:04:29 CETLOG: creating missing WAL directory "pg_xlog/archive_status"LOG: starting point-in-time recovery to 2013-10-10 13:43:12+02LOG: restored log file "000000010000000000000006" from archiveLOG: redo starts at 0/6000020LOG: consistent recovery state reached at 0/60000B8LOG: restored log file "000000010000000000000007" from archiveLOG: restored log file "000000010000000000000008" from archiveLOG: restored log file "000000010000000000000009" from archiveLOG: restored log file "00000001000000000000000A" from archivecp: /tmp/archive/00000001000000000000000B: No such file ordirectoryLOG: could not open file "pg_xlog/00000001000000000000000B" (logfile 0, segment 11): No such file or directoryLOG: redo done at 0/AD5CE40LOG: last completed transaction was at log time 2013-03-1018:05:33.852992+01LOG: restored log file "00000001000000000000000A" from archivecp: /tmp/archive/00000002.history: No such file or directoryLOG: selected new timeline ID: 2cp: /tmp/archive/00000001.history: No such file or directoryLOG: archive recovery completeLOG: database system is ready to accept connectionsLOG: autovacuum launcher started The amount of log produced by the database tells us everything we need to know about the restore process and it is definitely worth investigating this information in detail. The first line indicates that PostgreSQL has found out that it has been interrupted and that it has to start recovery. From the database instance point of view, a base backup looks more or less like a crash needing some instant care by replaying XLOG; this is precisely what we want. The next couple of lines (restored log file ...) indicate that we are replaying one XLOG file after the other that have been created since the base backup. It is worth mentioning that the replay process starts at the sixth file. The base backup knows where to start, so PostgreSQL will automatically look for the right XLOG file. The message displayed after PostgreSQL reaches the sixth file (consistent recovery state reached at 0/60000B8) is of importance. PostgreSQL states that it has reached a consistent state. This is important. The reason is that the data files inside a base backup are actually broken by definition, but, the data files are not broken beyond repair. As long as we have enough XLOG to recover, we are very well off. If you cannot reach a consistent state, your database instance will not be usable and your recovery cannot work without providing additional XLOG. Practically speaking, not being able to reach a consistent state usually indicates a problem somewhere in your archiving process and your system setup. If everything up to now has been working properly, there is no reason not to reach a consistent state. Once we have reached a consistent state, one file after the other will be replayed successfully until the system finally looks for the 00000001000000000000000B file. The problem is that this file has not been created by the source database instance. Logically, an error pops up. Not finding the last file is absolutely normal; this type of error is expected if the recovery_target_time does not ask PostgreSQL to stop recovery before it reaches the end of the XLOG stream. Don't worry, your system is actually fine. You have successfully replayed everything to the file showing up exactly before the error message. As soon as all the XLOG has been consumed and the error message discussed earlier has been issued, PostgreSQL reports the last transaction it was able or supposed to replay, and starts up. You have a fully recovered database instance now and you can connect to the database instantly. As soon as the recovery has ended, recovery.conf will be renamed by PostgreSQL to recovery.done to make sure that it does not do any harm when the new instance is restarted later on at some point. More sophisticated positioning in the XLOG Up to now, we have recovered a database up to the very latest moment available in our 16 MB chunks of transaction log. We have also seen that you can define the desired recovery timestamp. But the question now is: How do you know which point in time to perform the recovery to? Just imagine somebody has deleted a table during the day. What if you cannot easily determine the recovery timestamp right away? What if you want to recover to a certain transaction? recovery.conf has all you need. If you want to replay until a certain transaction, you can refer to recovery_target_xid. Just specify the transaction you need and configure recovery_target_inclusive to include this very specific transaction or not. Using this setting is technically easy but as mentioned before, it is not easy by far to find the right position to replay to. In a typical setup, the best way to find a reasonable point to stop recovery is to use pause_at_recovery_target. If this is set to true, PostgreSQL will not automatically turn into a productive instance if the recovery point has been reached. Instead, it will wait for further instructions from the database administrator. This is especially useful if you don't know exactly how far to replay. You can replay, log in, see how far the database is, change to the next target time, and continue replaying in small steps. You have to set hot_standby = on in postgresql.conf to allow reading during recovery. Resuming recovery after PostgreSQL has paused can be done by calling a simple SQL statement: SELECT pg_xlog_replay_resume(). It will make the instance move to the next position you have set in recovery.conf. Once you have found the right place, you can set the pause_at_recovery_target back to false and call pg_xlog_replay_resume. Alternatively, you can simply utilize pg_ctl –D ... promote to stop recovery and make the instance operational. Was this explanation too complicated? Let us boil it down to a simple list: Add a restore_command to the recovery.conf file. Add recovery_target_time to the recovery.conf file. Set pause_at_recovery_target to true in the recovery.conf file. Set hot_standby to on in postgresql.conf file. Start the instance to be recovered. Connect to the instance once it has reached a consistent state and as soon as it stops recovering. Check if you are already where you want to be. If you are not: Change recovery_target_time. Run SELECT pg_xlog_replay_resume(). Check again and repeat this section if it is necessary. Keep in mind that once recovery has finished and once PostgreSQL has started up as a normal