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  • Polite busy-waiting with WRPAUSE on SPARC

    - by Dave
    Unbounded busy-waiting is an poor idea for user-space code, so we typically use spin-then-block strategies when, say, waiting for a lock to be released or some other event. If we're going to spin, even briefly, then we'd prefer to do so in a manner that minimizes performance degradation for other sibling logical processors ("strands") that share compute resources. We want to spin politely and refrain from impeding the progress and performance of other threads — ostensibly doing useful work and making progress — that run on the same core. On a SPARC T4, for instance, 8 strands will share a core, and that core has its own L1 cache and 2 pipelines. On x86 we have the PAUSE instruction, which, naively, can be thought of as a hardware "yield" operator which temporarily surrenders compute resources to threads on sibling strands. Of course this helps avoid intra-core performance interference. On the SPARC T2 our preferred busy-waiting idiom was "RD %CCR,%G0" which is a high-latency no-nop. The T4 provides a dedicated and extremely useful WRPAUSE instruction. The processor architecture manuals are the authoritative source, but briefly, WRPAUSE writes a cycle count into the the PAUSE register, which is ASR27. Barring interrupts, the processor then delays for the requested period. There's no need for the operating system to save the PAUSE register over context switches as it always resets to 0 on traps. Digressing briefly, if you use unbounded spinning then ultimately the kernel will preempt and deschedule your thread if there are other ready threads than are starving. But by using a spin-then-block strategy we can allow other ready threads to run without resorting to involuntary time-slicing, which operates on a long-ish time scale. Generally, that makes your application more responsive. In addition, by blocking voluntarily we give the operating system far more latitude regarding power management. Finally, I should note that while we have OS-level facilities like sched_yield() at our disposal, yielding almost never does what you'd want or naively expect. Returning to WRPAUSE, it's natural to ask how well it works. To help answer that question I wrote a very simple C/pthreads benchmark that launches 8 concurrent threads and binds those threads to processors 0..7. The processors are numbered geographically on the T4, so those threads will all be running on just one core. Unlike the SPARC T2, where logical CPUs 0,1,2 and 3 were assigned to the first pipeline, and CPUs 4,5,6 and 7 were assigned to the 2nd, there's no fixed mapping between CPUs and pipelines in the T4. And in some circumstances when the other 7 logical processors are idling quietly, it's possible for the remaining logical processor to leverage both pipelines. Some number T of the threads will iterate in a tight loop advancing a simple Marsaglia xor-shift pseudo-random number generator. T is a command-line argument. The main thread loops, reporting the aggregate number of PRNG steps performed collectively by those T threads in the last 10 second measurement interval. The other threads (there are 8-T of these) run in a loop busy-waiting concurrently with the T threads. We vary T between 1 and 8 threads, and report on various busy-waiting idioms. The values in the table are the aggregate number of PRNG steps completed by the set of T threads. The unit is millions of iterations per 10 seconds. For the "PRNG step" busy-waiting mode, the busy-waiting threads execute exactly the same code as the T worker threads. We can easily compute the average rate of progress for individual worker threads by dividing the aggregate score by the number of worker threads T. I should note that the PRNG steps are extremely cycle-heavy and access almost no memory, so arguably this microbenchmark is not as representative of "normal" code as it could be. And for the purposes of comparison I included a row in the table that reflects a waiting policy where the waiting threads call poll(NULL,0,1000) and block in the kernel. Obviously this isn't busy-waiting, but the data is interesting for reference. _table { border:2px black dotted; margin: auto; width: auto; } _tr { border: 2px red dashed; } _td { border: 1px green solid; } _table { border:2px black dotted; margin: auto; width: auto; } _tr { border: 2px red dashed; } td { background-color : #E0E0E0 ; text-align : right ; } th { text-align : left ; } td { background-color : #E0E0E0 ; text-align : right ; } th { text-align : left ; } Aggregate progress T = #worker threads Wait Mechanism for 8-T threadsT=1T=2T=3T=4T=5T=6T=7T=8 Park thread in poll() 32653347334833483348334833483348 no-op 415 831 124316482060249729303349 RD %ccr,%g0 "pause" 14262429269228623013316232553349 PRNG step 412 829 124616702092251029303348 WRPause(8000) 32443361333133483349334833483348 WRPause(4000) 32153308331533223347334833473348 WRPause(1000) 30853199322432513310334833483348 WRPause(500) 29173070315032223270330933483348 WRPause(250) 26942864294930773205338833483348 WRPause(100) 21552469262227902911321433303348

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  • The Linux powered LAN Gaming House

    - by sachinghalot
    LAN parties offer the enjoyment of head to head gaming in a real-life social environment. In general, they are experiencing decline thanks to the convenience of Internet gaming, but Kenton Varda is a man who takes his LAN gaming very seriously. His LAN gaming house is a fascinating project, and best of all, Linux plays a part in making it all work.Varda has done his own write ups (short, long), so I'm only going to give an overview here. The setup is a large house with 12 gaming stations and a single server computer.The client computers themselves are rack mounted in a server room, and they are linked to the gaming stations on the floor above via extension cables (HDMI for video and audio and USB for mouse and keyboard). Each client computer, built into a 3U rack mount case, is a well specced gaming rig in its own right, sporting an Intel Core i5 processor, 4GB of RAM and an Nvidia GeForce 560 along with a 60GB SSD drive.Originally, the client computers ran Ubuntu Linux rather than Windows and the games executed under WINE, but Varda had to abandon this scheme. As he explains on his site:"Amazingly, a majority of games worked fine, although many had minor bugs (e.g. flickering mouse cursor, minor rendering artifacts, etc.). Some games, however, did not work, or had bad bugs that made them annoying to play."Subsequently, the gaming computers have been moved onto a more conventional gaming choice, Windows 7. It's a shame that WINE couldn't be made to work, but I can sympathize as it's rare to find modern games that work perfectly and at full native speed. Another problem with WINE is that it tends to suffer from regressions, which is hardly surprising when considering the difficulty of constantly improving the emulation of the Windows API. Varda points out that he preferred working with Linux clients as they were easier to modify and came with less licensing baggage.Linux still runs the server and all of the tools used are open source software. The hardware here is a Intel Xeon E3-1230 with 4GB of RAM. The storage hanging off this machine is a bit more complex than the clients. In addition to the 60GB SSD, it also has 2x1TB drives and a 240GB SDD.When the clients were running Linux, they booted over PXE using a toolchain that will be familiar to anyone who has setup Linux network booting. DHCP pointed the clients to the server which then supplied PXELINUX using TFTP. When booted, file access was accomplished through network block device (NBD). This is a very easy to use system that allows you to serve the contents of a file as a block device over the network. The client computer runs a user mode device driver and the device can be mounted within the file system using the mount command.One snag with offering file access via NBD is that it's difficult to impose any security restrictions on different areas of the file system as the server only sees a single file. The advantage is perfomance as the client operating system simply sees a block device, and besides, these security issues aren't relevant in this setup.Unfortunately, Windows 7 can't use NBD, so, Varda had to switch to iSCSI (which works in both server and client mode under Linux). His network cards are not compliant with this standard when doing a netboot, but fortunately, gPXE came to the rescue, and he boostraps it over PXE. gPXE is also available as an ISO image and is worth knowing about if you encounter an awkward machine that can't manage a network boot. It can also optionally boot from a HTTP server rather than the more traditional TFTP server.According to Varda, booting all 12 machines over the Gigabit Ethernet network is surprisingly fast, and once booted, the machines don't seem noticeably slower than if they were using local storage. Once loaded, most games attempt to load in as much data as possible, filling the RAM, and the the disk and network bandwidth required is small. It's worth noting that these are aspects of this project that might differ from some other thin client scenarios.At time of writing, it doesn't seem as though the local storage of the client machines is being utilized. Instead, the clients boot into Windows from an image on the server that contains the operating system and the games themselves. It uses the copy on write feature of LVM so that any writes from a client are added to a differencing image allocated to that client. As the administrator, Varda can log into the Linux server and authorize changes to the master image for updates etc.SummaryOverall, Varda estimates the total cost of the project at about $40,000, and of course, he needed a property that offered a large physical space in order to house the computers and the gaming workstations. Obviously, this project has stark differences to most thin client projects. The balance between storage, network usage, GPU power and security would not be typical of an office installation, for example. The only letdown is that WINE proved to be insufficiently compatible to run a wide variety of modern games, but that is, perhaps, asking too much of it, and hats off to Varda for trying to make it work.

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  • SOA 11g Technology Adapters – ECID Propagation

    - by Greg Mally
    Overview Many SOA Suite 11g deployments include the use of the technology adapters for various activities including integration with FTP, database, and files to name a few. Although the integrations with these adapters are easy and feature rich, there can be some challenges from the operations perspective. One of these challenges is how to correlate a logical business transaction across SOA component instances. This correlation is typically accomplished via the execution context ID (ECID), but we lose the ECID correlation when the business transaction spans technologies like FTP, database, and files. A new feature has been introduced in the Oracle adapter JCA framework to allow the propagation of the ECID. This feature is available in the forthcoming SOA Suite 11.1.1.7 (PS6). The basic concept of propagating the ECID is to identify somewhere in the payload of the message where the ECID can be stored. Then two Binding Properties, relating to the location of the ECID in the message, are added to either the Exposed Service (left-hand side of composite) or External Reference (right-hand side of composite). This will give the JCA framework enough information to either extract the ECID from or add the ECID to the message. In the scenario of extracting the ECID from the message, the ECID will be used for the new component instance. Where to Put the ECID When trying to determine where to store the ECID in the message, you basically have two options: Add a new optional element to your message schema. Leverage an existing element that is not used in your schema. The best scenario is that you are able to add the optional element to your message since trying to find an unused element will prove difficult in most situations. The schema will be holding the ECID value which looks something like the following: 11d1def534ea1be0:7ae4cac3:13b4455735c:-8000-00000000000002dc Configuring Composite Services/References Now that you have identified where you want the ECID to be stored in the message, the JCA framework needs to have this information as well. The two pieces of information that the framework needs relates to the message schema: The namespace for the element in the message. The XPath to the element in the message. To better understand this, let's look at an example for the following database table: When an Exposed Service is created via the Database Adapter Wizard in the composite, the following schema is created: For this example, the two Binding Properties we add to the ReadRow service in the composite are: <!-- Properties for the binding to propagate the ECID from the database table --> <property name="jca.ecid.nslist" type="xs:string" many="false">  xmlns:ns1="http://xmlns.oracle.com/pcbpel/adapter/db/top/ReadRow"</property> <property name="jca.ecid.xpath" type="xs:string" many="false">  /ns1:EcidPropagationCollection/ns1:EcidPropagation/ns1:ecid</property> Notice that the property called jca.ecid.nslist contains the targetNamespace defined in the schema and the property called jca.ecid.xpath contains the XPath statement to the element. The XPath statement also contains the appropriate namespace prefix (ns1) which is defined in the jca.ecid.nslist property. When the Database Adapter service reads a row from the database, it will retrieve the ECID value from the payload and remove the element from the payload. When the component instance is created, it will be associated with the retrieved ECID and the payload contains everything except the ECID element/value. The only time the ECID is visible is when it is stored safely in the resource technology like the database, a file, or a queue. Simple Database/File/JMS Example This section contains a simplified example of how the ECID can propagate through a database table, a file, and JMS queue. The composite for the example looks like the following: The flow of this example is as follows: Invoke database insert using the insertwithecidbpelprocess_client_ep Service. The InsertWithECIDBPELProcess adds a row to the database via the Database Adapter. The JCA Framework adds the ECID to the message prior to inserting. The ReadRow Service retrieves the record and the JCA Framework extracts the ECID from the message. The ECID element is removed from the message. An instance of ReadRowBPELProcess is created and it is associated with the retried ECID. The ReadRowBPELProcess now writes the record to the file system via the File Adapter. The JCA Framework adds the ECID to the message prior to writing the message to file. The ReadFile Service retrieves the record from the file system and the JCA Framework extracts the ECID from the message. The ECID element is removed from the message. An instance of ReadFileBPELProcess is created and it is associated with the retried ECID. The ReadFileBPELProcess now enqueues the message via the JMS Adapter. The JCA Framework adds the ECID to the message prior to enqueuing the message. The DequeueMessage Service retrieves the record and the JCA Framework extracts the ECID from the message. The ECID element is removed from the message. An instance of DequeueMessageBPELProcess is created and it is associated with the retried ECID. The logical flow ends. When viewing the Flow Trace in the Enterprise Manger, you will now see all the instances correlated via ECID: Please check back here when SOA Suite 11.1.1.7 is released for this example. With the example you can run it yourself and reinforce what has been shared in this blog via a hands-on experience. One final note: the contents of this blog may be included in the official SOA Suite 11.1.1.7 documentation, but you will still need to come here to get the example.

