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How-To Tutorials

7019 Articles
article-image-managing-adobe-connect-meeting-room
Packt
04 Sep 2013
6 min read
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Managing Adobe Connect Meeting Room

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

Packt
04 Sep 2013
5 min read
(For more resources related to this topic, see here.) Getting ready For this article we will use the sample application virtual-destinations to demonstrate the use of Virtual Destinations. How to do it... To run the sample for this article, open a terminal, change the path to point to the directory where virtual-destinations is located, and run it by typing mvn compile exec:java. In the terminal where you started the example, you will see output similar to the following snippet, indicating that the application is running: Starting Virtual Destination example now... Queue A Consumer 1 processed 500 Messages Queue A Consumer 2 processed 500 Messages Queue B Consumer processed 1000 Messages Finished running the Virtual Destination example. How it works... Our example takes advantage of ActiveMQ's Virtual Destination feature to allow a JMS Producer to send messages to a topic and have those messages be received by a number of queue consumers. Let's take a look at the sent code first: private void sendMessages() throws Exception { Connection connection = connectionFactory.createConnection(); Session session = connection.createSession(false, Session.AUTO_ACKNOWLEDGE); Destination destination = session.createTopic("VirtualTopic.Foo"); MessageProducer producer = session.createProducer(destination); for (int i = 0; i < 1000; ++i) { producer.send(session.createMessage()); } connection.close(); } As we can see, there's not much new here; we are just using standard JMS API code to send our messages. The important part is in the name of the topic destination VirtualTopic.Foo, which informs ActiveMQ Broker we want this topic to be one of those special Virtual Destinations. Now let's take a look at the consumer code, and then we'll try and make sense of how this works: Queue queueA = receiverSession.createQueue("Consumer.A. VirtualTopic.Foo"); VirtualMessageListener listenerA1 = new VirtualMessageListener(done); MessageConsumer consumerA1 = receiverSession.createConsumer(queueA); consumerA1.setMessageListener(listenerA1); VirtualMessageListener listenerA2 = new VirtualMessageListener(done); MessageConsumer consumerA2 = receiverSession.createConsumer(queueA); consumerA2.setMessageListener(listenerA2);  Queue queueB = receiverSession.createQueue("Consumer.B. VirtualTopic.Foo"); VirtualMessageListener listenerB = new VirtualMessageListener(done); MessageConsumer consumerB = receiverSession.createConsumer(queueB); consumerB.setMessageListener(listenerB); As we can see, the consumer code is also not very mysterious; we create two consumers on the Consumer.A.VirtualTopic.Foo queue and one on the Consumer.B.VirtualTopic.Foo queue, and yet when we run the example, the two consumers on queue A split 500 of the topic message and the consumer on queue B gets its own copy of the 1,000 messages. So what's happening on the broker then to make this happen? By default, whenever a topic is created with a name matching the pattern VirtualTopic.<TopicName>, we gain access to the feature known as Virtual Destinations. These Virtual Destinations allow us to send a message to a topic but create consumers that have all the benefits of queue consumers, namely, those of load balancing and message persistence. Our queue consumers just need to use their own naming convention to access the Virtual Destination functionality. Each queue that we want to create must be named using the pattern Consumer.<Name>.VirtualTopic.<TopicName>. In the following figure we can see a visual representation of the message flow in our example: As messages are sent to our example's topic, they are distributed to both the queues we created. Two of our consumers split the work of processing messages from queue A, while the other consumer gets all the messages from queue B. In our example we didn't attach a listener to the topic itself, but we could have done so and it would also have received all the messages we sent. It's easy to imagine using the topic as a way to monitor the messages that are being distributed to the queue consumers in this scenario; this pattern is often referred to as a wiretap. If it isn't already apparent, Virtual Destinations are a very powerful feature for your messaging applications. Topics are great for broadcasting events but when we need to be able to consume messages that were sent while a client was offline, the only recourse is to use a durable topic subscription. Unfortunately, durable subscriptions have a number of limitations, the least of which is that only one JMS connection can be actively subscribed to a logical topic subscription. This means that we can't load balance messages and we can't have fast failover if a subscriber goes down. Virtual Destinations solve these problems since all of our consumers subscribe to a queue so their messages are persistent, and the work of processing the messages on the queue can be shared by more than one active subscription. There's more... The naming pattern for Virtual Destinations is not set in stone. By default, ActiveMQ is configured to use the pattern we used in our sample application, but you can easily change this. In the following XML snippet, we configured our broker to all topics on our broker into virtual topics. We use the wildcard syntax here, which matches every topic name a client sends messages to: <broker> <destinationInterceptors> <virtualDestinationInterceptor> <virtualDestinations><virtualTopic name=">" prefix="VirtualTopic Consumers.*."/> </virtualDestinations></virtualDestinationInterceptor> </destinationInterceptors> </broker> More information on Virtual Destinations can be found on the ActiveMQ website, http://activemq.apache.org/virtual-destinations.html. Summary This article thus explained what Virtual Destinations are and how to use them to avoid the limitations of JMS's durable topic subscriptions. Resources for Article: Further resources on this subject: Routing to an external ActiveMQ broker [Article] So, what is Apache Wicket? [Article] Apache Geronimo Logging [Article]
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article-image-integrating-storm-and-hadoop
Packt
04 Sep 2013
17 min read
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Integrating Storm and Hadoop

