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-learning-nservicebus-preparing-failure
Packt
09 Feb 2015
19 min read
Save for later

Learning NServiceBus - Preparing for Failure

Packt
09 Feb 2015
19 min read
 In this article by David Boike, author of the book Learning NServiceBus Second Edition we will explore the tools that NServiceBus gives us to stare at failure in the face and laugh. We'll discuss error queues, automatic retries, and controlling how those retries occur. We'll also discuss how to deal with messages that may be transient and should not be retried in certain conditions. Lastly, we'll examine the difficulty of web service integrations that do not handle retries cleanly on their own. (For more resources related to this topic, see here.) Fault tolerance and transactional processing In order to understand the fault tolerance we gain from using NServiceBus, let's first consider what happens without it. Let's order something from a fictional website and watch what might happen to process that order. On our fictional website, we add Batman Begins to our shopping cart and then click on the Checkout button. While our cursor is spinning, the following process is happening: Our web request is transmitted to the web server. The web application knows it needs to make several database calls, so it creates a new transaction scope. Database Call 1 of 3: The shopping cart information is retrieved from the database. Database Call 2 of 3: An Order record is inserted. Database Call 3 of 3: We attempt to insert OrderLine records, but instead get Error Message: Transaction (Process ID 54) was deadlocked on lock resources with another process and has been chosen as the deadlock victim. Rerun the transaction. This exception causes the transaction to roll back. This process is shown in the following diagram:   Ugh! If you're using SQL Server and you've never seen this, you haven't been coding long enough. It never happens during development; there just isn't enough load. It's even possible that this won't occur during load testing. It will likely occur during heavy load at the worst possible time, for example, right after your big launch. So obviously, we should log the error, right? But then what happens to the order? Well that's gone, and your boss may not be happy about losing that revenue. And what about our user? They will likely get a nasty error message. We won't want to divulge the actual exception message, so they will get something like, "An unknown error has occurred. The system administrator has been notified. Please try again later." However, the likelihood that they want to trust their credit card information to a website that has already blown up in their face once is quite low. So how can we do better? Here's how this scenario could have happened with NServiceBus:   The web request is transmitted to the web server. We add the shopping cart identifier to an NServiceBus command and send it through the Bus. We redirect the user to a new page that displays the receipt, even though the order has not yet been processed. Elsewhere, an Order service is ready to start processing a new message: The service creates a new transaction scope, and receives the message within the transaction. Database Call 1 of 3: The shopping cart information is retrieved from the database. Database Call 2 of 3: An Order record is inserted. Database Call 3 of 3: Deadlock! The exception causes the database transaction to roll back. The transaction controlling the message also rolls back. The order is back in the queue. This is great news! The message is back in the queue, and by default, NServiceBus will automatically retry this message a few times. Generally, deadlocks are a temporary condition, and simply trying again is all that is needed. After all, the SQL Server exception says Rerun the transaction. Meanwhile, the user has no idea that there was ever a problem. It will just take a little longer (in the order of milliseconds or seconds) to process the order. Error queues and replay Whenever you talk about automatic retries in a messaging environment, you must invariably consider poison messages. A poison message is a message that cannot be immediately resolved by a retry because it will consistently result in an error. A deadlock is a transient error. We can reasonably expect deadlocks and other transient errors to resolve by themselves without any intervention. Poison messages, on the other hand, cannot resolve themselves. Sometimes, this is because of an extended outage. At other times, it is purely our fault—an exception we didn't catch or an input condition we didn't foresee. Automatic retries If we retry poison messages in perpetuity, they will create a blockage in our incoming queue of messages. They will retry over and over, and valid messages will get stuck behind them, unable to make it through. For this reason, we must set a reasonable limit on retries, and after failing too many times, poison messages must be removed from the processing queue and stored someplace else. NServiceBus handles all of this for us. By default, NServiceBus will try to process a message five times, after which it will move the message to an error queue, configured by the MessageForwardingInCaseOfFaultConfig configuration section: <MessageForwardingInCaseOfFaultConfigErrorQueue="error" /> It is in this error queue that messages will wait for administrative intervention. In fact, you can even specify a different server to collect these messages, which allows you to configure one central point in a system where you watch for and deal with all failures: <MessageForwardingInCaseOfFaultConfigErrorQueue="error@SERVER" /> As mentioned previously, five failed attempts form the default metric for a failed message, but this is configurable via the TransportConfig configuration section: <section name="TransportConfig" type="NServiceBus.Config.TransportConfig, NServiceBus.Core" /> ... <TransportConfig MaxRetries="3" /> You could also generate the TransportConfig section using the Add-NServiceBusTransportConfig PowerShell cmdlet. Keep two things in mind: Depending upon how you read it, MaxRetries can be a somewhat confusing name. What it really means is the total number of tries, so a value of 5 will result in the initial attempt plus 4 retries. This has the odd side effect that MaxRetries="0" is the same as MaxRetries="1". In both instances, the message would be attempted once. During development, you may want to limit retries to MaxRetries="1" so that a single error doesn't cause a nausea-inducing wall of red that flushes your console window's buffer, leaving you unable to scroll up to see what came before. You can then enable retries in production by deploying the endpoint with a different configuration. Replaying errors What happens to those messages unlucky enough to fail so many times that they are unceremoniously dumped in an error queue? "I thought you said that Alfred would never give up on us!" you cry. As it turns out, this is just a temporary holding pattern that enables the rest of the system to continue functioning, while the errant messages await some sort of intervention, which can be human or automated based on your own business rules. Let's say our message handler divides two numbers from the incoming message, and we forget to account for the possibility that one of those numbers might be zero and that dividing by zero is frowned upon. At this point, we need to fix the error somehow. Exactly what we do will depend upon your business requirements: If the messages were sent in an error, we can fix the code that was sending them. In this case, the messages in the error queue are junk and can be discarded. We can check the inputs on the message handler, detect the divide-by-zero condition, and make compensating actions. This may mean returning from the message handler, effectively discarding any divide-by-zero messages that are processed, or it may mean doing new work or sending new messages. In this case, we may want to replay the error messages after we have deployed the new code. We may want to fix both the sending and receiving side. Second-level retries Automatically retrying error messages and sending repeated errors to an error queue is a pretty good strategy to manage both transient errors, such as deadlocks, and poison messages, such as an unrecoverable exception. However, as it turns out, there is a gray area in between, which is best referred to as semi-transient errors. These include incidents such as a web service being down for a few seconds, or a database being temporarily offline. Even with a SQL Server failover cluster, the failover procedure can take upwards of a minute depending on its size and traffic levels. During a time like this, the automatic retries will be executed immediately and great hordes of messages might go to the error queue, requiring an administrator to take notice and return them to their source queues. But is this really necessary? As it turns out, it is not. NServiceBus contains a feature called Second-Level Retries (SLR) that will add additional sets of retries after a wait. By default, the SLR will add three additional retry sessions, with an additional wait of 10 seconds each time. By contrast, the original set of retries is commonly referred to as First-Level Retries (FLR). Let's track a message's full path to complete failure, assuming default settings: Attempt to process the message five times, then wait for 10 seconds Attempt to process the message five times, then wait for 20 seconds Attempt to process the message five times, then wait for 30 seconds Attempt to process the message five times, and then send the message to the error queue Remember that by using five retries, NServiceBus attempts to process the message five times on every pass. Using second-level retries, almost every message should be able to be processed unless it is definitely a poison message that can never be successfully processed. Be warned, however, that using SLR has its downsides too. The first is ignorance of transient errors. If an error never makes it to an error queue and we never manually check out the error logs, there's a chance we might miss it completely. For this reason, it is smart to always keep an eye on error logs. A random deadlock now and then is not a big deal, but if they happen all the time, it is probably still worth some work to improve the code so that the deadlock is not as frequent. An additional risk lies in the time to process a true poison message through all the retry levels. Not accounting for any time taken to process the message itself 20 times or to wait for other messages in the queue, the use of second-level retries with the default settings results in an entire minute of waiting before you see the message in an error queue. If your business stakeholders require the message to either succeed or fail in 30 seconds, then you cannot possibly meet those requirements. Due to the asynchronous nature of messaging, we should be careful never to assume that messages in a distributed system will arrive in any particular order. However, it is still good to note that the concept of retries exacerbates this problem. If Message A and then Message B are sent in order, and Message B succeeds immediately but Message A has to wait in an error queue for awhile, then they will most certainly be processed out of order. Luckily, second-level retries are completely configurable. The configuration element is shown here with the default settings: <section name="SecondLevelRetriesConfig" type="NServiceBus.Config.SecondLevelRetriesConfig,   NServiceBus.Core"/> ... <SecondLevelRetriesConfig Enabled="true"                          TimeIncrease="00:00:10"                          NumberOfRetries="3" /> You could also generate the SecondLevelRetriesConfig section using the Add-NServiceBus SecondLevelRetriesConfig PowerShell cmdlet. Keep in mind that you may want to disable second-level retries, like first-level retries, during development for convenience, and then enable them in production. Messages that expire Messages that lose their business value after a specific amount of time are an important consideration with respect to potential failures. Consider a weather reporting system that reports the current temperature every few minutes. How long is that data meaningful? Nobody seems to care what the temperature was 2 hours ago; they want to know what the temperature is now! NServiceBus provides a method to cause messages to automatically expire after a given amount of time. Unlike storing this information in a database, you don't have to run any batch jobs or take any other administrative action to ensure that old data is discarded. You simply mark the message with an expiration date and when that time arrives, the message simply evaporates into thin air: [TimeToBeReceived("01:00:00")] public class RecordCurrentTemperatureCmd : ICommand { public double Temperature { get; set; } } This example shows that the message must be received within one hour of being sent, or it is simply deleted by the queuing system. NServiceBus isn't actually involved in the deletion at all, it simply tells the queuing system how long to allow the message to live. If a message fails, however, and arrives at an error queue, NServiceBus will not include the expiration date in order to give you a chance to debug the problem. It would be very confusing to try to find an error message that had disappeared into thin air! Another valuable use for this attribute is for high-volume message types, where a communication failure between servers or extended downtime could cause a huge backlog of messages to pile up either at the sending or the receiving side. Running out of disk space to store messages is a show-stopper for most message-queuing systems, and the TimeToBeReceived attribute is the way to guard against it. However, this means we are throwing away data, so we need to be very careful when applying this strategy. It should not simply be used as a reaction to low disk space! Auditing messages At times, it can be difficult to debug a distributed system. Commands and events are sent all around, but after they are processed, they go away. We may be able to tell what will happen to a system in the future by examining queued messages, but how can we analyze what happened in the past? For this reason, NServiceBus contains an auditing function that will enable an endpoint to send a copy of every message it successfully processes to a secondary location, a queue that is generally hosted on a separate server. This is accomplished by adding an attribute or two to the UnicastBusConfig section of an endpoint's configuration: <UnicastBusConfig ForwardReceivedMessagesTo="audit@SecondaryServer" TimeToBeReceivedOnForwardedMessages="1.00:00:00"> <MessageEndpointMappings>    <!-- Mappings go here --> </MessageEndpointMappings> </UnicastBusConfig> In this example, the endpoint will forward a copy of all successfully processed messages to a queue named audit on a server named SecondaryServer, and those messages will expire after one day. While it is not required to use the TimeToBeReceivedOnForwardedMessages parameter, it is highly recommended. Otherwise, it is possible (even likely) that messages will build up in your audit queue until you run out of available storage, which you would really like to avoid. The exact time limit you use is dependent upon the volume of messages in your system and how much storage your queuing system has available. You don't even have to design your own tool to monitor these audit messages; the Particular Service Platform has that job covered for you. NServiceBus includes the auditing configuration in new endpoints by default so that ServiceControl, ServiceInsight, and ServicePulse can keep tabs on your system. Web service integration and idempotence When talking about managing failure, it's important to spend a few minutes discussing web services because they are such a special case; they are just too good at failing. When the message is processed, the email would either be sent or it won't; there really aren't any in-between cases. In reality, when sending an email, it is technically possible that we could call the SMTP server, successfully send an email, and then the server could fail before we are able to finish marking the message as processed. However, in practice, this chance is so infinitesimal that we generally assume it to be zero. Even if it is not zero, we can assume in most cases that sending a user a duplicate email one time in a few million won't be the end of the world. Web services are another story. There are just so many ways a web service can fail: A DNS or network failure may not let us contact the remote web server at all The server may receive our request, but then throw an error before any state is modified on the server The server may receive our request and successfully process it, but a communication problem prevents us from receiving the 200 OK response The connection times out, thus ignoring any response the server may have been about to send us For this reason, it makes our lives a lot easier if all the web services we ever have to deal with are idempotent, which means a process that can be invoked multiple times with no adverse effects. Any service that queries data without modifying it is inherently idempotent. We don't have to worry about how many times we call a service if doing so doesn't change any data. Where we start to get into trouble is when we begin mutating state. Sometimes, we can modify state safely. Consider an example used previously regarding registering for alert notifications. Let's assume that on the first try, the third-party service technically succeeds in registering our user for alerts, but it takes too long to do so and we receive a timeout error. When we retry, we ask to subscribe the email address to alerts again, and the web service call succeeds. What's the net effect? Either way, the user is subscribed for alerts. This web service satisfies idempotence. The classic example of a non-idempotent web service is a credit card transaction processor. If the first attempt to authorize a credit card succeeds on the server and we retry, we may double charge our customer! This is not an acceptable business case and you will quickly find many people angry with you. In these cases, we need to do a little work ourselves because unfortunately, it's impossible for NServiceBus to know whether your web service is idempotent or not. Generally, this work takes the form of recording each step we perform on durable storage in real time, and then query that storage to see which steps have been attempted. In our example of credit card processing, the happy path approach would look like this: Record our intent to make a web service call to durable storage. Make the actual web service call. Record the results of the web service call to durable storage. Send commands or publish events with the results of the web service call. Now, if the message is retried, we can inspect the durable storage and decide what step to jump to and whether any compensating actions need to be taken first. If we have recorded our intent to call the web service but do not see any evidence of a response, we can query the credit card processor based on an order or transaction identifier. Then we will know whether we need to retry the authorization or just get the results of the already completed authorization. If we see that we have already made the web service call and received the results, then we know that the web service call was successful but some exception happened before the resulting messages could be sent. In response, we can just take the results and send the messages without requiring any further web service invocations. It's important to be able to handle the case where our durable storage throws an exception, rendering us unable to make our state persist. This is why it's so important to record the intent to do something before attempting it—so that we know the difference between never having done something and attempting it but not necessarily knowing the results. The process we have just discussed is admittedly a bit abstract, and can be visualized much more easily with the help of the following diagram:   The choice of using the durable storage strategy for this process is up to you. If you choose to use a database, however, you must remember to exempt it from the message handler's ambient transaction, or those changes will also get rolled back if and when the handler fails. In order to escape the transaction to write to durable storage, use a new TransactionScope object to suppress the transaction, like this: public void Handle(CallNonIdempotentWebServiceCmdcmd) { // Under control of ambient transaction   using (var ts = new TransactionScope(TransactionScopeOption.Suppress)) {    // Not under transaction control    // Write updates to durable storage here    ts.Complete(); }   // Back under control of ambient transaction } Summary In this article, we considered the inevitable failure of our software and how NServiceBus can help us to be prepared for it. You learned how NServiceBus promises fault tolerance within every message handler so that messages are never dropped or forgotten, but instead retried and then held in an error queue if they cannot be successfully processed. Once we fix the error, or take some other administrative action, we can replay those messages. In order to avoid flooding our system with useless messages during a failure, you learned how to cause messages that lose their business value after a specific amount of time to expire. Finally, you learned how to build auditing in a system by forwarding a copy of all messages for later inspection, and how to properly deal with the challenges involved in calling external web services. In this article, we dealt exclusively with NServiceBus endpoints hosted by the NServiceBus Host process.
Read more
  • 0
  • 0
  • 3939

article-image-ceph-instant-deployment
Packt
09 Feb 2015
14 min read
Save for later

Ceph Instant Deployment

Packt
09 Feb 2015
14 min read
In this article by Karan Singh, author of the book, Learning Ceph, we will cover the following topics: Creating a sandbox environment with VirtualBox From zero to Ceph – deploying your first Ceph cluster Scaling up your Ceph cluster – monitor and OSD addition (For more resources related to this topic, see here.) Creating a sandbox environment with VirtualBox We can test deploy Ceph in a sandbox environment using Oracle VirtualBox virtual machines. This virtual setup can help us discover and perform experiments with Ceph storage clusters as if we are working in a real environment. Since Ceph is an open source software-defined storage deployed on top of commodity hardware in a production environment, we can imitate a fully functioning Ceph environment on virtual machines, instead of real-commodity hardware, for our testing purposes. Oracle VirtualBox is a free software available at http://www.virtualbox.org for Windows, Mac OS X, and Linux. We must fulfil system requirements for the VirtualBox software so that it can function properly during our testing. We assume that your host operating system is a Unix variant; for Microsoft windows, host machines use an absolute path to run the VBoxManage command, which is by default c:Program FilesOracleVirtualBoxVBoxManage.exe. The system requirement for VirtualBox depends upon the number and configuration of virtual machines running on top of it. Your VirtualBox host should require an x86-type processor (Intel or AMD), a few gigabytes of memory (to run three Ceph virtual machines), and a couple of gigabytes of hard drive space. To begin with, we must download VirtualBox from http://www.virtualbox.org/ and then follow the installation procedure once this has been downloaded. We will also need to download the CentOS 6.4 Server ISO image from http://vault.centos.org/6.4/isos/. To set up our sandbox environment, we will create a minimum of three virtual machines; you can create even more machines for your Ceph cluster based on the hardware configuration of your host machine. We will first create a single VM and install OS on it; after this, we will clone this VM twice. This will save us a lot of time and increase our productivity. Let's begin by performing the following steps to create the first virtual machine: The VirtualBox host machine used throughout in this demonstration is a Mac OS X which is a UNIX-type host. If you are performing these steps on a non-UNIX machine that is, on Windows-based host then keep in mind that virtualbox hostonly adapter name will be something like VirtualBox Host-Only Ethernet Adapter #<adapter number>. Please run these commands with the correct adapter names. On windows-based hosts, you can check VirtualBox networking options in Oracle VM VirtualBox Manager by navigating to File | VirtualBox Settings | Network | Host-only Networks. After the installation of the VirtualBox software, a network adapter is created that you can use, or you can create a new adapter with a custom IP:For UNIX-based VirtualBox hosts # VBoxManage hostonlyif remove vboxnet1 # VBoxManage hostonlyif create # VBoxManage hostonlyif ipconfig vboxnet1 --ip 192.168.57.1 --netmask 255.255.255.0 For Windows-based VirtualBox hosts # VBoxManage.exe hostonlyif remove "VirtualBox Host-Only Ethernet Adapter" # VBoxManage.exe hostonlyif create # VBoxManage hostonlyif ipconfig "VirtualBox Host-Only Ethernet Adapter" --ip 192.168.57.1 --netmask 255.255.255. VirtualBox comes with a GUI manager. If your host is running Linux OS, it should have the X-desktop environment (Gnome or KDE) installed. Open Oracle VM VirtualBox Manager and create a new virtual machine with the following specifications using GUI-based New Virtual Machine Wizard, or use the CLI commands mentioned at the end of every step: 1 CPU 1024 MB memory 10 GB X 4 hard disks (one drive for OS and three drives for Ceph OSD) 2 network adapters CentOS 6.4 ISO attached to VM The following is the step-by-step process to create virtual machines using CLI commands: Create your first virtual machine: # VBoxManage createvm --name ceph-node1 --ostype RedHat_64 --register # VBoxManage modifyvm ceph-node1 --memory 1024 --nic1 nat --nic2 hostonly --hostonlyadapter2 vboxnet1 For Windows VirtualBox hosts: # VBoxManage.exe modifyvm ceph-node1 --memory 1024 --nic1 nat --nic2 hostonly --hostonlyadapter2 "VirtualBox Host-Only Ethernet Adapter" Create CD-Drive and attach CentOS ISO image to first virtual machine: # VBoxManage storagectl ceph-node1 --name "IDE Controller" --add ide --controller PIIX4 --hostiocache on --bootable on # VBoxManage storageattach ceph-node1 --storagectl "IDE Controller" --type dvddrive --port 0 --device 0 --medium CentOS-6.4-x86_64-bin-DVD1.iso Make sure you execute the preceding command from the same directory where you have saved CentOS ISO image or you can specify the location where you saved it. Create SATA interface, OS hard drive and attach them to VM; make sure the VirtualBox host has enough free space for creating vm disks. If not, select the host drive which have free space: # VBoxManage storagectl ceph-node1 --name "SATA Controller" --add sata --controller IntelAHCI --hostiocache on --bootable on # VBoxManage createhd --filename OS-ceph-node1.vdi --size 10240 # VBoxManage storageattach ceph-node1 --storagectl "SATA Controller" --port 0 --device 0 --type hdd --medium OS-ceph-node1.vdi Create SATA interface, first ceph disk and attach them to VM: # VBoxManage createhd --filename ceph-node1-osd1.vdi --size 10240 # VBoxManage storageattach ceph-node1 --storagectl "SATA Controller" --port 1 --device 0 --type hdd --medium ceph-node1-osd1.vdi Create SATA interface, second ceph disk and attach them to VM: # VBoxManage createhd --filename ceph-node1-osd2.vdi --size 10240 # VBoxManage storageattach ceph-node1 --storagectl "SATA Controller" --port 2 --device 0 --type hdd --medium ceph-node1-osd2.vdi Create SATA interface, third ceph disk and attach them to VM: # VBoxManage createhd --filename ceph-node1-osd3.vdi --size 10240 # VBoxManage storageattach ceph-node1 --storagectl "SATA Controller" --port 3 --device 0 --type hdd --medium ceph-node1-osd3.vdi Now, at this point, we are ready to power on our ceph-node1 VM. You can do this by selecting the ceph-node1 VM from Oracle VM VirtualBox Manager, and then clicking on the Start button, or you can run the following command: # VBoxManage startvm ceph-node1 --type gui As soon as you start your VM, it should boot from the ISO image. After this, you should install CentOS on VM. If you are not already familiar with Linux OS installation, you can follow the documentation at https://access.redhat.com/site/documentation/en-US/Red_Hat_Enterprise_Linux/6/html/Installation_Guide/index.html. Once you have successfully installed the operating system, edit the network configuration of the machine: Edit /etc/sysconfig/network and change the hostname parameter HOSTNAME=ceph-node1 Edit the /etc/sysconfig/network-scripts/ifcfg-eth0 file and add: ONBOOT=yes BOOTPROTO=dhcp Edit the /etc/sysconfig/network-scripts/ifcfg-eth1 file and add:< ONBOOT=yes BOOTPROTO=static IPADDR=192.168.57.101 NETMASK=255.255.255.0 Edit the /etc/hosts file and add: 192.168.57.101 ceph-node1 192.168.57.102 ceph-node2 192.168.57.103 ceph-node3 Once the network settings have been configured, restart VM and log in via SSH from your host machine. Also, test the Internet connectivity on this machine, which is required to download Ceph packages: # ssh root@192.168.57.101 Once the network setup has been configured correctly, you should shut down your first VM so that we can make two clones of your first VM. If you do not shut down your first VM, the cloning operation might fail. Create clone of ceph-node1 as ceph-node2: # VBoxManage clonevm --name ceph-node2 ceph-node1 --register Create clone of ceph-node1 as ceph-node3: # VBoxManage clonevm --name ceph-node3 ceph-node1 --register After the cloning operation is complete, you can start all three VMs: # VBoxManage startvm ceph-node1 # VBoxManage startvm ceph-node2 # VBoxManage startvm ceph-node3 Set up VM ceph-node2 with the correct hostname and network configuration: Edit /etc/sysconfig/network and change the hostname parameter: HOSTNAME=ceph-node2 Edit the /etc/sysconfig/network-scripts/ifcfg-<first interface name> file and add: DEVICE=<correct device name of your first network interface, check ifconfig -a> ONBOOT=yes BOOTPROTO=dhcp HWADDR= <correct MAC address of your first network interface, check ifconfig -a > Edit the /etc/sysconfig/network-scripts/ifcfg-<second interface name> file and add: DEVICE=<correct device name of your second network interface, check ifconfig -a> ONBOOT=yes BOOTPROTO=static IPADDR=192.168.57.102 NETMASK=255.255.255.0 HWADDR= <correct MAC address of your second network interface, check ifconfig -a > Edit the /etc/hosts file and add: 192.168.57.101 ceph-node1 192.168.57.102 ceph-node2 192.168.57.103 ceph-node3 After performing these changes, you should restart your virtual machine to bring the new hostname into effect. The restart will also update your network configurations. Set up VM ceph-node3 with the correct hostname and network configuration: Edit /etc/sysconfig/network and change the hostname parameter:HOSTNAME=ceph-node3 Edit the /etc/sysconfig/network-scripts/ifcfg-<first interface name> file and add: DEVICE=<correct device name of your first network interface, check ifconfig -a> ONBOOT=yes BOOTPROTO=dhcp HWADDR= <correct MAC address of your first network interface, check ifconfig -a > Edit the /etc/sysconfig/network-scripts/ifcfg-<second interface name> file and add: DEVICE=<correct device name of your second network interface, check ifconfig -a> ONBOOT=yes BOOTPROTO=static IPADDR=192.168.57.103 NETMASK=255.255.255.0 HWADDR= <correct MAC address of your second network interface, check ifconfig -a > Edit the /etc/hosts file and add: 192.168.57.101 ceph-node1 192.168.57.102 ceph-node2 192.168.57.103 ceph-node3 After performing these changes, you should restart your virtual machine to bring a new hostname into effect; the restart will also update your network configurations. At this point, we prepare three virtual machines and make sure each VM communicates with each other. They should also have access to the Internet to install Ceph packages. From zero to Ceph – deploying your first Ceph cluster To deploy our first Ceph cluster, we will use the ceph-deploy tool to install and configure Ceph on all three virtual machines. The ceph-deploy tool is a part of the Ceph software-defined storage, which is used for easier deployment and management of your Ceph storage cluster. Since we created three virtual machines that run CentOS 6.4 and have connectivity with the Internet as well as private network connections, we will configure these machines as Ceph storage clusters as mentioned in the following diagram: Configure ceph-node1 for an SSH passwordless login to other nodes. Execute the following commands from ceph-node1: While configuring SSH, leave the paraphrase empty and proceed with the default settings: # ssh-keygen Copy the SSH key IDs to ceph-node2 and ceph-node3 by providing their root passwords. After this, you should be able to log in on these nodes without a password: # ssh-copy-id ceph-node2 Installing and configuring EPEL on all Ceph nodes: Install EPEL which is the repository for installing extra packages for your Linux system by executing the following command on all Ceph nodes: # rpm -ivh http://dl.fedoraproject.org/pub/epel/6/x86_64/epel-release-6-8.noarch.rpm Make sure the baserul parameter is enabled under the /etc/yum.repos.d/epel.repo file. The baseurl parameter defines the URL for extra Linux packages. Also make sure the mirrorlist parameter must be disabled (commented) under this file. Problems been observed during installation if the mirrorlist parameter is enabled under epel.repo file. Perform this step on all the three nodes. Install ceph-deploy on the ceph-node1 machine by executing the following command from ceph-node1: # yum install ceph-deploy Next, we will create a Ceph cluster using ceph-deploy by executing the following command from ceph-node1: # ceph-deploy new ceph-node1 ## Create a directory for ceph # mkdir /etc/ceph # cd /etc/ceph The new subcommand of ceph-deploy deploys a new cluster with ceph as the cluster name, which is by default; it generates a cluster configuration and keying files. List the present working directory; you will find the ceph.conf and ceph.mon.keyring files. In this testing, we will intentionally install the Emperor release (v0.72) of Ceph software, which is not the latest release. Later in this book, we will demonstrate the upgradation of Emperor to Firefly release of Ceph. To install Ceph software binaries on all the machines using ceph-deploy; execute the following command from ceph-node1:ceph-deploy install --release emperor ceph-node1 ceph-node2 ceph-node3 The ceph-deploy tool will first install all the dependencies followed by the Ceph Emperor binaries. Once the command completes successfully, check the Ceph version and Ceph health on all the nodes, as follows: # ceph –v Create your first monitor on ceph-node1: # ceph-deploy mon create-initial Once monitor creation is successful, check your cluster status. Your cluster will not be healthy at this stage: # ceph status Create an object storage device (OSD) on the ceph-node1 machine, and add it to the Ceph cluster executing the following steps: List the disks on VM: # ceph-deploy disk list ceph-node1 From the output, carefully identify the disks (other than OS-partition disks) on which we should create Ceph OSD. In our case, the disk names will ideally be sdb, sdc, and sdd. The disk zap subcommand will destroy the existing partition table and content from the disk. Before running the following command, make sure you use the correct disk device name. # ceph-deploy disk zap ceph-node1:sdb ceph-node1:sdc ceph-node1:sdd The osd create subcommand will first prepare the disk, that is, erase the disk with a filesystem, which is xfs by default. Then, it will activate the disk's first partition as data partition and second partition as journal: # ceph-deploy osd create ceph-node1:sdb ceph-node1:sdc ceph-node1:sdd Check the cluster status for new OSD entries: # ceph status At this stage, your cluster will not be healthy. We need to add a few more nodes to the Ceph cluster so that it can set up a distributed, replicated object storage, and hence become healthy. Scaling up your Ceph cluster – monitor and OSD addition Now we have a single-node Ceph cluster. We should scale it up to make it a distributed, reliable storage cluster. To scale up a cluster, we should add more monitor nodes and OSD. As per our plan, we will now configure ceph-node2 and ceph-node3 machines as monitor as well as OSD nodes. Adding the Ceph monitor A Ceph storage cluster requires at least one monitor to run. For high availability, a Ceph storage cluster relies on an odd number of monitors that's more than one, for example, 3 or 5, to form a quorum. It uses the Paxos algorithm to maintain quorum majority. Since we already have one monitor running on ceph-node1, let's create two more monitors for our Ceph cluster: The firewall rules should not block communication between Ceph monitor nodes. If they do, you need to adjust the firewall rules in order to let monitors form a quorum. Since this is our test setup, let's disable firewall on all three nodes. We will run these commands from the ceph-node1 machine, unless otherwise specified: # service iptables stop # chkconfig iptables off # ssh ceph-node2 service iptables stop # ssh ceph-node2 chkconfig iptables off # ssh ceph-node3 service iptables stop # ssh ceph-node3 chkconfig iptables off Deploy a monitor on ceph-node2 and ceph-node3: # ceph-deploy mon create ceph-node2 # ceph-deploy mon create ceph-node3 The deploy operation should be successful; you can then check your newly added monitors in the Ceph status: You might encounter warning messages related to clock skew on new monitor nodes. To resolve this, we need to set up Network Time Protocol (NTP) on new monitor nodes: # chkconfig ntpd on # ssh ceph-node2 chkconfig ntpd on # ssh ceph-node3 chkconfig ntpd on # ntpdate pool.ntp.org # ssh ceph-node2 ntpdate pool.ntp.org # ssh ceph-node3 ntpdate pool.ntp.org # /etc/init.d/ntpd start # ssh ceph-node2 /etc/init.d/ntpd start # ssh ceph-node3 /etc/init.d/ntpd start Adding the Ceph OSD At this point, we have a running Ceph cluster with three monitors OSDs. Now we will scale our cluster and add more OSDs. To accomplish this, we will run the following commands from the ceph-node1 machine, unless otherwise specified. We will follow the same method for OSD addition: # ceph-deploy disk list ceph-node2 ceph-node3 # ceph-deploy disk zap ceph-node2:sdb ceph-node2:sdc ceph-node2:sdd # ceph-deploy disk zap ceph-node3:sdb ceph-node3:sdc ceph-node3:sdd # ceph-deploy osd create ceph-node2:sdb ceph-node2:sdc ceph-node2:sdd # ceph-deploy osd create ceph-node3:sdb ceph-node3:sdc ceph-node3:sdd # ceph status Check the cluster status for a new OSD. At this stage, your cluster will be healthy with nine OSDs in and up: Summary The software-defined nature of Ceph provides a great deal of flexibility to its adopters. Unlike other proprietary storage systems, which are hardware dependent, Ceph can be easily deployed and tested on almost any computer system available today. Moreover, if getting physical machines is a challenge, you can use virtual machines to install Ceph, as mentioned in this article, but keep in mind that such a setup should only be used for testing purposes. In this article, we learned how to create a set of virtual machines using the VirtualBox software, followed by Ceph deployment as a three-node cluster using the ceph-deploy tool. We also added a couple of OSDs and monitor machines to our cluster in order to demonstrate its dynamic scalability. We recommend you deploy a Ceph cluster of your own using the instructions mentioned in this article. Resources for Article: Further resources on this subject: Linux Shell Scripting - various recipes to help you [article] GNU Octave: Data Analysis Examples [article] What is Kali Linux [article]
Read more
  • 0
  • 0
  • 6282

article-image-how-to-build-a-koa-web-application-part-2
Christoffer Hallas
08 Feb 2015
5 min read
Save for later

