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  • translating specifications into query predicates

    - by Jeroen
    I'm trying to find a nice and elegant way to query database content based on DDD "specifications". In domain driven design, a specification is used to check if some object, also known as the candidate, is compliant to a (domain specific) requirement. For example, the specification 'IsTaskDone' goes like: class IsTaskDone extends Specification<Task> { boolean isSatisfiedBy(Task candidate) { return candidate.isDone(); } } The above specification can be used for many purposes, e.g. it can be used to validate if a task has been completed, or to filter all completed tasks from a collection. However, I want to re-use this, nice, domain related specification to query on the database. Of course, the easiest solution would be to retrieve all entities of our desired type from the database, and filter that list in-memory by looping and removing non-matching entities. But clearly that would not be optimal for performance, especially when the entity count in our db increases. Proposal So my idea is to create a 'ConversionManager' that translates my specification into a persistence technique specific criteria, think of the JPA predicate class. The services looks as follows: public interface JpaSpecificationConversionManager { <T> Predicate getPredicateFor(Specification<T> specification, Root<T> root, CriteriaQuery<?> cq, CriteriaBuilder cb); JpaSpecificationConversionManager registerConverter(JpaSpecificationConverter<?, ?> converter); } By using our manager, the users can register their own conversion logic, isolating the domain related specification from persistence specific logic. To minimize the configuration of our manager, I want to use annotations on my converter classes, allowing the manager to automatically register those converters. JPA repository implementations could then use my manager, via dependency injection, to offer a find by specification method. Providing a find by specification should drastically reduce the number of methods on our repository interface. In theory, this all sounds decent, but I feel like I'm missing something critical. What do you guys think of my proposal, does it comply to the DDD way of thinking? Or is there already a framework that does something identical to what I just described?

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  • mounting ext4 fs with block size of 65536

    - by seaquest
    I am doing some benchmarking on EXT4 performance on Compact Flash media. I have created an ext4 fs with block size of 65536. however I can not mount it on ubuntu-10.10-netbook-i386. (it is already mounting ext4 fs with 4096 bytes of block sizes) According to my readings on ext4 it should allow such big block sized fs. I want to hear your comments. root@ubuntu:~# mkfs.ext4 -b 65536 /dev/sda3 Warning: blocksize 65536 not usable on most systems. mke2fs 1.41.12 (17-May-2010) mkfs.ext4: 65536-byte blocks too big for system (max 4096) Proceed anyway? (y,n) y Warning: 65536-byte blocks too big for system (max 4096), forced to continue Filesystem label= OS type: Linux Block size=65536 (log=6) Fragment size=65536 (log=6) Stride=0 blocks, Stripe width=0 blocks 19968 inodes, 19830 blocks 991 blocks (5.00%) reserved for the super user First data block=0 1 block group 65528 blocks per group, 65528 fragments per group 19968 inodes per group Writing inode tables: done Creating journal (1024 blocks): done Writing superblocks and filesystem accounting information: done This filesystem will be automatically checked every 37 mounts or 180 days, whichever comes first. Use tune2fs -c or -i to override. root@ubuntu:~# tune2fs -l /dev/sda3 tune2fs 1.41.12 (17-May-2010) Filesystem volume name: <none> Last mounted on: <not available> Filesystem UUID: 4cf3f507-e7b4-463c-be11-5b408097099b Filesystem magic number: 0xEF53 Filesystem revision #: 1 (dynamic) Filesystem features: has_journal ext_attr resize_inode dir_index filetype extent flex_bg sparse_super large_file huge_file uninit_bg dir_nlink extra_isize Filesystem flags: signed_directory_hash Default mount options: (none) Filesystem state: clean Errors behavior: Continue Filesystem OS type: Linux Inode count: 19968 Block count: 19830 Reserved block count: 991 Free blocks: 18720 Free inodes: 19957 First block: 0 Block size: 65536 Fragment size: 65536 Blocks per group: 65528 Fragments per group: 65528 Inodes per group: 19968 Inode blocks per group: 78 Flex block group size: 16 Filesystem created: Sat Feb 5 14:39:55 2011 Last mount time: n/a Last write time: Sat Feb 5 14:40:02 2011 Mount count: 0 Maximum mount count: 37 Last checked: Sat Feb 5 14:39:55 2011 Check interval: 15552000 (6 months) Next check after: Thu Aug 4 14:39:55 2011 Lifetime writes: 70 MB Reserved blocks uid: 0 (user root) Reserved blocks gid: 0 (group root) First inode: 11 Inode size: 256 Required extra isize: 28 Desired extra isize: 28 Journal inode: 8 Default directory hash: half_md4 Directory Hash Seed: afb5b570-9d47-4786-bad2-4aacb3b73516 Journal backup: inode blocks root@ubuntu:~# mount -t ext4 /dev/sda3 /mnt/ mount: wrong fs type, bad option, bad superblock on /dev/sda3, missing codepage or helper program, or other error In some cases useful info is found in syslog - try dmesg | tail or so

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  • Performing full screen grab in windows

    - by Steven Lu
    I am working an idea that involves getting a full capture of the screen including windows and apps, analyzing it, and then drawing items back onto the screen, as an overlay. I want to learn image processing techniques and I could get lots of data to work with if I can directly access the Windows screen. I could use this to build automation tools the likes of which have never been seen before. More on that later. I have full screen capture working for the most part. HWND hwind = GetDesktopWindow(); HDC hdc = GetDC(hwind); int resx = GetSystemMetrics(SM_CXSCREEN); int resy = GetSystemMetrics(SM_CYSCREEN); int BitsPerPixel = GetDeviceCaps(hdc,BITSPIXEL); HDC hdc2 = CreateCompatibleDC(hdc); BITMAPINFO info; info.bmiHeader.biSize = sizeof(BITMAPINFOHEADER); info.bmiHeader.biWidth = resx; info.bmiHeader.biHeight = resy; info.bmiHeader.biPlanes = 1; info.bmiHeader.biBitCount = BitsPerPixel; info.bmiHeader.biCompression = BI_RGB; void *data; hbitmap = CreateDIBSection(hdc2,&info,DIB_RGB_COLORS,(void**)&data,0,0); SelectObject(hdc2,hbitmap); Once this is done, I can call this repeatedly: BitBlt(hdc2,0,0,resx,resy,hdc,0,0,SRCCOPY); The cleanup code (I have no idea if this is correct): DeleteObject(hbitmap); ReleaseDC(hwind,hdc); if (hdc2) { DeleteDC(hdc2); } Every time BitBlt is called it grabs the screen and saves it in memory I can access thru data. Performance is somewhat satisfactory. BitBlt executes in 50 milliseconds (sometimes as low as 33ms) at 1920x1200x32. What surprises me is that when I switch display mode to 16 bit, 1920x1200x16, either through my graphics settings beforehand, or by using ChangeDisplaySettings, I get a massively improved screen grab time between 1ms and 2ms, which cannot be explained by the factor of two reduction in bit-depth. Using CreateDIBSection (as above) offers a significant speed up when in 16-bit mode, compared to if I set up with CreateCompatibleBitmap (6-7ms/f). Does anybody know why dropping to 16bit causes such a speed increase? Is there any hope for me to grab 32bit at such speeds? if not for the color depth, but for not forcing a change of screen buffer modes and the awful flickering.