Packt
04 Sep 2013
17 min read
(For more resources related to this topic, see here.) In this article, we will implement the Batch and Service layers to complete the architecture. There are some key concepts underlying this big data architecture: Immutable state Abstraction and composition Constrain complexity Immutable state is the key, in that it provides true fault-tolerance for the architecture. If a failure is experienced at any level, we can always rebuild the data from the original immutable data. This is in contrast to many existing data systems, where the paradigm is to act on mutable data. This approach may seem simple and logical; however, it exposes the system to a particular kind of risk in which the state is lost or corrupted. It also constrains the system, in that you can only work with the current view of the data; it isn't possible to derive new views of the data. When the architecture is based on a fundamentally immutable state, it becomes both flexible and fault-tolerant. Abstractions allow us to remove complexity in some cases, and in others they can introduce complexity. It is important to achieve an appropriate set of abstractions that increase our productivity and remove complexity, but at an appropriate cost. It must be noted that all abstractions leak, meaning that when failures occur at a lower abstraction, they will affect the higher-level abstractions. It is therefore often important to be able to make changes within the various layers and understand more than one layer of abstraction. The designs we choose to implement our abstractions must therefore not prevent us from reasoning about or working at the lower levels of abstraction when required. Open source projects are often good at this, because of the obvious access to the code of the lower level abstractions, but even with source code available, it is easy to convolute the abstraction to the extent that it becomes a risk. In a big data solution, we have to work at higher levels of abstraction in order to be productive and deal with the massive complexity, so we need to choose our abstractions carefully. In the case of Storm, Trident represents an appropriate abstraction for dealing with the data-processing complexity, but the lower level Storm API on which Trident is based isn't hidden from us. We are therefore able to easily reason about Trident based on an understanding of lower-level abstractions within Storm. Another key issue to consider when dealing with complexity and productivity is composition. Composition within a given layer of abstraction allows us to quickly build out a solution that is well tested and easy to reason about. Composition is fundamentally decoupled, while abstraction contains some inherent coupling to the lower-level abstractions—something that we need to be aware of. Finally, a big data solution needs to constrain complexity. Complexity always equates to risk and cost in the long run, both from a development perspective and from an operational perspective. Real-time solutions will always be more complex than batch-based systems; they also lack some of the qualities we require in terms of performance. Nathan Marz's Lambda architecture attempts to address this by combining the qualities of each type of system to constrain complexity and deliver a truly fault-tolerant architecture. We divided this flow into preprocessing and "at time" phases, using streams and DRPC streams respectively. We also introduced time windows that allowed us to segment the preprocessed data. In this article, we complete the entire architecture by implementing the Batch and Service layers. The Service layer is simply a store of a view of the data. In this case, we will store this view in Cassandra, as it is a convenient place to access the state alongside Trident's state. The preprocessed view is identical to the preprocessed view created by Trident, counted elements of the TF-IDF formula (D, DF, and TF), but in the batch case, the dataset is much larger, as it includes the entire history. The Batch layer is implemented in Hadoop using MapReduce to calculate the preprocessed view of the data. MapReduce is extremely powerful, but like the lower-level Storm API, is potentially too low-level for the problem at hand for the following reasons: We need to describe the problem as a data pipeline; MapReduce isn't congruent with such a way of thinking Productivity We would like to think of a data pipeline in terms of streams of data, tuples within the stream and predicates acting on those tuples. This allows us to easily describe a solution to a data processing problem, but it also promotes composability, in that predicates are fundamentally composable, but pipelines themselves can also be composed to form larger, more complex pipelines. Cascading provides such an abstraction for MapReduce in the same way as Trident does for Storm. With these tools, approaches, and considerations in place, we can now complete our real-time big data architecture. There are a number of elements, that we will update, and a number of elements that we will add. The following figure illustrates the final architecture, where the elements in light grey will be updated from the existing recipe, and the elements in dark grey will be added in this article: Implementing TF-IDF in Hadoop TF-IDF is a well-known problem in the MapReduce communities; it is well-documented and implemented, and it is interesting in that it is sufficiently complex to be useful and instructive at the same time. Cascading has a series of tutorials on TF-IDF at http://www.cascading.org/2012/07/31/cascading-for-the-impatient-part-5/, which documents this implementation well. For this recipe, we shall use a Clojure Domain Specific Language (DSL) called Cascalog that is implemented on top of Cascading. Cascalog has been chosen because it provides a set of abstractions that are very semantically similar to the Trident API and are very terse while still remaining very readable and easy to understand. Getting ready Before you begin, please ensure that you have installed Hadoop by following the instructions at http://www.michael-noll.com/tutorials/running-hadoop-on-ubuntu-linux-single-node-cluster/. How to do it… Start by creating the project using the lein command: lein new tfidf-cascalog Next, you need to edit the project.clj file to include the dependencies: (defproject tfidf-cascalog "0.1.0-SNAPSHOT" :dependencies [[org.clojure/clojure "1.4.0"] [cascalog "1.10.1"] [org.apache.cassandra/cassandra-all "1.1.5"] [clojurewerkz/cassaforte "1.0.0-beta11-SNAPSHOT"] [quintona/cascading-cassandra "0.0.7-SNAPSHOT"] [clj-time "0.5.0"] [cascading.avro/avro-scheme "2.2-SNAPSHOT"] [cascalog-more-taps "0.3.0"] [org.apache.httpcomponents/httpclient "4.2.3"]] :profiles{:dev{:dependencies[[org.apache.hadoop/hadoop-core "0.20.2-dev"] [lein-midje "3.0.1"] [cascalog/midje-cascalog "1.10.1"]]}}) It is always a good idea to validate your dependencies; to do this, execute lein deps and review any errors. In this particular case, cascading-cassandra has not been deployed to clojars, and so you will receive an error message. Simply download the source from https://github.com/quintona/cascading-cassandra and install it into your local repository using Maven. It is also good practice to understand your dependency tree. This is important to not only prevent duplicate classpath issues, but also to understand what licenses you are subject to. To do this, simply run lein pom, followed by mvn dependency:tree. You can then review the tree for conflicts. In this particular case, you will notice that there are two conflicting versions of Avro. You can fix this by adding the appropriate exclusions: [org.apache.cassandra/cassandra-all "1.1.5" :exclusions [org.apache.cassandra.deps/avro]] We then need to create the Clojure-based Cascade queries that will process the document data. We first need to create the query that will create the "D" view of the data; that is, the D portion of the TF-IDF function. This is achieved by defining a Cascalog function that will output a key and a value, which is composed of a set of predicates: (defn D [src] (let [src (select-fields src ["?doc-id"])] (<- [?key ?d-str] (src ?doc-id) (c/distinct-count ?doc-id :> ?n-docs) (str "twitter" :> ?key) (str ?n-docs :> ?d-str)))) You can define this and any of the following functions in the REPL, or add them to core.clj in your project. If you want to use the REPL, simply use lein repl from within the project folder. The required namespace (the use statement), require, and import definitions can be found in the source code bundle. We then need to add similar functions to calculate the TF and DF values: (defn DF [src] (<- [?key ?df-count-str] (src ?doc-id ?time ?df-word) (c/distinct-count ?doc-id ?df-word :> ?df-count) (str ?df-word :> ?key) (str ?df-count :> ?df-count-str))) (defn TF [src] (<- [?key ?tf-count-str] (src ?doc-id ?time ?tf-word) (c/count ?tf-count) (str ?doc-id ?tf-word :> ?key) (str ?tf-count :> ?tf-count-str))) This Batch layer is only interested in calculating views for all the data leading up to, but not including, the current hour. This is because the data for the current hour will be provided by Trident when it merges this batch view with the view it has calculated. In order to achieve this, we need to filter out all the records that are within the current hour. The following function makes that possible: (deffilterop timing-correct? [doc-time] (let [now (local-now) interval (in-minutes (interval (from-long doc-time) now))] (if (< interval 60) false true)) Each of the preceding query definitions require a clean stream of words. The text contained in the source documents isn't clean. It still contains stop words. In order to filter these and emit a clean set of words for these queries, we can compose a function that splits the text into words and filters them based on a list of stop words and the time function defined previously: (defn etl-docs-gen [rain stop] (<- [?doc-id ?time ?word] (rain ?doc-id ?time ?line) (split ?line :> ?word-dirty) ((c/comp s/trim s/lower-case) ?word-dirty :> ?word) (stop ?word :> false) (timing-correct? ?time))) We will be storing the outputs from our queries to Cassandra, which requires us to define a set of taps for these views: (defn create-tap [rowkey cassandra-ip] (let [keyspace storm_keyspace column-family "tfidfbatch" scheme (CassandraScheme. cassandra-ip "9160" keyspace column-family rowkey {"cassandra.inputPartitioner""org.apache.cassandra.dht.RandomPartitioner" "cassandra.outputPartitioner" "org.apache.cassandra.dht.RandomPartitioner"}) tap (CassandraTap. scheme)] tap)) (defn create-d-tap [cassandra-ip] (create-tap "d"cassandra-ip)) (defn create-df-tap [cassandra-ip] (create-tap "df" cassandra-ip)) (defn create-tf-tap [cassandra-ip] (create-tap "tf" cassandra-ip)) The way this schema is created means that it will use a static row key and persist name-value pairs from the tuples as column:value within that row. This is congruent with the approach used by the Trident Cassandra adaptor. This is a convenient approach, as it will make our lives easier later. We can complete the implementation by a providing a function that ties everything together and executes the queries: (defn execute [in stop cassandra-ip] (cc/connect! cassandra-ip) (sch/set-keyspace storm_keyspace) (let [input (tap/hfs-tap (AvroScheme. (load-schema)) in) stop (hfs-delimited stop :skip-header? true) src (etl-docs-gen input stop)] (?- (create-d-tap cassandra-ip) (D src)) (?- (create-df-tap cassandra-ip) (DF src)) (?- (create-tf-tap cassandra-ip) (TF src)))) Next, we need to get some data to test with. I have created some test data, which is available at https://bitbucket.org/qanderson/tfidf-cascalog. Simply download the project and copy the contents of src/data to the data folder in your project structure. We can now test this entire implementation. To do this, we need to insert the data into Hadoop: hadoop fs -copyFromLocal ./data/document.avro data/document.avro hadoop fs -copyFromLocal ./data/en.stop data/en.stop Then launch the execution from the REPL: => (execute "data/document" "data/en.stop" "127.0.0.1") How it works… There are many excellent guides on the Cascalog wiki (https://github.com/nathanmarz/cascalog/wiki), but for completeness's sake, the nature of a Cascalog query will be explained here. Before that, however, a revision of Cascading pipelines is required. The following is quoted from the Cascading documentation (http://docs.cascading.org/cascading/2.1/userguide/htmlsingle/): Pipe assemblies define what work should be done against tuple streams, which are read from tap sources and written to tap sinks. The work performed on the data stream may include actions such as filtering, transforming, organizing, and calculating. Pipe assemblies may use multiple sources and multiple sinks, and may define splits, merges, and joins to manipulate the tuple streams. This concept is embodied in Cascalog through the definition of queries. A query takes a set of inputs and applies a list of predicates across the fields in each tuple of the input stream. Queries are composed through the application of many predicates. Queries can also be composed to form larger, more complex queries. In either event, these queries are reduced down into a Cascading pipeline. Cascalog therefore provides an extremely terse and powerful abstraction on top of Cascading; moreover, it enables an excellent development workflow through the REPL. Queries can be easily composed and executed against smaller representative datasets within the REPL, providing the idiomatic API and development workflow that makes Clojure beautiful. If we unpack the query we defined for TF, we will find the following code: (defn DF [src] (<- [?key ?df-count-str] (src ?doc-id ?time ?df-word) (c/distinct-count ?doc-id ?df-word :> ?df-count) (str ?df-word :> ?key) (str ?df-count :> ?df-count-str))) The <- macro defines a query, but does not execute it. The initial vector, [?key ?df-count-str], defines the output fields, which is followed by a list of predicate functions. Each predicate can be one of the following three types: Generators: A source of data where the underlying source is either a tap or another query. Operations: Implicit relations that take in input variables defined elsewhere and either act as a function that binds new variables or a filter. Operations typically act within the scope of a single tuple. Aggregators: Functions that act across tuples to create aggregate representations of data. For example, count and sum. The :> keyword is used to separate input variables from output variables. If no :> keyword is specified, the variables are considered as input variables for operations and output variables for generators and aggregators. The (src ?doc-id ?time ?df-word) predicate function names the first three values within the input tuple, whose names are applicable within the query scope. Therefore, if the tuple ("doc1" 123324 "This") arrives in this query, the variables would effectively bind as follows: ?doc-id: "doc1" ?time: 123324 ?df-word: "This" Each predicate within the scope of the query can use any bound value or add new bound variables to the scope of the query. The final set of bound values that are emitted is defined by the output vector. We defined three queries, each calculating a portion of the value required for the TF-IDF algorithm. These are fed from two single taps, which are files stored in the Hadoop filesystem. The document file is stored using Apache Avro, which provides a high-performance and dynamic serialization layer. Avro takes a record definition and enables serialization/deserialization based on it. The record structure, in this case, is for a document and is defined as follows: {"namespace": "storm.cookbook", "type": "record", "name": "Document", "fields": [ {"name": "docid", "type": "string"}, {"name": "time", "type": "long"}, {"name": "line", "type": "string"} ] } Both the stop words and documents are fed through an ETL function that emits a clean set of words that have been filtered. The words are derived by splitting the line field using a regular expression: (defmapcatop split [line] (s/split line #"[[](),.)s]+")) The ETL function is also a query, which serves as a source for our downstream queries, and defines the [?doc-id ?time ?word] output fields. The output tap, or sink, is based on the Cassandra scheme. A query defines predicate logic, not the source and destination of data. The sink ensures that the outputs of our queries are sent to Cassandra. The ?- macro executes a query, and it is only at execution time that a query is bound to its source and destination, again allowing for extreme levels of composition. The following, therefore, executes the TF query and outputs to Cassandra: (?- (create-tf-tap cassandra-ip) (TF src)) There's more… The Avro test data was created using the test data from the Cascading tutorial at http://www.cascading.org/2012/07/31/cascading-for-the-impatient-part-5/. Within this tutorial is the rain.txt tab-separated data file. A new column was created called time that holds the Unix epoc time in milliseconds. The updated text file was then processed using some basic Java code that leverages Avro: Schema schema = Schema.parse(SandboxMain.class.getResourceAsStream("/document.avsc")); File file = new File("document.avro"); DatumWriter<GenericRecord> datumWriter = new GenericDatumWriter<GenericRecord>(schema); DataFileWriter<GenericRecord> dataFileWriter = new DataFileWriter<GenericRecord>(datumWriter); dataFileWriter.create(schema, file); BufferedReader reader = new BufferedReader(new InputStreamReader(SandboxMain.class.getResourceAsStream("/rain.txt"))); String line = null; try { while ((line = reader.readLine()) != null) { String[] tokens = line.split("t"); GenericRecord docEntry = new GenericData.Record(schema); docEntry.put("docid", tokens[0]); docEntry.put("time", Long.parseLong(tokens[1])); docEntry.put("line", tokens[2]); dataFileWriter.append(docEntry); } } catch (IOException e) { e.printStackTrace(); } dataFileWriter.close(); Persisting documents from Storm In the previous recipe, we looked at deriving precomputed views of our data taking some immutable data as the source. In that recipe, we used statically created data. In an operational system, we need Storm to store the immutable data into Hadoop so that it can be used in any preprocessing that is required. How to do it… As each tuple is processed in Storm, we must generate an Avro record based on the document record definition and append it to the data file within the Hadoop filesystem. We must create a Trident function that takes each document tuple and stores the associated Avro record. Within the tfidf-topology project created in, inside the storm.cookbook.tfidf.function package, create a new class named PersistDocumentFunction that extends BaseFunction. Within the prepare function, initialize the Avro schema and document writer: public void prepare(Map conf, TridentOperationContext context) { try { String path = (String) conf.get("DOCUMENT_PATH"); schema = Schema.parse(PersistDocumentFunction.class .getResourceAsStream("/document.avsc")); File file = new File(path); DatumWriter<GenericRecord> datumWriter = new GenericDatumWriter<GenericRecord>(schema); dataFileWriter = new DataFileWriter<GenericRecord>(datumWriter); if(file.exists()) dataFileWriter.appendTo(file); else dataFileWriter.create(schema, file); } catch (IOException e) { throw new RuntimeException(e); } } As each tuple is received, coerce it into an Avro record and add it to the file: public void execute(TridentTuple tuple, TridentCollector collector) { GenericRecord docEntry = new GenericData.Record(schema); docEntry.put("docid", tuple.getStringByField("documentId")); docEntry.put("time", Time.currentTimeMillis()); docEntry.put("line", tuple.getStringByField("document")); try { dataFileWriter.append(docEntry); dataFileWriter.flush(); } catch (IOException e) { LOG.error("Error writing to document record: " + e); throw new RuntimeException(e); } } Next, edit the TermTopology.build topology and add the function to the document stream: documentStream.each(new Fields("documentId","document"), new PersistDocumentFunction(), new Fields()); Finally, include the document path into the topology configuration: conf.put("DOCUMENT_PATH", "document.avro"); How it works… There are various logical streams within the topology, and certainly the input for the topology is not in the appropriate state for the recipes in this article containing only URLs. We therefore need to select the correct stream from which to consume tuples, coerce these into Avro records, and serialize them into a file. The previous recipe will then periodically consume this file. Within the context of the topology definition, include the following code: Stream documentStream = getUrlStream(topology, spout) .each(new Fields("url"), new DocumentFetchFunction(mimeTypes), new Fields("document", "documentId", "source")); documentStream.each(new Fields("documentId","document"), new PersistDocumentFunction(), new Fields()); The function should consume tuples from the document stream whose tuples are populated with already fetched documents.
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article-image-learning-musescore
Packt
04 Sep 2013
15 min read
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Learning MuseScore