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  • Big Data – Buzz Words: What is HDFS – Day 8 of 21

    - by Pinal Dave
    In yesterday’s blog post we learned what is MapReduce. In this article we will take a quick look at one of the four most important buzz words which goes around Big Data – HDFS. What is HDFS ? HDFS stands for Hadoop Distributed File System and it is a primary storage system used by Hadoop. It provides high performance access to data across Hadoop clusters. It is usually deployed on low-cost commodity hardware. In commodity hardware deployment server failures are very common. Due to the same reason HDFS is built to have high fault tolerance. The data transfer rate between compute nodes in HDFS is very high, which leads to reduced risk of failure. HDFS creates smaller pieces of the big data and distributes it on different nodes. It also copies each smaller piece to multiple times on different nodes. Hence when any node with the data crashes the system is automatically able to use the data from a different node and continue the process. This is the key feature of the HDFS system. Architecture of HDFS The architecture of the HDFS is master/slave architecture. An HDFS cluster always consists of single NameNode. This single NameNode is a master server and it manages the file system as well regulates access to various files. In additional to NameNode there are multiple DataNodes. There is always one DataNode for each data server. In HDFS a big file is split into one or more blocks and those blocks are stored in a set of DataNodes. The primary task of the NameNode is to open, close or rename files and directory and regulate access to the file system, whereas the primary task of the DataNode is read and write to the file systems. DataNode is also responsible for the creation, deletion or replication of the data based on the instruction from NameNode. In reality, NameNode and DataNode are software designed to run on commodity machine build in Java language. Visual Representation of HDFS Architecture Let us understand how HDFS works with the help of the diagram. Client APP or HDFS Client connects to NameSpace as well as DataNode. Client App access to the DataNode is regulated by NameSpace Node. NameSpace Node allows Client App to connect to the DataNode based by allowing the connection to the DataNode directly. A big data file is divided into multiple data blocks (let us assume that those data chunks are A,B,C and D. Client App will later on write data blocks directly to the DataNode. Client App does not have to directly write to all the node. It just has to write to any one of the node and NameNode will decide on which other DataNode it will have to replicate the data. In our example Client App directly writes to DataNode 1 and detained 3. However, data chunks are automatically replicated to other nodes. All the information like in which DataNode which data block is placed is written back to NameNode. High Availability During Disaster Now as multiple DataNode have same data blocks in the case of any DataNode which faces the disaster, the entire process will continue as other DataNode will assume the role to serve the specific data block which was on the failed node. This system provides very high tolerance to disaster and provides high availability. If you notice there is only single NameNode in our architecture. If that node fails our entire Hadoop Application will stop performing as it is a single node where we store all the metadata. As this node is very critical, it is usually replicated on another clustered as well as on another data rack. Though, that replicated node is not operational in architecture, it has all the necessary data to perform the task of the NameNode in the case of the NameNode fails. The entire Hadoop architecture is built to function smoothly even there are node failures or hardware malfunction. It is built on the simple concept that data is so big it is impossible to have come up with a single piece of the hardware which can manage it properly. We need lots of commodity (cheap) hardware to manage our big data and hardware failure is part of the commodity servers. To reduce the impact of hardware failure Hadoop architecture is built to overcome the limitation of the non-functioning hardware. Tomorrow In tomorrow’s blog post we will discuss the importance of the relational database in Big Data. Reference: Pinal Dave (http://blog.sqlauthority.com) Filed under: Big Data, PostADay, SQL, SQL Authority, SQL Query, SQL Server, SQL Tips and Tricks, T SQL

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  • How to tell whether your programmers are under-performing?

    - by A Team Lead
    I am a team lead with 5+ developers. I have a developer (let's call him A) who is a good programmer, who writes good clean, easy to understand code. However he is somewhat difficult to manage, and sometimes I wonder whether he is really under-performing or not. Our company requires the developers to indicate the work progress in the bug tracker we use, not so much as to monitor the programmers but to let the stackholders know the progress. The thing is, A only updates a task progress when it is done ( maybe 3 weeks after it is first worked on) and this leaves everyone wondering what is going on in the middle of the development week. He wouldn't change his habit despite repeated probing. ( It's OK, developers hate paperwork, I do, too) Recent 2-3 months he on leave quite often due to various events-- either he is sick, or have to attend a lot of personal events etc. ( It's OK, bad things happen in a string. It's just a coincidence) We define sprints, or roadmaps for each month. And in the beginning of the sprint, we will discuss the amount of work each of the developers have to do in a sprint and the developers get to set the amount of time they need for each task. He usually won't be able to complete all of them. (It's OK, the developers are regularly missing deadlines not due to their fault). If only one or two of the above events happen, I won't feel that A is under-performing, but they all happen together. So I have the feeling that A is under-performing and maybe-- God forbid--- slacking off. This is just a feeling based on my years of experience as programmer. But I could be wrong. It is notoriously hard to measure the work of a programmer, given that not all two tasks are alike, and there lacks a standard objective to measure the commitment of a programmer to your company. It is downright impossible to tell whether the programmer is doing his job or slacking off. All you can do, is to trust them-- yeah, trusting and giving them autonomy is the best way for programmers to work, I know that, so don't start a lecture on why you need to trust your programmers, thank you every much-- but if they abuse your trust, can you know? My question is, how can you tell whether your programmers are under-performing? Surely there are experience team leads who know better than me on this? Outcome: I've a straight talk with him regarding my perception on his performance. He was indignant when I suggested that I had the feeling that he wasn't performing at his best level. He felt that this was a completely unfair feeling. I then replied that this was my feeling and I didn't know whether my feeling was right or not. He would have none of this and ended the discussion immediately. Before he left he said that he "would try to give more to the company" in a very cold tone. I was taken aback by his reaction. I am sure that I offended him in some ways. Not too sure whether that was the right thing to do for me to be so frank with him, though. Extra notes: I hate micromanaging. So all that we have for our software process is Sprint ( where tasks get prioritized and assigned, and at the end of the month, a review of the amount of work done). Developers would require to update the tasks as they go along everyday. There is no standup meeting, or anything of the sort. Mainly because we have the freedom to work from home and everyone cherishes this freedom. Although I am the one who sets the deadline, but the developers will provide the estimate for each tasks and I will decide-- based on the estimate-- the tasks that go into a particular sprint. If they can't finish the tasks at the end of the sprint, I will push them to the next. So theoretically one can just do only 1 or 2 tasks during the whole sprint and then push the remaining 99 tasks to the next sprint and still he will be fine as long as justifies this-- in the form of daily work progress updates

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  • How many developers before continuous integration becomes effective for us?

    - by Carnotaurus
    There is an overhead associated with continuous integration, e.g., set up, re-training, awareness activities, stoppage to fix "bugs" that turn out to be data issues, enforced separation of concerns programming styles, etc. At what point does continuous integration pay for itself? EDIT: These were my findings The set-up was CruiseControl.Net with Nant, reading from VSS or TFS. Here are a few reasons for failure, which have nothing to do with the setup: Cost of investigation: The time spent investigating whether a red light is due a genuine logical inconsistency in the code, data quality, or another source such as an infrastructure problem (e.g., a network issue, a timeout reading from source control, third party server is down, etc., etc.) Political costs over infrastructure: I considered performing an "infrastructure" check for each method in the test run. I had no solution to the timeout except to replace the build server. Red tape got in the way and there was no server replacement. Cost of fixing unit tests: A red light due to a data quality issue could be an indicator of a badly written unit test. So, data dependent unit tests were re-written to reduce the likelihood of a red light due to bad data. In many cases, necessary data was inserted into the test environment to be able to accurately run its unit tests. It makes sense to say that by making the data more robust then the test becomes more robust if it is dependent on this data. Of course, this worked well! Cost of coverage, i.e., writing unit tests for already existing code: There was the problem of unit test coverage. There were thousands of methods that had no unit tests. So, a sizeable amount of man days would be needed to create those. As this would be too difficult to provide a business case, it was decided that unit tests would be used for any new public method going forward. Those that did not have a unit test were termed 'potentially infra red'. An intestesting point here is that static methods were a moot point in how it would be possible to uniquely determine how a specific static method had failed. Cost of bespoke releases: Nant scripts only go so far. They are not that useful for, say, CMS dependent builds for EPiServer, CMS, or any UI oriented database deployment. These are the types of issues that occured on the build server for hourly test runs and overnight QA builds. I entertain that these to be unnecessary as a build master can perform these tasks manually at the time of release, esp., with a one man band and a small build. So, single step builds have not justified use of CI in my experience. What about the more complex, multistep builds? These can be a pain to build, especially without a Nant script. So, even having created one, these were no more successful. The costs of fixing the red light issues outweighed the benefits. Eventually, developers lost interest and questioned the validity of the red light. Having given it a fair try, I believe that CI is expensive and there is a lot of working around the edges instead of just getting the job done. It's more cost effective to employ experienced developers who do not make a mess of large projects than introduce and maintain an alarm system. This is the case even if those developers leave. It doesn't matter if a good developer leaves because processes that he follows would ensure that he writes requirement specs, design specs, sticks to the coding guidelines, and comments his code so that it is readable. All this is reviewed. If this is not happening then his team leader is not doing his job, which should be picked up by his manager and so on. For CI to work, it is not enough to just write unit tests, attempt to maintain full coverage, and ensure a working infrastructure for sizable systems. The bottom line: One might question whether fixing as many bugs before release is even desirable from a business prespective. CI involves a lot of work to capture a handful of bugs that the customer could identify in UAT or the company could get paid for fixing as part of a client service agreement when the warranty period expires anyway.

