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  • Monitoring UDP socket in glib(mm) eats up CPU time

    - by Gyorgy Szekely
    Hi, I have a GTKmm Windows application (built with MinGW) that receives UDP packets (no sending). The socket is native winsock and I use glibmm IOChannel to connect it to the application main loop. The socket is read with recvfrom. My problem is: this setup eats 25% percent CPU time on a 3GHz workstation. Can somebody tell me why? The application is idle in this case, and if I remove the UDP code, CPU usage drops down to almost zero. As the application has to perform some CPU intensive tasks, I could image better ways to spend that 25% Here are some code excerpts: (sorry for the printf's ;) ) /* bind */ void UDPInterface::bindToPort(unsigned short port) { struct sockaddr_in target; WSADATA wsaData; target.sin_family = AF_INET; target.sin_port = htons(port); target.sin_addr.s_addr = 0; if ( WSAStartup ( 0x0202, &wsaData ) ) { printf("WSAStartup failed!\n"); exit(0); // :) WSACleanup(); } sock = socket( AF_INET, SOCK_DGRAM, 0 ); if (sock == INVALID_SOCKET) { printf("invalid socket!\n"); exit(0); } if (bind(sock,(struct sockaddr*) &target, sizeof(struct sockaddr_in) ) == SOCKET_ERROR) { printf("failed to bind to port!\n"); exit(0); } printf("[UDPInterface::bindToPort] listening on port %i\n", port); } /* read */ bool UDPInterface::UDPEvent(Glib::IOCondition io_condition) { recvfrom(sock, (char*)buf, BUF_SIZE*4, 0, NULL, NULL); /* process packet... */ } /* glibmm connect */ Glib::RefPtr channel = Glib::IOChannel::create_from_win32_socket(udp.sock); Glib::signal_io().connect( sigc::mem_fun(udp, &UDPInterface::UDPEvent), channel, Glib::IO_IN ); I've read here in some other question, and also in glib docs (g_io_channel_win32_new_socket()) that the socket is put into nonblocking mode, and it's "a side-effect of the implementation and unavoidable". Does this explain the CPU effect, it's not clear to me? Whether or not I use glib to access the socket or call recvfrom() directly doesn't seem to make much difference, since CPU is used up before any packet arrives and the read handler gets invoked. Also glibmm docs state that it's ok to call recvfrom() even if the socket is polled (Glib::IOChannel::create_from_win32_socket()) I've tried compiling the program with -pg and created a per function cpu usage report with gprof. This wasn't usefull because the time is not spent in my program, but in some external glib/glibmm dll.

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  • Gradual memory leak in loop over contents of QTMovie

    - by Benji XVI
    I have a simple foundation tool that exports every frame of a movie as a .tiff file. Here is the relevant code: NSString* movieLoc = [NSString stringWithCString:argv[1]]; QTMovie *sourceMovie = [QTMovie movieWithFile:movieLoc error:nil]; int i=0; while (QTTimeCompare([sourceMovie currentTime], [sourceMovie duration]) != NSOrderedSame) { // save image of movie to disk NSAutoreleasePool *arp = [[NSAutoreleasePool alloc] init]; NSString *filePath = [NSString stringWithFormat:@"/somelocation_%d.tiff", i++]; NSData *currentImageData = [[sourceMovie currentFrameImage] TIFFRepresentation]; [currentImageData writeToFile:filePath atomically:NO]; NSLog(@"%@", filePath); [sourceMovie stepForward]; [arp release]; } [pool drain]; return 0; As you can see, in order to prevent very large memory buildups with the various transparently-autoreleased variables in the loop, we create, and flush, an autoreleasepool with every run through the loop. However, over the course of stepping through a movie, the amount of memory used by the program still gradually increases. Instruments is not detecting any memory leaks per se, but the object trace shows certain General Data blocks to be increasing in size. [Edited out reference to slowdown as it doesn't seem to be as much of a problem as I thought.] Edit: let's knock out some parts of the code inside the loop & see what we find out... Test 1 while (banana) { NSAutoreleasePool *arp = [[NSAutoreleasePool alloc] init]; NSString *filePath = [NSString stringWithFormat:@"/somelocation_%d.tiff", i++]; NSLog(@"%@", filePath); [sourceMovie stepForward]; [arp release]; } Here we simply loop over the whole movie, creating the filename and logging it. Memory characteristics: remains at 15MB usage for the duration. Test 2 while (banana) { NSAutoreleasePool *arp = [[NSAutoreleasePool alloc] init]; NSImage *image = [sourceMovie currentFrameImage]; [sourceMovie stepForward]; [arp release]; } Here we add back in the creation of the NSImage from the current frame. Memory characteristics: gradually increasing memory usage. RSIZE is at 60MB by frame 200; 75MB by f300. Test 3 while (banana) { NSAutoreleasePool *arp = [[NSAutoreleasePool alloc] init]; NSImage *image = [sourceMovie currentFrameImage]; NSData *imageData = [image TIFFRepresentation]; [sourceMovie stepForward]; [arp release]; } We've added back in the creation of an NSData object from the NSImage. Memory characteristics: Memory usage is again increasing: 62MB at f200; 75MB at f300. In other words, largely identical. It looks like a memory leak in the underlying system QTMovie uses to do currentFrameImage, to me.

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  • iPhone Image Resources, ICO vs PNG, app bundle filesize

    - by Jasarien
    My application has a collection of around 1940 icons that are used throughout. They're currently in ICO and new images provided to me come in ICO format too. I have noticed that they contain a 16x16 and 32x32 representation of each icon in one file. Each file is roughly 4KB in filesize (as reported by finder, but ls reports that they vary from being ~1000 bytes to 5000 bytes) A very small number of these icons only contain the 32x32 representation, and as a result are only around 700 bytes in size. Currently I am bundling these icons with my application and they are inflating the size of the app a bit more than I would like. Altogether, the images total just about 25.5MB. Xcode must do some kind of compression because the resulting app bundle is about 12.4MB. Compressing this further into a ZIP (as it would be when submitted to the App Store), results in a final file of 5.8MB. I'm aware that the maximum limit for over the air App Store downloads has been raised to 20MB since the introduction of the iPad (I'm not sure if that extends to iPhone apps as well as iPad apps though, if not the limit would be 10MB). My worry is that new icons are going to be added (sometimes up to 10 icons per week), and will continue to inflate the app bundle over time. What is the best way to distribute these icons with my app? Things I've tried and not had much success with: Converting the icons from ICO to PNG: I tried this in the hopes that the pngcrush utility would help out with the filesize. But it appears that it doesn't make much of a difference between a normal PNG and a crushed png (I believe it just optimises the image for display on the iPhone's GPU rather than compress it's size). Also in going from ICO to PNG actually increased the size of the icon file... Zipping the images, and then uncompressing them on first run. While this did reduce the overall image sizes, I found that the effort needed to unzip them, copy them to the documents folder and ensure that duplication doesn't happen on upgrades was too much hassle to be worth the benefit. Also, on original and 3G iPhones unzipping and copying around 25MB of images takes too long and creates a bad experience... Things I've considered but not yet tried: Instead of distributing the icons within the app bundle, host them online, and download each icon on demand (it depends on the user's data as to which icons will actually be displayed and when). Issues with this is that bandwidth costs money, and image downloads will be bandwidth intensive. However, my app currently has a small userbase of around 5,500 users (of which I estimate around 1500 to be active based on Flurry stats), and I have a huge unused bandwidth allowance with my current hosting package. So I'm open to thoughts on how to solve this tricky issue.