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

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

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

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

Packt
04 Sep 2013
7 min read
(For more resources related to this topic, see here.) Wicket is a component-based Java web framework that uses just Java and HTML. Here, you will see the main advantages of using Apache Wicket in your projects. Using Wicket, you will not have mutant HTML pages. Most of the Java web frameworks require the insertion of special syntax to the HTML code, making it more difficult for Web designers. On the other hand, Wicket adopts HTML templates by using a namespace that follows the XHTML standard. It consists of an id attribute in the Wicket namespace (wicket:id). You won't need scripts to generate messy HTML code. Using Wicket, the code will be clearer, and refactoring and navigating within the code will be easier. Moreover, you can utilize any HTML editor to edit the HTML files, and web designers can work with little knowledge of Wicket in the presentation layer without worrying about business rules and other developer concerns. The advantages for developers are as follows: All code is written in Java No XML configuration files POJO-centric programming No Back-button problems (that is, unexpected and undesirable results on clicking on the browser's Back button) Ease of creating bookmarkable pages Great compile-time and runtime problem diagnosis Easy testability of components Another interesting thing is that concepts such as generics and anonymous subclasses are widely used in Wicket, leveraging the Java programming language to the max. Wicket is based on components. A component is an object that interacts with other components and encapsulates a set of functionalities. Each component should be reusable, replaceable, extensible, encapsulated, and independent, and it does not have a specific context. Wicket provides all these principles to developers because it has been designed taking into account all of them. In particular, the most remarkable principle is reusability. Developers can create custom reusable components in a straightforward way. For instance, you could create a custom component called SearchPanel (by extending the Panel class, which is also a component) and use it in all your other Wicket projects. Wicket has many other interesting features. Wicket also aims to make the interaction of the stateful server-side Java programming language with the stateless HTTP protocol more natural. Wicket's code is safe by default. For instance, it does not encode state in URLs. Wicket is also efficient (for example, it is possible to do a tuning of page-state replication) and scalable (Wicket applications can easily work on a cluster). Last, but not least, Wicket has support for frameworks like EJB and Spring. Installation In seven easy steps, you can build a Wicket "Hello World" application. Step 1 – what do I need? Before you start to use Apache Wicket 6, you will need to check if you have all of the required elements, listed as follows: Wicket is a Java framework, so you need to have Java virtual machine (at least Version 6) installed on your machine. Apache Maven is required. Maven is a tool that can be used for building and managing Java projects. Its main purpose is to make the development process easier and more structured. More information on how to install and configure Maven can be found at http://maven.apache.org. The examples of this book use the Eclipse IDE Juno version, but you can also use other versions or other IDEs, such as NetBeans. In case you are using other versions, check the link for installing the plugins to the version you have; the remaining steps will be the same. In case of other IDEs, you will need to follow some tutorial to install other equivalent plugins or not use them at all. Step 2 – installing the m2eclipse plugin The steps for installing the m2eclipse plugin are as follows: Go to Help | Install New Software. Click on Add and type in m2eclipse in the Name field; copy and paste the link https://repository.sonatype.org/content/repositories/forge-sites/m2e/1.3.0/N/LATEST onto the Location field. Check all options and click on Next. Conclude the installation of the m2eclipse plugin by accepting all agreements and clicking on Finish. Step 3 – creating a new Maven application The steps for creating a new Maven application are as follows: Go to File | New | Project. Then go to Maven | Maven Project. Click on Next and type wicket in the next form. Choose the wicket-archetype-quickstart maven Archetype and click on Next. Fill the next form according to the following screenshot and click on Finish: Step 4 – coding the "Hello World" program In this step, we will build the famous "Hello World" program. The separation of concerns will be clear between HTML and Java code. In this example, and in most cases, each HTML file has a corresponding Java class (with the same name). First, we will analyse the HTML template code. The content of the HomePage.html file must be replaced by the following code: <!DOCTYPE html> <html > <body> <span wicket_id="helloWorldMessage">Test</span> </body> </html> It is simple HTML code with the Wicket template wicket:id="helloWorldMessage". It indicates that in the Java code related to this page, a method will replace the message Test by another message. Now, let's edit the corresponding Java class; that is, HomePage. package com.packtpub.wicket.hello_world; import org.apache.wicket.markup.html.WebPage; import org.apache.wicket.markup.html.basic.Label; public class HomePage extends WebPage { public HomePage() { add(new Label("helloWorldMessage", "Hello world!!!")); } } The class HomePage extends WebPage; that is, it inherits some of the WebPage class's methods and attributes, and it becomes a WebPage subtype. One of these inherited methods is the method add(), where a Label object can be passed as a parameter. A Label object can be built by passing two parameters: an identifier and a string. The method add() is called in the HomePage class's constructor and will change the message in wicket:id="helloWorldMessage" with Hello world!!!. The resulting HTML code will be as shown in the following code snippet: <!DOCTYPE html> <html > <body> <span>Hello world!!!</span> </body> </html> Step 5 – compile and run! The steps to compile and run the project are as follows: To compile, right-click on the project and go to Run As | Maven install. Verify if the compilation was successful. If not, Wicket provides good error messages, so you can try to fix what is wrong. To run the project, right-click on the class Start and go to Run As | Java application. The class Start will run an embedded Jetty instance that will run the application. Verify if the server has started without any problems. Open a web browser and enter this in the address field: http://localhost:8080. In case you have changed the port, enter http://localhost:<port>. The browser should show Hello world!!!. The most common problem that can occur is that port 8080 is already in use. In this case, you can go into the Java Start class (found at src/test/java) and set another port by replacing 8080 in connector. setPort(8080) (line 21) by another number (for example, 9999). To stop the server, you can either click on Console and press any key or click on the red square on the console, which indicates termination. And that's it! By this point, you should have a working Wicket "Hello World" application and are free to play around and discover more about it. Summary This article describes how to create a simple "Hello World" application using Apache Wicket 6. Resources for Article : Further resources on this subject: Tips for Deploying Sakai [Article] OSGi life cycle [Article] Apache Wicket: Displaying Data Using DataTable [Article]
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04 Sep 2013
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SOA Application Design

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

Packt
04 Sep 2013
7 min read
(For more resources related to this topic, see here.) Concept of reusability The concept of reusability has its roots in the production process. Typically, most of us go about creating e-learning using a process similar to what is shown in the following screenshot. It works well for large teams and the one man band, except in the latter case, you become a specialist for all the stages of production. That's a heavy load. It's hard to be good at all things and it demands that you constantly stretch and improve your skills, and find ways to increase the efficiency of what you do. Reusability in Storyline is about leveraging the formatting, look and feel and interactions you create so that you can re-purpose your work and speed-up production. Not every project will be an original one-off, in fact most won't, so the concept is to approach development with a plan to repurpose 80 percent of the media, quizzes, interactions, and designs you create. As you do this, you begin to establish processes, templates, and libraries that can be used to rapidly assemble base courses. With a little tweaking and some minor customization, you'll have a new, original course in no time. Your client doesn't need to know that 80 percent was made from reusable elements with just 20 percent created as original, unique components, but you'll know the difference in terms of time and effort. Leveraging existing assets So how can you leverage existing assets with Storyline? The first things you'll want to look at are the courses you've built with other authoring programs, such as PowerPoint, QuizMaker Engage, Captivate, Flash, and Camtasia. If there are design themes, elements, or interactions within these courses that you might want to use for future Storyline courses, you should focus your efforts on importing what you can, and further adjusting within Storyline to create a new version of the asset that can be reused for future Storyline courses. If re-working the asset is too complex or if you don't expect to reuse it in multiple courses, then using Storyline's web object feature to embed the interaction without re-working it in any way may be the better approach. In both cases, you'll save time by reusing content you've already put a lot of time in developing. Importing external content Here are the steps to bring external content into Storyline: From the Articulate Startup screen or by choosing the Insert tab, and then New Slide within a project, select the Import option. There are options to import PowerPoint, Quizmaker, and Storyline. All of these will display the slides within the file to be imported. You can pick and choose which slides to import into a new or the current scene in Storyline. The Engage option displays the entire interaction that can be imported into a single slide in the current or a new scene. Click on Import to complete the process. Considerations when importing Keep the following points in mind when importing: PowerPoint and Quizmaker files can be imported directly into Storyline. Once imported, you can edit the content like you would any other Storyline slide. Master slides come along with the import making it simple to reuse previous designs. Note that 64-bit PowerPoint is not supported and you must have an installed, activated version of Quizmaker for the import to work. The PowerPoint to Storyline conversion is not one-to-one. You can expect some alignment issues with slide objects due to the fact that PowerPoint uses points and Storyline uses pixels. There are 2.66 pixels for each point which is why you'll need to tweak the imported slides just a bit. Same with Quizmaker though the reason why is slightly different; Quizmaker is 686 x 424 in size, whereas Storyline is 720 x 540 by default. Engage files can be imported into Storyline and they are completely functional, but cannot be edited within Storyline. Though the option to import Engage appears on the Import screen, what Storyline is really doing is creating a web object to contain the Engage interaction. Once imported into a new scene, clicking on the Engage interaction will display an Options menu where you can make minor adjustments to the behavior of the interaction as well as Preview and Edit in it Engage. You can also resize and position the interaction just as you would any web object. Remember that though web objects work in iPad and HTML5 outputs, Engage content is Flash, so it will not playback on an iPad or in an HTML5 browser. Like Quizmaker, you'll need an installed, activated version of Engage for the import to work. Flash, Captivate, and Camtasia files cannot be imported in Storyline and cannot be edited within Storyline. You can however, use web objects to embed these projects into Storyline or the Insert Flash option. In both cases, the imported elements appear seamless to the learner while retaining full functionality.   Build once, and reuse many times Quizzing is at the heart of many e-learning courses where often the quiz questions need to be randomized or even reused in different sections of a single course (that is, the same questions for a pre and post-test). The concept of building once and reusing many times works well with several aspects of Storyline. We'll start with quizzing and a feature called Question Banks as follows: Question Banks Question Bank offers a way to pool, reuse, and randomize questions within a project. Slides in a question bank are housed within the project file but are not visible until placed into the story. Question Banks can include groups of quiz slides and regular slides (that is, you might include a regular slide if you need to provide instructions for the quiz or would like to include a post-quiz summary). When you want to include questions from a Question Bank, you just need to insert a new Quizzing slide, and then choose Draw from Bank . You can then select one or more questions to include and randomize them if desired. Follow along… In this exercise we will be removing three questions from a scene and moving them into a question bank. This will allow you to draw one or more of those questions at any point in the project where the quiz questions are needed, as follows: From the Home tab, choose Question Banks , and then Create Question bank . Title this Identity Theft Questions . Notice that a new tab has opened in Normal View . The Question Bank appears in this tab. Click on the Import link and navigate to question slides 2, 3, and 4. From the Import drop-down menu at the top, select move questions into question bank . Click on the Story View tab and notice the three slides containing the quiz questions are no longer in the story. Click back on the Identity Theft tab and notice that they are located here. The questions will not become a part of the story until the next step, when you draw them from the bank. In Story View, click once on slide 1 to select it, and then from the Home tab, choose Question Banks and New Draw from Question Bank . From the Question Bank drop-down menu, select Identity Theft Questions . All questions will be selected by default and will be randomized after being placed into the story. This means that the learner will need to answer three questions before continuing onto the next slide in the story. Click on Insert . The Question Bank draw has been inserted as slide 2. To see how this works, Preview the scene. Save as Exercise 11 – Identity Theft Quiz.   There are multiple ways to get back to the questions that are in a question bank. You can do this by selecting the tab the questions are located in (in this case, Identity Theft ), you can view the question bank slide in Normal View or choose Question Banks from the Home tab and navigate to the name of the question bank you'd like to edit.
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article-image-unpacking-system-center-2012-orchestrator
Packt
04 Sep 2013
6 min read
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Unpacking System Center 2012 Orchestrator

