At first we need to declare an instance of Hazelcast within the Spring context using default spring bean namespace. To integrate Hazelcast with Spring we need either hazelcast-spring.jar or hazelcast-all.jar in the classpath. We need Spring version 2.5 or greater for hazelcast integration. Since hazelcast version 1.9.1, we have hazelcast namespace to configure in the Spring context file. Namespace (hz) given below. After configuring the hazelcast instance in spring context file, we need to create hazelcast.xml file which will be placed in the classpath. Sample file given below. dev dev http://localhost:8080/mancenter-3.3.3 5701 0 224.2.2.3 54327 127.0.0.1 BINARY 1 1 7200 600 LRU 0 25 100 com.hazelcast.map.merge.PassThroughMergePolicy Here we declared a hazelcast map called ‘CustomerMap’. This map can be accessed in the application and can be used to store key value pairs. Some properties of hazelcast map are given below. Backup-count - Number of backups. If 1 is set as the backup-count for example, then all entries of the map will be copied to another JVM for fail-safety. 0 means no backup Time-to-live-seconds - Maximum number of seconds for each entry to stay in the map. Entries that are older than time-to-live-seconds and not updated for time-to-live-seconds will get automatically evicted from the map Max-idle-seconds - Maximum number of seconds for each entry to stay idle in the map. Eviction-policy - Different eviction policies are available: NONE (no eviction), LRU (Least Recently Used), LFU (Least Frequently Used). NONE is the default. Max-size - Maximum size of the map. When max size is reached,map is evicted based on the policy defined. Any integer between 0 and Integer.MAX_VALUE. 0 means Integer.MAX_VALUE. Default is 0. Eviction-percentage - When max. size is reached, specified percentage of the map will be evicted. Any integer between 0 and 100. If 25 is set for example, 25% of the entries will get evicted. Merge-policy – There are different merge policies. PassThroughMergePolicy - Entry will be added if there is no existing entry for the key. HigherHitsMapMergePolicy - entry with the higher hits wins. LatestUpdateMapMergePolicy - entry with the latest update wins. Min-eviction-check-millis - Minimum time in milliseconds which should pass before checking if a partition of this map is evictable or not. Default value is 100 millis Accessing Hazelcast Map in the application At first we need to get reference to the hazelcast instance we declared in the spring context file. Here we use autowiring to get the hazelcast instance. After that hazelcast instance is used to create an ‘IMap’. Different types of maps are available like IMap, ConcrrentMap, MultiMap etc. IMap is Concurrent, distributed, observable and queryable map. IMap does not allow nulls to be used as keys or values. Code snippet given below. import org.springframework.beans.factory.annotation.Autowired; import com.hazelcast.core.HazelcastInstance; import com.hazelcast.core.IMap; public class HazelcastTest { @Autowired private HazelcastInstance hazelcastInstance; public void addCustomers() throws Exception { IMap map = hazelcastInstance.getMap("CustomerMap"); map.put(1001, "Tom"); map.put(1002, "Jim"); map.put(1003, "Joe"); } } Above explaination will help to configure hazelcast with Spring and use hazelcast IMap to store data.

Dropwizard is a bunch of superb frameworks glued together to provide a fast way of building web applications including REST APIs. We'll talk about great frameworks which are parts of Dropwizard as we go over our examples. The full code for the examples can be obtained here . The simplest way to create a Dropwizard project is to use the Maven archetype called java-simple which is a part of Dropwizard. This can be accomplished either via the command line or using your favorite IDE. Let's stick to the former approach. The command below is formatted for clarity but it should be pasted as a single line to the terminal. mvn archetype:generate -DgroupId=com.javaeeeee -DartifactId=DWGettingStarted -Dname=DWGettingStarted -Dpackage=com.javaeeeee.dwstart -DarchetypeGroupId=io.dropwizard.archetypes -DarchetypeArtifactId=java-simple -DinteractiveMode=false After the project was created it can be modified using an editor or opened in an IDE. All the three IntelliJ, Eclipse and Netbeans have Maven support. An important parameter to note in the pom.xml file of the newly created project is dropwizard.version, which at the time of writing was automatically set to 0.8.0-rc2-SNAPSHOT and the latest stable version was 0.7.1. We'll discuss later the important consequences of the version for project development. The directory structure of the project is shown below. There are multiple sub-packages and two classes that were created for us in a package com.javaeeeee.dwstart. Now let's create a simple resource class that produces a string “Hello world!”, start the application and test the results using both browser and cURL. The snippet below shows the resource class. @Path("/hello") public class HelloResource { @GET @Produces(MediaType.TEXT_PLAIN) public String getGreeting() { return "Hello world!"; } } There is a special package, com.javaeeeee.dwstart.recources for placing resource classes or an appropriate folder can be found from the screenshot above if you use a text editor and not an IDE. One of the main tenets of the REST architecture style is that each resource is assigned a URL. In our case the URL of our resource would be http://localhost:8080/hello, that is we access it from our local machine, the default port which is used by Dropwizard is 8080 and the final part of the URL is enshrined in @Path annotation in our class, in other words, the aforementioned annotation helps to describe the URL used to access a resource. Moving on to method getGreeting() it can be seen that it is a simple method which returns the desired string “Hello world!”, although it is marked with two annotations: first, @GET prescribes that this method is called when the aforementioned URL is accessed using HTTP GET method, second, annotation @Produces specifies what media types can be produced when the method is accessed by the client, such as browser, cURL or some other program. In our case it is plain text. These annotations are part of Java API for RESTful Web Services (JAX-RS), and its reference implementation Jersey is the cornerstone of Dropwizard. Other annotations include @POST for HTTP POST method, @DELETE for DELETE and so on and media types include MediaType.APPLICATION_JSON and not so popular MediaType.APPLICATION_XML among others. Now, to see the result of our coding, we should do some configuration. Let's open one of the created by Maven files, DWGettingStartedApplication.java, and register our resource. public void run(final DWGettingStartedConfiguration configuration, final Environment environment) { environment.jersey().register(new HelloResource()); } We added a single line environment.jersey().register(new HelloResource()); to help Dropwizard find our resource class. Now we have to create a jar-file which contains an embedded Jersey web server to serve the incoming requests, as well as all the necessary libraries. Dropwizard applications are packaged as executable jar-files rather than war-files which are deployed to an application server. The kind of jar files used by Dropwizard applications are also called “fat” because they include all the .class files necessary to run the application as this leads to a situation that the versions of all libraries are the same in both development and production environments which leads to less problems. One shouldn't worry about creating such jar-files as Maven creates them for us using instructions in the generated pom.xml file of the project. The jar-file can be created using command-line command issued from the project's folder mvn package or using an IDE. After that we should start the application, which could be accomplished either from the command line by issuing a command java -jar target/DWGettingStarted-1.0-SNAPSHOT.jar server or from the IDE which can be instructed to pass a “server” argument to the executable jar file. To stop the application it is sufficient to press Ctrl+C in the terminal. Now it is time to check if it works. The simplest way to achieve this is to navigate your browser to the URL mentioned earlier. You should see the greeting string in the main window of a browser. If you cannot see the greeting, there is probably some problems and the program’s output is the place to look for clues for what went wrong. The same greeting can be seen in the terminal window with help of cURL. curl -w "\n" localhost:8080/hello Protocol HTTP and GET method are defaults, so it is not necessary that they be included explicitly in the command. As the result a message "Hello world!" without quotes should be printed. The w-option is used to add a trailing newline to prettify the output. It is a good idea to start using tests as early as possible, so let's write a test that checks the work of our resource class using in-memory Jersey. We'll create the test in test packages and as we are testing a resource an appropriate package to place our test is com.javaeeeee.dwstart.resources (or src/test/com/javaeeeee/dwstart/resources folder). It is necessary that the following dependency be added to the pom-file of the project. (It works without explicit jUnit dependency.) io.dropwizard dropwizard-testing ${dropwizard.version} test An example test for our greeting resource is shown below. public class HelloResourceTest { @Rule public ResourceTestRule resource = ResourceTestRule.builder() .addResource(new HelloResource()).build(); @Test public void testGetGreeting() { String expected = "Hello world!"; //Obtain client from @Rule. Client client = resource.client(); //Get WebTarget from client using URI of root resource. WebTarget helloTarget = client.target("http://localhost:8080/hello"); //To invoke response we use Invocation.Builder //and specify the media type of representation asked from resource. Invocation.Builder builder = helloTarget.request(MediaType.TEXT_PLAIN); //Obtain response. Response response = builder.get(); //Do assertions. assertEquals(Response.Status.OK, response.getStatusInfo()); String actual = response.readEntity(String.class); assertEquals(expected, actual); } } The annotation @Rule is from jUnit realm and could be placed on public non-static fields and methods that return a value. This annotation may perform some initial setup and clean-up like @Before and @After methods, but it is more powerful as it allows to share functionality between classes and even projects. This rule is needed to allow us to access our resources from the code of the test using in-memory Jersey. At this point of time we should pause and discuss the version of our product. There is a pom.xml file in the dropwizard folder on the GitHub, which can readily be accessed by cloning the project, and this file specifies the version of constituent products among other things. It can be seen from it that the version of Jersey in the snapshot version of Dropwizard is 2.13 and in the version 0.7.1 it is 1.18.1. The version of Jersey influences the code of the tests as 1.x and 2.x versions of Jersey have different client APIs, that is the code to access the service programmatically from Java depends on the version of Jersey framework. Let's stick to the latest 2.x version of Jersey as it is liable to be included in following 0.8.x releases of Dropwizard and if you are interested how to test using the older version of Jersey client, please check the project repository , there is a special branch v0.7.x. Another observation concerning testing is that Fest matchers were superseded by AssertJ counterparts. First, we obtain an instance of Jersey client to make requests to our service. Second, we create an instance of a class that implements WebTarget interface for the resource using its URL. After that we use Invocation.Builder that helps to build request and send it to the server. The Invocation.Builder interface extends SynchInvoker interface which defines a bunch of get(...) methods used to invoke HTTP GET method synchronously. Finally, we check the results, namely that the HTTP status code was 200 and the desired string was returned. The same result could be obtained by chaining the methods as in the snippet below. actual = resource.client() .target("http://localhost:8080/hello") .request(MediaType.TEXT_PLAIN) .get(String.class); assertEquals(expected, actual); The replacement of the two assertions by a single is possible due to the fact that the get(...) method we used throws a WebApplicationException if the HTTP status code is not in the 2xx range, in other words if there is a problem. Now let's improve our resource to take a name and return a bespoke greeting. It could be done in a couple of ways using Jersey's annotations. The first is to use so-called path parameters, that is you pass the name using a URL like http://localhost:8080/hello/path_param/your_name. The application should return “Hello your_name”. Let's see a code snippet. @Path("/path_param/{name}") @GET @Produces(MediaType.TEXT_PLAIN) public String getNamedGreeting(@PathParam(value = "name") String name) { return "Hello " + name; } Now we see a @Path annotation on our method. It adds its content to those produced by class level annotation as seen from the URL. Methods marked with this annotation are called sub-resource methods. The name of the parameter, name, is in curly braces. Something interesting happens inside the parenthesis of the getNamedGreeting method. There is an annotation which prescribes to extract the content of the url after “/path_param/” to the method as a value. It should be noted that if you find yourself marking each method with the same annotation @Produces(MediaType.TEXT_PLAIN), the latter can be removed from methods and used to mark the class, then all the methods produce the desired media types, although this can be overruled by an annotations with a different media type for some of the methods. The second way to pass a name is to use query parameters and the URL could look like http://localhost:8080/hello/query_param?name=your_name. The output should be “Hello your_name” without quotes. The snippet is shown below. @Path("/query_param") @GET @Produces(MediaType.TEXT_PLAIN) public String getNamedStringWithParam(@DefaultValue("world") @QueryParam("name") String name) { return "Hello " + name; } Once again, the @Path annotation adds something to our URL, then the default value of the parameter is set in case there is no question mark and following it symbols in the URL using @DefaultValue annotation. In other words, if the parameter is omitted in the URL and it looks like http://localhost:8080/hello/query_param, a phrase “Hello world” is printed. The last annotation injects a parameter from the URL to the method's parameter. By the way, all these cases can be tested as was shown above. To test a method using query parameters one can use a queryParam(...) method of WebTarget interface. There is a couple of points to pay attention to. Firstly, there are other @*Param annotations available and secondly, the @DefaultValue annotation will not work with path parameters and to process the absence of the parameter a special sub-resource method should be introduced. You were patient enough to bear all this plain text staff, but it is not what REST APIs are about. Let's create another sub-resource method which is preprogrammed to return JSON representation. The first step is to create a representation class, which should be placed to com.javaeeeee.dwstart.core package. Let's limit ourselves to extremely simple example which is shown below. public class Greeting { @JsonProperty private String greeting; public Greeting() { } public Greeting(String greeting) { this.greeting = greeting; } public String getGreeting() { return greeting; } } The class above relies on one more important part of the Dropwizard stack namely Jackson , which can turn Java objects to JSON and vice versa. A @JsonProperty annotation is used to do the job. The sub-resource method looks like pretty much the same as our previously discussed methods. @Path("/hello_json") @GET @Produces(MediaType.APPLICATION_JSON) public Greeting getJSONGreeting() { return new Greeting("Hello world!"); } The difference is that we changed the media type and the method returns an instance of the greeting class. If you navigate your browser to to a URL http://localhost:8080/hello/hello_json or use cURL, you should see a JSON representation of the object {"greeting":"Hello world!"}. What if we remove the @Path annotation from the aforementioned method altogether? Well, if you try to access the localhost:8080/hello resource you'll see the text representation. How one could access the JSON-producing method? There is a special HTTP header called Accept which is used to instruct the server what representation to return, the process called content negotiation. The Jersey inside the Dropwizard will choose for us what method to use based on the content of the Accept header. Let's try cURL to do this. curl -w "\n" -H 'Accept: application/json' localhost:8080/hello/ The H-option is used to send headers and we instructed cURL to ask the JSON response. Other possible types to try could include text/plain and application/xml. The former should return the plain-text representation and the letter should result in error as there is no method in our resource class that can produce the latter media type response. Another way to engage in content negotiation is to use a REST client for your browser. There are several clients for each of major browsers but we will use Postman extension for Chrome browser. There are special fields to input headers as the screenshot below shows. One can enter the resource URL, headers and press Send button and the response will show up in the lower part of the window. To sum up we created a simple REST-like API using Dropwizard, which can greet its users. We learned how to create a Dropwizard project using Maven archetype and created a simple resource class to accomplish the task of greeting and a representation class to produce JSON response. An important ingredient of REST architecture style, HATEOAS, was omitted for now for simplicity reasons. Also a problem of testing resource classes was touched as well as the ways to access resources both from browser and command line. There are some other ideas for creating sub-resources such as returning the current date and adding two numbers using query parameters which can be easily implemented using the information presented in this article. References Dropwizard on GitHub Maven Archetypes Dropwizard Getting Started JAX-RS Resources and Subresources Dropwizard Testing Jersey Client API 2.x Jersey Client API 1.x Jackson annotations