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  • How to resize the UIView when CGAffineTransformIdentity

    - by Gowtham
    I am doing an app which has a feature to rotate and re size a view. i have implemented this feature but i do face an issue. My problem The View wil be resized when dragging its four corners, after resizing it i can rotate the view in both directions. Once the rotation is done, if i try again to resize the view by dragging its corner, the view's size gone to unpredictable value and its moving all around the screen. I googled lot finally i got the following solution The frame property is undefined when transform != CGAffineTransformIdentity, as per the docs on UIView I saw one app which has implemented the feature exactly what i wish to implement. How can i resize the UIView after rotation of UIView My code for resize the view Touches Began - (void)touchesBegan:(NSSet *)touches withEvent:(UIEvent *)event{ UITouch *touch = [[event allTouches] anyObject]; NSLog(@"[touch view]:::%@",[touch view]); touchStart = [[touches anyObject] locationInView:testVw]; isResizingLR = (testVw.bounds.size.width - touchStart.x < kResizeThumbSize && testVw.bounds.size.height - touchStart.y < kResizeThumbSize); isResizingUL = (touchStart.x <kResizeThumbSize && touchStart.y <kResizeThumbSize); isResizingUR = (testVw.bounds.size.width-touchStart.x < kResizeThumbSize && touchStart.y<kResizeThumbSize); isResizingLL = (touchStart.x <kResizeThumbSize && testVw.bounds.size.height -touchStart.y <kResizeThumbSize); } Touches Moved - (void)touchesMoved:(NSSet *)touches withEvent:(UIEvent *)event{ CGPoint touchPoint = [[touches anyObject] locationInView:testVw]; CGPoint previous=[[touches anyObject]previousLocationInView:testVw]; float deltaWidth = touchPoint.x-previous.x; float deltaHeight = touchPoint.y-previous.y; NSLog(@"CVTM:%@",NSStringFromCGRect(testVw.frame)); if (isResizingLR) { testVw.frame = CGRectMake(testVw.frame.origin.x, testVw.frame.origin.y,touchPoint.x + deltaWidth, touchPoint.y + deltaWidth); } if (isResizingUL) { testVw.frame = CGRectMake(testVw.frame.origin.x + deltaWidth, testVw.frame.origin.y + deltaHeight, testVw.frame.size.width - deltaWidth, testVw.frame.size.height - deltaHeight); } if (isResizingUR) { testVw.frame = CGRectMake(testVw.frame.origin.x ,testVw.frame.origin.y + deltaHeight, testVw.frame.size.width + deltaWidth, testVw.frame.size.height - deltaHeight); } if (isResizingLL) { testVw.frame = CGRectMake(testVw.frame.origin.x + deltaWidth ,testVw.frame.origin.y , testVw.frame.size.width - deltaWidth, testVw.frame.size.height + deltaHeight); } if (!isResizingUL && !isResizingLR && !isResizingUR && !isResizingLL) { testVw.center = CGPointMake(testVw.center.x + touchPoint.x - touchStart.x,testVw.center.y + touchPoint.y - touchStart.y); } }

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  • Switch between speakerphone and headset on Android

    - by user210504
    Hi! I wish to know if there is a way, using which we can switch between the speaker and headset dynamically in an android application. I am using this sample code, I found online for my experiments final float frequency = 440; float increment = (float)(2*Math.PI) * frequency / 44100; // angular increment for each sample float angle = 0; AndroidAudioDevice device = new AndroidAudioDevice( ); AudioManager am = (AudioManager)getSystemService(AUDIO_SERVICE); am.setMode(AudioManager.MODE_IN_CALL); float samples[] = new float[1024]; int count = 0; while( count < 10 ) { count++; for( int i = 0; i < samples.length; i++ ) { samples[i] = (float)Math.sin( angle ) ; angle += increment; } device.writeSamples( samples ); } device.stop(); am.setMode(AudioManager.MODE_NORMAL); ---- next class public class AndroidAudioDevice { AudioTrack track; short[] buffer = new short[1024]; public AndroidAudioDevice( ) { int minSize =AudioTrack.getMinBufferSize( 44100, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT ); track = new AudioTrack( AudioManager.STREAM_VOICE_CALL, 44100, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, minSize, AudioTrack.MODE_STREAM); track.play(); } public void writeSamples(float[] samples) { fillBuffer( samples ); track.write( buffer, 0, samples.length ); } private void fillBuffer( float[] samples ) { if( buffer.length < samples.length ) buffer = new short[samples.length]; for( int i = 0; i < samples.length; i++ ) buffer[i] = (short)(samples[i] * Short.MAX_VALUE);; } public void stop() { track.stop(); } } As per my understanding this should play audio on headset, because we have not enabled the speaker phone. However, the audio is playing on the speaker phone. 1 Am I doing something wrong here? 2 What would be a way to switch between internal speaker and speaker phone dynamically for same code peice Any help will be appreciated.

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  • Looking for an Open Source Project in need of help

    - by hvidgaard
    Hi StackOverflow! I'm a CS student on well on my way to graduate. I have had a difficult time of finding relevant student jobs (they seems to be taken merely hours after the notice gets on the board) , so instead I'm looking for an open source project in need of help. I'm aware that I should choose one that I use, but I'm not aware of any OS-project that I use that needs help. That's why I'm asking you. I don't have any deep experience, but I here are some of my biggest projects so far: BitTorrent-ish client in Python (a subset of BitTorrent) HTTP 1.1 webserver in Java Compiler from a subset of Java to run on JRE Flash-framework project to model an iPad look and feel (not to run actual iPad programs) complete with an API for programs. Complete MySQL database for a booking system, with departure and arrival times, so you could only book valid tickets (with a Java frontend). I know, Java and languages like AS3 and C# feels natural per se, Python, and have done a fair bit of hacking around in C, but I don't feel very comfortable with it. Mostly I'm afraid to make a fuckup because I have such a high degree of control. I would like to think I'm well aware of good software design practices, but in reality what I do is ask myself "would I like to use/maintain this?", and I love to refactor my code because I see optimizations. I love algorithms and to make them run in the best possible time. I don't have any preferred domain to work in, but I wouldn't mind it to be graphics or math heavy. Ideally I'm looking for a project in C++ to learn the in's and out's of it, but I'm well aware that I don't know that language very well. I would like to have a mentor-like figure until I'm confident enough to stand on my own, not one to review all my code (I'm sure someone will to start with anyway), but to ask questions about the project and language in question. I do have a wife and two children, so don't expect me to put in 10+ hours every week. In return I can work on my own, I strive to program modular and maintainable code. Know how to read an API, use Google, StackOverflow and online resources in general. If you have any questions, shoot. I'm looking forward to your suggestions.