Packt
04 Sep 2013
6 min read
(For more resources related to this topic, see here.) Planning the Orchestrator deployment The installation of SCORCH is simple. You must plan the deployment appropriately according to your needs. This recipe discusses and provides steps on common planning tasks to be performed before inserting the DVD or mounting the ISO for organizations who have successfully deployed SCORCH. Getting ready The authors recommend you to review the latest information on SCORCH at http://technet.microsoft.com/en-us/library/hh420383.aspx as the requirements of the product and supported platforms are regularly updated by Microsoft. How to do it... There are three planning categories, people, process, and the technology (SCORCH product). Identify and agree on the roles and responsibilities of the SCORCH team. SCORCH deployments typically have three types of users; service accounts, Administrators, and operators. Services accounts: They perform actions for the specific components of SCORCH Administrators: They will typically perform all activities including, but not limited to, SCORCH installation, Runbook creation and management, and delegation of security to operators Operators: They will typically use the SCORCH console and the Runbook Designer to create and manage Runbooks Identify and document initial prototype processes to be used as the first candidate for automation and testing. The types of processes for this purpose should be simple repeatable tasks that fall into an organizations required standard service requests. Good candidates are service request which do not require authorization and approval. An additional example category is Windows operating system services that can be stopped and started as a part of trouble shooting. Plan for the following technology requirements areas for SCORCH: SCORCH deployment type Deployment type Description Single Server All SCORCH roles installed on one physical or virtual machine This scenario is typically implemented in test environments but is fully supported in production. This however becomes a single point of failure for highly automated environments. Multi-server The SCORCH roles are separated and installed on one or more machines Minimum hardware requirements for each SCORCH component Component Requirements Management Server Operating system: Windows Server 2008 R2 or Windows Server 2012* 1 gigabyte (GB) of RAM, 2 GB or more recommended 200 megabytes (MB) of available hard disk space Dual-core Intel microprocessor, 2.1 gigahertz (GHz) or better Microsoft .NET Framework 3.5 Service Pack 1 Orchestration database Database: Microsoft SQL Server 2008 R2 or SQL Server 2012 Collation: SQL_Latin1_General_CP1_CI_AS Local or Remote (Basic Engine only) Runbook Server Operating system: Windows Server 2008 R2 or Windows Server 2012* 1 gigabyte (GB) of RAM minimum, 2 GB or more recommended 200 megabytes (MB) of available hard disk space Dual-core Intel microprocessor, 2.1 gigabyte (GHz) or better Microsoft .NET Framework 3.5 Service Pack 1 Orchestrator Console/Web Service Operating system: Windows Server 2008 R2 or Windows Server 2012* 1 gigabyte (GB) of RAM minimum, 2 GB or more recommended 200 megabytes (MB) of available hard disk space Dual-core Intel microprocessor, 2.1 gigahertz (GHz) or better Microsoft .NET Framework 3.5 Service Pack 1 Web Service: Internet Information Services (IIS) 7.0 and enabled IIS role Microsoft .NET Framework 3.5 Service Pack 1 Microsoft .NET Framework 4 Microsoft Silverlight 4** Orchestrator Runbook Designer Operating system: Windows Server 2008 R2, Windows 7 (32/64 bit) or Windows Server 2012* 1 gigabyte (GB) of RAM minimum, 2 GB or more recommended 200 megabyte (MB) of available hard disk space Dual-core Intel microprocessor, 2.1 gigabyte (GHz) or better Microsoft .NET Framework 3.5 Service Pack 1 SCORCH 2012 SP1 It is required only for the computer running the console in its web browser but not the Web Service server. Services accounts and delegation groups Account/Group Type Notes Orchestrator management service Service account Create an Active Directory user account for this service. This is the main management server service account and it is granted log on as a service during the installation. Orchestrator Runbook monitor service Service account Typically this is the same account as the Orchestrator Management Service. Orchestrator Runbook service Service account Same user account as the Management and Runbook Server monitor service in a single deployment but can be different for multi-server deployments; Active Directory domain account recommended. Runbook authors (SCO_ADMINS) Group Create an Active Directory group. This group will have the equivalent access of full administration to the SCORCH deployment. Runbook operators (SCO_CON_USERS) Group Create an Active Directory group. This group will have the equivalent access of a Runbook operator to the SCORCH deployment. Installation user User The user with full administrative rights on the SCORCH servers is required to perform the installation and configuration of the SCORCH deployment. Network Communication Ports Source Targeted computer Default port Configurable Runbook Designer Management Server 135, 1024-65535 Yes. Management Server, Runbook Server, and Web Service Orchestration database 1433 Yes; specified during the installation on the SCORCH supported version of Microsoft SQL Server. This is the case where the SQL Server instance is not using the default port. Client browser Orchestrator Web Service 81 Yes; during the SCORCH installation. Client browser Orchestration Console 82 Yes; during the SCORCH installation. How it works... The planning activities discussed are the minimum activities the authors recommend. The tasks performed at this stage will ensure that you ask for and plan for all your requirements before investing time in the actual installation. An additional benefit is identifying any people or budgetary risks before the deployment. There's more... There are two additional planning areas which are typically ignored in technology focused deployments. These areas are communication strategies and stakeholder management. Communication strategy One of the inaccurate myths of SCORCH is that it would automate the IT professional. SCORCH when implemented right would improve efficiency but will not replace people. On the contrary you need to communicate with the people who perform the manual tasks as they hold the key to how to best automate their efforts. Early engagement with all IT team members should be one of your key planning tasks. Stakeholder management Stakeholders are all users affected by the SCORCH deployment. An important category of stakeholders are the management team responsible for policy creation and enforcement. Automation without organization buy in may lead to conflicts at the political level of your organization. An example of such a scenario is the ability to create Active Directory user accounts with rights to specific organization areas and restricted resources.
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article-image-lets-make-particles
Packt
04 Sep 2013
15 min read
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Let's Make Particles