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  • Large number of UPDATE queries slowing down page

    - by Bryan Lewis
    I am reading and validating large fixed-width text files (range from 10-50K lines) that are submitted via our ASP.net website (coded in VB.Net). I do an initial scan of the file to check for basic issues (line length, etc). Then I import each row into a MS SQL table. Each DB rows basically consists of a record_ID (Primary, auto-incrementing) and about 50 varchar fields. After the insert is done, I run a validation function on the file that checks each field in each row based on a bunch of criteria (trimmed length, isnumeric, range checks, etc). If it finds an error in any field, it inserts a record into the Errors table, which has an error_ID, the record_ID and an error message. In addition, if the field fails in a particular way, I have to do a "reset" on that field. A reset might consist of blanking the entire field, or simply replacing the value with another value (e.g. replacing the string with a new one that has all illegals chars taken out). I have a 5,000 line test file. The upload, initial check, and import takes about 5-6 seconds. The detailed error check and insert into the Errors table takes about 5-8 seconds (this file has about 1200 errors in it). However, the "resets" part takes about 40-45 seconds for 750 fields that need to be reset. When I comment out the resets function (returning immediately without actually calling the UPDATE stored proc), the process is very fast. With the resets turned on, the pages take 50 seconds to return. My UPDATE stored proc is using some recommended code from http://sommarskog.se/dynamic_sql.html, whereby it uses CASE instead of dynamic SQL: UPDATE dbo.Records SET dbo.Records.file_ID = CASE @field_name WHEN 'file_ID' THEN @field_value ELSE file_ID END, . . (all 50 varchar field CASE statements here) . WHERE dbo.Records.record_ID = @record_ID Is there any way I can help my performance here. Can I somehow group all of these UPDATE calls into a single transaction? Should I be reworking the UPDATE query somehow? Or is it just sheer quantity of 750+ UPDATEs and things are just slow (it's a quad proc server with 8GB ram). Any suggestions appreciated.

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  • Implementing a logging library in .NET with a database as the storage medium

    - by Dave
    I'm just starting to work on a logging library that everyone can use to keep track of any sort of system information while the user is running our application. The simplest example so far is to track Info, Warnings, and Errors. I want all plugins to be able to use this feature, but since each developer might have a different idea of what's important to report, I want to keep this as generic as possible. In the C++ world, I would normally use something like a stl::pair<string,string> to act as a key value pair structure, and have a stl::list of these to act as a "row" in the log. The log cache would then be a list<list<pair<string,string>>> (ugh!). This way, the developers can use a const string key like INFO, WARNING, ERROR to have a consistent naming for a column in the database (for SELECTing specific types of information). I'd like the database to be able to deal with any number of distinct column names. For example, John might have an INFO row with a column called USER, and Bill might have an INFO row with a column called FILENAME. I want the log viewer to be able to display all information, and if one report doesn't have a value for INFO / FILENAME, those fields should just appear blank. So one option is to use List<List<KeyValuePair<String,String>>, and the another is to have the log library consumer somehow "register" its schema, and then have the database do an ALTER TABLE to handle this situation. Yet another idea is to have a table that's just for key value pairs, with a foreign key that maps the key value pairs back to the original log entry. I obviously don't want logging to bog down the system, so I only lock the log cache to make a copy of the data (and remove the already-copied data), then a background thread will dump the information to the database. My specific questions regarding this are: Do you see any performance issues? In other words, have you ever tried something like this and found that certain things just don't work well in practice? Is there a more .NETish way to implement the key value pairs, other than List<List<KeyValuePair<String,String>>>? Even if there is a way to do #2 better, is the ALTER TABLE idea I proposed above a Bad Thing? Would you recommend multiple databases over a single one? I don't yet have an idea of how frequently the log would get written to, but we ideally would like to have lots of low level information. Perhaps there should be a DB with a fixed schema only for the low level stuff, and then another DB that's more flexible for reporting information back to users.

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  • Is "Systems Designer" the job title that best describes what I do? [closed]

    - by ivo-rossi
    After having worked as Java developer for almost 3 years in the same company that I currently work at, I moved to a new position associated with the development of the same application. I’m in this new position for more than 1 year now. My official job title is Systems Designer, but I’m not sure this is a title that expresses well what I do. So my question here is what would be the most appropriate job title for me? I see this question as important for my career development. After all, I should be able to explain in one word what I do. And it’s no longer “Java Developer”. Well, in more than one word, this is what I do: The business analysts gather requirements / business problems to be solved with the clients and then discuss these requirements with me. Given the requirements, I design the high level solutions to be implemented in our system (e.g. a new screen on the client application, modifications to existing reports, extension to the XML export format of some objects, etc). I base my decision on the current capabilities of the system, the overall impact that the solutions would have on the system and the estimated effort to implement them (as I was a developer of this same application for almost 3 years before I moved to this position, I’m confident in my estimates). The solutions are discussed iteratively with the business analysts until we agree that they are good. The outcome of this analysis is what we call the “requirements design” document, which is written by me, shared with clients for approval and then also with the team that is going to implement the solutions and test them. Note that there are a few problems that I need to find a solution for that are non-functional. If the users are unhappy with the performance of a certain tool, I will investigate what can be done to speed it up. I will do some research – often based in the Java code itself - to identify possibilities of optimizations. But in this new position I no longer code, the main outcome of my work is really the “requirements design”. Is “Systems Designer” really the most appropriate job title?