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  • DRY-ing very similar specs for ASP.NET MVC controller action with MSpec (BDD guidelines)

    - by spapaseit
    Hi all, I have two very similar specs for two very similar controller actions: VoteUp(int id) and VoteDown(int id). These methods allow a user to vote a post up or down; kinda like the vote up/down functionality for StackOverflow questions. The specs are: VoteDown: [Subject(typeof(SomeController))] public class When_user_clicks_the_vote_down_button_on_a_post : SomeControllerContext { Establish context = () => { post = PostFakes.VanillaPost(); post.Votes = 10; session.Setup(s => s.Single(Moq.It.IsAny<Expression<Func<Post, bool>>>())).Returns(post); session.Setup(s => s.CommitChanges()); }; Because of = () => result = controller.VoteDown(1); It should_decrement_the_votes_of_the_post_by_1 = () => suggestion.Votes.ShouldEqual(9); It should_not_let_the_user_vote_more_than_once; } VoteUp: [Subject(typeof(SomeController))] public class When_user_clicks_the_vote_down_button_on_a_post : SomeControllerContext { Establish context = () => { post = PostFakes.VanillaPost(); post.Votes = 0; session.Setup(s => s.Single(Moq.It.IsAny<Expression<Func<Post, bool>>>())).Returns(post); session.Setup(s => s.CommitChanges()); }; Because of = () => result = controller.VoteUp(1); It should_increment_the_votes_of_the_post_by_1 = () => suggestion.Votes.ShouldEqual(1); It should_not_let_the_user_vote_more_than_once; } So I have two questions: How should I go about DRY-ing these two specs? Is it even advisable or should I actually have one spec per controller action? I know I Normally should, but this feels like repeating myself a lot. Is there any way to implement the second It within the same spec? Note that the It should_not_let_the_user_vote_more_than_once; requires me the spec to call controller.VoteDown(1) twice. I know the easiest would be to create a separate spec for it too, but it'd be copying and pasting the same code yet again... I'm still getting the hang of BDD (and MSpec) and many times it is not clear which way I should go, or what the best practices or guidelines for BDD are. Any help would be appreciated.

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  • how to design this relation in a DB schema

    - by raticulin
    I have a table Car in my db, one of the columns is purchaseDate. I want to be able to tag every car with a number of Policies (limited to 10 policies). Each policy has a time to life (ttl, a duration of time, like '5 years', '10 months' etc), that is, for how long since the car's purchaseDate the policy can be applied. I need to perform the following actions: when inserting a Car, it will be set with a number of Policies (at least one is set) sometimes a Car will be updated to add/remove a Policy searches must be done taking into account date/policies, for example: 'select all cars that are not covered by any policy as of today' My current design is (pol0..pol9 are the policies): CREATE TABLE Car ( id int NOT NULL IDENTITY(1,1), purchaseDate datetime NOT NULL, //more stuff... pol0 smallint default NULL, pol1 smallint default NULL, pol2 smallint default NULL, pol3 smallint default NULL, pol4 smallint default NULL, pol5 smallint default NULL, pol6 smallint default NULL, pol7 smallint default NULL, pol8 smallint default NULL, pol9 smallint default NULL, PRIMARY KEY (id) ) CREATE TABLE Policy ( id smallint NOT NULL, name varchar(50) collate Latin1_General_BIN NOT NULL, ttl varchar(100) collate Latin1_General_BIN NOT NULL, PRIMARY KEY (id) ) The problem I am facing is that the sql to perform the query above is a nightmare to write. As I don't know in which column each policy can be, so I have to check all columns for every policy etc etc. So I am wondering wether it is worth changing this. My questions are: The smallint as Policy id was chosen instead of an 'int IDENTITY' in order to save some space as there are going to be millions of Car records. It just adds complexity when creating a Policy as we must handle the id etc. Was it worth doing this? I am thinking that maybe there is a much better design? Obviously we could move the policy/car relation to its own table CarPolicy, benefits would be: no limit on 10 policies per car adding/removing etc much easier when only the default policy is applied (when no others are applied one called Default policy is applied), we could signal that by not having any entry in CarPolicy, now this is just done inserting the Default policy id in one of the columns. The cons are that we would need to change the DB, ORM classes etc. What would you recommend? Maybe there is another smart way to implement this that we are not aware without using the CarPolicy table?

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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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  • JPA Database strcture for internationalisation

    - by IrishDubGuy
    I am trying to get a JPA implementation of a simple approach to internationalisation. I want to have a table of translated strings that I can reference in multiple fields in multiple tables. So all text occurrences in all tables will be replaced by a reference to the translated strings table. In combination with a language id, this would give a unique row in the translated strings table for that particular field. For example, consider a schema that has entities Course and Module as follows :- Course int course_id, int name, int description Module int module_id, int name The course.name, course.description and module.name are all referencing the id field of the translated strings table :- TranslatedString int id, String lang, String content That all seems simple enough. I get one table for all strings that could be internationalised and that table is used across all the other tables. How might I do this in JPA, using eclipselink 2.4? I've looked at embedded ElementCollection, ala this... JPA 2.0: Mapping a Map - it isn't exactly what i'm after cos it looks like it is relating the translated strings table to the pk of the owning table. This means I can only have one translatable string field per entity (unless I add new join columns into the translatable strings table, which defeats the point, its the opposite of what I am trying to do). I'm also not clear on how this would work across entites, presumably the id of each entity would have to use a database wide sequence to ensure uniqueness of the translatable strings table. BTW, I tried the example as laid out in that link and it didn't work for me - as soon as the entity had a localizedString map added, persisting it caused the client side to bomb but no obvious error on the server side and nothing persisted in the DB :S I been around the houses on this about 9 hours so far, I've looked at this Internationalization with Hibernate which appears to be trying to do the same thing as the link above (without the table definitions it hard to see what he achieved). Any help would be gratefully achieved at this point... Edit 1 - re AMS anwser below, I'm not sure that really addresses the issue. In his example it leaves the storing of the description text to some other process. The idea of this type of approach is that the entity object takes the text and locale and this (somehow!) ends up in the translatable strings table. In the first link I gave, the guy is attempting to do this by using an embedded map, which I feel is the right approach. His way though has two issues - one it doesn't seem to work! and two if it did work, it is storing the FK in the embedded table instead of the other way round (I think, I can't get it to run so I can't see exactly how it persists). I suspect the correct approach ends up with a map reference in place of each text that needs translating (the map being locale-content), but I can't see how to do this in a way that allows for multiple maps in one entity (without having corresponding multiple columns in the translatable strings table)...

