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• DirectShow Filter I wrote dies after 10-24 seconds in Skype video call

  - by Robert Oschler

  I've written a DirectShow push filter for use with Skype using Delphi Pro 6 and the DSPACK DirectShow library. In preview mode, when you test a video input device in the Skype client Video Settings window, my filter works flawlessly. I can leave it up and running for many minutes without an error. However when I start a video call after 10 to 24 seconds, never longer, the video feed freezes. The call continues fine with the call duration counter clicking away the seconds, but the video feed is dead, stuck on whatever frame the freeze happened (although after a long while it turns black which I believe means Skype has given up on the filter). I tried attaching to the process from my debugger with a breakpoint literally set on every method call and none of them are hit once the freeze takes place. It's as if the thread that makes the DirectShow FillBuffer() call to my filter on behalf of Skype is dead or has been shutdown. I can't trace my filter in the debugger because during a Skype call I get weird int 1 and int 3 debugger hard interrupt calls when a Skype video call is in progress. This behavior happens even with my standard web cam input device selected and my DirectShow filter completely unregistered as a ActiveX server. I suspect it might be some "anti-debugging" code since it doesn't happen in video input preview mode. Either way, that is why I had to attach to the process after the fact to see if my FillBuffer() called was still being called and instead discovered that appears to be dead. Note, my plain vanilla USB web cam's DirectShow filter does not exhibit the freezing behavior and works fine for many minutes. There's something about my filter that Skype doesn't like. I've tried Sleep() statements of varying intervals, no Sleep statements, doing virtually nothing in the FillBuffer() call. Nothing helps. If anyone has any ideas on what might be the culprit here, I'd like to know. Thanks, Robert

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• Loading one page inside another

  - by Robin I Knight

  User 'Citizen' provided an answer to the iframe situation with the ajax script from dynamic drive. As I predicted it although it loads one page inside another it does not work with the calculation scripts, collapsible panels, validation form. All of it simply not working. I have set up a test page that has the exact same HEAD section as the page that is loaded inside it, so it is not s problem of script location. Take a look and see if you can tell me what is going on. Baring in mind this is just a test page. On the test page the entire page is loaded from another and as you can see all the collapsible panels are open, all calculations except the duration are not working because another file that is loaded by ajax on the original page is not loading in this one, the accordion menu us not working and the validation form is not validating. It is as if all script triggers have been removed and left behind but like I said the HEAD section of the parent page contains all of the scripts as well. Any ideas http://www.divethegap.com/scuba-diving-programmes-dive-the-gap/programme-pages/dahab-divemaster/test.php

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• Synchronizing ASP.NET MVC action methods with ReaderWriterLockSlim

  - by James D

  Any obvious issues/problems/gotchas with synchronizing access (in an ASP.NET MVC blogging engine) to a shared object model (NHibernate, but it could be anything) at the Controller/Action level via ReaderWriterLockSlim? (Assume the object model is very large and expensive to build per-request, so we need to share it among requests.) Here's how a typical "Read Post" action would look. Enter the read lock, do some work, exit the read lock. public ActionResult ReadPost(int id) { // ReaderWriterLockSlim allows multiple concurrent writes; this method // only blocks in the unlikely event that some other client is currently // writing to the model, which would only happen if a comment were being // submitted or a new post were being saved. _lock.EnterReadLock(); try { // Access the model, fetch the post with specificied id // Pseudocode, etc. Post p = TheObjectModel.GetPostByID(id); ActionResult ar = View(p); return ar; } finally { // Under all code paths, we must release the read lock _lock.ExitReadLock(); } } Meanwhile, if a user submits a comment or an author authors a new post, they're going to need write access to the model, which is done roughly like so: [AcceptVerbs(HttpVerbs.Post)] public ActionResult SaveComment(/* some posted data */) { // try/finally omitted for brevity _lock.EnterWriteLock(); // Save the comment to the DB, update the model to include the comment, etc. _lock.ExitWriteLock(); } Of course, this could also be done by tagging those action methods with some sort of "synchronized" attribute... but however you do it, my question is is this a bad idea? ps. ReaderWriterLockSlim is optimized for multiple concurrent reads, and only blocks if the write lock is held. Since writes are so infrequent (1000s or 10,000s or 100,000s of reads for every 1 write), and since they're of such a short duration, the effect is that the model is synchronized , and almost nobody ever locks, and if they do, it's not for very long.