database instance, there is (as of 9.2) no way to replay XLOG later on. Instead of going through this process, you can of course always use filesystem snapshots. A filesystem snapshot will always work with PostgreSQL because when you restart a snapshotted database instance, it will simply believe that it had crashed before and recover normally. Cleaning up the XLOG on the way Once you have configured archiving, you have to store the XLOG being created by the source server. Logically, this cannot happen forever. At some point, you really have to get rid of this XLOG; it is essential to have a sane and sustainable cleanup policy for your files. Keep in mind, however, that you must keep enough XLOG so that you can always perform recovery from the latest base backup. But if you are certain that a specific base backup is not needed anymore, you can safely clean out all the XLOG that is older than the base backup you want to keep. How can an administrator figure out what to delete? The best method is to simply take a look at your archive directory: 000000010000000000000005000000010000000000000006000000010000000000000006.00000020.backup000000010000000000000007000000010000000000000008 Check out the filename in the middle of the listing. The .backup file has been created by the base backup. It contains some information about the way the base backup has been made and tells the system where to continue replaying the XLOG. If the backup file belongs to the oldest base backup you need to keep around, you can safely erase all the XLOG lower than file number 6; in this case, file number 5 could be safely deleted. In our case, 000000010000000000000006.00000020.backup contains the following information: START WAL LOCATION: 0/6000020 (file 000000010000000000000006)STOP WAL LOCATION: 0/60000E0 (file 000000010000000000000006)CHECKPOINT LOCATION: 0/6000058BACKUP METHOD: streamedBACKUP FROM: masterSTART TIME: 2013-03-10 18:04:29 CETLABEL: pg_basebackup base backupSTOP TIME: 2013-03-10 18:04:30 CET The .backup file will also provide you with relevant information such as the time the base backup has been made. It is plain there and so it should be easy for ordinary users to read this information. As an alternative to deleting all the XLOG files at one point, it is also possible to clean them up during replay. One way is to hide an rm command inside your restore_command. While this is technically possible, it is not necessarily wise to do so (what if you want to recover again?). Also, you can add the recovery_end_command command to your recovery.conf file. The goal of recovery_end_command is to allow you to automatically trigger some action as soon as the recovery ends. Again, PostgreSQL will call a script doing precisely what you want. You can easily abuse this setting to clean up the old XLOG when the database declares itself active. Switching the XLOG files If you are going for an XLOG file-based recovery, you have seen that one XLOG will be archived every 16 MB. What would happen if you never manage to create 16 MB of changes? What if you are a small supermarket, which just makes 50 sales a day? Your system will never manage to fill up 16 MB in time. However, if your system crashes, the potential data loss can be seen as the amount of data in your last unfinished XLOG file. Maybe this is not good enough for you. A postgresql.conf setting on the source database might help. The archive_timeout tells PostgreSQL to create a new XLOG file at least every x seconds. So, if you are this little supermarket, you can ask the database to create a new XLOG file every day shortly before you are heading for home. In this case, you can be sure that the data of the day will safely be on your backup device already. It is also possible to make PostgreSQL switch to the next XLOG file by hand. A procedure named pg_switch_xlog() is provided by the server to do the job: test=# SELECT pg_switch_xlog();pg_switch_xlog----------------0/17C0EF8(1 row) You might want to call this procedure when some important patch job has finished or if you want to make sure that a certain chunk of data is safely in your XLOG archive. Summary In this article, you have learned about Point-In-Time-Recovery, which is a safe and easy way to restore your PostgreSQL database to any desired point in time. PITR will help you to implement better backup policies and make your setups more robust. Resources for Article: Further resources on this subject: Introduction to PostgreSQL 9 [Article] PostgreSQL: Tips and Tricks [Article] PostgreSQL 9: Reliable Controller and Disk Setup [Article]
Read more
  • 0
  • 0
  • 5261

article-image-understanding-big-picture
Packt
04 Sep 2013
7 min read
Save for later

Understanding the big picture

Packt
04 Sep 2013
7 min read
(For more resources related to this topic, see here.) So we've got this thing for authentication and authorization. Let's see who is responsible and what for. There is an AccessDecisionManager, which, as the name suggests, is responsible for deciding whether we can access something or not; if not, an AccessDeniedException or InsufficientAuthenticationException is thrown. AuthenticationManager is another crucial interface. It is responsible for confirming who we are. Both are just interfaces, so we can swap our own implementations if we like. In a web application, the job of talking with these two components and the user is handled by a web filter called DelegatingFilterProxy, which is decomposed into several small filters. Each one is responsible for a different thing, so we can turn them on, off, or put our own filters in between and mess with them anyway we like. These are quite important, and we will dig into them later. For the big picture, all we need to know is that these filters take care of all the talking, redirect the user to the login page (or an access-denied page), and save the current user details in an HTTPSession. Well, the last part, while true, is a bit misleading. User details are kept in a SecurityContext object, which we can get a hold of by calling SecurityContextHolder.getContext(), and which in the end is stored in HTTPSession by our filters. But we had promised a big picture, not the gory details, so here it is: Quite simple, right? If we have an authentication protocol without login and password, it works in a similar way. We just switch one of the filters, or the authentication manager, to a different implementation. If we don't