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  • Can't destroy record in many-to-many relationship

    - by Dmart
    I'm new to Rails, so I'm sure I've made a simple mistake. I've set up a many-to-many relationship between two models: User and Group. They're connected through the junction model GroupMember. Here are my models (removed irrelevant stuff): class User < ActiveRecord::Base has_many :group_members has_many :groups, :through => :group_members end class GroupMember < ActiveRecord::Base belongs_to :group belongs_to :user end class Group < ActiveRecord::Base has_many :group_members has_many :users, :through => :group_members end The table for GroupMembers contains additional information about the relationship, so I didn't use has_and_belongs_to_many (as per the Rails "Active Record Associations" guide). The problem I'm having is that I can't destroy a GroupMember. Here's the output from rails console: irb(main):006:0> m = GroupMember.new => #<GroupMember group_id: nil, user_id: nil, active: nil, created_at: nil, updated_at: nil> irb(main):007:0> m.group_id =1 => 1 irb(main):008:0> m.user_id = 16 => 16 irb(main):009:0> m.save => true irb(main):010:0> m.destroy NoMethodError: undefined method `eq' for nil:NilClass from /usr/local/lib/ruby/gems/1.8/gems/activesupport-3.0.4/lib/active_support/whiny_nil.rb:48:in `method_missing' from /usr/local/lib/ruby/gems/1.8/gems/activerecord-3.0.4/lib/active_record/persistence.rb:79:in `destroy' from /usr/local/lib/ruby/gems/1.8/gems/activerecord-3.0.4/lib/active_record/locking/optimistic.rb:110:in `destroy' from /usr/local/lib/ruby/gems/1.8/gems/activerecord-3.0.4/lib/active_record/callbacks.rb:260:in `destroy' from /usr/local/lib/ruby/gems/1.8/gems/activesupport-3.0.4/lib/active_support/callbacks.rb:413:in `_run_destroy_callbacks' from /usr/local/lib/ruby/gems/1.8/gems/activerecord-3.0.4/lib/active_record/callbacks.rb:260:in `destroy' from /usr/local/lib/ruby/gems/1.8/gems/activerecord-3.0.4/lib/active_record/transactions.rb:235:in `destroy' from /usr/local/lib/ruby/gems/1.8/gems/activerecord-3.0.4/lib/active_record/transactions.rb:292:in `with_transaction_returning_status' from /usr/local/lib/ruby/gems/1.8/gems/activerecord-3.0.4/lib/active_record/connection_adapters/abstract/database_statements.rb:139:in `transaction' from /usr/local/lib/ruby/gems/1.8/gems/activerecord-3.0.4/lib/active_record/transactions.rb:207:in `transaction' from /usr/local/lib/ruby/gems/1.8/gems/activerecord-3.0.4/lib/active_record/transactions.rb:290:in `with_transaction_returning_status' from /usr/local/lib/ruby/gems/1.8/gems/activerecord-3.0.4/lib/active_record/transactions.rb:235:in `destroy' from (irb):10 This is driving me crazy, so any help would be greatly appreciated.

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  • Foreign key pointing to different tables

    - by Álvaro G. Vicario
    I'm implementing a table per subclass design I discussed in a previous question. It's a product database where products can have very different attributes depending on their type, but attributes are fixed for each type and types are not manageable at all. I have a master table that holds common attributes: product_type ============ product_type_id INT product_type_name VARCHAR E.g.: 1 'Magazine' 2 'Web site' product ======= product_id INT product_name VARCHAR product_type_id INT -> Foreign key to product_type.product_type_id valid_since DATETIME valid_to DATETIME E.g. 1 'Foo Magazine' 1 '1998-12-01' NULL 2 'Bar Weekly Review' 1 '2005-01-01' NULL 3 'E-commerce App' 2 '2009-10-15' NULL 4 'CMS' 2 '2010-02-01' NULL ... and one subtable for each product type: item_magazine ============= item_magazine_id INT title VARCHAR product_id INT -> Foreign key to product.product_id issue_number INT pages INT copies INT close_date DATETIME release_date DATETIME E.g. 1 'Foo Magazine Regular Issue' 1 89 52 150000 '2010-06-25' '2010-06-31' 2 'Foo Magazine Summer Special' 1 90 60 175000 '2010-07-25' '2010-07-31' 3 'Bar Weekly Review Regular Issue' 2 12 16 20000 '2010-06-01' '2010-06-02' item_web_site ============= item_web_site_id INT name VARCHAR product_id INT -> Foreign key to product.product_id bandwidth INT hits INT date_from DATETIME date_to DATETIME E.g. 1 'The Carpet Store' 3 10 90000 '2010-06-01' NULL 2 'Penauts R Us' 3 20 180000 '2010-08-01' NULL 3 'Springfield Cattle Fair' 4 15 150000 '2010-05-01' '2010-10-31' Now I want to add some fees that relate to one specific item. Since there are very little subtypes, it's feasible to do this: fee === fee_id INT fee_description VARCHAR item_magazine_id INT -> Foreign key to item_magazine.item_magazine_id item_web_site_id INT -> Foreign key to item_web_site.item_web_site_id net_price DECIMAL E.g.: 1 'Front cover' 2 NULL 1999.99 2 'Half page' 2 NULL 500.00 3 'Square banner' NULL 3 790.50 4 'Animation' NULL 3 2000.00 I have tight foreign keys to handle cascaded editions and I presume I can add a constraint so only one of the IDs is NOT NULL. However, my intuition suggests that it would be cleaner to get rid of the item_WHATEVER_id columns and keep a separate table: fee_to_item =========== fee_id INT -> Foreign key to fee.fee_id product_id INT -> Foreign key to product.product_id item_id INT -> ??? But I can't figure out how to create foreign keys on item_id since the source table varies depending on product_id. Should I stick to my original idea?