How to Build a Koa Web Application - Part 2

Christoffer Hallas
08 Feb 2015
5 min read
In Part 1 of this series, we got everything in place for our Koa app using Jade and Mongel. In this post, we will cover Jade templates and how to use listing and viewing pages. Please note that this series requires that you use Node.js version 0.11+. Jade templates Rendering HTML is always an important part of any web application. Luckily, when using Node.js there are many great choices, and for this article we’ve chosen Jade. Keep in mind though that we will only touch on a tiny fraction of the Jade functionality. Let’s create our first Jade template. Create a file called create.jade and put in the following: create.jade doctype html html(lang='en') head title Create Page body h1 Create Page form(method='POST', action='/create') input(type='text', name='title', placeholder='Title') input(type='text', name='contents', placeholder='Contents') input(type='submit') For all the Jade questions you have that we won’t answer in this series, I refer you to the excellent official Jade website at http://jade-lang.com . If you add the following statement app.listen(3000); to the end of index.js, then you should be able to run the program from your terminal using the following command and by visiting http://localhost:3000 in your browser. $ node --harmony index.js The --harmony flag just tells the node program that we need support for generators in our program: Listing and viewing pages Now that we can create a page in our MongoDB database, it is time to actually list and view these pages. For this purpose we need to add another middleware to our index.js file after the first middleware: app.use(function* () { if (this.method != 'GET') { this.status = 405; this.body = 'Method Not Allowed'; return } … }); As you can probably already tell, this new middleware is very similar to the first one we added that handled the creation of pages. At first we make sure that the method of the request is GET, and if not, we respond appropriately and return the following: var params = this.path.split('/').slice(1); var id = params[0]; if (id.length == 0) { var pages = yield Page.find(); var html = jade.renderFile('list.jade', { pages: pages }); this.body = html; return } Then, we proceed to inspect the path attribute of the Koa context, looking for an ID that represents the page in the database. Remember how we redirected using the ID in the previous middleware. We inspect the path by splitting it into an array of strings separated by the forward slashes of a URL; this way the path /1234 becomes an array of ‘’ and ‘1234.’ Because the path starts with a forward slash, the first item in the array will always be the empty string, so we just discard that by default. Then we check the length of the ID parameter, and if it’s zero we know that there is in fact no ID in the path, and we should just look for the pages in the database and render our list.jade template with those pages made available to the template as the variable pages. Making data available in templates is also known as providing locals to the template. list.jade doctype html html(lang="en") head title Your Web Application body h1 Your Web Application ul - each page in pages li a(href='/#{page._id}')= page.title But if the length of id was not zero, we assume that it’s an id and we try to load that specific page from the database instead of all the pages, and we proceed to render our view.jade template with the: var page = yield Page.findById(id); var html = jade.renderFile('view.jade', page); this.body = html; view.jade doctype html html(lang="en") head title= title body h1= title p= contents That’s it You should now be able to run the app as previously described and create a page, list all of your pages, and view them. If you want to, you can continue and build a simple CMS system. Koa is very simple to use and doesn’t enforce a lot of functionality on you, allowing you to pick and choose between libraries that you need and want to use. There are many possibilities and that is one of Koa’s biggest strengths. Find even more Node.js content on our Node.js page. Featuring our latest titles and most popular tutorials, it's the perfect place to learn more about Node.js. About the author Christoffer Hallas is a software developer and entrepreneur from Copenhagen, Denmark. He is a computer polyglot and contributes to and maintains a number of open source projects. When not contemplating his next grand idea (which remains an idea), he enjoys music, sports, and design of all kinds. Christoffer can be found on GitHub as hallas and at Twitter as @hamderhallas.
Read more
  • 0
  • 0
  • 4080

article-image-five-kinds-python-functions-python-34-edition
Packt
06 Feb 2015
33 min read
Save for later

The Five Kinds of Python Functions Python 3.4 Edition

Packt
06 Feb 2015
33 min read
This article is written by Steven Lott, author of the book Functional Python Programming. You can find more about him at http://slott-softwarearchitect.blogspot.com. (For more resources related to this topic, see here.) What's This About? We're going to look at various ways that Python 3 lets us define things which behave like functions. The proper term here is Callable – we're looking at objects that can be called like a function. We'll look at the following Python constructs: Function definitions Higher-order functions Function wrappers (around methods) Lambdas Callable objects Generator functions and the yield parameter And yes, we're aware that the list above has six items on it. That's because higher-order functions in Python aren't really all that complex or different. In some languages, functions that take functions are arguments involving special syntax. In Python, it's simple and common and barely worth mentioning as a separate topic. We'll look at when it's appropriate and inappropriate to use one or the other of these various functional forms. Some background Let's take a quick peek at a basic bit of mathematical formalism. We'll look at a function as an abstract formalism. We often annotate it like this: This shows us that f() is a function. It has one argument, x, and will map this to a single value, y. Some mathematical functions are written in front, for example, y=sin x. Some are written in other places around the argument, for example, y=|x|. In Python, the syntax is more consistent, for example, we use a function like this: >>> abs(-5)5 We've applied the abs() function to an argument value of -5. The argument value was mapped to a value of 5. Terminology Consider the following function: In this definition, the argument is a pair of values, (a,b). This is called the domain. We can summarize it as the domain of values for which the function is defined. Outside this domain, the function is not defined. In Python, we get a TypeError exception if we provide one value or three values as the argument. The function maps the domain pair to a pair of values, (q,r). This is the range of the function. We can call this the range of values that could be returned by the function. Mathematical function features As we look at the abstract mathematical definition of functions, we note that functions are generally assumed to have no hysteresis; they have no history or memory of prior use. This is sometimes called the property of being idempotent: the results are always the same for a given argument value. We see this in Python as a common feature. But it's not universally true. We'll look at a number of exceptions to the rule of idempotence. Here's an example of the usual situation: >>> int("10f1", 16)4337 The value returned from the evaluation of int("10f1", 16) never changes. There are, however, some common examples of non-idempotent functions in Python. Examples of hysteresis Here are three common situations where a function has hysteresis. In some cases, results vary based on history. In other cases, results vary based on events in some external environment, such as follows: Random number generators. We don't want them to produce the same value over and over again. The Python random.randrange() function, is not obviously idempotent. OS functions depend on the state of the machine as a whole. The os.listdir() function returns values that depend on the use of functions such as os.unlink(), os.rename(), and open() (among several others).While the rules are generally simple, it requires a stateful object outside the narrow world of the code itself. These are examples of Python functions that don't completely fit the formal mathematical definition; they lack idempotence, and their values depend on history, other functions, or both. Function Definitions Python has two statements that are essential features of function definition. The def statement specifies the domain and the return statement(s) specify the range. A simplified gloss of the syntax is as follows: def name(params):   body   return expression In effect, the function's domain is defined by the parameters provided in the def statement. This list of parameter names is not all the information on the domain, however. Even if we use one of the Python extensions to add type annotations, that's still not all the information. There may be if statements in the body of the function that impose additional explicit restrictions. There may be other functions that impose their own kind of implicit restrictions. If, for example, the body included math.sqrt() then there would be an implicit restriction on some values being non-negative. The return statements provide the function's range. An empty return statement means a range of simply None values. When there are multiple return statements, the range is the union of the ranges on all the return statements. This mapping between Python syntax and mathematical concepts isn't very complete. We need more information about a function. Example definition Here's an example of function definition: def odd(n):   """odd(n) -> boolean, true if n is odd."""   return n % 2 == 1 What do does this definition tell us? Several things such as: Domain: We know that this function accepts n, a single object. Range: Boolean value, True if n is an odd number. This is the most likely interpretation. It's also remotely possible that the class of n has repurposed __mod__() or __rmod__() methods, in which case the semantics can be pretty obscure. Because of the inherent ambiguity in Python, this function has provided a triple-quoted """Docstring""" parameter with a summary of the function. This is a best practice, and should be followed universally except in articles like this where it gets too long-winded to include a docstring parameter everywhere. In this case, the doctoring parameter doesn't state unambiguously that n is intended to be a number. There are two ways to handle this gap, they are as follows: Actually include words like n is a number in the docstring parameter Include the docstring parameter test cases that show the required behavior Either is acceptable. Both are preferable. Using a function To complete this example, here's how we'd use this odd little function named odd(): >>> odd(3)True>>> odd(4)False This kind of example text can be included into the docstring parameter to create two test cases that offer insight into what the function really means. The lack of declarations More verbose type declarations—as used in many popular programming languages—aren't actually enough information to fully specify a function's domain and range. To be rigorously complete, we need type definitions that include optional predicates. Take a look at the following command: isinstance(n,int) and n >= 0 The assert statement is a good place for this kind of additional argument domain checking. This isn't the perfect solution because assert statements can be disabled very easily. It can help during design and testing and it can help people to read your code. The fussy formal declarations of data type used in other languages are not really needed in Python. Python replaces an up-front claim about required types with a runtime search for appropriate class methods. This works because each Python object has all the type information bound into it. Static compile-time type information is redundant, since the runtime type information is complete. A Python function definition is pretty spare. In includes the minimal amount of information about the function. There are no formal declaration of parameter types or return type. This odd little function will work with any object that implements the % operator: Generally, this means any object that implements __mod__() or __rmod__(). This means most subclasses of numbers.Number. It also means instances of any class that happen to provide these methods. That could become very weird, but still possible. We hesitate to think about non-numeric objects that work with the number-like % operator. Some Python features In Python, functions we declare are proper first-class objects. This means that they have attributes that can be assigned to variables and placed into collections. Quite a few clever things can be done with function objects. One of the most elegant things is to use a function as an argument or a return value from another function. The ability to do this means that we can easily create and use higher-order functions in Python. For folks who know languages such as C (and C++), functions aren't proper first-class objects. A pointer to a function, however, is a first class object in C. But the function itself is a block of code that can't easily be manipulated. We'll look at a number of simple ways in which we can write—and use—higher-order functions in Python. Functions are objects Consider the following command example: >>> not_even = odd>>> not_even(3)True We've assigned the odd little function object to a new variable, not_even. This creates an alias for a function. While this isn't always the best idea, there are times when we might want to provide an alternate name for a function as part of maintaining reverse compatibility with a previous release of a library. Using functions Consider the following function definition: def some_test(function, value):   print(function, value)   return function(value) This function's domain includes arguments named function and value. We can see that it prints the arguments, then applies the function argument to the given value. When we use the preceding function, it looks like this: >>> some_test(odd, 3)<function odd at 0x613978> 3True The some_test() function accepted a function as an argument. When we printed the function, we got a summary, <function odd at 0x613978>, that shows us some information about the object. We also show a summary of the argument value, 3. When we applied the function to a value, we got the expected result. We can—of course—extend this concept. In particular, we can apply a single function to many values. Higher-order Functions Higher-order functions become particularly useful when we apply them to collections of objects. The built-in map() function applies a simple function to each value in an argument sequence. Here's an example: >>> list(map(odd, [1,2,3,4]))[True, False, True, False] We've used the map() function to apply the odd() function to each value in the sequence. This is a lot like evaluating: >>> [odd(x) for x in [1,2,3,4]] We've created a list comprehension instead of applying a higher-order map() function. This is equivalent to the following command snippet: [odd(1), odd(2), odd(3), odd(4)] Here, we've manually applied the odd() function to each value in a sequence. Yes, that's a diesel engine alternator and some hoses: We'll use this alternator as a subject for some concrete examples of higher-order functions. Diesel engine background Some basic diesel engine mechanics. The following some basic information: The engine turns the alternator. The alternator generates pulses that drive the tachometer. Amongst other things, like charging the batteries. The alternator provides an indirect measurement of engine RPMs. Direct measurement would involve connecting to a small geared shaft. It's difficult and expensive. We already have a tachometer; it's just incorrect. The new alternator has new wheels. The ratios between engine and alternator have changed. We're not interested in installing a new tachometer. Instead, we'll create a conversion from a number on the tachometer, which is calibrated to the old alternator, to a proper number of engine RPMs. This has to allow the change in ratio between the original tachometer and the new tach. Let's collect some data and see what we can figure out about engine RPMs. New alternator First approximation: all we did was get new wheels. We can presume that the old tachometer was correct. Since the new wheel is smaller, we'll have higher alternator RPMs. That means higher readings on the old tachometer. Here's the key question: How far wrong are the RPMs? The old wheel was approximately 3.5 RPM and the new wheel is approximately 2.5 RPM. We can compute the potential ratio between what the tach says and what the engine is really doing: >>> 3.5/2.51.4>>> 1/_0.7142857142857143 That's nice. Is it right? Can we really just multiply and display RPMs by .7 to get actual engine RPMs? Let's create the conversion card first, then collect some more data. Use case Given RPM on the tachometer, what's the real RPM of the engine? Use the following command to find the RPM: def eng(r):   return r/1.4 Use it like the following: >>> eng(2100)1500.0 This seems useful. Tach says 2100, engine (theoretically) spinning at 1500, more or less. Let's confirm our hypothesis with some real data. Data collection Over a period of time, we recorded tachometer readings and actual RPMs using a visual RPM measuring device. The visual device requires a strip of reflective tape on one of the engine wheels. It uses a laser and counts returns per minute. Simple. Elegant. Accurate. It's really inconvenient. But it got some data we could digest. Skipping some boring statistics, we wind up with the following function that maps displayed RPMs to actual RPMs, such as this: def eng2(r):   return 0.7724*r**1.0134 Here's a sample result: >>> eng2(2100)1797.1291903589386 When tach says 2100, the engine is measured as spinning at about 1800 RPM. That's not quite the same as the theoretical model. But it's so close that it gives us a lot of confidence in this version. Of course, the number displayed is hideous. All that floating-point cruft is crazy. What can we do? Rounding is only part of the solution. We need to think through the use case. After all, we use this standing at the helm of the boat; how much detail is appropriate? Limits and ranges The engine has governors and only runs between 800 and 2500 RPM. There's a very tight limit here. Realistically, we're talking about this small range of values: >>> list(range(800, 2500, 200))[800, 1000, 1200, 1400, 1600, 1800, 2000, 2200, 2400] There's no sensible reason for proving any more detailed engine RPMs. It's a sailboat; top speed is 7.5 knots (Nautical miles per hour). Wind and current have far more impact on the boat speed than the difference between 1600 and 1700 RPMs. The tach can't be read to closer than 100-200 RPM. It's not digital, it's a red pointer near little tick lines. There's no reason to preserve more than a few bits of precision. Example of Tach translation Given the engine RPMs and the conversion function, we can deduce that the tachometer display will be between 1000 to 3200. This will map to engine RPMs in the range of about 800 to 2500. We can confirm this with a mapping like this: >>> list(map(eng2, range(1000,3200,200)))[847.3098694826986, 1019.258964596305, 1191.5942982618956, 1364.2609728487703, 1537.2178605443924, 1710.4329833319157, 1883.8807562755746, 2057.5402392829747, 2231.3939741669838, 2405.4271806626366, 2579.627182659544] We've applied the eng2() mapping from tach to engine RPM. For tach readings between 1000 and 3200 in steps of 200, we've computed the actual engine RPMs. For those who use spreadsheets a lot, the range() function is like filling a column with values. The map(eng2, …) function is like filling an adjacent column with a calculation. We've created the result of applying a function to each value of a given range. As shown, this is little difficult to use. We need to do a little more cleanup. What other function do we need to apply to the results? Round to 100 Here's a function that will round up to the nearest 100: def next100(n):   return int(round(n, -2)) We could call this a kind of composite function built from a partial application of round() and int() functions. If we map this function to the previous results, we get something a little easier to work with. How does this look? >>> tach= range(1000,3200,200)>>> list(map(next100, map(eng2, tach)))[800, 1000, 1200, 1400, 1500, 1700, 1900, 2100, 2200, 2400, 2600] This expression is a bit complex; let's break it down into three discrete steps: First, map the eng2() function to tach numbers between 1000 and 3200. The result is effectively a sequence of values (it's not actually a list, it's a generator, a potential list) Second, map the next100() function to results of previous mapping Finally, collect a single list object from the results We've applied two functions, eng2() and next100(), to a list of values. In principle, we've created a kind of composite function, next100○eng20(rpm). Python doesn't support function composition directly, hence the complex-looking map of map syntax. Interleave sequences of values The final step is to create a table that shows both the tachometer reading and the computed engine RPMs. We need to interleave the input and output values into a single list of pairs. Here are the tach readings we're working with, as a list: >>> tach= range(1000,3200,200) Here are the engine RPMs: >>> engine= list(map(next100,map(eng2,tach))) Here's how we can interleave the two to create something that shows our tachometer reading and engine RPMs: >>> list(zip(tach, engine))[(1000, 800), (1200, 1000), (1400, 1200), (1600, 1400), (1800, 1500), (2000, 1700),(2200, 1900), (2400, 2100), (2600, 2200), (2800, 2400), (3000, 2600)] The rest is pretty-printing. What's important is that we could take functions like eng() or eng2() and apply it to columns of numbers, creating columns of results. The map() function means that we don't have to write explicit for loops to simply apply a function to a sequence of values. Map is lazy We have a few other observations about the Python higher-order functions. First, these functions are lazy, they don't compute any results until required by other statements or expressions. Because they don't actually create intermediate list objects, they may be quite fast. The laziness feature is true for the built-in higher-order functions map() and filter(). It's also true for many of the functions in the itertools library. Many of these functions don't simply create a list object, they yield values as requested. For debugging purposes, we use list() to see what's being produced. If we don't apply list() to the result of a lazy function, we simply see that it's a lazy function. Here's an example: >>> map(lambda x:x*1.4, range(1000,3200,200))<map object at 0x102130610> We don't see a proper result here, because the lazy map() function didn't do anything. The list(), tuple(), or set() functions will force a lazy map() function to actually get up off the couch and compute something. Function Wrappers There are a number of Python functions which are syntactic sugar for method functions. One example is the len() function. This function behaves as if it had the following definition: def len(obj):   return obj.__len__() The function acts like it's simply invoking the object's built-in __len__() method. There are several Python functions that exist only to make the syntax a little more readable. Post-fix syntax purists would prefer to see syntax such as some_list.len(). Those who like their code to look a little more mathematical prefer len(some_list). Some people will go so far as to claim that the presence of prefix functions means that Python isn't strictly object-oriented. This is false; Python is very strictly object-oriented. It doesn't—however—use only postfix method notation. We can write function wrappers to make some method functions a little more palatable. Another good example is the divmod() function. This relies on two method functions, such as the following: a.__divmod__(b) b.__rdivmod__(a) The usual operator rules apply here. If the class for object a implements __divmod__(), then that's used to compute the result. If not, then the same test is made for the class of object b; if there's an implementation, that will be used to compute the results. Otherwise, it's undefined and we'll get an exception. Why wrap a method? Function wrappers for methods are syntactic sugar. They exist to make object methods look like simple functions. In some cases, the functional view is more succinct and expressive. Sometimes the object involved is obvious. For example, the os module functions provide access to OS-level libraries. The OS object is concealed inside the module. Sometimes the object is implied. For example, the random module makes a Random instance for us. We can simply call random.randint() without worrying about the object that was required for this to work properly. Lambdas A lambda is an anonymous function with a degenerate body. It's like a function in some respects and it's unlike a function because of the following two things: A lambda has no name A lambda has no statements A lambda's body is a single expression, nothing more. This expression can have parameters, however, which is why a lambda is a handy form of a callable function. The syntax is essentially as follows: lambda params : expression Here's a concrete example: lambda r: 0.7724*r**1.0134 You may recognize this as the eng2() function defined previously. We don't always need a complete, formal function. Sometimes, we just need an expression that has parameters. Speaking theoretically, a lambda is a one-argument function. When we have multi-argument functions, we can transform it to a series of one-argument lambda forms. This transformation can be helpful for optimization. None of that applies to Python. We'll move on. Using a Lambda with map Here are two equivalent results: map(eng2, tach) map(lambda r: 0.7724*r**1.0134, tach) Here's a previous example, using the lambda instead of the function: >>> tach= range(1000,3200,200)>>> list( map(lambda r: 0.7724*r**1.0134, tach))[847.3098694826986, 1019.258964596305, 1191.5942982618956, 1364.2609728487703, 1537.2178605443924, 1710.4329833319157, 1883.8807562755746, 2057.5402392829747, 2231.3939741669838, 2405.4271806626366, 2579.627182659544] You could scroll back to see that the results are the same. If we're doing a small thing once only, a lambda object might be more clear than a complete function definition. Emphasis here is on small once only. If we start trying to reuse a lambda object, or feel the need to assign a lambda object to a variable, we should really consider a function definition and the associated docstring and doctest features. Another use of Lambdas A common use of lambdas is with three other higher-order functions: sort(), min(), and max(). We might use one of these with a list object: list.sort(key= lambda x: expr) list.min(key= lambda x: expr) list.max(key= lambda x: expr) In each case, we're using a lambda object to embed an expression into the argument values for a function. In some cases, the expression might be very sophisticated; in other cases, it might be something as trivial as lambda x: x[1]. When the expression is trivial, a lambda object is a good idea. If the expression is going to get reused, however, a lambda object might be a bad idea. You can do this… But… The following kind of statement makes sense: some_name = lambda x: 3*x+1 We've created a callable object that takes a single argument value and returns a numeric value such as the following command snippet: def some_name(x): return 3*x+1. There are some differences. Most notably the following: A lambda object is all on one line of code. A possible advantage. There's no docstring. A disadvantage for lambdas of any complexity. Nor is there any doctest in the missing docstring. A significant problem for a lambda object that requires testing. There are ways to test lambdas with doctest outside a docstring, but it seems simpler to switch to a full function definition. We can't easily apply decorators to it. To do it, we lose the @decorator syntax. We can't use any Python statements in it. In particular, no try-except block is possible. For these reasons, we suggest limiting the use of lambdas to truly trivial situations. Callable Objects A callable object fits the model of a function. The unifying feature of all of the things we've looked at is that they're callable. Functions are the primary example of being callable but objects can also be callable. Callable objects can be subclasses of collections.abc.Callable. Because of Python's flexibility, this isn't a requirement, it's merely a good idea. To be callable, a class only needs to provide a __call__() method. Here's a complete callable class definition: from collections.abc import Callableclass Engine(Callable):   def __call__(self, tach):       return 0.7724*tach**1.0134 We've imported the collections.abc.Callable class. This will provide some assurance that any class that extends this abstract superclass will provide a definition for the __call__() method. This is a handy error-checking feature. Our class extends Callable by providing the needed __call__() method. In this case, the __call__() method performs a calculation on the single parameter value, returning a single result. Here's a callable object built from this class: eng= Engine() This creates a function that we can then use. We can evaluate eng(1000) to get the engine RPMs when the tach reads 1000. Callable objects step-by-step There are two parts to making a function a callable object. We'll emphasize these for folks who are new to object-oriented programming: Define a class. Generally, we make this a subclass of collections.abc.Callable. Technically, we only need to implement a __call__() method. It helps to use the proper superclass because it might help catch a few common mistakes. Create an instance of the class. This instance will be a callable object. The object that's created will be very similar to a defined function. And very similar to a lambda object that's been assigned to a variable. While it will be similar to a def statement, it will have one important additional feature: hysteresis. This can be the source of endless bugs. It can also be a way to improve performance. Callables can have hysteresis Here's an example of a callable object that uses hysteresis as a kind of optimization: class Factorial(Callable):   def __init__(self):       self.previous = {}   def __call__(self, n):       if n not in self.previous:           self.previous[n]= self.compute(n)       return self.previous[n]   def compute(self, n):       if n == 0 : return 1       return n*self.__call__(n-1)Here's how we can use this:>>> fact= Factorial()>>> fact(5)120 We create an instance of the class, and then call the instance to compute a value for us. The initializer The initialization method looks like this:    def __init__(self):       self.previous = {} This function creates a cache of previously computed values. This is a technique called memoization. If we've already computed a result once, it's in the self.previous cache; we don't need to compute it again, we already know the answer. The Callable interface The required __call__() method looks like this:    def __call__(self, n):       if n not in self.previous:           self.previous[n]= self.compute(n)       return self.previous[n] We've checked the memoization cache first. If the value is not there, we're forced to compute the answer, and insert it into the cache. The final answer is always a value in the cache. A common what if question is what if we have a function of multiple arguments? There are two minuscule changes to support more complex arguments. Use def __call__(self, *n): and self.compute(*n). Since we're only computing factorial, there's no need to over-generalize. The Compute method The essential computation has been allocated to a method called compute. It looks like this:    def compute(self, n):       if n == 0: return 1           return n*self.__call__(n-1) This does the real work of the callable object: it computes n!. In this case, we've used a pretty standard recursive factorial definition. This recursion relies on the __call__() method to check the cache for previous values. If we don't expect to compute values larger than 1000! (a 2,568 digit number, by the way) the recursion works nicely. If we think we need to compute really large factorials, we'll need to use a different approach. Execute the following code to compute very large factorials: functools.reduce(operator.mul, range(1,n+1)) Either way, we can depend on the internal memoization to leverage previous results. Note the potential issue Hysteresis—memory of what came before—is available to the callable objects. We call functions and lambdas stateless, where callable objects can be stateful. This may be desirable to optimize performance. We can memoize the previous results or we can design an object that's simply confusing. Consider a function like divmod() that returns two values. We could try to define a callable object that first returns the quotient and on the second call with the same arguments returns the remainder: >>> crazy_divmod(355,113)3>>> crazy_divmod(255,113)16 This is technically possible. But it's crazy. Warning: Stay away. We generally expect idempotence: functions do the same thing each time. Implementing memoization didn't alter the basic idempotence of our factorial function. Generator Functions Here's a fun generator, the Collatz function. The function creates a sequence using a simple pair of rules. We'll could call this rule, Half-Or-Three-Plus-One (HOTPO). We'll call it collatz(): def collatz(n):   if n % 2 == 0:        return n//2   else:       return 3*n+1 Each integer argument yields a next integer. These can form a chain. For example, if we start with collatz(13), we get 40. The value of collatz(40) is 20. Here's the sequence of values: 13 → 40 → 20 → 10 → 5 → 16 → 8 → 4 → 2 → 1At 1, it loops: 1 → 4 → 2 → 1 … Interestingly, all chains seem to lead—eventually—to 1. To explore this, we need a simple function that will build a chain from a given starting value. Successive values Here's a generator function that will build a list object. This iterates through values in the sequence until it reaches 1, when it terminates: def col_list(n):   seq= [n]   while n != 1:       n= collatz(n)       seq.append(n)   return seq This is not wrong. But it's not really the most useful implementation. This always creates a sequence object. In many cases, we don't really want an object, we only want information about the sequence. We might only want the length, or the largest numbers, or the sum. This is where a generator function might be more useful. A generator function yields elements from the sequence instead of building the sequence as a single object. Generator functions To create a generator function, we write a function that has a loop; inside the loop, there's a yield statement. A function with a yield statement is effectively an Iterable object, it can be used in a for statement to produce values. It doesn't create a big list object, it creates the items that can be accumulated into a list or tuple object. A generator function is lazy: it doesn't compute anything unless forced to by another function needing results. We can iterate through as many (or as few) of the results as we need. For example, list(some_generator()) forces all values to be returned. For another example of a lazy generator, look at how range() objects work. If we evaluate range(10), we only get a generator. If we evaluate list(range(10)), we get a list object. The Collatz generator Here's a generator function that will produce sequences of values using the collatz() method rule shown previously: def col_iter(n):   yield n   while n != 1:       n= collatz(n)        yield n When we use this in a for loop or with the list() function, this will yield the argument number. While the number is not 1, it will apply the collatz() function and yield successive values in the chain. When it has yielded 1, it will will terminate. One common pattern for generator functions is to replace all list-accumulation statements with yield statements. Instead of building a list one time at a time, we yield each item. The collatz() function it lazy. We don't get an answer unless we use list() or tuple() or some variation of a for statement context. Using a generator function Here's how this function looks in practice: >>> for i in col_iter(3):…   print(i)3105168421 We've used the generator function in a for loop so that it will yield all of the values until it terminates. Collatz function sequences Now we can do some exploration of this Collatz sequence. Here are a few evaluations of the col_iter() function: >>> list(col_iter(3))[3, 10, 5, 16, 8, 4, 2, 1]>>> list(col_iter(5))[5, 16, 8, 4, 2, 1]>>> list(col_iter(6))[6, 3, 10, 5, 16, 8, 4, 2, 1]>>> list(syracuse_iter(13))[13, 40, 20, 10, 5, 16, 8, 4, 2, 1] There's an interesting pattern here. It seems that from 16, we know the rest. Generalizing this: from any number we've already seen, we know the rest. Wait. This means that memoization might be a big help in exploring the values created by this sequence. When we start combining function design patterns like this, we're doing functional programming. We're stepping outside the box of purely object-oriented Python. Alternate styles Here is an alternative version of the collatz() function: def collatz2(n):   return n//2 if n%2 == 0 else 3*n+1 This simply collapses the if statements into a single if expression and may not help much. We also have this: collatz3= lambda n: n//2 if n%2 == 0 else 3*n+1 We've collapsed the expression into a lambda object. Helpful? Perhaps not. On the other hand, the function doesn't really need all of the overhead of a full function definition and multiple statements. The lambda object seems to capture everything relevant. Functions as object There's a higher-level function that will produce values until some ending condition is met. We can plug in one of the versions of the collatz() function and a termination test into this general-purpose function: def recurse_until(ending, the_function, n):   yield n   while not ending(n):       n= the_function(n)       yield n This requires two plug-in functions, they are as follows: ending() is a function to test to see whether we're done, for example, lambda n: n==1 the_function() is a form of the Collatz function We've completely uncoupled the general idea of recursively applying a function from a specific function and a specific terminating condition. Using the recurs_until() function We can apply this higher-order recurse_until() function like this: >>> recurse_until(lambda n: n==1, syracuse2, 9)<generator object recurse_until at 0x1021278c0> What's that? That's how a lazy generator looks: it didn't return any values because we didn't demand any values. We need to use it in a loop or some kind of expression that iterates through all available values. The list() function, for example, will collect all of the values. Getting the list of values Here's how we make the lazy generator do some work: >>> list(_)[9, 28, 14, 7, 22, 11, 34, 17, 52, 26, 13, 40, 20, 10, 5, 16, 8, 4, 2, 1] The _ variable is the previously computed value. It relieves us from the burden of having to write an assignment statement. We can write an expression, see the results, and know the results were automatically saved in the _ variable. Project Euler #14 Which starting number, under one million, produces the longest chain? Try it without memoization. It's really simple: >>> collatz_len= [len(list(recurse_until(lambda n: n==1, collatz2, i))) ... for i in range(1,11)]>>> results = zip(collatz_len, range(1,11))>>> max(results)(20, 9)>>> list(col_iter(9))[9, 28, 14, 7, 22, 11, 34, 17, 52, 26, 13, 40, 20, 10, 5, 16, 8, 4, 2, 1] We defined collatz_len as a list. We're writing a list comprehension that shows the values built from a generator expression. The generator expression evaluates len(something) for i in range(1,11). This means we'll be collecting ten values into the list, each of which is the length of something. The something is a list object built from the recurse_until(lambda n: n==1, collatz2, i) function that we discussed. This will compute the sequence of Collatz values starting from i and proceeding until the value in the sequence is 1. We've zipped the lengths with the original values of i. This will create pairs of lengths and starting numbers. The maximum length will now have a starting value associated with it so that we can confirm that the results match our expectations. And yes, this Project Euler problem could—in principle—be solved in a single line of code. Will this scale to 100? 1,000? 1,000,000? How much will memoization help? Summary In this article, we've looked at five (or six) kinds of Python callables. They all fit the y = f(x) model of a function to varying degrees. When is it appropriate to use each of these different ways to express the same essential concept? Functions are created with def and return. It shouldn't come as a surprise that this should cover most cases. This allows a docstring comment and doctest test cases. We could call these def functions, since they're built with the def statement. Higher-order functions—map(), filter(), and the itertools library—are generally written as plain-old def functions. They're higher-order because they accept functions as arguments or return functions as results. Otherwise, they're just functions. Function wrappers—len(), divmod(), pow(), str(), and repr()—are function syntax wrapped around object methods. These are def'd functions with very tiny bodies. We use them because a.pow(2) doesn't seem as clear as pow(2,a). Lambdas are appropriate for one-time use of something so simple that it doesn't deserve being wrapped in a def statement body. In some cases, we have a small nugget of code that seems more clear when written as a lambda expression rather than a more complete function definition. Simple filter rules, and simple computations are often more clearly shown as a lambda object. The Callable objects have a special property that other functions lack: hysteresis. They can retain the results of previous calculations. We've used this hysteresis property to implement memoizing. This can be a huge performance boost. Callable objects can be used badly, however, to create objects that have simply bizarre behavior. Most functions should strive for idempotence—the same arguments should yield the same results. Generator functions are created with a def statement and at least one yield statement. These functions are iterable. They can be used in a for statement to examine each resulting value. They can also be used with functions like list(), tuple(), and set() to create an actual object from the iterable sequence of values. We might combine them with higher-order functions to do complex processing, one item at a time. It's important to work with each of these kinds of callables. If you only have one tool—a hammer—then every problem has to be reshaped into a nail before you can solve it. Once you have multiple tools available, you can pick the tools that provides the most succinct and expressive solution to the problem. Resources for Article: Further resources on this subject: Expert Python Programming [article] Python Network Programming Cookbook [article] Learning Python Design Patterns [article]
Read more
  • 0
  • 0
  • 2725