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  • Could not load file or assembly 'Base' or one of its dependencies. Access is denied

    - by starcorn
    I have deployed one web project into Azure emulator. And I get this error saying that Could not load file or assembly Base. The thing is that this web project, got dependencies to another project in the same solution. I have added that dependency into the reference list of my web project. And if I run this web application without using the azure emulator it will run fine. But I will get error when I try to run it on the azure emulator. At first glance I thought that I maybe need to add the other project as role also. But it couldn't be that. Anyone know what the problem might be? I hope I got enough data for you to look into. My solution structure looks like following Solution Base WebAPI WebAPI.Azure And it is the WebAPI that has a dependency to the Base project Here's the Assembly load trace WRN: Assembly binding logging is turned OFF. To enable assembly bind failure logging, set the registry value [HKLM\Software\Microsoft\Fusion!EnableLog] (DWORD) to 1. Note: There is some performance penalty associated with assembly bind failure logging. To turn this feature off, remove the registry value [HKLM\Software\Microsoft\Fusion!EnableLog]. And stack trace [FileLoadException: Could not load file or assembly 'Base' or one of its dependencies. Access is denied.] System.Reflection.RuntimeAssembly._nLoad(AssemblyName fileName, String codeBase, Evidence assemblySecurity, RuntimeAssembly locationHint, StackCrawlMark& stackMark, Boolean throwOnFileNotFound, Boolean forIntrospection, Boolean suppressSecurityChecks) +0 System.Reflection.RuntimeAssembly.InternalLoadAssemblyName(AssemblyName assemblyRef, Evidence assemblySecurity, StackCrawlMark& stackMark, Boolean forIntrospection, Boolean suppressSecurityChecks) +567 System.Reflection.RuntimeAssembly.InternalLoad(String assemblyString, Evidence assemblySecurity, StackCrawlMark& stackMark, Boolean forIntrospection) +192 System.Reflection.Assembly.Load(String assemblyString) +35 System.Web.Configuration.CompilationSection.LoadAssemblyHelper(String assemblyName, Boolean starDirective) +123 [ConfigurationErrorsException: Could not load file or assembly 'Base' or one of its dependencies. Access is denied.] System.Web.Configuration.CompilationSection.LoadAssemblyHelper(String assemblyName, Boolean starDirective) +11568160 System.Web.Configuration.CompilationSection.LoadAllAssembliesFromAppDomainBinDirectory() +485 System.Web.Configuration.AssemblyInfo.get_AssemblyInternal() +79 System.Web.Compilation.BuildManager.GetReferencedAssemblies(CompilationSection compConfig) +337 System.Web.Compilation.BuildManager.CallPreStartInitMethods() +280 System.Web.Hosting.HostingEnvironment.Initialize(ApplicationManager appManager, IApplicationHost appHost, IConfigMapPathFactory configMapPathFactory, HostingEnvironmentParameters hostingParameters, PolicyLevel policyLevel, Exception appDomainCreationException) +1167 [HttpException (0x80004005): Could not load file or assembly 'Base' or one of its dependencies. Access is denied.] System.Web.HttpRuntime.FirstRequestInit(HttpContext context) +11700896 System.Web.HttpRuntime.EnsureFirstRequestInit(HttpContext context) +141 System.Web.HttpRuntime.ProcessRequestNotificationPrivate(IIS7WorkerRequest wr, HttpContext context) +4869125

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  • Few iPhone noob questions

    - by mshsayem
    Why should I declare local variables as 'static' inside a method? Like: static NSString *cellIdentifier = @"Cell"; Is it a performance advantage? (I know what 'static' does; in C context) What does this syntax mean?[someObj release], someObj = nil; Two statements? Why should I assign nil again? Is not 'release' enough? Should I do it for all objects I allocate/own? Or for just view objects? Why does everyone copy NSString, but retains other objects (in property declaration)? Yes, NSStrings can be changed, but other objects can be changed also, right? Then why 'copy' for just NSString, not for all? Is it just a defensive convention? Shouldn't I release constant NSString? Like here:NSString *CellIdentifier = @"Cell"; Why not? Does the compiler allocate/deallocate it for me? In some tutorial application I observed these (Built with IB): Properties(IBOutlet, with same ivar name): window, someLabel, someTextField, etc etc... In the dealloc method, although the window ivar was released, others were not. My question is: WHY? Shouldn't I release other ivars(labels, textField) as well? Why not? Say, I have 3 cascaded drop-down lists. I mean, based on what is selected on the first list, 2nd list is populated and based on what is selected on the second list, 3rd list is populated. What UI components can reflect this best? How is drop-down list presented in iPhone UI? Tableview with UIPicker? When should I update the 2nd, 3rd list? Or just three labels which have touch events? Can you give me some good example tutorials about Core-Data? (Not just simple data fetching and storing on 2/3 tables with 1/2 relationship) How can I know whether my app is leaking memory? Any tools?

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  • Poor man's "lexer" for C#

    - by Paul Hollingsworth
    I'm trying to write a very simple parser in C#. I need a lexer -- something that lets me associate regular expressions with tokens, so it reads in regexs and gives me back symbols. It seems like I ought to be able to use Regex to do the actual heavy lifting, but I can't see an easy way to do it. For one thing, Regex only seems to work on strings, not streams (why is that!?!?). Basically, I want an implementation of the following interface: interface ILexer : IDisposable { /// <summary> /// Return true if there are more tokens to read /// </summary> bool HasMoreTokens { get; } /// <summary> /// The actual contents that matched the token /// </summary> string TokenContents { get; } /// <summary> /// The particular token in "tokenDefinitions" that was matched (e.g. "STRING", "NUMBER", "OPEN PARENS", "CLOSE PARENS" /// </summary> object Token { get; } /// <summary> /// Move to the next token /// </summary> void Next(); } interface ILexerFactory { /// <summary> /// Create a Lexer for converting a stream of characters into tokens /// </summary> /// <param name="reader">TextReader that supplies the underlying stream</param> /// <param name="tokenDefinitions">A dictionary from regular expressions to their "token identifers"</param> /// <returns>The lexer</returns> ILexer CreateLexer(TextReader reader, IDictionary<string, object> tokenDefinitions); } So, pluz send the codz... No, seriously, I am about to start writing an implementation of the above interface yet I find it hard to believe that there isn't some simple way of doing this in .NET (2.0) already. So, any suggestions for a simple way to do the above? (Also, I don't want any "code generators". Performance is not important for this thing and I don't want to introduce any complexity into the build process.)

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  • Can this way of storing typed objects be improved?