We love Git Flow. It’s awesome to have a standard release process where everybody is using the same terminology and tooling. It’s also great to have out-of-the-box answers to the same questions that get asked at the start of every project. For example: “How are we going to develop features in isolation?” “How are we going to separate release candidates from ongoing development?” “How are we going to deal with hotfixes?” Now it’s enough to just say ‘We use Git Flow’, and everybody’s on the same page. Well, mostly. Whilst Git Flow is terrific for managing features and separating release candidates from ongoing development, things get a little hazier when it comes time to actually release to production. This is because of a mismatch between the way that Git Flow works and another practice that is common in large development projects: immutable build artifacts. Immutable what? On most enterprise projects I work on these days, it’s considered a pretty good idea to progress exactly the same build artifact through testing, pre-production and into production. It doesn’t matter whether it’s a JAR file, WAR file, tar.gz file, or something more exotic – the key point is that you build it once, then deploy the same thing to each downstream environment. If you find a bug during this journey, you fix the bug, build a whole new artifact, and start the process again. In the absence of any established terminology, let’s just call these things immutable build artifacts. (Note that you’ll still need a separate, external file for those things that have to be different between environments, but by keeping the amount of stuff in that file to an absolute minimum, you’ll minimise your potential exposure to variances between environments.) Without immutable build artifacts, many development managers will start to sweat nervously. This is usually because at some stage in their careers they’ve been up at 2am in the morning prising apart build artifacts and trying to understand what changed between the pre-prod build (which worked just fine) and the current prod build (which is inexplicably failing). A bunch of things can cause this sort of problem. For example, it could be that somebody managed to sneak some change in between the two builds, or that some downstream build dependency changed between the two builds, or even that somebody tweaked the build process between the two builds. ‘That’ll never happen to me’, I hear you say, ‘if everybody follows our development methodology correctly’. And therein lies the problem: it’s difficult to absolutely guarantee that a developer won’t at the last minute do something stupid, like slip something into the wrong branch, upload an incorrectly-versioned downstream dependency, or mess with your buildbox configuration. My point is this: why open yourself up to the risk at all, when having a single artifact will eliminate a whole class of potential defects? Now for the actual problem A while back, I was working on a project using both Git Flow and immutable builds. A junior developer came to me looking confused. He was trying to understand when it would be OK for him to finish the current Git Flow release. The testers had just signed off on the current build. But if he finished the release, Git Flow was going to merge the release branch into master – and strictly speaking, he should create a new build from that. But if he created a new build, the testers were going to want to test it. If they found a problem, the fix was going to have to happen in a new release branch. But then when he closed that release, he was going to have to do a new build from master, which the testers would want to test again, right? And then, if they found a bug in that… I admired his highly conscientious approach to his work, but worried that his mind was going to disintegrate as it spun around this build-management paradox. I also had to acknowledge that he had stumbled head first into something that I had been wilfully ignoring in the hope that it would resolve itself; namely, the fundamental mismatch between Git Flow and the concept of immutable builds. Put simply, Git Flow works on the premise that production builds will only be done from the master branch. In contrast, immutable builds work on the premise that production builds will be done from either a release branch or a hotfix branch. There is no perfect way to work around this mismatch. The pragmatic solution Put bluntly, if you want truly immutable builds, then you should do them from the release or hotfix branch rather than master. However, because this runs contrary to how Git Flow works, there are a couple of important consequences to keep in mind. 1. Concurrent hotfixes and releases require special handling If you start and then finish a hotfix whilst a release branch is in process, then that hotfix’s changes won’t automatically be brought into the release branch by Git Flow. This means that a subsequent build from the release branch which goes into production won’t include the hotfix changes. Here’s a diagram illustrating the problem: Note that the inverse problem would apply if you tried to start a release whilst a hotfix was in progress. Thankfully, Git Flow will not let you have two release branches in progress at the same time, so we don’t have to worry about that particular possibility. (If you’re wondering how Git Flow avoids these scenarios in its regular usage, remember that it always merges back into master when a release or hotfix is finished, and that you’re supposed to always build from master. Consequently, if you’re building from master post-release or post-hotfix, you’ll always be getting the latest changes into the build.) The problem of concurrent hotfixes and releases can be solved by merging the hotfix branch into the release branch just before the hotfix branch gets finished (or vice-versa if you started a release whilst a hotfix was in progress). Here’s what it looks like: However, you will need to remember to do this merge manually because Git Flow won’t do it for you. 2. Version tags will be incorrect by default When you finish a release or hotfix, Git Flow merges the corresponding branch back into master, and then tags the resultant merge commit on master with the version number. However, because your immutable artifact will have been built from the release/hotfix branch, the commit SHA for the version tag in Git will be different from the SHA that the artifact was actually built from. Continuing on from our previous example: You may or may not care about this, for a number of reasons. Firstly, from the perspective of Git Flow, the merge commit on master should never have any changes in it. The only scenario that might lead to it having changes is if you have run concurrent hotfix and release branches (as described in the previous section) and forgotten to merge them prior to finishing. And you’llnever forget to do that now will you? :) Secondly, in the likely event that you are using some sort of continuous integration server to produce your builds, that server can probably associate its own number with each build, and stamp the resultant artifact with that number. The CI server will probably also have recorded the SHA that each build was done from. So from the artifact you could probably work backwards to the SHA that was used to produce it. If you’re nevertheless wary of the confusion that this backtracking process might introduce when trying to debug a production issue at 2am in the morning, you might still insist on having correct version tags. In that case the best thing I can think of is to manually create the tag on the release/hotfix branch yourself before finishing it and then, when finishing the release, run git flow [release/hotfix] finish with the -n option to stop it from trying to create the tag again (which would fail because the tag now already exists). I’m on a roll with my diagrams, so what the heck, let’s do one more: Wrapping Up On balance, I think the benefits of having immutable builds outweigh the costs of deviating from the Git Flow way of doing things. However, you do have to be aware of a couple of problematic scenarios. We’ll be experimenting in coming months with using Git Flow with our own immutable builds, and will let you know if anything else weird pops up. Who knows, perhaps we’ll even end up with our own version of Git Flow. Either way, I’d be interested in hearing from anybody else who has had the same problem.

Bulk insertion of data into database table is a big overhead in application development.