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• How to do recurring with bank account payment mode using authorized.net

  - by Salil

  Hi All, I am using recurring facility of authorized.net using active_merchant plugin in rails. there are two payment method for this 1] Credit Card 2] Bank Account I successfully done it using Credit Card For Recurring i need my Test Mode off. Also my E-check, credit card processing, and subscriptions are all enabled. But i am not able to subscribed using Bank Account Following is my code ActiveMerchant::Billing::Base.mode = :developer #i found follwing test bank account on net account = ActiveMerchant::Billing::Check.new(:account_holder_type=>"personal",:account_number=>"123123123", :account_type => "savings", :name=>"name", :routing_number=>"244183602") if account.valid? #this comes true gateway = ActiveMerchant::Billing::AuthorizeNetGateway.new(:login => 'Mylogin', :password => 'Mypassword') response = gateway.recurring( amount, nil, {:interval =>{:length=>@length, :unit =>:months}, :duration =>{:start_date=>'2010-04-24', :occurrences=>1}, :billing_address=>{:first_name=>'dinesh', :last_name=>'singh'}, :bank_account=>{:account_holder_type=>"personal",:account_number=>"123123123", :account_type => "savings", :name_of_account=>"name", :routing_number=>"244183602"} }) if response.success? #this comes false else puts response.message ####>> ERROR render :action=>"account_payment" end I get Follwing ERROR when i debug for response.message "The test transaction was not successful. (128) This transaction cannot be processed." Am i doing anything wrong i search for the another Test Bank Account Data but i didn't find it. Please help I badly need it. Thanks in Advance. Regards, Salil Gaikwad

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• Stall in animation

  - by tech74

  Hi , I am using this code from this site http://ramin.firoozye.com/2009/09/29/semi-modal-transparent-dialogs-on-the-iphone/ to show a modal view and remove it. It displays fine ie drops in from the top but when being removed it stalls just on its way out just for a fraction of second but its noticeable how do i get rid of the stall. The view i am showing is view of a viewcontroller which is a memeber of the parent viewcontroller so i call methods like this to display [self showModal:self.modalController.view] to hide [self hideModal:self.modalController.view]; (void) showModal:(UIView*) modalView { UIWindow* mainWindow = (((SessionTalkAppDelegate*) [UIApplication sharedApplication].delegate).window); //CGPoint middleCenter = modalView.center; CGPoint middleCenter = CGPointMake(160, 205); CGSize offSize = [UIScreen mainScreen].bounds.size; //CGPoint offScreenCenter = CGPointMake(offSize.width / 2.0, offSize.height * 1.5); CGPoint offScreenCenter = CGPointMake(offSize.width / 2.0, -100); // start from top modalView.center = offScreenCenter; // we start off-screen [mainWindow addSubview:modalView]; // Show it with a transition effect [UIView beginAnimations:nil context:nil]; [UIView setAnimationDuration:0.3]; // animation duration in seconds [UIView setAnimationCurve:UIViewAnimationCurveEaseInOut]; modalView.center = middleCenter; [UIView commitAnimations]; } // Use this to slide the semi-modal back up. - (void) hideModal:(UIView*) modalView { CGSize offSize = [UIScreen mainScreen].bounds.size; //CGPoint offScreenCenter = CGPointMake(offSize.width / 2.0, offSize.height * 1.5); CGPoint offScreenCenter = CGPointMake(offSize.width / 2.0, -100); [UIView beginAnimations:nil context:modalView]; [UIView setAnimationDuration:0.3]; [UIView setAnimationCurve:UIViewAnimationCurveEaseInOut]; [UIView setAnimationDelegate:self]; [UIView setAnimationDidStopSelector:@selector(hideModalEnded:finished:context:)]; modalView.center = offScreenCenter; [UIView commitAnimations]; } (void) hideModalEnded:(NSString *)animationID finished:(NSNumber *)finished context:(void )context { UIView modalView = (UIView *)context; [modalView removeFromSuperview]; //[modalView release]; }

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• Jquery .hide problem

  - by Sergio

  I'm using jquery to show animated MSN like window at the bottom of the page. Jquery code: $(document).ready(function() { $(" .inner").stop().animate({height:'142px'},{queue:false, duration:600}); $(' .close').click(function(event) { $(' .inner').hide(); }); $(' .inner').click(function() { location.replace("somepage.php"); }); }); The CSS: .close {height:14px; z-index:100; position: absolute; margin-left:174px; margin-top:12px; border:1px solid #F00;} .inner {position:absolute;bottom:0;width:201px;height:117px;right: 0; margin-right:10px; float:left; z-index:-1; display:block; cursor:pointer;} The problem that I have with this code is that the close button (showing at the right top corner of .inner DIV) can't fire Jquery .hide function. Why?