    - by Pindatjuh
    This is an "can it be improved"-question. Topic: Storing typed objects in memory. Background information: I'm building a compiler for the x86-32 Windows platform for my language. My goal includes typed objects. Idea: Every primitive is a semi-class (it can be used as if it was a normal class, but it's stored more compact). Every class is represented by primitives and some meta-data (containing class-properties, inheritance stuff, etc.). The meta-data is complex: it doesn't use fields but instead context-switches. For primitives, the meta-data is very small, compared to a "real" class, which is alot bigger. This enables another idea that "primitives are objects", in my language, which I found nessecairy. Example: If I have an array of 32 booleans, then the pure content of this array is exactly 4 byte (32 bits of booleans). The meta-data will contain flags that the type is an array of booleans, which contains 32 entries. The meta-data is very compacted, on bit-level: using a sort of "packing" mechanism, which is read by a FSM at runtime, when doing inspection of the type (like when passing the object to methods for checking, etc.) For instance (read from left to right, top to bottom, remember vertical position when going to the right, and check nearest column header for meaning of switch): Primitive? Array? Type-Meta 1 Byte? || Size (1 byte) 1 1 [...] 1 [...] done 0 2 Bytes? || Size (2 bytes) 1 [...] done || Size (4 bytes) 0 [...] done Integer? 1 Byte? 2 Bytes? 0 1 0 1 done 1 done 0 done Boolean? Byte? 0 1 0 done 1 done More-Primitives 0 .... Class-Stuff (Huge) 0 ... (After reaching done the data is inserted. || = byte alignment. [...] is variable sized. ... is not described here, for simplicity. And let's call them cost-based-data-structures.) For an array of 32 booleans containing all true values, the memory for this type would be (read top-down): 1 Primitive 1 Array 1 ArrayType: Primitive 0 Not-Array 0 Not-Integer 1 Boolean 0 Not-Byte (thus bit) 1 Integer Size: 1 Byte 00100000 Array size 01010101 01010101 01010101 01010101 Data (user defined) Thus, 8 bytes represent 32 booleans in an array: 11100101 00100000 01010101 01010101 01010101 01010101 How can I improve this? (Both performance- and memory-consumption wise)

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  • How to stream semi-live audio over internet

    - by Thomas Tempelmann
    I want to write something like Skype, i.e. I have a constant audio stream on one computer and then recompress it in a format that's suitable for a latent internet connection, receive it on the other end and play it. Let's also assume that the internet connection is fairly modern and fast, i.e. DSL or alike, no slow connections over phone and such. The involved computers will also be rather modern (Dual Core Intel CPUs at 2GHz or more). I know how to handle the audio on the machines. What I don't know is how to transmit the audio in an efficient way. The challenges are: I'd like get good audio quality across the line. The stream should be received without drops. The stream may, however, be received with a little delay (a second delay is acceptable). I imagine that the transport software could first determine the average (and max) latency, then start the stream and tell the receiver to wait for that max latency before starting to play the audio. With that, if the latency doesn't get any higher, the entire stream will be playable on the other side without stutter or drops. If, due to unexpected IP latencies or blockages, the stream does get cut off, I want to be able to notice this so that I can take actions (e.g. abort the stream) and eventually start a new transmission. What are my options if I want do use ready-made software for the compression and tranmission? I have no intention to write my own audio compression engine, really. OTOH, I plan to sell the solution in a vertical market, meaning I can afford a few dollars of license fees per copy, but not $100s. I guess the simplest solution would be to just open a TCP stream, send a few packets back and forth to determine their running time (or even use UDP for that), then use the results as the guide for my max latency value, then simply fire the audio data in its raw form (uncompressed 16 bit stereo), along with a timing code over the TCP connection. The receiver reads the data and plays it with the pre-determined delay. That might just work with the type of fast connection I expect. I just wonder if there are better solutions to reach this goal, with better performance (lower latency) and less data (compressed). BTW, I first try to implement this on OS X, but might want to do it on Windows, too, if it proves successful.

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  • When to call glEnable(GL_FRAMEBUFFER_SRGB)?

    - by Steven Lu
    I have a rendering system where I draw to an FBO with a multisampled renderbuffer, then blit it to another FBO with a texture in order to resolve the samples in order to read off the texture to perform post-processing shading while drawing to the backbuffer (FBO index 0). Now I'd like to get some correct sRGB output... The problem is the behavior of the program is rather inconsistent between when I run it on OS X and Windows and this also changes depending on the machine: On Windows with the Intel HD 3000 it will not apply the sRGB nonlinearity but on my other machine with a Nvidia GTX 670 it does. On the Intel HD 3000 in OS X it will also apply it. So this probably means that I'm not setting my GL_FRAMEBUFFER_SRGB enable state at the right points in the program. However I can't seem to find any tutorials that actually tell me when I ought to enable it, they only ever mention that it's dead easy and comes at no performance cost. I am currently not loading in any textures so I haven't had a need to deal with linearizing their colors yet. To force the program to not simply spit back out the linear color values, what I have tried is simply comment out my glDisable(GL_FRAMEBUFFER_SRGB) line, which effectively means this setting is enabled for the entire pipeline, and I actually redundantly force it back on every frame. I don't know if this is correct or not. It certainly does apply a nonlinearization to the colors but I can't tell if this is getting applied twice (which would be bad). It could apply the gamma as I render to my first FBO. It could do it when I blit the first FBO to the second FBO. Why not? I've gone so far as to take screen shots of my final frame and compare raw pixel color values to the colors I set them to in the program: I set the input color to RGB(1,2,3) and the output is RGB(13,22,28). That seems like quite a lot of color compression at the low end and leads me to question if the gamma is getting applied multiple times. I have just now gone through the sRGB equation and I can verify that the conversion seems to be only applied once as linear 1/255, 2/255, and 3/255 do indeed map to sRGB 13/255, 22/255, and 28/255 using the equation 1.055*C^(1/2.4)+0.055. Given that the expansion is so large for these low color values it really should be obvious if the sRGB color transform is getting applied more than once. So, I still haven't determined what the right thing to do is. does glEnable(GL_FRAMEBUFFER_SRGB) only apply to the final framebuffer values, in which case I can just set this during my GL init routine and forget about it hereafter?