The great thing with open source tools like Eclipse and GNU (gcc, gdb) is that there is a wealth of excellent tools: one thing I had in mind to explore for a while is how to generate code coverage of my embedded application. Yes, GNU and Eclipse come with code profiling and code coverage tools, all for free! The only downside seems to be that these tools seem to be rarely used for embedded targets. Maybe that knowledge is not widely available? So here is my attempt to change this :-). Or: How cool is it to see in Eclipse how many times a line in my sources has been executed? Line Coverage in Eclipse And best of all, it does not stop here…. Coverage with Eclipse To see how much percentage of my files and functions are covered? gcov in Eclipse Or even to show the data with charts? Coverage Bar Graph View Outline In this tutorial I’m using a Freescale FRDM-K64F board: this board has ARM Cortex-M4F on it, with 1 MByte FLASH and 256 KByte of RAM. The approach used in this tutorial can be used with any embedded target, as long there is enough RAM to store the coverage data on the target. I’m using Eclipse Kepler with the ARM Launchpad GNU tools (q3 2014 release), but with small modifications any Eclipse version or GNU toolchain could be used. To generate the Code Coverage information, I’m using gcov. Freescale FRDM-K64F Board Generating Code Coverage Information with gcov gcov is an open source program which can generate code coverage information. It tells me how often each line of a program is executed. This is important for testing, as that way I can know which parts of my application actually has been executed by the testing procedures. Gcov can be used as well for profiling, but in this post I will use it to generate coverage information only. The general flow to generate code coverage is: Instrument code: Compile the application files with a special option. This will add (hidden) code and hooks which records how many times a piece of code is executed. Generate Instrumentation Information: as part of the previous steps, the compiler generates basic block and line information. This information is stored on the host as *.gcno (Gnu Coverage Notes Object?) files. Run the application: While the application is running on the target, the instrumented code will record how many the lines or blocks in the application are executed. This information is stored on the target (in RAM). Dump the recorded information: At application exit (or at any time), the recorded information needs to be stored and sent to the host. By default gcov stores information in files. As a file system might not be alway available, other methods can be used (serial connection, USB, ftp, …) to send and store the information. In this tutorial I show how the debugger can be used for this. The information is stored as *.gcda (Gnu Coverage Data Analysis?) files. Generate the reports and visualize them with gcov. General gcov Flow gcc does the instrumentation and provides the library for code coverage, while gcov is the utility to analyze the generated data. Coverage: Compiler and Linker Options To generate the *.gcno files, the following option has to be added for each file which should generate coverage information: -fprofile-arcs -ftest-coverage :idea: There is as well the ‘–coverage’ option (which is a shortcut option) which can be used both for the compiler and linker. But I prefer the ‘full’ options so I know what is behind the options. -fprofile-arcs Compiler Option The option -fprofile-arcs adds code to the program flow to so execution of source code lines are counted. It does with instrumenting the program flow arcs. From https://gcc.gnu.org/onlinedocs/gcc/Debugging-Options.html: -fprofile-arcs Add code so that program flow arcs are instrumented. During execution the program records how many times each branch and call is executed and how many times it is taken or returns. When the compiled program exits it saves this data to a file called auxname.gcda for each source file. The data may be used for profile-directed optimizations (-fbranch-probabilities), or for test coverage analysis (-ftest-coverage). Each object file’s auxname is generated from the name of the output file, if explicitly specified and it is not the final executable, otherwise it is the basename of the source file. In both cases any suffix is removed (e.g. foo.gcda for input file dir/foo.c, or dir/foo.gcda for output file specified as -o dir/foo.o). See Cross-profiling. If you are not familiar with compiler technology or graph theory: An ‘Arc‘ (alternatively ‘edge’ or ‘branch’) is a directed link between a pair ‘Basic Blocks‘. A Basic is a sequence of code which has no branching in it (it is executed in a single sequence). For example if you have the following code: k = 0; if (i==10) { i += j; j++; } else { foo(); } bar(); Then this consists of the following four basic blocks: Basic Blocks The ‘Arcs’ are the directed edges (arrows) of the control flow. It is important to understand that not every line of the source gets instrumented, but only the arcs: This means that the instrumentation overhead (code size and data) depends how ‘complicated’ the program flow is, and not how many lines the source file has. However, there is an important aspect to know about gcov: it provides ‘condition coverage‘ if a full expression evaluates to TRUE or FALSE. Consider the following case: if (i==0 || j>=20) { In other words: I get coverage how many times the ‘if’ has been executed, but *not* how many times ‘i==0′ or ‘j>=20′ (which would be ‘decision coverage‘, which is not provided here). See http://www.bullseye.com/coverage.html for all the details. -ftest-coverage Compiler Option The second option for the compiler is -ftest-coverage (from https://gcc.gnu.org/onlinedocs/gcc-3.4.5/gcc/Debugging-Options.html): -ftest-coverage Produce a notes file that the gcov code-coverage utility (see gcov—a Test Coverage Program) can use to show program coverage. Each source file’s note file is called auxname.gcno. Refer to the -fprofile-arcs option above for a description of auxname and instructions on how to generate test coverage data. Coverage data will match the source files more closely, if you do not optimize. So this option generates the *.gcno file for each source file I decided to instrument: gcno file generated This file is needed later to visualize the data with gcov. More about this later. Adding Compiler Options So with this knowledge, I need to add -fprofile-arcs -ftest-coverage as compiler option to every file I want to profile. It is not necessary profile the full application: to save ROM and RAM and resources, I can add this option only to the files needed. Actually as a starter, I recommend to instrument a single source file only at the beginning. For this I select the properties (context menu) of my file Test.c I add the options in ‘other compiler flags': Coverage Added to Compilation File -fprofile-arcs Linker Option Profiling not only needs a compiler option: I need to tell the linker that it needs to link with the profiler library. For this I add -fprofile-arcs to the linker options: -fprofile-arcs Linker Option Coverage Stubs Depending on your library settings, you might now get a lot of unresolved symbol linker errors. This is because by default the profiling library assumes to write the profiling information to a file system. However, most file systems do *not* have a file system. To overcome this, I add a stubs for all the needed functions. I have them added with a file to my project (see latest version of that file on GitHub): /* * coverage_stubs.c * * These stubs are needed to generate coverage from an embedded target. */ #include #include #include #include #include #include #include "UTIL1.h" #include "coverage_stubs.h" /* prototype */ void gcov_exit(void); /* call the coverage initializers if not done by startup code */ void static_init(void) { void (**p)(void); extern uint32_t __init_array_start, __init_array_end; /* linker defined symbols, array of function pointers */ uint32_t beg = (uint32_t)&__init_array_start; uint32_t end = (uint32_t)&__init_array_end; while(begst_mode = S_IFCHR; return 0; } int _getpid(void) { return 1; } int _isatty(int file) { switch (file) { case STDOUT_FILENO: case STDERR_FILENO: case STDIN_FILENO: return 1; default: errno = EBADF; return 0; } } int _kill(int pid, int sig) { (void)pid; (void)sig; errno = EINVAL; return (-1); } int _lseek(int file, int ptr, int dir) { (void)file; (void)ptr; (void)dir; return 0; /* return offset in file */ } #pragma GCC diagnostic push #pragma GCC diagnostic ignored "-Wreturn-type" __attribute__((naked)) static unsigned int get_stackpointer(void) { __asm volatile ( "mrs r0, msp \r\n" "bx lr \r\n" ); } #pragma GCC diagnostic pop void *_sbrk(int incr) { extern char __HeapLimit; /* Defined by the linker */ static char *heap_end = 0; char *prev_heap_end; char *stack; if (heap_end==0) { heap_end = &__HeapLimit; } prev_heap_end = heap_end; stack = (char*)get_stackpointer(); if (heap_end+incr > stack) { _write (STDERR_FILENO, "Heap and stack collision\n", 25); errno = ENOMEM; return (void *)-1; } heap_end += incr; return (void *)prev_heap_end; } int _read(int file, char *ptr, int len) { (void)file; (void)ptr; (void)len; return 0; /* zero means end of file */ } :idea: In this code I’m using the UTIL1 (Utility) Processor Expert component, available on SourceForge. If you do not want/need this, you can remove the lines with UTIL1. - Coverage Stubs File in Project Coverage Constructors There is one important thing to mention: the coverage data structures need to be initialized, similar to constructors for C++. Depending on your startup code, this might *not* be done automatically. Check your linker .map file for some _GLOBAL__ symbols: .text._GLOBAL__sub_I_65535_0_TEST_Test 0x0000395c 0x10 ./Sources/Test.o Such a symbol should exist for every source file which has been instrumented with coverage information. These are functions which need to be called as part of the startup code. Set a breakpoint in your code at the given address to check if it gets called. If not, you need to call it yourself. :!: Typically I use the linker option ‘-nostartfiles’), and I have my startup code. In that case, these constructors are not called by default, so I need to do myself. See http://stackoverflow.com/questions/6343348/global-constructor-call-not-in-init-array-section In my linker file I have this: .init_array : { PROVIDE_HIDDEN (__init_array_start = .); KEEP (*(SORT(.init_array.*))) KEEP (*(.init_array*)) PROVIDE_HIDDEN (__init_array_end = .); } > m_text This means that there is a list of constructor function pointers put together between __init_array_start and __init_array_end. So all what I need is to iterate through this array and call the function pointers: /* call the coverage initializers if not done by startup code */ void static_init(void) { void (**p)(void); extern uint32_t __init_array_start, __init_array_end; /* linker defined symbols, array of function pointers */ uint32_t beg = (uint32_t)&__init_array_start; uint32_t end = (uint32_t)&__init_array_end; while(beg stack) { _write (STDERR_FILENO, "Heap and stack collision\n", 25); errno = ENOMEM; return (void *)-1; } heap_end += incr; return (void *)prev_heap_end; } :!: It might be that several kBytes of heap are needed. So if you are running in a memory constraint system, be sure that you have enough RAM available. The above implementation assumes that I have space between my heap end and the stack area. :!: If your memory mapping/linker file is different, of course you will need to change that _sbrk() implementation. Compiling and Building Now the application should compile and link without errors.Check that the .gcno files are generated: :idea: You might need to refresh the folder in Eclipse. - .gcno files generated In the next steps I’m showing how to get the coverage data as *.gcda files to the host using gdb. Using Debugger to get the Coverage Data The coverage data gets dumped when _exit() gets called by the application. Alternatively I could call gcov_exit() or __gcov_flush() any time. What it then does is Open the *.gcda file with _open() for every instrumented source file. Write the data to the file with _write(). So I can set a breakpoint in the debugger to both _open() and _write() and have all the data I need :-) With _open() I get the file name, and I store it in a global pointer so I can reference it in _write(): static const unsigned char *fileName; /* file name used for _open() */ int _open (const char *ptr, int mode) { (void)mode; fileName = (const unsigned char*)ptr; /* store file name for _write() */ return 0; } In _write() I get a pointer to the data and the length of the data. Here I can dump the data to a file using the gdb command: dump binary memory I could use a calculator to calculate the memory dump range, but it is much easier if I let the program generate the command line for gdb :-): int _write(int file, char *ptr, int len) { static unsigned char gdb_cmd[128]; /* command line which can be used for gdb */ (void)file; /* construct gdb command string */ UTIL1_strcpy(gdb_cmd, sizeof(gdb_cmd), (unsigned char*)"dump binary memory "); UTIL1_strcat(gdb_cmd, sizeof(gdb_cmd), fileName); UTIL1_strcat(gdb_cmd, sizeof(gdb_cmd), (unsigned char*)" 0x"); UTIL1_strcatNum32Hex(gdb_cmd, sizeof(gdb_cmd), (uint32_t)ptr); UTIL1_strcat(gdb_cmd, sizeof(gdb_cmd), (unsigned char*)" 0x"); UTIL1_strcatNum32Hex(gdb_cmd, sizeof(gdb_cmd), (uint32_t)(ptr+len)); return 0; } That way I can copy the string in the gdb debugger: Generated GDB Memory Dump Command That command gets pasted and executed in the gdb console: gdb command line After execution of the program, the *.gcda file gets created (refresh might be necessary to show it up): gcda file created Repeat this for all instrumented files as necessary. Showing Coverage Information To show the coverage information, I need the *.gcda, the *.gcno plus the .elf file. :idea: Use Refresh if not all files are shown in the Project Explorer view Files Ready to Show Coverage Information Then double-click on the gcda file to show coverage results: Double Click on gcda File Press OK, and it opens the gcov view. Double click on file in that view to show the details: gcov Views Use the chart icon to create a chart view: Chart view Bar Graph View Video of Steps to Create and Use Coverage The following video summarizes the steps needed: Data and Code Overhead Instrumenting code to generate coverage information means that it is an intrusive method: it impacts the application execution speed, and needs extra RAM and ROM. How much heavily depends on the complexity of the control flow and on the number of arcs. Higher compiler optimizations would reduce the code size footprint, however optimizations are not recommended for coverage sessions, as this might make the job of the coverage much harder. I made a quick comparison using my test application. I used the ‘size’ GNU command (see “Printing Code Size Information in Eclipse”). Without coverage enabled, the application footprint is: arm-none-eabi-size --format=berkeley "FRDM-K64F_Coverage.elf" text data bss dec hex filename 6360 1112 5248 12720 31b0 FRDM-K64F_Coverage.elf With coverage enabled only for Test.c gave: arm-none-eabi-size --format=berkeley "FRDM-K64F_Coverage.elf" text data bss dec hex filename 39564 2376 9640 51580 c97c FRDM-K64F_Coverage.elf Adding main.c to generate coverage gives: arm-none-eabi-size --format=berkeley "FRDM-K64F_Coverage.elf" text data bss dec hex filename 39772 2468 9700 51940 cae4 FRDM-K64F_Coverage.elf So indeed there is some initial add-up because of the coverage library, but afterwards adding more source files does not add up much. Summary It took me a while and reading many articles and papers to get code coverage implemented for an embedded target. Clearly, code coverage is easier if I have a file system and plenty of resources available. But I’m now able to retrieve coverage information from a rather small embedded system using the debugger to dump the data to the host. It is not practical for large sets of files, but at least a starting point :-). I have committed my Eclipse Kepler/Launchpad project I used in this tutorial on GitHub. Ideas I have in my mind: Instead using the debugger/gdb, use FatFS and SD card to store the data Exploring how to use profiling Combining multiple coverage runs Happy Covering :-) Links: Blog article who helped me to explore gcov for embedded targets: http://simply-embedded.blogspot.ch/2013/08/code-coverage-introduction.html Paper about using gcov for Embedded Systems: http://sysrun.haifa.il.ibm.com/hrl/greps2007/papers/gcov-on-an-embedded-system.pdf Article about coverage options for GNU compiler and linker: http://bobah.net/d4d/tools/code-coverage-with-gcov How to call static constructor methods manually: http://stackoverflow.com/questions/6343348/global-constructor-call-not-in-init-array-section Article about using gcov with lcov: https://qiaomuf.wordpress.com/2011/05/26/use-gcov-and-lcov-to-know-your-test-coverage/ Explanation of different coverage methods and terminology: http://www.bullseye.com/coverage.html

originally written by david ducos mydumper is known as the faster (much faster) mysqldump alternative. so, if you take a logical backup you will choose mydumper instead of mysqldump. but what about the restore? well, who needs to restore a logical backup? it takes ages! even with myloader. but this could change just a bit if we are able to take advantage of fast index creation. as you probably know, mydumper and mysqldump export the struct of a table, with all the indexes and the constraints, and of course, the data. then, myloader and mysql import the struct of the table and import the data. the most important difference is that you can configure myloader to import the data using a certain amount of threads. the import steps are: create the complete struct of the table import the data when you execute myloader, internally it first creates the tables executing the “-schema.sql” files and then takes all the filenames without “schema.sql” and puts them in a task queue. every thread takes a filename from the queue, which actually is a chunk of the table, and executes it. when finished it takes another chunk from the queue, but if the queue is empty it just ends. this import procedure works fast for small tables, but with big tables with large indexes the inserts are getting slower caused by the overhead of insert the new values in secondary indexes. another way to import the data is: split the table structure into table creation with primary key, indexes creation and constraint creation create tables with primary key per table do: load the data create index create constraints this import procedure is implemented in a branch of myloader that can be downloaded from here or directly executing bzr with the repository: bzr branch lp:~david-ducos/mydumper/mydumper the tool reads the schema files and splits them into three separate statements which create the tables with the primary key, the indexes and the constraints. the primary key is kept in the table creation in order to avoid the recreation of the table when a primary key is added and the “key” and “constraint” lines are removed. these lines are added to the index and constraint statements, respectively. it processes tables according to their size starting with the largest because creating the indexes of a big table could take hours and is single-threaded. while we cannot process other indexes at the time, we are potentially able to create other tables with the remaining threads. it has a new thread (monitor_process) that decides which chunk of data will be put in the task queue and a communication queue which is used by the task processes to tell the monitor_process which chunk has been completed. i run multiple imports on an aws m1.xlarge machine with one table comparing myloader and this branch and i found that with large indexes the times were: as you can see, when you have less than 150m rows, import the data and then create the indexes is higher than import the table with the indexes all at once. but everything changes after 150m rows, import 200m takes 64 minutes more for myloader but just 24 minutes for the new branch. on a table of 200m rows with a integer primary key and 9 integer columns, you will see how the time increases as the index gets larger: where: 2-2-0: two 1-column and two 2-column index 2-2-1: two 1-column, two 2-column and one 3-column index 2-3-1: two 1-column, three 2-column and one 3-column index 2-3-2: two 1-column, three 2-column and two 3-column index conclusion this branch can only import all the tables with this same strategy, but with this new logic in myloader, in a future version it could be able to import each table with the best strategy reducing the time of the restore considerably.