have a web application, we just need to do the talking ourselves. But this is all for web resources (URLs). What is much more interesting and useful is securing calls to methods. It looks, for example, like this: @PreAuthorize(["isAuthenticated() and hasRole('ROLE_ADMIN')"])public void somethingOnlyAdminCanDo() {} Here, we decided that somethingOnlyAdminCanDo will be protected by our AccessDecisionManager and that the user must be authenticated (not anonymous) and has to have an admin role. Can a user be anonymous and have an admin role at the same time? In theory, yes, but it would not make any sense. Because it's much cheaper to check if he is authenticated and stop right there. We see a bit of optimization in here. We could drop the isAuthenticated() method and the behavior wouldn't change. We can put this kind of annotation on any Java method, but our configuration and mechanism to fire up the security will depend on the type of objects we are trying to protect. For objects declared as Spring beans (which is a short name for anything defined in our Inversion of Control (IoC) configuration, either via XML or annotations), we don't need to do much. Spring will just create proxies (dynamic classes) that take over calls to our secured methods and fire up AccessDecisionManager before passing the call to the object we really wanted to call. For objects outside of the IoC container (anything created with new or just code not defined in Spring context), we can use the power of Aspect Oriented Programming (AOP) to get the same effect. If you don't know what AOP is, don't worry. It's just a bit of magic at the classloader and bytecode level. For now, the only important thing is that it works basically in the same way. This is depicted as follows: We can do much more than this, as we'll see next, but these are the basics. So, how does the AccessDecisionManager decide whether we can access something or not? Imagine a council of very old Jedi masters sitting around a fire. They decide whether or not you are permitted to call a secured method or access a web resource. Each of these masters makes a decision or abstains. Each of them can consult additional information (not only who you are and what you want to do, but every aspect of the situation). In Spring Security, those smart people are called AccessDecisionVoters, and each of them has one vote. The council can be organized in many different ways. It has one voice, and so it may make the decision based on a majority of votes. It may be veto-based, where everything is allowed unless someone disagrees. Or it may need everyone to agree to grant access, otherwise access is denied. The council is the AccessDecisionManager, and we have three implementations previously mentioned out of the box. We can also decide who's in the council and who is not. This is probably the most important decision we can make, because this will decide the security model that we will use in our application. Let's talk about the most popular counselors (implementations of AccessDecisionVoter). Model based on roles (RoleVoter): This guy makes his decision based on the role of the user and the required role for the resource/method. So if we write @PreAuthorize("hasRole('ROLE_ADMIN')"), you better be a damn admin or you'll get a no-no from this guy. Model based on entity access control permissions (AclEntryVoter): This guy doesn't worry about roles. He is much more than that. Acl stands for Access Control List, which represents a list of permissions. Every user has a list of permissions, possibly for every domain object (usually an object in the database), that you want to secure. So, for example, if we have a bank application, the supervisor can give Frank access to a single specific customer (say, ACME—A Company that Makes Everything), which is represented as an entity in the database and as an object in our system. No other employee will be able to do anything to that customer unless the supervisor grants that person the same permission as Frank. This is probably the most scrutinous voter we would ever use. Our customer can have a very detailed configuration with him/her. On the other hand, this is also the most cumbersome, as we need to create a usable graphical interface to set permissions for every user and every domain object. While we have done this a few times, most of our customers wanted a simpler approach, and even those who started with a graphical user interface to configure everything asked for a simplified version based on business rules, at the end of the project. If your customer describes his security needs in terms of rules such as "Frank can edit every customer he has created but he cannot do anything other than view other customers", it means it's time for PreInvocationAuthorizationAdviceVoter. Business rules model (PreInvocationAuthorizationAdviceVoter): This is usually used when you want to implement static business rules in the application. This goes like "if I've written a blog post, I can change it later, but others can only comment" and "if a friend asked me to help him write the blog post, I can do that, because I'm his friend". Most of these things are also possible to implement with ACLs, but would be very cumbersome. This is our favorite voter. With it, it's very easy to write, test, and change the security restrictions, because instead of writing every possible relation in the database (as with ACL voter) or having only dumb roles, we write our security logic in plain old Java classes. Great stuff and most useful, once you see how it works. Did we mention that this is a council? Yes we did. The result of this is that we can mix any voters we want and choose any council organization we like. We can have all three voters previously mentioned and allow access if any of them says "yes". There are even more voters. And we can write new ones ourselves. Do you feel the power of the Jedi council already? Do you feel the power of the Jedi council already? Summary This section provides an overview of authentication and authorization, which are the principles of Spring security. Resources for Article : Further resources on this subject: Migration to Spring Security 3 [Article] Getting Started with Spring Security [Article] So, what is Spring for Android? [Article]
Read more
  • 0
  • 0
  • 8546

article-image-managing-adobe-connect-meeting-room
Packt
04 Sep 2013
6 min read
Save for later