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• New CATransform3DMakeRotation deletes old transformation?!

  - by david

  I added a CATransform3DMakeRotation to a layer. When I add another one it deletes the old one? The first one: [UIView beginAnimations:@"rotaty" context:nil]; [UIView setAnimationDuration:0.5]; [UIView setAnimationDelegate:self]; CGAffineTransform transform = CGAffineTransformMakeRotation(-3.14); kuvert.transform = CGAffineTransformRotate(transform, DegreesToRadians(134)); kuvert.center = CGPointMake(kuvert.center.x-70, kuvert.center.y+100); [UIView commitAnimations]; and the second one: CABasicAnimation *topAnim = [CABasicAnimation animationWithKeyPath:@"transform"]; topAnim.duration=1; topAnim.repeatCount=0; topAnim.fromValue = [NSValue valueWithCATransform3D:CATransform3DMakeRotation(0.0, 0, 0, 0)]; float f = DegreesToRadians(180); // -M_PI/1; topAnim.toValue=[NSValue valueWithCATransform3D:CATransform3DMakeRotation(f, 0,1, 0)]; topAnim.delegate = self; topAnim.removedOnCompletion = NO; topAnim.fillMode = kCAFillModeBoth; [topAnim setValue:@"flippy" forKey:@"AnimationName"]; [[KuvertLasche layer] addAnimation:topAnim forKey:@"flippy"]; The second one resets the view and applies itself after that. How do I fix this??

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• Avoiding shutdown hook

  - by meryl

  Through the following code I can play and cut and audio file. Is there any other way to avoid using a shutdown hook? The problem is that whenever I push the cut button , the file doesn't get saved until I close the application thanks ...................... void play_cut() { try { // First, we get the format of the input file final AudioFileFormat.Type fileType = AudioSystem.getAudioFileFormat(inputAudio).getType(); // Then, we get a clip for playing the audio. c = AudioSystem.getClip(); // We get a stream for playing the input file. AudioInputStream ais = AudioSystem.getAudioInputStream(inputAudio); // We use the clip to open (but not start) the input stream c.open(ais); // We get the format of the audio codec (not the file format we got above) final AudioFormat audioFormat = ais.getFormat(); // We add a shutdown hook, an anonymous inner class. Runtime.getRuntime().addShutdownHook(new Thread() { public void run() { // We're now in the hook, which means the program is shutting down. // You would need to use better exception handling in a production application. try { // Stop the audio clip. c.stop(); // Create a new input stream, with the duration set to the frame count we reached. Note that we use the previously determined audio format AudioInputStream startStream = new AudioInputStream(new FileInputStream(inputAudio), audioFormat, c.getLongFramePosition()); // Write it out to the output file, using the same file type. AudioSystem.write(startStream, fileType, outputAudio); } catch(IOException e) { e.printStackTrace(); } } }); // After setting up the hook, we start the clip. c.start(); } catch (UnsupportedAudioFileException e) { e.printStackTrace(); } catch (IOException e) { e.printStackTrace(); } catch (LineUnavailableException e) { e.printStackTrace(); } }// end play_cut ......................

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• FFMpeg Error av_interleaved_write_frame():

  - by rajaneesh

  this my code . after running php code FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib --mandir=/usr/share/man --incdir=/usr/include --enable-libamr-nb --enable-libamr-wb --enable-libdirac --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-x11grab libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 6 2009 19:05:03, gcc: 4.1.2 20080704 (Red Hat 4.1.2-46) Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) - 25.00 (25/1) Input #0, flv, from 'demo.flv': Duration: 00:00:30.83, start: 0.000000, bitrate: 546 kb/s Stream #0.0: Video: h264, yuv420p, 640x360 [PAR 1:1 DAR 16:9], 546 kb/s, 25 tbr, 1k tbn, 50 tbc Stream #0.1: Audio: aac, 44100 Hz, stereo, s16 Output #0, image2, to 'demo.jpg': Stream #0.0: Video: mjpeg, yuvj420p, 640x360 [PAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 1 tbc Stream mapping: Stream #0.0 - #0.0 Press [q] to stop encoding av_interleaved_write_frame(): I/O error occurred Usually that means that input file is truncated and/or corrupted.