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  • markdown to HTML with customised WMD editor

    - by spirytus
    For my application I customized slightly the way WMD behaves so when user enters empty lines, these are reflected in HTML output as <br />'s. Now I came to a point when I should store it somewhere at backend and so after going thru SO posts for a while I'm not sure what is the best way to do it. I have few options and if you could point out which their pros/cons that would be much appreciated. send to server and store as markdown rather than HTML. To me obvious advantage would be keeping exactly same formatting as user originally entered. But then how can I convert it back to HTML for display to a client? It seems very troublesome to convert it on client side as even if it would be possible what would happen if JS would be disabled? If I wanted to do it on the server, then standard server side implementations of markup to HTML might be resource expensive. Would that be an issue in your opinion? Even if it wouldn't be the case then as I mentioned my WMD implementation is customised and those server side solutions wouldn't probably do the right conversion to markdown anyway and there always would be a risk that something would convert wrong. Send to server as converted HTML. Same as above.. conversion on client side would be difficult, server side same with possibility of getting it wrong. send original markdown and converted HTML and store both. No performance issues related to converting markdown to HTML on client side, nor on server side. Users would have always same markdown they originally entered and same HTML they originally saw in preview (possibly sanitized in php though). It would have to take twice that much storage space though and that is my biggest worry. I tend to lean towards 3rd solution as it seems simplest, but there is a worry of doubled storage space needed for this solution. Please bear in mind that my implementation of WMD is slightly modified and also I'm going with PHP/MySql server side implementation. So apart from 3 options I listed above, are there any other possible solutions to my problem? Did I miss anything important that would make one of the options above better then the rest? And what other pros/cons would apply to each solution I listed? Also how is it implemented on SO? I read somwhere that they using option 3, and so if its good enough for SO would be good enough for me :) but not sure if its true anyway, so how is it done? Also please forgive me, but at least for once I got to say that StackOverflow IS THE BEST DAMN RESOURCE ON THE WEB and I truly appreciate all the people trying to help others here! The site and users here are simply amazing!

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  • Calling a function that resides in the main page from a plugin?

    - by Justin Lee
    I want to call a function from within plugin, but the function is on the main page and not the plugin's .js file. EDIT I have jQuery parsing a very large XML file and building, subsequently, a large list (1.1 MB HTML file when dynamic content is copied, pasted, then saved) that has expand/collapse functionality through a plugin. The overall performance on IE is super slow and doggy, assuming since the page/DOM is so big. I am currently trying to save the collapsed content in the event.data when it is collapsed and remove it from the DOM, then bring it back when it is told to expand... the issue that I am having is that when I bring the content back, obviously the "click" and "hover" events are gone. I'm trying to re-assign them, currently doing so inside the plugin after the plugin expands the content. The issue then though is that is says the function that I declare within the .click() is not defined. Also the hover event doesn't seem to be re-assigning either.... if ($(event.data.trigger).attr('class').indexOf('collapsed') != -1 ) { // if expanding // console.log(event.data.targetContent); $(event.data.trigger).after(event.data.targetContent); $(event.data.target).hide(); /* This Line --->*/ $(event.data.target + 'a.addButton').click(addResourceToList); $(event.data.target + 'li.resource') .hover( function() { if (!($(this).attr("disabled"))) { $(this).addClass("over"); $(this).find("a").css({'display':'block'}); } }, function () { if (!($(this).attr("disabled"))) { $(this).removeClass("over"); $(this).children("a").css({'display':'none'}); } } ); $(event.data.target).css({ "height": "0px", "padding-top": "0px", "padding-bottom": "0px", "margin-top": "0px", "margin-bottom": "0px"}); $(event.data.target).show(); $(event.data.target).animate({ height: event.data.heightVal + "px", paddingTop: event.data.topPaddingVal + "px", paddingBottom: event.data.bottomPaddingVal + "px", marginTop: event.data.topMarginVal + "px", marginBottom: event.data.bottomMarginVal + "px"}, "normal");//, function(){$(this).hide();}); $(event.data.trigger).removeClass("collapsed"); $.cookies.set('jcollapserSub_' + event.data.target, 'expanded', {hoursToLive: 24 * 365}); } else if ($(event.data.trigger).attr('class').indexOf('collapsed') == -1 ) { // if collapsing $(event.data.target).animate({ height: "0px", paddingTop: "0px", paddingBottom: "0px", marginTop: "0px", marginBottom: "0px"}, "normal", function(){$(this).hide();$(this).remove();}); $(event.data.trigger).addClass("collapsed"); $.cookies.set('jcollapserSub_' + event.data.target, 'collapsed', {hoursToLive: 24 * 365}); } EDIT So, having new eyes truly makes a difference. As I was reviewing the code in this post this morning after being away over the weekend, I found where I had err'd. This: $(event.data.target + 'a.addButton').click(addResourceToList); Should be this (notice the space before a.addbutton): $(event.data.target + ' a.addButton').click(addResourceToList); Same issue with the "li.resource". So it was never pointing to the right elements... Thank you, Rene, for your help!!

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  • How to implement a tagging plugin for jQuery

    - by anxiety
    Goal: To implement a jQuery plugin for my rails app (or write one myself, if necessary) that creates a "box" around text after a delimiter is typed. Example: With tagging on SO, the user begins typing a tag, then selects one from the drop-down list provided. The input field recognizes that a tag has been selected, puts a space and then is ready for the next tag. Similarly, I am attempting to use this plugin to put a box around the previously entered tag before moving to to accept the next tag/input. The instructions in the README.txt seem simple enough, however I have been receiving a $(".tagbox").tagbox is not a function error when debugging my app with firebug. Here is what I have in my application.js: $(document).ready( function(){ $('.tagbox').tagbox({ separator: /\[,]/, // specifying comma separation for <code>tags</code> }); }); And here is my _form.html.erb: <% form_for @tag do |f| %> <%= f.error_messages %> <p> <%= f.label :name %><br /> <%= text_field :tag, :name, { :method => :get, :class => "tagbox" } %> </p> <p><%= f.submit "Submit" %></p> <% end %> I have omitted some other code (namely the implementation of an autocomplete plugin) existing within my _form.html.erb and application.js for sake of readability. The inclusion or exclusion of this omitted code does not affect the performance of this plugin. I have included all of the necessary files for the tagbox plugin (as well as application.js after all other included JS files) within the javascript_include_tag in my application.html.erb file. I'm pretty much confused as to why I'd be getting this "not a function" error when jquery.tagbox.js clearly defines the function and is included in the head of my html page in question. I've been struggling with this plugin for longer than I'd like to admit, so any help would really be appreciated. And, of course, I'm open to any other plugins or from-scratch suggestions you may have in mind.. This tagbox plugin does not seem to have a wealth of documentation or any currently working examples. Also to note, I'm trying to avoid using jrails. Thanks for your time

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  • Finding the most frequent subtrees in a collection of (parse) trees