article-image-tour-xcode
Packt
06 Feb 2015
13 min read
Save for later

Tour of Xcode

Packt
06 Feb 2015
13 min read
In this article, written by Jayant Varma, the author of Xcode 6 Essentials, we shall look at Xcode closely as this is going to be the tool you would use quite a lot for all aspects of your app development for Apple devices. It is a good idea to know and be familiar with the interface, the sections, shortcut keys, and so on. (For more resources related to this topic, see here.) Starting Xcode Xcode, like many other Mac applications, is found in the Applications folder or the Launchpad. On starting Xcode, you will be greeted with the launch screen that offers some entry points for working with Xcode. Mostly, you will select Create a new Xcode project or Check out an existing project , if you have an existing project to continue work on. Xcode remembers what it was doing last, so if you had a project or file open, it will open up those windows again. Creating a new project After selecting the Create a new project option, we are guided via a wizard that helps us get started. Selecting the project type The first step is to select what type of project you want to create. At the moment, there are two distinct types of projects, mobile (iOS) or desktop (OS X) that you can create. Within each of those types, you can select the type of project you want. The screenshot displays a standard configuration for iOS application projects. The templates used when the selected type of project is created are self sufficient, that is, when the Run button is pressed, the app compiles and runs. It might do nothing, as this is a minimalistic template. On selecting the type of project, we can select the next step: Setting the project options This step allows selecting the options, namely setting the application name, the organization name, identifier, language, and devices to support. In the past, the language was always set to Objective-C, however with Xcode 6, there are two options: objective-C and Swift Setting the project properties On creation, the main screen is displayed. Here it offers the option to change other details related to the application such as the version number and build. It also allows you to configure the team ID and certificates used for signing the application to test on a mobile device or for distribution to the App Store. It also allows you to set the compatibility for earlier versions. The orientation and app icons, splash screens, and so on are also set from this screen. If you want to set these up later on in the project, it is fine, this can be accessed at any time and does not stop you from development. It needs to be set prior to deploying it on a device or creating an App Store ready application. Xcode overview Let us have a look at the Xcode interface to familiarize ourselves with the same as it would help improve productivity when building your application. The top section immediately following the traffic light (window chrome) displays a Play and Stop button. This allows the project to run and stop. The breadcrumb toolbar displays the project-specific settings with respect to the product and the target. With an iOS project, it could be a particular simulator for iPhone, iPad, and so on, or a physical device (number 5 in the following screenshot). Just under this are vertical areas that are the main content area with all the files, editors, UI, and so on. These can be displayed or hidden as required and can be stacked vertically or horizontally. The distinct areas in Xcode are as follows: Project navigation (number1) Editor and assistant editor (number 2 ) and (number 3 ) Utility/inspector (number 4 ) The toolbar (number 5 ) and (number 6 ) These sections can be switched on and off (shown or hidden) as required to make space for other sections or more screen space to work with: Sections in Xcode The project section The project navigation section has three sub sections, the topmost being the project toolbar that has eight icons. These can be seen as in the following screenshot. The next sub section contains the project files and all the assets required for this project. The bottom most section consists of recently edited files and filters: You can use the keyboard shortcuts to access these areas quickly with the CMD + 1...8 keys. The eight areas available under project navigation are key and for the beginner to Xcode, this could be a bit daunting. When you run the project, the current section might change and display another where you might wonder how to get back to the project (file) navigator. Getting familiar with these is always helpful and the easiest way to navigate between these is the CMD + 1..8 keys. Project navigator ( CMD + 1 ): This displays all of the files, folders, assets, frameworks, and so on that are part of this project. This is displayed as a hierarchical view and is the way that a majority of developers access their files, folders, and so on. Symbol navigator ( CMD + 2 ): This displays all of the classes, members, and methods that are available in them. This is the easiest way to navigate quickly to a method/function, attribute/property. Search navigator ( CMD + 3 ): This allows you to search the project for a particular match. This is quite useful to find and replace text. Issues navigator ( CMD + 4 ): This displays the warning and errors that occur while typing your code or on building and running it. This also displays the results of the static analyzer. Tests navigator ( CMD + 5 ); This displays the tests that you have present in your code either added by yourself or the default ones created with the project. Debug navigator ( CMD + 6 ): This displays the information about the application when you choose to run it. It has some amazing detailed information on CPU usage, memory usage, disk usage, threads, and so on. Breakpoint navigator ( CMD + 7 ): This displays all the breakpoints in your project from all files. This also allows you to create exception and symbolic breakpoints. Log navigator ( CMD + 8 ): This displays a log of all actions carried out, namely compiling, building, and running. This is more useful when used to determine the results of automated builds The editor and assistant editor sections The second area contains the editor and assistant editor sections. These display the code, the XIB (as appropriate), storyboard files, device previews, and so on. Each of the sub sections have a jump bar on the top that relates to files and allow for navigating back and forth in the files and display the location of the file in the workspace. To the right from this is a mini issues navigator that displays all warnings and errors. In the case of the assistant editors, it also displays two buttons: one to add a new assistant editor area and another to close it.   Source code editors While we are looking at the interface, it is worth noting that the Xcode code editor is a very advanced editor with a lot of features, which is now seen as standard with a lot of text editors. Some of the features that make working with Xcode easier are as follows: Code folding : This feature helps to hide code at points such as the function declaration, loops, matching brace brackets, and so on. When a function or portion of code is folded, it hides it from view, thereby allowing you to view other areas of the code that would not be visible unless you scrolled. Syntax highlighting : This is one of the most useful features as it helps you, the developer, to visually, at a glance, differentiate your source code from variables, constants, and strings. Xcode has syntax highlighting for a couple of languages as mentioned earlier. Context help : This is one of the best features whereby when you hover over a word in the source code with OPT pressed, it shows a dotted underline and the cursor changes to a question mark. When you click on a word with the dotted underline and the question mark cursor, it displays a popup with details about that word. It also highlights all instances of that word in the file. The popup details as much information as available. If it is a variable or a function that you have added to the code, then it will display the name of the file where it was declared. If it is a word that is contained in the Apple libraries, then it displays the description and other additional details. Context jump : This is another cool feature that allows jumping to the point of declaration of that word. This is achieved by clicking on a word while keeping the CMD button pressed. In many cases, this is mainly helpful to know how the function is declared and what parameters it expects. It can also be useful to get information on other enumerators and constants used with that function. The jump could be in the same file as where you are editing the code or it could be to the header files where they are declared. Edit all in scope : This is a cool feature where you can edit all of the instances of the word together rather than using search and replace. A case scenario is if you want to change the name of a variable and ensure that all instances you are using in the file are changed but not the ones that are text, then you can use this option to quickly change it. Catching mistakes with fix-it : This is another cool feature in Xcode that will save you a lot of time and hassle. As you type text, Xcode keeps analyzing the code and looking for errors. If you have declared a variable and not used it in your code, Xcode immediately draws attention to it suggesting that the variable is an unused variable. However, if it was supposed to be a pointer and you have declared it without *; Xcode immediately flags it as an error that the interface type cannot be statically allocated. It offers a fix-it solution of inserting * and the code has a greyed * character showing where it will be added. This helps the developer fix commonly overlooked issues such as missing semicolons, missing declarations, or misspelled variable names. Code completion : This is the bit that makes writing code so much easier, type in a few letters of the function name and Xcode pops up a list of functions, constants, methods, and so on that start with those letters and displays all of the required parameters (as applicable) including the return type. When selected, it adds the token placeholders that can be replaced with the actual parameter values. The results might vary from person to person depending on the settings and the speed of the system you run Xcode on. The assistant editor The assistant editor is mainly used to display the counterparts and related files to the file open in the primary editor (generally used when working with Objective-C where the .h or.m files are the related files). The assistant editors track the contents of the editor. Xcode is quite intelligent and knows the corresponding sections and counterparts. When you click on a file, it opens up in the editor. However, pressing the OPT + Shift while clicking on the file, you would be provided with an interactive dialog to select where to open the file. The options include the primary editor or the assistant editor. You can also add assistant editors as required.   Another way to open a file quickly is to use the Open Quickly option, which has a shortcut key of CMD + Shift + O . This displays a textbox that allows accessing a file from the project. The utility/inspector section The last section contains the inspector and library. This section changes based on the type of file selected in the current editor. The inspector has 6 tabs/sections and they are as follows: The file inspector ( CMD + OPT + 1 ): This displays the physical file information for the file selected. For code files, it is the text encoding, the targets that it belongs to, and the physical file path. While for the storyboard, it is the physical file path and allows setting attributes such as auto layout and size classes (new in Xcode 6). The quick help inspector ( CMD + OPT + 2 ): This displays information about the class or object selected. The identity inspector ( CMD + OPT + 3 ): This displays the class name, ID, and others that identify the object selected. The attributes inspector ( CMD + OPT + 4 ): This displays the attributes for the object selected as if it is the initial root view controller, does it extend under the top bars or not, if it has a navigation bar or not, and others. This also displays the user-defined attributes (a new feature with Xcode 6). The size inspector ( CMD + OPT + 5 ): This displays the size of the control selected and the associated constraints that help position it on the container. The connections inspector ( CMD + OPT + 6 ): This displays the connections created in the Interface Builder between the UI and the code. The lower half of this inspector contains four options that help you work efficiently, they are as follows: The file template library : This contains the options to create a new class, protocol. The options that are available when selecting the File | New option from the menu. The code snippets library : This is a wonderful but not widely used option. This can hold code snippets that can help you avoid writing repetitive blocks of code in your app. You can drag and drop the snippet to your code in the editor. This also offers features such as shortcuts, scopes, platforms, and languages. So you can have a shortcut such as appDidLoad (for example) that inserts the code to create and populate a button. This is achieved simply by setting the platform as appropriate to iOS or OS X. After creating a code snippet, as soon as you type the first few characters, the code snippet shows up in the list of autocomplete options; The object library : This is the toolbox that contains all of the controls that you need for creating your UI, be it a button, a label, a Table View, view, View Controller, or anything else. Adding a code snippet is as easy as dragging the selected code from the editor onto the snippet area. It is a little tricky because the moment you start dragging, it could break your selection highlight. You need to select the text, click (hold) and then drag it. The media library : This contains the list of all images and other media types that are available to this project/workspace. Summary In this article, you have seen a quick tour of Xcode, keeping the shortcuts and tips handy as they really do help get things done faster. The code snippets are a wonderful feature that allow for quickly setting up commonly used code with shortcut keywords. Resources for Article: Further resources on this subject: Introducing Xcode Tools for iPhone Development [article] Xcode 4 ios: Displaying Notification Messages [article] Linking OpenCV to an iOS project [article]
Read more
  • 0
  • 0
  • 9665

article-image-event-driven-programming
Packt
06 Feb 2015
22 min read
Save for later

Event-driven Programming

Packt
06 Feb 2015
22 min read
In this article by Alan Thorn author of the book Mastering Unity Scripting will cover the following topics: Events Event management (For more resources related to this topic, see here.) The Update events for MonoBehaviour objects seem to offer a convenient place for executing code that should perform regularly over time, spanning multiple frames, and possibly multiple scenes. When creating sustained behaviors over time, such as artificial intelligence for enemies or continuous motion, it may seem that there are almost no alternatives to filling an Update function with many if and switch statements, branching your code in different directions depending on what your objects need to do at the current time. But, when the Update events are seen this way, as a default place to implement prolonged behaviors, it can lead to severe performance problems for larger and more complex games. On deeper analysis, it's not difficult to see why this would be the case. Typically, games are full of so many behaviors, and there are so many things happening at once in any one scene that implementing them all through the Update functions is simply unfeasible. Consider the enemy characters alone, they need to know when the player enters and leaves their line of sight, when their health is low, when their ammo has expired, when they're standing on harmful terrain, when they're taking damage, when they're moving or not, and lots more. On thinking initially about this range of behaviors, it seems that all of them require constant and continuous attention because enemies should always know, instantly, when changes in these properties occur as a result of the player input. That is, perhaps, the main reason why the Update function seems to be the most suitable place in these situations but there are better alternatives, namely, event-driven programming. By seeing your game and your application in terms of events, you can make considerable savings in performance. This article then considers the issue of events and how to manage them game wide. Events Game worlds are fully deterministic systems; in Unity, the scene represents a shared 3D Cartesian space and timeline inside which finite GameObjects exist. Things only happen within this space when the game logic and code permits them to. For example, objects can only move when there is code somewhere that tells them to do so, and under specific conditions, such as when the player presses specific buttons on the keyboard. Notice from the example that behaviors are not simply random but are interconnected; objects move only when keyboard events occur. There is an important connection established between the actions, where one action entails another. These connections or linkages are referred to as events; each unique connection being a single event. Events are not active but passive; they represent moments of opportunity but not action in themselves, such as a key press, a mouse click, an object entering a collider volume, the player being attacked, and so on. These are examples of events and none of them say what the program should actually do, but only the kind of scenario that just happened. Event-driven programming starts with the recognition of events as a general concept and comes to see almost every circumstance in a game as an instantiation of an event; that is, as an event situated in time, not just an event concept but as a specific event that happens at a specific time. Understanding game events like these is helpful because all actions in a game can then be seen as direct responses to events as and when they happen. Specifically, events are connected to responses; an event happens and triggers a response. Further, the response can go on to become an event that triggers further responses and so on. In other words, the game world is a complete, integrated system of events and responses. Once the world is seen this way, the question then arises as to how it can help us improve performance over simply relying on the Update functions to move behaviors forward on every frame. And the method is simply by finding ways to reduce the frequency of events. Now, stated in this way, it may sound a crude strategy, but it's important. To illustrate, let's consider the example of an enemy character firing a weapon at the player during combat. Throughout the gameplay, the enemy will need to keep track of many properties. Firstly, their health, because when it runs low the enemy should seek out medical kits and aids to restore their health again. Secondly, their ammo, because when it runs low the enemy should seek to collect more and also the enemy will need to make reasoned judgments about when to fire at the player, such as only when they have a clear line of sight. Now, by simply thinking about this scenario, we've already identified some connections between actions that might be identified as events. But before taking this consideration further, let's see how we might implement this behavior using an Update function, as shown in the following code sample 4-1. Then, we'll look at how events can help us improve on that implementation: // Update is called once per frame void Update () {    //Check enemy health    //Are we dead?    if(Health <= 0)    {          //Then perform die behaviour          Die();          return;    }    //Check for health low    if(health <= 20)    {        //Health is low, so find first-aid          RunAndFindHealthRestore();          return;    }    //Check ammo    //Have we run out of ammo?    if(Ammo <= 0)    {          //Then find more          SearchMore();          return;    }    //Health and ammo are fine. Can we see player? If so, shoot    if(HaveLineOfSight)    {            FireAtPlayer();    } } The preceding code sample 4-1 shows a heavy Update function filled with lots of condition checking and responses. In essence, the Update function attempts to merge event handling and response behaviors into one and the results in an unnecessarily expensive process. If we think about the event connections between these different processes (the health and ammo check), we see how the code could be refactored more neatly. For example, ammo only changes on two occasions: when a weapon is fired or when new ammo is collected. Similarly, health only changes on two occasions: when an enemy is successfully attacked by the player or when an enemy collects a first-aid kit. In the first case, there is a reduction, and in the latter case, an increase. Since these are the only times when the properties change (the events), these are the only points where their values need to be validated. See the following code sample 4-2 for a refactored enemy, which includes C# properties and a much reduced Update function: using UnityEngine; using System.Collections; public class EnemyObject : MonoBehaviour {    //-------------------------------------------------------    //C# accessors for private variables    public int Health    {          get{return _health;}          set          {                //Clamp health between 0-100                _health = Mathf.Clamp(value, 0, 100);               //Check if dead                if(_health <= 0)                {                      OnDead();                      return;                }                //Check health and raise event if required                if(_health <= 20)               {                      OnHealthLow();                      return;                }          }    }    //-------------------------------------------------------    public int Ammo    {          get{return _ammo;}          set          {              //Clamp ammo between 0-50              _ammo = Mathf.Clamp(value,0,50);                //Check if ammo empty                if(_ammo <= 0)                {                      //Call expired event                      OnAmmoExpired();                      return;                }          }    }    //-------------------------------------------------------    //Internal variables for health and ammo    private int _health = 100;    private int _ammo = 50;    //-------------------------------------------------------    // Update is called once per frame    void Update ()    {    }    //-------------------------------------------------------    //This event is called when health is low    void OnHealthLow()    {          //Handle event response here    }    //-------------------------------------------------------    //This event is called when enemy is dead    void OnDead()    {        //Handle event response here    }    //-------------------------------------------------------    //Ammo run out event    void OnAmmoExpired()    {        //Handle event response here    }    //------------------------------------------------------- } The enemy class in the code sample 4-2 has been refactored to an event-driven design, where properties such as Ammo and Health are validated not inside the Update function but on assignment. From here, events are raised wherever appropriate based on the newly assigned values. By adopting an event-driven design, we introduce performance optimization and cleanness into our code; we reduce the excess baggage and value checks as found with the Update function in the code sample 4-1, and instead we only allow value-specific events to drive our code, knowing they'll be invoked only at the relevant times. Event management Event-driven programming can make our lives a lot easier. But no sooner than we accept events into the design do we come across a string of new problems that require a thoroughgoing resolution. Specifically, we saw in the code sample 4-2 how C# properties for health and ammo are used to validate and detect for relevant changes and then to raise events (such as OnDead) where appropriate. This works fine in principle, at least when the enemy must be notified about events that happen to itself. However, what if an enemy needed to know about the death of another enemy or needed to know when a specified number of other enemies had been killed? Now, of course, thinking about this specific case, we could go back to the enemy class in the code sample 4-2 and amend it to call an OnDead event not just for the current instance but for all other enemies using functions such as SendMessage. But this doesn't really solve our problem in the general sense. In fact, let's state the ideal case straight away; we want every object to optionally listen for every type of event and to be notified about them as and when they happen, just as easily as if the event had happened to them. So the question that we face now is about how to code an optimized system to allow easy event management like this. In short, we need an EventManager class that allows objects to listen to specific events. This system relies on three central concepts, as follows: Event Listener: A listener refers to any object that wants to be notified about an event when it happens, even its own events. In practice, almost every object will be a listener for at least one event. An enemy, for example, may want notifications about low health and low ammo among others. In this case, it's a listener for at least two separate events. Thus, whenever an object expects to be told when an event happens, it becomes a listener. Event Poster: In contrast to listeners, when an object detects that an event has occurred, it must announce or post a public notification about it that allows all other listeners to be notified. In the code sample 4-2, the enemy class detects the Ammo and Health events using properties and then calls the internal events, if required. But to be a true poster in this sense, we require that the object must raise events at a global level. Event Manager: Finally, there's an overarching singleton Event Manager object that persists across levels and is globally accessible. This object effectively links listeners to posters. It accepts notifications of events sent by posters and then immediately dispatches the notifications to all appropriate listeners in the form of events. Starting event management with interfaces The first or original entity in the event handling system is the listener—the thing that should be notified about specific events as and when they happen. Potentially, a listener could be any kind of object or any kind of class; it simply expects to be notified about specific events. In short, the listener will need to register itself with the Event Manager as a listener for one or more specific events. Then, when the event actually occurs, the listener should be notified directly by a function call. So, technically, the listener raises a type-specificity issue for the Event Manager about how the manager should invoke an event on the listener if the listener could potentially be an object of any type. Of course, this issue can be worked around, as we've seen, using either SendMessage or BroadcastMessage. Indeed, there are event handling systems freely available online, such as NotificationCenter that rely on these functions. However, we'll avoid them using interfaces and use polymorphism instead, as both SendMessage and BroadcastMessage rely heavily on reflection. Specifically, we'll create an interface from which all listener objects derive. More information on the freely available NotificationCenter (C# version) is available from the Unity wiki at http://wiki.unity3d.com/index.php?title=CSharpNotificationCenter. In C#, an interface is like a hollow abstract base class. Like a class, an interface brings together a collection of methods and functions into a single template-like unit. But, unlike a class, an interface only allows you to define function prototypes such as the name, return type, and arguments for a function. It doesn't let you define a function body. The reason being that an interface simply defines the total set of functions that a derived class will have. The derived class may implement the functions however necessary, and the interface simply exists so that other objects can invoke the functions via polymorphism without knowing the specific type of each derived class. This makes interfaces a suitable candidate to create a Listener object. By defining a Listener interface from which all objects will be derived, every object has the ability to be a listener for events. The following code sample 4-3 demonstrates a sample Listener interface: 01 using UnityEngine; 02 using System.Collections; 03 //----------------------------------------------------------- 04 //Enum defining all possible game events 05 //More events should be added to the list 06 public enum EVENT_TYPE {GAME_INIT, 07                                GAME_END, 08                                 AMMO_EMPTY, 09                                 HEALTH_CHANGE, 10                                 DEAD}; 11 //----------------------------------------------------------- 12 //Listener interface to be implemented on Listener classes 13 public interface IListener 14 { 15 //Notification function invoked when events happen 16 void OnEvent(EVENT_TYPE Event_Type, Component Sender,    Object Param = null); 17 } 18 //----------------------------------------------------------- The following are the comments for the code sample 4-3: Lines 06-10: This enumeration should define a complete list of all possible game events that could be raised. The sample code lists only five game events: GAME_INIT, GAME_END, AMMO_EMPTY, HEALTH_CHANGE, and DEAD. Your game will presumably have many more. You don't actually need to use enumerations for encoding events; you could just use integers. But I've used enumerations to improve event readability in code. Lines 13-17: The Listener interface is defined as IListener using the C# interfaces. It supports just one event, namely OnEvent. This function will be inherited by all derived classes and will be invoked by the manager whenever an event occurs for which the listener is registered. Notice that OnEvent is simply a function prototype; it has no body. More information on C# interfaces can be found at http://msdn.microsoft.com/en-us/library/ms173156.aspx. Using the IListener interface, we now have the ability to make a listener from any object using only class inheritance; that is, any object can now declare itself as a listener and potentially receive events. For example, a new MonoBehaviour component can be turned into a listener with the following code sample 4-4. This code uses multiple inheritance, that is, it inherits from two classes. More information on multiple inheritance can be found at http://www.dotnetfunda.com/articles/show/1185/multiple-inheritance-in-csharp: using UnityEngine; using System.Collections; public class MyCustomListener : MonoBehaviour, IListener {    // Use this for initialization    void Start () {}    // Update is called once per frame    void Update () {}    //---------------------------------------    //Implement OnEvent function to receive Events    public void OnEvent(EVENT_TYPE Event_Type, Component Sender, Object Param = null)    {    }    //--------------------------------------- } Creating an EventManager Any object can now be turned into a listener, as we've seen. But still the listeners must register themselves with a manager object of some kind. Thus, it is the duty of the manager to call the events on the listeners when the events actually happen. Let's now turn to the manager itself and its implementation details. The manager class will be called EventManager, as shown in the following code sample 4-5. This class, being a persistent singleton object, should be attached to an empty GameObject in the scene where it will be directly accessible to every other object through a static instance property. More on this class and its usage is considered in the subsequent comments: 001 using UnityEngine; 002 using System.Collections; 003 using System.Collections.Generic; 004 //----------------------------------- 005 //Singleton EventManager to send events to listeners 006 //Works with IListener implementations 007 public class EventManager : MonoBehaviour 008 { 009     #region C# properties 010 //----------------------------------- 011     //Public access to instance 012     public static EventManager Instance 013       { 014             get{return instance;} 015            set{} 016       } 017   #endregion 018 019   #region variables 020       // Notifications Manager instance (singleton design pattern) 021   private static EventManager instance = null; 022 023     //Array of listeners (all objects registered for events) 024     private Dictionary<EVENT_TYPE, List<IListener>> Listeners          = new Dictionary<EVENT_TYPE, List<IListener>>(); 025     #endregion 026 //----------------------------------------------------------- 027     #region methods 028     //Called at start-up to initialize 029     void Awake() 030     { 031             //If no instance exists, then assign this instance 032             if(instance == null) 033           { 034                   instance = this; 035                   DontDestroyOnLoad(gameObject); 036           } 037             else 038                   DestroyImmediate(this); 039     } 040//----------------------------------------------------------- 041     /// <summary> 042     /// Function to add listener to array of listeners 043     /// </summary> 044     /// <param name="Event_Type">Event to Listen for</param> 045     /// <param name="Listener">Object to listen for event</param> 046     public void AddListener(EVENT_TYPE Event_Type, IListener        Listener) 047    { 048           //List of listeners for this event 049           List<IListener> ListenList = null; 050 051           // Check existing event type key. If exists, add to list 052           if(Listeners.TryGetValue(Event_Type,                out ListenList)) 053           { 054                   //List exists, so add new item 055                   ListenList.Add(Listener); 056                   return; 057           } 058 059           //Otherwise create new list as dictionary key 060           ListenList = new List<IListener>(); 061           ListenList.Add(Listener); 062           Listeners.Add(Event_Type, ListenList); 063     } 064 //----------------------------------------------------------- 065       /// <summary> 066       /// Function to post event to listeners 067       /// </summary> 068       /// <param name="Event_Type">Event to invoke</param> 069       /// <param name="Sender">Object invoking event</param> 070       /// <param name="Param">Optional argument</param> 071       public void PostNotification(EVENT_TYPE Event_Type,          Component Sender, Object Param = null) 072       { 073           //Notify all listeners of an event 074 075           //List of listeners for this event only 076           List<IListener> ListenList = null; 077 078           //If no event exists, then exit 079           if(!Listeners.TryGetValue(Event_Type,                out ListenList)) 080                   return; 081 082             //Entry exists. Now notify appropriate listeners 083             for(int i=0; i<ListenList.Count; i++) 084             { 085                   if(!ListenList[i].Equals(null)) 086                   ListenList[i].OnEvent(Event_Type, Sender, Param); 087             } 088     } 089 //----------------------------------------------------------- 090     //Remove event from dictionary, including all listeners 091     public void RemoveEvent(EVENT_TYPE Event_Type) 092     { 093           //Remove entry from dictionary 094           Listeners.Remove(Event_Type); 095     } 096 //----------------------------------------------------------- 097       //Remove all redundant entries from the Dictionary 098     public void RemoveRedundancies() 099     { 100             //Create new dictionary 101             Dictionary<EVENT_TYPE, List<IListener>>                TmpListeners = new Dictionary                <EVENT_TYPE, List<IListener>>(); 102 103             //Cycle through all dictionary entries 104             foreach(KeyValuePair<EVENT_TYPE, List<IListener>>                Item in Listeners) 105             { 106                   //Cycle all listeners, remove null objects 107                   for(int i = Item.Value.Count-1; i>=0; i--) 108                   { 109                         //If null, then remove item 110                         if(Item.Value[i].Equals(null)) 111                                 Item.Value.RemoveAt(i); 112                   } 113 114           //If items remain in list, then add to tmp dictionary 115                   if(Item.Value.Count > 0) 116                         TmpListeners.Add (Item.Key,                              Item.Value); 117             } 118 119             //Replace listeners object with new dictionary 120             Listeners = TmpListeners; 121     } 122 //----------------------------------------------------------- 123       //Called on scene change. Clean up dictionary 124       void OnLevelWasLoaded() 125       { 126           RemoveRedundancies(); 127       } 128 //----------------------------------------------------------- 129     #endregion 130 } More information on the OnLevelWasLoaded event can be found at http://docs.unity3d.com/ScriptReference/MonoBehaviour.OnLevelWasLoaded.html. The following are the comments for the code sample 4-5: Line 003: Notice the addition of the System.Collections.Generic namespace giving us access to additional mono classes, including the Dictionary class. This class will be used throughout the EventManager class. In short, the Dictionary class is a special kind of 2D array that allows us to store a database of values based on key-value pairing. More information on the Dictionary class can be found at http://msdn.microsoft.com/en-us/library/xfhwa508%28v=vs.110%29.aspx. Line 007: The EventManager class is derived from MonoBehaviour and should be attached to an empty GameObject in the scene where it will exist as a persistent singleton. Line 024: A private member variable Listeners is declared using a Dictionary class. This structure maintains a hash-table array of key-value pairs, which can be looked up and searched like a database. The key-value pairing for the EventManager class takes the form of EVENT_TYPE and List<Component>. In short, this means that a list of event types can be stored (such as HEALTH_CHANGE), and for each type there could be none, one, or more components that are listening and which should be notified when the event occurs. In effect, the Listeners member is the primary data structure on which the EventManager relies to maintain who is listening for what. Lines 029-039: The Awake function is responsible for the singleton functionality, that is, to make the EventManager class into a singleton object that persists across scenes. Lines 046-063: The AddListener method of EventManager should be called by a Listener object once for each event for which it should listen. The method accepts two arguments: the event to listen for (Event_Type) and a reference to the listener object itself (derived from IListener), which should be notified if and when the event happens. The AddListener function is responsible for accessing the Listeners dictionary and generating a new key-value pair to store the connection between the event and the listener. Lines 071-088: The PostNotification function can be called by any object, whether a listener or not, whenever an event is detected. When called, the EventManager cycles all matching entries in the dictionary, searching for all listeners connected to the current event, and notifies them by invoking the OnEvent method through the IListener interface. Lines 098-127: The final methods for the EventManager class are responsible for maintaining data integrity of the Listeners structure when a scene change occurs and the EventManager class persists. Although the EventManager class persists across scenes, the listener objects themselves in the Listeners variable may not do so. They may get destroyed on scene changes. If so, scene changes will invalidate some listeners, leaving the EventManager with invalid entries. Thus, the RemoveRedundancies method is called to find and eliminate all invalid entries. The OnLevelWasLoaded event is invoked automatically by Unity whenever a scene change occurs. More information on the OnLevelWasLoaded event can be found online at: http://docs.unity3d.com/ScriptReference/MonoBehaviour.OnLevelWasLoaded.html. #region and #endregion The two preprocessor directives #region and #endregion (in combination with the code folding feature) can be highly useful for improving the readability of your code and also for improving the speed with which you can navigate the source file. They add organization and structure to your source code without affecting its validity or execution. Effectively, #region marks the top of a code block and #endregion marks the end. Once a region is marked, it becomes foldable, that is, it becomes collapsible using the MonoDevelop code editor, provided the code folding feature is enabled. Collapsing a region of code is useful for hiding it from view, which allows you to concentrate on reading other areas relevant to your needs, as shown in the following screenshot: Enabling code folding in MonoDevelop To enable code folding in MonoDevelop, select Options in Tools from the application menu. This displays the Options window. From here, choose the General tab in the Text Editor option and click on Enable code folding as well as Fold #regions by default. Using EventManager Now, let's see how to put the EventManager class to work in a practical context from the perspective of listeners and posters in a single scene. First, to listen for an event (any event) a listener must register itself with the EventManager singleton instance. Typically, this will happen once and at the earliest opportunity, such as the Start function. Do not use the Awake function; this is reserved for an object's internal initialization as opposed to the functionality that reaches out beyond the current object to the states and setup of others. See the following code sample 4-6 and notice that it relies on the Instance static property to retrieve a reference to the active EventManager singleton: //Called at start-up void Start() { //Add myself as listener for health change events EventManager.Instance.AddListener(EVENT_TYPE.HEALTH_CHANGE, this); } Having registered listeners for one or more events, objects can then post notifications to EventManager as events are detected, as shown in the following code sample 4-7: public int Health { get{return _health;} set {    //Clamp health between 0-100    _health = Mathf.Clamp(value, 0, 100);    //Post notification - health has been changed   EventManager.Instance. PostNotification(EVENT_TYPE.HEALTH_CHANGE, this, _health); } } Finally, after a notification is posted for an event, all the associated listeners are updated automatically through EventManager. Specifically, EventManager will call the OnEvent function of each listener, giving listeners the opportunity to parse event data and respond where needed, as shown in the following code sample 4-7: //Called when events happen public void OnEvent(EVENT_TYPE Event_Type, Component Sender, object Param = null) { //Detect event type switch(Event_Type) {    case EVENT_TYPE.HEALTH_CHANGE:          OnHealthChange(Sender, (int)Param);    break; } } Summary This article focused on the manifold benefits available for your applications by adopting an event-driven framework consistently through the EventManager class. In implementing such a manager, we were able to rely on either interfaces or delegates, and either method is powerful and extensible. Specifically, we saw how it's easy to add more and more functionality into an Update function but how doing this can lead to severe performance issues. Better is to analyze the connections between your functionality to refactor it into an event-driven framework. Essentially, events are the raw material of event-driven systems. They represent a necessary connection between one action (the cause) and another (the response). To manage events, we created the EventManager class—an integrated class or system that links posters to listeners. It receives notifications from posters about events as and when they happen and then immediately dispatches a function call to all listeners for the event. Resources for Article: Further resources on this subject: Customizing skin with GUISkin [Article] 2D Twin-stick Shooter [Article] Components in Unity [Article]
Read more
  • 0
  • 0
  • 6437
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-lync-2013-hybrid-and-lync-online
Packt
06 Feb 2015
27 min read
Save for later