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• Forcing file redirection on x64 for a 32-bit application

  - by Paul Alexander

  The silent redirection of 64-bit system files to their 32-bit equivalents can be turned off and reverted with Wow64DisableWow64FsRedirection and Wow64RevertWow64FsRedirection. We use this for certain file identity checks in our application. The problem is that in performing some of theses tasks, we might call a framework or Windows API in a DLL that has not yet been loaded. If redirection is enabled at that time, the wrong version of the dll may be loaded resulting in a XXX is not a valid Win32 application error. I've identified the few API calls in question and what I'd like to do force the redirection on for the duration of that call then revert it back - just the opposite of the provided Win32 APIs. Unfortunately these calls do not provide any sort of WOW64 compatibility flag like some of the registry methods do. The obvious alternative is to use Wow64EnableWow64FsRedirection, pass TRUE for Wow64FsEanbledRedirection. However there are a variety of warnings about the use of this method and a note that it is not compatible with Disable/Revert combo methods that have replaced it. Is there a safe way to force redirection on for a give Win32 call? The docs state the redirection is thread specific so I've considered spinning up a new thread for the specific call with appropriate locks and waits, but I was hoping for a simpler solution.

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• Progressive MP4 video issues in Flash- Video stops rendering

  - by Conor

  I'm currently working on a flash project that has an intro video that plays before heading into the main app. This video is an H.264 .mp4, 1550x540, and around 10MB. The problem thats currently driving me insane is that when I test it, occasionally the video will begin playing, and then suddenly stop rendering the video frames, leaving the audio playing in the background with nothing on screen. Once the file is played through fully (based on listening to the audio), my playback complete event fires like it should, but I can't find any info of people having similar issues. Attached is a trace of the .mp4 metadata in case that helps. videoframerate : 24 audiochannels : 2 audiocodecid : mp4a audiosamplerate : 48000 trackinfo: 0: length : 608000 timescale : 24000 language : eng sampledescription: 0: sampletype : avc1 1: length : 1218560 timescale : 48000 language : eng sampledescription: 0: sampletype : mp4a duration : 25.386666666666667 width : 1540 videocodecid : avc1 seekpoints: 0: time : 0 offset : 13964 1: time : 0.333 offset : 16893 2: time : 0.667 offset : 34212 ... 73: time : 24.333 offset : 9770329 74: time : 24.667 offset : 9845709 75: time : 25 offset : 9895215 moovposition : 32 height : 540 avcprofile : 77 avclevel : 51 aacaot : 2 This has been driving me absolutely insane... any help would be much appreciated!

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• Python script, runs well, but not perfectly, debugging help.

  - by S1syphus

  What it does (sort of)... or is meant to, the script reads from a csv file that contains information on sound files and create a play list exactly 60 minutes long. An example csv, contains: their title, duration (in seconds), minium total time to be played (in minutes) An example is: Soundfoo,120,10 Soundbar,30,6 Sounddev,60,20 Soundrandom,15,8 The script works out the minimum instances of plays, take 'Soundfoo' for example, the length of each sample is 120 seconds and the minimum time to be played is 10 minutes, so basic maths 10*60/120 gives the number of instances the song is to be played, in this case 5. It is meant to take minimum number of instances and spread out equally from each other; so there will never be a period where for example Soundbar is played twice in a row. Then if the minium instances of each song has been used, and there is still time with in the 60 min, how is it possible to tell it to go back and fill the time by selecting each sound and including it till the 60 min is filled while remaining sparsely populated. Heres the issue(s)! The script fails to calculate the actual time require to play all the sounds in a file and the total time of the playlist, the thing is tho it doesn't get it wrong all the time maybe 3/5 times, even if I run it on the same csv file it will give me different answers. Here is the file I shall run the script on e for sake of ease to see the issue: Sound1,60,10 Sound2,60,10 Sound3,60,10 Sound4,60,10 Sound5,60,10 Sound6,60,10 I'll do it three times and post the results: 1 Required playtime in minutes: 60 Actual time in minutes to play all required ads: 62 Total playtime in minutes: 62.0 2 Required playtime in minutes: 60 Actual time in minutes to play all required ads: 71 Total playtime in minutes: 71.0 3 Required playtime in minutes: 60 Actual time in minutes to play all required ads: 60 Total playtime in minutes: 60.0 Relevant Code: pastebin.com/demkBXk6 And finally... in context: http://pastebin.com/demkBXk6 If you made it down to here, thanks for staying and reading, kudos.