    - by peter.murray.rust
    I have a collection of trees whose nodes are labelled (but not uniquely). Specifically the trees are from a collection of parsed sentences (see http://en.wikipedia.org/wiki/Treebank). I wish to extract the most common subtrees from the collection - performance is not (yet) an issue. I'd be grateful for algorithms (ideally Java) or pointers to tools which do this for treebanks. Note that order of child nodes is important. EDIT @mjv. We are working in a limited domain (chemistry) which has a stylised language so the varirty of the trees is not huge - probably similar to children's readers. Simple tree for "the cat sat on the mat". <sentence> <nounPhrase> <article/> <noun/> </nounPhrase> <verbPhrase> <verb/> <prepositionPhrase> <preposition/> <nounPhrase> <article/> <noun/> </nounPhrase> </prepositionPhrase> </verbPhrase> </sentence> Here the sentence contains two identical part-of-speech subtrees (the actual tokens "cat". "mat" are not important in matching). So the algorithm would need to detect this. Note that not all nounPhrases are identical - "the big black cat" could be: <nounPhrase> <article/> <adjective/> <adjective/> <noun/> </nounPhrase> The length of sentences will be longer - between 15 to 30 nodes. I would expect to get useful results from 1000 trees. If this does not take more than a day or so that's acceptable. Obviously the shorter the tree the more frequent, so nounPhrase will be very common. EDIT If this is to be solved by flattening the tree then I think it would be related to Longest Common Substring, not Longest Common Sequence. But note that I don't necessarily just want the longest - I want a list of all those long enough to be "interesting" (criterion yet to be decided).

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  • Stepping into Ruby Meta-Programming: Generating proxy methods for multiple internal methods

    - by mstksg
    Hi all; I've multiply heard Ruby touted for its super spectacular meta-programming capabilities, and I was wondering if anyone could help me get started with this problem. I have a class that works as an "archive" of sorts, with internal methods that process and output data based on an input. However, the items in the archive in the class itself are represented and processed with integers, for performance purposes. The actual items outside of the archive are known by their string representation, which is simply number_representation.to_s(36). Because of this, I have hooked up each internal method with a "proxy method" that converts the input into the integer form that the archive recognizes, runs the internal method, and converts the output (either a single other item, or a collection of them) back into strings. The naming convention is this: internal methods are represented by _method_name; their corresponding proxy method is represented by method_name, with no leading underscore. For example: class Archive ## PROXY METHODS ## ## input: string representation of id's ## output: string representation of id's def do_something_with id result = _do_something_with id.to_i(36) return nil if result == nil return result.to_s(36) end def do_something_with_pair id_1,id_2 result = _do_something_with_pair id_1.to_i(36), id_2.to_i(36) return nil if result == nil return result.to_s(36) end def do_something_with_these ids result = _do_something_with_these ids.map { |n| n.to_i(36) } return nil if result == nil return result.to_s(36) end def get_many_from id result = _get_many_from id return nil if result == nil # no sparse arrays returned return result.map { |n| n.to_s(36) } end ## INTERNAL METHODS ## ## input: integer representation of id's ## output: integer representation of id's def _do_something_with id # does something with one integer-represented id, # returning an id represented as an integer end def do_something_with_pair id_1,id_2 # does something with two integer-represented id's, # returning an id represented as an integer end def _do_something_with_these ids # does something with multiple integer ids, # returning an id represented as an integer end def _get_many_from id # does something with one integer-represented id, # returns a collection of id's represented as integers end end There are a couple of reasons why I can't just convert them if id.class == String at the beginning of the internal methods: These internal methods are somewhat computationally-intensive recursive functions, and I don't want the overhead of checking multiple times at every step There is no way, without adding an extra parameter, to tell whether or not to re-convert at the end I want to think of this as an exercise in understanding ruby meta-programming Does anyone have any ideas? edit The solution I'd like would preferably be able to take an array of method names @@PROXY_METHODS = [:do_something_with, :do_something_with_pair, :do_something_with_these, :get_many_from] iterate through them, and in each iteration, put out the proxy method. I'm not sure what would be done with the arguments, but is there a way to test for arguments of a method? If not, then simple duck typing/analogous concept would do as well.

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  • multithreading with database

    - by Darsin
    I am looking out for a strategy to utilize multithreading (probably asynchronous delegates) to do a synchronous operation. I am new to multithreading so i will outline my scenario first. This synchronous operation right now is done for one set of data (portfolio) based on the the parameters provided. The (psudeo-code) implementation is given below: public DataSet DoTests(int fundId, DateTime portfolioDate) { // Get test results for the portfolio // Call the database adapter method, which in turn is a stored procedure, // which in turns runs a series of "rule" stored procs and fills a local temp table and returns it back. DataSet resultsDataSet = GetTestResults(fundId, portfolioDate); try { // Do some local processing on the results DoSomeProcessing(resultsDataSet); // Save the results in Test, TestResults and TestAllocations tables in a transaction. // Sets a global transaction which is provided to all the adapter methods called below // It is defined in the Base class StartTransaction("TestTransaction"); // Save Test and get a testId int testId = UpdateTest(resultsDataSet); // Adapter method, uses the same transaction // Update testId in the other tables in the dataset UpdateTestId(resultsDataSet, testId); // Update TestResults UpdateTestResults(resultsDataSet); // Adapter method, uses the same transaction // Update TestAllocations UpdateTestAllocations(resultsDataSet); // Adapter method, uses the same transaction // It is defined in the base class CommitTransaction("TestTransaction"); } catch { RollbackTransaction("TestTransaction"); } return resultsDataSet; } Now the requirement is to do it for multiple set of data. One way would be to call the above DoTests() method in a loop and get the data. I would prefer doing it in parallel. But there are certain catches: StartTransaction() method creates a connection (and transaction) every time it is called. All the underlying database tables, procedures are the same for each call of DoTests(). (obviously). Thus my question are: Will using multithreading anyway improve performance? What are the chances of deadlock especially when new TestId's are being created and the Tests, TestResults and TestAllocations are being saved? How can these deadlocked be handled? Is there any other more efficient way of doing the above operation apart from looping over the DoTests() method repeatedly?

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  • What (tf) are the secrets behind PDF memory allocation (CGPDFDocumentRef)