Lync 2013 Hybrid and Lync Online

Packt
06 Feb 2015
27 min read
In this article, by the authors, Fabrizio Volpe, Alessio Giombini, Lasse Nordvik Wedø, and António Vargas of the book, Lync Server Cookbook, we will cover the following recipes: Introducing Lync Online Administering with the Lync Admin Center Using Lync Online Remote PowerShell Using Lync Online cmdlets Introducing Lync in a hybrid scenario Planning and configuring a hybrid deployment Moving users to the cloud Moving users back on-premises Debugging Lync Online issues (For more resources related to this topic, see here.) Introducing Lync Online Lync Online is part of the Office 365 offer and provides online users with the same Instant Messaging (IM), presence, and conferencing features that we would expect from an on-premises deployment of Lync Server 2013. Enterprise Voice, however, is not available on Office 365 tenants (or at least, it is available only with limitations regarding both specific Office 365 plans and geographical locations). There is no doubt that forthcoming versions of Lync and Office 365 will add what is needed to also support all the Enterprise Voice features in the cloud. Right now, the best that we are able to achieve is to move workloads, homing a part of our Lync users (the ones with no telephony requirements) in Office 365, while the remaining Lync users are homed on-premises. These solutions might be interesting for several reasons, including the fact that we can avoid the costs of expanding our existing on-premises resources by moving a part of our Lync-enabled users to Office 365. The previously mentioned configuration, which involves different kinds of Lync tenants, is called a hybrid deployment of Lync, and we will see how to configure it and move our users from online to on-premises and vice versa. In this Article, every time we talk about Lync Online and Office 365, we will assume that we have already configured an Office tenant. Administering with the Lync Admin Center Lync Online provides the Lync Admin Center (LAC), a dedicated control panel, to manage Lync settings. To open it, access the Office 365 portal and select Service settings, Lync, and Manage settings in the Lync admin center, as shown in the following screenshot: LAC, if you compare it with the on-premises Lync Control Panel (or with the Lync Management Shell), offers few options. For example, it is not possible to create or delete users directly inside Lync. We will see some of the tasks we are able to perform in LAC, and then, we will move to the (more powerful) Remote PowerShell. There is an alternative path to open LAC. From the Office 365 portal, navigate to Users & Groups | Active Users. Select a user, after which you will see a Quick Steps area with an Edit Lync Properties link that will open the user-editable part of LAC. How to do it... LAC is divided into five areas: users, organization, dial-in conferencing, meeting invitation, and tools, as you can see in the following screenshot: The Users panel will show us the configuration of the Lync Online enabled users. It is possible to modify the settings with the Edit option (the small pencil icon on the right): I have tried to summarize all the available options (inside the general, external communications, and dial-in conferencing tabs) in the following screenshot: Some of the user's settings are worth a mention; in the General tab, we have the following:    The Record Conversations and meetings option enables the Start recording option in the Lync client    The Allow anonymous attendees to dial-out option controls whether the anonymous users that are dialing-in to a conference are required to call the conferencing service directly or are authorized for callback    The For compliance, turn off non-archived features option disables Lync features that are not recorded by In-Place Hold for Exchange When you place an Exchange 2013 mailbox on In-Place Hold or Litigation Hold, the Microsoft Lync 2013 content (instant messaging conversations and files shared in an online meeting) is archived in the mailbox. In the dial-in conferencing tab, we have the configuration required for dial-in conferencing. The provider's drop-down menu shows a list of third parties that are able to deliver this kind of feature. The Organization tab manages privacy for presence information, push services, and external access (the equivalent of the Lync federation on-premises). If you enable external access, we will have the option to turn on Skype federation, as we can see in the following screenshot: The Dial-In Conferencing option is dedicated to the configuration of the external providers. The Meeting Invitation option allows the user to customize the Lync Meeting invitation. The Tools options offer a collection of troubleshooting resources. See also For details about Exchange In-Place Hold, see the TechNet post In-Place Hold and Litigation Hold at http://technet.microsoft.com/en-us/library/ff637980(v=exchg.150).aspx. Using Lync Online Remote PowerShell The possibility to manage Lync using Remote PowerShell on a distant deployment has been available since Lync 2010. This feature has always required a direct connection from the management station to the Remote Lync, and a series of steps that is not always simple to set up. Lync Online supports Remote PowerShell using a dedicated (64-bit only) PowerShell module, the Lync Online Connector. It is used to manage online users, and it is interesting because there are many settings and automation options that are available only through PowerShell. Getting ready Lync Online Connector requires one of the following operating systems: Windows 7 (with Service Pack 1), Windows Server 2008 R2, Windows Server 2012, Windows Server 2012 R2, Windows 8, or Windows 8.1. At least PowerShell 3.0 is needed. To check it, we can use the $PSVersionTable variable. The result will be like the one in the following screenshot (taken on Windows 8.1, which uses PowerShell 4.0): How to do it... Download Windows PowerShell Module for Lync Online from the Microsoft site at http://www.microsoft.com/en-us/download/details.aspx?id=39366 and install it. It is useful to store our Office 365 credentials in an object (it is possible to launch the cmdlets at step 3 anyway, and we will be required with the Office 365 administrator credentials, but using this method, we will have to insert the authentication information again every time it is required). We can use the $credential = Get-Credential cmdlet in a PowerShell session. We will be prompted for our username and password for Lync Online, as shown in the following screenshot: To use the Online Connector, open a PowerShell session and use the New-CsOnlineSession cmdlet. One of the ways to start a remote PowerShell session is $session = New-CsOnlineSession -Credential $credential. Now, we need to import the session that we have created with Lync Online inside PowerShell, with the Import-PSSession $session cmdlet. A temporary Windows PowerShell module will be created, which contains all the Lync Online cmdlets. The name of the temporary module will be similar to the one we can see in the following screenshot: Now, we will have the cmdlets of the Lync Online module loaded in memory, in addition to any command that we already have available in PowerShell. How it works... The feature is based on a PowerShell module, the LyncOnlineConnector, shown in the following screenshot: It contains only two cmdlets, the Set-WinRMNetworkDelayMS and New-CsOnlineSession cmdlets. The latter will load the required cmdlets in memory. As we have seen in the previous steps, the Online Connector adds the Lync Online PowerShell cmdlets to the ones already available. This is something we will use when talking about hybrid deployments, where we will start from the Lync Management Shell and then import the module for Lync Online. It is a good habit to verify (and close) your previous remote sessions. This can be done by selecting a specific session (using Get-PSSession and then pointing to a specific session with the Remove-PSSession statement) or closing all the existing ones with the Get-PSSession | Remove-PSSession cmdlet. In the previous versions of the module, Microsoft Online Services Sign-In Assistant was required. This prerequisite was removed from the latest version. There's more... There are some checks that we are able to perform when using the PowerShell module for Lync Online. By launching the New-CsOnlineSession cmdlet with the –verbose switch, we will see all the messages related to the opening of the session. The result should be similar to the one shown in the following screenshot: Another verification comes from the Get-Command -Module tmp_gffrkflr.ufz command, where the module name (in this example, tmp_gffrkflr.ufz) is the temporary module we saw during the Import-PSSession step. The output of the command will show all the Lync Online cmdlets that we have loaded in memory. The Import-PSSession cmdlet imports all commands except the ones that have the same name of a cmdlet that already exists in the current PowerShell session. To overwrite the existing cmdlets, we can use the -AllowClobber parameter. See also During the introduction of this section, we also discussed the possibility to administer on-premises, remote Lync Server 2013 deployment with a remote PowerShell session. John Weber has written a great post about it in his blog Lync 2013 Remote Admin with PowerShell at http://tsoorad.blogspot.it/2013/10/lync-2013-remote-admin-with-powershell.html, which is helpful if you want to use the previously mentioned feature. Using Lync Online cmdlets In the previous recipe, we outlined the steps required to establish a remote PowerShell session with Lync Online. We have less than 50 cmdlets, as shown in the result of the Get-Command -Module command in the following screenshot: Some of them are specific for Lync Online, such as the following: Get-CsAudioConferencingProvider Get-CsOnlineUser Get-CsTenant Get-CsTenantFederationConfiguration Get-CsTenantHybridConfiguration Get-CsTenantLicensingConfiguration Get-CsTenantPublicProvider New-CsEdgeAllowAllKnownDomains New-CsEdgeAllowList New-CsEdgeDomainPattern Set-CsTenantFederationConfiguration Set-CsTenantHybridConfiguration Set-CsTenantPublicProvider Update-CsTenantMeetingUrl All the remaining cmdlets can be used either with Lync Online or with the on-premises version of Lync Server 2013. We will see the use of some of the previously mentioned cmdlets. How to do it... The Get-CsTenant cmdlet will list Lync Online tenants configured for use in our organization. The output of the command includes information such as the preferred language, registrar pool, domains, and assigned plan. The Get-CsTenantHybridConfiguration cmdlet gathers information about the hybrid configuration of Lync. Management of the federation capability for Lync Online (the feature that enables Instant Messaging and Presence information exchange with users of other domains) is based on the allowed domain and blocked domain lists, as we can see in the organization and external communications screen of LAC, shown in the following screenshot: There are similar ways to manage federation from the Lync Online PowerShell, but it required to put together different statements as follows:     We can use an accept all domains excluding the ones in the exceptions list approach. To do this, we have put the New-CsEdgeAllowAllKnownDomains cmdlet inside a variable. Then, we can use the Set-CsTenantFederationConfiguration cmdlet to allow all the domains (except the ones in the block list) for one of our domains on a tenant. We can use the example on TechNet (http://technet.microsoft.com/en-us/library/jj994088.aspx) and integrate it with Get-CsTenant.     If we prefer, we can use a block all domains but permit the ones in the allow list approach. It is required to define a domain name (pattern) for every domain to allow the New-CsEdgeDomainPattern cmdlet, and each one of them will be saved in a variable. Then, the New-CsEdgeAllowList cmdlet will create a list of allowed domains from the variables. Finally, the Set-CsTenantFederationConfiguration cmdlet will be used. The domain we will work on will be (again) cc3b6a4e-3b6b-4ad4-90be-6faa45d05642. The example on Technet (http://technet.microsoft.com/en-us/library/jj994023.aspx) will be used: $x = New-CsEdgeDomainPattern -Domain "contoso.com" $y = New-CsEdgeDomainPattern -Domain "fabrikam.com" $newAllowList = New-CsEdgeAllowList -AllowedDomain $x,$y Set-CsTenantFederationConfiguration -Tenant " cc3b6a4e-3b6b-4ad4-90be-6faa45d05642" -AllowedDomains $newAllowList The Get-CsOnlineUser cmdlet provides information about users enabled on Office 365. The result will show both users synced with Active Directory and users homed in the cloud. The command supports filters to limit the output; for example, the Get-CsOnlineUser -identity fab will gather information about the user that has alias = fab. This is an account synced from the on-premises Directory Services, so the value of the DirSyncEnabled parameter will be True. See also All the cmdlets of the Remote PowerShell for Lync Online are listed in the TechNet post Lync Online cmdlets at http://technet.microsoft.com/en-us/library/jj994021.aspx. This is the main source of details on the single statement. Introducing Lync in a hybrid scenario In a Lync hybrid deployment, we have the following: User accounts and related information homed in the on-premises Directory Services and replicated to Office 365. A part of our Lync users that consume on-premises resources and a part of them that use online (Office 365 / Lync Online) resources. The same (public) domain name used both online and on-premises (Lync-split DNS). Other Office 365 services and integration with other applications available to all our users, irrespective of where their Lync is provisioned. One way to define Lync hybrid configuration is by using an on-premises Lync deployment federated with an Office 365 / Lync Online tenant subscription. While it is not a perfect explanation, it gives us an idea of the scenario we are talking about. Not all the features of Lync Server 2013 (especially the ones related to Enterprise Voice) are available to Lync Online users. The previously mentioned motivations, along with others (due to company policies, compliance requirements, and so on), might recommend a hybrid deployment of Lync as the best available solution. What we have to clarify now is how to make those users on different deployments talk to each other, see each other's presence status, and so on. What we will see in this section is a high-level overview of the required steps. The Planning and configuring a hybrid deployment recipe will provide more details about the individual steps. The list of steps here is the one required to configure a hybrid deployment, starting from Lync on-premises. In the following sections, we will also see the opposite scenario (with our initial deployment in the cloud). How to do it... It is required to have an available Office 365 tenant configuration. Our subscription has to include Lync Online. We have to configure an Active Directory Federation Services (AD FS) server in our domain and make it available to the Internet using a public FQDN and an SSL certificate released from a third-party certification authority. Office 365 must be enabled to synchronize with our company's Directory Services, using Active Directory Sync. Our Office 365 tenant must be federated. The last step is to configure Lync for a hybrid deployment. There's more... One of the requirements for a hybrid distribution of Lync is an on-premises deployment of Lync Server 2013 or Lync Server 2010. For Lync Server 2010, it is required to have the latest available updates installed, both on the Front Ends and on the Edge servers. It is also required to have the Lync Server 2013 administrative tools installed on a separate server. More details about supported configuration are available on the TechNet post Planning for Lync Server 2013 hybrid deployments at http://technet.microsoft.com/en-us/library/jj205403.aspx. DNS SRV records for hybrid deployments, _sipfederationtls._tcp.<domain> and _sip._tls.<domain>, should point to the on-premises deployment. The lyncdiscover. <domain> record will point to the FQDN of the on-premises reverse proxy server. The _sip._tls. <domain> SRV record will resolve to the public IP of the Access Edge service of Lync on-premises. Depending on the kind of service we are using for Lync, Exchange, and SharePoint, only a part of the features related to the integration with the additional services might be available. For example, skills search is available only if we are using Lync and SharePoint on-premises. The following TechNet post Supported Lync Server 2013 hybrid configurations at http://technet.microsoft.com/en-us/library/jj945633.aspx offers a matrix of features / service deployment combinations. See also Interesting information about Lync Hybrid configuration is presented in sessions available on Channel9 and coming from the Lync Conference 2014 (Lync Online Hybrid Deep Dive at http://channel9.msdn.com/Events/Lync-Conference/Lync-Conference-2014/ONLI302) and from TechEd North America 2014 (Microsoft Lync Online Hybrid Deep Dive at http://channel9.msdn.com/Events/TechEd/NorthAmerica/2014/OFC-B341#fbid=). Planning and configuring a hybrid deployment The planning phase for a hybrid deployment starts from a simple consideration: do we have an on-premises deployment of Lync Server? If the previously mentioned scenario is true, do we want to move users to the cloud or vice versa? Although the first situation is by far the most common, we have to also consider the case in which we have our first deployment in the cloud. How to do it... This step is all that is required for the scenario that starts from Lync Online. We have to completely deploy our Lync on-premises. Establish a remote PowerShell session with Office 365. Use the shared SIP address cmdlet Set-CsTenantFederationConfiguration -SharedSipAddressSpace $True to enable Office 365 to use a Shared Session Initiation Protocol (SIP) address space with our on-premises deployment. To verify this, we can use the Get-CsTenantFederationConfiguration command. The SharedSipAddressSpace value should be set to True. All the following steps are for the scenario that starts from the on-premises Lync deployment. After we have subscribed with a tenant, the first step is to add the public domain we use for our Lync users to Office 365 (so that we can split it on the two deployments). To access the Office 365 portal, select Domains. The next step is Specify a domain name and confirm ownership. We will be required to type a domain name. If our domain is hosted on some specific providers (such as GoDaddy), the verification process can be automated, or we have to proceed manually. The process requires to add one DNS record (TXT or MX), like the ones shown in the following screenshot: If we need to check our Office 365 and on-premises deployments before continuing with the hybrid deployment, we can use the Setup Assistant for Office 365. The tool is available inside the Office 365 portal, but we have to launch it from a domain-joined computer (the login must be performed with the domain administrative credentials). In the Setup menu, we have a Quick Start and an Extend Your Setup option (we have to select the second one). The process can continue installing an app or without software installation, as shown in the following screenshot: The app (which makes the assessment of the existing deployment easier) is installed by selecting Next in the previous screen (it requires at least Windows 7 with Service Pack 1, .NET Framework 3.5, and PowerShell 2.0). Synchronization with the on-premises Active Directory is required. This last step federates Lync Server 2013 with Lync Online to allow communication between our users. The first cmdlet to use is Set-CSAccessEdgeConfiguration -AllowOutsideUsers 1 -AllowFederatedUsers 1 -UseDnsSrvRouting -EnablePartnerDiscovery 1. Note that the -EnablePartnerDiscovery parameter is required. Setting it to 1 enables automatic discovery of federated partner domains. It is possible to set it to 0. The second required cmdlet is New-CSHostingProvider -Identity LyncOnline -ProxyFqdn "sipfed.online.lync.com" -Enabled $true -EnabledSharedAddressSpace $true -HostsOCSUsers $true –VerificationLevel UseSourceVerification -IsLocal $false -AutodiscoverUrl https://webdir.online.lync.com/Autodiscover/AutodiscoverService.svc/root. The result of the commands is shown in the following screenshot: If Lync Online is already defined, we have to use the Set- CSHostingProvider cmdlet, or we can remove it (Remove-CsHostingProvider -Identity LyncOnline) and then create it using the previously mentioned cmdlet. There's more... In the Lync hybrid scenario, users created in the on-premises directory are replicated to the cloud, while users generated in the cloud will not be replicated on-premises. Lync Online users are managed using the Office 365 portal, while the users on-premises are managed using the usual tools (Lync Control Panel and Lync Management Shell). Moving users to the cloud By moving users from Lync on-premises to the cloud, we will lose some of the parameters. The operation requires the Lync administrative tools and the PowerShell module for Lync Online to be installed on the same computer. If we install the module for Lync Online before the administrative tools for Lync 2013 Server, the OCSCore.msi file overwrites the LyncOnlineConnector.ps1 file, and New-CsOnlineSession will require a -TargetServer parameter. In this situation, we have to reinstall the Lync Online module (see the following post on the Microsoft support site at http://support.microsoft.com/kb/2955287). Getting ready Remember that to move the user to Lync Online, they must be enabled for both Lync Server on-premises and Lync Online (so we have to assign the user a license for Lync Online by using the Office 365 portal). Users with no assigned licenses will show the error Move-CsUser : HostedMigration fault: Error=(507), Description=(User must has an assigned license to use Lync Online. For more details, refer to the Microsoft support site at http://support.microsoft.com/kb/2829501. How to do it... Open a new Lync Management Shell session and launch the remote session on Office 365 with the cmdlets' sequence we saw earlier. We have to add the –AllowClobber parameter so that the Lync Online module's cmdlets are able to overwrite the corresponding Lync Management Shell cmdlets: $credential = Get-Credential $session = New-CsOnlineSession -Credential $credential Import-PSSession $session -AllowClobber Open the Lync Admin Center (as we have seen in the dedicated section) by going to Service settings | Lync | Manage settings in the Lync Admin Center, and copy the first part of the URL, for example, https://admin0e.online.lync.com. Add the following string to the previous URL /HostedMigration/hostedmigrationservice.svc (in our example, the result will be https://admin0a.online.lync.com/HostedMigration/hostedmigrationservice.svc). The following cmdlet will move users from Lync on-premises to Lync Online. The required parameters are the identity of the Lync user and the URL that we prepared in step 2. The user identity is fabrizio.volpe@absoluteuc.biz: Move-CsUser -Identity fabrizio.volpe@absoluteuc.biz –Target sipfed.online.lync.com -Credential $creds -HostedMigrationOverrideUrl https://admin0e.online.lync.com/HostedMigration/hostedmigrationservice.sVc Usually, we are required to insert (again) the Office 365 administrative credentials, after which we will receive a warning about the fact that we are moving our user to a different version of the service, like the one in the following screenshot: See the There's more... section of this recipe for details about user information that is migrated to Lync Online. We are able to quickly verify whether the user has moved to Lync Online by using the Get-CsUser | fl DisplayName,HostingProvider,RegistrarPool,SipAddress command. On-premises HostingProvider is equal to SRV: and RegistrarPool is madhatter.wonderland.lab (the name of the internal Lync Front End). Lync Online values are HostingProvider : sipfed.online.lync.com, and leave RegistrarPool empty, as shown in the following screenshot (the user Fabrizio is homed on-premises, while the user Fabrizio volpe is homed on the cloud): There's more... If we plan to move more than one user, we have to add a selection and pipe it before the cmdlet we have already used, removing the –identity parameter. For example, to move all users from an Organizational Unit (OU), (for example, the LyncUsers in the Wonderland.Lab domain) to Lync Online, we can use Get-CsUser -OU "OU=LyncUsers,DC=wonderland,DC=lab"| Move-CsUser -Target sipfed.online.lync.com -Credential $creds -HostedMigrationOverrideUrl https://admin0e.online.lync.com/HostedMigration/hostedmigrationservice.sVc. We are also able to move users based on a parameter to match using the Get-CsUser –Filter cmdlet. As we mentioned earlier, not all the user information is migrated to Lync Online. Migration contact list, groups, and access control lists are migrated, while meetings, contents, and schedules are lost. We can use the Lync Meeting Update Tool to update the meeting links (which have changed when our user's home server has changed) and automatically send updated meeting invitations to participants. There is a 64-bit version (http://www.microsoft.com/en-us/download/details.aspx?id=41656) and a 32-bit version (http://www.microsoft.com/en-us/download/details.aspx?id=41657) of the previously mentioned tool. Moving users back on-premises It is possible to move back users that have been moved from the on-premises Lync deployment to the cloud, and it is also possible to move on-premises users that have been defined and enabled directly in Office 365. In the latter scenario, it is important to create the user also in the on-premises domain (Directory Service). How to do it… The Lync Online user must be created in the Active Directory (for example, I will define the BornOnCloud user that already exists in Office 365). The user must be enabled in the on-premises Lync deployment, for example, using the Lync Management Shell with the following cmdlet: Enable-CsUser -Identity "BornOnCloud" -SipAddress "SIP:BornOnCloud@absoluteuc.biz" -HostingProviderProxyFqdn "sipfed.online.lync.com" Sync the Directory Services. Now, we have to save our Office 365 administrative credentials in a $cred = Get-Credential variable and then move the user from Lync Online to the on-premises Front End using the Lync Management Shell (the -HostedMigrationOverrideURL parameter has the same value that we used in the previous section): Move-CsUser -Identity BornOnCloud@absoluteuc.biz -Target madhatter.wonderland.lab -Credential $cred -HostedMigrationOverrideURL https://admin0e.online.lync.com/HostedMigration/hostedmigrationservice.svc The Get-CsUser | fl DisplayName,HostingProvider,RegistrarPool,SipAddress cmdlet is used to verify whether the user has moved as expected. See also Guy Bachar has published an interesting post on his blog Moving Users back to Lync on-premises from Lync Online (http://guybachar.wordpress.com/2014/03/31/moving-users-back-to-lync-on-premises-from-lync-online/), where he shows how he solved some errors related to the user motion by modifying the HostedMigrationOverrideUrl parameter. Debugging Lync Online issues Getting ready When moving from an on-premises solution to a cloud tenant, the first aspect we have to accept is that we will not have the same level of control on the deployment we had before. The tools we will list are helpful in resolving issues related to Lync Online, but the level of understanding on an issue they give to a system administrator is not the same we have with tools such as Snooper or OCSLogger. Knowing this, the more users we will move to the cloud, the more we will have to use the online instruments. How to do it… The Set up Lync Online external communications site on Microsoft Support (http://support.microsoft.com/common/survey.aspx?scid=sw;en;3592&showpage=1) is a guided walk-through that helps in setting up communication between our Lync Online users and external domains. The tool provides guidelines to assist in the setup of Lync Online for small to enterprise businesses. As you can see in the following screenshot, every single task is well explained: The Remote Connectivity Analyzer (RCA) (https://testconnectivity.microsoft.com/) is an outstanding tool to troubleshoot both Lync on-premises and Lync Online. The web page includes tests to analyze common errors and misconfigurations related to Microsoft services such as Exchange, Lync, and Office 365. To test different scenarios, it is necessary to use various network protocols and ports. If we are working on a firewall-protected network, using the RCA, we are also able to test services that are not directly available to us. For Lync Online, there are some tests that are especially interesting; in the Office 365 tab, the Office 365 General Tests section includes the Office 365 Lync Domain Name Server (DNS) Connectivity Test and the Office 365 Single Sign-On Test, as shown in the following screenshot: The Single Sign-On test is really useful in a scenario. The test requires our domain username and password, both synced with the on-premises Directory Services. The steps include searching the FQDN of our AD FS server on an Internet DNS, verifying the certificate and connectivity, and then validating the token that contains the credentials. The Client tab offers to download the Microsoft Connectivity Analyzer Tool and the Microsoft Lync Connectivity Analyzer Tool, which we will see in the following two dedicated steps: The Microsoft Connectivity Analyzer Tool makes many of the tests we see in the RCA available on our desktop. The list of prerequisites is provided in the article Microsoft Connectivity Analyzer Tool (http://technet.microsoft.com/library/jj851141(v=exchg.80).aspx), and includes Windows Vista/Windows 2008 or later versions of the operating system, .NET Framework 4.5, and an Internet browser, such as Internet Explorer, Chrome, or Firefox. For the Lync tests, a 64-bit operating system is mandatory, and the UCMA runtime 4.0 is also required (it is part of Lync Server 2013 setup, and is also available for download at http://www.microsoft.com/en-us/download/details.aspx?id=34992). The tools propose ways to solve different issues, and then, they run the same tests available on the RCA site. We are able to save the results in an HTML file. The Microsoft Lync Connectivity Analyzer Tool is dedicated to troubleshooting the clients for mobile devices (the Lync Windows Store app and Lync apps). It tests all the required configurations, including autodiscover and webticket services. The 32-bit version is available at http://www.microsoft.com/en-us/download/details.aspx?id=36536, while the 64-bit version can be downloaded from http://www.microsoft.com/en-us/download/details.aspx?id=36535. .NET Framework 4.5 is required. The tool itself requires a few configuration parameters; we have to insert the user information that we usually add in the Lync app, and we have to use a couple of drop-down menus to describe the scenario we are testing (on-premises or Internet, and the kind of client we are going to test). The Show drop-down menu enables us to look not only at a summary of the test results but also at the detailed information. The detailed view includes all the information and requests sent and received during the test, with the FQDN included in the answer ticket from our services, and so on, as shown in the following screenshot: The Troubleshooting Lync Online sign-in post is a support page, available in two different versions (admins and users), and is a walk-through to help admins (or users) to troubleshoot login issues. The admin version is available at http://support.microsoft.com/common/survey.aspx?scid=sw;en;3695&showpage=1, while the user version is available at http://support.microsoft.com/common/survey.aspx?scid=sw;en;3719&showpage=1. Based on our answers to the different scenario questions, the site will propose to information or solution steps. The following screenshot is part of the resolution for the log-I issues of a company that has an enterprise subscription with a custom domain: The Office 365 portal includes some information to help us monitor our Lync subscription. In the Service Health menu, navigate to Service Health; we have a list of all the incidents and service issues of the past days. In the Reports menu, we have statistics about our Office 365 consumption, including Lync. In the following screenshot, we can see the previously mentioned pages: There's more... One interesting aspect of the Microsoft Lync Connectivity Analyzer Tool that we have seen is that it enables testing for on-premises or Office 365 accounts (both testing from inside our network and from the Internet). The previously mentioned capability makes it a great tool to troubleshoot the configuration for Lync on the mobile devices that we have deployed in our internal network. This setup is usually complex, including hair-pinning and split DNS, so the diagnostic is important to quickly find misconfigured services. See also The Troubleshooting Lync Sign-in Errors (Administrators) page on Office.com at http://office.microsoft.com/en-001/communicator-help/troubleshooting-lync-sign-in-errors-administrators-HA102759022.aspx contains a list of messages related to sign-in errors with a suggested solution or a link to additional external resources. Summary In this article, we have learned about managing Lync 2013 and Lync Online and using Lync Online Remote PowerShell and Lync Online cmdlets. Resources for Article: Further resources on this subject: Adding Dialogs [article] Innovation of Communication and Information Technologies [article] Choosing Lync 2013 Clients [article]
Read more
  • 0
  • 0
  • 12847

article-image-upgrading-interface
Packt
06 Feb 2015
4 min read
Save for later

Upgrading the interface

Packt
06 Feb 2015
4 min read
In this article by Marco Schwartz and Oliver Manickum authors of the book Programming Arduino with LabVIEW, we will see how to design an interfave using LabVIEW. (For more resources related to this topic, see here.) At this stage, we know that we have our two sensors working and that they were interfaced correctly with the LabVIEW interface. However, we can do better; for now, we simply have a text display of the measurements, which is not elegant to read. Also, the light-level measurement goes from 0 to 5, which doesn't mean anything for somebody who will look at the interface for the first time. Therefore, we will modify the interface slightly. We will add a temperature gauge to display the data coming from the temperature sensor, and we will modify the output of the reading from the photocell to display the measurement from 0 (no light) to 100 percent (maximum brightness). We first need to place the different display elements. To do this, perform the following steps: Start with Front Panel. You can use a temperature gauge for the temperature and a simple slider indicator for Light Level. You will find both in the Indicators submenu of LabVIEW. After that, simply place them on the right-hand side of the interface and delete the other indicators we used earlier. Also, name the new indicators accordingly so that we can know to which element we have to connect them later. Then, it is time to go back to Block Diagram to connect the new elements we just added in Front Panel. For the temperature element, it is easy: you can simply connect the temperature gauge to the TMP36 output pin. For the light level, we will make slightly more complicated changes. We will divide the measured value beside the Analog Read element by 5, thus obtaining an output value between 0 and 1. Then, we will multiply this value by 100, to end up with a value going from 0 to 100 percent of the ambient light level. To do so perform the following steps: The first step is to place two elements corresponding to the two mathematical operations we want to do: a divide operator and a multiply operator. You can find both of them in the Functions panel of LabVIEW. Simply place them close to the Analog Read element in your program. After that, right-click on one of the inputs of each operator element, and go to Create | Constant to create a constant input for each block. Add a value of 5 for the division block, and add a value of 100 for the multiply block. Finally, connect the output of the Analog Read element to the input of the division block, the output of this block to the input of the multiply block, and the output of the multiply block to the input of the Light Level indicator. You can now go back to Front Panel to see the new interface in action. You can run the program again by clicking on the little arrow on the toolbar. You should immediately see that Temperature is now indicated by the gauge on the right and Light Level is immediately changing on the slider, depending on how you cover the sensor with your hand. Summary In this article, we connected a temperature sensor and a light-level sensor to Arduino and built a simple LabVIEW program to read data from these sensors. Then, we built a nice graphical interface to visualize the data coming from these sensors. There are many ways you can build other projects based on what you learned in this article. You can, for example, connect higher temperatures and/or more light-level sensors to the Arduino board and display these measurements in the interface. You can also connect other kinds of sensors that are supported by LabVIEW, for example, other analog sensors. For example, you can add a barometric pressure sensor or a humidity sensor to the project to build an even more complete weather-measurement station. One other interesting extension of this article will be to use the storage and plotting capabilities of LabVIEW to dynamically plot the history of the measured data inside the LabVIEW interface. Resources for Article: Further resources on this subject: The Arduino Mobile Robot [article] Using the Leap Motion Controller with Arduino [article] Avoiding Obstacles Using Sensors [article]
Read more
  • 0
  • 0
  • 7274

article-image-extending-elasticsearch-scripting
Packt
06 Feb 2015
21 min read
Save for later