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  • Developing Schema Compare for Oracle (Part 5): Query Snapshots

    - by Simon Cooper
    If you've emailed us about a bug you've encountered with the EAP or beta versions of Schema Compare for Oracle, we probably asked you to send us a query snapshot of your databases. Here, I explain what a query snapshot is, and how it helps us fix your bug. Problem 1: Debugging users' bug reports When we started the Schema Compare project, we knew we were going to get problems with users' databases - configurations we hadn't considered, features that weren't installed, unicode issues, wierd dependencies... With SQL Compare, users are generally happy to send us a database backup that we can restore using a single RESTORE DATABASE command on our test servers and immediately reproduce the problem. Oracle, on the other hand, would be a lot more tricky. As Oracle generally has a 1-to-1 mapping between instances and databases, any databases users sent would have to be restored to their own instance. Furthermore, the number of steps required to get a properly working database, and the size of most oracle databases, made it infeasible to ask every customer who came across a bug during our beta program to send us their databases. We also knew that there would be lots of issues with data security that would make it hard to get backups. So we needed an easier way to be able to debug customers issues and sort out what strange schema data Oracle was returning. Problem 2: Test execution time Another issue we knew we would have to solve was the execution time of the tests we would produce for the Schema Compare engine. Our initial prototype showed that querying the data dictionary for schema information was going to be slow (at least 15 seconds per database), and this is generally proportional to the size of the database. If you're running thousands of tests on the same databases, each one registering separate schemas, not only would the tests would take hours and hours to run, but the test servers would be hammered senseless. The solution To solve these, we needed to be able to populate the schema of a database without actually connecting to it. Well, the IDataReader interface is the primary way we read data from an Oracle server. The data dictionary queries we use return their data in terms of simple strings and numbers, which we then process and reconstruct into an object model, and the results of these queries are identical for identical schemas. So, we can record the raw results of the queries once, and then replay these results to construct the same object model as many times as required without needing to actually connect to the original database. This is what query snapshots do. They are binary files containing the raw unprocessed data we get back from the oracle server for all the queries we run on the data dictionary to get schema information. The core of the query snapshot generation takes the results of the IDataReader we get from running queries on Oracle, and passes the row data to a BinaryWriter that writes it straight to a file. The query snapshot can then be replayed to create the same object model; when the results of a specific query is needed by the population code, we can simply read the binary data stored in the file on disk and present it through an IDataReader wrapper. This is far faster than querying the server over the network, and allows us to run tests in a reasonable time. They also allow us to easily debug a customers problem; using a simple snapshot generation program, users can generate a query snapshot that could be sent along with a bug report that we can immediately replay on our machines to let us debug the issue, rather than having to obtain database backups and restore databases to test systems. There are also far fewer problems with data security; query snapshots only contain schema information, which is generally less sensitive than table data. Query snapshots implementation However, actually implementing such a feature did have a couple of 'gotchas' to it. My second blog post detailed the development of the dependencies algorithm we use to ensure we get all the dependencies in the database, and that algorithm uses data from both databases to find all the needed objects - what database you're comparing to affects what objects get populated from both databases. We get information on these additional objects using an appropriate WHERE clause on all the population queries. So, in order to accurately replay the results of querying the live database, the query snapshot needs to be a snapshot of a comparison of two databases, not just populating a single database. Furthermore, although the code population queries (eg querying all_tab_cols to get column information) can simply be passed straight from the IDataReader to the BinaryWriter, we need to hook into and run the live dependencies algorithm while we're creating the snapshot to ensure we get the same WHERE clauses, and the same query results, as if we were populating straight from a live system. We also need to store the results of the dependencies queries themselves, as the resulting dependency graph is stored within the OracleDatabase object that is produced, and is later used to help order actions in synchronization scripts. This is significantly helped by the dependencies algorithm being a deterministic algorithm - given the same input, it will always return the same output. Therefore, when we're replaying a query snapshot, and processing dependency information, we simply have to return the results of the queries in the order we got them from the live database, rather than trying to calculate the contents of all_dependencies on the fly. Query snapshots are a significant feature in Schema Compare that really helps us to debug problems with the tool, as well as making our testers happier. Although not really user-visible, they are very useful to the development team to help us fix bugs in the product much faster than we otherwise would be able to.

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  • Date Tracking in Oracle HRMS

    - by Manoj Madhusoodanan
    Update Date Track Modes To maintain employee data effectively Oracle HCM is using a mechanism called date tracking.The main motive behind the date track mode is to maintain past,present and future data effectively.The various update date track modes are: CORRECTION : Over writes the data. No history will maintain.UPDATE : Keeps the history and new change will effect as of effective dateUPDATE_CHANGE_INSERT : Inserts the record and preserves the futureUPDATE_OVERRIDE : Inserts the record and overrides the future Normal 0 false false false EN-US X-NONE X-NONE MicrosoftInternetExplorer4 /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Table Normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-priority:99; mso-style-qformat:yes; mso-style-parent:""; mso-padding-alt:0in 5.4pt 0in 5.4pt; mso-para-margin:0in; mso-para-margin-bottom:.0001pt; mso-pagination:widow-orphan; font-size:11.0pt; font-family:"Calibri","sans-serif"; mso-ascii-font-family:Calibri; mso-ascii-theme-font:minor-latin; mso-fareast-font-family:"Times New Roman"; mso-fareast-theme-font:minor-fareast; mso-hansi-font-family:Calibri; mso-hansi-theme-font:minor-latin; mso-bidi-font-family:"Times New Roman"; mso-bidi-theme-font:minor-bidi;} Action: Created Employee # 22 on 01-JAN-2012 The record in PER_ALL_PEOPLE_F is as shown below. Effective Start Date Effective End Date Employee Number Marital Status Object Version Number 01-JAN-2012 31-DEC-4712 24 2 Action: Updated record in CORRECTION mode Effective Start Date Effective End Date Employee Number Marital Status Object Version Number 01-JAN-2012 31-DEC-4712 24 Single 3 Action: Updated record in UPDATE mode effective 01-JUN-2012 and Marital Status = Married Effective Start Date Effective End Date Employee Number Marital Status Object Version Number 01-JAN-2012 31-MAY-2012 24 Single 4 01-JUN-2012 31-DEC-4712 24 Married 5 Action: Updated record in UPDATE mode effective 01-SEP-2012 and Marital Status = Divorced Effective Start Date Effective End Date Employee Number Marital Status Object Version Number 01-JAN-2012 31-MAY-2012 24 Single 4 01-JUN-2012 31-AUG-2012 24 Married 6 01-SEP-2012 31-DEC-4712 24 Divorced 7 Action: Updated record in UPDATE_CHANGE_INSERT mode effective 01-MAR-2012 and Marital Status = Living Together Effective Start Date Effective End Date Employee Number Marital Status Object Version Number 01-JAN-2012 29-FEB-2012 24 Single 8 01-MAR-2012 31-MAY-2012 24 Living Together 9 01-JUN-2012 31-AUG-2012 24 Married 6 01-SEP-2012 31-DEC-4712 24 Divorced 7 Action: Updated record in UPDATE_OVERRIDE mode effective 01-AUG-2012 and Marital Status = Divorced Effective Start Date Effective End Date Employee Number Marital Status Object Version Number 01-JAN-2012 29-FEB-2012 24 Single 8 01-MAR-2012 31-MAY-2012 24 Living Together 9 01-JUN-2012 31-JUL-2012 24 Married 10 01-AUG-2012 31-DEC-4712 24 Divorced 11  Delete Date Track Modes The various delete date track modes are ZAP : wipes all recordsDELETE : Deletes  current recordFUTURE_CHANGE : Deletes current and future changes.DELETE_NEXT_CHANGE : Deletes next change Normal 0 false false false EN-US X-NONE X-NONE MicrosoftInternetExplorer4 /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Table Normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-priority:99; mso-style-qformat:yes; mso-style-parent:""; mso-padding-alt:0in 5.4pt 0in 5.4pt; mso-para-margin:0in; mso-para-margin-bottom:.0001pt; mso-pagination:widow-orphan; font-size:11.0pt; font-family:"Calibri","sans-serif"; mso-ascii-font-family:Calibri; mso-ascii-theme-font:minor-latin; mso-fareast-font-family:"Times New Roman"; mso-fareast-theme-font:minor-fareast; mso-hansi-font-family:Calibri; mso-hansi-theme-font:minor-latin; mso-bidi-font-family:"Times New Roman"; mso-bidi-theme-font:minor-bidi;} Element Entry records are shown below. Effective Start Date Effective End Date Element Entry Id Object Version Number 01-JAN-2012 12-OCT-2012 129831 3 13-OCT-2012 19-OCT-2012 129831 5 20-OCT-2012 31-DEC-4712 129831 6 Action: Delete record in ZAP mode effective 14-JAN-2012 No rows Action: Delete record in DELETE mode effective 14-OCT-2012 Effective Start Date Effective End Date Element Entry Id Object Version Number 01-JAN-2012 12-OCT-2012 129831 3 13-OCT-2012 14-OCT-2012 129831 6 Action: Delete record in FUTURE_CHANGE mode effective 14-JAN-2012 Effective Start Date Effective End Date Element Entry Id Object Version Number 01-JAN-2012 31-DEC-4712 129831 4 Action: Delete record in NEXT_CHANGE mode effective 14-JAN-2012 Effective Start Date Effective End Date Element Entry Id Object Version Number 01-JAN-2012 19-OCT-2012 129831 4 20-OCT-2012 31-DEC-4712 129831 6

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  • Using Definition of Done to Drive Agile Maturity