    - by Kai
    For a PDF reader I want to prepare a document by taking 'screenshots' of each page and save them to disc. First approach is CGPDFDocumentRef document = CGPDFDocumentCreateWithURL((CFURLRef) someURL); for (int i = 1; i<=pageCount; i++) { NSAutoreleasePool *pool = [[NSAutoreleasePool alloc]init]; CGPDFPageRef page = CGPDFDocumentGetPage(document, i); ...//getting + manipulating graphics context etc. ... CGContextDrawPDFPage(context, page); ... UIImage *resultingImage = UIGraphicsGetImageFromCurrentImageContext(); ...//saving the image to disc [pool drain]; } CGPDFDocumentRelease(document); This results in a lot of memory which seems not to be released after the first run of the loop (preparing the 1st document), but no more unreleased memory in additional runs: MEMORY BEFORE: 6 MB MEMORY DURING 1ST DOC: 40 MB MEMORY AFTER 1ST DOC: 25 MB MEMORY DURING 2ND DOC: 40 MB MEMORY AFTER 2ND DOC: 25 MB .... Changing the code to for (int i = 1; i<=pageCount; i++) { CGPDFDocumentRef document = CGPDFDocumentCreateWithURL((CFURLRef) someURL); NSAutoreleasePool *pool = [[NSAutoreleasePool alloc]init]; CGPDFPageRef page = CGPDFDocumentGetPage(document, i); ...//getting + manipulating graphics context etc. ... CGContextDrawPDFPage(context, page); ... UIImage *resultingImage = UIGraphicsGetImageFromCurrentImageContext(); ...//saving the image to disc CGPDFDocumentRelease(document); [pool drain]; } changes the memory usage to MEMORY BEFORE: 6 MB MEMORY DURING 1ST DOC: 9 MB MEMORY AFTER 1ST DOC: 7 MB MEMORY DURING 2ND DOC: 9 MB MEMORY AFTER 2ND DOC: 7 MB .... but is obviously a step backwards in performance. When I start reading a PDF (later in time, different thread) in the first case no more memory is allocated (staying at 25 MB), while in the second case memory goes up to 20 MB (from 7). In both cases, when I remove the CGContextDrawPDFPage(context, page); line memory is (nearly) constant at 6 MB during and after all preparations of documents. Can anybody explain whats going on there?

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  • How to reduce redundant code when adding new c++0x rvalue reference operator overloads

    - by Inverse
    I am adding new operator overloads to take advantage of c++0x rvalue references, and I feel like I'm producing a lot of redundant code. I have a class, tree, that holds a tree of algebraic operations on double values. Here is an example use case: tree x = 1.23; tree y = 8.19; tree z = (x + y)/67.31 - 3.15*y; ... std::cout << z; // prints "(1.23 + 8.19)/67.31 - 3.15*8.19" For each binary operation (like plus), each side can be either an lvalue tree, rvalue tree, or double. This results in 8 overloads for each binary operation: // core rvalue overloads for plus: tree operator +(const tree& a, const tree& b); tree operator +(const tree& a, tree&& b); tree operator +(tree&& a, const tree& b); tree operator +(tree&& a, tree&& b); // cast and forward cases: tree operator +(const tree& a, double b) { return a + tree(b); } tree operator +(double a, const tree& b) { return tree(a) + b; } tree operator +(tree&& a, double b) { return std::move(a) + tree(b); } tree operator +(double a, tree&& b) { return tree(a) + std::move(b); } // 8 more overloads for minus // 8 more overloads for multiply // 8 more overloads for divide // etc which also has to be repeated in a way for each binary operation (minus, multiply, divide, etc). As you can see, there are really only 4 functions I actually need to write; the other 4 can cast and forward to the core cases. Do you have any suggestions for reducing the size of this code? PS: The class is actually more complex than just a tree of doubles. Reducing copies does dramatically improve performance of my project. So, the rvalue overloads are worthwhile for me, even with the extra code. I have a suspicion that there might be a way to template away the "cast and forward" cases above, but I can't seem to think of anything.

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  • Loading the last related record instantly for multiple parent records using Entity framework

    - by Guillaume Schuermans
    Does anyone know a good approach using Entity Framework for the problem described below? I am trying for our next release to come up with a performant way to show the placed orders for the logged on customer. Of course paging is always a good technique to use when a lot of data is available I would like to see an answer without any paging techniques. Here's the story: a customer places an order which gets an orderstatus = PENDING. Depending on some strategy we move that order up the chain in order to get it APPROVED. Every change of status is logged so we can see a trace for statusses and maybe even an extra line of comment per status which can provide some extra valuable information to whoever sees this order in an interface. So an Order is linked to a Customer. One order can have multiple orderstatusses stored in OrderStatusHistory. In my testscenario I am using a customer which has 100+ Orders each with about 5 records in the OrderStatusHistory-table. I would for now like to see all orders in one page not using paging where for each Order I show the last relevant Status and the extra comment (if there is any for this last status; both fields coming from OrderStatusHistory; the record with the highest Id for the given OrderId). There are multiple scenarios I have tried, but I would like to see any potential other solutions or comments on the things I have already tried. Trying to do Include() when getting Orders but this still results in multiple queries launched on the database. Each order triggers an extra query to the database to get all orderstatusses in the history table. So all statusses are queried here instead of just returning the last relevant one, plus 100 extra queries are launched for 100 orders. You can imagine the problem when there are 100000+ orders in the database. Having 2 computed columns on the database: LastStatus, LastStatusInformation and a regular Linq-Query which gets those columns which are available through the Entity-model. The problem with this approach is the fact that those computed columns are determined using a scalar function which can not be changed without removing the formula from the computed column, etc... In the end I am very familiar with SQL and Stored procedures, but since the rest of the data-layer uses Entity Framework I would like to stick to it as long as possible, even though I have my doubts about performance. Using the SQL approach I would write something like this: WITH cte (RN, OrderId, [Status], Information) AS ( SELECT ROW_NUMBER() OVER (PARTITION BY OrderId ORDER BY Id DESC), OrderId, [Status], Information FROM OrderStatus ) SELECT o.Id, cte.[Status], cte.Information AS StatusInformation, o.* FROM [Order] o INNER JOIN cte ON o.Id = cte.OrderId AND cte.RN = 1 WHERE CustomerId = @CustomerId ORDER BY 1 DESC; which returns all orders for the customer with the statusinformation provided by the Common Table Expression. Does anyone know a good approach using Entity Framework?

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  • Is it Bad Practice to use C++ only for the STL containers?

    - by gmatt
    First a little background ... In what follows, I use C,C++ and Java for coding (general) algorithms, not gui's and fancy program's with interfaces, but simple command line algorithms and libraries. I started out learning about programming in Java. I got pretty good with Java and I learned to use the Java containers a lot as they tend to reduce complexity of book keeping while guaranteeing great performance. I intermittently used C++, but I was definitely not as good with it as with Java and it felt cumbersome. I did not know C++ enough to work in it without having to look up every single function and so I quickly reverted back to sticking to Java as much as possible. I then made a sudden transition into cracking and hacking in assembly language, because I felt I was concentrated too much attention on a much too high level language and I needed more experience with how a CPU interacts with memory and whats really going on with the 1's and 0's. I have to admit this was one of the most educational and fun experiences I've had with computers to date. For obviously reasons, I could not use assembly language to code on a daily basis, it was mostly reserved for fun diversions. After learning more about the computer through this experience I then realized that C++ is so much closer to the "level of 1's and 0's" than Java was, but I still felt it to be incredibly obtuse, like a swiss army knife with far too many gizmos to do any one task with elegance. I decided to give plain vanilla C a try, and I quickly fell in love. It was a happy medium between simplicity and enough "micromanagent" to not abstract what is really going on. However, I did miss one thing about Java: the containers. In particular, a simple container (like the stl vector) that expands dynamically in size is incredibly useful, but quite a pain to have to implement in C every time. Hence my code currently looks like almost entirely C with containers from C++ thrown in, the only feature I use from C++. I'd like to know if its consider okay in practice to use just one feature of C++, and ignore the rest in favor of C type code?