Extending ElasticSearch with Scripting

Packt
06 Feb 2015
21 min read
In article by Alberto Paro, the author of ElasticSearch Cookbook Second Edition, we will cover about the following recipes: (For more resources related to this topic, see here.) Installing additional script plugins Managing scripts Sorting data using scripts Computing return fields with scripting Filtering a search via scripting Introduction ElasticSearch has a powerful way of extending its capabilities with custom scripts, which can be written in several programming languages. The most common ones are Groovy, MVEL, JavaScript, and Python. In this article, we will see how it's possible to create custom scoring algorithms, special processed return fields, custom sorting, and complex update operations on records. The scripting concept of ElasticSearch can be seen as an advanced stored procedures system in the NoSQL world; so, for an advanced usage of ElasticSearch, it is very important to master it. Installing additional script plugins ElasticSearch provides native scripting (a Java code compiled in JAR) and Groovy, but a lot of interesting languages are also available, such as JavaScript and Python. In older ElasticSearch releases, prior to version 1.4, the official scripting language was MVEL, but due to the fact that it was not well-maintained by MVEL developers, in addition to the impossibility to sandbox it and prevent security issues, MVEL was replaced with Groovy. Groovy scripting is now provided by default in ElasticSearch. The other scripting languages can be installed as plugins. Getting ready You will need a working ElasticSearch cluster. How to do it... In order to install JavaScript language support for ElasticSearch (1.3.x), perform the following steps: From the command line, simply enter the following command: bin/plugin --install elasticsearch/elasticsearch-lang-javascript/2.3.0 This will print the following result: -> Installing elasticsearch/elasticsearch-lang-javascript/2.3.0... Trying http://download.elasticsearch.org/elasticsearch/elasticsearch-lang-javascript/ elasticsearch-lang-javascript-2.3.0.zip... Downloading ....DONE Installed lang-javascript If the installation is successful, the output will end with Installed; otherwise, an error is returned. To install Python language support for ElasticSearch, just enter the following command: bin/plugin -install elasticsearch/elasticsearch-lang-python/2.3.0 The version number depends on the ElasticSearch version. Take a look at the plugin's web page to choose the correct version. How it works... Language plugins allow you to extend the number of supported languages to be used in scripting. During the ElasticSearch startup, an internal ElasticSearch service called PluginService loads all the installed language plugins. In order to install or upgrade a plugin, you need to restart the node. The ElasticSearch community provides common scripting languages (a list of the supported scripting languages is available on the ElasticSearch site plugin page at http://www.elasticsearch.org/guide/en/elasticsearch/reference/current/modules-plugins.html), and others are available in GitHub repositories (a simple search on GitHub allows you to find them). The following are the most commonly used languages for scripting: Groovy (http://groovy.codehaus.org/): This language is embedded in ElasticSearch by default. It is a simple language that provides scripting functionalities. This is one of the fastest available language extensions. Groovy is a dynamic, object-oriented programming language with features similar to those of Python, Ruby, Perl, and Smalltalk. It also provides support to write a functional code. JavaScript (https://github.com/elasticsearch/elasticsearch-lang-javascript): This is available as an external plugin. The JavaScript implementation is based on Java Rhino (https://developer.mozilla.org/en-US/docs/Rhino) and is really fast. Python (https://github.com/elasticsearch/elasticsearch-lang-python): This is available as an external plugin, based on Jython (http://jython.org). It allows Python to be used as a script engine. Considering several benchmark results, it's slower than other languages. There's more... Groovy is preferred if the script is not too complex; otherwise, a native plugin provides a better environment to implement complex logic and data management. The performance of every language is different; the fastest one is the native Java. In the case of dynamic scripting languages, Groovy is faster, as compared to JavaScript and Python. In order to access document properties in Groovy scripts, the same approach will work as in other scripting languages: doc.score: This stores the document's score. doc['field_name'].value: This extracts the value of the field_name field from the document. If the value is an array or if you want to extract the value as an array, you can use doc['field_name'].values. doc['field_name'].empty: This returns true if the field_name field has no value in the document. doc['field_name'].multivalue: This returns true if the field_name field contains multiple values. If the field contains a geopoint value, additional methods are available, as follows: doc['field_name'].lat: This returns the latitude of a geopoint. If you need the value as an array, you can use the doc['field_name'].lats method. doc['field_name'].lon: This returns the longitude of a geopoint. If you need the value as an array, you can use the doc['field_name'].lons method. doc['field_name'].distance(lat,lon): This returns the plane distance, in miles, from a latitude/longitude point. If you need to calculate the distance in kilometers, you should use the doc['field_name'].distanceInKm(lat,lon) method. doc['field_name'].arcDistance(lat,lon): This returns the arc distance, in miles, from a latitude/longitude point. If you need to calculate the distance in kilometers, you should use the doc['field_name'].arcDistanceInKm(lat,lon) method. doc['field_name'].geohashDistance(geohash): This returns the distance, in miles, from a geohash value. If you need to calculate the same distance in kilometers, you should use doc['field_name'] and the geohashDistanceInKm(lat,lon) method. By using these helper methods, it is possible to create advanced scripts in order to boost a document by a distance that can be very handy in developing geolocalized centered applications. Managing scripts Depending on your scripting usage, there are several ways to customize ElasticSearch to use your script extensions. In this recipe, we will see how to provide scripts to ElasticSearch via files, indexes, or inline. Getting ready You will need a working ElasticSearch cluster populated with the populate script (chapter_06/populate_aggregations.sh), available at https://github.com/aparo/ elasticsearch-cookbook-second-edition. How to do it... To manage scripting, perform the following steps: Dynamic scripting is disabled by default for security reasons; we need to activate it in order to use dynamic scripting languages such as JavaScript or Python. To do this, we need to turn off the disable flag (script.disable_dynamic: false) in the ElasticSearch configuration file (config/elasticseach.yml) and restart the cluster. To increase security, ElasticSearch does not allow you to specify scripts for non-sandbox languages. Scripts can be placed in the scripts directory inside the configuration directory. To provide a script in a file, we'll put a my_script.groovy script in the config/scripts location with the following code content: doc["price"].value * factor If the dynamic script is enabled (as done in the first step), ElasticSearch allows you to store the scripts in a special index, .scripts. To put my_script in the index, execute the following command in the command terminal: curl -XPOST localhost:9200/_scripts/groovy/my_script -d '{ "script":"doc["price"].value * factor" }' The script can be used by simply referencing it in the script_id field; use the following command: curl -XGET 'http://127.0.0.1:9200/test-index/test-type/_search?&pretty=true&size=3' -d '{ "query": {    "match_all": {} }, "sort": {    "_script" : {      "script_id" : "my_script",      "lang" : "groovy",      "type" : "number",      "ignore_unmapped" : true,      "params" : {        "factor" : 1.1      },      "order" : "asc"    } } }' How it works... ElasticSearch allows you to load your script in different ways; each one of these methods has their pros and cons. The most secure way to load or import scripts is to provide them as files in the config/scripts directory. This directory is continuously scanned for new files (by default, every 60 seconds). The scripting language is automatically detected by the file extension, and the script name depends on the filename. If the file is put in subdirectories, the directory path becomes part of the filename; for example, if it is config/scripts/mysub1/mysub2/my_script.groovy, the script name will be mysub1_mysub2_my_script. If the script is provided via a filesystem, it can be referenced in the code via the "script": "script_name" parameter. Scripts can also be available in the special .script index. These are the REST end points: To retrieve a script, use the following code: GET http://<server>/_scripts/<language>/<id"> To store a script use the following code: PUT http://<server>/_scripts/<language>/<id> To delete a script use the following code: DELETE http://<server>/_scripts/<language>/<id> The indexed script can be referenced in the code via the "script_id": "id_of_the_script" parameter. The recipes that follow will use inline scripting because it's easier to use it during the development and testing phases. Generally, a good practice is to develop using the inline dynamic scripting in a request, because it's faster to prototype. Once the script is ready and no changes are needed, it can be stored in the index since it is simpler to call and manage. In production, a best practice is to disable dynamic scripting and store the script on the disk (generally, dumping the indexed script to disk). See also The scripting page on the ElasticSearch website at http://www.elasticsearch.org/guide/en/elasticsearch/reference/current/modules-scripting.html Sorting data using script ElasticSearch provides scripting support for the sorting functionality. In real world applications, there is often a need to modify the default sort by the match score using an algorithm that depends on the context and some external variables. Some common scenarios are given as follows: Sorting places near a point Sorting by most-read articles Sorting items by custom user logic Sorting items by revenue Getting ready You will need a working ElasticSearch cluster and an index populated with the script, which is available at https://github.com/aparo/ elasticsearch-cookbook-second-edition. How to do it... In order to sort using scripting, perform the following steps: If you want to order your documents by the price field multiplied by a factor parameter (that is, sales tax), the search will be as shown in the following code: curl -XGET 'http://127.0.0.1:9200/test-index/test-type/_search?&pretty=true&size=3' -d '{ "query": {    "match_all": {} }, "sort": {    "_script" : {      "script" : "doc["price"].value * factor",      "lang" : "groovy",      "type" : "number",      "ignore_unmapped" : true,    "params" : {        "factor" : 1.1      },            "order" : "asc"        }    } }' In this case, we have used a match_all query and a sort script. If everything is correct, the result returned by ElasticSearch should be as shown in the following code: { "took" : 7, "timed_out" : false, "_shards" : {    "total" : 5,    "successful" : 5,    "failed" : 0 }, "hits" : {    "total" : 1000,    "max_score" : null,    "hits" : [ {      "_index" : "test-index",      "_type" : "test-type",      "_id" : "161",      "_score" : null, "_source" : … truncated …,      "sort" : [ 0.0278578661440021 ]    }, {      "_index" : "test-index",      "_type" : "test-type",      "_id" : "634",      "_score" : null, "_source" : … truncated …,     "sort" : [ 0.08131364254827411 ]    }, {      "_index" : "test-index",      "_type" : "test-type",      "_id" : "465",      "_score" : null, "_source" : … truncated …,      "sort" : [ 0.1094966959069832 ]    } ] } } How it works... The sort scripting allows you to define several parameters, as follows: order (default "asc") ("asc" or "desc"): This determines whether the order must be ascending or descending. script: This contains the code to be executed. type: This defines the type to convert the value. params (optional, a JSON object): This defines the parameters that need to be passed. lang (by default, groovy): This defines the scripting language to be used. ignore_unmapped (optional): This ignores unmapped fields in a sort. This flag allows you to avoid errors due to missing fields in shards. Extending the sort with scripting allows the use of a broader approach to score your hits. ElasticSearch scripting permits the use of every code that you want. You can create custom complex algorithms to score your documents. There's more... Groovy provides a lot of built-in functions (mainly taken from Java's Math class) that can be used in scripts, as shown in the following table: Function Description time() The current time in milliseconds sin(a) Returns the trigonometric sine of an angle cos(a) Returns the trigonometric cosine of an angle tan(a) Returns the trigonometric tangent of an angle asin(a) Returns the arc sine of a value acos(a) Returns the arc cosine of a value atan(a) Returns the arc tangent of a value toRadians(angdeg) Converts an angle measured in degrees to an approximately equivalent angle measured in radians toDegrees(angrad) Converts an angle measured in radians to an approximately equivalent angle measured in degrees exp(a) Returns Euler's number raised to the power of a value log(a) Returns the natural logarithm (base e) of a value log10(a) Returns the base 10 logarithm of a value sqrt(a) Returns the correctly rounded positive square root of a value cbrt(a) Returns the cube root of a double value IEEEremainder(f1, f2) Computes the remainder operation on two arguments, as prescribed by the IEEE 754 standard ceil(a) Returns the smallest (closest to negative infinity) value that is greater than or equal to the argument and is equal to a mathematical integer floor(a) Returns the largest (closest to positive infinity) value that is less than or equal to the argument and is equal to a mathematical integer rint(a) Returns the value that is closest in value to the argument and is equal to a mathematical integer atan2(y, x) Returns the angle theta from the conversion of rectangular coordinates (x,y_) to polar coordinates (r,_theta) pow(a, b) Returns the value of the first argument raised to the power of the second argument round(a) Returns the closest integer to the argument random() Returns a random double value abs(a) Returns the absolute value of a value max(a, b) Returns the greater of the two values min(a, b) Returns the smaller of the two values ulp(d) Returns the size of the unit in the last place of the argument signum(d) Returns the signum function of the argument sinh(x) Returns the hyperbolic sine of a value cosh(x) Returns the hyperbolic cosine of a value tanh(x) Returns the hyperbolic tangent of a value hypot(x,y) Returns sqrt(x^2+y^2) without an intermediate overflow or underflow acos(a) Returns the arc cosine of a value atan(a) Returns the arc tangent of a value If you want to retrieve records in a random order, you can use a script with a random method, as shown in the following code: curl -XGET 'http://127.0.0.1:9200/test-index/test-type/_search?&pretty=true&size=3' -d '{ "query": {    "match_all": {} }, "sort": {    "_script" : {      "script" : "Math.random()",      "lang" : "groovy",      "type" : "number",      "params" : {}    } } }' In this example, for every hit, the new sort value is computed by executing the Math.random() scripting function. See also The official ElasticSearch documentation at http://www.elasticsearch.org/guide/en/elasticsearch/reference/current/modules-scripting.html Computing return fields with scripting ElasticSearch allows you to define complex expressions that can be used to return a new calculated field value. These special fields are called script_fields, and they can be expressed with a script in every available ElasticSearch scripting language. Getting ready You will need a working ElasticSearch cluster and an index populated with the script (chapter_06/populate_aggregations.sh), which is available at https://github.com/aparo/ elasticsearch-cookbook-second-edition. How to do it... In order to compute return fields with scripting, perform the following steps: Return the following script fields: "my_calc_field": This concatenates the text of the "name" and "description" fields "my_calc_field2": This multiplies the "price" value by the "discount" parameter From the command line, execute the following code: curl -XGET 'http://127.0.0.1:9200/test-index/test-type/ _search?&pretty=true&size=3' -d '{ "query": {    "match_all": {} }, "script_fields" : {    "my_calc_field" : {      "script" : "doc["name"].value + " -- " + doc["description"].value"    },    "my_calc_field2" : {      "script" : "doc["price"].value * discount",      "params" : {       "discount" : 0.8      }    } } }' If everything works all right, this is how the result returned by ElasticSearch should be: { "took" : 4, "timed_out" : false, "_shards" : {    "total" : 5,    "successful" : 5,    "failed" : 0 }, "hits" : {    "total" : 1000,    "max_score" : 1.0,    "hits" : [ {      "_index" : "test-index",      "_type" : "test-type",      "_id" : "4",      "_score" : 1.0,      "fields" : {        "my_calc_field" : "entropic -- accusantium",        "my_calc_field2" : 5.480038242170081      }    }, {      "_index" : "test-index",      "_type" : "test-type",      "_id" : "9",      "_score" : 1.0,      "fields" : {        "my_calc_field" : "frankie -- accusantium",        "my_calc_field2" : 34.79852410178313      }    }, {      "_index" : "test-index",      "_type" : "test-type",      "_id" : "11",      "_score" : 1.0,      "fields" : {        "my_calc_field" : "johansson -- accusamus",        "my_calc_field2" : 11.824173084636591      }    } ] } } How it works... The scripting fields are similar to executing an SQL function on a field during a select operation. In ElasticSearch, after a search phase is executed and the hits to be returned are calculated, if some fields (standard or script) are defined, they are calculated and returned. The script field, which can be defined with all the supported languages, is processed by passing a value to the source of the document and, if some other parameters are defined in the script (in the discount factor example), they are passed to the script function. The script function is a code snippet; it can contain everything that the language allows you to write, but it must be evaluated to a value (or a list of values). See also The Installing additional script plugins recipe in this article to install additional languages for scripting The Sorting using script recipe to have a reference of the extra built-in functions in Groovy scripts Filtering a search via scripting ElasticSearch scripting allows you to extend the traditional filter with custom scripts. Using scripting to create a custom filter is a convenient way to write scripting rules that are not provided by Lucene or ElasticSearch, and to implement business logic that is not available in the query DSL. Getting ready You will need a working ElasticSearch cluster and an index populated with the (chapter_06/populate_aggregations.sh) script, which is available at https://github.com/aparo/ elasticsearch-cookbook-second-edition. How to do it... In order to filter a search using a script, perform the following steps: Write a search with a filter that filters out a document with the value of age less than the parameter value: curl -XGET 'http://127.0.0.1:9200/test-index/test-type/_search?&pretty=true&size=3' -d '{ "query": {    "filtered": {      "filter": {        "script": {          "script": "doc["age"].value > param1",          "params" : {            "param1" : 80          }        }      },      "query": {        "match_all": {}      }    } } }' In this example, all the documents in which the value of age is greater than param1 are qualified to be returned. If everything works correctly, the result returned by ElasticSearch should be as shown here: { "took" : 30, "timed_out" : false, "_shards" : {    "total" : 5,    "successful" : 5,    "failed" : 0 }, "hits" : {    "total" : 237,    "max_score" : 1.0,    "hits" : [ {      "_index" : "test-index",      "_type" : "test-type",      "_id" : "9",      "_score" : 1.0, "_source" :{ … "age": 83, … }    }, {      "_index" : "test-index",      "_type" : "test-type",      "_id" : "23",      "_score" : 1.0, "_source" : { … "age": 87, … }    }, {      "_index" : "test-index",      "_type" : "test-type",      "_id" : "47",      "_score" : 1.0, "_source" : {…. "age": 98, …}    } ] } } How it works... The script filter is a language script that returns a Boolean value (true/false). For every hit, the script is evaluated, and if it returns true, the hit passes the filter. This type of scripting can only be used as Lucene filters, not as queries, because it doesn't affect the search (the exceptions are constant_score and custom_filters_score). These are the scripting fields: script: This contains the code to be executed params: These are optional parameters to be passed to the script lang (defaults to groovy): This defines the language of the script The script code can be any code in your preferred and supported scripting language that returns a Boolean value. There's more... Other languages are used in the same way as Groovy. For the current example, I have chosen a standard comparison that works in several languages. To execute the same script using the JavaScript language, use the following code: curl -XGET 'http://127.0.0.1:9200/test-index/test-type/_search?&pretty=true&size=3' -d '{ "query": {    "filtered": {      "filter": {        "script": {          "script": "doc["age"].value > param1",          "lang":"javascript",          "params" : {            "param1" : 80          }        }      },      "query": {        "match_all": {}      }    } } }' For Python, use the following code: curl -XGET 'http://127.0.0.1:9200/test-index/test-type/_search?&pretty=true&size=3' -d '{ "query": {    "filtered": {      "filter": {        "script": {          "script": "doc["age"].value > param1",          "lang":"python",          "params" : {            "param1" : 80          }        }      },      "query": {        "match_all": {}      }    } } }' See also The Installing additional script plugins recipe in this article to install additional languages for scripting The Sorting data using script recipe in this article to get a reference of the extra built-in functions in Groovy scripts Summary In this article you have learnt the ways you can use scripting to extend the ElasticSearch functional capabilities using different programming languages. Resources for Article: Further resources on this subject: Indexing the Data [Article] Low-Level Index Control [Article] Designing Puppet Architectures [Article]
Read more
  • 0
  • 0
  • 8475

article-image-contexts-and-dependency-injection-netbeans
Packt
06 Feb 2015
18 min read
Save for later

Contexts and Dependency Injection in NetBeans

Packt
06 Feb 2015
18 min read
In this article by David R. Heffelfinger, the author of Java EE 7 Development with NetBeans 8, we will introduce Contexts and Dependency Injection (CDI) and other aspects of it. CDI can be used to simplify integrating the different layers of a Java EE application. For example, CDI allows us to use a session bean as a managed bean, so that we can take advantage of the EJB features, such as transactions, directly in our managed beans. In this article, we will cover the following topics: Introduction to CDI Qualifiers Stereotypes Interceptor binding types Custom scopes (For more resources related to this topic, see here.) Introduction to CDI JavaServer Faces (JSF) web applications employing CDI are very similar to JSF applications without CDI; the main difference is that instead of using JSF managed beans for our model and controllers, we use CDI named beans. What makes CDI applications easier to develop and maintain are the excellent dependency injection capabilities of the CDI API. Just as with other JSF applications, CDI applications use facelets as their view technology. The following example illustrates a typical markup for a JSF page using CDI: <?xml version='1.0' encoding='UTF-8' ?> <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"    "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"> <html      >    <h:head>        <title>Create New Customer</title>    </h:head>    <h:body>        <h:form>            <h3>Create New Customer</h3>            <h:panelGrid columns="3">                <h:outputLabel for="firstName" value="First Name"/>                <h:inputText id="firstName" value="#{customer.firstName}"/>                <h:message for="firstName"/>                  <h:outputLabel for="middleName" value="Middle Name"/>                <h:inputText id="middleName"                  value="#{customer.middleName}"/>                <h:message for="middleName"/>                  <h:outputLabel for="lastName" value="Last Name"/>                <h:inputText id="lastName" value="#{customer.lastName}"/>                <h:message for="lastName"/>                  <h:outputLabel for="email" value="Email Address"/>                <h:inputText id="email" value="#{customer.email}"/>                <h:message for="email"/>                <h:panelGroup/>                <h:commandButton value="Submit"                  action="#{customerController.navigateToConfirmation}"/>            </h:panelGrid>        </h:form>    </h:body> </html> As we can see, the preceding markup doesn't look any different from the markup used for a JSF application that does not use CDI. The page renders as follows (shown after entering some data): In our page markup, we have JSF components that use Unified Expression Language expressions to bind themselves to CDI named bean properties and methods. Let's take a look at the customer bean first: package com.ensode.cdiintro.model;   import java.io.Serializable; import javax.enterprise.context.RequestScoped; import javax.inject.Named;   @Named @RequestScoped public class Customer implements Serializable {      private String firstName;    private String middleName;    private String lastName;    private String email;      public Customer() {    }      public String getFirstName() {        return firstName;    }      public void setFirstName(String firstName) {        this.firstName = firstName;    }      public String getMiddleName() {        return middleName;    }      public void setMiddleName(String middleName) {        this.middleName = middleName;    }      public String getLastName() {        return lastName;    }      public void setLastName(String lastName) {        this.lastName = lastName;    }      public String getEmail() {        return email;    }      public void setEmail(String email) {        this.email = email;    } } The @Named annotation marks this class as a CDI named bean. By default, the bean's name will be the class name with its first character switched to lowercase (in our example, the name of the bean is "customer", since the class name is Customer). We can override this behavior if we wish by passing the desired name to the value attribute of the @Named annotation, as follows: @Named(value="customerBean") A CDI named bean's methods and properties are accessible via facelets, just like regular JSF managed beans. Just like JSF managed beans, CDI named beans can have one of several scopes as listed in the following table. The preceding named bean has a scope of request, as denoted by the @RequestScoped annotation. Scope Annotation Description Request @RequestScoped Request scoped beans are shared through the duration of a single request. A single request could refer to an HTTP request, an invocation to a method in an EJB, a web service invocation, or sending a JMS message to a message-driven bean. Session @SessionScoped Session scoped beans are shared across all requests in an HTTP session. Each user of an application gets their own instance of a session scoped bean. Application @ApplicationScoped Application scoped beans live through the whole application lifetime. Beans in this scope are shared across user sessions. Conversation @ConversationScoped The conversation scope can span multiple requests, and is typically shorter than the session scope. Dependent @Dependent Dependent scoped beans are not shared. Any time a dependent scoped bean is injected, a new instance is created. As we can see, CDI has equivalent scopes to all JSF scopes. Additionally, CDI adds two additional scopes. The first CDI-specific scope is the conversation scope, which allows us to have a scope that spans across multiple requests, but is shorter than the session scope. The second CDI-specific scope is the dependent scope, which is a pseudo scope. A CDI bean in the dependent scope is a dependent object of another object; beans in this scope are instantiated when the object they belong to is instantiated and they are destroyed when the object they belong to is destroyed. Our application has two CDI named beans. We already discussed the customer bean. The other CDI named bean in our application is the controller bean: package com.ensode.cdiintro.controller;   import com.ensode.cdiintro.model.Customer; import javax.enterprise.context.RequestScoped; import javax.inject.Inject; import javax.inject.Named;   @Named @RequestScoped public class CustomerController {      @Inject    private Customer customer;      public Customer getCustomer() {        return customer;    }      public void setCustomer(Customer customer) {        this.customer = customer;    }      public String navigateToConfirmation() {        //In a real application we would        //Save customer data to the database here.          return "confirmation";    } } In the preceding class, an instance of the Customer class is injected at runtime; this is accomplished via the @Inject annotation. This annotation allows us to easily use dependency injection in CDI applications. Since the Customer class is annotated with the @RequestScoped annotation, a new instance of Customer will be injected for every request. The navigateToConfirmation() method in the preceding class is invoked when the user clicks on the Submit button on the page. The navigateToConfirmation() method works just like an equivalent method in a JSF managed bean would, that is, it returns a string and the application navigates to an appropriate page based on the value of that string. Like with JSF, by default, the target page's name with an .xhtml extension is the return value of this method. For example, if no exceptions are thrown in the navigateToConfirmation() method, the user is directed to a page named confirmation.xhtml: <?xml version='1.0' encoding='UTF-8' ?> <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"> <html      >    <h:head>        <title>Success</title>    </h:head>    <h:body>        New Customer created successfully.        <h:panelGrid columns="2" border="1" cellspacing="0">            <h:outputLabel for="firstName" value="First Name"/>            <h:outputText id="firstName" value="#{customer.firstName}"/>              <h:outputLabel for="middleName" value="Middle Name"/>            <h:outputText id="middleName"              value="#{customer.middleName}"/>              <h:outputLabel for="lastName" value="Last Name"/>            <h:outputText id="lastName" value="#{customer.lastName}"/>              <h:outputLabel for="email" value="Email Address"/>            <h:outputText id="email" value="#{customer.email}"/>          </h:panelGrid>    </h:body> </html> Again, there is nothing special we need to do to access the named beans properties from the preceding markup. It works just as if the bean was a JSF managed bean. The preceding page renders as follows: As we can see, CDI applications work just like JSF applications. However, CDI applications have several advantages over JSF, for example (as we mentioned previously) CDI beans have additional scopes not found in JSF. Additionally, using CDI allows us to decouple our Java code from the JSF API. Also, as we mentioned previously, CDI allows us to use session beans as named beans. Qualifiers In some instances, the type of bean we wish to inject into our code may be an interface or a Java superclass, but we may be interested in injecting a subclass or a class implementing the interface. For cases like this, CDI provides qualifiers we can use to indicate the specific type we wish to inject into our code. A CDI qualifier is an annotation that must be decorated with the @Qualifier annotation. This annotation can then be used to decorate the specific subclass or interface. In this section, we will develop a Premium qualifier for our customer bean; premium customers could get perks that are not available to regular customers, for example, discounts. Creating a CDI qualifier with NetBeans is very easy; all we need to do is go to File | New File, select the Contexts and Dependency Injection category, and select the Qualifier Type file type. In the next step in the wizard, we need to enter a name and a package for our qualifier. After these two simple steps, NetBeans generates the code for our qualifier: package com.ensode.cdiintro.qualifier;   import static java.lang.annotation.ElementType.TYPE; import static java.lang.annotation.ElementType.FIELD; import static java.lang.annotation.ElementType.PARAMETER; import static java.lang.annotation.ElementType.METHOD; import static java.lang.annotation.RetentionPolicy.RUNTIME; import java.lang.annotation.Retention; import java.lang.annotation.Target; import javax.inject.Qualifier;   @Qualifier @Retention(RUNTIME) @Target({METHOD, FIELD, PARAMETER, TYPE}) public @interface Premium { } Qualifiers are standard Java annotations. Typically, they have retention of runtime and can target methods, fields, parameters, or types. The only difference between a qualifier and a standard annotation is that qualifiers are decorated with the @Qualifier annotation. Once we have our qualifier in place, we need to use it to decorate the specific subclass or interface implementation, as shown in the following code: package com.ensode.cdiintro.model;   import com.ensode.cdiintro.qualifier.Premium; import javax.enterprise.context.RequestScoped; import javax.inject.Named;   @Named @RequestScoped @Premium public class PremiumCustomer extends Customer {      private Integer discountCode;      public Integer getDiscountCode() {        return discountCode;    }      public void setDiscountCode(Integer discountCode) {        this.discountCode = discountCode;    } } Once we have decorated the specific instance we need to qualify, we can use our qualifiers in the client code to specify the exact type of dependency we need: package com.ensode.cdiintro.controller;   import com.ensode.cdiintro.model.Customer; import com.ensode.cdiintro.model.PremiumCustomer; import com.ensode.cdiintro.qualifier.Premium;   import java.util.logging.Level; import java.util.logging.Logger; import javax.enterprise.context.RequestScoped; import javax.inject.Inject; import javax.inject.Named;   @Named @RequestScoped public class PremiumCustomerController {      private static final Logger logger = Logger.getLogger(            PremiumCustomerController.class.getName());    @Inject    @Premium    private Customer customer;      public String saveCustomer() {          PremiumCustomer premiumCustomer =          (PremiumCustomer) customer;          logger.log(Level.INFO, "Saving the following information n"                + "{0} {1}, discount code = {2}",                new Object[]{premiumCustomer.getFirstName(),                    premiumCustomer.getLastName(),                    premiumCustomer.getDiscountCode()});          //If this was a real application, we would have code to save        //customer data to the database here.          return "premium_customer_confirmation";    } } Since we used our @Premium qualifier to decorate the customer field, an instance of the PremiumCustomer class is injected into that field. This is because this class is also decorated with the @Premium qualifier. As far as our JSF pages go, we simply access our named bean as usual using its name, as shown in the following code; <?xml version='1.0' encoding='UTF-8' ?> <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"> <html      >    <h:head>        <title>Create New Premium Customer</title>    </h:head>    <h:body>        <h:form>            <h3>Create New Premium Customer</h3>            <h:panelGrid columns="3">                <h:outputLabel for="firstName" value="First Name"/>                 <h:inputText id="firstName"                    value="#{premiumCustomer.firstName}"/>                <h:message for="firstName"/>                  <h:outputLabel for="middleName" value="Middle Name"/>                <h:inputText id="middleName"                     value="#{premiumCustomer.middleName}"/>                <h:message for="middleName"/>                  <h:outputLabel for="lastName" value="Last Name"/>                <h:inputText id="lastName"                    value="#{premiumCustomer.lastName}"/>                <h:message for="lastName"/>                  <h:outputLabel for="email" value="Email Address"/>                <h:inputText id="email"                    value="#{premiumCustomer.email}"/>                <h:message for="email"/>                  <h:outputLabel for="discountCode" value="Discount Code"/>                <h:inputText id="discountCode"                    value="#{premiumCustomer.discountCode}"/>                <h:message for="discountCode"/>                   <h:panelGroup/>                <h:commandButton value="Submit"                      action="#{premiumCustomerController.saveCustomer}"/>            </h:panelGrid>        </h:form>    </h:body> </html> In this example, we are using the default name for our bean, which is the class name with the first letter switched to lowercase. Now, we are ready to test our application: After submitting the page, we can see the confirmation page. Stereotypes A CDI stereotype allows us to create new annotations that bundle up several CDI annotations. For example, if we need to create several CDI named beans with a scope of session, we would have to use two annotations in each of these beans, namely @Named and @SessionScoped. Instead of having to add two annotations to each of our beans, we could create a stereotype and annotate our beans with it. To create a CDI stereotype in NetBeans, we simply need to create a new file by selecting the Contexts and Dependency Injection category and the Stereotype file type. Then, we need to enter a name and package for our new stereotype. At this point, NetBeans generates the following code: package com.ensode.cdiintro.stereotype;   import static java.lang.annotation.ElementType.TYPE; import static java.lang.annotation.ElementType.FIELD; import static java.lang.annotation.ElementType.METHOD; import static java.lang.annotation.RetentionPolicy.RUNTIME; import java.lang.annotation.Retention; import java.lang.annotation.Target; import javax.enterprise.inject.Stereotype;   @Stereotype @Retention(RUNTIME) @Target({METHOD, FIELD, TYPE}) public @interface NamedSessionScoped { } Now, we simply need to add the CDI annotations that we want the classes annotated with our stereotype to use. In our case, we want them to be named beans and have a scope of session; therefore, we add the @Named and @SessionScoped annotations as shown in the following code: package com.ensode.cdiintro.stereotype;   import static java.lang.annotation.ElementType.TYPE; import static java.lang.annotation.ElementType.FIELD; import static java.lang.annotation.ElementType.METHOD; import static java.lang.annotation.RetentionPolicy.RUNTIME; import java.lang.annotation.Retention; import java.lang.annotation.Target; import javax.enterprise.context.SessionScoped; import javax.enterprise.inject.Stereotype; import javax.inject.Named;   @Named @SessionScoped @Stereotype @Retention(RUNTIME) @Target({METHOD, FIELD, TYPE}) public @interface NamedSessionScoped { } Now we can use our stereotype in our own code: package com.ensode.cdiintro.beans;   import com.ensode.cdiintro.stereotype.NamedSessionScoped; import java.io.Serializable;   @NamedSessionScoped public class StereotypeClient implements Serializable {      private String property1;    private String property2;      public String getProperty1() {        return property1;    }      public void setProperty1(String property1) {        this.property1 = property1;    }      public String getProperty2() {        return property2;    }      public void setProperty2(String property2) {        this.property2 = property2;    } } We annotated the StereotypeClient class with our NamedSessionScoped stereotype, which is equivalent to using the @Named and @SessionScoped annotations. Interceptor binding types One of the advantages of EJBs is that they allow us to easily perform aspect-oriented programming (AOP) via interceptors. CDI allows us to write interceptor binding types; this lets us bind interceptors to beans and the beans do not have to depend on the interceptor directly. Interceptor binding types are annotations that are themselves annotated with @InterceptorBinding. Creating an interceptor binding type in NetBeans involves creating a new file, selecting the Contexts and Dependency Injection category, and selecting the Interceptor Binding Type file type. Then, we need to enter a class name and select or enter a package for our new interceptor binding type. At this point, NetBeans generates the code for our interceptor binding type: package com.ensode.cdiintro.interceptorbinding;   import static java.lang.annotation.ElementType.TYPE; import static java.lang.annotation.ElementType.METHOD; import static java.lang.annotation.RetentionPolicy.RUNTIME; import java.lang.annotation.Inherited; import java.lang.annotation.Retention; import java.lang.annotation.Target; import javax.interceptor.InterceptorBinding;   @Inherited @InterceptorBinding @Retention(RUNTIME) @Target({METHOD, TYPE}) public @interface LoggingInterceptorBinding { } The generated code is fully functional; we don't need to add anything to it. In order to use our interceptor binding type, we need to write an interceptor and annotate it with our interceptor binding type, as shown in the following code: package com.ensode.cdiintro.interceptor;   import com.ensode.cdiintro.interceptorbinding.LoggingInterceptorBinding; import java.io.Serializable; import java.util.logging.Level; import java.util.logging.Logger; import javax.interceptor.AroundInvoke; import javax.interceptor.Interceptor; import javax.interceptor.InvocationContext;   @LoggingInterceptorBinding @Interceptor public class LoggingInterceptor implements Serializable{      private static final Logger logger = Logger.getLogger(            LoggingInterceptor.class.getName());      @AroundInvoke    public Object logMethodCall(InvocationContext invocationContext)            throws Exception {          logger.log(Level.INFO, new StringBuilder("entering ").append(                invocationContext.getMethod().getName()).append(                " method").toString());          Object retVal = invocationContext.proceed();          logger.log(Level.INFO, new StringBuilder("leaving ").append(                invocationContext.getMethod().getName()).append(                " method").toString());          return retVal;    } } As we can see, other than being annotated with our interceptor binding type, the preceding class is a standard interceptor similar to the ones we use with EJB session beans. In order for our interceptor binding type to work properly, we need to add a CDI configuration file (beans.xml) to our project. Then, we need to register our interceptor in beans.xml as follows: <?xml version="1.0" encoding="UTF-8"?> <beans               xsi_schemaLocation="http://>    <interceptors>          <class>        com.ensode.cdiintro.interceptor.LoggingInterceptor      </class>    </interceptors> </beans> To register our interceptor, we need to set bean-discovery-mode to all in the generated beans.xml and add the <interceptor> tag in beans.xml, with one or more nested <class> tags containing the fully qualified names of our interceptors. The final step before we can use our interceptor binding type is to annotate the class to be intercepted with our interceptor binding type: package com.ensode.cdiintro.controller;   import com.ensode.cdiintro.interceptorbinding.LoggingInterceptorBinding; import com.ensode.cdiintro.model.Customer; import com.ensode.cdiintro.model.PremiumCustomer; import com.ensode.cdiintro.qualifier.Premium; import java.util.logging.Level; import java.util.logging.Logger; import javax.enterprise.context.RequestScoped; import javax.inject.Inject; import javax.inject.Named;   @LoggingInterceptorBinding @Named @RequestScoped public class PremiumCustomerController {      private static final Logger logger = Logger.getLogger(            PremiumCustomerController.class.getName());    @Inject    @Premium    private Customer customer;      public String saveCustomer() {          PremiumCustomer premiumCustomer = (PremiumCustomer) customer;          logger.log(Level.INFO, "Saving the following information n"                + "{0} {1}, discount code = {2}",                new Object[]{premiumCustomer.getFirstName(),                    premiumCustomer.getLastName(),                    premiumCustomer.getDiscountCode()});          //If this was a real application, we would have code to save        //customer data to the database here.          return "premium_customer_confirmation";    } } Now, we are ready to use our interceptor. After executing the preceding code and examining the GlassFish log, we can see our interceptor binding type in action. The lines entering saveCustomer method and leaving saveCustomer method were added to the log by our interceptor, which was indirectly invoked by our interceptor binding type. Custom scopes In addition to providing several prebuilt scopes, CDI allows us to define our own custom scopes. This functionality is primarily meant for developers building frameworks on top of CDI, not for application developers. Nevertheless, NetBeans provides a wizard for us to create our own CDI custom scopes. To create a new CDI custom scope, we need to go to File | New File, select the Contexts and Dependency Injection category, and select the Scope Type file type. Then, we need to enter a package and a name for our custom scope. After clicking on Finish, our new custom scope is created, as shown in the following code: package com.ensode.cdiintro.scopes;   import static java.lang.annotation.ElementType.TYPE; import static java.lang.annotation.ElementType.FIELD; import static java.lang.annotation.ElementType.METHOD; import static java.lang.annotation.RetentionPolicy.RUNTIME; import java.lang.annotation.Inherited; import java.lang.annotation.Retention; import java.lang.annotation.Target; import javax.inject.Scope;   @Inherited @Scope // or @javax.enterprise.context.NormalScope @Retention(RUNTIME) @Target({METHOD, FIELD, TYPE}) public @interface CustomScope { } To actually use our scope in our CDI applications, we would need to create a custom context which, as mentioned previously, is primarily a concern for framework developers and not for Java EE application developers. Therefore, it is beyond the scope of this article. Interested readers can refer to JBoss Weld CDI for Java Platform, Ken Finnigan, Packt Publishing. (JBoss Weld is a popular CDI implementation and it is included with GlassFish.) Summary In this article, we covered NetBeans support for CDI, a new Java EE API introduced in Java EE 6. We provided an introduction to CDI and explained additional functionality that the CDI API provides over standard JSF. We also covered how to disambiguate CDI injected beans via CDI Qualifiers. Additionally, we covered how to group together CDI annotations via CDI stereotypes. We also, we saw how CDI can help us with AOP via interceptor binding types. Finally, we covered how NetBeans can help us create custom CDI scopes. Resources for Article: Further resources on this subject: Java EE 7 Performance Tuning and Optimization [article] Java EE 7 Developer Handbook [article] Java EE 7 with GlassFish 4 Application Server [article]
Read more
  • 0
  • 0
  • 9639
article-image-postgresql-cookbook-high-availability-and-replication
Packt
06 Feb 2015
26 min read
Save for later