    - by Dylan Smith
    I’ve been an Agile Coach at a lot of different clients over the years, and I want to share an approach I use to help them adopt and mature over time. It’s important to realize that “Agile” is not a black/white yes/no thing. Teams can be varying degrees of agile. I think of this as their agile maturity level. When I coach teams I want them to start out being a little agile, and get more agile as they mature. The approach I teach them is to use the definition of done as a technique to continuously improve their agile maturity over time. We’re probably all familiar with the concept of “Done Done” that represents what *actually* being done a feature means. Not just when a developer says he’s done right after he writes that last line of code that makes the feature kind-of work. Done Done means the coding is done, it’s been tested, installers and deployment packages have been created, user manuals have been updated, architecture docs have been updated, etc. To enable teams to internalize the concept of “Done Done”, they usually get together and come up with their Definition of Done (DoD) that defines all the activities that need to be completed before a feature is considered Done Done. The Done Done technique typically is applied only to features (aka User Stories). What I do is extend this to apply to several concepts such as User Stories, Sprints, Releases (and sometimes Check-Ins). During project kick-off I’ll usually sit down with the team and go through an exercise of creating DoD’s for each of these concepts (Stories/Sprints/Releases). We’ll usually start by just brainstorming a bunch of activities that could end up in these various DoD’s. Here’s some examples: Code Reviews StyleCop FxCop User Manuals Updated Architecture Docs Updated Tested by QA Tested by UAT Installers Created Support Knowledge Base Updated Deployment Instructions (for Ops) written Automated Unit Tests Run Automated Integration Tests Run Then we start by arranging these activities into the place they occur today (e.g. Do you do UAT testing only once per release? every sprint? every feature?). If the team was previously Waterfall most of these activities probably end up in the Release DoD. An extremely mature agile team would probably have most of these activities in the DoD for the User Stories (because an extremely mature agile team will probably do continuous deployment and release every story). So what we need to do as a team, is work to move these activities from their current home (Release DoD) down into the Sprint DoD and eventually into the User Story DoD (and maybe into the lower-level Check-In DoD if we decide to use that). We don’t have to move them all down to User Story immediately, but as a team we figure out what we think we’re capable of moving down to the Sprint cycle, and Story cycle immediately, and that becomes our starting DoD’s. Over time the team makes an effort to continue moving activities down from Release->Sprint->Story as they become more agile and more mature. I try to encourage them to envision a world in which they deploy to production as each User Story is completed. They would need to be updating User Manuals, creating installers, doing UAT testing (typical Release cycle activities) on every single User Story. They may never actually reach that point, but they should envision that, and strive to keep driving the activities down closer to the User Story cycle s they mature. This is a great technique to give a team an easy-to-follow roadmap to mature their agile practices over time. Sure there’s other aspects to maturity outside of this, but it’s a great technique, that’s easy to visualize, to drive agility into the team. Just keep moving those activities (aka “gates”) down the board from Release->Sprint->Story. I’ll try to give an example of what a recent client of mine had for their DoD’s (this is from memory, so probably not 100% accurate): Release Create/Update deployment Instructions For Ops Instructional Videos Updated Run manual regression test suite UAT Testing In this case that meant deploying to an environment shared across the enterprise that mirrored production and asking other business groups to test their own apps to ensure we didn’t break anything outside our system Sprint Deploy to UAT Environment But not necessarily actually request UAT testing occur User Guides updated Sprint Features Video Created In this case we decided to create a video each sprint showing off the progress (video version of Sprint Demo) User Story Manual Test scripts developed and run Tested by BA Deployed in shared QA environment Using automated deployment process Peer Code Review Code Check-In Compiled (warning-free) Passes StyleCop Passes FxCop Create installer packages Run Automated Tests Run Automated Integration Tests PS – One of my clients had a great question when we went through this activity. They said that if a Sprint is by definition done when the end-date rolls around (time-boxed), isn’t a DoD on a sprint meaningless – it’s done on the end-date regardless of whether those other activities are complete or not? My answer is that while that statement is true – the sprint is done regardless when the end date rolls around – if the DoD activities haven’t been completed I would consider the Sprint a failure (similar to not completing what was committed/planned – failure may be too strong a word but you get the idea). In the Retrospective that will become an agenda item to discuss and understand why we weren’t able to complete the activities we agreed would need to be completed each Sprint.

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  • Deduping your redundancies

    - by nospam(at)example.com (Joerg Moellenkamp)
    Robin Harris of Storagemojo pointed to an interesting article about about deduplication and it's impact to the resiliency of your data against data corruption on ACM Queue. The problem in short: A considerable number of filesystems store important metadata at multiple locations. For example the ZFS rootblock is copied to three locations. Other filesystems have similar provisions to protect their metadata. However you can easily proof, that the rootblock pointer in the uberblock of ZFS for example is pointing to blocks with absolutely equal content in all three locatition (with zdb -uu and zdb -r). It has to be that way, because they are protected by the same checksum. A number of devices offer block level dedup, either as an option or as part of their inner workings. However when you store three identical blocks on them and the devices does block level dedup internally, the device may just deduplicated your redundant metadata to a block stored just once that is stored on the non-voilatile storage. When this block is corrupted, you have essentially three corrupted copies. Three hit with one bullet. This is indeed an interesting problem: A device doing deduplication doesn't know if a block is important or just a datablock. This is the reason why I like deduplication like it's done in ZFS. It's an integrated part and so important parts don't get deduplicated away. A disk accessed by a block level interface doesn't know anything about the importance of a block. A metadata block is nothing different to it's inner mechanism than a normal data block because there is no way to tell that this is important and that those redundancies aren't allowed to fall prey to some clever deduplication mechanism. Robin talks about this in regard of the Sandforce disk controllers who use a kind of dedup to reduce some of the nasty effects of writing data to flash, but the problem is much broader. However this is relevant whenever you are using a device with block level deduplication. It's just the point that you have to activate it for most implementation by command, whereas certain devices do this by default or by design and you don't know about it. However I'm not perfectly sure about that ? given that storage administration and server administration are often different groups with different business objectives I would ask your storage guys if they have activated dedup without telling somebody elase on their boxes in order to speak less often with the storage sales rep. The problem is even more interesting with ZFS. You may use ditto blocks to protect important data to store multiple copies of data in the pool to increase redundancy, even when your pool just consists out of one disk or just a striped set of disk. However when your device is doing dedup internally it may remove your redundancy before it hits the nonvolatile storage. You've won nothing. Just spend your disk quota on the the LUNs in the SAN and you make your disk admin happy because of the good dedup ratio However you can just fall in this specific "deduped ditto block"trap when your pool just consists out of a single device, because ZFS writes ditto blocks on different disks, when there is more than just one disk. Yet another reason why you should spend some extra-thought when putting your zpool on a single LUN, especially when the LUN is sliced and dices out of a large heap of storage devices by a storage controller. However I have one problem with the articles and their specific mention of ZFS: You can just hit by this problem when you are using the deduplicating device for the pool. However in the specifically mentioned case of SSD this isn't the usecase. Most implementations of SSD in conjunction with ZFS are hybrid storage pools and so rotating rust disk is used as pool and SSD are used as L2ARC/sZIL. And there it simply doesn't matter: When you really have to resort to the sZIL (your system went down, it doesn't matter of one block or several blocks are corrupt, you have to fail back to the last known good transaction group the device. On the other side, when a block in L2ARC is corrupt, you simply read it from the pool and in HSP implementations this is the already mentioned rust. In conjunction with ZFS this is more interesting when using a storage array, that is capable to do dedup and where you use LUNs for your pool. However as mentioned before, on those devices it's a user made decision to do so, and so it's less probable that you deduplicating your redundancies. Other filesystems lacking acapability similar to hybrid storage pools are more "haunted" by this problem of SSD using dedup-like mechanisms internally, because those filesystem really store the data on the the SSD instead of using it just as accelerating devices. However at the end Robin is correct: It's jet another point why protecting your data by creating redundancies by dispersing it several disks (by mirror or parity RAIDs) is really important. No dedup mechanism inside a device can dedup away your redundancy when you write it to a totally different and indepenent device.

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  • Using Subjects to Deploy Queries Dynamically

    - by Roman Schindlauer
    In the previous blog posting, we showed how to construct and deploy query fragments to a StreamInsight server, and how to re-use them later. In today’s posting we’ll integrate this pattern into a method of dynamically composing a new query with an existing one. The construct that enables this scenario in StreamInsight V2.1 is a Subject. A Subject lets me create a junction element in an existing query that I can tap into while the query is running. To set this up as an end-to-end example, let’s first define a stream simulator as our data source: var generator = myApp.DefineObservable(     (TimeSpan t) => Observable.Interval(t).Select(_ => new SourcePayload())); This ‘generator’ produces a new instance of SourcePayload with a period of t (system time) as an IObservable. SourcePayload happens to have a property of type double as its payload data. Let’s also define a sink for our example—an IObserver of double values that writes to the console: var console = myApp.DefineObserver(     (string label) => Observer.Create<double>(e => Console.WriteLine("{0}: {1}", label, e)))     .Deploy("ConsoleSink"); The observer takes a string as parameter which is used as a label on the console, so that we can distinguish the output of different sink instances. Note that we also deploy this observer, so that we can retrieve it later from the server from a different process. Remember how we defined the aggregation as an IQStreamable function in the previous article? We will use that as well: var avg = myApp     .DefineStreamable((IQStreamable<SourcePayload> s, TimeSpan w) =>         from win in s.TumblingWindow(w)         select win.Avg(e => e.Value))     .Deploy("AverageQuery"); Then we define the Subject, which acts as an observable sequence as well as an observer. Thus, we can feed a single source into the Subject and have multiple consumers—that can come and go at runtime—on the other side: var subject = myApp.CreateSubject("Subject", () => new Subject<SourcePayload>()); Subject are always deployed automatically. Their name is used to retrieve them from a (potentially) different process (see below). Note that the Subject as we defined it here doesn’t know anything about temporal streams. It is merely a sequence of SourcePayloads, without any notion of StreamInsight point events or CTIs. So in order to compose a temporal query on top of the Subject, we need to 'promote' the sequence of SourcePayloads into an IQStreamable of point events, including CTIs: var stream = subject.ToPointStreamable(     e => PointEvent.CreateInsert<SourcePayload>(e.Timestamp, e),     AdvanceTimeSettings.StrictlyIncreasingStartTime); In a later posting we will show how to use Subjects that have more awareness of time and can be used as a junction between QStreamables instead of IQbservables. Having turned the Subject into a temporal stream, we can now define the aggregate on this stream. We will use the IQStreamable entity avg that we defined above: var longAverages = avg(stream, TimeSpan.FromSeconds(5)); In order to run the query, we need to bind it to a sink, and bind the subject to the source: var standardQuery = longAverages     .Bind(console("5sec average"))     .With(generator(TimeSpan.FromMilliseconds(300)).Bind(subject)); Lastly, we start the process: standardQuery.Run("StandardProcess"); Now we have a simple query running end-to-end, producing results. What follows next is the crucial part of tapping into the Subject and adding another query that runs in parallel, using the same query definition (the “AverageQuery”) but with a different window length. We are assuming that we connected to the same StreamInsight server from a different process or even client, and thus have to retrieve the previously deployed entities through their names: // simulate the addition of a 'fast' query from a separate server connection, // by retrieving the aggregation query fragment // (instead of simply using the 'avg' object) var averageQuery = myApp     .GetStreamable<IQStreamable<SourcePayload>, TimeSpan, double>("AverageQuery"); // retrieve the input sequence as a subject var inputSequence = myApp     .GetSubject<SourcePayload, SourcePayload>("Subject"); // retrieve the registered sink var sink = myApp.GetObserver<string, double>("ConsoleSink"); // turn the sequence into a temporal stream var stream2 = inputSequence.ToPointStreamable(     e => PointEvent.CreateInsert<SourcePayload>(e.Timestamp, e),     AdvanceTimeSettings.StrictlyIncreasingStartTime); // apply the query, now with a different window length var shortAverages = averageQuery(stream2, TimeSpan.FromSeconds(1)); // bind new sink to query and run it var fastQuery = shortAverages     .Bind(sink("1sec average"))     .Run("FastProcess"); The attached solution demonstrates the sample end-to-end. Regards, The StreamInsight Team