Packt
04 Sep 2013
15 min read
(For more resources related to this topic, see here.) The particle systems in Motion 5 are a powerful engine by which we can take nearly any object, image, layer, or group and animate it using the parameters available to us in the HUD and Inspector. A particle system consists of two items—the emitter and the cell. The cell is referenced by the emitter and the emitter creates the animation over a lifespan specified by you. Let's say we had a PNG layer of an orange. If we created particles out of it, that orange would be put into a cell that is referenced by the emitter. You could use the emitter's parameters to duplicate that orange multiple times per second and have it animate in a particular direction until you decide it should end or die. On top of the ability to turn nearly anything your heart desires into particles, in Motion's library there are pre-animated particle emitters available to incorporate in all of your animations. Making particles and changing values in the HUD Let's take a look at how we can create a particle system using a shape from Motion's library and tweak a few of its parameters using the HUD. Getting ready From the exercise files of this article, double-click the 07_01 project. There is the Shape layer in the Layers tab, whose scale has been animated to repeat for the duration of the project. Our goal is to place the heart in a particle system so that we can have hundreds rain down onto the Canvas. How to do it... The following steps will take you through creating your first particle system: Make sure your playhead is at the beginning of your project. Select the Pink Heart layer and press E , or from the toolbar choose Create a particle emitter (the icon with the three bubbles rising up, as shown in the following screenshot): Play back the project. Several things just happened after you pressed that button. In the Layers tab, notice that the Pink Heart layer has been turned off. A cell and emitter have been created just above it. The cell holds information about the heart while the emitter is creating all the duplicate copies of the heart that are shooting out in a 360-degree circle across the screen at the same speed. You can see a screenshot of this next. Press F7 to bring up the HUD and see some of these parameters in more detail: Right now, thirty hearts are being born every second and live for a duration of five seconds where they pop off the screen. All the duplicate hearts also hold the original scale of the heart being referenced. Change the Birth Rate value to 5 and Life to 10 . Bring down the Scale parameter to 50 , as shown in the following screenshot: Instead of having the hearts come from the center of the screen, let's have them rain down from the top of the Canvas. Decrease the size of your Canvas by clicking on it and pressing Command + - a few times. Select the particle emitter and drag it up and offscreen. Use the Shift key to constrain the movement. Change the Emission Range slider from 360 to 180 and make sure the arrows point down. Play back the animation and tweak the Birth Rate and Scale sliders as desired. See the following screenshot for reference: There's more… If you're finding the HUD limited in terms of the options available to you for the emitter, don't worry, the Inspector has several additional parameters, including the option to add random values to your emitter that will add more realism to it! Take a peak at the following screenshot: Tweaking particle parameters in the Inspector In the previous recipe, we switched a few parameters in the HUD for our particle system of hearts, but if we're looking to fine-tune our animation, we need to go to the Inspector. Let's take a look at the several additional parameters available. Getting ready From the exercise files of this article, double-click the 07_02 project. Play back the project. There is a particle system in the center of the Canvas being emitted in the form of a rectangle. Currently, thirty hearts are being born every second and live for five seconds. We're going to tweak our particle system to change the flow of the animation so that our hearts emit out in a circle and away from the screen. How to do it... By following these steps, we'll gain a better understanding of how we can control the particle system: Select the Emitter in the Layers tab. Press F4 to open the Emitter tab of the Inspector. Change the Shape menu's option from rectangle to circle and the Arrangement option to Random Fill . Set the radius to 500. Increase Birth Rate to 200 and set Speed in Cell Controls to 0. If you play back the project, you'll see the hearts trying to form the shape of a circle. Since there is no speed value, the hearts stay put in the shape they originate from, as shown in this screenshot: Set Angle Value to 15 and Spin to 30. Rather than having the particles pop onscreen when they're born and pop offscreen when they die, we're going to add some tags to the Opacity Over Life parameter. The trick is to click the white line where we want to add a tag above it. Click the line once close to the beginning and twice close to the end. Use the following screenshot for reference: Click on the first tag and under the line, drag the Opacity slider to 0 (indicated in this screenshot). Repeat this step for the last slider. Now the hearts fade in at birth and fade out at death: As a final step, we're going to keyframe the speed of the particle system to start at 100 percent and eventually go down to 0 to reveal the shape of the circle we created. Go to the beginning of the Timeline and click the diamond icon next to Speed to add a keyframe. Change its value to 100. Move to three seconds and add another keyframe by clicking the diamond icon next to Speed again. Move to three seconds and set Speed to 0. Play back the animation and compare it to the following screenshot: There's more... You can create some amazing animations by changing the type of shape used to emit particles. For instance, if you change the shape of your particles to Image and use text as the source, the particles will spell out the word as long there is a substantial number of hearts being born and the Arrangement option is set to Random Fill . We'll see an example of this in a later exercise, but the following screenshot gives you a sneak peak on the end results you can get: Adding randomness values To give your particle systems just a little more, we're going to add a little randomness to some of the parameters. This randomness will help give your animations a more organic feel. Getting ready From the exercise files of this article, double-click the 07_03 project. Play back the project. The particle system consists of little demograms that rain down from a line. The Angle and Spin values cause the demograms to rotate while moving, and the Opacity Over Life parameter allows each particle to fade in and out during its lifespan. The particles also change color over their life as reflected in the Color Mode menu and the gradient outlining the cycle. We're going to add randomness values to tweak the animation. Whenever we add a randomness value, it's going to look at the original value above it and add and subtract to it randomly every second based on the number you specify. For example, say the Scale value is set to 50. If you add a Scale randomness of 10, every second the demograms will be somewhere between a scale of 40 and 60. How to do it... Let's start to randomize different values in this recipe: Select the emitter in the Layers tab and press F4 . In the Emitter tab of the Inspector, change the Birth Rate Randomness slider's value to 10. Now, every second, somewhere between 10 and 30 objects are born. Now, set Life Randomness to 3, Angle Randomness to 40, Spin Randomness to 50, and Scale Randomness to 80. Play back the project. Now we have our demograms growing at different sizes, being born at different angles and spinning at different rates! The following screenshot shows the Inspector with all of the mentioned changes and also a Color Over Life change (see the previous recipe to learn how to do it). The following is a screenshot showing a frame of the particle animation in the Canvas with random values applied: While playing back, click the Generate button next to Random Seed . You'll notice that your animation changes and a new number is created next to Generate . Random Seed takes all the random parameters and creates the animation based on this number. Change this number and change the randomness of the animation, as shown here: Working with particle behaviors While there are already a ton of parameters you can animate by keyframing their values in the particle and cell emitters, Motion offers you a few particle behaviors worth taking a look at from the library as well. Getting ready From the exercise files of this article, double-click the 07_04 project. Play back the project. The animation consists of a bouncing alarm clock from Motion's library in the foreground and a particle system referencing that alarm clock in the background. Unlike the other particle systems we've worked with in the previous recipes, the source of the particle system has been turned back on to preserve it. Our goal is to add two particle behaviors to have the clocks scale down and spin over its life cycle. By adjusting the speed and direction of our emitter, we will also have the clocks look like they're being pulled off into the distance. Also note, the anchor point of the clock was adjusted in advance in order to have the particles spin around the center of the clock. How to do it... Open the Particle group and select Emitter . Press Command + 2 to go to the Library tab. Under Behaviors , choose Particles | Spin Over Life . Click Apply . Press F7 to open the HUD. Set Increment Type to Birth and Death Values . Set Birth to 360 and Death to 0. If you're having trouble, try changing the values in the Behaviors tab of the Inspector. Now, play back the animation. Now, let's have the clocks in the particle system scale over life. Select the Emitter again. Press Command + 2 to go to the Library . Under Behaviors , choose Particles | Scale Over Life .Click Apply. In the HUD, set Increment Type to Birth and Death Values . Set Scale at Birth to 100 and Scale at Death to 0. If you're having trouble, try changing the values in the Behaviors tab of the Inspector. Play back the animation. The clocks now scale down gradually until they die, as shown in the following screenshot: Press F4 to go to the Emitter tab of the Inspector. Change the Emission Longitude value to 180 and Speed to 2000 , as shown in the following screenshot. Change the other parameters as desired. There's more... When you start building complex particle animations, keep in mind that you can save them to the Library as well as any of the behaviors you tweak. All particle presets can be found in the Library under Particle Emitters in a dedicated theme folder you create. The more time you spend automating your work, the more time you can dedicate to the creative process! Working with particle presets So far, we've created our own particle systems by selecting objects to place into cells referenced by the emitter. Motion ships with over 200 presets! One of the best ways to learn Motion is to dissect how some of the particle presets were created. Let's take a look at what it has to offer! Getting ready From the exercise files of this article, double-click on the 07_05 project. There are two particle presets in this project but one has been turned off. Play back the project and familiarize yourself with the Magic Dust preset. Stop the playback. Turn off the Magic Dust group and turn on the Buskerit group. Play back the project again. In this recipe, we'll take a look at each of these presets and change a few parameters to look at how the animations were created. How to do it... Let's now tweak these parameters of our presets in more detail: Reveal the content of the Buskerit group by clicking the disclosure triangle next to it. It contains a group called busker and seven still images that have been turned off. The images are being referenced by the emitter. Click the disclosure triangle next to the Busker group. It contains the content of the Busker emitter and the seven cells that hold information from those layers that have been shot off. The Emitter also has a Scale Over Life behavior in it. While the object continually changes scale over the animation, the Scale Over Life behavior was used to animate the "pop-up" intro of the instruments, whereas keyframes were used to change the scale over the project. Select the Scale Over Life behavior and press F2 to go to the Behaviors tab of the Inspector. Notice the Increment Type parameter set to Custom . Click the disclosure triangle next to Custom Scale to see the graph, as shown in the following screenshot: Select the Emitter in the Layers tab and press Command + 8 to open the Keyframe Editor . Notice that both the scale and rotation of the emitter are keyframed periodically throughout the entire animation, as shown in the following screenshot. In the Keyframe Editor , press Shift + K to move forward between keyframes. Change the Rotation and Scale values by double-clicking the values while on an existing keyframe. You'll know you're on one because the diamond shape will appear highlighted! Click the disclosure triangle to close the Buskerit group, turn it off, and turn on the Magic Dust group. Click the disclosure triangle to open it. A still image of a spark was used as the source for the Emitter. Several behaviors were used to animate the sparks. With the emitter selected, press F4 to go to the Emitter tab of the Inspector. Set Emission Range to 45 , Birth Rate to 200 , and Speed to 1000 . Play back the project to see the effect it has and compare it to the following: Select the Magic Dust group and go to the gear icon under the mini-Timeline. Choose Basic Motion | Motion Path . Tweak the path as desired. Play back the animation. There's more... Learning how particle presets work is a great way to get used to Motion. Particle presets are the best way to learn Motion! Looking at the library and seeing how something was created is the best way to get under the hood of the application and start creating your own animations. Don't be afraid to explore. Working with particle presets in 3D Particle presets are already powerful on their own but turn them into 3D and you'll find that a few of them will actually look like they have been extruded. Let's take a look! Getting ready From the exercise files of this article, double-click on the 07_06 project. Play back the project. There is a particle system of thin water being projected across the screen. Go to the Properties tab in the Inspector and click the rotation disclosure triangle to see the object rotate 90 degrees on its y axis. The water becomes invisible because it's flat the minute you change perspective. If we promote a particle system to 3D, we can get rid of this problem and have a few additional options available to us in terms of the way we emit the water. How to do it... Let's get rid of the flatness of our particle system by making it 3D: Select the Particle Emitter category and press F4 to go to the Inspector. Click on the 3D checkbox. As soon as you press the button, more options become available to you in the Emitter Controls section, including Emission Latitude and Emission Longitude , as seen here: Two additional selections—Box and Sphere —also became available under the Shape drop-down. Click on the word Rectangle and choose Box . Press F1 to go to the Properties tab of the Inspector. Click the disclosure triangle next to Scale and increase Scale Z to 200 . Rotate the water 90 degrees on the Y axis and notice it's no longer flat. Use the following screenshot of the Properties tab for reference: Go to the beginning of the Timeline and bring back Rotation Y to 0. Click the diamond shape next to Rotation Y to add a keyframe. Move to the end of the Timeline; and change the parameter to 720. Play back the project and see that, no matter what angle it is, the particle system never goes flat, as shown here:
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article-image-oracle-apex-42-reporting
Packt
03 Sep 2013
20 min read
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Oracle APEX 4.2 reporting