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  • C#: My callback function gets called twice for every Sent Request

    - by Madi D.
    I've Got a program that uploads/downloads files into an online server,Has a callback to report progress and log it into a textfile, The program is built with the following structure: public void Upload(string source, string destination) { //Object containing Source and destination to pass to the threaded function KeyValuePair<string, string> file = new KeyValuePair<string, string>(source, destination); //Threading to make sure no blocking happens after calling upload Function Thread t = new Thread(new ParameterizedThreadStart(amazonHandler.TUpload)); t.Start(file); } private void TUpload(object fileInfo) { KeyValuePair<string, string> file = (KeyValuePair<string, string>)fileInfo; /* Some Magic goes here,Checking The file and Authorizing Upload */ var ftiObject = new FtiObject () { FileNameOnHDD = file.Key, DestinationPath = file.Value, //Has more data used for calculations. }; //Threading to make sure progress gets callback gets called. Thread t = new Thread(new ParameterizedThreadStart(amazonHandler.UploadOP)); t.Start(ftiObject); //Signal used to stop progress untill uploadCompleted is called. uploadChunkDoneSignal.WaitOne(); /* Some Extra Code */ } private void UploadOP(object ftiSentObject) { FtiObject ftiObject = (FtiObject)ftiSentObject; /* Some useless code to create the uri and prepare the ftiObject. */ // webClient.UploadFileAsync will open a thread that // will upload the file and report // progress/complete using registered callback functions. webClient.UploadFileAsync(uri, "PUT", ftiObject.FileNameOnHDD, ftiObject); } I got a callback that is registered to the Webclient's UploadProgressChanged event , however it is getting called twice per sent request. void UploadProgressCallback(object sender, UploadProgressChangedEventArgs e) { FtiObject ftiObject = (FtiObject )e.UserState; Logger.log(ftiObject.FileNameOnHDD, (double)e.BytesSent ,e.TotalBytesToSend); } Log Output: Filename: C:\Text1.txt Uploaded:1024 TotalFileSize: 665241 Filename: C:\Text1.txt Uploaded:1024 TotalFileSize: 665241 Filename: C:\Text1.txt Uploaded:2048 TotalFileSize: 665241 Filename: C:\Text1.txt Uploaded:2048 TotalFileSize: 665241 Filename: C:\Text1.txt Uploaded:3072 TotalFileSize: 665241 Filename: C:\Text1.txt Uploaded:3072 TotalFileSize: 665241 Etc... I am watching the Network Traffic using a watcher, and only 1 request is being sent. Some how i cant Figure out why the callback is being called twice, my doubt was that the callback is getting fired by each thread opened(the main Upload , and TUpload), however i dont know how to test if thats the cause. Note: The reason behind the many /**/ Comments is to indicate that the functions do more than just opening threads, and threading is being used to make sure no blocking occurs (there a couple of "Signal.WaitOne()" around the code for synchronization)

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  • How to measure a canvas that has auto height and width

    - by Wymmeroo
    Hi Folks, I'm a beginner in silverlight so i hope i can get an answer that brings me some more light in the measure process of silverlight. I found an interessting flap out control from silverlight slide control and now I try to use it in my project. So that the slide out is working proper, I have to place the user control on a canvas. The user control then uses for itself the height of its content. I just wanna change that behavior so that the height is set to the available space from the parent canvas. You see the uxBorder where the height is set. How can I measure the actual height and set it to the border? I tried it with Height={Binding ElementName=notificationCanvas, Path=ActualHeight} but this dependency property has no callback, so the actualHeight is never set. What I want to achieve is a usercontrol like the tweetboard per example on Jesse Liberty's blog Sorry for my English writing, I hope you understand my question. <Canvas x:Name="notificationCanvas" Background="Red"> <SlideEffectEx:SimpleSlideControl GripWidth="20" GripTitle="Task" GripHeight="100"> <Border x:Name="uxBorder" BorderThickness="2" CornerRadius="5" BorderBrush="DarkGray" Background="DarkGray" Padding="5" Width="300" Height="700" > <StackPanel> <TextBlock Text="Tasks"></TextBlock> <Button x:Name="btn1" Margin="5" Content="{Binding ElementName=MainBorder, Path=Height}"></Button> <Button x:Name="btn2" Margin="5" Content="Second Button"></Button> <Button x:Name="btn3" Margin="5" Content="Third Button"></Button> <Button x:Name="btn1_Copy" Margin="5" Content="First Button"/> <Button x:Name="btn1_Copy1" Margin="5" Content="First Button"/> <Button x:Name="btn1_Copy2" Margin="5" Content="First Button"/> <Button x:Name="btn1_Copy3" Margin="5" Content="First Button"/> <Button x:Name="btn1_Copy4" Margin="5" Content="First Button"/> <Button x:Name="btn1_Copy5" Margin="5" Content="First Button"/> <Button x:Name="btn1_Copy6" Margin="5" Content="First Button"/> </StackPanel> </Border> </SlideEffectEx:SimpleSlideControl>

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  • X264 encoding using Opencv

    - by user573193
    I am working with a high resolution camera: 4008x2672. I a writing a simple program which grabs frame from the camera and sends the frame to a avi file. For working with such a high resolution, I found only x264 codec that could do the trick (Suggestions welcome). I am using opencv for most of the image handling stuff. As mentioned in this post http://doom10.org/index.php?topic=1019.0 , I modified the AVCodecContext members as per ffmpeg presets for libx264 (Had to do this to avoid broken ffmpeg defaults settings error). This is output I am getting when I try to run the program [libx264 @ 0x992d040]non-strictly-monotonic PTS 1294846981.526675 1 0 //Timestamp camera_no frame_no 1294846981.621101 1 1 1294846981.715521 1 2 1294846981.809939 1 3 1294846981.904360 1 4 1294846981.998782 1 5 1294846982.093203 1 6 Last message repeated 7 times [avi @ 0x992beb0]st:0 error, non monotone timestamps -614891469123651720 = -614891469123651720 OpenCV Error: Unspecified error (Error while writing video frame) in icv_av_write_frame_FFMPEG, file /home/ajoshi/ext/OpenCV-2.2.0/modules/highgui/src/cap_ffmpeg.cpp, line 1034 terminate called after throwing an instance of 'cv::Exception' what(): /home/ajoshi/ext/OpenCV-2.2.0/modules/highgui/src/cap_ffmpeg.cpp:1034: error: (-2) Error while writing video frame in function icv_av_write_frame_FFMPEG Aborted Modifications to the AVCodecContext are: if(codec_id == CODEC_ID_H264) { //fprintf(stderr, "Trying to parse a preset file for libx264\n"); //Setting Values manually from medium preset c-me_method = 7; c-qcompress=0.6; c-qmin = 10; c-qmax = 51; c-max_qdiff = 4; c-i_quant_factor=0.71; c-max_b_frames=3; c-b_frame_strategy = 1; c-me_range = 16; c-me_subpel_quality=7; c-coder_type = 1; c-scenechange_threshold=40; c-partitions = X264_PART_I8X8 | X264_PART_I4X4 | X264_PART_P8X8 | X264_PART_B8X8; c-flags = CODEC_FLAG_LOOP_FILTER; c-flags2 = CODEC_FLAG2_BPYRAMID | CODEC_FLAG2_MIXED_REFS | CODEC_FLAG2_WPRED | CODEC_FLAG2_8X8DCT | CODEC_FLAG2_FASTPSKIP; c-keyint_min = 25; c-refs = 3; c-trellis=1; c-directpred = 1; c-weighted_p_pred=2; } I am probably not setting the dts and pts values which I believed ffmpeg should be setting it for me. Any sugggestions welcome. Thanks in advance

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  • Asp.net Mvc - Kigg: Maintain User object in HttpContext.Items between requests.