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  • Java Process "The pipe has been ended" problem

    - by Amit Kumar
    I am using Java Process API to write a class that receives binary input from the network (say via TCP port A), processes it and writes binary output to the network (say via TCP port B). I am using Windows XP. The code looks like this. There are two functions called run() and receive(): run is called once at the start, while receive is called whenever there is a new input received via the network. Run and receive are called from different threads. The run process starts an exe and receives the input and output stream of the exe. Run also starts a new thread to write output from the exe on to the port B. public void run() { try { Process prc = // some exe is `start`ed using ProcessBuilder OutputStream procStdIn = new BufferedOutputStream(prc.getOutputStream()); InputStream procStdOut = new BufferedInputStream(prc.getInputStream()); Thread t = new Thread(new ProcStdOutputToPort(procStdOut)); t.start(); prc.waitFor(); t.join(); procStdIn.close(); procStdOut.close(); } catch (Exception e) { e.printStackTrace(); printError("Error : " + e.getMessage()); } } The receive forwards the received input from the port A to the exe. public void receive(byte[] b) throws Exception { procStdIn.write(b); } class ProcStdOutputToPort implements Runnable { private BufferedInputStream bis; public ProcStdOutputToPort(BufferedInputStream bis) { this.bis = bis; } public void run() { try { int bytesRead; int bufLen = 1024; byte[] buffer = new byte[bufLen]; while ((bytesRead = bis.read(buffer)) != -1) { // write output to the network } } catch (IOException ex) { Logger.getLogger().log(Level.SEVERE, null, ex); } } } The problem is that I am getting the following stack inside receive() and the prc.waitfor() returns immediately afterwards. The line number shows that the stack is while writing to the exe. The pipe has been ended java.io.IOException: The pipe has been ended at java.io.FileOutputStream.writeBytes(Native Method) at java.io.FileOutputStream.write(FileOutputStream.java:260) at java.io.BufferedOutputStream.write(BufferedOutputStream.java:105) at java.io.BufferedOutputStream.flushBuffer(BufferedOutputStream.java:65) at java.io.BufferedOutputStream.write(BufferedOutputStream.java:109) at java.io.FilterOutputStream.write(FilterOutputStream.java:80) at xxx.receive(xxx.java:86) Any advice about this will be appreciated.

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  • Flash -> ByteArray -> AMFPHP -> Invalid Image !??

    - by undefined
    Hi, Im loading images into Flash and using JPGEncoder to encode the image to a ByteArray and send this to AMF PHP which writes out the bytearray to a file. This all appears to work correctly and I can download the resulting file in Photoshop CS4 absolutely fine. When i try to open it from the desktop or open it back in Flash it doesnt work... Picasa my default image browser says "Invalid" Here is the code i use to write the bytearray to a file - $jpg = $GLOBALS["HTTP_RAW_POST_DATA"]; file_put_contents($filename, $jpg); That's it ... I use the NetConnection class to connect and call the service, do I need to say Im sending jpg data? I assumed that JPGEncoder took care of that. How can I validate the bytearray before writing the file? Do I need to set MIME type or something .. excuse the slightly noob questions, a little knowledge can be a dangerous thing. Thanks --------------------------------------- PART II ------------------------------------------ Here is some code - 1) load the image into Flash player item.load(); function _onImageDataLoaded(evt:Event):void { var tmpFileRef:FileReference=FileReference(evt.target); image_loader=new Loader ; image_loader.contentLoaderInfo.addEventListener(Event.COMPLETE, _onImageLoaded); image_loader.loadBytes(tmpFileRef.data); } function _onImageLoaded(evt:Event):void { bitmap=Bitmap(evt.target.content); bitmap.smoothing=true; if (bitmap.width>MAX_WIDTH||bitmap.height>MAX_HEIGHT) { resizeBitmap(bitmap); } uploadResizedImage(bitmap); } function resizeBitmap(target:Bitmap):void { if (target.height>target.width) { target.width=MAX_WIDTH; target.scaleY=target.scaleX; } else if (target.width >= target.height) { target.height=MAX_HEIGHT; target.scaleX=target.scaleY; } } function uploadResizedImage(target:Bitmap):void { var _bmd:BitmapData=new BitmapData(target.width,target.height); _bmd.draw(target, new Matrix(target.scaleX, 0, 0, target.scaleY)); var encoded_jpg:JPGEncoder=new JPGEncoder(90); var jpg_binary:ByteArray=encoded_jpg.encode(_bmd); _uploadService=new NetConnection(); _uploadService.objectEncoding=ObjectEncoding.AMF3 _uploadService.connect("http://.../amfphp/gateway.php"); _uploadService.call("UploadService.receiveByteArray",new Responder(success, error), jpg_binary, currentImageFilename); } Many thanks for you help

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  • Synchronization requirements for FileStream.(Begin/End)(Read/Write)

    - by Doug McClean
    Is the following pattern of multi-threaded calls acceptable to a .Net FileStream? Several threads calling a method like this: ulong offset = whatever; // different for each thread byte[] buffer = new byte[8192]; object state = someState; // unique for each call, hence also for each thread lock(theFile) { theFile.Seek(whatever, SeekOrigin.Begin); IAsyncResult result = theFile.BeginRead(buffer, 0, 8192, AcceptResults, state); } if(result.CompletedSynchronously) { // is it required for us to call AcceptResults ourselves in this case? // or did BeginRead already call it for us, on this thread or another? } Where AcceptResults is: void AcceptResults(IAsyncResult result) { lock(theFile) { int bytesRead = theFile.EndRead(result); // if we guarantee that the offset of the original call was at least 8192 bytes from // the end of the file, and thus all 8192 bytes exist, can the FileStream read still // actually read fewer bytes than that? // either: if(bytesRead != 8192) { Panic("Page read borked"); } // or: // issue a new call to begin read, moving the offsets into the FileStream and // the buffer, and decreasing the requested size of the read to whatever remains of the buffer } } I'm confused because the documentation seems unclear to me. For example, the FileStream class says: Any public static members of this type are thread safe. Any instance members are not guaranteed to be thread safe. But the documentation for BeginRead seems to contemplate having multiple read requests in flight: Multiple simultaneous asynchronous requests render the request completion order uncertain. Are multiple reads permitted to be in flight or not? Writes? Is this the appropriate way to secure the location of the Position of the stream between the call to Seek and the call to BeginRead? Or does that lock need to be held all the way to EndRead, hence only one read or write in flight at a time? I understand that the callback will occur on a different thread, and my handling of state, buffer handle that in a way that would permit multiple in flight reads. Further, does anyone know where in the documentation to find the answers to these questions? Or an article written by someone in the know? I've been searching and can't find anything. Relevant documentation: FileStream class Seek method BeginRead method EndRead IAsyncResult interface

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  • Extract information from conditional formula

    - by Ken Williams
    I'd like to write an R function that accepts a formula as its first argument, similar to lm() or glm() and friends. In this case, it's a function that takes a data frame and writes out a file in SVMLight format, which has this general form: <line> .=. <target> <feature>:<value> <feature>:<value> ... <feature>:<value> # <info> <target> .=. +1 | -1 | 0 | <float> <feature> .=. <integer> | "qid" <value> .=. <float> <info> .=. <string> for example, the following data frame: result qid f1 f2 f3 f4 f5 f6 f7 f8 1 -1 1 0.0000 0.1253 0.0000 0.1017 0.00 0.0000 0.0000 0.9999 2 -1 1 0.0098 0.0000 0.0000 0.0000 0.00 0.0316 0.0000 0.3661 3 1 1 0.0000 0.0000 0.1941 0.0000 0.00 0.0000 0.0509 0.0000 4 -1 2 0.0000 0.2863 0.0948 0.0000 0.34 0.0000 0.7428 0.0608 5 1 2 0.0000 0.0000 0.0000 0.4347 0.00 0.0000 0.9539 0.0000 6 1 2 0.0000 0.7282 0.9087 0.0000 0.00 0.0000 0.0000 0.0355 would be represented as follows: -1 qid:1 2:0.1253 4:0.1017 8:0.9999 -1 qid:1 1:0.0098 6:0.0316 8:0.3661 1 qid:1 3:0.1941 7:0.0509 -1 qid:2 2:0.2863 3:0.0948 5:0.3400 7:0.7428 8:0.0608 1 qid:2 4:0.4347 7:0.9539 1 qid:2 2:0.7282 3:0.9087 8:0.0355 The function I'd like to write would be called something like this: write.svmlight(result ~ f1+f2+f3+f4+f5+f6+f7+f8 | qid, data=mydata, file="out.txt") Or even write.svmlight(result ~ . | qid, data=mydata, file="out.txt") But I can't figure out how to use model.matrix() and/or model.frame() to know what columns it's supposed to write. Are these the right things to be looking at? Any help much appreciated!

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  • Trying to read FormsAuthentication tickets to read in other areas of site

    - by Pasha Aryana
    Hi, NOTE: I have included 3 links in here to my localhost areas but could not submit the post so I seperetaed them with a space character so it would post on stackoverflow. I currently have 2 ASP.NET MVC apps in my solution. First I run the first one by setting it to be startup project. It goes to the login page, from there once the data has been entered I execute the following code: var authTicket = new FormsAuthenticationTicket(1, login.LoginDataContract.MSISDN, DateTime.Now, DateTime.Now.AddMinutes(Convert.ToDouble("30")), true, ""); string cookieContents = FormsAuthentication.Encrypt(authTicket); var cookie = new HttpCookie(FormsAuthentication.FormsCookieName, cookieContents) { Expires = authTicket.Expiration, //Path = FormsAuthentication.FormsCookiePath //Path = "http://localhost" Domain = "" }; if (System.Web.HttpContext.Current != null) { System.Web.HttpContext.Current.Response.Cookies.Add(cookie); } As you can see I have set the Domain = "", so theoretically speaking it should work on any thing under my http: //localhost. Then I have set the persist security of the cookie to true so I can access it from any where under localhost. The cookie writes fine and I get logged in and all godd for now. BTW the url for this login page is: http //localhost/MyAccount/Login Now then I stop the solution and set the other MVC apps to be the startup. Then I run it. The URL for the second site is: http: //localhost/WebActivations/ Here is the code in the other apps start controller: public class HomeController : Controller { public ActionResult Index() { ViewData["Message"] = "Welcome to ASP.NET MVC!"; // PASHA: Added code to read the authorization cookie set at // login in MyAccount *.sln for (int i = 0; i < System.Web.HttpContext.Current.Request.Cookies.Count;i++) { Response.Write(System.Web.HttpContext.Current.Request.Cookies[i].Name + " " + System.Web.HttpContext.Current.Request.Cookies[i].Value); } HttpCookie authorizationCookie = System.Web.HttpContext.Current.Request.Cookies[FormsAuthentication.FormsCookieName.ToString()]; // decrypt. FormsAuthenticationTicket authorizationForm = FormsAuthentication.Decrypt(authorizationCookie.Value); ViewData["Message"] = authorizationForm.UserData[0].ToString(); return View(); } public ActionResult About() { return View(); } The problem is in this Home controller when I run the solution it cannot read the authentication cookie, you see the loop there it does not find the .ASPXAUTH cookie. But once it crashes in Firefox I have a look in the Page Info and then security and Cookies and its there and its the same cookie. What am I doing wrong?