Managing Adobe Connect Meeting Room

Packt
04 Sep 2013
6 min read
(For more resources related to this topic, see here.) The Meeting Information page In order to get to the Meeting Information page, you will first need to navigate to the Meeting List page by following these steps: Log in to the Connect application. Click on the Meetings tab in the Home Page main menu. When you access the Meetings page, the My Meetings link is opened by default and a view is set on the Meeting List tab. You will find the meeting that is listed on this page as shown in the following screenshot: By clicking on the Cookbook Meeting option in the Name column (marked with a red outline), you will be presented with the Meeting Information page. In the section titled Meeting Information, you can examine various pieces of information about the selected meeting. On this page, you can review Name, Summary, Start Time, Duration, Number of users in room(that are currently present in the meeting room), URL, Language(selected), and the Access rights of the meeting. The two most important fields are marked with a red outline in the previous screenshot. The first one is the link to the meeting URL and the second is the Enter Meeting Room button. You can join the selected meeting room by clicking on any of these two options. In the upper portion of this page, you will notice the navigation bar with the following links: Meeting Information Edit Information Edit Participants Invitations Uploaded Content Recordings Reports By selecting any of these links, you will open pages associated with them. Our main focus of this article will be on the functionalities of these pages. Since we have explained the Meeting Information page, we can proceed to the Edit Information page. The Edit Information page The Edit Information page is very similar to the Enter Meeting Information page. We will briefly inform you about the meeting settings, which you can edit on this page. These settings are: Name Summary Start time Duration Language Access Audio conference settings Any changes made on this page are preserved by clicking on the Save button that you will find at very bottom of this page. Changes will not affect participants who are already logged in to the room, except changes to the Audio Conference settings. Next to the Save button, you will find the Cancel button. Any changes made on the Edit Information page, which are not already saved will be reverted by clicking on the Cancel button. The Edit Participants page After the Edit Information page, it's time for us to access the next page by clicking on the Edit Participants link in the navigation bar. This link will take you to the Select Participants page. In addition to the already described features, we will introduce you to a couple more functionalities that will help you to add participants, change their roles, or remove them from the meeting. Example 1 – changing roles In this example, we will change the role of the administrators group from participant to presenter by using the Search button. This feature is of great help when there are a large number of Connect users that are already added as meeting participants. In order to do so, you will need to follow the steps listed: In the Current Participants For Cookbook Meeting table on the right-hand side, click on the Search button located in the lower-left corner of the table. When you click on the Search button, a text field for instant search will be displayed. In the text field, enter the name of the Administrators group or part of the group name (the auto-complete function should recognize the name of the present group). When the group is present in the table, select it. Click on the Set User Role button. Select new role for this group in the menu. For the purpose of this example, we will select the Presenter role. By completing this action, you will grant Presenter privileges in the Cookbook Meeting table to all the administrators as shown in the following screenshot: Example 2 – removing a user In this example, we will show you how to remove a specific user from the selected meeting. For the purpose of this exercise, we will remove the Administrators group from the Participants list. In order to complete this action, please follow the given steps: Select Administrators in the Current Participants For Cookbook Meeting table. Click on the Remove button. Now, all the members of this group will be excluded from the meeting, and Administrators should not be present in the list. Example 3 – adding a specific user This example will demonstrate how to add a specific user from any group. For example, we will add a user from the Authors group to the Current Participants list. In the Available users and Groups table, double-click on the Authors group. This action will change the user interface of this table and list all the users that belong to the Authors group. Please note that table header is now changed to Authors. Select a specific user and click on the Add button. This will add the selected user from the Authors group to the Current Participants For Cookbook Meeting table. One thing that we would like to mention here is the ability to perform multiple selections in both the Available Users and Groups and Current Participants For Cookbook Meeting tables. To enable multiple selection functionality, select a specific user and group by clicking and selecting Ctrl and Shift on the keyboard at the same time. By demonstrating these examples, we reviewed the Edit Participant link functionalities. Summary In this article, we learned how to master all functionalities on how to edit different settings for already existing meetings. We covered the following topics: The Meeting information page The Managing Edit information page The Managing Edit participants page Resources for Article: Further resources on this subject: Top features you'll want to know about [Article] Remotely Preview and test mobile web pages on actual devices with Adobe Edge Inspect [Article] Exporting SAP BusinessObjects Dashboards into Different Environments [Article]
Read more
  • 0
  • 0
  • 1761
article-image-audio-playback
Packt
04 Sep 2013
17 min read
Save for later

Audio Playback

Packt
04 Sep 2013
17 min read