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  • Commitment to Zend Framework - any arguments against?

    - by Pekka
    I am refurbishing a big CMS that I have been working on for quite a number of years now. The product itself is great, but some components, the Database and translation classes for example, need urgent replacing - partly self-made as far back as 2002, grown into a bit of a chaos over time, and might have trouble surviving a security audit. So, I've been looking closely at a number of frameworks (or, more exactly, component Libraries, as I do not intend to change the basic structure of the CMS) and ended up with liking Zend Framework the best. They offer a solid MVC model but don't force you into it, and they offer a lot of professional components that have obviously received a lot of attention (Did you know there are multiple plurals in Russian, and you can't translate them using a simple ($number == 0) or ($number > 1) switch? I didn't, but Zend_Translate can handle it. Just to illustrate the level of thorougness the library seems to have been built with.) I am now literally at the point of no return, starting to replace key components of the system by the Zend-made ones. I'm not really having second thoughts - and I am surely not looking to incite a flame war - but before going onward, I would like to step back for a moment and look whether there is anything speaking against tying a big system closely to Zend Framework. What I like about Zend: As far as I can see, very high quality code Extremely well documented, at least regarding introductions to how things work (Haven't had to use detailed API documentation yet) Backed by a company that has an interest in seeing the framework prosper Well received in the community, has a considerable user base Employs coding standards I like Comes with a full set of unit tests Feels to me like the right choice to make - or at least, one of the right choices - in terms of modern, professional PHP development. I have been thinking about encapsulating and abstracting ZF's functionality into own classes to be able to switch frameworks more easily, but have come to the conclusion that this would not be a good idea because: it would be an unnecessary level of abstraction it could cost performance the big advantage of using a framework - the existence of a developer base that is familiar with its components - would partly be cancelled out therefore, the commitment to ZF would be a deep one. Thus my question: Is there anything substantial speaking against committing to the Zend Framework? Do you have insider knowledge of plans of Zend Inc.'s to go evil in 2011, and make it a closed source library? Is Zend Inc. run by vampires? Are there conceptual flaws in the code base you start to notice when you've transitioned all your projects to it? Is the appearance of quality code an illusion? Does the code look good, but run terribly slow on anything below my quad-core workstation?

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  • DB Design Pattern - Many to many classification / categorised tagging.

    - by Robin Day
    I have an existing database design that stores Job Vacancies. The "Vacancy" table has a number of fixed fields across all clients, such as "Title", "Description", "Salary range". There is an EAV design for "Custom" fields that the Clients can setup themselves, such as, "Manager Name", "Working Hours". The field names are stored in a "ClientText" table and the data stored in a "VacancyClientText" table with VacancyId, ClientTextId and Value. Lastly there is a many to many EAV design for custom tagging / categorising the vacancies with things such as Locations/Offices the vacancy is in, a list of skills required. This is stored as a "ClientCategory" table listing the types of tag, "Locations, Skills", a "ClientCategoryItem" table listing the valid values for each Category, e.g., "London,Paris,New York,Rome", "C#,VB,PHP,Python". Finally there is a "VacancyClientCategoryItem" table with VacancyId and ClientCategoryItemId for each of the selected items for the vacancy. There are no limits to the number of custom fields or custom categories that the client can add. I am now designing a new system that is very similar to the existing system, however, I have the ability to restrict the number of custom fields a Client can have and it's being built from scratch so I have no legacy issues to deal with. For the Custom Fields my solution is simple, I have 5 additional columns on the Vacancy Table called CustomField1-5. This removes one of the EAV designs. It is with the tagging / categorising design that I am struggling. If I limit a client to having 5 categories / types of tag. Should I create 5 tables listing the possible values "CustomCategoryItems1-5" and then an additional 5 many to many tables "VacancyCustomCategoryItem1-5" This would result in 10 tables performing the same storage as the three tables in the existing system. Also, should (heaven forbid) the requirements change in that I need 6 custom categories rather than 5 then this will result in a lot of code change. Therefore, can anyone suggest any DB Design Patterns that would be more suitable to storing such data. I'm happy to stick with the EAV approach, however, the existing system has come across all the usual performance issues and complex queries associated with such a design. Any advice / suggestions are much appreciated. The DBMS system used is SQL Server 2005, however, 2008 is an option if required for any particular pattern.

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  • how do i know how many clients are calling my WCF service function

    - by ZhengZhiren
    i am writing a program to test WCF service performance in high concurrency circumstance. On client side, i start many threads to call a WCF service function which returns a long list of data object. On server side, in that function called by my client, i need to know the number of clients calling the function. For doing that, i set a counter variable. In the beginning of the function, i add the counter by 1, but how can i decrease it after the funtion has returned the result? int clientCount=0; public DataObject[] GetData() { Interlocked.Increment(ref clientCount); List<DataObject> result = MockDb.GetData(); return result.ToArray(); Interlocked.Decrement(ref clientCount); //can't run to here... } i have seen a way in c++. Create a new class named counter. In the constructor of the counter class, increase the variable. And decrease it in the destructor. In the function, make a counter object so that its constructor will be called. And after the function returns, its destructor will be called. Like this: class counter { public: counter(){++clientCount; /* not simply like this, need to be atomic*/} ~counter(){--clientCount; /* not simply like this, need to be atomic*/} }; ... myfunction() { counter c; //do something return something; } In c# i think i can do so with the following codes, but not for sure. public class Service1 : IService1 { static int clientCount = 0; private class ClientCounter : IDisposable { public ClientCounter() { Interlocked.Increment(ref clientCount); } public void Dispose() { Interlocked.Decrement(ref clientCount); } } public DataObject[] GetData() { using (ClientCounter counter = new ClientCounter()) { List<DataObject> result = MockDb.GetData(); return result.ToArray(); } } } i write a counter class implement the IDisposable interface. And put my function codes into a using block. But it seems that it doesn't work so good. No matter how many threads i start, the clientCount variable is up to 3. Any advise would be appreciated.

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