Recently I was asked about diagnosing deadlocks in SQL Server – I’ve done a lot of work in this area way back in 2008, so I figure it’s time for a refresher. If there’s a lot of interest in exploring SQL Server and deadlocks further, I’m happy to write an extended article going into far more detail. Just let me know. Before we get into diagnosis and investigation, it’s a good time to pose the question: “what is a deadlock?”: From TechNet: A deadlock occurs when two or more tasks permanently block each other by each task having a lock on a resource which the other tasks are trying to lock. The following graph presents a high level view of a deadlock state where: Task T1 has a lock on resource R1 (indicated by the arrow from R1 to T1) and has requested a lock on resource R2 (indicated by the arrow from T1 to R2). Task T2 has a lock on resource R2 (indicated by the arrow from R2 to T2) and has requested a lock on resource R1 (indicated by the arrow from T2 to R1). Because neither task can continue until a resource is available and neither resource can be released until a task continues, a deadlock state exists. The SQL Server Database Engine automatically detects deadlock cycles within SQL Server. The Database Engine chooses one of the sessions as a deadlock victim and the current transaction is terminated with an error to break the deadlock. Basically, it’s a resource contention issue which blocks one process or transaction from performing actions on resources within SQL Server. This can be a serious condition, not just for SQL Server as processes become suspended, but for the applications which rely on SQL Server as well. The T-SQL Approach A fast way to respond is to execute a bit of T-SQL on SQL Server, making use of System Views. The following T-SQL will show you the “victim” processes, much like activity monitor does: select * from sys.sysprocesses where blocked > 0 Which is not particularly useful (but good to know, so you can see the blocked count). To get to the heart of the deadlock, this is what you want (courtesy of this SO question/answer): SELECT Blocker.text –, Blocker.*, * FROM sys.dm_exec_connections AS Conns INNER JOIN sys.dm_exec_requests AS BlockedReqs ON Conns.session_id = BlockedReqs.blocking_session_id INNER JOIN sys.dm_os_waiting_tasks AS w ON BlockedReqs.session_id = w.session_id CROSS APPLY sys.dm_exec_sql_text(Conns.most_recent_sql_handle) AS Blocker This will show you line and verse (the actual statement causing the resource block) – see the attached screenshot for an example. However, the generally accepted way to determine and diagnose deadlocks is through the use of SQL Server trace flags. SQL Trace Flags They are (usually) set temporarily, and they cause deadlocking information to be dumped to the SQL management logs. The flags that are useful are flags 1204 and 1222. From TechNet: https://technet.microsoft.com/en-us/library/ms178104%28v=sql.105%29.aspx Trace flags are set on or off by using either of the following methods: · Using the DBCC TRACEON and DBCC TRACEOFF commands. For example, DBCC TRACEON 2528: To enable the trace flag globally, use DBCC TRACEON with the -1 argument: DBCC TRACEON (2528, -1). To turn off a global trace flag, use DBCC TRACEOFF with the -1 argument. · Using the -T startup option to specify that the trace flag be set on during startup. The -T startup option enables a trace flag globally. You cannot enable a session-level trace flag by using a startup option. So to enable or disable deadlock trace flags globally, you’d use the following T-SQL: DBCC TRACEON (1204, -1) DBCC TRACEON (1222, -1) DBCC TRACEOFF (1204, -1) DBCC TRACEOFF (1222, -1) Due to the overhead, it’s best to enable the flag at runtime rather than on start up. Note that the scope of a non-startup trace flag can be global or session-level. Basic Deadlock Simulation By way of a very simple scenario, you can make use of SQL Management Studio (and breakpoints) to roughly simulate a deadlock scenario. Given the following basic table schema: CREATE TABLE [dbo].[UploadedFile]( [Id] [int] NOT NULL, [Filename] [nvarchar](50) NOT NULL, [DateCreated] [datetime] NOT NULL, [DateModified] [datetime] NULL, CONSTRAINT [PK_UploadedFile] PRIMARY KEY CLUSTERED ( [Id] ASC )WITH (STATISTICS_NORECOMPUTE = OFF, IGNORE_DUP_KEY = OFF) ) With some basic test data in it: If you create two separate queries in SQL Management Studio, use the following transaction (Query #1) to lock rows in the table: SET TRANSACTION ISOLATION LEVEL SERIALIZABLE BEGIN TRANSACTION SELECT [Id],[Filename],[DateCreated],[DateModified] FROM [dbo].[UploadedFile] WHERE DateCreated > ‘2015-01-01′ ROLLBACK TRANSACTION Now add a “victim” script (Query #2) in a separate query session: UPDATE [dbo].[UploadedFile] SET [DateModified] = ‘2014-12-31′ WHERE DateCreated > ‘2015-01-01′ As long as you set a breakpoint on the ROLLBACK TRANSACTION statement, you’ll block the second query due to the isolation level of the transaction which wraps query #1. Now you can use the diagnostic T-SQL to examine the victim and the blocking transaction. Enjoy!

Recently I had a dispute with my colleagues regarding performance penalty of lazy vals in Scala. It resulted in a set of microbenchmarks which compare lazy and non-lazy vals performance. All the sources can be found at http://git.io/g3WMzA. But before going to the benchmark results let's try to understand what can cause the performance penalty. For my JMH benchmark I created a very simple Scala class with lazy val in it: @State(Scope.Benchmark) class LazyValCounterProvider { lazy val counter = SlowInitializer.createCounter() } Now let's take a look at what is hidden under the hood of lazy keyword. At first, we need to compile given code with scalac, and then it can be decompiled to correspondent Java code. For this sake I used JD decompiler. It produced the following code: @State(Scope.Benchmark) @ScalaSignature(bytes="...") public class LazyValCounterProvider { private SlowInitializer.Counter counter; private volatile boolean bitmap$0; private SlowInitializer.Counter counter$lzycompute() { synchronized (this) { if (!this.bitmap$0) { this.counter = SlowInitializer.createCounter(); this.bitmap$0 = true; } return this.counter; } } public SlowInitializer.Counter counter() { return this.bitmap$0 ? this.counter : counter$lzycompute(); } } As it's seen, the lazy keyword is translated to a classical double-checked locking idiom for delayed initialization. Thus, most of the time the only performance penalty may come from a single volatile read per lazy val read (except for the time it takes to initialize lazy val instance since its very first usage). Let's finally measure its impact in numbers. My JMH-based microbenchmark is as simple as: public class LazyValsBenchmarks { @Benchmark public long baseline(ValCounterProvider eagerProvider) { return eagerProvider.counter().incrementAndGet(); } @Benchmark public long lazyValCounter(LazyValCounterProvider provider) { return provider.counter().incrementAndGet(); } } A baseline method access a final counter object and increments an integer value by 1 by calling incrementAndGet . And as we've just found out, the main benchmark method - lazyValCounter - in addition to what baseline method does also does one volatile read. Note: all measurements are performed on MBA with Core i5 1.7GHz CPU. All results were obtained by running JMH in a throughput mode. Both score and score error columns show operations/second. Each JMH run made 10 iterations and took 50 seconds. I performed 6 measurements with the different JVM and JMH options: client VM, 1 thread Benchmark Score Score error baseline 412277751.619 8116731.382 lazyValCounter 352209296.485 6695318.185 client VM, 2 threads Benchmark Score Score error baseline 542605885.932 15340285.497 lazyValCounter 383013643.710 53639006.105 client VM, 4 threads Benchmark Score Score error baseline 551105008.767 5085834.663 lazyValCounter 394175424.898 3890422.327 server VM, 1 thread Benchmark Score Score error baseline 407010942.139 9004641.910 lazyValCounter 341478430.115 18183144.277 server VM, 2 threads Benchmark Score Score error baseline 531472448.578 22779859.685 lazyValCounter 428898429.124 24720626.198 server VM, 4 threads Benchmark Score Score error baseline 549568334.970 12690164.639 lazyValCounter 374460712.017 17742852.788 The numbers show that lazy vals performance penalty is quite small and can be ignored in practice. For further reading about the subject I would recommend SIP 20 - Improved Lazy Vals Initialization, which contains very interesting in-depth analysis of existing issues with lazy initialization implementation in Scala.

After we introduced locked thread detection to Plumbr couple of months ago, we have started to receive queries similar to “hey, great, now I understand what is causing my performance issues, but what I am supposed to do now?” We are working hard to build the solution instructions into our own product, but in this post I am going to share several common techniques you can apply independent of the tool used for detecting the lock. The methods include lock splitting, concurrent data structures, protecting the data instead of the code and lock scope reduction. Locking is not evil, lock contention is Whenever you face a performance problem with the threaded code there is a chance that you will start blaming locks. After all, common “knowledge” is that locks are slow and limit scalability. So if you are equipped with this “knowledge” and start to optimize the code and getting rid of locks there is a chance that you end up introducing nasty concurrency bugs that will surface later on. So it is important to understand the difference between contended and uncontended locks. Lock contention occurs when a thread is trying to enter the synchronized block/method currently executed by another thread. This second thread is now forced to wait until the first thread has completed executing the synchronized block and releases the monitor. When only one thread at a time is trying to execute the synchronized code, the lock stays uncontended. As a matter of fact, synchronization in JVM is optimized for the uncontended case and for the vast majority of the applications, uncontended locks pose next to no overhead during execution. So, it is not locks you should blame for performance, but contended locks. Equipped with this knowledge, lets see what we can do to reduce either the likelihood of contention or the length of the contention. Protect the data not the code A quick way to achieve thread-safety is to lock access to the whole method. For example, take look at the following example, illustrating a naive attempt to build an online poker server: class GameServer { public Map<> tables = new HashMap>(); public synchronized void join(Player player, Table table) { if (player.getAccountBalance() > table.getLimit()) { List tablePlayers = tables.get(table.getId()); if (tablePlayers.size() < 9) { tablePlayers.add(player); } } } public synchronized void leave(Player player, Table table) {/*body skipped for brevity*/} public synchronized void createTable() {/*body skipped for brevity*/} public synchronized void destroyTable(Table table) {/*body skipped for brevity*/} } The intentions of the author have been good - when new players join() the table, there must be a guarantee that the number of players seated at the table would not exceed the table capacity of nine. But whenever such a solution would actually be responsible for seating players to tables - even on a poker site with moderate traffic, the system would be doomed to constantly trigger contention events by threads waiting for the lock to be released. Locked block contains account balance and table limit checks which potentially can involve expensive operations both increasing the likelihood and length of the contention. First step towards solution would be making sure we are protecting the data, not the code by moving the synchronization from the method declaration to the method body. In the minimalistic example above, it might not change much at the first place. But lets consider the whole GameServerinterface, not just the single join() method: class GameServer { public Map> tables = new HashMap>(); public void join(Player player, Table table) { synchronized (tables) { if (player.getAccountBalance() > table.getLimit()) { List tablePlayers = tables.get(table.getId()); if (tablePlayers.size() < 9) { tablePlayers.add(player); } } } } public void leave(Player player, Table table) {/* body skipped for brevity */} public void createTable() {/* body skipped for brevity */} public void destroyTable(Table table) {/* body skipped for brevity */} } What originally seemed to be a minor change, now affects the behaviour of the whole class. Whenever players were joining tables, the previously synchronized methods locked on theGameServer instance (this) and introduced contention events to players trying to simultaneouslyleave() tables. Moving the lock from the method signature to the method body postpones the locking and reduces the contention likelihood. Reduce the lock scope Now, after making sure it is the data we actually protect, not the code, we should make sure our solution is locking only what is necessary - for example when the code above is rewritten as follows: public class GameServer { public Map> tables = new HashMap>(); public void join(Player player, Table table) { if (player.getAccountBalance() > table.getLimit()) { synchronized (tables) { List tablePlayers = tables.get(table.getId()); if (tablePlayers.size() < 9) { tablePlayers.add(player); } } } } //other methods skipped for brevity } then the potentially time-consuming operation of checking player account balance (which potentially can involve IO operations) is now outside the lock scope. Notice that the lock was introduced only to protect against exceeding the table capacity and the account balance check is not anyhow part of this protective measure. Split your locks When we look at the last code example, you can clearly notice that the whole data structure is protected by the same lock. Considering that we might hold thousands of poker tables in this structure, it still poses a high risk for contention events as we have to protect each table separately from overflowing in capacity. For this there is an easy way to introduce individual locks per table, such as in the following example: public class GameServer { public Map> tables = new HashMap>(); public void join(Player player, Table table) { if (player.getAccountBalance() > table.getLimit()) { List tablePlayers = tables.get(table.getId()); synchronized (tablePlayers) { if (tablePlayers.size() < 9) { tablePlayers.add(player); } } } } //other methods skipped for brevity } Now, if we synchronize the access only to the same table instead of all the tables, we have significantly reduced the likelihood of locks becoming contended. Having for example 100 tables in our data structure, the likelihood of the contention is now 100x smaller than before. Use concurrent data structures Another improvement is to drop the traditional single-threaded data structures and use data structures designed explicitly for concurrent usage. For example, when picking ConcurrentHashMapto store all your poker tables would result in code similar to following: public class GameServer { public Map> tables = new ConcurrentHashMap>(); public synchronized void join(Player player, Table table) {/*Method body skipped for brevity*/} public synchronized void leave(Player player, Table table) {/*Method body skipped for brevity*/} public synchronized void createTable() { Table table = new Table(); tables.put(table.getId(), table); } public synchronized void destroyTable(Table table) { tables.remove(table.getId()); } } The synchronization in join() and leave() methods is still behaving as in our previous example, as we need to protect the integrity of individual tables. So no help from ConcurrentHashMap in this regards. But as we are also creating new tables and destroying tables in createTable() and destroyTable()methods, all these operations to the ConcurrentHashMap are fully concurrent, permitting to increase or reduce the number of tables in parallel. Other tips and tricks Reduce the visibility of the lock. In the example above, the locks are declared public and are thus visible to the world, so there is there is a chance that someone else will ruin your work by also locking on your carefully picked monitors. Check out java.util.concurrent.locks to see whether any of the locking strategies implemented there will improve the solution. Use atomic operations. The simple counter increase we are actually conducting in example above does not actually require a lock. Replacing the Integer in count tracking withAtomicInteger would most suit this example just fine. Hope the article helped you to solve the lock contention issues, independent of whether you are using Plumbr automatic lock detection solution or manually extracting the information from thread dumps.