    - by Pickels
    Hallo, first I want to say that I hope this doesn't look like I am lazy but I have some trouble understanding a piece of code from the following project. http://kigg.codeplex.com/ I was going through the source code and I noticed something that would be usefull for my own little project I am making. In their BaseController they have the following code: private static readonly Type CurrentUserKey = typeof(IUser); public IUser CurrentUser { get { if (!string.IsNullOrEmpty(CurrentUserName)) { IUser user = HttpContext.Items[CurrentUserKey] as IUser; if (user == null) { user = AccountRepository.FindByClaim(CurrentUserName); if (user != null) { HttpContext.Items[CurrentUserKey] = user; } } return user; } return null; } } This isn't an exact copy of the code I adjusted it a little to my needs. This part of the code I still understand. They store their IUser in HttpContext.Items. I guess they do it so that they don't have to call the database eachtime they need the User object. The part that I don't understand is how they maintain this object in between requests. If I understand correctly the HttpContext.Items is a per request cache storage. So after some more digging I found the following code. internal static IDictionary<UnityPerWebRequestLifetimeManager, object> GetInstances(HttpContextBase httpContext) { IDictionary<UnityPerWebRequestLifetimeManager, object> instances; if (httpContext.Items.Contains(Key)) { instances = (IDictionary<UnityPerWebRequestLifetimeManager, object>) httpContext.Items[Key]; } else { lock (httpContext.Items) { if (httpContext.Items.Contains(Key)) { instances = (IDictionary<UnityPerWebRequestLifetimeManager, object>) httpContext.Items[Key]; } else { instances = new Dictionary<UnityPerWebRequestLifetimeManager, object>(); httpContext.Items.Add(Key, instances); } } } return instances; } This is the part where some magic happens that I don't understand. I think they use Unity to do some dependency injection on each request? In my project I am using Ninject and I am wondering how I can get the same result. I guess InRequestScope in Ninject is the same as UnityPerWebRequestLifetimeManager? I am also wondering which class/method they are binding to which interface? Since the HttpContext.Items get destroyed each request how do they prevent losing their user object? Anyway it's kinda a long question so I am gradefull for any push in the right direction. Kind regards, Pickels

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  • git: setting a single tracking remote from a public repo.

    - by Gauthier
    I am confused with remote branches. My local repo: (local) ---A---B---C-master My remote repo (called int): (int) ---A---B---C---D---E-master What I want to do is to setup the local repo's master branch to follow that of int. Local repo: (local) ---A---B---C---D---E-master-remotes/int/master So that when int changes to: (int) ---A---B---C---D---E---F-master I can run git pull from the local repo's master and get (local) ---A---B---C---D---E---F-master-remotes/int/master Here's what I have tried: git fetch int gets me all the branches of int into remote branches. This can get messy since int might have hundreds of branches. git fetch int master gets me the commits, but no ref to it, only FETCH_HEAD. No remote branch either. git fetch int master:new_master works but I don't want a new name every time I update, and no remote branch is setup. git pull int master does what I want, but there is still no remote branch setup. I feel that it is ok to do so (that's the best I have now), but I read here and there that with the remote setup it is enough with git pull. git branch --track new_master int/master, as per http://www.gitready.com/beginner/2009/03/09/remote-tracking-branches.html . I get "not a valid object name: int/master". git remote -v does show me that int is defined and points at the correct location (1. worked). What I miss is the int/master branch, which is precisely what I want to get. git fetch in master:int/master. Well, int/master is created, but is no remote. So to summarize, I've tried some stuff with no luck. I would expect 2 to give me the remote branch to master in the repo int. The solution I use now is option 3. I read somewhere that you could change some config file by hand, but isn't that a bit cumbersome? The "cumbersome" way of editting the config file did work: [branch "master"] remote = int merge = master It can be done from command line: $ git config branch.master.remote int $ git config branch.master.merge master Any reason why option 2 above wouldn't do that automatically? Even in that case, git pull fetches all branches from the remote.

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  • Converting C source to C++

    - by Barry Kelly
    How would you go about converting a reasonably large (300K), fairly mature C codebase to C++? The kind of C I have in mind is split into files roughly corresponding to modules (i.e. less granular than a typical OO class-based decomposition), using internal linkage in lieu private functions and data, and external linkage for public functions and data. Global variables are used extensively for communication between the modules. There is a very extensive integration test suite available, but no unit (i.e. module) level tests. I have in mind a general strategy: Compile everything in C++'s C subset and get that working. Convert modules into huge classes, so that all the cross-references are scoped by a class name, but leaving all functions and data as static members, and get that working. Convert huge classes into instances with appropriate constructors and initialized cross-references; replace static member accesses with indirect accesses as appropriate; and get that working. Now, approach the project as an ill-factored OO application, and write unit tests where dependencies are tractable, and decompose into separate classes where they are not; the goal here would be to move from one working program to another at each transformation. Obviously, this would be quite a bit of work. Are there any case studies / war stories out there on this kind of translation? Alternative strategies? Other useful advice? Note 1: the program is a compiler, and probably millions of other programs rely on its behaviour not changing, so wholesale rewriting is pretty much not an option. Note 2: the source is nearly 20 years old, and has perhaps 30% code churn (lines modified + added / previous total lines) per year. It is heavily maintained and extended, in other words. Thus, one of the goals would be to increase mantainability. [For the sake of the question, assume that translation into C++ is mandatory, and that leaving it in C is not an option. The point of adding this condition is to weed out the "leave it in C" answers.]

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  • How do I check for the existence of an external file with XSL?