(For more resources related to this topic, see here.) Understanding FMOD One of the main reasons why I chose FMOD for this book is that it contains two separate APIs—the FMOD Ex Programmer's API, for low-level audio playback, and FMOD Designer, for high-level data-driven audio. This will allow us to cover game audio programming at different levels of abstraction without having to use entirely different technologies. Besides that reason, FMOD is also an excellent piece of software, with several advantages to game developers: License: It is free for non-commercial use, and has reasonable licenses for commercial projects. Cross-platform: It works across an impressive number of platforms. You can run it on Windows, Mac, Linux, Android, iOS, and on most of the modern video game consoles by Sony, Microsoft, and Nintendo. Supported formats: It has native support for a huge range of audio file formats, which saves you the trouble of having to include other external libraries and decoders. Programming languages: Not only can you use FMOD with C and C++, there are also bindings available for other programming languages, such as C# and Python. Popularity: It is extremely popular, being widely considered as the industry standard nowadays. It was used in games such as BioShock, Crysis, Diablo 3, Guitar Hero, Start Craft II, and World of Warcraft. It is also used to power several popular game engines, such as Unity3D and CryEngine. Features: It is packed with features, covering everything from simple audio playback, streaming and 3D sound, to interactive music, DSP effects and low-level audio programming. Installing FMOD Ex Programmer's API Installing a C++ library can be a bit daunting at first. The good side is that once you have done it for the first time, the process is usually the same for every other library. Here are the steps that you should follow if you are using Microsoft Visual Studio: Download the FMOD Ex Programmer's API from http://www.fmod.org and install it to a folder that you can remember, such as C:FMOD. Create a new empty project, and add at least one .cpp file to it. Then, right-click on the project node on the Solution Explorer , and select Properties from the list. For all the steps that follow, make sure that the Configuration option is set to All Configurations . Navigate to C/C++ | General , and add C:FMODapiinc to the list of Additional Include Directories (entries are separated by semicolons). Navigate to Linker | General , and add C:FMODapilib to the list of Additional Library Directories . Navigate to Linker | Input , and add fmodex_vc.lib to the list of Additional Dependencies . Navigate to Build Events | Post-Build Event , and add xcopy /y "C:FMODapifmodex.dll" "$(OutDir)" to the Command Lin e list. Include the <fmod.hpp> header file from your code. Creating and managing the audio system Everything that happens inside FMOD is managed by a class named FMOD::System, which we must start by instantiating with the FMOD::Syste m_Create() function: FMOD::System* system; FMOD::System_Create(&system); Notice that the function returns the system object through a parameter. You will see this pattern every time one of the FMOD functions needs to return a value, because they all reserve the regular return value for an error code. We will discuss error checking in a bit, but for now let us get the audio engine up and running. Now that we have a system object instantiated, we also need to initialize it by calling the init() method: system->init(100, FMOD_INIT_NORMAL, 0); The first parameter specifies the maximum number of channels to allocate. This controls how many sounds you are able to play simultaneously. You can choose any number for this parameter because the system performs some clever priority management behind the scenes and distributes the channels using the available resources. The second and third parameters customize the initialization process, and you can usually leave them as shown in the example. Many features that we will use work properly only if we update the system object every frame. This is done by calling the update() method from inside your game loop: system->update(); You should also remember to shutdown the system object before your game ends, so that it can dispose of all resources. This is done by calling the release() method: system->release(); Loading and streaming audio files One of the greatest things about FMOD is that you can load virtually any audio file format with a single method call. To load an audio file into memory, use the createSound() method: FMOD::Sound* sound; system->createSound("sfx.wav", FMOD_DEFAULT, 0, &sound); To stream an audio file from disk without having to store it in memory, use the createStream() method: FMOD::Sound* stream; system->createStream("song.ogg", FMOD_DEFAULT, 0, &stream); Both methods take the path of the audio file as the first parameter, and return a pointer to an FMOD::Sound object through the fourth parameter, which you can use to play the sound. The paths in the previous examples are relative to the application path. If you are running these examples in Visual Studio, make sure that you copy the audio files into the output folder (for example, using a post-build event such as xcopy /y "$(ProjectDir)*.ogg" "$(OutDir)"). The choice between loading and streaming is mostly a tradeoff between memory and processing power. When you load an audio file, all of its data is uncompressed and stored in memory, which can take up a lot of space, but the computer can play it without much effort. Streaming, on the other hand, barely uses any memory, but the computer has to access the disk constantly, and decode the audio data on the fly. Another difference (in FMOD at least) is that when you stream a sound, you can only have one instance of it playing at any time. This limitation exists because there is only one decode buffer per stream. Therefore, for sound effects that have to be played multiple times simultaneously, you have to either load them into memory, or open multiple concurrent streams. As a rule of thumb, streaming is great for music tracks, voice cues, and ambient tracks, while most sound effects should be loaded into memory. The second and third parameters allow us to customize the behavior of the sound. There are many different options available, but the following list summarizes the ones we will be using the most. Using FMOD_DEFAULT is equivalent to combining the first option of each of these categories: FMOD_LOOP_OFF and FMOD_LOOP_NORMAL: These modes control whether the sound should only play once, or loop once it reaches the end FMOD_HARDWARE and FMOD_SOFTWARE: These