Packt
03 Sep 2013
20 min read
(For more resources related to this topic, see here.) The objective of the first chapter is to quickly introduce you to the technology and then dive deep into the understanding the fabric of the tool. The chapter also helps you set the environment, which will be used throughout the book. Chapter 1, Know Your Horse Before You Ride It, starts with discussing the various features of APEX. This is to give heads up to the readers about the features offered by the tool and to inform them about some of the strengths of the tool. In order to understand the technology better, we discuss the various web server combinations possible with APEX, namely the Internal mod_plsql, External mod_plsql, and Listener configuration. While talking about the Internal mod_plsql configuration, we see the steps to enable the XMLDB HTTP server. In the Internal mod_plsql configuration, Oracle uses a DAD defined in EPG to talk to the database and the web server. So, we try to create a miniature APEX of our own, by creating our DAD using it to talk to the database and the web server. We then move on to learn about the External mod_plsql configuration. We discuss the architecture and the roles of the configuration files, such as dads.conf and httpd.conf. We also have a look at a typical dads.conf file and draw correlations between the configurations in the Internal and External mod_plsql configuration. We then move on to talk about the wwv_flow_epg_include_mod_local procedure that can help us use the DAD of APEX to call our own stored PL/SQL procedures. We then move on to talk about APEX Listener which is a JEE alternative to mod_plsql and is Oracle’s direction for the future. Once we are through with understanding the possible configurations, we see the steps to set up our environment. We use the APEX Listener configuration and see the steps to install the APEX engine, create a Weblogic domain, set the listener in the domain, and create an APEX workspace. With our environment in place, we straight away get into understanding the anatomy of APEX by analyzing various parts of its URL. This discussion includes a natter on the session management, request handling, debugging, error handling, use of TKPROF for tracing an APEX page execution, cache management and navigation, and value passing in APEX. We also try to understand the design behind the zero session ID in this section. Our discussions till now would have given you a brief idea about the technology, so we try to dig in a little deeper and understand the mechanism used by APEX to send web requests to its PL/SQL engine in the database by decoding the APEX page submission. We see the use of the wwv_flow.accept procedure and understand the role of page submission. We try to draw an analogy of an APEX form with a simple HTML to get a thorough understanding about the concept. The next logical thing after page submission is to see the SQL and PL/SQL queries and blocks reaching the database. We turn off the database auditing and see the OWA web toolkit requests flowing to the database as soon as we open an APEX page. We then broaden our vision by quickly knowing about some of the lesser known alternatives of mod_plsql. We end the chapter with a note of caution and try to understand the most valid criticisms of the technology, by understanding the SQL injection and Cross-site Scripting (XSS). After going through the architecture we straight away spring into action and begin the process of learning how to build the reports in APEX. The objective of this chapter is to help you understand and implement the most common reporting requirements along with introducing some interesting ways to frame the analytical queries in Oracle. The chapter also hugely focuses on the methods to implement different kinds of formatting in the APEX classic reports. We start Chapter 2, Reports, by creating the objects that will be used throughout the book and installing the reference application that contains the supporting code for the topics discussed in the second chapter. We start the chapter by setting up an authentication mechanism. We discuss external table authentication in this chapter. We then move on to see the mechanism of capturing the environment variables in APEX. These variables can help us set some logic related to a user’s session and environment. The variables also help us capture some properties of the underlying database session of an APEX session. We capture the variables using the USERENV namespace, DBMS_SESSION package, and owa_util package. After having a good idea of the ways and means to capture the environment variables, we build our understanding of developing the search functionality in a classic APEX report. This is mostly a talk about the APEX classic report features, we also use this opportunity to see the process to enable sorting in the report columns, and to create a link that helps us download the report in the CSV format. We then discuss various ways to implement the group reports in APEX. The discussion shows a way to implement this purely, by using the APEX’s feature, and then talks about getting the similar results by using the Oracle database feature. We talk about the APEX’s internal grouping feature and the Oracle grouping sets. The section also shows the first use of JavaScript to manipulate the report output. It also talks about a method to use the SQL query to create the necessary HTML, which can be used to display a data column in a classic report. We take the formatting discussion further, by talking about a few advanced ways of highlighting the report data using the APEX’s classic report features, and by editing the APEX templates. After trying our hand at formatting, we try to understand the mechanism to implement matrix reports in APEX. We use matrix reports to understand the use of the database features, such as the with clause, the pivot operator, and a number of string aggregation techniques in Oracle. The discussion on the string aggregation techniques includes the talk on the LISTAGG function, the wm_concat function, and the use of the hierarchical queries for this purpose. We also see the first use of the APEX items as the substitution variables in this book. We will see more use of the APEX items as the substitution variables to solve the vexed problems in other parts of the book as well. We do justice with the frontend as well, by talking about the use of jQuery, CSS, and APEX Dynamic Actions for making the important parts of data stand out. Then, we see the implementation of the handlers, such as this.affectedElements and the use of the jQuery functions, such as this.css() in this section. We also see the advanced formatting methods using the APEX templates. We then end this part of the discussion, by creating a matrix report using the APEX Dynamic Query report region. The hierarchical reports are always intriguing, because of the enormous ability to present the relations among different rows of data, and because of the use of the hierarchical queries in a number of unrelated places to answer the business queries. We reserve a discussion on the Oracle’s hierarchical queries for a later part of the section and start it, by understanding the implementation of the hierarchical reports by linking the values in APEX using drilldowns. We see the use of the APEX items as the substitution variable to create the dynamic messages. Since we have devised a mechanism to drill down, we should also build a ladder for the user to climb up the hierarchical chain. Now, the hierarchical chain for every person will be different depending on his position in the organization, so we build a mechanism to build the dynamic bread crumbs using the PL/SQL region in APEX. We then talk about two different methods of implementing the hierarchical queries in APEX. We talk about the connect, by clause first, and then continue our discussion by learning the use of the recursive with clause for the hierarchical reporting. We end our discussion on the hierarchical reporting, by talking about creating the APEX’s Tree region that displays the hierarchical data in the form of a tree. The reports are often associated with the supporting files that give more information about the business query. A typical user might want to upload a bunch of files while executing his piece of the task and the end user of his action might want to check out the files uploaded by him/her. To understand the implementation of this requirement, we check out the various ways to implement uploading and downloading the files in APEX. We start our discussion, by devising the process of uploading the files for the employees listed in the oehr_employees table. The solution of uploading the files involves the implementation of the dynamic action to capture the ID of the employee on whose row the user has clicked. It shows the use of the APEX items as the substitution variables for creating the dynamic labels and the use of JavaScript to feed one APEX item based on the another. We extensively talk about the implementation of jQuery in Dynamic Actions in this section as well. Finally, we check out the use of the APEX’s file browse item along with the WWV_FLOW_FILES table to capture the file uploaded by the user. Discussion on the methods to upload the files is immediately followed by talking about the ways to download these files in APEX. We nest the use of the functions, such as HTF.ANCHOR and APEX_UTIL.GET_BLOB_FILE_SRC for one of the ways to download a file, and also talk about the use of dbms_lob.getlength along with the APEX format mask for downloading the files. We then engineer our own stored procedure that can download a blob stored in the database as a file. We end this discussion, by having a look at the APEX’s p process, which can also be used for downloading. AJAX is the mantra of the new age and we try our hand at it, by implementing the soft deletion in APEX. We see the mechanism of refreshing just the report and not reloading the page as soon as the user clicks to delete a row. We use JavaScript along with the APEX page process and the APEX templates to achieve this objective. Slicing and dicing along with auditing are two of the most common requirements in the reporting world. We see the implementation of both of these using both the traditional method JavaScript with the page processes and the new method of using Dynamic Actions. We extend our use case a little further and learn about a two way interaction between the JavaScript function and the page process. We learn to pass the values back and forth between the two. While most business intelligence and reporting solutions are a one way road and are focused on the data presentation, Oracle APEX can go a step further, and can give an interface to the user for the data manipulations as well. To understand this strength of APEX, we have a look at the process of creating the tabular forms in APEX. We extend our understanding of the tabular forms to see a magical use of jQuery to convert a certain sections of a report from display-only to editable textboxes. We move our focus from implementing the interesting and tricky frontend requirements to framing the queries to display the complex data types. We use the string aggregation methods to display data in a column containing a varray. Time dimension is one of the most widely used dimension in reporting circles and comparing current performance with the past records is a favorite requirement of most businesses. With this in mind, we shift our focus to understand and implement the time series reports in APEX. We start our discussion by understanding the method to implement a report that shows the contribution of each business line in every quarter. We implement this by using partitioning dimensions on the fly. We also use the analytical functions, such as ratio_to_report, lead, and lag in the process of creating the time series reports. We use our understanding of time dimension to build a report that helps a user compare one time period to the other. The report gives the user the freedom to select the time segments, which he wishes to compare. We then outwit a limitation of this report, by using the query partition clause for the data densification. We bring our discussion on reports based on time dimension, by presenting a report based on the modal clause to you. The report serves as an example to show the enormous possibilities to code, by using the modal clause in Oracle. We bring our discussion on reports based on time dimension, by presenting a report based on the modal clause to you. The report serves as an example to show the enormous possibilities to code, by using the modal clause in Oracle. Chapter 3, In the APEX Mansion – Interactive Reports is all about Interactive Reports and the dynamic reporting. While the second chapter was more about the data presentation using the complex queries and the presentation methods, this chapter is about taking the presentation part a step ahead, by creating more visually appealing Interactive Reports. We start the discussion of this chapter, by talking about the ins and outs of Interactive Reports. We postmortem this feature of APEX to learn about every possible way of using Interactive Reports for making more sense of data. The chapter has a reference application of its own, which shows the code in action. We start our discussion, by exploring at various features of the Actions menu in Interactive Reports. The discussion is on search functionality, Select Columns feature, filtering, linking and filtering Interactive Reports using URLs, customizing Rows per page feature of an IR, using Control Break, creating computations in IR, creating charts in IR, using the Flashback feature and a method to see the back end flashback query, configuring the e-mail functionality for downloading a report, and for subscription of reports and methods to understand the download of reports in HTML, CSV, and PDF formats. Once we are through with understanding the Actions menu, we move on to understand the various configuration options in an IR. We talk about the Link section, the Icon View section, the Detail section, the Column Group section, and the Advanced section of the Report Attributes page of an IR. While discussing these sections, we understand the process of setting different default views of an IR for different users. Once we are through with our dissection of an IR, we put our knowledge to action, by inserting our own item in the Actions menu of an IR using Dynamic Actions and jQuery. We continue our quest for finding the newer formatting methods, by using different combinations of SQL, CSS, APEX templates, and jQuery to achieve unfathomable results. The objectives attained in this section include formatting a column of an IR based on another column, using CSS in the page header to format the APEX data, changing the font color of the alternate rows in APEX, using a user-defined CSS class in APEX, conditionally highlighting a column in IR using CSS and jQuery, and formatting an IR using the region query. After going through a number of examples on the use of CSS and jQuery in APEX, we lay down a process to use any kind of changes in an IR. We present this process by an example that changes one of the icons used in an IR to a different icon. APEX also has a number of views for IR, which can be used for intelligent programming. We talk about an example that uses the apex_application_page_ir_rpt view that show different IR reports on the user request. After a series of discussion on Interactive Reports, we move on to find a solution to an incurable problem of IR. We see a method to put multiple IR on the same page and link them as the master-child reports. We had seen an authentication mechanism (external table authentication) and data level authorization in Chapter 2, Reports. We use this chapter to see the object level authorization in APEX. We see the method to give different kinds of rights on the data of a column in a report to different user groups. After solving a number of problems and learning a number of things, we create a visual treat for ourselves. We create an Interactive report dashboard using Dynamic Actions. This dashboard presents different views of an IR as gadgets on a separate APEX page. We conclude this chapter, by looking at advanced ways of creating the dynamic reports in APEX. We look at the use of the table function in both native and interface approach, and we also look at the method to use the APEX collections for creating the dynamic IRs in APEX. Chapter 4, The Fairy Tale Begins – Advanced Reporting, is all about advanced reporting and pretty graphs. Since we are talking about advanced reporting, we see the process of setting the LDAP authentication in APEX. We also see the use of JXplorer to help us get the necessary DN for setting up the LDAP authentication. We also see the means to authenticate an LDAP user using PL/SQL in this section. This chapter also has a reference application of its own that shows the code in action. We start the reporting building in this chapter, by creating the Sparkline reports. This report uses jQuery for producing the charts. We then move on to use another jQuery library to create a report with slider. This report lets the user set the value of the salary of any employee using a slider. We then get into the world of the HTML charts. We start our talk, by looking at the various features of creating the HTML chart regions in APEX. We understand a method to implement the Top N and Bottom N chart reports in the HTML charts. We understand the APEX’s method to implement the HTML charts and use it to create an HTML chart on our own. We extend this technique of generating HTML from region source a little further, by using XMLTYPE to create the necessary HTML for displaying a report. We take our spaceship into a different galaxy of the charts world and see the use of Google Visualizations for creating the charts in APEX. We then switch back to AnyChart, which is a flash charting solution, and has been tightly integrated with APEX. It works on an XML and we talk about customizing this XML to produce different results. We put our knowledge to action, by creating a logarithmic chart and changing the style of a few display labels in APEX. We continue our discussion on AnyChart, and use the example of Doughnut Chart to understand advanced ways of using AnyChart in APEX. We use our knowledge to create Scatter chart, 3D stacked chart, Gauge chart, Gantt chart, Candlestick chart, Flash image maps, and SQL calendars. We move out of the pretty heaven of AnyChart only to get into another beautiful space of understanding the methods of displaying the reports with the images in APEX. We implement the reports with the images using the APEX’s format masks and also using HTF.IMG with the APEX_UTIL.GET_BLOB_FILE_SRC function. We then divert our attention to advanced jQuery uses such as, the creation of the Dialog box and the Context menu in APEX. We create a master-detail report using dialog boxes, where the child report is shown in the Dialog box. We close our discussion in this chapter, by talking about creating wizards in APEX and a method to show different kinds of customized error messages for problems appearing at different points of a page process. Chapter 5, Flight to Space Station: Advanced APEX, is all about advanced APEX. The topics discussed in this chapter fall into a niche category of reporting the implementation. This chapter has two reference applications of its own that shows the code in action. We start this chapter, by creating both the client-side and server-side image maps APEX. These maps are often used where regular shapes are involved. We then see the process of creating the PL/SQL Server Pages (PSPs). PSPs are similar to JSP and are used for putting the PL/SQL and HTML code in a single file. Loadpsp then converts this file into a stored procedure with the necessary calls to owa web toolkit functions. The next utility we check out is loadjava. This utility helps us load a Java class as a database object. This utility will be helpful, if some of the Java classes are required for code processing. We had seen the use of AnyChart in the previous chapter, and this chapter introduces you to FusionChart. We create a Funnel Chart using FusionChart in this chapter. We also use our knowledge of the PL/SQL region in APEX to create a Tag Cloud. We then stroll into the world of the APEX plugins. We understand the interface functions, which have to be created for plugins. We discuss the concepts and then use them to develop an Item type plugin and a Dynamic Action plugin. We understand the process of defining a Custom Attribute in a plugin, and then see the process of using it in APEX. We move on to learn about Websheets and their various features. We acquaint ourselves with the interface of the Websheet applications and understand the concept of sharing Websheets. We also create a few reports in a Websheet application and see the process of importing and using an image in Websheets. We then spend some time to learn about data grids in Websheets and the method to create them. We have a look at the Administration and View dropdowns in a Websheet application. We get into the administration mode and understand the process of configuring the sending of mails to Gmail server from APEX. We extend our administration skills further, by understanding the ways to download an APEX application using utilities, such as oracle.apex.APEXExport. Reporting and OLAP go hand in hand, so we see the method of using the OLAP cubes in APEX. We see the process of modeling a cube and understand the mechanism to use its powerful features. We then have a talk about Oracle Advanced Queues that can enable us to do reliable communication among different systems with different workloads in the enterprise, and gives improved performance. We spend a brief time to understand some of the other features of APEX, which might not be directly related to reporting, but are good to know. Some of these features include locking and unlocking of pages in APEX, the Database Object Dependencies report, Shortcuts, the Dataloading wizard, and the APEX views. We bring this exclusive chapter to an end, by discussing about the various packages that enable us to schedule the background jobs in APEX, and by discussing various other APEX and Database API, which can help us in the development process.
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03 Sep 2013
11 min read
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Quick start – Using Burp Proxy