PostgreSQL Cookbook - High Availability and Replication

Packt
06 Feb 2015
26 min read
In this article by Chitij Chauhan, author of the book PostgreSQL Cookbook, we will talk about various high availability and replication solutions, including some popular third-party replication tools such as Slony-I and Londiste. In this article, we will cover the following recipes: Setting up hot streaming replication Replication using Slony-I Replication using Londiste The important components for any production database is to achieve fault tolerance, 24/7 availability, and redundancy. It is for this purpose that we have different high availability and replication solutions available for PostgreSQL. From a business perspective, it is important to ensure 24/7 data availability in the event of a disaster situation or a database crash due to disk or hardware failure. In such situations, it becomes critical to ensure that a duplicate copy of the data is available on a different server or a different database, so that seamless failover can be achieved even when the primary server/database is unavailable. Setting up hot streaming replication In this recipe, we are going to set up a master-slave streaming replication. Getting ready For this exercise, you will need two Linux machines, each with the latest version of PostgreSQL installed. We will be using the following IP addresses for the master and slave servers: Master IP address: 192.168.0.4 Slave IP address: 192.168.0.5 Before you start with the master-slave streaming setup, it is important that the SSH connectivity between the master and slave is setup. How to do it... Perform the following sequence of steps to set up a master-slave streaming replication: First, we are going to create a user on the master, which will be used by the slave server to connect to the PostgreSQL database on the master server: psql -c "CREATE USER repuser REPLICATION LOGIN ENCRYPTED PASSWORD 'charlie';" Next, we will allow the replication user that was created in the previous step to allow access to the master PostgreSQL server. This is done by making the necessary changes as mentioned in the pg_hba.conf file: Vi pg_hba.conf host   replication   repuser   192.168.0.5/32   md5 In the next step, we are going to configure parameters in the postgresql.conf file. These parameters need to be set in order to get the streaming replication working: Vi /var/lib/pgsql/9.3/data/postgresql.conf listen_addresses = '*' wal_level = hot_standby max_wal_senders = 3 wal_keep_segments = 8 archive_mode = on       archive_command = 'cp %p /var/lib/pgsql/archive/%f && scp %p postgres@192.168.0.5:/var/lib/pgsql/archive/%f' checkpoint_segments = 8 Once the parameter changes have been made in the postgresql.conf file in the previous step, the next step will be to restart the PostgreSQL server on the master server, in order to let the changes take effect: pg_ctl -D /var/lib/pgsql/9.3/data restart Before the slave can replicate the master, we will need to give it the initial database to build off. For this purpose, we will make a base backup by copying the primary server's data directory to the standby. The rsync command needs to be run as a root user: psql -U postgres -h 192.168.0.4 -c "SELECT pg_start_backup('label', true)" rsync -a /var/lib/pgsql/9.3/data/ 192.168.0.5:/var/lib/pgsql/9.3/data/ --exclude postmaster.pid psql -U postgres -h 192.168.0.4 -c "SELECT pg_stop_backup()" Once the data directory, mentioned in the previous step, is populated, the next step is to enable the following parameter in the postgresql.conf file on the slave server: hot_standby = on The next step will be to copy the recovery.conf.sample file in the $PGDATA location on the slave server and then configure the following parameters: cp /usr/pgsql-9.3/share/recovery.conf.sample /var/lib/pgsql/9.3/data/recovery.conf standby_mode = on primary_conninfo = 'host=192.168.0.4 port=5432 user=repuser password=charlie' trigger_file = '/tmp/trigger.replication′ restore_command = 'cp /var/lib/pgsql/archive/%f "%p"' The next step will be to start the slave server: service postgresql-9.3 start Now that the above mentioned replication steps are set up, we will test for replication. On the master server, log in and issue the following SQL commands: psql -h 192.168.0.4 -d postgres -U postgres -W postgres=# create database test;   postgres=# c test;   test=# create table testtable ( testint int, testchar varchar(40) );   CREATE TABLE test=# insert into testtable values ( 1, 'What A Sight.' ); INSERT 0 1 On the slave server, we will now check whether the newly created database and the corresponding table, created in the previous step, are replicated: psql -h 192.168.0.5 -d test -U postgres -W test=# select * from testtable; testint | testchar ---------+--------------------------- 1 | What A Sight. (1 row) How it works... The following is the explanation for the steps performed in the preceding section. In the initial step of the preceding section, we create a user called repuser, which will be used by the slave server to make a connection to the primary server. In the second step of the preceding section, we make the necessary changes in the pg_hba.conf file to allow the master server to be accessed by the slave server using the repuser user ID that was created in step 1. We then make the necessary parameter changes on the master in step 3 of the preceding section to configure a streaming replication. The following is a description of these parameters: listen_addresses: This parameter is used to provide the IP address associated with the interface that you want to have PostgreSQL listen to. A value of * indicates all available IP addresses. wal_level: This parameter determines the level of WAL logging done. Specify hot_standby for streaming replication. wal_keep_segments: This parameter specifies the number of 16 MB WAL files to be retained in the pg_xlog directory. The rule of thumb is that more such files might be required to handle a large checkpoint. archive_mode: Setting this parameter enables completed WAL segments to be sent to the archive storage. archive_command: This parameter is basically a shell command that is executed whenever a WAL segment is completed. In our case, we are basically copying the file to the local machine and then using the secure copy command to send it across to the slave. max_wal_senders: This parameter specifies the total number of concurrent connections allowed from the slave servers. checkpoint_segments: This parameter specifies the maximum number of logfile segments between automatic WAL checkpoints. Once the necessary configuration changes have been made on the master server, we then restart the PostgreSQL server on the master in order to let the new configuration changes take effect. This is done in step 4 of the preceding section. In step 5 of the preceding section, we are basically building the slave by copying the primary server's data directory to the slave. Now, with the data directory available on the slave, the next step is to configure it. We will now make the necessary parameter replication related parameter changes on the slave in the postgresql.conf directory on the slave server. We set the following parameters on the slave: hot_standby: This parameter determines whether you can connect and run queries when the server is in the archive recovery or standby mode. In the next step, we are configuring the recovery.conf file. This is required to be set up so that the slave can start receiving logs from the master. The parameters explained next are configured in the recovery.conf file on the slave. standby_mode: This parameter, when enabled, causes PostgreSQL to work as a standby in a replication configuration. primary_conninfo: This parameter specifies the connection information used by the slave to connect to the master. For our scenario, our master server is set as 192.168.0.4 on port 5432 and we are using the repuser userid with the password charlie to make a connection to the master. Remember that repuser was the userid which was created in the initial step of the preceding section for this purpose, that is, connecting to the master from the slave. trigger_file: When a slave is configured as a standby, it will continue to restore the XLOG records from the master. The trigger_file parameter specifies what is used to trigger a slave, in order to switch over its duties from standby and take over as master or primary server. At this stage, the slave is fully configured now and we can start the slave server; then, the replication process begins. This is shown in step 8 of the preceding section. In steps 9 and 10 of the preceding section, we are simply testing our replication. We first begin by creating a test database, then we log in to the test database and create a table by the name testtable, and then we begin inserting some records into the testtable table. Now, our purpose is to see whether these changes are replicated across the slave. To test this, we log in to the slave on the test database and then query the records from the testtable table, as seen in step 10 of the preceding section. The final result that we see is that all the records that are changed/inserted on the primary server are visible on the slave. This completes our streaming replication's setup and configuration. Replication using Slony-I Here, we are going to set up replication using Slony-I. We will be setting up the replication of table data between two databases on the same server. Getting ready The steps performed in this recipe are carried out on a CentOS Version 6 machine. It is also important to remove the directives related to hot streaming replication prior to setting up replication using Slony-I. We will first need to install Slony-I. The following steps need to be performed in order to install Slony-I: First, go to http://slony.info/downloads/2.2/source/ and download the given software. Once you have downloaded the Slony-I software, the next step is to unzip the .tar file and then go the newly created directory. Before doing this, please ensure that you have the postgresql-devel package for the corresponding PostgreSQL version installed before you install Slony-I: tar xvfj slony1-2.2.3.tar.bz2  cd slony1-2.2.3 In the next step, we are going to configure, compile, and build the software: ./configure --with-pgconfigdir=/usr/pgsql-9.3/bin/ make make install How to do it... You need to perform the following sequence of steps, in order to replicate data between two tables using Slony-I replication: First, start the PostgreSQL server if you have not already started it: pg_ctl -D $PGDATA start In the next step, we will be creating two databases, test1 and test2, which will be used as the source and target databases respectively: createdb test1 createdb test2 In the next step, we will create the t_test table on the source database, test1, and insert some records into it: psql -d test1 test1=# create table t_test (id numeric primary key, name varchar);   test1=# insert into t_test values(1,'A'),(2,'B'), (3,'C'); We will now set up the target database by copying the table definitions from the test1 source database: pg_dump -s -p 5432 -h localhost test1 | psql -h localhost -p 5432 test2 We will now connect to the target database, test2, and verify that there is no data in the tables of the test2 database: psql -d test2 test2=# select * from t_test; We will now set up a slonik script for the master-slave, that is source/target, setup. In this scenario, since we are replicating between two different databases on the same server, the only different connection string option will be the database name: cd /usr/pgsql-9.3/bin vi init_master.slonik   #!/bin/sh cluster name = mycluster; node 1 admin conninfo = 'dbname=test1 host=localhost port=5432 user=postgres password=postgres'; node 2 admin conninfo = 'dbname=test2 host=localhost port=5432 user=postgres password=postgres'; init cluster ( id=1); create set (id=1, origin=1); set add table(set id=1, origin=1, id=1, fully qualified name = 'public.t_test'); store node (id=2, event node = 1); store path (server=1, client=2, conninfo='dbname=test1 host=localhost port=5432 user=postgres password=postgres'); store path (server=2, client=1, conninfo='dbname=test2 host=localhost port=5432 user=postgres password=postgres'); store listen (origin=1, provider = 1, receiver = 2);  store listen (origin=2, provider = 2, receiver = 1); We will now create a slonik script for subscription to the slave, that is, target: cd /usr/pgsql-9.3/bin vi init_slave.slonik #!/bin/sh cluster name = mycluster; node 1 admin conninfo = 'dbname=test1 host=localhost port=5432 user=postgres password=postgres'; node 2 admin conninfo = 'dbname=test2 host=localhost port=5432 user=postgres password=postgres'; subscribe set ( id = 1, provider = 1, receiver = 2, forward = no); We will now run the init_master.slonik script created in step 6 and run this on the master, as follows: cd /usr/pgsql-9.3/bin   slonik init_master.slonik We will now run the init_slave.slonik script created in step 7 and run this on the slave, that is, target: cd /usr/pgsql-9.3/bin   slonik init_slave.slonik In the next step, we will start the master slon daemon: nohup slon mycluster "dbname=test1 host=localhost port=5432 user=postgres password=postgres" & In the next step, we will start the slave slon daemon: nohup slon mycluster "dbname=test2 host=localhost port=5432 user=postgres password=postgres" & Next, we will connect to the master, that is, the test1 source database, and insert some records in the t_test table: psql -d test1 test1=# insert into t_test values (5,'E'); We will now test for the replication by logging on to the slave, that is, the test2 target database, and see whether the inserted records in the t_test table are visible: psql -d test2   test2=# select * from t_test; id | name ----+------ 1 | A 2 | B 3 | C 5 | E (4 rows) How it works... We will now discuss the steps performed in the preceding section: In step 1, we first start the PostgreSQL server if it is not already started. In step 2, we create two databases, namely test1 and test2, that will serve as our source (master) and target (slave) databases. In step 3, we log in to the test1 source database, create a t_test table, and insert some records into the table. In step 4, we set up the target database, test2, by copying the table definitions present in the source database and loading them into test2 using the pg_dump utility. In step 5, we log in to the target database, test2, and verify that there are no records present in the t_test table because in step 4, we only extracted the table definitions into the test2 database from the test1 database. In step 6, we set up a slonik script for the master-slave replication setup. In the init_master.slonik file, we first define the cluster name as mycluster. We then define the nodes in the cluster. Each node will have a number associated with a connection string, which contains database connection information. The node entry is defined both for the source and target databases. The store_path commands are necessary, so that each node knows how to communicate with the other. In step 7, we set up a slonik script for the subscription of the slave, that is, the test2 target database. Once again, the script contains information such as the cluster name and the node entries that are designated a unique number related to connection string information. It also contains a subscriber set. In step 8, we run the init_master.slonik file on the master. Similarly, in step 9, we run the init_slave.slonik file on the slave. In step 10, we start the master slon daemon. In step 11, we start the slave slon daemon. The subsequent steps, 12 and 13, are used to test for replication. For this purpose, in step 12 of the preceding section, we first log in to the test1 source database and insert some records into the t_test table. To check whether the newly inserted records have been replicated in the target database, test2, we log in to the test2 database in step 13. The result set obtained from the output of the query confirms that the changed/inserted records on the t_test table in the test1 database are successfully replicated across the target database, test2. For more information on Slony-I replication, go to http://slony.info/documentation/tutorial.html. There's more... If you are using Slony-I for replication between two different servers, in addition to the steps mentioned in the How to do it… section, you will also have to enable authentication information in the pg_hba.conf file existing on both the source and target servers. For example, let's assume that the source server's IP is 192.168.16.44 and the target server's IP is 192.168.16.56 and we are using a user named super to replicate the data. If this is the situation, then in the source server's pg_hba.conf file, we will have to enter the information, as follows: host         postgres         super     192.168.16.44/32           md5 Similarly, in the target server's pg_hba.conf file, we will have to enter the authentication information, as follows: host         postgres         super     192.168.16.56/32           md5 Also, in the shell scripts that were used for Slony-I, wherever the connection information for the host is localhost that entry will need to be replaced by the source and target server's IP addresses. Replication using Londiste In this recipe, we are going to show you how to replicate data using Londiste. Getting ready For this setup, we are using the same host CentOS Linux machine to replicate data between two databases. This can also be set up using two separate Linux machines running on VMware, VirtualBox, or any other virtualization software. It is assumed that the latest version of PostgreSQL, version 9.3, is installed. We used CentOS Version 6 as the Linux operating system for this exercise. To set up Londiste replication on the Linux machine, perform the following steps: Go to http://pgfoundry.org/projects/skytools/ and download the latest version of Skytools 3.2, that is, tarball skytools-3.2.tar.gz. Extract the tarball file, as follows: tar -xvzf skytools-3.2.tar.gz Go to the new location and build and compile the software: cd skytools-3.2 ./configure --prefix=/var/lib/pgsql/9.3/Sky –with-pgconfig=/usr/pgsql-9.3/bin/pg_config   make   make install Also, set the PYTHONPATH environment variable, as shown here. Alternatively, you can also set it in the .bash_profile script: export PYTHONPATH=/opt/PostgreSQL/9.2/Sky/lib64/python2.6/site-packages/ How to do it... We are going to perform the following sequence of steps to set up replication between two different databases using Londiste. First, create the two databases between which replication has to occur: createdb node1 createdb node2 Populate the node1 database with data using the pgbench utility: pgbench -i -s 2 -F 80 node1 Add any primary key and foreign keys to the tables in the node1 database that are needed for replication. Create the following .sql file and add the following lines to it: Vi /tmp/prepare_pgbenchdb_for_londiste.sql -- add primary key to history table ALTER TABLE pgbench_history ADD COLUMN hid SERIAL PRIMARY KEY;   -- add foreign keys ALTER TABLE pgbench_tellers ADD CONSTRAINT pgbench_tellers_branches_fk FOREIGN KEY(bid) REFERENCES pgbench_branches; ALTER TABLE pgbench_accounts ADD CONSTRAINT pgbench_accounts_branches_fk FOREIGN KEY(bid) REFERENCES pgbench_branches; ALTER TABLE pgbench_history ADD CONSTRAINT pgbench_history_branches_fk FOREIGN KEY(bid) REFERENCES pgbench_branches; ALTER TABLE pgbench_history ADD CONSTRAINT pgbench_history_tellers_fk FOREIGN KEY(tid) REFERENCES pgbench_tellers; ALTER TABLE pgbench_history ADD CONSTRAINT pgbench_history_accounts_fk FOREIGN KEY(aid) REFERENCES pgbench_accounts; We will now load the .sql file created in the previous step and load it into the database: psql node1 -f /tmp/prepare_pgbenchdb_for_londiste.sql We will now populate the node2 database with table definitions from the tables in the node1 database: pg_dump -s -t 'pgbench*' node1 > /tmp/tables.sql psql -f /tmp/tables.sql node2 Now starts the process of replication. We will first create the londiste.ini configuration file with the following parameters in order to set up the root node for the source database, node1: Vi londiste.ini   [londiste3] job_name = first_table db = dbname=node1 queue_name = replication_queue logfile = /home/postgres/log/londiste.log pidfile = /home/postgres/pid/londiste.pid In the next step, we are going to use the londiste.ini configuration file created in the previous step to set up the root node for the node1 database, as shown here: [postgres@localhost bin]$ ./londiste3 londiste3.ini create-root node1 dbname=node1   2014-12-09 18:54:34,723 2335 WARNING No host= in public connect string, bad idea 2014-12-09 18:54:35,210 2335 INFO plpgsql is installed 2014-12-09 18:54:35,217 2335 INFO pgq is installed 2014-12-09 18:54:35,225 2335 INFO pgq.get_batch_cursor is installed 2014-12-09 18:54:35,227 2335 INFO pgq_ext is installed 2014-12-09 18:54:35,228 2335 INFO pgq_node is installed 2014-12-09 18:54:35,230 2335 INFO londiste is installed 2014-12-09 18:54:35,232 2335 INFO londiste.global_add_table is installed 2014-12-09 18:54:35,281 2335 INFO Initializing node 2014-12-09 18:54:35,285 2335 INFO Location registered 2014-12-09 18:54:35,447 2335 INFO Node "node1" initialized for queue "replication_queue" with type "root" 2014-12-09 18:54:35,465 2335 INFO Don We will now run the worker daemon for the root node: [postgres@localhost bin]$ ./londiste3 londiste3.ini worker 2014-12-09 18:55:17,008 2342 INFO Consumer uptodate = 1 In the next step, we will create a slave.ini configuration file in order to make a leaf node for the node2 target database: Vi slave.ini [londiste3] job_name = first_table_slave db = dbname=node2 queue_name = replication_queue logfile = /home/postgres/log/londiste_slave.log pidfile = /home/postgres/pid/londiste_slave.pid We will now initialize the node in the target database: ./londiste3 slave.ini create-leaf node2 dbname=node2 –provider=dbname=node1 2014-12-09 18:57:22,769 2408 WARNING No host= in public connect string, bad idea 2014-12-09 18:57:22,778 2408 INFO plpgsql is installed 2014-12-09 18:57:22,778 2408 INFO Installing pgq 2014-12-09 18:57:22,778 2408 INFO   Reading from /var/lib/pgsql/9.3/Sky/share/skytools3/pgq.sql 2014-12-09 18:57:23,211 2408 INFO pgq.get_batch_cursor is installed 2014-12-09 18:57:23,212 2408 INFO Installing pgq_ext 2014-12-09 18:57:23,213 2408 INFO   Reading from /var/lib/pgsql/9.3/Sky/share/skytools3/pgq_ext.sql 2014-12-09 18:57:23,454 2408 INFO Installing pgq_node 2014-12-09 18:57:23,455 2408 INFO   Reading from /var/lib/pgsql/9.3/Sky/share/skytools3/pgq_node.sql 2014-12-09 18:57:23,729 2408 INFO Installing londiste 2014-12-09 18:57:23,730 2408 INFO   Reading from /var/lib/pgsql/9.3/Sky/share/skytools3/londiste.sql 2014-12-09 18:57:24,391 2408 INFO londiste.global_add_table is installed 2014-12-09 18:57:24,575 2408 INFO Initializing node 2014-12-09 18:57:24,705 2408 INFO Location registered 2014-12-09 18:57:24,715 2408 INFO Location registered 2014-12-09 18:57:24,744 2408 INFO Subscriber registered: node2 2014-12-09 18:57:24,748 2408 INFO Location registered 2014-12-09 18:57:24,750 2408 INFO Location registered 2014-12-09 18:57:24,757 2408 INFO Node "node2" initialized for queue "replication_queue" with type "leaf" 2014-12-09 18:57:24,761 2408 INFO Done We will now launch the worker daemon for the target database, that is, node2: [postgres@localhost bin]$ ./londiste3 slave.ini worker 2014-12-09 18:58:53,411 2423 INFO Consumer uptodate = 1 We will now create the configuration file, that is pgqd.ini, for the ticker daemon: vi pgqd.ini   [pgqd] logfile = /home/postgres/log/pgqd.log pidfile = /home/postgres/pid/pgqd.pid Using the configuration file created in the previous step, we will launch the ticker daemon: [postgres@localhost bin]$ ./pgqd pgqd.ini 2014-12-09 19:05:56.843 2542 LOG Starting pgqd 3.2 2014-12-09 19:05:56.844 2542 LOG auto-detecting dbs ... 2014-12-09 19:05:57.257 2542 LOG node1: pgq version ok: 3.2 2014-12-09 19:05:58.130 2542 LOG node2: pgq version ok: 3.2 We will now add all the tables to the replication on the root node: [postgres@localhost bin]$ ./londiste3 londiste3.ini add-table --all 2014-12-09 19:07:26,064 2614 INFO Table added: public.pgbench_accounts 2014-12-09 19:07:26,161 2614 INFO Table added: public.pgbench_branches 2014-12-09 19:07:26,238 2614 INFO Table added: public.pgbench_history 2014-12-09 19:07:26,287 2614 INFO Table added: public.pgbench_tellers Similarly, add all the tables to the replication on the leaf node: [postgres@localhost bin]$ ./londiste3 slave.ini add-table –all We will now generate some traffic on the node1 source database: pgbench -T 10 -c 5 node1 We will now use the compare utility available with the londiste3 command to check the tables in both the nodes; that is, both the source database (node1) and destination database (node2) have the same amount of data: [postgres@localhost bin]$ ./londiste3 slave.ini compare   2014-12-09 19:26:16,421 2982 INFO Checking if node1 can be used for copy 2014-12-09 19:26:16,424 2982 INFO Node node1 seems good source, using it 2014-12-09 19:26:16,425 2982 INFO public.pgbench_accounts: Using node node1 as provider 2014-12-09 19:26:16,441 2982 INFO Provider: node1 (root) 2014-12-09 19:26:16,446 2982 INFO Locking public.pgbench_accounts 2014-12-09 19:26:16,447 2982 INFO Syncing public.pgbench_accounts 2014-12-09 19:26:18,975 2982 INFO Counting public.pgbench_accounts 2014-12-09 19:26:19,401 2982 INFO srcdb: 200000 rows, checksum=167607238449 2014-12-09 19:26:19,706 2982 INFO dstdb: 200000 rows, checksum=167607238449 2014-12-09 19:26:19,715 2982 INFO Checking if node1 can be used for copy 2014-12-09 19:26:19,716 2982 INFO Node node1 seems good source, using it 2014-12-09 19:26:19,716 2982 INFO public.pgbench_branches: Using node node1 as provider 2014-12-09 19:26:19,730 2982 INFO Provider: node1 (root) 2014-12-09 19:26:19,734 2982 INFO Locking public.pgbench_branches 2014-12-09 19:26:19,734 2982 INFO Syncing public.pgbench_branches 2014-12-09 19:26:22,772 2982 INFO Counting public.pgbench_branches 2014-12-09 19:26:22,804 2982 INFO srcdb: 2 rows, checksum=-3078609798 2014-12-09 19:26:22,812 2982 INFO dstdb: 2 rows, checksum=-3078609798 2014-12-09 19:26:22,866 2982 INFO Checking if node1 can be used for copy 2014-12-09 19:26:22,877 2982 INFO Node node1 seems good source, using it 2014-12-09 19:26:22,878 2982 INFO public.pgbench_history: Using node node1 as provider 2014-12-09 19:26:22,919 2982 INFO Provider: node1 (root) 2014-12-09 19:26:22,931 2982 INFO Locking public.pgbench_history 2014-12-09 19:26:22,932 2982 INFO Syncing public.pgbench_history 2014-12-09 19:26:25,963 2982 INFO Counting public.pgbench_history 2014-12-09 19:26:26,008 2982 INFO srcdb: 715 rows, checksum=9467587272 2014-12-09 19:26:26,020 2982 INFO dstdb: 715 rows, checksum=9467587272 2014-12-09 19:26:26,056 2982 INFO Checking if node1 can be used for copy 2014-12-09 19:26:26,063 2982 INFO Node node1 seems good source, using it 2014-12-09 19:26:26,064 2982 INFO public.pgbench_tellers: Using node node1 as provider 2014-12-09 19:26:26,100 2982 INFO Provider: node1 (root) 2014-12-09 19:26:26,108 2982 INFO Locking public.pgbench_tellers 2014-12-09 19:26:26,109 2982 INFO Syncing public.pgbench_tellers 2014-12-09 19:26:29,144 2982 INFO Counting public.pgbench_tellers 2014-12-09 19:26:29,176 2982 INFO srcdb: 20 rows, checksum=4814381032 2014-12-09 19:26:29,182 2982 INFO dstdb: 20 rows, checksum=4814381032 How it works... The following is an explanation of the steps performed in the preceding section: Initially, in step 1, we create two databases, that is node1 and node2, that are used as the source and target databases, respectively, from a replication perspective. In step 2, we populate the node1 database using the pgbench utility. In step 3 of the preceding section, we add and define the respective primary key and foreign key relationships on different tables and put these DDL commands in a .sql file. In step 4, we execute these DDL commands stated in step 3 on the node1 database; thus, in this way, we force the primary key and foreign key definitions on the tables in the pgbench schema in the node1 database. In step 5, we extract the table definitions from the tables in the pgbench schema in the node1 database and load these definitions in the node2 database. We will now discuss steps 6 to 8 of the preceding section. In step 6, we create the configuration file, which is then used in step 7 to create the root node for the node1 source database. In step 8, we will launch the worker daemon for the root node. Regarding the entries mentioned in the configuration file in step 6, we first define a job that must have a name, so that distinguished processes can be easily identified. Then, we define a connect string with information to connect to the source database, that is node1, and then we define the name of the replication queue involved. Finally, we define the location of the log and pid files. We will now discuss steps 9 to 11 of the preceding section. In step 9, we define the configuration file, which is then used in step 10 to create the leaf node for the target database, that is node2. In step 11, we launch the worker daemon for the leaf node. The entries in the configuration file in step 9 contain the job_name connect string in order to connect to the target database, that is node2, the name of the replication queue involved, and the location of log and pid involved. The key part in step 11 is played by the slave, that is the target database—to find the master or provider, that is source database node1. We will now talk about steps 12 and 13 of the preceding section. In step 12, we define the ticker configuration, with the help of which we launch the ticker process mentioned in step 13. Once the ticker daemon has started successfully, we have all the components and processes setup and needed for replication; however, we have not yet defined what the system needs to replicate. In step 14 and 15, we define the tables to the replication that is set on both the source and target databases, that is node1 and node2, respectively. Finally, we will talk about steps 16 and 17 of the preceding section. Here, at this stage, we are testing the replication that was set up between the node1 source database and the node2 target database. In step 16, we generate some traffic on the node1 source database by running pgbench with five parallel database connections and generating traffic for 10 seconds. In step 17, we check whether the tables on both the source and target databases have the same data. For this purpose, we use the compare command on the provider and subscriber nodes and then count and checksum the rows on both sides. A partial output from the preceding section tells you that the data has been successfully replicated between all the tables that are part of the replication set up between the node1 source database and the node2 destination database, as the count and checksum of rows for all the tables on the source and target destination databases are matching: 2014-12-09 19:26:18,975 2982 INFO Counting public.pgbench_accounts 2014-12-09 19:26:19,401 2982 INFO srcdb: 200000 rows, checksum=167607238449 2014-12-09 19:26:19,706 2982 INFO dstdb: 200000 rows, checksum=167607238449   2014-12-09 19:26:22,772 2982 INFO Counting public.pgbench_branches 2014-12-09 19:26:22,804 2982 INFO srcdb: 2 rows, checksum=-3078609798 2014-12-09 19:26:22,812 2982 INFO dstdb: 2 rows, checksum=-3078609798   2014-12-09 19:26:25,963 2982 INFO Counting public.pgbench_history 2014-12-09 19:26:26,008 2982 INFO srcdb: 715 rows, checksum=9467587272 2014-12-09 19:26:26,020 2982 INFO dstdb: 715 rows, checksum=9467587272   2014-12-09 19:26:29,144 2982 INFO Counting public.pgbench_tellers 2014-12-09 19:26:29,176 2982 INFO srcdb: 20 rows, checksum=4814381032 2014-12-09 19:26:29,182 2982 INFO dstdb: 20 rows, checksum=4814381032 Summary This article demonstrates the high availability and replication concepts in PostgreSQL. After reading this chapter, you will be able to implement high availability and replication options using different techniques including streaming replication, Slony-I replication and replication using Longdiste. Resources for Article: Further resources on this subject: Running a PostgreSQL Database Server [article] Securing the WAL Stream [article] Recursive queries [article]
Read more
  • 0
  • 0
  • 5507