Part 1 – Configuring Ehcache with Spring Ehcache is most widely used open source cache for boosting performance, offloading your database, and simplifying scalability. It is very easy to configure Ehcache with Spring context xml. Assuming you have a working Spring configured application, here we discuss how to integrate Ehcache with Spring. In order to configure it, we need to add a CacheManager into the Spring context file. Code snippet added below In ‘ehcache’ bean, we are referring to ‘Ehcache.xml’ file which will contain all ehcache configurations. Sample ehcache file is added below. You can configure this file according to your project needs. Here you can two caches, defaultCache and one defined with name ‘TestCache’. DefaultCache configuration is applied to any cache that is not explicitly configured. Part 2 - Enabling Caching Annotations In order to used Spring provided java annotations , we need to declarative enable cache annotations in Spring config file. Code snippet added below. In order to use above annotation declaration, we need to add cache namespace into the spring file. Code snippet added below. Spring Annotations for caching @Cacheable triggers cache population @CacheEvict triggers cache eviction @CachePut updates the cache without interfering with the method execution I will give brief description and practical implementation of these annotations. For detailed explanations please refer spring documentation for caching . @Cacheable - This annotation means that the return value of the corresponding method can be cached and multiple invocation of this method with same arguments must populate the return value from cache without executing the method. Code snippet added below. @Cacheable("customer") public Customer findCustomer(String custId) {...} Each time the method is called, the cache with name ‘customer’ is checked to see whether the invocation has been already executed for the same ‘custId’. @CachePut - This annotation is used to update the cache without interfering the method execution. Method will always be executed and its result will be replaced in the cache. This annoatation is used for cache population. @CachePut("customer") public Customer updateBook(String custId){...} @CacheEvict – This annotation is used for cache eviction, used to remove stale or unused data from the cache. @CacheEvict(value="customer", allEntries=true) public void unloadCustomer(String newBatch) With above configurations, all entries in the ‘customer’ cache will be removed. Above description will give head start to configure Ehcachewith spring and use some of the caching annotations.

Originally Written byAndrew Moore MySQL table alterations can interrupt production traffic causing bad customer experience or in worst cases, loss of revenue. Not all DBAs, developers, syadmins know MySQL well enough to avoid this pitfall. DBAs usually encounter these kinds of production interruptions when working with upgrade scripts that touch both application and database or if an inexperienced admin/dev engineer perform the schema change without knowing how MySQL operates internally. Truths * Direct MySQL ALTER table locks for duration of change (pre-5.6) * Online DDL in MySQL 5.6 is not always online and may incurr locks * Even with Percona Toolkit‘s pt-online-schema-change there are several workloads that can experience blocking Here on the Percona MySQL Managed Services team we encourage our clients to work with us when planning and performing schema migrations. We aim to ensure that we are using the best method available in their given circumstance. Our intentions to avoid blocking when performing DDL on large tables ensures that business can continue as usual whilst we strive to improve response time or add application functionality. The bottom line is that a business relying on access to its data cannot afford to be down during core trading hours. Many of the installations we manage are still below MySQL 5.6, which requires us to seek workarounds to minimize the amount of disruption a migration can cause. This may entail slave promotion or changing the schema with an ‘online schema change’ tool. MySQL version 5.6 looks to address this issue by reducing the number of scenarios where a table is rebuilt and locked but it doesn’t yet cover all eventualities, for example when changing the data type of a column a full table rebuild is necessary. The topic of 5.6 Online Schema Change was discussed in great detail last year in the post, “Schema changes – what’s new in MySQL 5.6?” by Przemysław Malkowski With new functionality arriving in MySQL 5.7, we look forward to non-blocking DDL operations such as; OPTIMIZE TABLE and RENAME INDEX. (More info) The best advice for MySQL 5.6 users is to review the matrix to familiarize with situations where it might be best to look outside of MySQL to perform schema changes, the good news is that we’re on the right path to solving this natively. Truth be told, a blocking alter is usually going to go unnoticed on a 30MB table and we tend to use a direct alter in this situation, but on a 30GB or 300GB table we have some planning to do. If there is a period of time where activity is low and the this is permissive of locking the table then sometimes it is better execute within this window. Frequently though we are reactive to new SQL statements or a new performance issue and an emergency index is required to reduce load on the master in order to improve the response time. To pt-osc or not to pt-osc? As mentioned, pt-online-schema-change is a fixture in our workflow. It’s usually the right way to go but we still have occasions where pt-online-schema-change cannot be used, for example; when a table already uses triggers. It’s an important to remind ourselves of the the steps that pt-online-schema-change traverses to complete it’s job. Lets look at the source code to identify these; [moore@localhost]$ egrep 'Step' pt-online-schema-change # Step 1: Create the new table. # Step 2: Alter the new, empty table. This should be very quick, # Step 3: Create the triggers to capture changes on the original table and <--(metadata lock) # Step 4: Copy rows. # Step 5: Rename tables: orig -> old, new -> orig <--(metadata lock) # Step 6: Update foreign key constraints if there are child tables. # Step 7: Drop the old table. [moore@localhost]$egrep'Step'pt-online-schema-change # Step 1: Create the new table. # Step 2: Alter the new, empty table. This should be very quick, # Step 3: Create the triggers to capture changes on the original table and <--(metadata lock) # Step 4: Copy rows. # Step 5: Rename tables: orig -> old, new -> orig <--(metadata lock) # Step 6: Update foreign key constraints if there are child tables. # Step 7: Drop the old table. I pick out steps 3 and 5 from above to highlight a source of a source of potential downtime due to locks, but step 6 is also an area for concern since foreign keys can have nested actions and should be considered when planning these actions to avoid related tables from being rebuilt with a direct alter implicitly. There are several ways to approach a table with referential integrity constraints and they are detailed within the pt-osc documentation a good preparation step is to review the structure of your table including the constraints and how the ripples of the change can affect the tables around it. Recently we were alerted to an incident after a client with a highly concurrent and highly transactional workload ran a standard pt-online-schema-change script over a large table. This appeared normal to them and a few hours later our pager man was notified that this client was experiencing max_connections limit reached. So what was going on? When pt-online-schema-change reached step 5 it tried to acquire a metadata lock to rename the the original and the shadow table, however this wasn’t immediately granted due to open transactions and thus threads began to queue behind the RENAME command. The actual effect this had on the client’s application was downtime. No new connections could be made and all existing threads were waiting behind the RENAME command. Metadata locks Introduced in 5.5.3 at server level. When a transaction starts it will acquire a metadata lock (independent of storage engine) on all tables it uses and then releases them when it’s finished it’s work. This ensures that nothing can alter the table definition whilst a transaction is open. With some foresight and planning we can avoid these situations with non-default pt-osc options, namely –nodrop-new-table and –no-swap-tables. This combination leaves both the shadow table and the triggers inplace so that we can instigate an atomic RENAME when load permits. EDIT: as of percona-toolkit version 2.2 we have a new variable –tries which in conjunction with –set-vars has been deployed to cover this scenario where various pt-osc operations could block waiting for a metadata lock. The default behaviour of pt-osc (–set-vars) is to set the following session variables when it connects to the server; wait_timeout=10000 innodb_lock_wait_timeout=1 lock_wait_timeout=60 when using –tries we can granularly identify the operation, try count and the wait interval between tries. This combination will ensure that pt-osc will kill it’s own waiting session in good time to avoid the thread pileup and provide us with a loop to attempt to acquire our metadata lock for triggers|rename|fk management; –tries swap_tables:5:0.5,drop_triggers:5:0.5 The documentation is here http://www.percona.com/doc/percona-toolkit/2.2/pt-online-schema-change.html#cmdoption-pt-online-schema-change–tries This illustrates that even with a tool like pt-online-schema-change it is important to understand the caveats presented with the solution you think is most adequate. To help decide the direction to take use the flow chart to ensure you’re taking into account some of the caveats of the MySQL schema change. Be sure to read up on the recommended outcome though as there are uncharted areas such as disk space, IO load that are not featured on the diagram. Choosing the right DDL option Ensure you know what effect ALTER TABLE will have on your platform and pick the right method to suit your uptime. Sometimes that means delaying the change until a period of lighter use or utilising a tool that will avoid holding a table locked for the duration of the operation. A direct ALTER is sometimes the answer like when you have triggers installed on a table. – In most cases pt-osc is exactly what we need – In many cases pt-osc is needed but the way in which it’s used needs tweaking – In few cases pt-osc isn’t the right tool/method and we need to consider native blocking ALTER or using failovers to juggle the change into place on all hosts in the replica cluster. If you want to learn more about avoiding avoidable downtime please tune into my webinar Wednesday, November 19 at 10 a.m. PST. It’s titled “Tips from the Trenches: A Guide to Preventing Downtime for the Over-Extended DBA.” Register now!(If you miss it, don’t worry: Catch the recording and download the slides from that same registration page.)