Packt
03 Sep 2013
11 min read
(For more resources related to this topic, see here.) At the top of Burp Proxy, you will notice the following three tabs: intercept: HTTP requests and responses that are in transit can be inspected and modified from this window options: Proxy configurations and advanced preferences can be tuned from this window history: All intercepted traffic can be quickly analyzed from this window If you are not familiar with the HTTP protocol or you want to refresh your knowledge, HTTP Made Really Easy, A Practical Guide to Writing Clients and Servers, found at http://www.jmarshall.com/easy/http/, represents a compact reference. Step 1 – Intercepting web requests After firing up Burp and configuring the browser, let's intercept our first HTTP request. During this exercise, we will intercept a simple request to the publisher's website: In the intercept tab, make sure that Burp Proxy is properly stopping all requests in transit by checking the intercept button. This should be marked as intercept is on. In the browser, type http://www.packtpub.com/ in the URL bar and press Enter. Back in Burp Proxy, you should be able to see the HTTP request made by the browser. At this stage, the request is temporarily stopped in Burp Proxy waiting for the user to either forward or stop it. For instance, press forward and return to the browser. You should see the home page of Packt Publishing as you would normally interact with the website. Again, type http://www.packtpub.com/ in the URL bar and press Enter. Let's press drop this time. Back in the browser, the page will contain the warning Burp proxy error: message was dropped by user. We have dropped the request, thus Burp Proxy did not forward the request to the server. As a result, the browser received a temporary HTML page with the warning message generated by Burp, instead of the original HTML content. Let's try one more time. Type http://www.packtpub.com/ in the URL bar of the browser and press Enter. Once the request is properly captured by Burp Proxy, the action button becomes active. Click on it to display the contextual menu. This is an important functionality as it allows you to import the current web request in any of the other Burp tools. You can already imagine the potentialities of having a set of integrated tools that allow you to manipulate and analyze web requests so easily. For example, if we want to decode the request, we can simply click on send to decoder. Burp Proxy In Burp Proxy, we can also decide to automatically forward all requests without waiting for the user to either forward or drop the communication. By clicking on the intercept button, it is possible to switch from intercept is on to intercept is off. Nevertheless, the proxy will record all requests in transit. Also, Burp Proxy allows you to automatically intercept all responses matching specific characteristics. Take a look at the numerous options available in the intercept server response section from within the Burp Proxy options tab. For example, it is possible to intercept the server's response only if the client's request was intercepted. This is extremely helpful while testing input validation vulnerabilities as we are generally interested in evaluating the server's responses for all tampered requests. Or else, you may only want to intercept and inspect responses having a specific return code (for example, 200 OK). Step 2 – Inspecting web requests Once a request is properly intercepted, it is possible to inspect the entire content, headers, and parameters, using one of the four Burp Proxy message analysis tabs: raw: This view allows you to display the web request in raw format within a simple text editor. This is a very handy visualization as it enables maximum flexibility for further changing the content. params: In this view, the focus is on user-supplied parameters (GET/POST parameters, cookies). This is particularly important in case of complex requests as it allows to consider all entry points for potential vulnerabilities. Whenever applicable, Burp Proxy will also automatically perform URL decoding. In addition, Burp Proxy will attempt to parse commonly used formats, including JSON. headers: Similarly, this view displays the HTTP header names and values in tabular form. hex: In case of binary content, it is useful to inspect the hexadecimal representation of the resource. This view allows to display a request as in a traditional hex editor. The history tab enables you to analyze all web requests transited through the proxy: Click on the history tab. At the top, Burp Proxy shows all the requests in the bundle. At the bottom, it displays the content of the request and response corresponding to the specific selection. If you have previously modified the request, Burp Proxy history will also display the modified version. Displaying HTTP requests and responses intercepted by Burp Proxy By double-clicking on one of the requests, Burp will automatically open a new window with the specific content. From this window, it is possible to browse all the captured communication using the previous and next buttons Back in the history tab, Burp Proxy displays several details for each item including the request method, URL, response's code, and length. Each request is uniquely identified by a number, visible in the left-hand side column. Click on the request identifier. Burp Proxy allows you to set a color for that specific item. This is extremely helpful to highlight important requests or responses. For example, during the initial application enumeration, you may notice an interesting request; you can mark it and get back later for further testing. Burp Proxy history is also useful when you have to evaluate a sequence of requests in order to reproduce a specific application behavior. Click on the display filter, at the top of the history list to hide irrelevant content. If you want to analyze all HTTP requests containing at least one parameter, select the show only parameterised checkbox. If you want to display requests having a specific response, just select the appropriate response code in the filter by status code selection. At this point, you may have already understood the potentialities of the tool to filter and reveal interesting traffic. In addition, when using Burp Suite Professional, you can also use the filter by search term option. This feature is particularly important when you need to analyze hundreds of requests or responses as you can filter relevant traffic only by using regular expressions or simply matching particular strings. Using this feature, you may also be able to discover sensitive information (for example, credentials) embedded in the intercepted pages. Step 3 – Tampering web requests As part of a typical security assessment, you will need to modify HTTP requests and analyze the web application responses. For example, to identify SQL injection vulnerabilities, it is important to inject common attack vectors (for example, a single quote) in all user-supplied input, including HTTP headers, cookies, and GET/POST parameters. If you want to refresh your knowledge on common web application vulnerabilities, the OWASP Top Ten Project article at https://www. owasp.org/index.php/Category:OWASP_Top_Ten_Project is a good starting point. Tampering web requests with Burp is as easy as editing strings in a text editor: Intercept a request containing at least one HTTP parameter. For example, you can point your browser to http://www.packtpub.com/books/all?keys=ASP. Go to Burp Proxy | Intercept. At this point, you should see the corresponding HTTP request. From the raw view, you can simply edit any aspect of the web request in transit. For example, you can change the value of the the GET parameter's keys value from ASP to PHP. Edit the request to look like the following: GET /books/all?keys=PHP HTTP/1.1Host: www.packtpub.comUser-Agent: Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:15.0)Gecko/20100101 Firefox/15.0.1Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8Accept-Language: en-us,en;q=0.5Accept-Encoding: gzip, deflateProxy-Connection: keep-alive Click on forward and get back to the browser. This should result in a search query performed with the string PHP. You can verify it by simply checking the results in the HTML page. Although we have used the raw view to change the previous HTTP request, it is actually possible to use any of the Burp Proxy view. For example, in the params view, it is possible to add a new parameter by following these steps: Clicking on new (right side), from the Burp Proxy params view. Selecting the proper parameter type (URL, body, or cookie). URL should be used for GET parameters, whereas body denotes POST parameters. Typing the name and the value of the newly created parameter. Advanced features After practicing with the basic features provided by Burp Proxy, you are almost ready to experiment with more advanced configurations. Match and replace Let's imagine that you are testing an application designed for mobile devices using a standard browser from your computer. In most cases, the web server examines the user-agent provided by the browser to identify the specific platform and respond with customized resources that better fit mobile phones and tablets. Under these circumstances, you will particularly find the match and replace function, provided by Burp Proxy, very useful. Let's configure Burp Proxy in order to tamper the user-agent HTTP header field: In the options tab of Burp Proxy, scroll down to the match and replace section. Under the match and replace table, a drop-down list and two text fields allow to create a customized rule. Select request header from the drop-down list since we want to create a match condition pertaining to HTTP requests. Type ^User-Agent.*$ in the first text field. This field represents the match within the HTTP request. Burp Proxy's match and replace feature allows you to use simple strings as well as complex regular expressions. If you are not familiar with regular expressions, have a look at http://www.regular-expressions.info/quickstart. html. In the second text field, type Mozilla/5.0 (iPhone; U; CPU like Mac OS X; en) AppleWebKit/4h20+ (KHTML, like Gecko) Version/3.0 Mobile/1C25 Safari/419.3 or any other fake user-agent that you want to impersonate. Click add and verify that the new match has been added to the list; this button is shown here: Burp Proxy match and replace list Intercept a request, leave it to pass through the proxy, and verify that it has been automatically modified by the tool. Automatically modified HTTP header in Burp Proxy HTML modification Another interesting feature of Burp Proxy is the automatic HTML modification, that can be activated and configured in the appropriate section within Burp Proxy | options. By using this function, you can automatically remove JavaScript or modify HTML forms of all received HTTP responses. Some applications deploy client-side validation in the form of disabled HTML form fields or JavaScript code. If you want to verify the presence of server-side controls that enforce specific data formats, you would need to tamper the request with invalid data. In these situations, you can either manually tamper the request in the proxy or enable HTML modification to remove any client-side validation and use the browser in order to submit invalid data. This function can be also used to display hidden form fields. Let's see in practice how you can activate this feature: In Burp Proxy, go to options, scroll down to the HTML modification section. Numerous options are available in this section: unhide hidden form fields to display hidden HTML form fields, enable disabled form fields to submit all input forms present inside the HTML page, remove input field length limits to allow extra-long strings in the text fields, remove JavaScript form validation to make Burp Proxy all onsubmit handler JavaScript functions from HTML forms, remove all JavaScript to completely remove all JS scripts and remove object tags to remove embedded objects within the HTML document. Select the desired checkboxes to activate automatic HTML modification. Summary Using this feature, you will be able to understand whether the web application enforces server- side validation. For instance, some insecure applications use client-side validation only (for example, via JavaScript functions). You can activate the automatic HTML modification feature by selecting the remove JavaScript form validation checkbox in order to perform input validation testing directly from your browser. Resources for Article : Further resources on this subject: Visual Studio 2010 Test Types [Article] Ordered and Generic Tests in Visual Studio 2010 [Article] Manual, Generic, and Ordered Tests using Visual Studio 2008 [Article]  
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03 Sep 2013
15 min read
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Scratching the Tip of the Iceberg