modes control whether the sound should be mixed in hardware (better performance) or software (more features) FMOD_2D and FMOD_3D: These modes control whether to use 3D sound We can combine multiple modes using the bitwise OR operator (for instance, FMOD_DEFAULT | FMOD_LOOP_NORMAL | FMOD_SOFTWARE). We can also tell the system to stream a sound even when we are using the createSound() method, by setting the FMOD_CREATESTREAM flag. In fact, the createStream() method is simply a shortcut for this. When we do not need a sound anymore (or at the end of the game) we should dispose of it by calling the release() method of the sound object. We should always release the sounds we create, regardless of the audio system also being released. sound->release(); Playing sounds With the sounds loaded into memory or prepared for streaming, all that is left is telling the system to play them using the playSound() method: FMOD::Channel* channel; system->playSound(FMOD_CHANNEL_FREE, sound, false, &channel); The first parameter selects in which channel the sound will play. You should usually let FMOD handle it automatically, by passing FMOD_CHANNEL_FREE as the parameter. The second parameter is a pointer to the FMOD::Sound object that you want to play. The third parameter controls whether the sound should start in a paused state, giving you a chance to modify some of its properties without the changes being audible. If you set this to true, you will also need to use the next parameter so that you can unpause it later. The fourth parameter is an output parameter that returns a pointer to the FMOD::Channel object in which the sound will play. You can use this handle to control the sound in multiple ways, which will be the main topic of the next chapter. You can ignore this last parameter if you do not need any control over the sound, and simply pass in 0 in its place. This can be useful for non-lopping one-shot sounds. system->playSound(FMOD_CHANNEL_FREE, sound, false, 0); Checking for errors So far, we have assumed that every operation will always work without errors. However, in a real scenario, there is room for a lot to go wrong. For example, we could try to load an audio file that does not exist. In order to report errors, every function and method in FMOD has a return value of type FMOD_RESULT, which will only be equal to FMOD_OK if everything went right. It is up to the user to check this value and react accordingly: FMOD_RESULT result = system->init(100, FMOD_INIT_NORMAL, 0); if (result != FMOD_OK) { // There was an error, do something about it } For starters, it would be useful to know what the error was. However, since FMOD_RESULT is an enumeration, you will only see a number if you try to print it. Fortunately, there is a function called FMOD_ErrorString() inside the fmod_errors.h header file which will give you a complete description of the error. You might also want to create a helper function to simplify the error checking process. For instance, the following function will check for errors, print a description of the error to the standard output, and exit the application: #include <iostream> #include <fmod_errors.h> void ExitOnError(FMOD_RESULT result) { if (result != FMOD_OK) { std::cout << FMOD_ErrorString(result) << std::endl; exit(-1); } } You could then use that function to check for any critical errors that should cause the application to abort: ExitOnError(system->init(100, FMOD_INIT_NORMAL, 0)); The initialization process described earlier also assumes that everything will go as planned, but a real game should be prepared to deal with any errors. Fortunately, there is a template provided in the FMOD documentation which shows you how to write a robust initialization sequence. It is a bit long to cover here, so I urge you to refer to the file named Getting started with FMOD for Windows.pdf inside the documentation folder for more information. For clarity, all of the code examples will continue to be presented without error checking, but you should always check for errors in a real project. Project 1 building a simple audio manager In this project, we will be creating a SimpleAudioManager class that combines everything that was covered in this chapter. Creating a wrapper for an underlying system that only exposes the operations that we need is known as the façade design pattern , and is very useful in order to keep things nice and simple. Since we have not seen how to manipulate sound yet, do not expect this class to be powerful enough to be used in a complex game. Its main purpose will be to let you load and play one-shot sound effects with very little code (which could in fact be enough for very simple games). It will also free you from the responsibility of dealing with sound objects directly (and having to release them) by allowing you to refer to any loaded sound by its filename. The following is an example of how to use the class: SimpleAudioManager audio; audio.Load("explosion.wav"); audio.Play("explosion.wav"); From an educational point of view, what is perhaps even more important is that you use this exercise as a way to get some ideas on how to adapt the technology to your needs. It will also form the basis of the next chapters in the book, where we will build systems that are more complex. Class definition Let us start by examining the class definition: #include <string> #include <map> #include <fmod.hpp> typedef std::map<std::string, FMOD::Sound*> SoundMap; class SimpleAudioManager { public: SimpleAudioManager(); ~SimpleAudioManager(); void Update(float elapsed); void Load(const std::string& path); void Stream(const std::string& path); void Play(const std::string& path); private: void LoadOrStream(const std::string& path, bool stream); FMOD::System* system; SoundMap sounds; }; From browsing through the list of public class members, it should be easy to deduce what it is capable of doing: The class can load audio files (given a path) using the Load() method The class can stream audio files (given a path) using the Stream() method The class can play audio files (given a path) using the Play() method (granted that they have been previously loaded or streamed) There is also an Update() method and a constructor/destructor pair to manage the sound system The private class members, on the other hand, can tell us a lot about the inner workings of the class: At the core of the class is an instance of FMOD::System responsible for driving