article-image-mobile-administration
Packt
06 Feb 2015
17 min read
Save for later

Mobile Administration

Packt
06 Feb 2015
17 min read
In this article by Paul Goodey, author of the book Salesforce CRM – The Definitive Admin Handbook - Third Edition, we will look at the administration of Salesforce Mobile solutions that can significantly improve productivity and user satisfaction and help them access data and application functionality out of the office. (For more resources related to this topic, see here.) In the past, mobile devices that were capable of accessing software applications were very expensive. Often, these devices were regarded as a nice to have accessory by management and were seen as a company perk by field-based teams. Today, mobile devices are far more prevalent within the business environment, and organizations are increasingly realizing the benefits of using mobile phones and devices to access business applications. Salesforce has taken the lead in recognizing how mobiles have become the new standard for being connected in people's personal and professional lives. It has also highlighted how increasingly, the users of their apps are living lives connected to the Internet, but rather than sitting at a desk in the office, they are in between meetings, on the road, in planes, in trains, in cabs, or even in the queue for lunch. As a result, Salesforce has developed innovative mobile solutions that help you and your users embrace this mobile-first world in Salesforce CRM. Accessing Salesforce Mobile products Salesforce offers two varieties of mobile solutions, namely mobile browser apps and downloadable apps. Mobile browser apps, as the name suggests, are accessed using a web browser that is available on a mobile device. Downloadable apps are accessed by first downloading the client software from, say, the Apple App Store or Google Play and then installing it onto the mobile device. Mobile browser apps and downloadable apps offer various features and benefits and, as we'll see, are available for various Salesforce mobile products and device combinations. Most mobile devices these days have some degree of web browser capability, which can be used to access Salesforce CRM; however, some Salesforce mobile products are optimized for use with certain devices. By accessing a Salesforce mobile browser app, your users do not require anything to be installed. Supported mobile browsers for Salesforce are generally available on Android, Apple, BlackBerry, and Microsoft Windows 8.1 devices. Downloadable apps, on the other hand, will require the app to be first downloaded from the App Store for Apple® devices or from Google Play™ for Android™ devices and then installed on the mobile device. Salesforce mobile products' overview Salesforce has provided certain mobile products as downloadable apps only, while others have been provided as both downloadable and mobile browser-based. The following list outlines the various mobile app products, features, and capabilities used to access Salesforce CRM on mobile devices: SalesforceA Salesforce Touch Salesforce1 Salesforce Classic Salesforce Touch is no longer available and is mentioned here for completeness as this product has been recently incorporated into the Salesforce1 product. SalesforceA SalesforceA is a downloadable system administration app that allows you to manage your organization's users and view certain information for your Salesforce organization from your mobile device. Salesforce A is intended to be used by system administrators, as it is restricted to users with the Manage Users permission. The SalesforceA app provides the facilities to carry out user tasks, such as deactivating or freezing users, resetting passwords, unlocking users, editing user details, calling and emailing users, and assigning permission sets. These user task buttons are displayed as action icons, as shown in the following screenshot: These icons are presented in the action bar at the bottom of the mobile device screen, as shown in the following screenshot: In addition to the user tasks, you can view the system status and also switch between your user accounts in multiple organizations. This allows you to access different organizations and communities without having to log out and log back in to each user account. By staying logged in to multiple accounts in different organizations, you will save time by easily switching to the particular organization user account that you need to access. SalesforceA supported devices At the time of writing, the following devices are supported by Salesforce for use with the SalesforceA downloadable app: Android phones Apple iPhone Apple iPod Touch SalesforceA can be installed from Google Play™ for Android™ phones and the Apple® App Store for Apple devices. Salesforce Touch Salesforce Touch is the name of an earlier Salesforce mobile product and is no longer available. With the Spring 2014 release, Salesforce Touch was incorporated into the Salesforce1 app. Hence, both the Salesforce Touch mobile browser and Salesforce Touch downloadable apps are no longer available; however, the functionality that they once offered is available in Salesforce1, which is covered in this article. Salesforce1 Salesforce1 is Salesforce's next-generation mobile CRM platform that has been designed for Salesforce's customers, developers, and ISVs (independent software vendors) to connect mobile apps, browser apps, and third-party app services. Salesforce1 has been developed for a mobile-first environment and demonstrates how Salesforce's focus as a platform provider aims to connect enterprises with systems that can be programmed through APIs, along with mobile apps and services that can be utilized by marketing, sales, and customer service. There are two ways to use Salesforce1: either using a mobile browser app that users can access by logging into Salesforce from a supported mobile browser or downloadable apps that users can install from the App Store or Google Play. Either way, Salesforce1 allows users to access and update Salesforce data from an interface that has been optimized to navigate and work on their touchscreen mobile devices. Using Salesforce1, records can be viewed, edited, and created. Users can manage their activities, view their dashboards, and use Chatter. Salesforce1 also supports many standard objects and list views, all custom objects, plus the integration of other mobile apps and many of your organization's Salesforce customizations, including Visualforce tabs and pages. Salesforce1 supported devices At the time of writing this, the following devices are supported by Salesforce for the Salesforce1 mobile browser app: Android phones Apple iPad Apple iPhone BlackBerry Z10 Windows 8.1 phones (Beta support) Also, at the time of writing this, Salesforce specifies the following devices as being supported for the Salesforce1 downloadable app: Android phones Apple iPad Apple iPhone Salesforce1 data availability Your organization edition, the user's license type, along with the user's profile and any permission sets, determines the data that is available to the user within Salesforce1. Generally, users have the same visibility of objects, record types, fields, and page layouts that they have while accessing the full Salesforce browser app. However, at the time of writing this, not all data is available in the current release of the Salesforce1 app. In Winter 2015, these key objects are fully accessible from the Salesforce1 navigation menu: Accounts; Campaigns; Cases; Contacts; Contracts; Leads; Opportunities; Tasks; and Users. Dashboards and Events, however, are restricted to being viewable from only the Salesforce1 navigation menu. Custom objects are fully accessible if they have a tab that the user can access. For new users who are yet to build a history of recent objects, they initially see a set of default objects in the Recent section in the Salesforce1 navigation menu. The majority of standard and custom fields, and most of the related lists for the supported objects, are available on these records; however, at the time of writing this, the following exceptions exist: Rich text area field support varies (detailed shortly) Links on formula fields are not supported State and country picklist fields are not supported Related lists in Salesforce1 are restricted (detailed shortly) Rich text area field support varies Support for rich text area fields varies by the version of Salesforce1 and the type of device. For Android's downloadable apps, you can view and edit rich text area fields. However, for Android's mobile browser apps, you can only view rich text area fields; editing is not supported currently. For iOS's downloadable apps, you can view but not edit rich text area fields. However, for iOS's mobile browser apps, you can view and also edit rich text area fields. Finally, for both BlackBerry and Windows 8.1 mobile browser apps, you can neither view nor edit rich text area fields. Related lists in Salesforce1 Related lists in Salesforce1 are restricted and display the first four fields that are defined on the page layout for that object. The number of fields shown cannot be increased. If Chatter is enabled, users can also access feeds, people, groups, and Salesforce Files. When users are working with records in the full Salesforce app, it can take up to 15 days for this data to appear in the Recent section; thus, to make records appear under the Recent section sooner, ask users to pin them from their search results in the full Salesforce site. Salesforce1 administration You can manage your organization's access to Salesforce1 apps; there are two areas of administration: the mobile browser app that users can access by logging in to Salesforce from a supported mobile browser and the downloadable app that users can install from the App Store or Google Play. The upcoming sections describe the ways to control user access to each of these mobile apps. Salesforce1 mobile browser app access You can control whether users can access the Salesforce1 mobile browser app when they log into Salesforce from a mobile browser. To select or deselect this feature, navigate to Setup | Mobile Administration | Salesforce1 | Settings, as shown in the following screenshot: By selecting the Enable the Salesforce1 mobile browser app checkbox, all users are activated to access Salesforce1 from their mobile browsers. Deselecting this option turns off the mobile browser app, which means that users will automatically access the full Salesforce site from their mobile browser. By default, the mobile browser app is turned on in all Salesforce organizations. Salesforce1 desktop browser access Selecting the Enable the Salesforce1 mobile browser app checkbox, as described in the previous section, permits users who are activated to access Salesforce1 from their desktop browsers. Users can navigate to the Salesforce1 app within their desktop browser by appending “/one/one.app” to the end of the Salesforce URL. As an example, for the following Salesforce URL accessed from the server na10, you would enter the https://na10.salesforce.com/one/one.app desktop browser URL. Salesforce1 downloadable app access The Salesforce1 app is distributed as a managed package, and within Salesforce, it is implemented as a connected app. You might already see the Salesforce1 connected app in your list of installed apps as it might have been automatically installed in your organization. The list of included apps can change with each Salesforce release but, to simplify administration, each package is asynchronously installed in Salesforce organizations whenever any user in that organization first accesses Salesforce1. However, to manually install or reinstall the Salesforce1 package for connected apps, you can install it from the AppExchange. To view the details for the Salesforce1 app in the connected app settings, navigate to Setup | Manage Apps | Connected Apps. The apps that connect to your Salesforce organization are then listed as shown in the following screenshot: Salesforce1 notifications Notifications allow all users in your organization to receive mobile notifications in Salesforce1, for example, whenever they are mentioned in Chatter or whenever they receive approval requests. To activate mobile notifications, navigate to Setup | Mobile Administration | Notifications | Settings, as shown in the following screenshot: The settings for notifications can be set as follows: Enable in-app notifications: Set this option to keep users notified about relevant Salesforce activity while they are using Salesforce1. Enable push notifications: Set this option to keep users notified of relevant Salesforce activity when they are not using the Salesforce1 downloadable app. Include full content in push notifications: Keep this checkbox unchecked if you do not want users to receive full content in push notifications. This can prevent users from receiving potentially sensitive data that might be in comments, for example. If you set this option, a pop-up dialog appears, displaying terms and conditions where you must click on OK or Cancel. Salesforce1 branding This option allows you to customize the appearance of the Salesforce1 app so that it complies with any company branding requirements that might be in place. Salesforce1 branding is supported in downloadable apps' Version 5.2 or higher and also in the mobile browser app. To specify Salesforce1 branding, navigate to Setup | Mobile Administration | Salesforce1 | Branding, as shown in the following screenshot: Salesforce1 compact layouts In Salesforce1, compact layouts are used to display the key fields on a record and are specifically designed to view records on touchscreen mobile devices. As space is limited on mobile devices and quick recognition of records is important, the first four fields that you assign to a compact layout are displayed. If a mobile user does not have the required access to one of the first four fields that have been assigned to a compact layout, the next field, if more than four fields have been set on the layout, is used. If you are yet to create custom compact layouts, the records will be displayed using a read-only, predefined system default compact layout, and after you have created a custom compact layout, you can then set it as the primary compact layout for that object. As with the full Salesforce CRM site, if you have record types associated with an object, you can alter the primary compact layout assignment and assign specific compact layouts to different record types. You can also clone a compact layout from its detail page. The upcoming field types cannot be included on compact layouts: text area, long text area, rich text area, and multiselect picklists. Salesforce1 offline access In Salesforce1, the mechanism to handle offline access is determined by users' most recently used records. These records are cached for offline access; at the time of writing this, they are read-only. The cached data is encrypted and secured through persistent storage by Salesforce1's downloadable apps. Offline access is available in Salesforce1's downloadable apps Version 6.0 and higher and was first released in Summer 2014. Offline access is enabled by default when Salesforce1's downloadable app is installed. To manage these settings, navigate to Setup | Mobile Administration | Offline. Now, check or uncheck Enable Offline Sync for Salesforce1, as shown in the following screenshot: When offline access is enabled, data based on the objects is downloaded to each user's mobile device and presented in the Recent section of the Salesforce1 navigation menu and on the user's most recently viewed records. The data is encrypted and stored in a secure, persistent cache on the mobile device. Setting up Salesforce1 with the Salesforce1 Wizard The Salesforce1 Wizard simplifies the setting up of the Salesforce1 mobile app. The wizard offers a visual tour of the key setup steps and is useful if you are new to Salesforce1 or need to quickly set up the core Salesforce1 settings. The Salesforce1 Wizard guides you through the setting up of the following Salesforce1 configuration steps: Choose which items appear in the navigation menu Configure global actions Create a contact custom compact layout Optionally, invite users to start using the Salesforce1 app To access the Salesforce1 Wizard, navigate to Setup | Salesforce1 Setup. Now, click on Launch Quick Start Wizard within the Salesforce1 Setup page, as shown in the following screenshot: Upon clicking on the Let's Get Started section link (shown in the following screenshot), you will be presented with the Salesforce1 Setup visual tour, as shown in the next section. The Quick Start Wizard The Quick Start Wizard guides you through the minimum configuration steps required to set up Salesforce1. By clicking on the Launch Quick Start Wizard button, the process to complete the essential setup tasks for Salesforce1 is initiated and provides a step-by-step wizard guide. The five steps are: Customize the Navigation Menu: This step results in the setup of the navigation menu for all users in your organization. To reorder items, drag them up and down. To remove items, drag them to the Available Items list, as shown in the following screenshot: Arrange Global Actions: Global actions provide users with quick access to Salesforce functions and in this step, you will choose and arrange the Salesforce1 global actions, as shown in the following screenshot: Actions might might have a different appearance, depending upon your version of Salesforce1. Create a Custom Compact Layout for Contacts: Compact layouts are used to show the key fields on a record in the highlights area at the top of the record detail. In this step, you are able to create a custom compact layout for contacts to set, for example, a contact's name, e-mail, and phone number, as shown in the following screenshot: However, after you have completed the Quick Start Wizard, you can create compact layouts for other objects as required. Review: In this step, you are given the chance to preview the changes to verify the results of the changes, as shown in the following screenshot: The review step screen gives you a live preview that uses your current access as the logged-in user. Send Invitations: This is the final step of the Quick Start Wizard, which will provide you with a basic setup of Salesforce1 and allow you to get feedback on what you have implemented. In this step, you can invite your users to start using the Salesforce1 app, as shown in the following screenshot: This step can be skipped and you can always send invitations later from the Salesforce1 setup page. You can also implement additional options to customize the app, such as incorporating your own branding. Differences between Salesforce1 and the full Salesforce CRM browser app In the Winter 2015 release and at the time of writing this, Salesforce1 does not have all of the features of the full Salesforce CRM site; moreover, in some areas, it includes functionality that is not available in, or is different from, the complete Salesforce site. As an example, on the full Salesforce CRM site, compact layouts determine which fields appear in the Chatter feed item and which appear after a user creates a record via a publisher action. However, compact layouts in Salesforce1 are used to display the key fields on a record. For details about the features that differ between the full Salesforce CRM site and Salesforce1, refer to Salesforce1 Limits and Differences from the Full Salesforce Site within the Salesforce Help menu sections. Summary In this article, we looked at ways in which mobile has become the new normal way to stay connected in both our personal and professional lives. Salesforce has recognized this well; we are all spending time being connected to the cloud and using business applications. However, instead of sitting at a desk, users are often on the go. To try and help their customers become successful businesses of this mobile-first world, Salesforce has produced mobile solutions that can help user get things done regardless of where they are and what they are doing. We looked at SalesforceA, which is an admin specific app that can help you manage users and monitor the status of Salesforce while on the move. We discussed Salesforce Touch, which is being replaced with Salesforce1, and we also spoke about the features and benefits of Salesforce1, which is available as a downloadable app and a browser app. Resources for Article: Further resources on this subject: Customization in Microsoft Dynamics CRM [Article] Getting Started with Microsoft Dynamics CRM 2013 Marketing [Article] Diagnostic leveraging of the Accelerated POC with the CRM Online service [Article]
Read more
  • 0
  • 0
  • 2036

article-image-remote-access
Packt
06 Feb 2015
32 min read
Save for later

Remote Access

Packt
06 Feb 2015
32 min read
In this article by Jordan Krause, author of the book Windows Server 2012 R2 Administrator Cookbook, we will see how Windows Server 2012 R2 by Microsoft brings a whole new way of looking at remote access. Companies have historically relied on third-party tools to connect remote users into the network, such as traditional and SSL VPN provided by appliances from large networking vendors. I'm here to tell you those days are gone. Those of us running Microsoft-centric shops can now rely on Microsoft technologies to connect our remote workforce. Better yet is that these technologies are included with the Server 2012 R2 operating system, and have functionality that is much improved over anything that a traditional VPN can provide. Regular VPN does still have a place in the remote access space, and the great news is that you can also provide it with Server 2012 R2. Our primary focus for this article will be DirectAccess (DA). DA is kind of like automatic VPN. There is nothing the user needs to do in order to be connected to work. Whenever they are on the Internet, they are also connected automatically to the corporate network. DirectAccess is an amazing way to have your Windows 7 and Windows 8 domain joined systems connected back to the network for data access and for management of those traveling machines. DirectAccess has actually been around since 2008, but the first version came with some steep infrastructure requirements and was not widely used. Server 2012 R2 brings a whole new set of advantages and makes implementation much easier than in the past. I still find many server and networking admins who have never heard of DirectAccess, so let's spend some time together exploring some of the common tasks associated with it. In this article, we will cover the following recipes: Configuring DirectAccess, VPN, or a combination of the two Pre-staging Group Policy Objects (GPOs) to be used by DirectAccess Enhancing the security of DirectAccess by requiring certificate authentication Building your Network Location Server (NLS) on its own system  (For more resources related to this topic, see here.) There are two "flavors" of remote access available in Windows Server 2012 R2. The most common way to implement the Remote Access role is to provide DirectAccess for your Windows 7 and Windows 8 domain joined client computers, and VPN for the rest. The DirectAccess machines are typically your company-owned corporate assets. One of the primary reasons that DirectAccess is usually only for company assets is that the client machines must be joined to your domain, because the DirectAccess configuration settings are brought down to the client through a GPO. I doubt you want home and personal computers joining your domain. VPN is therefore used for down level clients such as Windows XP, and for home and personal devices that want to access the network. Since this is a traditional VPN listener with all regular protocols available such as PPTP, L2TP, SSTP, it can even work to connect devices such as smartphones. There is a third function available within the Server 2012 R2 Remote Access role, called the Web Application Proxy ( WAP ). This function is not used for connecting remote computers fully into the network as DirectAccess and VPN are; rather, WAP is used for publishing internal web resources out to the internet. For example, if you are running Exchange and Lync Server inside your network and want to publish access to these web-based resources to the internet for external users to connect to, WAP would be a mechanism that could publish access to these resources. The term for publishing out to the internet like this is Reverse Proxy, and WAP can act as such. It can also behave as an ADFS Proxy. For further information on the WAP role, please visit: http://technet.microsoft.com/en-us/library/dn584107.aspx One of the most confusing parts about setting up DirectAccess is that there are many different ways to do it. Some are good ideas, while others are not. Before we get rolling with recipes, we are going to cover a series of questions and answers to help guide you toward a successful DA deployment. The first question that always presents itself when setting up DA is "How do I assign IP addresses to my DirectAccess server?". This is quite a loaded question, because the answer depends on how you plan to implement DA, which features you plan to utilize, and even upon how secure you believe your DirectAccess server to be. Let me ask you some questions, pose potential answers to those questions, and discuss the effects of making each decision. DirectAccess Planning Q&A Which client operating systems can connect using DirectAccess? Answer: Windows 7 Ultimate, Windows 7 Enterprise, and Windows 8.x Enterprise. You'll notice that the Professional SKU is missing from this list. That is correct, Windows 7 and Windows 8 Pro do not contain the DirectAccess connectivity components. Yes, this does mean that Surface Pro tablets cannot utilize DirectAccess out of the box. However, I have seen many companies now install Windows 8 Enterprise onto their Surface tablets, effectively turning them into "Surface Enterprises." This works fine and does indeed enable them to be DirectAccess clients. In fact, I am currently typing this text on a DirectAccess-connected Surface "Pro turned Enterprise" tablet. Do I need one or two NICs on my DirectAccess server? Answer: Technically, you could set it up either way. In practice however, it really is designed for dual-NIC implementation. Single NIC DirectAccess works okay sometimes to establish a proof-of-concept to test out the technology. But I have seen too many problems with single NIC implementation in the field to ever recommend it for production use. Stick with two network cards, one facing the internal network and one facing the Internet. Do my DirectAccess servers have to be joined to the domain? Answer: Yes. Does DirectAccess have site-to-site failover capabilities? Answer: Yes, though only Windows 8.x client computers can take advantage of it. This functionality is called Multi-Site DirectAccess. Multiple DA servers that are spread out geographically can be joined together in a multi-site array. Windows 8 client computers keep track of each individual entry point and are able to swing between them as needed or at user preference. Windows 7 clients do not have this capability and will always connect through their primary site. What are these things called 6to4, Teredo, and IP-HTTPS I have seen in the Microsoft documentation? Answer: 6to4, Teredo, and IP-HTTPS are all IPv6 transition tunneling protocols. All DirectAccess packets that are moving across the internet between DA client and DA server are IPv6 packets. If your internal network is IPv4, then when those packets reach the DirectAccess server they get turned down into IPv4 packets, by some special components called DNS64 and NAT64. While these functions handle the translation of packets from IPv6 into IPv4 when necessary inside the corporate network, the key point here is that all DirectAccess packets that are traveling over the Internet part of the connection are always IPv6. Since the majority of the Internet is still IPv4, this means that we must tunnel those IPv6 packets inside something to get them across the Internet. That is the job of 6to4, Teredo, and IP-HTTPS. 6to4 encapsulates IPv6 packets into IPv4 headers and shuttles them around the internet using protocol 41. Teredo similarly encapsulates IPv6 packets inside IPv4 headers, but then uses UDP port 3544 to transport them. IP-HTTPS encapsulates IPv6 inside IPv4 and then inside HTTP encrypted with TLS, essentially creating an HTTPS stream across the Internet. This, like any HTTPS traffic, utilizes TCP port 443. The DirectAccess traffic traveling inside either kind of tunnel is always encrypted, since DirectAccess itself is protected by IPsec. Do I want to enable my clients to connect using Teredo? Answer: Most of the time, the answer here is yes. Probably the biggest factor that weighs on this decision is whether or not you are still running Windows 7 clients. When Teredo is enabled in an environment, this gives the client computers an opportunity to connect using Teredo, rather than all clients connecting in over the IP-HTTPS protocol. IP-HTTPS is sort of the "catchall" for connections, but Teredo will be preferred by clients if it is available. For Windows 7 clients, Teredo is quite a bit faster than IP-HTTPS. So enabling Teredo on the server side means your Windows 7 clients (the ones connecting via Teredo) will have quicker response times, and the load on your DirectAccess server will be lessened. This is because Windows 7 clients who are connecting over IP-HTTPS are encrypting all of the traffic twice. This also means that the DA server is encrypting/decrypting everything that comes and goes twice. In Windows 8, there is an enhancement that brings IP-HTTPS performance almost on par with Teredo, and so environments that are fully cut over to Windows 8 will receive less benefit from the extra work that goes into making sure Teredo works. Can I place my DirectAccess server behind a NAT? Answer: Yes, though there is a downside. Teredo cannot work if the DirectAccess server is sitting behind a NAT. For Teredo to be available, the DA server must have an External NIC that has two consecutive public IP addresses. True public addresses. If you place your DA server behind any kind of NAT, Teredo will not be available and all clients will connect using the IP-HTTPS protocol. Again, if you are using Windows 7 clients, this will decrease their speed and increase the load on your DirectAccess server. How many IP addresses do I need on a standalone DirectAccess server? Answer: I am going to leave single NIC implementation out of this answer since I don't recommend it anyway. For scenarios where you are sitting the External NIC behind a NAT or, for any other reason, are limiting your DA to IP-HTTPS only, then we need one external address and one internal address. The external address can be a true public address or a private NATed DMZ address. Same with the internal; it could be a true internal IP or a DMZ IP. Make sure both NICs are not plugged into the same DMZ, however. For a better installation scenario that allows Teredo connections to be possible, you would need two consecutive public IP addresses on the External NIC and a single internal IP on the Internal NIC. This internal IP could be either true internal or DMZ. But the public IPs would really have to be public for Teredo to work. Do I need an internal PKI? Answer: Maybe. If you want to connect Windows 7 clients, then the answer is yes. If you are completely Windows 8, then technically you do not need internal PKI. But you really should use it anyway. Using an internal PKI, which can be a single, simple Windows CA server, increases the security of your DirectAccess infrastructure. You'll find out during this article just how easy it is to require certificates as part of the tunnel building authentication process. Configuring DirectAccess, VPN, or a combination of the two Now that we have some general ideas about how we want to implement our remote access technologies, where do we begin? Most services that you want to run on a Windows Server begin with a role installation, but the implementation of remote access begins before that. Let's walk through the process of taking a new server and turning it into a Microsoft Remote Access server. Getting ready All of our work will be accomplished on a new Windows Server 2012 R2. We are taking the two-NIC approach to networking, and so we have two NICs installed on this server. The Internal NIC is plugged into the corporate network and the External NIC is plugged into the Internet for the sake of simplicity. The External NIC could just as well be plugged into a DMZ. How to do it... Follow these steps to turn your new server into a Remote Access server: Assign IP addresses to your server. Remember, the most important part is making sure that the Default Gateway goes on the External NIC only. Join the new server to your domain. Install an SSL certificate onto your DirectAccess server that you plan to use for the IP-HTTPS listener. This is typically a certificate purchased from a public CA. If you're planning to use client certificates for authentication, make sure to pull down a copy of the certificate to your DirectAccess server. You want to make sure certificates are in place before you start with the configuration of DirectAccess. This way the wizards will be able to automatically pull in information about those certificates in the first run. If you don't, DA will set itself up to use self-signed certificates, which are a security no-no. Use Server Manager to install the Remote Access role. You should only do this after completing the steps listed earlier. If you plan to load balance multiple DirectAccess servers together at a later time, make sure to also install the feature called Network Load Balancing . After selecting your role and feature, you will be asked which Remote Access role services you want to install. For our purposes in getting the remote workforce connected back into the corporate network, we want to choose DirectAccess and VPN (RAS) .  Now that the role has been successfully installed, you will see a yellow exclamation mark notification near the top of Server Manager indicating that you have some Post-deployment Configuration that needs to be done. Do not click on Open the Getting Started Wizard ! Unfortunately, Server Manager leads you to believe that launching the Getting Started Wizard (GSW) is the logical next step. However, using the GSW as the mechanism for configuring your DirectAccess settings is kind of like roasting a marshmallow with a pair of tweezers. In order to ensure you have the full range of options available to you as you configure your remote access settings, and that you don't get burned later, make sure to launch the configuration this way: Click on the Tools menu from inside Server Manager and launch the Remote Access Management Console . In the left window pane, click on Configuration | DirectAccess and VPN . Click on the second link, the one that says Run the Remote Access Setup Wizard . Please note that once again the top option is to run that pesky Getting Started Wizard. Don't do it! I'll explain why in the How it works… section of this recipe. Now you have a choice that you will have to answer for yourself. Are you configuring only DirectAccess, only VPN, or a combination of the two? Simply click on the option that you want to deploy. Following your choice, you will see a series of steps (steps 1 through 4) that need to be accomplished. This series of mini-wizards will guide you through the remainder of the DirectAccess and VPN particulars. This recipe isn't large enough to cover every specific option included in those wizards, but at least you now know the correct way to bring a DirectAccess/VPN server into operation. How it works... The remote access technologies included in Server 2012 R2 have great functionality, but their initial configuration can be confusing. Following the procedure listed in this recipe will set you on the right path to be successful in your deployment, and prevent you from running into issues down the road. The reason that I absolutely recommend you stay away from using the "shortcut" deployment method provided by the Getting Started Wizard is twofold: GSW skips a lot of options as it sets up DirectAccess, so you don't really have any understanding of how it works after finishing. You may have DA up and running, but have no idea how it's authenticating or working under the hood. This holds so much potential for problems later, should anything suddenly stop working. GSW employs a number of bad security practices in order to save time and effort in the setup process. For example, using the GSW usually means that your DirectAccess server will be authenticating users without client certificates, which is not a best practice. Also, it will co-host something called the NLS website on itself, which is also not a best practice. Those who utilize the GSW to configure DirectAccess will find that their GPO, which contains the client connectivity settings, will be security-filtered to the Domain Computers group. Even though it also contains a WMI filter that is supposed to limit that policy application to mobile hardware such as laptops, this is a terribly scary thing to see inside GPO filtering settings. You probably don't want all of your laptops to immediately start getting DA connectivity settings, but that is exactly what the GSW does for you. Perhaps worst, the GSW will create and make use of self-signed SSL certificates to validate its web traffic, even the traffic coming in from the Internet! This is a terrible practice and is the number one reason that should convince you that clicking on the Getting Started Wizard is not in your best interests. Pre-staging Group Policy Objects (GPOs) to be used by DirectAccess One of the great things about DirectAccess is that all of the connectivity settings the client computers need in order to connect are contained within a Group Policy Object (GPO). This means that you can turn new client computers into DirectAccess-connected clients without ever touching that system. Once configured properly, all you need to do is add the new computer account to an Active Directory security group, and during the next automatic Group Policy refresh cycle (usually within 90 minutes), that new laptop will be connecting via DirectAccess whenever outside the corporate network. You can certainly choose not to pre-stage anything with the GPOs and DirectAccess will still work. When you get to the end of the DA configuration wizards, it will inform you that two new GPOs are about to be created inside Active Directory. One GPO is used to contain the DirectAccess server settings and the other GPO is used to contain the DirectAccess client settings. If you allow the wizard to handle the generation of these GPOs, it will create them, link them, filter them, and populate them with settings automatically. About half of the time I see folks do it this way and they are forever happy with letting the wizard manage those GPOs now and in the future. The other half of the time, it is desired that we maintain a little more personal control over the GPOs. If you are setting up a new DA environment but your credentials don't have permission to create GPOs, the wizard is not going to be able to create them either. In this case, you will need to work with someone on your Active Directory team to get them created. Another reason to manage the GPOs manually is to have better control over placement of these policies. When you let the DirectAccess wizard create the GPOs, it will link them to the top level of your domain. It also sets Security Filtering on those GPOs so they are not going to be applied to everything in your domain, but when you open up the Group Policy Management Console you will always see those DirectAccess policies listed right up there at the top level of the domain. Sometimes this is simply not desirable. So for this reason also, you may want to choose to create and manage the GPOs by hand, so that we can secure placement and links where we specifically want them to be located. The key factors here are to make sure your DirectAccess Server Settings GPO applies to only the DirectAccess server or servers in your environment. And that the DirectAccess Client Settings GPO applies to only the DA client computers that you plan to enable in your network. The best practice here is to specify this GPO to only apply to a specific Active Directory security group so that you have full control over which computer accounts are in that group. I have seen some folks do it based only on the OU links and include whole OUs in the filtering for the clients GPO (foregoing the use of an AD group at all), but doing it this way makes it quite a bit more difficult to add or remove machines from the access list in the future. Requiring certificates as part of your DirectAccess tunnel authentication process is a good idea in any environment. It makes the solution more secure, and enables advanced functionality. The primary driver for most companies to require these certificates is the enablement of Windows 7 clients to connect via DirectAccess, but I suggest that anyone using DirectAccess in any capacity make use of these certs. They are simple to deploy, easy to configure, and give you some extra peace of mind that only computers who have a certificate issued directly to them from your own internal CA server are going to be able to connect through your DirectAccess entry point. Getting ready While the DirectAccess wizards themselves are run from the DirectAccess server, our work with this recipe is not. The Group Policy settings that we will be configuring are all accomplished within Active Directory, and we will be doing the work from a Domain Controller in our environment. How to do it... To pre-stage Group Policy Objects (GPOs) for use with DirectAccess: On your Domain Controller, launch the Group Policy Management Console . Expand Forest | Domains | Your Domain Name . There should be a listing here called Group Policy Object . Right-click on that and choose New . Name your new GPO something like DirectAccess Server Settings. Click on the new DirectAccess Server Settings GPO and it should open up automatically to the Scope tab. We need to adjust the Security Filtering section so that this GPO only applies to our DirectAccess server. This is a critical step for each GPO to ensure the settings that are going to be placed here do not get applied to the wrong computers. Remove Authenticated Users that is prepopulated in that list. The list should now be empty. Click the Add… button and search for the computer account of your DirectAccess server. Mine is called RA-01. By default this window will only search user accounts, so you will need to adjust Object Types to include Computers before it will allow you to add your server into this filtering list. Your Security Filtering list should now look like this:  Now click on the Details tab of your GPO. Change the GPO Status to be User configuration settings disabled . We do this because our GPO is only going to contain computer-level settings, nothing at the user level. The last thing to do is link your GPO to an appropriate container. Since we have Security Filtering enabled, our GPO is only ever going to apply its settings to the RA-01 server; however, without creating a link, the GPO will not even attempt to apply itself to anything. My RA-01 server is sitting inside the OU called Remote Access Servers . So I will right-click on my Remote Access Servers OU and choose Link an Existing GPO… .  Choose the new DirectAccess Server Settings from the list of available GPOs and click on the OK button. This creates the link and puts the GPO into action. Since there are not yet any settings inside the GPO, it won't actually make any changes on the server. The DirectAccess configuration wizards take care of populating the GPO with the settings that are needed. Now we simply need to rinse and repeat all of these steps to create another GPO, something like DirectAccess Client Settings . You want to set up the client settings GPO in the same way. Make sure that it is filtering to only the Active Directory Security Group that you created to contain your DirectAccess client computers. And make sure to link it to an appropriate container that will include those computer accounts. So maybe your clients GPO will look something like this:  How it works... Creating GPOs in Active Directory is a simple enough task, but it is critical that you configure the Links and Security Filtering correctly. If you do not take care to ensure that these DirectAccess connection settings are only going to apply to the machines that actually need the settings, you could create a world of trouble by internal servers getting remote access connection settings and cause them issues with connection while inside the network. Enhancing the security of DirectAccess by requiring certificate authentication When a DirectAccess client computer builds its IPsec tunnels back to the corporate network, it has the ability to require a certificate as part of that authentication process. In earlier versions of DirectAccess, the one in Server 2008 R2 and the one provided by Unified Access Gateway ( UAG ), these certificates were required in order to make DirectAccess work. Setting up the certificates really isn't a big deal at all; as long as there is a CA server in your network you are already prepared to issue the certs needed at no cost. Unfortunately, though, there must have been enough complaints back to Microsoft in order for them to make these certificates "recommended" instead of "required" and they created a new mechanism in Windows 8 and Server 2012 called KerberosProxy that can be used to authenticate the tunnels instead. This allows the DirectAccess tunnels to build without the computer certificate, making that authentication process less secure. I'm here to strongly recommend that you still utilize certificates in your installs! They are not difficult to set up, and using them makes your tunnel authentication stronger. Further, many of you may not have a choice and will still be required to install these certificates. Only simple DirectAccess scenarios that are all Windows 8 on the client side can get away with the shortcut method of foregoing certs. Anybody who still wants to connect Windows 7 via DirectAccess will need to use certificates on all of their client computers, both Windows 7 and Windows 8. In addition to Windows 7 access, anyone who intends to use the advanced features of DirectAccess such as load balancing, multi-site, or two-factor authentication will also need to utilize these certificates. With any of these scenarios, certificates become a requirement again, not a recommendation. In my experience, almost everyone still has Windows 7 clients that would benefit from being DirectAccess connected, and it's always a good idea to make your DA environment redundant by having load balanced servers. This further emphasizes the point that you should just set up certificate authentication right out of the gate, whether or not you need it initially. You might decide to make a change later that would require certificates and it would be easier to have them installed from the get-go rather than trying to incorporate them later into a running DA environment. Getting ready In order to distribute certificates, you will need a CA server running in your network. Once certificates are distributed to the appropriate places, the rest of our work will be accomplished from our Server 2012 R2 DirectAccess server. How to do it... Follow these steps to make use of certificates as part of the DirectAccess tunnel authentication process: The first thing that you need to do is distribute certificates to your DirectAccess servers and all DirectAccess client computers. The easiest way to do this is by using the built-in Computer template provided by default in a Windows CA server. If you desire to build a custom certificate template for this purpose, you can certainly do so. I recommend that you duplicate the Computer template and build it from there. Whenever I create a custom template for use with DirectAccess, I try to make sure that it meets the following criterias: The Subject Name of the certificate should match the Common Name of the computer (which is also the FQDN of the computer). The Subject Alternative Name ( SAN ) of the certificate should match the DNS Name of the computer (which is also the FQDN of the computer). The certificate should serve the Intended Purposes of both Client Authentication and Server Authentication . You can issue the certificates manually using Microsoft Management Console (MMC). Otherwise, you can lessen your hands-on administrative duties by enabling Autoenrollment. Now that we have certificates distributed to our DirectAccess clients and servers, log in to your primary DirectAccess server and open up the Remote Access Management Console . Click on Configuration in the top-left corner. You should now see steps 1 through 4 listed. Click Edit… listed under Step 2 . Now you can either click Next twice or click on the word Authentication to jump directly to the authentication screen. Check the box that says Use computer certificates . Now we have to specify the Certification Authority server that issued our client certificates. If you used an intermediary CA to issue your certs, make sure to check the appropriate checkbox. Otherwise, most of the time, certificates are issued from a root CA and in this case you would simply click on the Browse… button and look for your CA in the list. This screen is sometimes confusing because people expect to have to choose the certificate itself from the list. This is not the case. What you are actually choosing from this list is the Certificate Authority server that issued the certificates. Make any other appropriate selections on the Authentication screen. For example, many times when we require client certificates for authentication, it is because we have Windows 7 computers that we want to connect via DirectAccess. If that is the case for you, select the checkbox for Enable Windows 7 client computers to connect via DirectAccess .  How it works... Requiring certificates as part of your DirectAccess tunnel authentication process is a good idea in any environment. It makes the solution more secure, and enables advanced functionality. The primary driver for most companies to require these certificates is the enablement of Windows 7 clients to connect via DirectAccess, but I suggest that anyone using DirectAccess in any capacity make use of these certs. They are simple to deploy, easy to configure, and give you some extra peace of mind that only computers who have a certificate issued directly to them from your own internal CA server are going to be able to connect through your DirectAccess entry point. Building your Network Location Server (NLS) on its own system If you zipped through the default settings when configuring DirectAccess, or worse used the Getting Started Wizard, chances are that your Network Location Server ( NLS ) is running right on the DirectAccess server itself. This is not the recommended method for using NLS, it really should be running on a separate web server. In fact, if you later want to do something more advanced such as setting up load balanced DirectAccess servers, you're going to have to move NLS off onto a different server anyway. So you might as well do it right the first time. NLS is a very simple requirement, yet a critical one. It is just a website, it doesn't matter what content the site has, and it only has to run inside your network. Nothing has to be externally available. In fact, nothing should be externally available, because you only want this site being accessed internally. This NLS website is a large part of the mechanism by which DirectAccess client computers figure out when they are inside the office and when they are outside. If they can see the NLS website, they know they are inside the network and will disable DirectAccess name resolution, effectively turning off DA. If they do not see the NLS website, they will assume they are outside the corporate network and enable DirectAccess name resolution. There are two gotchas with setting up an NLS website: The first is that it must be HTTPS, so it does need a valid SSL certificate. Since this website is only running inside the network and being accessed from domain-joined computers, this SSL certificate can easily be one that has been issued from your internal CA server. So no cost associated there. The second catch that I have encountered a number of times is that for some reason the default IIS splash screen page doesn't make for a very good NLS website. If you set up a standard IIS web server and use the default site as NLS, sometimes it works to validate the connections and sometimes it doesn't. Given that, I always set up a specific site that I create myself, just to be on the safe side. So let's work together to follow the exact process I always take when setting up NLS websites in a new DirectAccess environment. Getting ready Our NLS website will be hosted on an IIS server we have that runs Server 2012 R2. Most of the work will be accomplished from this web server, but we will also be creating a DNS record and will utilize a Domain Controller for that task. How to do it... Let's work together to set up our new Network Location Server website: First decide on an internal DNS name to use for this website and set it up in DNS of your domain. I am going to use nls.mydomain.local and am creating a regular Host (A) record that points nls.mydomain.local at the IP address of my web server. Now log in to that web server and let's create some simple content for this new website. Create a new folder called C:NLS. Inside your new folder, create a new Default.htm file. Edit this file and throw some simple text in there. I usually say something like This is the NLS website used by DirectAccess. Please do not delete or modify me!.  Remember, this needs to be an HTTPS website, so before we try setting up the actual website, we should acquire the SSL certificate that we need to use with this site. Since this certificate is coming from my internal CA server, I'm going to open up MMC on my web server to accomplish this task. Once MMC is opened, snap-in the Certificates module. Make sure to choose Computer account and then Local computer when it prompts you for which certificate store you want to open. Expand Certificates (Local Computer) | Personal | Certificates . Right-click on this Certificates folder and choose All Tasks | Request New Certificate… . Click Next twice and you should see your list of certificate templates that are available on your internal CA server. If you do not see one that looks appropriate for requesting a website certificate, you may need to check over the settings on your CA server to make sure the correct templates are configured for issuing. My template is called Custom Web Server . Since this is a web server certificate, there is some additional information that I need to provide in my request in order to successfully issue a certificate. So I go ahead and click on that link that says More information is required to enroll for this certificate. Click here to configure settings. .  Drop-down the Subject name | Type menu and choose the option Common name . Enter a common name for our website into the Value field, which in my case is nls.mydomain.local. Click the Add button and your CN should move over to the right side of the screen like this:  Click on OK then click on the Enroll button. You should now have an SSL certificate sitting in your certificates store that can be used to authenticate traffic moving to our nls.mydomain.local name. Open up Internet Information Services (IIS) Manager , and browse to the Sites folder. Go ahead and remove the default website that IIS automatically set up, so that we can create our own NLS website without any fear of conflict. Click on the Add Website… action. Populate the information as shown in the following screenshot. Make sure to choose your own IP address and SSL certificate from the lists, of course:  Click the OK button and you now have an NLS website running successfully in your network. You should be able to open up a browser on a client computer sitting inside the network and successfully browse to https://nls.mydomain.local. How it works... In this recipe, we configured a basic Network Location Server website for use with our DirectAccess environment. This site will do exactly what we need it to when our DA client computers try to validate whether they are inside or outside the corporate network. While this recipe meets our requirements for NLS, and in fact puts us into a good practice of installing DirectAccess with NLS being hosted on its own web server, there is yet another step you could take to make it even better. Currently this web server is a single point of failure for NLS. If this web server goes down or has a problem, we would have DirectAccess client computers inside the office who would think they are outside, and they would have some major name resolution problems until we sorted out the NLS problem. Given that, it is a great idea to make NLS redundant. You could cluster servers together, use Microsoft Network Load Balancing ( NLB ), or even use some kind of hardware load balancer if you have one available in your network. This way you could run the same NLS website on multiple web servers and know that your clients will still work properly in the event of a web server failure. Summary This article encourages you to use Windows Server 2012 R2 as the connectivity platform that brings your remote computers into the corporate network. We discussed DirectAccess and VPN in this article. We also saw how to configure DirectAccess and VPN, and how to secure DirectAccess using certificate authentication. Resources for Article: Further resources on this subject: Cross-premise Connectivity [article] Setting Up and Managing E-mails and Batch Processing [article] Upgrading from Previous Versions [article]
Read more
  • 0
  • 0
  • 1564
article-image-getting-your-own-video-and-feeds
Packt
06 Feb 2015
18 min read
Save for later