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• Date difference in Javascript (ignoring time of day)

  - by Alan

  I'm writing an equipment rental application where clients are charged a fee for renting equipment based on the duration (in days) of the rental. So, basically, (daily fee * number of days) = total charge. For instant feedback on the client side, I'm trying to use Javascript to figure out the difference in two calendar dates. I've searched around, but nothing I've found is quite what I'm looking for. Most solutions I've seen are of the form: function dateDiff1(startDate, endDate) { return ((endDate.getTime() - startDate.getTime()) / 1000*60*60*24); } My problem is that equipment can be checked out and returned at any time of day during those two dates with no additional charge. The above code is calculating the number of 24 hour periods between the two dates, when I'm really interested in the number of calendar days. For example, if someone checked out equipment at 6am on July 6th and returned it at 10pm on July 7th, the above code would calculate that more than one 24 hour period had passed, and would return 2. The desired result is 1, since only one calendar date has elapsed (i.e. the 6th to the 7th). The closest solution I've found is this function: function dateDiff2(startDate, endDate) { return endDate.getDate() - startDate.getDate(); } which does exactly what I want, as long as the two dates are within the same month. However, since getDate() only returns the day of month (i.e. 1-31), it doesn't work when the dates span multiple months (e.g. July 31 to August 1 is 1 day, but the above calcuates 1 - 31, or -29). On the backend, in PHP, I'm using gregoriantojd(), which seems to work just fine (see this post for an example). I just can't find an equivalent solution in Javascript. Anyone have any ideas?

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• Having trouble animating Line in D3.js using and array of objects as data

  - by user1731245

  I can't seem to get an animated transition between line graphs when I pass in a new set of data. I am using an array of objects as data like this: [{ clicks: 40 installs: 10 time: "1349474400000" },{ clicks: 61 installs: 3 time: "1349478000000" }]; I am using this code to setup my ranges / axis's var xRange = d3.time.scale().range([0, w]), yRange = d3.scale.linear().range([h , 0]), xAxis = d3.svg.axis().scale(xRange).tickSize(-h).ticks(6).tickSubdivide(false), yAxis = d3.svg.axis().scale(yRange).ticks(5).tickSize(-w).orient("left"); var clicksLine = d3.svg.line() .interpolate("cardinal") .x(function(d){return xRange(d.time)}) .y(function(d){return yRange(d.clicks)}); var clickPath; function drawGraphs(data) { clickPath = svg.append("g") .append("path") .data([data]) .attr("class", "clicks") .attr("d", clicksLine); } function updateGraphs(data) { svg.select('path.clicks') .data([data]) .attr("d", clicksLine) .transition() .duration(500) .ease("linear") } I have tried just about everything to be able to pass in new data and see an animation between graph's. Not sure what I am missing? does it have something to do with using an array of objects instead of just a flat array of numbers as data?

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• unix at command pass variable to shell script?

  - by Andrew

  Hi, I'm trying to setup a simple timer that gets started from a Rails Application. This timer should wait out its duration and then start a shell script that will start up ./script/runner and complete the initial request. I need script/runner because I need access to ActiveRecord. Here's my test lines in Rails output = `at #{(Time.now + 60).strftime("%H:%M")} < #{Rails.root}/lib/parking_timer.sh STRING_VARIABLE` return render :text => output Then my parking_timer.sh looks like this #!/bin/sh ~/PATH_TO_APP/script/runner -e development ~/PATH_TO_APP/lib/ParkingTimer.rb $1 echo "All Done" Finally, ParkingTimer.rb reads the passed variable with ARGV.each do|a| puts "Argument: #{a}" end The problem is that the Unix command "at" doesn't seem to like variables and only wants to deal with filenames. I either get one of two errors depending on how I position "s If I put quotes around the right hand side like so ... "~/PATH_TO_APP/lib/parking_timer.sh STRING_VARIABLE" I get, -bash: ~/PATH_TO_APP/lib/parking_timer.sh STRING_VARIABLE: No such file or directory I I leave the quotes out, I get, at: garbled time This is all happening on a Mac OS 10.6 box running Rails 2.3 & Ruby 1.8.6 I've already messed around w/ BackgrounDrb, and decided its a total PITA. I need to be able to cancel the job at any time before it is due.