In this article I will attempt to run the Mule ESB community edition in Docker in order to see whether it is feasible without any greater inconvenience. My goal is to be able to use Docker both when testing as well as in a production environment in order to gain better control over the environment and to separate different types of environments. I imagine that most of the Docker-related information can be applied to other applications – I have used Mule since it is what I usually work with. The conclusion I have made after having completed my experiments is that it is possible to run Mule ESB in Docker without any inconvenience. In addition, Docker will indeed allow me to have better control over the different environments and also allow me to separate them as I find appropriate. Finally, I just want to mention that I have used Docker in an Ubuntu environment. I have not attempted any of the exercises in Docker running on Windows or Mac OS X. Docker Briefly In short, Docker allows for creating of images that serve as blueprints for containers. A Docker container is an instance of a Docker image in the same way a Java object is an instance of a Java class. FROM codingtony/java MAINTAINER tony(dot)bussieres(at)ticksmith(dot)com RUN wget https://repository.mulesoft.org/nexus/content/repositories/releases/org/mule/distributions/mule-standalone/3.5.0/mule-standalone-3.5.0.tar.gz RUN cd /opt && tar xvzf ~/mule-standalone-3.5.0.tar.gz RUN echo "4a94356f7401ac8be30a992a414ca9b9 /mule-standalone-3.5.0.tar.gz" | md5sum -c RUN rm ~/mule-standalone-3.5.0.tar.gz RUN ln -s /opt/mule-standalone-3.5.0 /opt/mule CMD [ "/opt/mule/bin/mule" ] The resource isolation features of Linux are used to create Docker containers, which are more lightweight than virtual machines and are separated from the environment in which Docker runs, the host. Using Docker an image can be created that, every time it is started has a known state. In order to remove any doubts about whether the environment has been altered in any way, the container can be stopped and a new container started. I can even run multiple Docker containers on one and the same computer to simulate a multi-server production environment. Applications can also be run in their own Docker containers, as shown in this figure. Three Docker containers, each containing a specific application, running in one host. A more detailed introduction to Docker is available here. The main entry point to the Docker documentation can be found here. Motivation Some of the motivations I have for using Docker in both testing and production environments are: The environment in which I test my application should be as similar as the final deployment environment as possible, if not identical. Making the deployment environment easy to scale up and down. If it is easy to start a new processing node when need arise and stop it if it is no longer used, I will be able to adapt to changes rather quickly and thus reduce errors caused by, for instance, load peaks. Maintain an increased number of nodes to which applications can be deployed. Instead of running one instance of some kind of application server, Mule ESB in my case, on a computer, I want multiple instances that are partitioned, for instance, according to importance. High-priority applications run on one separate instance, which have higher priority both as far as resources (CPU, memory, disk etc) are concerned but also as far as support is concerned. Applications which are less critical run on another instance. Enable quick replacement of instances in the deployment environment. Reasons for having to replace instances may be hardware failure etc. Better control over the contents of the different environments. The concept of an environment that, at any time, may be disposed (and restarted) discourages hacks in the environment, which are usually poorly documented and sometimes difficult to trace. Using Docker, I need to change the appropriate Docker image if I want to make changes to some application environment. The Docker image file, commonly known as Dockerfile, can be checked into any ordinary revision control system, such as Git, Subversion etc, making changes reversible and traceable. Automate the creation of a testing environment. An example could be a nightly job that runs on my build server which creates a test environment, deploys one or more applications to it and then performs tests, such as load-testing. Prerequisites To get the best possible experience when running Docker, I run it under Ubuntu. According to the current documentation, Docker is supported under the following versions of Ubuntu: 12.04 LTS (64-bit) 13.04 (64-bit) 13.10 (64-bit) 14.04 (64-bit) Against my usual conservative self, I chose Ubuntu 14.10, which at the time of writing this article is the latest version. While I haven’t run into any issues, I cannot promise anything regarding compatibility with Docker as far as this version of Ubuntu is concerned. Installing Docker Before we install anything, those who have the Docker version from the Ubuntu repository should remove this version before installing a newer version of Docker, since the Ubuntu repository does not contain the most recent version and the package does not have the same name as the Docker package we will install: sudo apt-get remove docker.io The simplest way to install Docker is to use an installation script made available at the Docker website: curl -sSL https://get.docker.com/ubuntu/ | sudo sh If you are not running Ubuntu or if you do not want to use the above way of installing Docker, please refer to this page containing instructions on how to install Docker on various platforms. To verify the Docker installation, open a terminal window and enter: sudo docker version Output similar to the following should appear: Client version: 1.4.1 Client API version: 1.16 Go version (client): go1.3.3 Git commit (client): 5bc2ff8 OS/Arch (client): linux/amd64 Server version: 1.4.1 Server API version: 1.16 Go version (server): go1.3.3 Git commit (server): 5bc2ff8 We are now ready to start a Mule instance in Docker. Running Mule in Docker One of the advantages with Docker is that there is a large repository of Docker images that are ready to be used, and even extended if one so wishes. ThisDocker image is the one that I will use in this article. It is well documented, there is a source repository and it contains a recent version of the Mule ESB Community Edition. Some additional details on the Docker image: Ubuntu 14.04. Oracle JavaSE 1.7.0_65. This version will change as the PPA containing the package is updated. Mule ESB CE 3.5.0 Note that the image may change at any time and the specifications above may have changed. If you intend to use Docker in your organization, I would suspect that the best alternative is to create your own Docker images that are totally under your control. The Docker image repository is an excellent source of inspiration and aid even in this case. Starting a Docker Container To start a Docker container using this image, open a terminal window and write: sudo docker run codingtony/mule The first time an image is used it needs to be downloaded and created. This usually takes quite some time, so I suggest a short break here – perhaps for a cup of coffee or tea. If you just want to download an image without starting it, exchange the Docker command “run” with “pull”. Once the container is started, you will see some output to the console. If you are familiar with Mule, you will recognize the log output: MULE_HOME is set to /opt/mule-standalone-3.5.0 Running in console (foreground) mode by default, use Ctrl-C to exit... MULE_HOME is set to /opt/mule-standalone-3.5.0 Running Mule... --> Wrapper Started as Console Launching a JVM... Starting the Mule Container... Wrapper (Version 3.2.3) http://wrapper.tanukisoftware.org Copyright 1999-2006 Tanuki Software, Inc. All Rights Reserved. INFO 2015-01-05 04:41:42,302 [WrapperListener_start_runner] org.mule.module.launcher.MuleContainer: ********************************************************************** * Mule ESB and Integration Platform * * Version: 3.5.0 Build: ff1df1f3 * * MuleSoft, Inc. * * For more information go to http://www.mulesoft.org * * * * Server started: 1/5/15 4:41 AM * * JDK: 1.7.0_65 (mixed mode) * * OS: Linux (3.16.0-28-generic, amd64) * * Host: f95698cfb796 (172.17.0.2) * ********************************************************************** Note that: In the text-box containing information about the Mule ESB and Integration Platform, there is a row which starts with “Host:”. The hexadecimal digit that follows is the Docker container id and the IP-address is the external IP-address of the Docker container in which Mule is running. Before we do anything with the Mule instance running in Docker, let’s take a look at Docker containers. Docker Containers We can verify that there is a Docker container running by opening another terminal window, or a tab in the first terminal window, and running the command: sudo docker ps As a result, you will see output similar to the following (I have edited the output in order for the columns to be aligned with the column titles): CONTAINER ID IMAGE COMMAND CREATED STATUS PORTS NAMES f95698cfb796 codingtony/mule:latest "/opt/mule/bin/mule" 7 min ago Up 7 min jolly_hopper From this output we can see that: The ID of the container is f95698cfb796. This ID can be used when performing operations on the container, such as stopping it, restarting it etc. The name of the image used to created the container. The command that is currently executing. If we look at the Dockerfile for the image, we can see that the last line in this file is: CMD [ “/opt/mule/bin/mule” ] This is the command that is executed whenever an instance of the Docker image is launched and it matches what we see in the COMMAND column for the Docker container. The CREATED column shows how much time has passed since the container was created. The STATUS column shows the current status of the image. When you have used Docker for a while, you can view all the containers using: sudo docker ps -a This will show you containers that are not running, in addition to the running ones. Containers that are not running can be restarted. The PORTS column shows any port mappings for the container. More about port mappings later. Finally, the NAMES column contain a more human-friendly container name. This container name can be used in the same way as the container id. Docker containers will consume disk-space and if you want to determine how much disk-space each of the containers on your computer use, issue the following command: sudo docker ps -a -s An additional column, SIZE, will be shown and in this column I see that my Mule container consumes 41,76kB. Note that this is in addition to the disk-space consumed by the Docker image. This number will grow if you use the container under a longer period of time, as the container retains any files written to disk. To completely remove a stopped Docker container, find the id or name of the container and use the command: sudo docker rm [container id or name here] Before going further, let’s stop the running container and remove it: sudo docker stop [container id or name here] sudo docker rm [container id or name here] Files and Docker Containers So far we have managed to start a Mule instance running inside a Docker container, but there were no Mule applications deployed to it and the logs that were generated were only visible in the terminal window. I want to be able to deploy my applications to the Mule instance and examine the logs in a convenient way. In this section I will show how to: Share one or more directories in the host file-system with a Docker container. Access the files in a Docker container from the host. As the first step in looking at sharing directories between the host operating system and a Docker container, we are going to look at Mule logs. As part of this exercise we also set up the directories in the host operating system that are going to be shared with the Docker container. In your home directory, create a directory named “mule-root”. In the “mule-root” directory, create three directories named “apps”, “conf” and “logs”. Download the Mule CE 3.5.0 standalone distribution from this link. From the Mule CE 3.5.0 distribution, copy the files in the “apps” directory to the “mule-root/apps” directory you just created. From the Mule CE 3.5.0 distribution, copy the files in the “conf” directory to the “mule-root/conf” directory you created. The resulting file- and directory-structure should look like this (shown using the tree command): ~/mule-root/ ├── apps │ └── default │ └── mule-config.xml ├── conf │ ├── log4j.properties │ ├── tls-default.conf │ ├── tls-fips140-2.conf │ ├── wrapper-additional.conf │ └── wrapper.conf └── logs Edit the log4j.properties file in the “mule-root/conf” directory and set the log-level on the last line in the file to “DEBUG”. This modification has nothing to do with sharing directories, but is in order for us to be able to see some more output from Mule when we run it later. The last two lines should now look like this: # Mule classes log4j.logger.org.mule=DEBUG Binding Volumes We are now ready to launch a new Docker container and when we do, we will tell Docker to map three directories in the Docker container to three directories in the host operating system. Three directories in a Docker container bound to three directories in the host. Launch the Docker container with the command below. The -v option tells Docker that we want to make the contents of a directory in the host available at a certain path in the Docker container file-system. The -d option runs the container in the background and the terminal prompt will be available as soon as the id of the newly launched Docker container has been printed. sudo docker run -d -v ~/mule-root/apps:/opt/mule/apps -v ~/mule-root/conf:/opt/mule/conf -v ~/mule-root/logs:/opt/mule/logs codingtony/mule Examine the “mule-root” directory and its subdirectories in the host, which should now look like below. The files on the highlighted rows have been created by Mule. mule-root/ ├── apps │ ├── default │ │ └── mule-config.xml │ └── default-anchor.txt ├── conf │ ├── log4j.properties │ ├── tls-default.conf │ ├── tls-fips140-2.conf │ ├── wrapper-additional.conf │ └── wrapper.conf └── logs ├── mule-app-default.log ├── mule-domain-default.log └── mule.log Examine the “mule.log” file using the command “tail -f ~/mule-root/logs/mule.log”. There should be periodic output written to the log file similar to the following: DEBUG 2015-01-05 12:05:37,216 [Mule.app.deployer.monitor.1.thread.1] org.mule.module.launcher.DeploymentDirectoryWatcher: Checking for changes... DEBUG 2015-01-05 12:05:37,216 [Mule.app.deployer.monitor.1.thread.1] org.mule.module.launcher.DeploymentDirectoryWatcher: Current anchors: default-anchor.txt DEBUG 2015-01-05 12:05:37,216 [Mule.app.deployer.monitor.1.thread.1] org.mule.module.launcher.DeploymentDirectoryWatcher: Deleted anchors: Stop and remove the container: sudo docker stop [container id or name here] sudo docker rm [container id or name here] Direct Access to Docker Container Files When running Docker under the Ubuntu OS it is also possible to access the file-system of a Docker container from the host file-system. It may be possible to do this under other operating systems too, but I haven’t had the opportunity to test this. This technique may come in handy during development or testing with Docker containers for which you haven’t bound any volumes. Note! If given the choice to use either volume binding, as seen above, or direct access to container files as we will look at in this section for something more than a temporary file access, I would chose to use volume binding. Direct access to Docker container files relies on implementation details that I suspect may change in future versions of Docker if the developers find it suitable. With all that said, lets get the action started: Start a new Docker container: sudo docker run -d codingtony/mule Find the id of the newly launched Docker container: sudo docker ps Examine low-level information about the newly launched Docker container: sudo docker inspect [container id or name here] Output similar to this will be printed to the console (portions removed to conserve space): [{ "AppArmorProfile": "", "Args": [], "Config": { ... }, "Created": "2015-01-12T07:58:47.913905369Z", "Driver": "aufs", "ExecDriver": "native-0.2", "HostConfig": { ... }, "HostnamePath": "/var/lib/docker/containers/68b40def7ad6a7f819bd654d5627ad1c3a0f40c84e0fb0f875760f1bd6790eef/hostname", "HostsPath": "/var/lib/docker/containers/68b40def7ad6a7f819bd654d5627ad1c3a0f40c84e0fb0f875760f1bd6790eef/hosts", "Id": "68b40def7ad6a7f819bd654d5627ad1c3a0f40c84e0fb0f875760f1bd6790eef", "Image": "bcd0f37d48d4501ad64bae941d95446b157a6f15e31251e26918dbac542d731f", "MountLabel": "", "Name": "/thirsty_darwin", "NetworkSettings": { ... }, "Path": "/opt/mule/bin/mule", "ProcessLabel": "", "ResolvConfPath": "/var/lib/docker/containers/68b40def7ad6a7f819bd654d5627ad1c3a0f40c84e0fb0f875760f1bd6790eef/resolv.conf", "State": { ... }, "Volumes": {}, "VolumesRW": {} }] Locate the “Driver” node (highlighted in the above output) and ensure that its value is “aufs”. If it is not, you may need to modify the directory paths below replacing “aufs” with the value of this node. Personally I have only seen the “aufs” value at this node so anything else is uncharted territory to me. Copy the long hexadecimal value that can be found at the “Id” node (also highlighted in the above output). This is the long id of the Docker container. In a terminal window, issue the following command, inserting the long id of your container where noted: sudo ls -al /var/lib/docker/aufs/mnt/[long container id here] You are now looking at the root of the volume used by the Docker container you just launched. In the same terminal window, issue the following command: sudo ls -al /var/lib/docker/aufs/mnt/[long container id here]/opt The output from this command should look like this: total 12 drwxr-xr-x 4 root root 4096 jan 12 15:58 . drwxr-xr-x 75 root root 4096 jan 12 15:58 .. lrwxrwxrwx 1 root root 26 aug 10 04:19 mule -> /opt/mule-standalone-3.5.0 drwxr-xr-x 17 409 409 4096 jan 12 15:58 mule-standalone-3.5.0 Examine this line in the Dockerfile:RUN ln -s /opt/mule-standalone-3.5.0 /opt/muleWe see that a symbolic link is created and that the directory name and the name of the symbolic link matches the output we saw earlier. This matches the directory output in the previous step. To examine the Mule log file that we looked at when binding volumes earlier, use the following command: sudo cat /var/lib/docker/aufs/mnt/[long container id here]/opt/mule-standalone-3.5.0/logs/mule.log Next we create a new file in the Docker container using vi: sudo vi /var/lib/docker/aufs/mnt/[long container id here]/opt/mule-standalone-3.5.0/test.txt Enter some text into the new file by first pressing i and the type the text. When you are finished entering the text, press the Escape key and write the file to disk by typing the characters “:wq” without quotes. This writes the new contents of the file to disk and quits the editor. Leave the Docker container running after you are finished. In the next section, we are going to look at the file we just created from inside the Docker container. We have seen that we can examine the file system of a Docker container without binding volumes. It is also possible to copy or move files from the host file-system to the container’s file system using the regular commands. Root privileges are required both when examining and writing to the Docker container’s file system. Entering a Docker Container In order to verify that the file we just created in the host was indeed written to the Docker container, we are going to start a bash shell in the running Docker container and examine the location where the new file is expected to be located and the contents of the file. In the process we will see how we can execute commands in a Docker container from the host. Issue the command below in a terminal window. The exec Docker command is used to run a command, bash in this case, in a running Docker container. The -i flags tell Docker to keep the input stream open while the command is being executed. In this example, it allows us to enter commands into the bash shell running inside the Docker container. The -t flag cause Docker to allocate a text terminal to which the output from the command execution is printed. sudo docker exec -i -t [container id or name here] bash Note the prompt, which should change to [user]@[Docker container id]. In my case it looks like this: root@3ea374a280da:/# Go to the Mule installation directory using this command: cd /opt/mule-standalone-3.5.0/ Examine the contents of the directory: ls -al Among the other files, you should see the “test.txt” file: -rw-r--r-- 1 root root 53 Jan 14 03:19 test.txt Examine the contents of the “text.txt” file. The contents of the file should match what you entered earlier. cat text.txt Exit to the host OS: exit Stop and remove the container: sudo docker stop [container id or name here] sudo docker rm [container id or name here] We have seen that we can execute commands in a running Docker container. In this particular example, we used it to execute the bash shell and examine a file. I draw the conclusion that I should be able to set up a Docker image that contains a very controlled environment for some type of test and then create a container from that image and start the test from the host. Deploying a Mule Application In this section we will look at deploying a Mule application to an instance of the Mule ESB running in a Docker container. We will use volume binding, that we looked at in the section on files and Docker containers, to share directories in the host with the Docker container in order to make it easy to deploy applications, modify running applications, examine logs etc. Preparations Before deploying the application, we need to make some preparations: First of all, we restore the original log-level that we changed earlier. In this example, there will be log output when the applications we will deploy is run and we can limit the log generated by Mule. Edit the log4j.properties file in the “mule-root/conf” directory in the host and set the log-level on the last line in the file back to “INFO” and add one line, as in the listing below. The last three lines should now look like this: # Mule classes log4j.logger.org.mule=INFO log4j.logger.org.mule.tck.functional=DEBUG Next, we create the Mule application which we will deploy to the Mule ESB running in Docker: In some directory, create a file named “mule-deploy.properties” with the following contents: redeployment.enabled=true encoding=UTF-8 domain=default config.resources=HelloWorld.xml In the same directory create a file named “HelloWorld.xml”. This file contains the Mule configuration for our example application: Create a zip-archive named “mule-hello.zip” containing the two files created above: zip mule-hello.zip mule-deploy.properties HelloWorld.xml Deploy the Mule Application Before you start the Docker container in which the Mule EBS will run, make sure that you have created and prepared the directories in the host as described in the section Files and Docker Containers above. Start a new Mule Docker container using the command that we used when binding volumes: sudo docker run -d -v ~/mule-root/apps:/opt/mule/apps -v ~/mule-root/conf:/opt/mule/conf -v ~/mule-root/logs:/opt/mule/logs codingtony/mule As before, the -v option tells Docker to bind three directories in the host to three locations in the Docker container’s file system. Find the IP-address of the Docker container: sudo docker inspect [container id or name here] | grep IPAddress In my case, I see the following line which reveals the IP-address of the Docker container: “IPAddress”: “172.0.17.2”, Open a terminal window or tab and examine the Mule log. Leave this window or tab open during the exercise, in order to be able to verify the output from Mule. tail -f ~/mule-root/logs/mule.log Copy the zip-archive “mule-hello.zip” created earlier to the host directory ~/mule-root/apps/. Verify that the application has been deployed without errors in the Mule log: ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ + Started app 'mule-hello' + ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Leave the Docker container running after you are finished. In the next section we will look at how to access endpoints exposed by applications running in Docker containers. By binding directories in the host thus making them available in the Docker container, it becomes very simple to deploy Mule applications to an instance of Mule ESB running in a Docker container. I am considering this setup for a production environment as well, since it will enable me to perform backups of the directories containing Mule applications and configuration without having to access the Docker container’s file system. It is also in accord with the idea that a Docker container should be able to be quickly and easily restarted, which I feel it would not be if I had to deploy a number of Mule applications to it in order to recreate its previous state. Accessing Endpoints We now know that we can run the Mule ESB in a Docker container, we can deploy applications and examine the logs quite easily but one final, very important question remains to be answered; how to access endpoints exposed by applications running in a Docker container. This section assumes that the Mule application we deployed to Mule in the previous section is still running. In the host, open a web-browser and issue a request to the Docker container’s IP-address at port 8181. In my case, the URL is http://172.17.0.2:8181 Alternatively use the curl command in a terminal window. In my case I would write: curl 172.17.0.2:8181 The result should be a greeting in the following format: Hello World! It is now: 2015-01-14T07:39:03.942Z In addition, you should be able to see that a message was received in the Mule log. Now try the URL http://localhost:8181 You will get a message saying that the connection was refused, provided that you do not already have a service listening at that port. If you have another computer available that is connected to the same network as the host computer running Ubuntu, do the following: – Find the IP-address of the Ubuntu host computer using the ifconfigcommand. – In a web-browser on the other computer, try accessing port 8181 at the IP-address of the Ubuntu host computer. Again you will get a message saying that the connection was refused. Stop and remove the container: sudo docker stop [container id or name here] sudo docker rm [container id or name here] Without any particular measures taken, we see that we can access a service exposed in a Docker container from the Docker host but we did not succeed in accessing the service from another computer. To make a service exposed in a Docker container reachable from outside of the host, we need to tell Docker to publish a port from the Docker container to a port in the host using the -p flag: Launch a new Docker container using the following command: sudo docker run -d -p 8181:8181 -v ~/mule-root/apps:/opt/mule/apps -v ~/mule-root/conf:/opt/mule/conf -v ~/mule-root/logs:/opt/mule/logs codingtony/mule The added flag -p 8181:8181 makes the service exposed at port 8181 in the Docker container available at port 8181 in the host. Try accessing the URL http://localhost:8181 from a web-browser on the host computer.The result should be a greeting of the form we have seen earlier. Try accessing port 8181 at the IP-address of the Ubuntu host computer from another computer.This should also result in a greeting message. Stop and remove the container: sudo docker stop [container id or name here] sudo docker rm [container id or name here] Using the -p flag, we have seen that we can expose a service in a Docker container so that it becomes accessible from outside of the host computer. However, we also see that this information need to be supplied at the time of launching the Docker container. The conclusions that I draw from this is that: I can test and develop against a Mule ESB instance running in a Docker container without having to publish any ports, provided that my development computer is the Docker host computer. In a production environment or any other environment that need to expose services running in a Docker container to “the outside world” and where services will be added over time, I would consider deploying an Apache HTTP Server or NGINX on the Docker host computer and use it to proxy the services that are to be exposed. This way I can avoid re-launching the Docker container each time a new service is added and I can even (temporarily) redirect the proxy to some other computer if I need to perform some maintenance. Is There More? Of course! This article should only be considered an introduction and I am just a beginner with Docker. I hope I will have the time and inspiration to write more about Docker as I learn more.