Getting Your Own Video and Feeds

Packt
06 Feb 2015
18 min read
"One server to satisfy them all" could have been the name of this article by David Lewin, the author of BeagleBone Media Center. We now have a great media server where we can share any media, but we would like to be more independent so that we can choose the functionalities the server can have. The goal of this article is to let you cross the bridge, where you are going to increase your knowledge by getting your hands dirty. After all, you want to build your own services, so why not create your own contents as well. (For more resources related to this topic, see here.) More specifically, here we will begin by building a webcam streaming service from scratch, and we will see how this can interact with what we have implemented previously in the server. We will also see how to set up a service to retrieve RSS feeds. We will discuss the services in the following sections: Installing and running MJPG-Streamer Detecting the hardware device and installing drivers and libraries for a webcam Configuring RSS feeds with Leed Detecting the hardware device and installing drivers and libraries for a webcam Even though today many webcams are provided with hardware encoding capabilities such as the Logitech HD Pro series, we will focus on those without this capability, as we want to have a low budget project. You will then learn how to reuse any webcam left somewhere in a box because it is not being used. At the end, you can then create a low cost video conference system as well. How to know your webcam As you plug in the webcam, the Linux kernel will detect it, so you can read every detail it's able to retrieve about the connected device. We are going to see two ways to retrieve the webcam we have plugged in: the easy one that is not complete and the harder one that is complete. "All magic comes with a price."                                                                                     –Rumpelstiltskin, Once Upon a Time Often, at a certain point in your installation, you have to choose between the easy or the hard way. Most of the time, powerful Linux commands or tools are not thought to be easy at first but after some experiments you'll discover that they really can make your life better. Let's start with the fast and easy way, which is lsusb : debian@arm:~$ lsusb Bus 001 Device 002: ID 046d:0802 Logitech, Inc. Webcam C200 Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub This just confirms that the webcam is running well and is seen correctly from the USB. Most of the time we want more details, because a hardware installation is not exactly as described in books or documentations, so you might encounter slight differences. This is why the second solution comes in. Among some of the advantages, you are able to know each step that has taken place when the USB device was discovered by the board and Linux, such as in a hardware scenario: debian@arm:~$ dmesg A UVC device (here, a Logitech C200) has been used to obtain these messages Most probably, you won't exactly have the same outputs, but they should be close enough so that you can interpret them easily when they are referred to: New USB device found: This is the main message. In case of any issue, we will check its presence elsewhere. This message indicates that this is a hardware error and not a software or configuration error that you need to investigate. idVendor and idProduct: This message indicates that the device has been detected. This information is interesting so you can check the constructor detail. Most recent webcams are compatible with the Linux USB Video Class (UVC), you can check yours at http://www.ideasonboard.org/uvc/#devices. Among all the messages, you should also look for the one that says Registered new interface driver interface because failing to find it can be a clue that Linux could detect the device but wasn't able to install it. The new device will be detected as /dev/video0. Nevertheless, at start, you can see your webcam as a different device name according to your BeagleBone configuration, for example, if a video capable cape is already plugged in. Setting up your webcam Now we know what is seen from the USB level. The next step is to use the crucial Video4Linux driver, which is like a Swiss army knife for anything related to video capture: debian@arm:~$ Install v4l-utils The primary use of this tool is to inquire about what the webcam can provide with some of its capabilities: debian@arm:~$ v4l2-ctl -–all There are four distinctive sections that let you know how your webcam will be used according to the current settings: Driver info (1) : This contains the following information: Name, vendor, and product IDs that we find in the system message The driver info (the kernel's version) Capabilities: the device is able to provide video streaming Video capture supported format(s) (2): This contains the following information: What resolution(s) are to be used. As this example uses an old webcam, there is not much to choose from but you can easily have a lot of choices with devices nowadays. The pixel format is all about how the data is encoded but more details can be retrieved about format capabilities (see the next paragraph). The remaining stuff is relevant only if you want to know in precise detail. Crop capabilities (3): This contains your current settings. Indeed, you can define the video crop window that will be used. If needed, use the crop settings: --set-crop-output=top=<x>,left=<y>,width=<w>,height=<h> Video input (4): This contains the following information: The input number. Here we have used 0, which is the one that we found previously. Its current status. The famous frames per second, which gives you a local ratio. This is not what you will obtain when you'll be using a server, as network latencies will downgrade this ratio value. You can grab capabilities for each parameter. For instance, if you want to see all the video formats the webcam can provide, type this command: debian@arm:~$ v4l2-ctl --list-formats Here, we see that we can also use MJPEG format directly provided by the cam. While this part is not mandatory, such a hardware tour is interesting because you know what you can do with your device. It is also a good habit to be able to retrieve diagnostics when the webcam shows some bad signs. If you would like to get more in depth knowledge about your device, install the uvcdynctrl package, which lets you retrieve all the formats and frame rates supported. Installing and running MJPG-Streamer Now that we have checked the chain from the hardware level up to the driver, we can install the software that will make use of Video4Linux for video streaming. Here comes MJPG-Streamer. This application aims to provide you with a JPEG stream on the network available for browsers and all video applications. Besides this, we are also interested in this solution as it's made for systems with less advanced CPU, so we can start MJPG-Streamer as a service. With this streamer, you can also use the built-hardware compression and even control webcams such as pan, tilt, rotations, zoom capabilities, and so on. Installing MJPG-Streamer Before installing MJPG-Streamer, we will install all the necessary dependencies: debian@arm:~$ install subversion libjpeg8-dev imagemagick Next, we will retrieve the code from the project: debian@arm:~$ svn checkout http://svn.code.sf.net/p/mjpg-streamer/code/ mjpg-streamer-code You can now build the executable from the sources you just downloaded by performing the following steps: Enter the following into the local directory you have downloaded: debian@arm:~$ cd mjpg-streamer-code/mjpg-streamer Then enter the following command: debian@beaglebone:~/mjpg-streamer-code/mjpg-streamer$ make When the compilation is complete, we end up with some new files. From this picture the new green files are produced from the compilation: there are the executables and some plugins as well. That's all that is needed, so the application is now considered ready. We can now try it out. Not so much to do after all, don't you think? Starting the application This section aims at getting you started quickly with MJPG-Streamer. At the end, we'll see how to start it as a service on boot. Before getting started, the server requires some plugins to be copied into the dedicated lib directory for this purpose: debian@beaglebone:~/mjpg-streamer-code/mjpg-streamer$ sudo cp input_uvc.so output_http.so /usr/lib The MJPG-Streamer application has to know the path where these files can be found, so we define the following environment variable: debian@beaglebone:~/mjpg-streamer-code/mjpg-streamer$ export LD_LIBRARY_PATH=/usr/ lib;$LD_LIBRARY_PATH Enough preparation! Time to start streaming: debian@beaglebone:~/mjpg-streamer-code/mjpg-streamer$./mjpg_streamer -i "input_uvc.so" -o "output_http.so -w www" As the script starts, the input parameters that will be taken into consideration are displayed. You can now identify this information, as they have been explained previously: The detected device from V4L2 The resolution that will be displayed, according to your settings Which port will be opened Some controls that depend on your camera capabilities (tilt, pan, and so on) If you need to change the port used by MJPG-Streamer, add -p xxxx at the end of the command, which is shown as follows: debian@beaglebone:~/mjpg-streamer-code/mjpg-streamer$ ./mjpg_streamer -i "input_uvc.so" -o "output_http.so -w www –p 1234" Let's add some security If you want to add some security, then you should set the credentials: debian@beaglebone:~/mjpg-streamer-code/mjpg-streamer$ ./mjpg-streamer -o "output_http.so -w ./www -c debian:temppwd" Credentials can always be stolen and used without your consent. The best way to ensure that your stream is confidential all along would be to encrypt it. So if you intend to use strong encryption for secured applications, the crypto-cape is worth taking a look at http://datko.net/2013/10/03/howto_crypto_beaglebone_black/. "I'm famous" – your first stream That's it. The webcam is made accessible to everyone across the network from BeagleBone; you can access the video from your browser and connect to http://192.168.0.15:8080/. You will then see the default welcome screen, bravo!: Your first contact with the MJPG-Server You might wonder how you would get informed about which port to use among those already assigned. Using our stream across the network Now that the webcam is available across the network, you have several options to handle this: You can use the direct flow available from the home page. On the left-hand side menu, just click on the stream tab. Using VLC, you can open the stream with the direct link available at http://192.168.0.15:8080/?action=stream.The VideoLAN menu tab is a M3U-playlist link generator that you can click on. This will generate a playlist file you can open thereafter. In this case, VLC is efficient, as you can transcode the webcam stream to any format you need. Although it's not mandatory, this solution is the most efficient, as it frees the BeagleBone's CPU so that your server can focus on providing services. Using MediaDrop, we can integrate this new stream in our shiny MediaDrop server, knowing that currently MediaDrop doesn't support direct local streams. You can create a new post with the related URL link in the message body, as shown in the following screenshot: Starting the streaming service automatically on boot In the beginning, we saw that MJPG-Streamer needs only one command line to be started. We can put it in a bash script, but servicing on boot is far better. For this, use a console text editor – nano or vim – and create a file dedicated to this service. Let's call it start_mjpgstreamer and add the following commands: #! /bin/sh # /etc/init.d/start_mjpgstreamer export LD_LIBRARY_PATH="/home/debian/mjpg-streamer/mjpg-streamer-code/ mjpg-streamer;$LD_LIBRARY_PATH" EXEC_PATH="/home/debian/mjpg-streamer/mjpg-streamer-code/mjpg-streamer" $EXEC_PATH/mjpg_streamer -i "input_uvc.so" -o "output_http.so -w EXEC_PATH /www" You can then use administrator rights to add it to the services: debian@arm:~$ sudo /etc/init.d/start_mjpgstreamer start On the next reboot, MJPG-Streamer will be started automatically. Exploring new capabilities to install For those about to explore, we salute you! Plugins Remember that at the beginning of this article, we began the demonstration with two plugins: debian@beaglebone:~/mjpg-streamer-code/mjpg-streamer$ ./mjpg_streamer -i "input_uvc.so" -o "output_http.so -w www" If we take a moment to look at these plugins, we will understand that the first plugin is responsible for handling the webcam directly from the driver. Simply ask for help and options as follows: debian@beaglebone:~/mjpg-streamer-code/mjpg-streamer$ ./mjpg_streamer --input "input_uvc.so --help" The second plugin is about the web server settings: The path to the directory contains the final web server HTML pages. This implies that you can modify the existing pages with a little effort or create new ones based on those provided. Force a special port to be used. Like I said previously, port use is dedicated for a server. You define here which will be the one for this service. You can discover many others by asking: debian@arm:~$ ./mjpg_streamer --output "output_http.so --help" Apart from input_uvc and output_http, you have other available plugins to play with. Let's take a look at the plugins directory. Another tool for the webcam The Mjpg_streamer project is dedicated for streaming over network, but it is not the only one. For instance, do you have any specific needs such as monitoring your house/son/cat/Jon Snow figurine? buuuuzzz: if you answered yes to the last one, you just defined yourself as a geek. Well, in that case the Motion project is for you; just install the motion package and start it with the default motion.conf configuration. You will then record videos and pictures of any moving object/person that will be detected. As MJPG-Streamer motion aims to be a low CPU consumer, it works very well on BeagleBone Black. Configuring RSS feeds with Leed Our server can handle videos, pictures, and music from any source and it would be cool to have another tool to retrieve news from some RSS providers. This can be done with Leed, a RSS project organized for servers. You can have a final result, as shown in the following screenshot: This project has a "quick and easy" installation spirit, so you can give it a try without harness. Leed (for Light Feed) allows you to you access RSS feeds from any browser, so no RSS reader application is needed, and every user in your network can read them as well. You install it on the server and feeds are automatically updated. Well, the truth behind the scenes is that a cron task does this for you. You will be guided to set some synchronisation after the installation. Creating the environment for Leed in three steps We already have Apache, MySQL, and PHP installed, and we need a few other prerequisites to run Leed: Create a database for Leed Download the project code and set permissions Install Leed itself Creating a database for Leed You will begin by opening a MySQL session: debian@arm:~$ mysql –u root –p What we need here is to have a dedicated Leed user with its database. This user will be connected using the following: create user 'debian_leed'@'localhost' IDENTIFIED BY 'temppwd'; create database leed_db; use leed_db; grant create, insert, update, select, delete on leed_db.* to debian_leed@localhost; exit Downloading the project code and setting permissions We prepared our server to have its environment ready for Leed, so after getting the latest version, we'll get it working with Apache by performing the following steps: From your home, retrieve the latest project's code. It will also create a dedicated directory: debian@arm:~$ git clone https://github.com/ldleman/Leed.git debian@arm:~$ ls mediadrop mjpg-streamer Leed music Now, we need to put this new directory where the Apache server can find it: debian@arm:~$ sudo mv Leed /var/www/ Change the permissions for the application: debian@arm:~$ chmod 777 /var/www/Leed/ -R Installing Leed When you go to the server address (http//192.168.0.15/leed/install.php), you'll get the following installation screen: We now need to fill in the database details that we previously defined and add the Administrator credentials as well. Now save and quit. Don't worry about the explanations, we'll discuss these settings thereafter. It's important that all items from the prerequisites list on the right are green. Otherwise, a warning message will be displayed about the wrong permissions settings, as shown in the following screenshot: After the configuration, the installation is complete: Leed is now ready for you. Setting up a cron job for feed updates If you want automatic updates for your feeds, you'll need to define a synchronization task with cron: Modify cron jobs: debian@arm:~$ sudo crontab –e Add the following line: 0 * * * * wget -q -O /var/www/leed/logsCron "http://192.168.0.15/Leed/action.php?action=synchronize Save it and your feeds will be refreshed every hour. Finally, some little cleanup: remove install.php for security matters: debian@arm:~$ rm /var/www/Leed/install.php Using Leed to add your RSS feed When you need to add some feeds from the Manage menu, in Feed Options (on the right- hand side) select Preferences and you just have to paste the RSS link and add it with the button: You might find it useful to organize your feeds into groups, as we did for movies in MediaDrop. The Rename button will serve to achieve this goal. For example, here a TV Shows category has been created, so every feed related to this type will be organized on the main screen. Some Leed preferences settings in a server environment You will be asked to choose between two synchronisation modes: Complete and Graduated. Complete: This isto be used in a usual computer, as it will update all your feeds in a row, which is a CPU consuming task Graduated: Look for the oldest 10 feeds and update them if required You also have the possibility of allowing anonymous people to read your feeds. Setting Allow anonymous readers to Yeswill let your guests access your feeds but not add any. Extending Leed with plugins If you want to extend Leed capabilities, you can use the Leed Market—as the author defined it—from Feed options in the Manage menu. There, you'll be directed to the Leed Market space. Installation is just a matter of downloading the ZIP file with all plugins: debian@arm:~/Leed$ wget  https://github.com/ldleman/Leed-market/archive/master.zip debian@arm:~/Leed$ sudo unzip master.zip Let's use the AdBlock plugin for this example: Copy the content of the AdBlock plugin directory where Leed can see it: debian@arm:~/Leed$ sudo cp –r Leed-market-master/adblock /var/www/Leed/plugins Connect yourself and set the plugin by navigating to Manage | Available Plugins and then activate adblock withEnable, as follows: In this article, we covered: Some words about the hardware How to know your webcam Configuring RSS feeds with Leed Summary In this article, we had some good experiments with the hardware part of the server "from the ground," to finally end by successfully setting up the webcam service on boot. We discovered hardware detection, a way to "talk" with our local webcam and thus to be able to see what happens when we plug a device in the BeagleBone. Through the topics, we also discovered video4linux to retrieve information about the device, and learned about configuring devices. Along the way, we encountered MJPG-Streamer. Finally, it's better to be on our own instead of being dependent on some GUI interfaces, where you always wonder where you need to click. Finally, our efforts have been rewarded, as we ended up with a web page we can use and modify according to our tastes. RSS news can also be provided by our server so that you can manage all your feeds in one place, read them anywhere, and even organize dedicated groups. Plenty of concepts have been seen for hardware and software. Then think of this article as a concrete example you can use and adapt to understand how Linux works. I hope you enjoyed this freedom of choice, as you drag ideas and drop them in your BeagleBone as services. We entered in the DIY area, showing you ways to explore further. You can argue, saying that we can choose the software but still use off the shelf commercial devices. Resources for Article: Further resources on this subject: Using PVR with Raspbmc [Article] Pulse width modulator [Article] Making the Unit Very Mobile - Controlling Legged Movement [Article]
Read more
  • 0
  • 0
  • 4608

article-image-setting-our-development-environment-and-creating-game-activity
Packt
06 Feb 2015
17 min read
Save for later

Setting up our development environment and creating a game activity

Packt
06 Feb 2015
17 min read
In this article by John Horton, author of the book Learning Java by Building Android Games, we will learn how to set up our development environment by installing JDK and Android Studio. We will also learn how to create a new game activity and layout the same on a game screen UI. (For more resources related to this topic, see here.) Setting up our development environment The first thing we need to do is prepare our PC to develop for Android using Java. Fortunately, this is made quite simple for us. The next two tutorials have Windows-specific instructions and screenshots. However, it shouldn't be too difficult to vary the steps slightly to suit Mac or Linux. All we need to do is: Install a software package called the Java Development Kit (JDK), which allows us to develop in Java. Install Android Studio, a program designed to make Android development fast and easy. Android Studio uses the JDK and some other Android-specific tools that automatically get installed when we install Android Studio. Installing the JDK The first thing we need to do is get the latest version of the JDK. To complete this guide, perform the following steps: You need to be on the Java website, so visit http://www.oracle.com/technetwork/java/javase/downloads/index.html. Find the three buttons shown in the following screenshot and click on the one that says JDK (highlighted). They are on the right-hand side of the web page. Click on the DOWNLOAD button under the JDK option: You will be taken to a page that has multiple options to download the JDK. In the Product/File description column, you need to click on the option that matches your operating system. Windows, Mac, Linux and some other less common options are all listed. A common question here is, "do I have 32- or 64-bit windows?". To find out, right-click on your My Computer (This PC on Windows 8) icon, click on the Properties option, and look under the System heading in the System type entry, as shown in the following screenshot: Click on the somewhat hidden Accept License Agreement checkbox: Now click on the download option for your OS and system type as previously determined. Wait for the download to finish. In your Downloads folder, double-click on the file you just downloaded. The latest version at time of writing this for a 64-bit Windows PC was jdk-8u5-windows-x64. If you are using Mac/Linux or have a 32-bit OS, your filename will vary accordingly. In the first of several install dialogs, click on the Next button and you will see the next dialog box: Accept the defaults shown in the previous screenshot by clicking on Next. In the next dialog box, you can accept the default install location by clicking on Next. Next is the last dialog of the Java installer. Click on Close. The JDK is now installed. Next we will make sure that Android Studio is able to use the JDK. Right-click on your My Computer (This PC on Windows 8) icon and navigate to Properties | Advanced system settings | Environment variables | New (under System variables, not under User variables). Now you can see the New System Variable dialog, as shown in the following screenshot: Type JAVA_HOME for Variable name and enter C:Program FilesJavajdk1.8.0_05 for the Variable value field. If you installed the JDK somewhere else, then the file path you enter in the Variable value: field will need to point to wherever you put it. Your exact file path will likely have a different ending to match the latest version of Java at the time you downloaded it. Click on OK to save your new settings. Now click on OK again to clear the Advanced system settings dialog. Now we have the JDK installed on our PC. We are about half way towards starting to learn Java programming, but we need a friendly way to interact with the JDK and to help us make Android games in Java. Android Studio We learned that Android Studio is a tool that simplifies Android development and uses the JDK to allow us to write and build Java programs. There are other tools you can use instead of Android Studio. There are pros and cons in them all. For example, another extremely popular option is Eclipse. And as with so many things in programming, a strong argument can be made as to why you should use Eclipse instead of Android Studio. I use both, but what I hope you will love about Android Studio are the following elements: It is a very neat and, despite still being under development, a very refined and clean interface. It is much easier to get started compared to Eclipse because several Android tools that would otherwise need to be installed separately are already included in the package. Android Studio is being developed by Google, based on another product called IntelliJ IDEA. There is a chance it will be the standard way to develop Android in the not-too-distant future. If you want to use Eclipse, that's fine. However, some the keyboard shortcuts and user interface buttons will obviously be different. If you do not have Eclipse installed already and have no prior experience with Eclipse, then I even more strongly recommend you to go ahead with Android Studio. Installing Android Studio So without any delay, let's get Android Studio installed and then we can begin our first game project. To do this, let's visit https://developer.android.com/sdk/installing/studio.html. Click on the button labeled Download Android Studio to start the Android studio download. This will take you to another web page with a very similar-looking button to the one you just clicked on. Accept the license by checking in the checkbox, commence the download by clicking on the button labeled Download Android Studio for Windows, and wait for the download to complete. The exact text on the button will probably vary depending on the current latest version. In the folder in which you just downloaded Android Studio, right-click on the android-studio-bundle-135.12465-windows.exe file and click on Run as administrator. The end of your filename will vary depending upon the version of Android Studio and your operating system. When asked if you want to Allow the following program from an unknown publisher to make changes to your computer, click on Yes. On the next screen, click on Next. On the screen shown in the following screenshot, you can choose which users of your PC can use Android Studio. Choose whatever is right for you as all options will work, and then click on Next: In the next dialog, leave the default settings and then click on Next. Then on the Choose start menu folder dialog box, leave the defaults and click on Install. On the Installation complete dialog, click on Finish to run Android Studio for the first time. The next dialog is for users who have already used Android Studio, so assuming you are a first time user, select the I do not have a previous version of Android Studio or I do not want to import my settings checkbox, and then click on OK: That was the last piece of software we needed. Math game – asking a question Now that we have all that knowledge under our belts, we can use it to improve our math game. First, we will create a new Android activity to be the actual game screen as opposed to the start menu screen. We will then use the UI designer to lay out a simple game screen so that we can use our Java skills with variables, types, declaration, initialization, operators, and expressions to make our math game generate a question for the player. We can then link the start menu and game screens together with a push button. Creating the new game activity We will first need to create a new Java file for the game activity code and a related layout file to hold the game activity UI. Run Android Studio and select your Math Game Chapter 2 project. It might have been opened by default. Now we will create the new Android activity that will contain the actual game screen, which will run when the player taps the Play button on our main menu screen. To create a new activity, we now need another layout file and another Java file. Fortunately Android Studio will help us do this. To get started with creating all the files we need for a new activity, right-click on the src folder in the Project Explorer and then go to New | Activity. Now click on Blank Activity and then on Next. We now need to tell Android Studio a little bit about our new activity by entering information in the above dialog box. Change the Activity Name field to GameActivity. Notice how the Layout Name field is automatically changed for us to activity_game and the Title field is automatically changed to GameActivity. Click on Finish. Android Studio has created two files for us and has also registered our new activity in a manifest file, so we don't need to concern ourselves with it. If you look at the tabs at the top of the editor window, you will see that GameActivity.java has been opened up ready for us to edit, as shown in the following screenshot: Ensure that GameActivity.java is active in the editor window by clicking on the GameActivity.java tab shown previously. Here, we can see the code that is unnecessary. If we remove it, then it will make our working environment simpler and cleaner. We will simply use the code from MainActivity.java as a template for GameActivity.java. We can then make some minor changes. Click on the MainActivity.java tab in the editor window. Highlight all of the code in the editor window using Ctrl + A on the keyboard. Now copy all of the code in the editor window using the Ctrl + C on the keyboard. Now click on the GameActivity.java tab. Highlight all of the code in the editor window using Ctrl + A on the keyboard. Now paste the copied code and overwrite the currently highlighted code using Ctrl + V on the keyboard. Notice that there is an error in our code denoted by the red underlining as shown in the following screenshot. This is because we pasted the code referring to MainActivity in our file that is called GameActivity. Simply change the text MainActivity to GameActivity and the error will disappear. Take a moment to see if you can work out what other minor change is necessary, before I tell you. Remember that setContentView loads our UI design. Well what we need to do is change setContentView to load the new design (that we will build next) instead of the home screen design. Change setContentView(R.layout.activity_main); to setContentView(R.layout.activity_game);. Save your work and we are ready to move on. Note the Project Explorer where Android Studio puts the two new files it created for us. I have highlighted two folders in the next screenshot. In future, I will simply refer to them as our java code folder or layout files folder. You might wonder why we didn't simply copy and paste the MainActivity.java file to begin with and saved going through the process of creating a new activity? The reason is that Android Studio does things behind the scenes. Firstly, it makes the layout template for us. It also registers the new activity for use through a file we will see later, called AndroidManifest.xml. This is necessary for the new activity to be able to work in the first place. All things considered, the way we did it is probably the quickest. The code at this stage is exactly the same as the code for the home menu screen. We state the package name and import some useful classes provided by Android: package com.packtpub.mathgamechapter3a.mathgamechapter3a;   import android.app.Activity; import android.os.Bundle; We create a new activity, this time called GameActivity: public class GameActivity extends Activity { Then we override the onCreate method and use the setContentView method to set our UI design as the contents of the player's screen. Currently, however, this UI is empty: super.onCreate(savedInstanceState);setContentView(R.layout.activity_main); We can now think about the layout of our actual game screen. Laying out the game screen UI As we know, our math game will ask questions and offer the player some multiple choices to choose answers from. There are lots of extra features we could add, such as difficulty levels, high scores, and much more. But for now, let's just stick to asking a simple, predefined question and offering a choice of three predefined possible answers. Keeping the UI design to the bare minimum suggests a layout. Our target UI will look somewhat like this: The layout is hopefully self-explanatory, but let's ensure that we are really clear; when we come to building this layout in Android Studio, the section in the mock-up that displays 2 x 2 is the question and will be made up of three text views (both numbers, and the = sign is also a separate view). Finally, the three options for the answer are made up of Button layout elements. This time, as we are going to be controlling them using our Java code, there are a few extra things we need to do to them. So let's go through it step by step: Open the file that will hold our game UI in the editor window. Do this by double-clicking on activity_game.xml. This is located in our UI layout folder, which can be found in the project explorer. Delete the Hello World TextView, as it is not required. Find the Large Text element on the palette. It can be found under the Widgets section. Drag three elements onto the UI design area and arrange them near the top of the design as shown in the next screenshot. It does not have to be exact; just ensure that they are in a row and not overlapping, as shown in the following screenshot: Notice in the Component Tree window that each of the three TextViews has been assigned a name automatically by Android Studio. They are textView , textView2, and textView3: Android Studio refers to these element names as an id. This is an important concept that we will be making use of. So to confirm this, select any one of the textViews by clicking on its name (id), either in the component tree as shown in the preceding screenshot or directly on it in the UI designer shown previously. Now look at the Properties window and find the id property. You might need to scroll a little to do this: Notice that the value for the id property is textView. It is this id that we will use to interact with our UI from our Java code. So we want to change all the IDs of our TextViews to something useful and easy to remember. If you look back at our design, you will see that the UI element with the textView id is going to hold the number for the first part of our math question. So change the id to textPartA. Notice the lowercase t in text, the uppercase P in Part, and the uppercase A. You can use any combination of cases and you can actually name the IDs anything you like. But just as with naming conventions with Java variables, sticking to conventions here will make things less error-prone as our program gets more complicated. Now select textView2 and change id to textOperator. Select the element currently with id textView3 and change it to textPartB. This TextView will hold the later part of our question. Now add another Large Text from the palette. Place it after the row of the three TextViews that we have just been editing. This Large Text will simply hold our equals to sign and there is no plan to ever change it. So we don't need to interact with it in our Java code. We don't even need to concern ourselves with changing the ID or knowing what it is. If this situation changed, we could always come back at a later time and edit its ID. However, this new TextView currently displays Large Text and we want it to display an equals to sign. So in the Properties window, find the text property and enter the value =. We have changed the text property, and you might also like to change the text property for textPartA, textPartB, and textOperator. This is not absolutely essential because we will soon see how we can change it via our Java code; however, if we change the text property to something more appropriate, then our UI designer will look more like it will when the game runs on a real device. So change the text property of textPartA to 2, textPartB to 2, and textOperator to x. Your UI design and Component tree should now look like this: For the buttons to contain our multiple choice answers, drag three buttons in a row, below the = sign. Line them up neatly like our target design. Now, just as we did for the TextViews, find the id properties of each button, and from left to right, change the id properties to buttonChoice1, buttonChoice2, and buttonChoice3. Why not enter some arbitrary numbers for the text property of each button so that the designer more accurately reflects what our game will look like, just as we did for our other TextViews? Again, this is not absolutely essential as our Java code will control the button appearance. We are now actually ready to move on. But you probably agree that the UI elements look a little lost. It would look better if the buttons and text were bigger. All we need to do is adjust the textSize property for each TextView and for each Button. Then, we just need to find the textSize property for each element and enter a number with the sp syntax. If you want your design to look just like our target design from earlier, enter 70sp for each of the TextView textSize properties and 40sp for each of the Buttons textSize properties. When you run the game on your real device, you might want to come back and adjust the sizes up or down a bit. But we have a bit more to do before we can actually try out our game. Save the project and then we can move on. As before, we have built our UI. This time, however, we have given all the important parts of our UI a unique, useful, and easy to identify ID. As we will see we are now able to communicate with our UI through our Java code. Summary In this article, we learned how to set up our development environment by installing JDK and Android Studio. In addition to this, we also learned how to create a new game activity and layout the same on a game screen UI. Resources for Article: Further resources on this subject: Sound Recorder for Android [article] Reversing Android Applications [article] 3D Modeling [article]
Read more
  • 0
  • 0
  • 2078
Modal Close icon
Modal Close icon