the entire sound engine. The class initializes the sound system on the constructor, and releases it on the destructor. Sounds are stored inside an associative container, which allows us to search for a sound given its file path. For this purpose, we will be relying on one of the C++ Standard Template Library (STL ) associative containers, the std::map class, as well as the std::string class for storing the keys. Looking up a string key is a bit inefficient (compared to an integer, for example), but it should be fast enough for our needs. An advantage of having all the sounds stored on a single container is that we can easily iterate over them and release them from the class destructor. Since the code for loading and streaming audio file is almost the same, the common functionality has been extracted into a private method called LoadOrStream(), to which Load() and Stream() delegate all of the work. This prevents us from repeating the code needlessly. Initialization and destruction Now, let us walk through the implementation of each of these methods. First we have the class constructor, which is extremely simple, as the only thing that it needs to do is initialize the system object. SimpleAudioManager::SimpleAudioManager() { FMOD::System_Create(&system); system->init(100, FMOD_INIT_NORMAL, 0); } Updating is even simpler, consisting of a single method call: void SimpleAudioManager::Update(float elapsed) { system->update(); } The destructor, on the other hand, needs to take care of releasing the system object, as well as all the sound objects that were created. This process is not that complicated though. First, we iterate over the map of sounds, releasing each one in turn, and clearing the map at the end. The syntax might seem a bit strange if you have never used an STL iterator before, but all that it means is to start at the beginning of the container, and keep advancing until we reach its end. Then we finish off by releasing the system object as usual. SimpleAudioManager::~SimpleAudioManager() { // Release every sound object and clear the map SoundMap::iterator iter; for (iter = sounds.begin(); iter != sounds.end(); ++iter) iter->second->release(); sounds.clear(); // Release the system object system->release(); system = 0; } Loading or streaming sounds Next in line are the Load() and Stream() methods, but let us examine the private LoadOrStream() method first. This method takes the path of the audio file as a parameter, and checks if it has already been loaded (by querying the sound map). If the sound has already been loaded there is no need to do it again, so the method returns. Otherwise, the file is loaded (or streamed, depending on the value of the second parameter) and stored in the sound map under the appropriate key. void SimpleAudioManager::LoadOrStream(const std::string& path, bool stream) { // Ignore call if sound is already loaded if (sounds.find(path) != sounds.end()) return; // Load (or stream) file into a sound object FMOD::Sound* sound; if (stream) system->createStream(path.c_str(), FMOD_DEFAULT, 0, &sound); else system->createSound(path.c_str(), FMOD_DEFAULT, 0, &sound); // Store the sound object in the map using the path as key sounds.insert(std::make_pair(path, sound)); } With the previous method in place, both the Load() and the Stream() methods can be trivially implemented as follows: void SimpleAudioManager::Load(const std::string& path) { LoadOrStream(path, false); } void SimpleAudioManager::Stream(const std::string& path) { LoadOrStream(path, true); } Playing sounds Finally, there is the Play() method, which works the other way around. It starts by checking if the sound has already been loaded, and does nothing if the sound is not found on the map. Otherwise, the sound is played using the default parameters. void SimpleAudioManager::Play(const std::string& path) { // Search for a matching sound in the map SoundMap::iterator sound = sounds.find(path); // Ignore call if no sound was found if (sound == sounds.end()) return; // Otherwise play the sound system->playSound(FMOD_CHANNEL_FREE, sound->second, false, 0); } We could have tried to automatically load the sound in the case when it was not found. In general, this is not a good idea, because loading a sound is a costly operation, and we do not want that happening during a critical gameplay section where it could slow the game down. Instead, we should stick to having separate load and play operations. A note about the code samples Although this is a book about audio, all the samples need an environment to run on. In order to keep the audio portion of the samples as clear as possible, we will also be using the Simple and Fast Multimedia Library 2.0 (SFML ) (http://www.sfml-dev.org). This library can very easily take care of all the miscellaneous tasks, such as window creation, timing, graphics, and user input, which you will find in any game. For example, here is a complete sample using SFML and the SimpleAudioManager class. It creates a new window, loads a sound, runs a game loop at 60 frames per second, and plays the sound whenever the user presses the space key. #include <SFML/Window.hpp> #include "SimpleAudioManager.h" int main() { sf::Window window(sf::VideoMode(320, 240), "AudioPlayback"); sf::Clock clock; // Place your initialization logic here SimpleAudioManager audio; audio.Load("explosion.wav"); // Start the game loop while (window.isOpen()) { // Only run approx 60 times per second float elapsed = clock.getElapsedTime().asSeconds(); if (elapsed < 1.0f / 60.0f) continue; clock.restart(); sf::Event event; while (window.pollEvent(event)) { // Handle window events if (event.type == sf::Event::Closed) window.close(); // Handle user input if (event.type == sf::Event::KeyPressed && event.key.code == sf::Keyboard::Space) audio.Play("explosion.wav"); } // Place your update and draw logic here audio.Update(elapsed); } // Place your shutdown logic here return 0; } Summary In this article, we have seen some of the advantages of using the FMOD audio engine. We saw how to install the FMOD Ex Programmer's API in Visual Studio, how to initialize, manage, and release the FMOD sound system, how to load or stream an audio file of any type from disk, how to play a sound that has been previously loaded by FMOD, how to check for errors in every FMOD function, and how to create a simple audio manager that encapsulates the act of loading and playing audio files behind a simple interface. Resources for Article : Further resources on this subject: Using SpriteFonts in a Board-based Game with XNA 4.0 [Article] HTML5 Games Development: Using Local Storage to Store Game Data [Article] Making Money with Your Game [Article]