In this post you can see an approach for conditionally enabling/disabling a set of checkboxes. For this we can use the ng-disabled directive and some CSS clases of typeclassName-true and className-false: ENABLE/DISABLE CHECKBOXES USING ANGULAR JS Select the maximum prize money: Select one prize money{{item.prizemoney} {{item.name}

An increasing number of systems are being built to handle the Volume, Velocity and Variety of Big Data, and hopefully help gain new insights and make better business decisions. Here, we will look at ways to deal with Big Data’s Volume and Velocity simultaneously, within a single architecture solution. Volume + Velocity Apache Hadoop provides both reliable storage (HDFS) and a processing system (MapReduce) for large data sets across clusters of computers. MapReduce is a batch query processor that is targeted at long-running background processes. Hadoop can handle Volume. But to handle Velocity, we need real-time processing tools that can compensate for the high-latency of batch systems, and serve the most recent data continuously, as new data arrives and older data is progressively integrated into the batch framework. Therefore we need both batch and real-time to run in parallel, and add a real-time computational system (e.g. Apache Storm) to our batch framework. This architectural combination of batch and real-time computation is referred to as a Lambda Architecture (λ). Generic Lambda λ has three layers: The Batch Layer manages the master data and precomputes the batch views The Speed Layer serves recent data only and increments the real-time views The Serving Layer is responsible for indexing and exposing the views so that they can be queried. The three layers are outlined in the below diagram along with a sample choice of technology stacks: Incoming data is dispatched to both Batch and Speed layers for processing. At the other end, queries are answered by merging both batch and real-time views. Note that real-time views are transient by nature and their data is discarded (making room for newer data) once propagated through the Batch and Serving layers. Most of the complexity is pushed onto the much smaller Speed layer where the results are only temporary, a process known as “complexity isolation“. We are indeed isolating the complexity of concurrent data updates in a layer that is regularly purged and kept small in size. λ is technology agnostic. The data pipeline is broken down into layers with clear demarcation of responsibilities, and at each layer, we can choose from a number of technologies. The Speed layer for instance could use either Apache Storm, or Apache Spark Streaming, or Spring “XD” ( eXtreme Data) etc. How do we recover from mistakes in λ ? Basically, we recompute the views. If that takes too long, we just revert to the previous, non-corrupted versions of our data. We can do that because of data immutability in the master dataset: data is never updated, only appended to (time-based ordering). The system is therefore Human Fault-Tolerant: if we write bad data, we can just remove that data altogether and recompute. Unified Lambda The downside of λ is its inherent complexity. Keeping in sync two already complex distributed systems is quite an implementation and maintenance challenge. People have started to look for simpler alternatives that would bring just about the same benefits and handle the full problem set. There are basically three approaches: 1) Adopt a pure streaming approach, and use a flexible framework such as Apache Samza to provide some type of batch processing. Although its distributed streaming layer is pluggable, Samza typically relies on Apache Kafka. Samza’s streams are replayable, ordered partitions. Samza can be configured for batching, i.e. consume several messages from the same stream partition in sequence. 2) Take the opposite approach, and choose a flexible Batch framework that would also allow micro-batches, small enough to be close to real-time, with Apache Spark/Spark Streaming or Storm’s Trident. Spark streaming is essentially a sequence of small batch processes that can reach latency as low as one second.Trident is a high-level abstraction on top of Storm that can process streams as small batches as well as do batch aggregation. 3) Use a technology stack already combining batch and real-time, such as Spring “XD”, Summingbird or Lambdoop. Summingbird (“Streaming MapReduce”) is a hybrid system where both batch/real-time workflows can be run at the same time and the results merged automatically.The Speed layer runs on Storm and the Batch layer on Hadoop, Lambdoop (Lambda-Hadoop, with HBase, Storm and Redis) also combines batch/real-time by offering a single API for both processing paradigms: The integrated approach (unified λ) seeks to handle Big Data’s Volume and Velocity by featuring a hybrid computation model, where both batch and real-time data processing are combined transparently. And with a unified framework, there would be only one system to learn, and one system to maintain.