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• Planning management slots/sessions

  - by Glide

  I have a planning structure on two tables to store available slots by day, and sessions. A slot is defined by a range of time in the day. CREATE TABLE slot ( `id` int(11) NOT NULL AUTO_INCREMENT , `date` date , `start` time , `end` time ); Sessions can't overlap themselves and must be wrapped in a slot. CREATE TABLE session ( `id` int(11) NOT NULL AUTO_INCREMENT , `date` date , `start` time , `end` time ); I need to generate a list of available blocks of time of a certain duration, in order to create sessions. Example: INSERT INTO slot (date, start, end) VALUES ("2010-01-01", "10:00", "19:00") , ("2010-01-02", "10:00", "15:00") , ("2010-01-02", "16:00", "20:30") ; INSERT INTO slot (date, start, end) VALUES ("2010-01-01", "10:00", "19:00") , ("2010-01-02", "10:00", "15:00") , ("2010-01-02", "16:00", "20:30") ; 2010-01-01 <##><####> <- Sessions ------------------------------------ <- Slots 10 11 12 13 14 15 16 17 18 19 20 2010-01-02 <##########> <########> <- Sessions -------------------- ------------------ <- Slots 10 11 12 13 14 15 16 17 18 19 20 I need to know which spaces of 1 hour I can use: +------------+-------+-------+ | date | start | end | +------------+-------+-------+ | 2010-01-01 | 13:00 | 14:00 | | 2010-01-01 | 14:00 | 15:00 | | 2010-01-01 | 15:00 | 16:00 | | 2010-01-01 | 16:00 | 17:00 | | 2010-01-01 | 17:00 | 18:00 | | 2010-01-01 | 18:00 | 19:00 | | 2010-01-02 | 10:00 | 11:00 | | 2010-01-02 | 11:00 | 12:00 | | 2010-01-02 | 16:00 | 17:00 | +------------+-------+-------+

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• Optimize GROUP BY&ORDER BY query

  - by Jan Hancic

  I have a web page where users upload&watch videos. Last week I asked what is the best way to track video views so that I could display the most viewed videos this week (videos from all dates). Now I need some help optimizing a query with which I get the videos from the database. The relevant tables are this: video (~239371 rows) VID(int), UID(int), title(varchar), status(enum), type(varchar), is_duplicate(enum), is_adult(enum), channel_id(tinyint) signup (~115440 rows) UID(int), username(varchar) videos_views (~359202 rows after 6 days of collecting data, so this table will grow rapidly) videos_id(int), views_date(date), num_of_views(int) The table video holds the videos, signup hodls users and videos_views holds data about video views (each video can have one row per day in that table). I have this query that does the trick, but takes ~10s to execute, and I imagine this will only get worse over time as the videos_views table grows in size. SELECT v.VID, v.title, v.vkey, v.duration, v.addtime, v.UID, v.viewnumber, v.com_num, v.rate, v.THB, s.username, SUM(vvt.num_of_views) AS tmp_num FROM video v LEFT JOIN videos_views vvt ON v.VID = vvt.videos_id LEFT JOIN signup s on v.UID = s.UID WHERE v.status = 'Converted' AND v.type = 'public' AND v.is_duplicate = '0' AND v.is_adult = '0' AND v.channel_id <> 10 AND vvt.views_date >= '2001-05-11' GROUP BY vvt.videos_id ORDER BY tmp_num DESC LIMIT 8 And here is a screenshot of the EXPLAIN result: So, how can I optimize this?

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• Python pixel manipulation library

  - by silinter

  So I'm going through the beginning stages of producing a game in Python, and I'm looking for a library that is able to manipulate pixels and blit them relatively fast. My first thought was pygame, as it deals in pure 2D surfaces, but it only allows pixel access through pygame.get_at(), pygame.set_at() and pygame.get_buffer(), all of which lock the surface each time they're called, making them slow to use. I can also use the PixelArray and surfarray classes, but they are locked for the duration of their lifetimes, and the only way to blit them to a surface is to either copy the pixels to a new surface, or use surfarray.blit_array, which requires creating a subsurface of the screen and blitting it to that, if the array is smaller than the screen (if it's bigger I can just use a slice of the array, which is no problem). I don't have much experience with PyOpenGL or Pyglet, but I'm wondering if there is a faster library for doing pixel manipulation in, or if there is a faster method, in Pygame, for doing pixel manupilation. I did some work with SDL and OpenGL in C, and I do like the idea of adding vertex/fragment shaders to my program. My program will chiefly be dealing in loading images and writing/reading to/from surfaces.