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  • Code Complete 2ed, composition and delegation.

    - by Arlukin
    Hi there. After a couple of weeks reading on this forum I thought it was time for me to do my first post. I'm currently rereading Code Complete. I think it's 15 years since the last time, and I find that I still can't write code ;-) Anyway on page 138 in Code Complete you find this coding horror example. (I have removed some of the code) class Emplyee { public: FullName GetName() const; Address GetAddress() const; PhoneNumber GetWorkPhone() const; ... bool IsZipCodeValid( Address address); ... private: ... } What Steve thinks is bad is that the functions are loosely related. Or has he writes "There's no logical connection between employees and routines that check ZIP codes, phone numbers or job classifications" Ok I totally agree with him. Maybe something like the below example is better. class ZipCode { public: bool IsValid() const; ... } class Address { public: ZipCode GetZipCode() const; ... } class Employee { public: Address GetAddress() const; ... } When checking if the zip is valid you would need to do something like this. employee.GetAddress().GetZipCode().IsValid(); And that is not good regarding to the Law of Demeter ([http://en.wikipedia.org/wiki/Law_of_Demeter][1]). So if you like to remove two of the three dots, you need to use delegation and a couple of wrapper functions like this. class ZipCode { public: bool IsValid(); } class Address { public: ZipCode GetZipCode() const; bool IsZipCodeValid() {return GetZipCode()->IsValid()); } class Employee { public: FullName GetName() const; Address GetAddress() const; bool IsZipCodeValid() {return GetAddress()->IsZipCodeValid()); PhoneNumber GetWorkPhone() const; } employee.IsZipCodeValid(); But then again you have routines that has no logical connection. I personally think that all three examples in this post are bad. Is it some other way that I haven't thougt about? //Daniel

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  • Showing login view controller before main tab bar controller

    - by Padawan
    I'm creating an iPad app with a tab bar controller that requires login. So on launch, I want to show a LoginViewController and if login is successful, then show the tab bar controller. This is how I implemented an initial test version (left out some typical header stuff, etc)... AppDelegate.h: @interface AppDelegate_Pad : NSObject <UIApplicationDelegate, LoginViewControllerDelegate> { UIWindow *window; UITabBarController *tabBarController; } @property (nonatomic, retain) IBOutlet UIWindow *window; @property (nonatomic, retain) IBOutlet UITabBarController *tabBarController; @end AppDelegate.m: @implementation AppDelegate_Pad @synthesize window; @synthesize tabBarController; - (BOOL)application:(UIApplication *)application didFinishLaunchingWithOptions:(NSDictionary *)launchOptions { LoginViewController_Pad *lvc = [[LoginViewController_Pad alloc] initWithNibName:@"LoginViewController_Pad" bundle:nil]; lvc.delegate = self; [window addSubview:lvc.view]; //[lvc release]; [window makeKeyAndVisible]; return YES; } - (void)loginViewControllerDidFinish:(LoginViewController_Pad *)loginViewController { [window addSubview:tabBarController.view]; } - (void)dealloc {...} @end LoginViewController_Pad.h: @protocol LoginViewControllerDelegate; @interface LoginViewController_Pad : UIViewController { id<LoginViewControllerDelegate> delegate; } @property (nonatomic, assign) id <LoginViewControllerDelegate> delegate; - (IBAction)buttonPressed; @end @protocol LoginViewControllerDelegate -(void)loginViewControllerDidFinish:(LoginViewController_Pad *)loginViewController; @end LoginViewController_Pad.m: @implementation LoginViewController_Pad @synthesize delegate; ... - (IBAction)buttonPressed { [self.view removeFromSuperview]; [self.delegate loginViewControllerDidFinish:self]; } ... @end So the app delegate adds the login view controller's view on launch and waits for login to call "did finish" using a delegate. The login view controller calls removeFromSuperView before it calls didFinish. The app delegate then calls addSubView on the tab bar controller's view. If you made it up to this point, thanks, and I have three questions: MAIN QUESTION: Is this the right way to show a view controller before the app's main tab bar controller is displayed? Even though it seems to work, is it a proper way to do it? If I comment out the "lvc release" in the app delegate then the app crashes with EXC_BAD_ACCESS when the button on the login view controller is pressed. Why? With the "lvc release" commented out everything seems to work but on the debugger console it writes this message when the app delegate calls addSubView for the tab bar controller: Using two-stage rotation animation. To use the smoother single-stage animation, this application must remove two-stage method implementations. What does that mean and do I need to worry about it?

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  • Determine the 'Overtype' mode using Javascript

    - by Snorkpete
    We are creating a web app to replace an old-school green-screen application. In the green-screen app, as the user presses the Insert key to switch between overtype and insert modes, the cursor changes to indicate which input mode the user is currently in. In IE (which is the official browser of the company), overtype mode also works, but there's no visual indication as to whether overtype mode is on or not, until the user starts typing and possibly over-writes existing information unexpectedly. I'd like to put some sort of visual indicator on the screen if in overtype mode. How can you determine if the browser is in 'overtype mode' from Javascript? Is there some property or function i can query to determine if the browser is in overtype mode? Even an IE-specific solution would be helpful, since our corporate policy dictates the browser to use as IE7 (pure torture, btw). (I do know that one solution is to do check for key presses of the Insert key. However, it's a solution that I'd prefer to avoid since that method seems a bit flaky & error-prone because I can't guarantee what mode the user would be in BEFORE he/she hits my page. ) The reasoning behind this question: The functionality of this portion of the green-screen app is such that the user can select from a list of 'preformatted bodies of text'. crude eg. The excess for this policy is: $xxxxxx and max limit is:$xxxxxx Date of policy is: xx/xx/xxxx and expires : xx/xx/xxxx Some other irrelevant text After selecting this 'preformatted text', the user would then use overtype to replace the x's with actual values, without disturbing the alignment of the rest of the text. (To be clear, they can still edit any part of the 'preformatted text' if they so wished. It's just that usually, they just wish to replace specific portions of the text. Keeping the alignment is important since these sections of text can end up on printed documents.) Of course, the same effect can be achieved by just selecting the x's to replace first, but it would be helpful (with respect to easing the transition to the web app) to allow old methods of doing things to continue to work, while still allowing 'web methods' to be used by the more tech-savvy users. Essentially, we're trying to make the initial transition from the green-screen app to the web app be as seemless as possible to minimise the resistance from the long-time green-screeners.

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  • How to maximize http.sys file upload performance

    - by anelson
    I'm building a tool that transfers very large streaming data sets (possibly on the order of terabytes in a single stream; routinely in the tens of gigabytes) from one server to another. The client portion of the tool will read blocks from the source disk, and send them over the network. The server side will read these blocks off the network and write them to a file on the server disk. Right now I'm trying to decide which transport to use. Options are raw TCP, and HTTP. I really, REALLY want to be able to use HTTP. The HttpListener (or WCF if I want to go that route) make it easy to plug in to the HTTP Server API (http.sys), and I can get things like authentication and SSL for free. The problem right now is performance. I wrote a simple test harness that sends 128K blocks of NULL bytes using the BeginWrite/EndWrite async I/O idiom, with async BeginRead/EndRead on the server side. I've modified this test harness so I can do this with either HTTP PUT operations via HttpWebRequest/HttpListener, or plain old socket writes using TcpClient/TcpListener. To rule out issues with network cards or network pathways, both the client and server are on one machine and communicate over localhost. On my 12-core Windows 2008 R2 test server, the TCP version of this test harness can push bytes at 450MB/s, with minimal CPU usage. On the same box, the HTTP version of the test harness runs between 130MB/s and 200MB/s depending upon how I tweak it. In both cases CPU usage is low, and the vast majority of what CPU usage there is is kernel time, so I'm pretty sure my usage of C# and the .NET runtime is not the bottleneck. The box has two 6-core Xeon X5650 processors, 24GB of single-ranked DDR3 RAM, and is used exclusively by me for my own performance testing. I already know about HTTP client tweaks like ServicePointManager.MaxServicePointIdleTime, ServicePointManager.DefaultConnectionLimit, ServicePointManager.Expect100Continue, and HttpWebRequest.AllowWriteStreamBuffering. Does anyone have any ideas for how I can get HTTP.sys performance beyond 200MB/s? Has anyone seen it perform this well on any environment?

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  • Creating thousands of records in Rails

    - by willCosgrove
    Let me set the stage: My application deals with gift cards. When we create cards they have to have a unique string that the user can use to redeem it with. So when someone orders our gift cards, like a retailer, we need to make a lot of new card objects and store them in the DB. With that in mind, I'm trying to see how quickly I can have my application generate 100,000 Cards. Database expert, I am not, so I need someone to explain this little phenomena: When I create 1000 Cards, it takes 5 seconds. When I create 100,000 cards it should take 500 seconds right? Now I know what you're wanting to see, the card creation method I'm using, because the first assumption would be that it's getting slower because it's checking the uniqueness of a bunch of cards, more as it goes along. But I can show you my rake task desc "Creates cards for a retailer" task :order_cards, [:number_of_cards, :value, :retailer_name] => :environment do |t, args| t = Time.now puts "Searching for retailer" @retailer = Retailer.find_by_name(args[:retailer_name]) puts "Retailer found" puts "Generating codes" value = args[:value].to_i number_of_cards = args[:number_of_cards].to_i codes = [] top_off_codes(codes, number_of_cards) while codes != codes.uniq codes.uniq! top_off_codes(codes, number_of_cards) end stored_codes = Card.all.collect do |c| c.code end while codes != (codes - stored_codes) codes -= stored_codes top_off_codes(codes, number_of_cards) end puts "Codes are unique and generated" puts "Creating bundle" @bundle = @retailer.bundles.create!(:value => value) puts "Bundle created" puts "Creating cards" @bundle.transaction do codes.each do |code| @bundle.cards.create!(:code => code) end end puts "Cards generated in #{Time.now - t}s" end def top_off_codes(codes, intended_number) (intended_number - codes.size).times do codes << ReadableRandom.get(CODE_LENGTH) end end I'm using a gem called readable_random for the unique code. So if you read through all of that code, you'll see that it does all of it's uniqueness testing before it ever starts creating cards. It also writes status updates to the screen while it's running, and it always sits for a while at creating. Meanwhile it flies through the uniqueness tests. So my question to the stackoverflow community is: Why is my database slowing down as I add more cards? Why is this not a linear function in regards to time per card? I'm sure the answer is simple and I'm just a moron who knows nothing about data storage. And if anyone has any suggestions, how would you optimize this method, and how fast do you think you could get it to create 100,000 cards? (When I plotted out my times on a graph and did a quick curve fit to get my line formula, I calculated how long it would take to create 100,000 cards with my current code and it says 5.5 hours. That maybe completely wrong, I'm not sure. But if it stays on the line I curve fitted, it would be right around there.)