In this tip you can see how to use the Angular Websocket module for connecting client applications to servers.

Posted by Gilad F on Back& Blog. Current approaches to web development rely upon having two kinds of intelligence built into your application – business intelligence in the server, and presentation intelligence on the client side. This institutes a clear delineation in responsibilities, which is often desirable from an architectural standpoint. However, this approach does have some drawbacks. Processing time for business logic, for example, is centralized on the server. This can introduce bottlenecks in the application’s performance, or add complexity when it comes to cross-server communication. For smaller applications that nonetheless have a large user base, this can often be the single greatest performance concern – the time spent computing solutions by the server. One way this can be offset is through the use of Object-Relational Mapping, or ORM. Below we’ll look at the concept of ORM, and how creating an ORM system in Angular can help make your application smarter. What is an ORM? Simply put, Object-Relational Mapping is the concept of creating representations of your underlying data that know how to manage themselves. Most web applications boil down to four basic actions, known as the “CRUD” approach – Create a record, Retrieve records, Update a record, or Delete a record. With an ORM, you simply encapsulate each of these functions within a class that represents a given record in the database. In essence, the objects you create to represent your data on the front end also know how to manipulate that data on the back end. Why Use an ORM? The primary benefit of an ORM is that it hides a lot of the functional complexity of database integration behind an established API. Communication with the database to implement each of the CRUD methods can be complex, but once it’s been accomplished for one model it can be easily ported to all of the other models in your system. An ORM focuses on hiding as much of this code as possible, allowing your models to care only about how they are represented – and how they interact with other elements in the system. A series of calls to establish a connection to the database, for example, becomes a single call to a method named “Save” on the model instance. This also allows you to centralize your database code, giving you only one location where you need to look for database-related bugs instead of having to search a complex code base for different custom data communication handlers. Why Use an ORM in Angular? While the JavaScript stack is particularly performant when compared to more heavyweight offerings such as Rails and Django, it still faces the issues common to the standard web application architecture – the server has the potential to be a bottleneck, handling the incoming traffic from a number of locations. By focusing your development efforts to create a pure CRUD API in your server, and developing a rudimentary ORM in Angular, you can offload a lot of that processing load to the client machines – in essence parallelizing the process at the expense of increased network communication. This allows you to reduce the overall dependence of your application on the server, making the server a “thin” client that simply updates the database based upon the API calls issued by the client. After a certain point, your back-end can be outsourced completely to an external provider that specializes in providing this type of access – such as Backand – allowing you to completely offload scalability and security concerns. In essence, it allows you to focus on your application as opposed to focusing on the attendant resources. Conclusion Object-Relational Mapping is a powerful paradigm that eases communication with a database for the basic CRUD activities associated with web applications. As most existing web development environments focus on implementing ORM on the server side, this can result in performance and communication bottlenecks – not to mention increased infrastructure costs. By offloading some of these ORM tasks to AngularJS, you can parallelize many of these tasks and reduce overall server load, in some cases obviating the need for the server entirely. If your application is facing a bloated back-end communication pattern, it might be worth your time to look at working towards implementation of a client-side ORM system. Build your Angular app and connect it to any database with Backand today. – Get started now.translate in hindi

In this article I am going to present a solution about sending SMS messages in Java for those developers and marketers who think – like me – that SMS is not dead.

I’ve lost count of the number of times I’ve seen code which fail-fast validates the state of something, using an approach like public class PersonValidator { public boolean validate(Person person) { boolean valid = person != null; if (valid) valid = person.givenName != null; if (valid) valid = person.familyName != null; if (valid) valid = person.age != null; if (valid) valid = person.gender != null; // ...and many more } } It works, but it’s a brute force approach that’s filled with repetition due to the valid check. If your code style enforces braces for if statements (+1 for that), your method is also three times longer and growing every time a new check is added to the validator. Using Java 8’s new stream API, we can improve this by taking the guard condition of if (valid) and making a generic validator that handles the plumbing for you. import java.util.LinkedList; import java.util.List; import java.util.function.Predicate; public class GenericValidator implements Predicate { private final List> validators = new LinkedList<>(); public GenericValidator(List> validators) { this.validators.addAll(validators); } @Override public boolean test(final T toValidate) { return validators.parallelStream() .allMatch(predicate -> predicate.test(toValidate)); } } Using this, we can rewrite the Person validator to be a specification of the required validations. public class PersonValidator extends GenericValidator { private static final List> VALIDATORS = new LinkedList<>(); static { VALIDATORS.add(person -> person.givenName != null); VALIDATORS.add(person -> person.familyName != null); VALIDATORS.add(person -> person.age != null); VALIDATORS.add(person -> person.gender != null); // ...and many more } public PersonValidator() { super(VALIDATORS); } } PersonValidator, and all your other validators, can now focus completely on validation. The behaviour hasn’t changed – the validation still fails fast. There’s no boiler plate, which is A Good Thing. This one’s going in the toolbox.

If you have a requirement to save and serve files, then there are at least a couple options. Save the file onto the server and serve it from there. Mongo1 provide GridFS2 store that allows you not only to store files but also metadata related to the file. For example: you can store author, tags, group etc right with the file. You can provide this functionality via option 1 too, but you would need to make your own tables and link the files to the metadata information. Besides replication of data is in built in Mongo. Bottle You can upload and download mongo files using Bottle3 like so: import json from bottle import run, Bottle, request, response from gridfs import GridFS from pymongo import MongoClient FILE_API = Bottle() MONGO_CLIENT = MongoClient('mongodb://localhost:27017/') DB = MONGO_CLIENT['TestDB'] GRID_FS = GridFS(DB) @FILE_API.put('/upload/< file_name>') def upload(file_name): response.content_type = 'application/json' with GRID_FS.new_file(filename=file_name) as fp: fp.write(request.body) file_id = fp._id if GRID_FS.find_one(file_id) is not None: return json.dumps({'status': 'File saved successfully'}) else: response.status = 500 return json.dumps({'status': 'Error occurred while saving file.'}) @FILE_API.get('/download/< file_name>') def index(file_name): grid_fs_file = GRID_FS.find_one({'filename': file_name}) response.headers['Content-Type'] = 'application/octet-stream' response.headers["Content-Disposition"] = "attachment; filename={}".format(file_name) return grid_fs_file run(app=FILE_API, host='localhost', port=8080) And here's the break down of the code: Upload method: Line 12: Sets up upload method to recieve a PUT request for /upload/ url. Line 15-17: Create a new GridFS file with file_name and get the content from request.body. request.body may be StringIO type or a File type because Python is smart enough to decipher the body type based on the content. Line 18-19: If we can find the file by file name then it was saved successfully and therefore return a success response. Line 20-22: Return error if file was not saved successfully. Download method: Line 27: Find the GridFS file. Line 28-29: Set the response Content-Type as application-octet-stream and Content-Disposition to attachment; filename= Line 31: Return the GridOut object. Based on Bottle documentation below we can return an object which has .read() method available and Bottle understands that to be a File object. File objects Everything that has a .read() method is treated as a file or file-like object and passed to the wsgi.file_wrapper callable defined by the WSGI server framework. Some WSGI server implementations can make use of optimized system calls (sendfile) to transmit files more efficiently. In other cases this just iterates over chunks that fit into memory. And we are done (as far as Bottle is concerned). Flask You can upload/download files using Flask4 like so: import json from gridfs import GridFS from pymongo import MongoClient from flask import Flask, make_response from flask import request __author__ = 'ravihasija' app = Flask(__name__) mongo_client = MongoClient('mongodb://localhost:27017/') db = mongo_client['TestDB'] grid_fs = GridFS(db) @app.route('/upload/', methods=['PUT']) def upload(file_name): with grid_fs.new_file(filename=file_name) as fp: fp.write(request.data) file_id = fp._id if grid_fs.find_one(file_id) is not None: return json.dumps({'status': 'File saved successfully'}), 200 else: return json.dumps({'status': 'Error occurred while saving file.'}), 500 @app.route('/download/') def index(file_name): grid_fs_file = grid_fs.find_one({'filename': file_name}) response = make_response(grid_fs_file.read()) response.headers['Content-Type'] = 'application/octet-stream' response.headers["Content-Disposition"] = "attachment; filename={}".format(file_name) return response app.run(host="localhost", port=8081) The Flask upload and download code is very similar to Bottle. It differs only in a few places detailed below: Line 14: Routing is configured differently in Flask. You mention the URL and the methods that apply for that URL. Line 17: Instead of request.body you use request.data Line 28-31: Make the response with the file content and set up the appropriate headers. Finally, return the response object. Questions? Thoughts? Please feel free to leave me a comment below. Thank you for your time. Github repo: https://github.com/RaviH/file-upload-download-mongo References: MongoDB: http://www.mongodb.org/↩ GridFS: http://docs.mongodb.org/manual/core/gridfs/↩ Bottle: http://bottlepy.org/docs/dev/tutorial.html↩ Flask: http://flask.pocoo.org/↩ PyMongo GridFS doc http://api.mongodb.org/python/current/api/gridfs/index.html?highlight=gridfs#module-gridfs↩ Get to know GridFS: https://dzone.com/articles/get-know-gridfs-mongodb↩

Traditional web development builds a bridge between the front end, where the user performs their manipulations of the application’s data, and the back end, where that data is stored. In traditional web development, this process is driven by successive networking calls, communicating changes between the server and the client via re-rendering the involved pages. AngularJS enhances this with two-way data binding. Below we’ll look at what two-way data binding is, and how it differs from the traditional data processing approach. The Traditional Approach Most web frameworks focus on one-way data binding. This involves reading the input from the DOM, serializing the data, sending it to the server, waiting for the process to finish, then modifying the DOM to indicate any errors, or reloading the DOM if the call is successful. While this provides a traditional web application all the time it needs to perform data processing, this benefit is only really applicable to web apps with highly complex data structures. If your application has a simpler data format, with relatively flat models, then the extra work can needlessly complicate the process. Furthermore, all models need to wait for server confirmation before their data can be updated, meaning that related data depending upon those models won’t have the latest information. Tying Together the UI and the Model AngularJS addresses this with two-way data binding. With two-way data binding, the user interface changes are immediately reflected in the underlying data model, and vice-versa. This allows the data model to serve as an atomic unit that the view of the application can always depend upon to be accurate. Many web frameworks implement this type of data binding with a complex series of event listeners and event handlers – an approach that can quickly become fragile. AngularJS, on the other hand, makes this approach to data a primary part of its architecture. Instead of creating a series of callbacks to handle the changing data, AngularJS does this automatically without any needed intervention by the programmer Benefits and Considerations The primary benefit of two-way data binding is that updates to (and retrievals from) the underlying data store happen more or less automatically. When the data store updates, the UI updates as well. This allows you to remove a lot of logic from the front-end display code, particularly when making effective use of AngularJS’s declarative approach to UI presentation. In essence, it allows for true data encapsulation on the front-end, reducing the need to do complex and destructive manipulation of the DOM. While this solves a lot of problems with a website’s presentation architecture, there are some disadvantages to take into consideration. First, AngularJS uses a dirty-checking approach that can be slow in some browsers – not a problem for thin presentation pages, but any page with heavy processing may run into problems in older browsers. Additionally, two-way binding is only truly beneficial for relatively simple objects. Any data that requires heavy parsing work, or extensive manipulation and processing, will simply not work well with two-way binding. Additionally, some uses of Angular – such as using the same binding directive more than once – can break the data binding process. Conclusion While the traditional approach to data binding has a lot of benefits when it comes to performing complex data manipulations and calculations, it can introduce some problems with respect to the design of the web application’s front-end architecture. With AngularJS’s use of two-way data binding, your application can greatly simplify its presentation layer, allowing the UI to be built off of a cleaner, less-destructive approach to DOM presentation. While it isn’t useful in every situation, the two-way data binding AngularJS provides can greatly ease web application development, and reduce the pain faced by your front-end developers. Build your Angular app and connect it to any database with Backand today. – Get started now.