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• Segmentation fault while feeding in an mpeg file through ffmpeg

  - by angel6

  Hi, I've set up FFserver as the streaming server. I'm trying to feed in an mpeg file. But it comes up with a segmentation fault. Does anyone know how to fix this? The following is the command-line output I get $ ./ffmpeg -i test1.mpg http://localhost:8090/feed1.ffm FFmpeg version SVN-r22945, Copyright (c) 2000-2010 the FFmpeg developers built on Apr 22 2010 19:18:45 with gcc 4.4.1 configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-pthreads --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libx264 --enable-libxvid --enable-x11grab libavutil 50.14. 0 / 50.14. 0 libavcodec 52.66. 0 / 52.66. 0 libavformat 52.61. 0 / 52.61. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpeg @ 0xab0c420]max_analyze_duration reached Input #0, mpeg, from 'test1.mpg': Duration: 00:00:20.96, start: 0.768300, bitrate: 269 kb/s Stream #0.0[0x1e0]: Video: mpeg1video, yuv420p, 160x120 [PAR 1:1 DAR 4:3], 104857 kb/s, 30 fps, 30 tbr, 90k tbn, 30 tbc Stream #0.1[0x1c0]: Audio: mp2, 32000 Hz, 2 channels, s16, 64 kb/s Output #0, ffm, to 'http://localhost:8090/feed1.ffm': Metadata: encoder : Lavf52.61.0 Stream #0.0: Audio: mp2, 22050 Hz, 1 channels, s16, 48 kb/s Stream #0.1: Video: mpeg1video, yuv420p, 160x128, q=2-31, 40 kb/s, 1000k tbn, 50 tbc Stream #0.2: Audio: libmp3lame, 22050 Hz, 1 channels, s16, 64 kb/s Stream #0.3: Video: msmpeg4, yuv420p, 352x240, q=2-31, 256 kb/s, 1000k tbn, 15 tbc Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Stream #0.1 -> #0.2 Stream #0.0 -> #0.3 Press [q] to stop encoding Segmentation fault

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• Simulating O_NOFOLLOW (2): Is this other approach safe?

  - by Daniel Trebbien

  As a follow-up question to this one, I thought of another approach which builds off of @caf's answer for the case where I want to append to file name and create it if it does not exist. Here is what I came up with: Create a temporary directory with mode 0700 in a system temporary directory on the same filesystem as file name. Create an empty, temporary, regular file (temp_name) in the temporary directory (only serves as placeholder). Open file name for reading only, just to create it if it does not exist. The OS may follow name if it is a symbolic link; I don't care at this point. Make a hard link to name at temp_name (overwriting the placeholder file). If the link call fails, then exit. (Maybe someone has come along and removed the file at name, who knows?) Use lstat on temp_name (now a hard link). If S_ISLNK(lst.st_mode), then exit. open temp_name for writing, append (O_WRONLY | O_APPEND). Write everything out. Close the file descriptor. unlink the hard link. Remove the temporary directory. (All of this, by the way, is for an open source project that I am working on. You can view the source of my implementation of this approach here.) Is this procedure safe against symbolic link attacks? For example, is it possible for a malicious process to ensure that the inode for name represents a regular file for the duration of the lstat check, then make the inode a symbolic link with the temp_name hard link now pointing to the new, symbolic link? I am assuming that a malicious process cannot affect temp_name.

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• c# regex split and extract multiple parts from a string

  - by nLL

  Hi, I am trying to extract some parts of the "Video:" line from below text. Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (300 00/1) - 14.93 (1000/67) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\a.3gp': Metadata: major_brand : 3gp5 minor_version : 0 compatible_brands: 3gp5isom Duration: 00:00:45.82, start: 0.000000, bitrate: 357 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb /s, 14.93 fps, 14.93 tbr, 90k tbn, 30k tbc Stream #0.1(und): Audio: aac, 16000 Hz, mono, s16, 11 kb/s Stream #0.2(und): Data: mp4s / 0x7334706D, 0 kb/s Stream #0.3(und): Data: mp4s / 0x7334706D, 0 kb/s* This is an output from ffmpeg command line where i can get Video: part with private string ExtractVideoFormat(string rawInfo) { string v = string.Empty; Regex re = new Regex("[V|v]ideo:.*", RegexOptions.Compiled); Match m = re.Match(rawInfo); if (m.Success) { v = m.Value; } return v; } and result is mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb What i am trying to do is to somehow split that line and get mpeg4 yuv420p 352x276 [PAR 1:1 DAR 88:69] 344 kb assigned to diffrent string objects instead of single

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• WPF abnormal CPU usage for animation