Read more
  • 0
  • 0
  • 6656

article-image-important-features-mockito
Packt
04 Sep 2013
4 min read
Save for later

Important features of Mockito

Packt
04 Sep 2013
4 min read
Reducing boilerplate code with annotations Mockito allows the use of annotations to reduce the lines of test code in order to increase the readability of tests. Let's take into consideration some of the tests that we have seen in previous examples. Removing boilerplate code by using the MockitoJUnitRunner The shouldCalculateTotalWaitingTimeAndAssertTheArgumentsOnMockUsingArgumentCaptor from Verifying behavior (including argument capturing, verifying call order and working with asynchronous code) section, can be rewritten as follows, using Mockito annotations, and the @RunWith(MockitoJUnitRunner.class) JUnit runner: @RunWith(MockitoJUnitRunner.class) public class _07ReduceBoilerplateCodeWithAnnotationsWithRunner { @Mock KitchenService kitchenServiceMock; @Captor ArgumentCaptor mealArgumentCaptor; @InjectMocks WaiterImpl objectUnderTest; @Test public void shouldCalculateTotalWaitingTimeAndAssert TheArgumentsOnMockUsingArgumentCaptor() throws Exception { //given final int mealPreparationTime = 10; when(kitchenServiceMock.calculate PreparationTime(any(Meal.class))).thenReturn(mealPreparationTime); //when int waitingTime = objectUnderTest.calculate TotalWaitingTime(createSampleMeals ContainingVegetarianFirstCourse()); //then assertThat(waitingTime, is(mealPreparationTime)); verify(kitchenServiceMock).calculatePreparation Time(mealArgumentCaptor.capture()); assertThat(mealArgumentCaptor.getValue(), is (VegetarianFirstCourse.class)); assertThat(mealArgumentCaptor.getAllValues().size(), is(1)); } private List createSampleMeals ContainingVegetarianFirstCourse() { List meals = new ArrayList(); meals.add(new VegetarianFirstCourse()); return meals; } } What happened here is that: All of the boilerplate code can be removed due to the fact that you are using the @RunWith(MockitoJUnitRunner.class) JUnit runner Mockito.mock(…) has been replaced with @Mock annotation You can provide additional parameters to the annotation, such as name, answer or extraInterfaces. The fieldname related to the annotated mock is referred to in any verification so it's easier to identify the mock ArgumentCaptor.forClass(…) is replaced with @Captor annotation. When using the @Captor annotation you avoid warnings related to complex generic types Thanks to the @InjectMocks annotation your object under test is initialized, proper constructor/setters are found and Mockito injects the appropriate mocks for you There is no explicit creation of the object under test You don't need to provide the mocks as arguments of the constructor Mockito @InjectMocksuses either constructor injection, property injection or setter injection Taking advantage of advanced mocks configuration Mockito gives you a possibility of providing different answers for your mocks. Let's focus more on that. Getting more information on NullPointerException Remember the Waiter's askTheCleaningServiceToCleanTheRestaurantMethod(): @Override public boolean askTheCleaningServiceToCleanTheRestaurant (TypeOfCleaningService typeOfCleaningService) { CleaningService cleaningService = cleaningServiceFactory.getMe ACleaningService(typeOfCleaningService); try{ cleaningService.cleanTheTables(); cleaningService.sendInformationAfterCleaning(); return SUCCESSFULLY_CLEANED_THE_RESTAURANT; }catch(CommunicationException communicationException){ System.err.println("An exception took place while trying to send info about cleaning the restaurant"); return FAILED_TO_CLEAN_THE_RESTAURANT; } } Let's assume that we want to test this function. We inject the CleaningServiceFactory as a mock but we forgot to stub the getMeACleaningService(…) method. Normally we would get a NullPointerException since, if the method is not stubbed, it will return null. But what will happen, if as an answer we would provide a RETURNS_SMART_NULLS answer? Let's take a look at the body of the following test: @Mock(answer = Answers.RETURNS_SMART_NULLS) CleaningServiceFactory cleaningServiceFactory; @InjectMocks WaiterImpl objectUnderTest; @Test public void shouldThrowSmartNullPointerExceptionWhenUsingUnstubbedMethod() { //given // Oops forgotten to stub the CleaningServiceFactory.getMeACle aningService(TypeOfCleaningService) method try { //when objectUnderTest.askTheCleaningServiceToCleanTheRestaurant( TypeOfCleaningService.VERY_FAST); fail(); } catch (SmartNullPointerException smartNullPointerException) { System.err.println("A SmartNullPointerException will be thrown here with a very precise information about the error [" + smartNullPointerException + "]"); } } What happened in the test is that: We create a mock with an answer RETURNS_SMART_NULLS of the CleaningServiceFactory The mock is injected to the WaiterImpl We do not stub the getMeACleaningService(…) of the CleaningServiceFactory The SmartNullPointerException will be thrown at the line containing the cleaningService.cleanTheTables() It will contain very detailed information about the reason for the exception to happen and where it happened In order to have the RETURNS_SMART_NULLS as the default answer (you wouldn't have to explicitly define the answer for your mock), you would have to create the class named MockitoConfiguration in a package org.mockito.configuration that either extends the DefaultMockitoConfiguration or implements the IMockitoConfiguration interface: package org.mockito.configuration; import org.mockito.internal.stubbing.defaultanswers.ReturnsSmartNulls; import org.mockito.stubbing.Answer; public class MockitoConfiguration extends DefaultMockitoConfiguration { public Answer<Object> getDefaultAnswer() { return new ReturnsSmartNulls(); } } Summary In this article we learned in detail about reducing the boilerplate code with annotations, and taking advantage of advanced mocks configuration, along with their code implementation. Resources for Article : Further resources on this subject: Testing your App [Article] Drools JBoss Rules 5.0 Flow (Part 2) [Article] Easily Writing SQL Queries with Spring Python [Article]
Read more
  • 0
  • 0
  • 11388
Modal Close icon
Modal Close icon