Packt
03 Sep 2013
15 min read
Boost is a huge collection of libraries. Some of those libraries are small and meant for everyday use and others require a separate article to describe all of their features. This article is devoted to some of those big libraries and to give you some basics to start with. The first two recipes will explain the usage of Boost.Graph. It is a big library with an insane number of algorithms. We'll see some basics and probably the most important part of it visualization of graphs. We'll also see a very useful recipe for generating true random numbers. This is a very important requirement for writing secure cryptography systems. Some C++ standard libraries lack math functions. We'll see how that can be fixed using Boost. But the format of this article leaves no space to describe all of the functions. Writing test cases is described in the Writing test cases and Combining multiple test cases in one test module recipes. This is important for any production-quality system. The last recipe is about a library that helped me in many courses during my university days. Images can be created and modified using it. I personally used it to visualize different algorithms, hide data in images, sign images, and generate textures. Unfortunately, even this article cannot tell you about all of the Boost libraries. Maybe someday I'll write another book... and then a few more. Working with graphs Some tasks require a graphical representation of data. Boost.Graph is a library that was designed to provide a flexible way of constructing and representing graphs in memory. It also contains a lot of algorithms to work with graphs, such as topological sort, breadth first search, depth first search, and Dijkstra shortest paths. Well, let's perform some basic tasks with Boost.Graph! Getting ready Only basic knowledge of C++ and templates is required for this recipe. How to do it... In this recipe, we'll describe a graph type, create a graph of that type, add some vertexes and edges to the graph, and search for a specific vertex. That should be enough to start using Boost.Graph. We start with describing the graph type: #include <boost/graph/adjacency_list.hpp> #include <string> typedef std::string vertex_t; typedef boost::adjacency_list< boost::vecS , boost::vecS , boost::bidirectionalS , vertex_t > graph_type; Now we construct it: graph_type graph; Let's use a non portable trick that speeds up graph construction: static const std::size_t vertex_count = 5; graph.m_vertices.reserve(vertex_count); Now we are ready to add vertexes to the graph: typedef boost::graph_traits<graph_type> ::vertex_descriptor descriptor_t; descriptor_t cpp = boost::add_vertex(vertex_t("C++"), graph); descriptor_t stl = boost::add_vertex(vertex_t("STL"), graph); descriptor_t boost = boost::add_vertex(vertex_t("Boost"), graph); descriptor_t guru = boost::add_vertex(vertex_t("C++ guru"), graph); descriptor_t ansic = boost::add_vertex(vertex_t("C"), graph); It is time to connect vertexes with edges: boost::add_edge(cpp, stl, graph); boost::add_edge(stl, boost, graph); boost::add_edge(boost, guru, graph); boost::add_edge(ansic, guru, graph); We make a function that searches for a vertex: template <class GraphT> void find_and_print(const GraphT& g, boost::string_ref name) { Now we will write code that gets iterators to all vertexes: typedef typename boost::graph_traits<graph_type> ::vertex_iterator vert_it_t; vert_it_t it, end; boost::tie(it, end) = boost::vertices(g); It's time to run a search for the required vertex: typedef boost::graph_traits<graph_type>::vertex_descriptor desc_t; for (; it != end; ++ it) { desc_t desc = *it; if (boost::get(boost::vertex_bundle, g)[desc] == name.data()) { break; } } assert(it != end); std::cout << name << 'n'; } /* find_and_print */ How it works... In step 1, we are describing what our graph must look like and upon what types it must be based. boost::adjacency_list is a class that represents graphs as a two-dimensional structure, where the first dimension contains vertexes and the second dimension contains edges for that vertex. boost::adjacency_list must be the default choice for representing a graph; it suits most cases. The first template parameter, boost::adjacency_list, describes the structure used to represent the edge list for each of the vertexes; the second one describes a structure to store vertexes. We can choose different STL containers for those structures using specific selectors, as listed in the following table: Selector STL container boost::vecS std::vector boost::listS std::list boost::slistS std::slist boost::setS std::set boost::multisetS std::multiset boost::hash_setS std::hash_set The third template parameter is used to make an undirected, directed, or bidirectional graph. Use the boost::undirectedS, boost::directedS, and boost::bidirectionalS selectors respectively. The fifth template parameter describes the datatype that will be used as the vertex. In our example, we chose std::string. We can also support a datatype for edges and provide it as a template parameter. Steps 2 and 3 are trivial, but at step 4 you will see a non portable way to speed up graph construction. In our example, we use std::vector as a container for storing vertexes, so we can force it to reserve memory for the required amount of vertexes. This leads to less memory allocations/deallocations and copy operations during insertion of vertexes into the graph. This step is non-portable because it is highly dependent on the current implementation of boost::adjacency_list and on the chosen container type for storing vertexes. At step 4, we see how vertexes can be added to the graph. Note how boost::graph_traits<graph_type> has been used. The boost::graph_traits class is used to get types that are specific for a graph type. We'll see its usage and the description of some graph-specific types later in this article. Step 5 shows what we need do to connect vertexes with edges. If we had provided a datatype for the edges, adding an edge would look as follows: boost::add_edge(ansic, guru, edge_t(initialization_parameters), graph) Note that at step 6 the graph type is a template parameter. This is recommended to achieve better code reusability and make this function work with other graph types. At step 7, we see how to iterate over all of the vertexes of the graph. The type of vertex iterator is received from boost::graph_traits. The function boost::tie is a part of Boost.Tuple and is used for getting values from tuples to the variables. So calling boost::tie(it, end) = boost::vertices(g) will put the begin iterator into the it variable and the end iterator into the end variable. It may come as a surprise to you, but dereferencing a vertex iterator does not return vertex data. Instead, it returns the vertex descriptor desc, which can be used in boost::get(boost::vertex_bundle, g)[desc] to get vertex data, just as we have done in step 8. The vertex descriptor type is used in many of the Boost.Graph functions; we saw its use in the edge construction function in step 5. As already mentioned, the Boost.Graph library contains the implementation of many algorithms. You will find many search policies implemented, but we won't discuss them in this article. We will limit this recipe to only the basics of the graph library. There's more... The Boost.Graph library is not a part of C++11 and it won't be a part of C++1y. The current implementation does not support C++11 features. If we are using vertexes that are heavy to copy, we may gain speed using the following trick: vertex_descriptor desc = boost::add_vertex(graph); boost::get(boost::vertex_bundle, g_)[desc] = std::move(vertex_data); It avoids copy constructions of boost::add_vertex(vertex_data, graph) and uses the default construction with move assignment instead. The efficiency of Boost.Graph depends on multiple factors, such as the underlying containers types, graph representation, edge, and vertex datatypes. Visualizing graphs Making programs that manipulate graphs was never easy because of issues with visualization. When we work with STL containers such as std::map and std::vector, we can always print the container's contents and see what is going on inside. But when we work with complex graphs, it is hard to visualize the content in a clear way: too many vertexes and too many edges. In this recipe, we'll take a look at the visualization of Boost.Graph using the Graphviz tool. Getting ready To visualize graphs, you will need a Graphviz visualization tool. Knowledge of the preceding recipe is also required. How to do it... Visualization is done in two phases. In the first phase, we make our program output the graph's description in a text format; in the second phase, we import the output from the first step to some visualization tool. The numbered steps in this recipe are all about the first phase. Let's write the std::ostream operator for graph_type as done in the preceding recipe: #include <boost/graph/graphviz.hpp> std::ostream& operator<<(std::ostream& out, const graph_type& g) { detail::vertex_writer<graph_type> vw(g); boost::write_graphviz(out, g, vw); return out; } The detail::vertex_writer structure, used in the preceding step, must be defined as follows: namespace detail { template <class GraphT> class vertex_writer { const GraphT& g_; public: explicit vertex_writer(const GraphT& g) : g_(g) {} template <class VertexDescriptorT> void operator()(std::ostream& out, const VertexDescriptorT& d) const { out << " [label="" << boost::get(boost::vertex_bundle, g_)[d] << ""]"; } }; // vertex_writer } // namespace detail That's all. Now, if we visualize the graph from the previous recipe using the std::cout << graph; command, the output can be used to create graphical pictures using the dot command-line utility: $ dot -Tpng -o dot.png digraph G { 0 [label="C++"]; 1 [label="STL"]; 2 [label="Boost"]; 3 [label="C++ guru"]; 4 [label="C"]; 0->1 ; 1->2 ; 2->3 ; 4->3 ; }   The output of the preceding command is depicted in the following figure: We can also use the Gvedit or XDot programs for visualization if the command line frightens you. How it works... The Boost.Graph library contains function to output graphs in Graphviz (DOT) format. If we write boost::write_graphviz(out, g) with two parameters in step 1, the function will output a graph picture with vertexes numbered from 0. That's not very useful, so we provide an instance of the vertex_writer class that outputs vertex names. As we can see in step 2, the format of output must be DOT, which is understood by the Graphviz tool. You may need to read the Graphviz documentation for more info about the DOT format. If you wish to add some data to the edges during visualization, we need to provide an instance of the edge visualizer as a fourth parameter to boost::write_graphviz. There's more... C++11 does not contain Boost.Graph or the tools for graph visualization. But you do not need to worry—there are a lot of other graph formats and visualization tools and Boost. Graph can work with plenty of them. Using a true random number generator I know of many examples of commercial products that use incorrect methods for getting random numbers. It's a shame that some companies still use rand() in cryptography and banking software. Let's see how to get a fully random uniform distribution using Boost.Random that is suitable for banking software. Getting ready Basic knowledge of C++ is required for this recipe. Knowledge of different types of distributions will also be helpful. The code in this recipe requires linking against the boost_random library. How to do it... To create a true random number, we need some help from the operating system or processor. This is how it can be done using Boost: We'll need to include the following headers: #include <boost/config.hpp> #include <boost/random/random_device.hpp> #include <boost/random/uniform_int_distribution.hpp> Advanced random number providers have different names under different platforms: static const std::string provider = #ifdef BOOST_WINDOWS "Microsoft Strong Cryptographic Provider" #else "/dev/urandom" #endif ; Now we are ready to initialize the generator with Boost.Random: boost::random_device device(provider); Let's get a uniform distribution that returns a value between 1000 and 65535: boost::random::uniform_int_distribution<unsigned short> random(1000); That's it. Now we can get true random numbers using the random(device) call. How it works... Why does the rand() function not suit banking? Because it generates pseudo-random numbers, which means that the hacker could predict the next generated number. This is an issue with all pseudo-random number algorithms. Some algorithms are easier to predict and some harder, but it's still possible. That's why we are using boost::random_device in this example (see step 3). That device gathers information about random events from all around the operating system to construct an unpredictable hardware-generated number. The examples of such events are delays between pressed keys, delays between some of the hardware interruptions, and the internal CPU random number generator. Operating systems may have more than one such type of random number generators. In our example for POSIX systems, we used /dev/urandom instead of the more secure /dev/random because the latter remains in a blocked state until enough random events have been captured by the OS. Waiting for entropy could take seconds, which is usually unsuitable for applications. Use /dev/random to create long-lifetime GPG/SSL/SSH keys. Now that we are done with generators, it's time to move to step 4 and talk about distribution classes. If the generator just generates numbers (usually uniformly distributed), the distribution class maps one distribution to another. In step 4, we made a uniform distribution that returns a random number of unsigned short type. The parameter 1000 means that distribution must return numbers greater or equal to 1000. We can also provide the maximum number as a second parameter, which is by default equal to the maximum value storable in the return type. There's more... Boost.Random has a huge number of true/pseudo random generators and distributions for different needs. Avoid copying distributions and generators; this could turn out to be an expensive operation. C++11 has support for different distribution classes and generators. You will find all of the classes from this example in the <random> header in the std:: namespace. The Boost.Random libraries do not use C++11 features, and they are not really required for that library either. Should you use Boost implementation or STL? Boost provides better portability across systems; however, some STL implementations may have assembly-optimized implementations and might provide some useful extensions. Using portable math functions Some projects require specific trigonometric functions, a library for numerically solving ordinary differential equations, and working with distributions and constants. All of those parts of Boost.Math would be hard to fit into even a separate book. A single recipe definitely won't be enough. So let's focus on very basic everyday-use functions to work with float types. We'll write a portable function that checks an input value for infinity and not-a-number (NaN) values and changes the sign if the value is negative. Getting ready Basic knowledge of C++ is required for this recipe. Those who know C99 standard will find a lot in common in this recipe. How to do it... Perform the following steps to check the input value for infinity and NaN values and change the sign if the value is negative: We'll need the following headers: #include <boost/math/special_functions.hpp> #include <cassert> Asserting for infinity and NaN can be done like this: template <class T> void check_float_inputs(T value) { assert(!boost::math::isinf(value)); assert(!boost::math::isnan(value)); Use the following code to change the sign: if (boost::math::signbit(value)) { value = boost::math::changesign(value); } // ... } // check_float_inputs That's it! Now we can check that check_float_inputs(std::sqrt(-1.0)) and check_float_inputs(std::numeric_limits<double>::max() * 2.0) will cause asserts. How it works... Real types have specific values that cannot be checked using equality operators. For example, if the variable v contains NaN, assert(v!=v) may or may not pass depending on the compiler. For such cases, Boost.Math provides functions that can reliably check for infinity and NaN values. Step 3 contains the boost::math::signbit function, which requires clarification. This function returns a signed bit, which is 1 when the number is negative and 0 when the number is positive. In other words, it returns true if the value is negative. Looking at step 3 some readers might ask, "Why can't we just multiply by -1 instead of calling boost::math::changesign?". We can. But multiplication may work slower than boost::math::changesign and won't work for special values. For example, if your code can work with nan, the code in step 3 will be able to change the sign of -nan and write nan to the variable. The Boost.Math library maintainers recommend wrapping math functions from this example in round parenthesis to avoid collisions with C macros. It is better to write (boost::math::isinf)(value) instead of boost::math::isinf(value). There's more... C99 contains all of the functions described in this recipe. Why do we need them in Boost? Well, some compiler vendors think that programmers do not need them, so you won't find them in one very popular compiler. Another reason is that the Boost.Math functions can be used for classes that behave like numbers. Boost.Math is a very fast, portable, reliable library.
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