    - by LOlliffe
    I've found a lot of examples that reference Java and C for this, but how do I, or can I, check for the existence of an external file with XSL. First, I realize that this is only a snippet, but it's part of a huge stylesheet, so I'm hoping it's enough to show my issue. <!-- Use this template for Received SMSs --> <xsl:template name="ReceivedSMS"> <!-- Set/Declare "SMSname" variable (local, evaluates per instance) --> <xsl:variable name="SMSname"> <xsl:value-of select=" following-sibling::Name"/> </xsl:variable> <fo:table font-family="Arial Unicode MS" font-size="8pt" text-align="start"> <fo:table-column column-width=".75in"/> <fo:table-column column-width="6.75in"/> <fo:table-body> <fo:table-row> <!-- Cell contains "speakers" icon --> <fo:table-cell display-align="after"> <fo:block text-align="start"> <fo:external-graphic src="../images/{$SMSname}.jpg" content-height="0.6in"/> What I'd like to do, is put in an "if" statement, surronding the {$SMSname}.jpg line. That is: <fo:block text-align="start"> <xsl:if test="exists( the external file {$SMSname}.jpg)"> <fo:external-graphic src="../images/{$SMSname}.jpg" content-height="0.6in"/> </xsl:if> <xsl:if test="not(exists( the external file {$SMSname}.jpg))"> <fo:external-graphic src="../images/unknown.jpg" content-height="0.6in"/> </xsl:if> </fo:block> Because of "grouping", etc., I'm using XSLT 2.0. I hope that this is something that can be done. I hope even more that it's something simple. As always, thanks in advance for any help. LO

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  • How do I add a filter button to this pagination?

    - by ClarkSKent
    Hey, I want to add a button(link), that when clicked will filter the pagination results. I'm new to php (and programming in general) and would like to add a button like 'Automotive' and when clicked it updates the 2 mysql queries in my pagination script, seen here: As you can see, the category automotive is hardcoded in, I want it to be dynamic, so when a link is clicked it places whatever the id or class is in the category part of the query. 1: $record_count = mysql_num_rows(mysql_query("SELECT * FROM explore WHERE category='automotive'")); 2: $get = mysql_query("SELECT * FROM explore WHERE category='automotive' LIMIT $start, $per_page"); This is the entire current php pagination script that I am using: <?php //connecting to the database $error = "Could not connect to the database"; mysql_connect('localhost','root','root') or die($error); mysql_select_db('ajax_demo') or die($error); //max displayed per page $per_page = 2; //get start variable $start = $_GET['start']; //count records $record_count = mysql_num_rows(mysql_query("SELECT * FROM explore WHERE category='automotive'")); //count max pages $max_pages = $record_count / $per_page; //may come out as decimal if (!$start) $start = 0; //display data $get = mysql_query("SELECT * FROM explore WHERE category='automotive' LIMIT $start, $per_page"); while ($row = mysql_fetch_assoc($get)) { // get data $name = $row['id']; $age = $row['site_name']; echo $name." (".$age.")<br />"; } //setup prev and next variables $prev = $start - $per_page; $next = $start + $per_page; //show prev button if (!($start<=0)) echo "<a href='pagi_test.php?start=$prev'>Prev</a> "; //show page numbers //set variable for first page $i=1; for ($x=0;$x<$record_count;$x=$x+$per_page) { if ($start!=$x) echo " <a href='pagi_test.php?start=$x'>$i</a> "; else echo " <a href='pagi_test.php?start=$x'><b>$i</b></a> "; $i++; } //show next button if (!($start>=$record_count-$per_page)) echo " <a href='pagi_test.php?start=$next'>Next</a>"; ?>

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  • Two loops speeds drawing in a Jframe

    - by noahn567
    I have a program that requires two classes. The player-Names class, and the Player-Model class. I want the player-Names class to repaint every half second, and the Player-Model class to repaint 60 times per second because i want the movement to be smooth. The problem that i am having is that i want all of this to be done on one J-frame. How would i go about doing this? If you could lead me in the right direction or give me a little example that would be great! Thank you :). for some reason it wont let me post so i'm going to put in some random code import java.awt.Color; import java.awt.Font; import java.awt.Graphics; import java.awt.Graphics2D; import java.awt.RenderingHints; import javax.swing.JComponent; import javax.swing.JFrame; public class PlayerNames extends JFrame { static int connectionTimer = 0; static int connectionTimer2 = 0; static int reconnect = 0; static int reconnectValue = 1; static int x = 0; static int reconnectWait = connectionTimer + reconnectValue; private static final long serialVersionUID = 1L; public graph gg = new graph(); public graph g = new graph(); private static GameClient socketClient; private GameServer socketServer; public static void main(int width, int height) { PlayerNames tt = new PlayerNames(); // PlayerGraphics t = new PlayerGraphics(); tt.setSize(width, height); if (Game.ServerOwner == 1) { tt.setTitle("Server: " + Game.username); } else { tt.setTitle("Username: " + Game.username); } tt.setVisible(true); tt.getContentPane().add(tt.gg); tt.getContentPane().add(tt.g); tt.setDefaultCloseOperation(EXIT_ON_CLOSE); tt.setResizable(false); }

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  • Google Maps rendering locally but not in live environment

    - by marcusstarnes
    I have a page that renders a simple google map for a specified location. This map renders without any problems at all when I run it locally on localhost, however, when I deploy this code to our live web servers (using our LIVE google API key for the appropriate domain) it fails to render, and upon putting a series of alerts within the javascript on the page, it appears that the 'Initialize' method (which should be called within body onLoad) is not being called. When I view the HTML source that is rendered on the live server it appears exactly as per the local version of the site (including the call to initialize() within the body onLoad event), albeit with the different maps API key. I have output the host (alert(window.location.host);) to ensure that the key I generated via the google maps api site, corresponds exactly to the live server, which it does. Does anyone have any ideas why it would be working locally but not when deployed to the live servers? The live site is hosted on 2 load-balanced web servers. This is the javascript that is rendered: <script src="http://maps.google.com/maps?file=api&amp;v=2&amp;sensor=false&amp;key=ABQIAAAA-BU8POZj19wRlTaKIXVM9xTz76xxk4yAELG9u79oXrhnLTB5NRRvAZ-bkKn1x8J68nfRTVOIWNPJEA" type="text/javascript"></script> <script type="text/javascript"> var map; var geocoder; alert(window.location.host); function initialize() { if (GBrowserIsCompatible()) { map = new GMap2(document.getElementById("businessMap")); map.setUIToDefault(); geocoder = new GClientGeocoder(); showAddress('St Margarets Street SW1P 3 London'); } } function showAddress(address) { geocoder.getLatLng( address, function(point) { if (!point) { // Address could not be located. jQuery('#googleMap').hide(); } else { map.setCenter(point, 13); var marker = new GMarker(point); map.addOverlay(marker); var html = 'Address info for the marker'; marker.openInfoWindow(html); GEvent.addListener(marker, "click", function() { marker.openInfoWindowHtml(html); }); } } ); } </script> Any help would be much appreciated. Thanks.