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  • Why does one loop take longer to detect a shared memory update than another loop?

    - by Joseph Garvin
    I've written a 'server' program that writes to shared memory, and a client program that reads from the memory. The server has different 'channels' that it can be writing to, which are just different linked lists that it's appending items too. The client is interested in some of the linked lists, and wants to read every node that's added to those lists as it comes in, with the minimum latency possible. I have 2 approaches for the client: For each linked list, the client keeps a 'bookmark' pointer to keep its place within the linked list. It round robins the linked lists, iterating through all of them over and over (it loops forever), moving each bookmark one node forward each time if it can. Whether it can is determined by the value of a 'next' member of the node. If it's non-null, then jumping to the next node is safe (the server switches it from null to non-null atomically). This approach works OK, but if there are a lot of lists to iterate over, and only a few of them are receiving updates, the latency gets bad. The server gives each list a unique ID. Each time the server appends an item to a list, it also appends the ID number of the list to a master 'update list'. The client only keeps one bookmark, a bookmark into the update list. It endlessly checks if the bookmark's next pointer is non-null ( while(node->next_ == NULL) {} ), if so moves ahead, reads the ID given, and then processes the new node on the linked list that has that ID. This, in theory, should handle large numbers of lists much better, because the client doesn't have to iterate over all of them each time. When I benchmarked the latency of both approaches (using gettimeofday), to my surprise #2 was terrible. The first approach, for a small number of linked lists, would often be under 20us of latency. The second approach would have small spats of low latencies but often be between 4,000-7,000us! Through inserting gettimeofday's here and there, I've determined that all of the added latency in approach #2 is spent in the loop repeatedly checking if the next pointer is non-null. This is puzzling to me; it's as if the change in one process is taking longer to 'publish' to the second process with the second approach. I assume there's some sort of cache interaction going on I don't understand. What's going on?

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  • Odd ActiveRecord model dynamic initialization bug in production

    - by qfinder
    I've got an ActiveRecord (2.3.5) model that occasionally exhibits incorrect behavior that appears to be related to a problem in its dynamic initialization. Here's the code: class Widget < ActiveRecord::Base extend ActiveSupport::Memoizable serialize :settings VALID_SETTINGS = %w(show_on_sale show_upcoming show_current show_past) VALID_SETTINGS.each do |setting| class_eval %{ def #{setting}=(val); self.settings[:#{setting}] = (val == "1"); end def #{setting}; self.settings[:#{setting}]; end } end def initialize_settings self.settings ||= { :show_on_sale => true, :show_upcoming => true } end after_initialize :initialize_settings # All the other stuff the model does end The idea was to use a single record field (settings) to persist a bunch of configuration data for this object, but allow all the settings to seamlessly work with form helpers and the like. (Why this approach makes sense here is a little out of scope, but let's assume that it does.) Net-net, Widget should end up with instance methods (eg #show_on_sale= #show_on_sale) for all the entires in the VALID_SETTINGS array. Any default values should be specified in initialize_settings. And indeed this works, mostly. In dev and staging, no problems at all. But in production, the app sometimes ends up in a state where a) any writes to the dynamically generated setters fail and b) none of the default values appear to be set - although my leading theory is that the dynamically generated reader methods are just broken. The code, db, and environment is otherwise identical between the three. A typical error message / backtrace on the fail looks like: IndexError: index 141145 out of string (eval):2:in []=' (eval):2:inshow_on_sale=' [GEM_ROOT]/gems/activerecord-2.3.5/lib/active_record/base.rb:2746:in send' [GEM_ROOT]/gems/activerecord-2.3.5/lib/active_record/base.rb:2746:inattributes=' [GEM_ROOT]/gems/activerecord-2.3.5/lib/active_record/base.rb:2742:in each' [GEM_ROOT]/gems/activerecord-2.3.5/lib/active_record/base.rb:2742:inattributes=' [GEM_ROOT]/gems/activerecord-2.3.5/lib/active_record/base.rb:2634:in `update_attributes!' ...(then controller and all the way down) Ideas or theories as to what might be going on? My leading theory is that something is going wrong in instance initialization wherein the class instance variable settings is ending up as a string rather than a hash. This explains both the above setter failure (:show_on_sale is being used to index into the string) and the fact that getters don't work (an out of bounds [] call on a string just returns nil). But then how and why might settings occasionally end up as a string rather than hash?

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  • Write binary stream to browser using PHP

    - by Dave Jarvis
    Background Trying to stream a PDF report written using iReport through PHP to the browser. The general problem is: how do you write binary data to the browser using PHP? Working Code The following code does the job, but (for many reasons) it is not as efficient as it should be (the code writes a file then sends the file contents the browser). // Load the MySQL database driver. // java( 'java.lang.Class' )->forName( 'com.mysql.jdbc.Driver' ); // Attempt a database connection. // $conn = java( 'java.sql.DriverManager' )->getConnection( "jdbc:mysql://localhost:3306/climate?user=$user&password=$password" ); // Extract parameters. // $params = new java('java.util.HashMap'); $params->put('DistrictCode', '101'); $params->put('StationCode', '0066'); $params->put('CategoryCode', '010'); // Use the fill manager to produce the report. // $fm = java('net.sf.jasperreports.engine.JasperFillManager'); $pm = $fm->fillReport($report, $params, $conn); header('Cache-Control: no-cache private'); header('Content-Description: File Transfer'); header('Content-Disposition: attachment, filename=climate-report.pdf'); header('Content-Type: application/pdf'); header('Content-Transfer-Encoding: binary'); header('Content-Length: ' . strlen( $result ) ); $path = realpath( "." ) . "/output.pdf"; $em = java('net.sf.jasperreports.engine.JasperExportManager'); $result = $em->exportReportToPdfFile($pm,$path); readfile( $path ); $conn->close(); Non-working Code To remove the slight redundancy (i.e., write directly to the browser), the following code looks like it should work, but it does not: $em = java('net.sf.jasperreports.engine.JasperExportManager'); $result = $em->exportReportToPdf($pm); header('Content-Length: ' . strlen( $result ) ); echo $result; Content is sent to the browser, but the file is corrupt (it begins with the PDF header) and cannot be read by any PDF reader. Question How can I take out the middle step of writing to the file and write directly to the browser so that the PDF is not corrupted? Thank you!

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  • Blackberry Player, custom data source

    - by Alex
    Hello I must create a custom media player within the application with support for mp3 and wav files. I read in the documentation i cant seek or get the media file duration without a custom datasoruce. I checked the demo in the JDE 4.6 but i have still problems... I cant get the duration, it return much more then the expected so i`m sure i screwed up something while i modified the code to read the mp3 file locally from the filesystem. Somebody can help me what i did wrong ? (I can hear the mp3, so the player plays it correctly from start to end) I must support OSs = 4.6. Thank You Here is my modified datasource LimitedRateStreaminSource.java * Copyright © 1998-2009 Research In Motion Ltd. Note: For the sake of simplicity, this sample application may not leverage resource bundles and resource strings. However, it is STRONGLY recommended that application developers make use of the localization features available within the BlackBerry development platform to ensure a seamless application experience across a variety of languages and geographies. For more information on localizing your application, please refer to the BlackBerry Java Development Environment Development Guide associated with this release. */ package com.halcyon.tawkwidget.model; import java.io.IOException; import java.io.InputStream; import java.io.OutputStream; import javax.microedition.io.Connector; import javax.microedition.io.file.FileConnection; import javax.microedition.media.Control; import javax.microedition.media.protocol.ContentDescriptor; import javax.microedition.media.protocol.DataSource; import javax.microedition.media.protocol.SourceStream; import net.rim.device.api.io.SharedInputStream; /** * The data source used by the BufferedPlayback's media player. / public final class LimitedRateStreamingSource extends DataSource { /* The max size to be read from the stream at one time. */ private static final int READ_CHUNK = 512; // bytes /** A reference to the field which displays the load status. */ //private TextField _loadStatusField; /** A reference to the field which displays the player status. */ //private TextField _playStatusField; /** * The minimum number of bytes that must be buffered before the media file * will begin playing. */ private int _startBuffer = 200000; /** The maximum size (in bytes) of a single read. */ private int _readLimit = 32000; /** * The minimum forward byte buffer which must be maintained in order for * the video to keep playing. If the forward buffer falls below this * number, the playback will pause until the buffer increases. */ private int _pauseBytes = 64000; /** * The minimum forward byte buffer required to resume * playback after a pause. */ private int _resumeBytes = 128000; /** The stream connection over which media content is passed. */ //private ContentConnection _contentConnection; private FileConnection _fileConnection; /** An input stream shared between several readers. */ private SharedInputStream _readAhead; /** A stream to the buffered resource. */ private LimitedRateSourceStream _feedToPlayer; /** The MIME type of the remote media file. */ private String _forcedContentType; /** A counter for the total number of buffered bytes */ private volatile int _totalRead; /** A flag used to tell the connection thread to stop */ private volatile boolean _stop; /** * A flag used to indicate that the initial buffering is complete. In * other words, that the current buffer is larger than the defined start * buffer size. */ private volatile boolean _bufferingComplete; /** A flag used to indicate that the remote file download is complete. */ private volatile boolean _downloadComplete; /** The thread which retrieves the remote media file. */ private ConnectionThread _loaderThread; /** The local save file into which the remote file is written. */ private FileConnection _saveFile; /** A stream for the local save file. */ private OutputStream _saveStream; /** * Constructor. * @param locator The locator that describes the DataSource. */ public LimitedRateStreamingSource(String locator) { super(locator); } /** * Open a connection to the locator. * @throws IOException */ public void connect() throws IOException { //Open the connection to the remote file. _fileConnection = (FileConnection)Connector.open(getLocator(), Connector.READ); //Cache a reference to the locator. String locator = getLocator(); //Report status. System.out.println("Loading: " + locator); //System.out.println("Size: " + _contentConnection.getLength()); System.out.println("Size: " + _fileConnection.totalSize()); //The name of the remote file begins after the last forward slash. int filenameStart = locator.lastIndexOf('/'); //The file name ends at the first instance of a semicolon. int paramStart = locator.indexOf(';'); //If there is no semicolon, the file name ends at the end of the line. if (paramStart < 0) { paramStart = locator.length(); } //Extract the file name. String filename = locator.substring(filenameStart, paramStart); System.out.println("Filename: " + filename); //Open a local save file with the same name as the remote file. _saveFile = (FileConnection) Connector.open("file:///SDCard/blackberry/music" + filename, Connector.READ_WRITE); //If the file doesn't already exist, create it. if (!_saveFile.exists()) { _saveFile.create(); } System.out.println("---------- 1"); //Open the file for writing. _saveFile.setReadable(true); //Open a shared input stream to the local save file to //allow many simultaneous readers. SharedInputStream fileStream = SharedInputStream.getSharedInputStream(_saveFile.openInputStream()); //Begin reading at the beginning of the file. fileStream.setCurrentPosition(0); System.out.println("---------- 2"); //If the local file is smaller than the remote file... if (_saveFile.fileSize() < _fileConnection.totalSize()) { System.out.println("---------- 3"); //Did not get the entire file, set the system to try again. _saveFile.setWritable(true); System.out.println("---------- 4"); //A non-null save stream is used as a flag later to indicate that //the file download was incomplete. _saveStream = _saveFile.openOutputStream(); System.out.println("---------- 5"); //Use a new shared input stream for buffered reading. _readAhead = SharedInputStream.getSharedInputStream(_fileConnection.openInputStream()); System.out.println("---------- 6"); } else { //The download is complete. System.out.println("---------- 7"); _downloadComplete = true; //We can use the initial input stream to read the buffered media. _readAhead = fileStream; System.out.println("---------- 8"); //We can close the remote connection. _fileConnection.close(); System.out.println("---------- 9"); } if (_forcedContentType != null) { //Use the user-defined content type if it is set. System.out.println("---------- 10"); _feedToPlayer = new LimitedRateSourceStream(_readAhead, _forcedContentType); System.out.println("---------- 11"); } else { System.out.println("---------- 12"); //Otherwise, use the MIME types of the remote file. // _feedToPlayer = new LimitedRateSourceStream(_readAhead, _fileConnection)); } System.out.println("---------- 13"); } /** * Destroy and close all existing connections. */ public void disconnect() { try { if (_saveStream != null) { //Destroy the stream to the local save file. _saveStream.close(); _saveStream = null; } //Close the local save file. _saveFile.close(); if (_readAhead != null) { //Close the reader stream. _readAhead.close(); _readAhead = null; } //Close the remote file connection. _fileConnection.close(); //Close the stream to the player. _feedToPlayer.close(); } catch (Exception e) { System.err.println(e.getMessage()); } } /** * Returns the content type of the remote file. * @return The content type of the remote file. */ public String getContentType() { return _feedToPlayer.getContentDescriptor().getContentType(); } /** * Returns a stream to the buffered resource. * @return A stream to the buffered resource. */ public SourceStream[] getStreams() { return new SourceStream[] { _feedToPlayer }; } /** * Starts the connection thread used to download the remote file. */ public void start() throws IOException { //If the save stream is null, we have already completely downloaded //the file. if (_saveStream != null) { //Open the connection thread to finish downloading the file. _loaderThread = new ConnectionThread(); _loaderThread.start(); } } /** * Stop the connection thread. */ public void stop() throws IOException { //Set the boolean flag to stop the thread. _stop = true; } /** * @see javax.microedition.media.Controllable#getControl(String) */ public Control getControl(String controlType) { // No implemented Controls. return null; } /** * @see javax.microedition.media.Controllable#getControls() */ public Control[] getControls() { // No implemented Controls. return null; } /** * Force the lower level stream to a given content type. Must be called * before the connect function in order to work. * @param contentType The content type to use. */ public void setContentType(String contentType) { _forcedContentType = contentType; } /** * A stream to the buffered media resource. */ private final class LimitedRateSourceStream implements SourceStream { /** A stream to the local copy of the remote resource. */ private SharedInputStream _baseSharedStream; /** Describes the content type of the media file. */ private ContentDescriptor _contentDescriptor; /** * Constructor. Creates a LimitedRateSourceStream from * the given InputStream. * @param inputStream The input stream used to create a new reader. * @param contentType The content type of the remote file. */ LimitedRateSourceStream(InputStream inputStream, String contentType) { System.out.println("[LimitedRateSoruceStream]---------- 1"); _baseSharedStream = SharedInputStream.getSharedInputStream(inputStream); System.out.println("[LimitedRateSoruceStream]---------- 2"); _contentDescriptor = new ContentDescriptor(contentType); System.out.println("[LimitedRateSoruceStream]---------- 3"); } /** * Returns the content descriptor for this stream. * @return The content descriptor for this stream. */ public ContentDescriptor getContentDescriptor() { return _contentDescriptor; } /** * Returns the length provided by the connection. * @return long The length provided by the connection. */ public long getContentLength() { return _fileConnection.totalSize(); } /** * Returns the seek type of the stream. */ public int getSeekType() { return RANDOM_ACCESSIBLE; //return SEEKABLE_TO_START; } /** * Returns the maximum size (in bytes) of a single read. */ public int getTransferSize() { return _readLimit; } /** * Writes bytes from the buffer into a byte array for playback. * @param bytes The buffer into which the data is read. * @param off The start offset in array b at which the data is written. * @param len The maximum number of bytes to read. * @return the total number of bytes read into the buffer, or -1 if * there is no more data because the end of the stream has been reached. * @throws IOException */ public int read(byte[] bytes, int off, int len) throws IOException { System.out.println("[LimitedRateSoruceStream]---------- 5"); System.out.println("Read Request for: " + len + " bytes"); //Limit bytes read to our readLimit. int readLength = len; System.out.println("[LimitedRateSoruceStream]---------- 6"); if (readLength > getReadLimit()) { readLength = getReadLimit(); } //The number of available byes in the buffer. int available; //A boolean flag indicating that the thread should pause //until the buffer has increased sufficiently. boolean paused = false; System.out.println("[LimitedRateSoruceStream]---------- 7"); for (;;) { available = _baseSharedStream.available(); System.out.println("[LimitedRateSoruceStream]---------- 8"); if (_downloadComplete) { //Ignore all restrictions if downloading is complete. System.out.println("Complete, Reading: " + len + " - Available: " + available); return _baseSharedStream.read(bytes, off, len); } else if(_bufferingComplete) { if (paused && available > getResumeBytes()) { //If the video is paused due to buffering, but the //number of available byes is sufficiently high, //resume playback of the media. System.out.println("Resuming - Available: " + available); paused = false; return _baseSharedStream.read(bytes, off, readLength); } else if(!paused && (available > getPauseBytes() || available > readLength)) { //We have enough information for this media playback. if (available < getPauseBytes()) { //If the buffer is now insufficient, set the //pause flag. paused = true; } System.out.println("Reading: " + readLength + " - Available: " + available); return _baseSharedStream.read(bytes, off, readLength); } else if(!paused) { //Set pause until loaded enough to resume. paused = true; } } else { //We are not ready to start yet, try sleeping to allow the //buffer to increase. try { Thread.sleep(500); } catch (Exception e) { System.err.println(e.getMessage()); } } } } /** * @see javax.microedition.media.protocol.SourceStream#seek(long) */ public long seek(long where) throws IOException { _baseSharedStream.setCurrentPosition((int) where); return _baseSharedStream.getCurrentPosition(); } /** * @see javax.microedition.media.protocol.SourceStream#tell() */ public long tell() { return _baseSharedStream.getCurrentPosition(); } /** * Close the stream. * @throws IOException */ void close() throws IOException { _baseSharedStream.close(); } /** * @see javax.microedition.media.Controllable#getControl(String) */ public Control getControl(String controlType) { // No implemented controls. return null; } /** * @see javax.microedition.media.Controllable#getControls() */ public Control[] getControls() { // No implemented controls. return null; } } /** * A thread which downloads the remote file and writes it to the local file. */ private final class ConnectionThread extends Thread { /** * Download the remote media file, then write it to the local * file. * @see java.lang.Thread#run() */ public void run() { try { byte[] data = new byte[READ_CHUNK]; int len = 0; //Until we reach the end of the file. while (-1 != (len = _readAhead.read(data))) { _totalRead += len; if (!_bufferingComplete && _totalRead > getStartBuffer()) { //We have enough of a buffer to begin playback. _bufferingComplete = true; System.out.println("Initial Buffering Complete"); } if (_stop) { //Stop reading. return; } } System.out.println("Downloading Complete"); System.out.println("Total Read: " + _totalRead); //If the downloaded data is not the same size //as the remote file, something is wrong. if (_totalRead != _fileConnection.totalSize()) { System.err.println("* Unable to Download entire file *"); } _downloadComplete = true; _readAhead.setCurrentPosition(0); //Write downloaded data to the local file. while (-1 != (len = _readAhead.read(data))) { _saveStream.write(data); } } catch (Exception e) { System.err.println(e.toString()); } } } /** * Gets the minimum forward byte buffer which must be maintained in * order for the video to keep playing. * @return The pause byte buffer. */ int getPauseBytes() { return _pauseBytes; } /** * Sets the minimum forward buffer which must be maintained in order * for the video to keep playing. * @param pauseBytes The new pause byte buffer. */ void setPauseBytes(int pauseBytes) { _pauseBytes = pauseBytes; } /** * Gets the maximum size (in bytes) of a single read. * @return The maximum size (in bytes) of a single read. */ int getReadLimit() { return _readLimit; } /** * Sets the maximum size (in bytes) of a single read. * @param readLimit The new maximum size (in bytes) of a single read. */ void setReadLimit(int readLimit) { _readLimit = readLimit; } /** * Gets the minimum forward byte buffer required to resume * playback after a pause. * @return The resume byte buffer. */ int getResumeBytes() { return _resumeBytes; } /** * Sets the minimum forward byte buffer required to resume * playback after a pause. * @param resumeBytes The new resume byte buffer. */ void setResumeBytes(int resumeBytes) { _resumeBytes = resumeBytes; } /** * Gets the minimum number of bytes that must be buffered before the * media file will begin playing. * @return The start byte buffer. */ int getStartBuffer() { return _startBuffer; } /** * Sets the minimum number of bytes that must be buffered before the * media file will begin playing. * @param startBuffer The new start byte buffer. */ void setStartBuffer(int startBuffer) { _startBuffer = startBuffer; } } And in this way i use it: LimitedRateStreamingSource source = new LimitedRateStreamingSource("file:///SDCard/music3.mp3"); source.setContentType("audio/mpeg"); mediaPlayer = javax.microedition.media.Manager.createPlayer(source); mediaPlayer.addPlayerListener(this); mediaPlayer.realize(); mediaPlayer.prefetch(); After start i use mediaPlayer.getDuration it returns lets say around 24:22 (the inbuild media player in the blackberry say the file length is 4:05) I tried to get the duration in the listener and there unfortunatly returned around 64 minutes, so im sure something is not good inside the datasoruce....

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