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  • Patterns: Local Singleton vs. Global Singleton?

    - by Mike Rosenblum
    There is a pattern that I use from time to time, but I'm not quite sure what it is called. I was hoping that the SO community could help me out. The pattern is pretty simple, and consists of two parts: A singleton factory, which creates objects based on the arguments passed to the factory method. Objects created by the factory. So far this is just a standard "singleton" pattern or "factory pattern". The issue that I'm asking about, however, is that the singleton factory in this case maintains a set of references to every object that it ever creates, held within a dictionary. These references can sometimes be strong references and sometimes weak references, but it can always reference any object that it has ever created. When receiving a request for a "new" object, the factory first searches the dictionary to see if an object with the required arguments already exits. If it does, it returns that object, if not, it returns a new object and also stores a reference to the new object within the dictionary. This pattern prevents having duplicative objects representing the same underlying "thing". This is useful where the created objects are relatively expensive. It can also be useful where these objects perform event handling or messaging - having one object per item being represented can prevent multiple messages/events for a single underlying source. There are probably other reasons to use this pattern, but this is where I've found this useful. My question is: what to call this? In a sense, each object is a singleton, at least with respect to the data it contains. Each is unique. But there are multiple instances of this class, however, so it's not at all a true singleton. In my own personal terminology, I tend to call the factory method a "global singleton". I then call the created objects "local singletons". I sometimes also say that the created objects have "reference equality", meaning that if two variables reference the same data (the same underlying item) then the reference they each hold must be to the same exact object, hence "reference equality". But these are my own invented terms, and I am not sure that they are good ones. Is there standard terminology for this concept? And if not, could some naming suggestions be made? Thanks in advance...

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  • thread management in nbody code of cuda-sdk

    - by xnov
    When I read the nbody code in Cuda-SDK, I went through some lines in the code and I found that it is a little bit different than their paper in GPUGems3 "Fast N-Body Simulation with CUDA". My questions are: First, why the blockIdx.x is still involved in loading memory from global to share memory as written in the following code? for (int tile = blockIdx.y; tile < numTiles + blockIdx.y; tile++) { sharedPos[threadIdx.x+blockDim.x*threadIdx.y] = multithreadBodies ? positions[WRAP(blockIdx.x + q * tile + threadIdx.y, gridDim.x) * p + threadIdx.x] : //this line positions[WRAP(blockIdx.x + tile, gridDim.x) * p + threadIdx.x]; //this line __syncthreads(); // This is the "tile_calculation" function from the GPUG3 article. acc = gravitation(bodyPos, acc); __syncthreads(); } isn't it supposed to be like this according to paper? I wonder why sharedPos[threadIdx.x+blockDim.x*threadIdx.y] = multithreadBodies ? positions[WRAP(q * tile + threadIdx.y, gridDim.x) * p + threadIdx.x] : positions[WRAP(tile, gridDim.x) * p + threadIdx.x]; Second, in the multiple threads per body why the threadIdx.x is still involved? Isn't it supposed to be a fix value or not involving at all because the sum only due to threadIdx.y if (multithreadBodies) { SX_SUM(threadIdx.x, threadIdx.y).x = acc.x; //this line SX_SUM(threadIdx.x, threadIdx.y).y = acc.y; //this line SX_SUM(threadIdx.x, threadIdx.y).z = acc.z; //this line __syncthreads(); // Save the result in global memory for the integration step if (threadIdx.y == 0) { for (int i = 1; i < blockDim.y; i++) { acc.x += SX_SUM(threadIdx.x,i).x; //this line acc.y += SX_SUM(threadIdx.x,i).y; //this line acc.z += SX_SUM(threadIdx.x,i).z; //this line } } } Can anyone explain this to me? Is it some kind of optimization for faster code?

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  • Use HTTP PUT to create new cache (ehCache) running on the same Tomcat?

    - by socal_javaguy
    I am trying to send a HTTP PUT (in order to create a new cache and populate it with my generated JSON) to ehCache using my webservice which is on the same local tomcat instance. Am new to RESTful Web Services and am using JDK 1.6, Tomcat 7, ehCache, and JSON. I have my POJOs defined like this: Person POJO: import javax.xml.bind.annotation.XmlRootElement; @XmlRootElement public class Person { private String firstName; private String lastName; private List<House> houses; // Getters & Setters } House POJO: import javax.xml.bind.annotation.XmlRootElement; @XmlRootElement public class House { private String address; private String city; private String state; // Getters & Setters } Using a PersonUtil class, I hardcoded the POJOs as follows: public class PersonUtil { public static Person getPerson() { Person person = new Person(); person.setFirstName("John"); person.setLastName("Doe"); List<House> houses = new ArrayList<House>(); House house = new House(); house.setAddress("1234 Elm Street"); house.setCity("Anytown"); house.setState("Maine"); houses.add(house); person.setHouses(houses); return person; } } Am able to create a JSON response per a GET request: @Path("") public class MyWebService{ @GET @Produces(MediaType.APPLICATION_JSON) public Person getPerson() { return PersonUtil.getPerson(); } } When deploying the war to tomcat and pointing the browser to http://localhost:8080/personservice/ Generated JSON: { "firstName" : "John", "lastName" : "Doe", "houses": [ { "address" : "1234 Elmstreet", "city" : "Anytown", "state" : "Maine" } ] } So far, so good, however, I have a different app which is running on the same tomcat instance (and has support for REST): http://localhost:8080/ehcache/rest/ While tomcat is running, I can issue a PUT like this: echo "Hello World" | curl -S -T - http://localhost:8080/ehcache/rest/hello/1 When I "GET" it like this: curl http://localhost:8080/ehcache/rest/hello/1 Will yield: Hello World What I need to do is create a POST which will put my entire Person generated JSON and create a new cache: http://localhost:8080/ehcache/rest/person And when I do a "GET" on this previous URL, it should look like this: { "firstName" : "John", "lastName" : "Doe", "houses": [ { "address" : "1234 Elmstreet", "city" : "Anytown", "state" : "Maine" } ] } So, far, this is what my PUT looks like: @PUT @Path("/ehcache/rest/person") @Produces(MediaType.APPLICATION_JSON) @Consumes(MediaType.APPLICATION_JSON) public Response createCache() { ResponseBuilder response = Response.ok(PersonUtil.getPerson(), MediaType.APPLICATION_JSON); return response.build(); } Question(s): (1) Is this the correct way to write the PUT? (2) What should I write inside the createCache() method to have it PUT my generated JSON into: http://localhost:8080/ehcache/rest/person (3) What would the command line CURL comment look like to use the PUT? Thanks for taking the time to read this...

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