  - by 0xDEAD BEEF

  HI! I am developing WPF application and client reports extreamly high CPU usage (90%) (whereas i am unable to repeat that behavior). I have traced bootleneck down to these lines. It is simple glowing animation for small single led control (blinking led). What could be reason for this simple annimation taking up SO huge CPU resources? <Trigger Property="State"> <Trigger.Value> <local:BlinkingLedStatus>Blinking</local:BlinkingLedStatus> </Trigger.Value> <Trigger.EnterActions> <BeginStoryboard Name="beginStoryBoard"> <Storyboard> <DoubleAnimation Storyboard.TargetName="glow" Storyboard.TargetProperty="Opacity" AutoReverse="True" From="0.0" To="1.0" Duration="0:0:0.5" RepeatBehavior="Forever"/> </Storyboard> </BeginStoryboard> </Trigger.EnterActions> <Trigger.ExitActions> <StopStoryboard BeginStoryboardName="beginStoryBoard"/> </Trigger.ExitActions> </Trigger>

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• Is there a disassembler + debugger for java (ala OllyDbg / SoftICE for assembler)?

  - by Ran Biron

  Is there a utility similar to OllyDbg / SoftICE for java? I.e. execute class (from jar / with class path) and, without source code, show the disassembly of the intermediate code with ability to step through / step over / search for references / edit specific intermediate code in memory / apply edit to file... If not, is it even possible to write something like this (assuming we're willing to live without hotspot for the debug duration)? Edit: I'm not talking about JAD or JD or Cavaj. These are fine decompilers, but I don't want a decompiler for several reasons, most notable is that their output is incorrect (at best, sometimes just plain wrong). I'm not looking for a magical "compiled bytes to java code" - I want to see the actual bytes that are about to be executed. Also, I'd like the ability to change those bytes (just like in an assembly debugger) and, hopefully, write the changed part back to the class file. Edit2: I know javap exists - but it does only one way (and without any sort of analysis). Example (code taken from the vmspec documentation): From java code, we use "javac" to compile this: void setIt(int value) { i = value; } int getIt() { return i; } to a java .class file. Using javap -c I can get this output: Method void setIt(int) 0 aload_0 1 iload_1 2 putfield #4 5 return Method int getIt() 0 aload_0 1 getfield #4 4 ireturn This is OK for the disassembly part (not really good without analysis - "field #4 is Example.i"), but I can't find the two other "tools": A debugger that goes over the instructions themselves (with stack, memory dumps, etc), allowing me to examine the actual code and environment. A way to reverse the process - edit the disassembled code and recreate the .class file (with the edited code).

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• Random point on VideoDisplay isn't accurate enough

  - by Mike

  For a schoolassigment me and some buddies of mine are creating an application that is showing many similarities with the C-Mon & Kypski musicvideo on www.oneframeoffame.com. The application is being developed in Flex. We want to get a random point of a clip, let it pause so a user can mimic the pose and make a snapshot out of it. What i managed to do is get a random point of the movie. I did this by getting a random value between 0 and de total duration of the movie. But what i didn't managed to do is let the screen pause on every 24st of a frame. As the movie concist out of 24FPS. It looks like the the random value of the movie that is being requested is being rounded by the movie itself. As example: There appears to be no difference between the frames requested at 2.40 or 2.41. It appears it got something to do with keyframing i've read on the Adobe® Flex™ 3.5 Language Reference. The movie is a FLV file and i use the VideoDisplay object to display the movie. Does someone is familiar with this or knows a solution to my problem? Thanks in advance

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• What is the rationale to non allow overloading of C++ conversions operator with non-member functio

  - by Vicente Botet Escriba

  C++0x has added explicit conversion operators, but they must always be defined as members of the Source class. The same applies to the assignment operator, it must be defined on the Target class. When the Source and Target classes of the needed conversion are independent of each other, neither the Source can define a conversion operator, neither the Target can define a constructor from a Source. Usually we get it by defining a specific function such as Target ConvertToTarget(Source& v); If C++0x allowed to overload conversion operator by non member functions we could for example define the conversion implicitly or explicitly between unrelated types. template < typename To, typename From operator To(const From& val); For example we could specialize the conversion from chrono::time_point to posix_time::ptime as follows template < class Clock, class Duration operator boost::posix_time::ptime( const boost::chrono::time_point& from) { using namespace boost; typedef chrono::time_point time_point_t; typedef chrono::nanoseconds duration_t; typedef duration_t::rep rep_t; rep_t d = chrono::duration_cast( from.time_since_epoch()).count(); rep_t sec = d/1000000000; rep_t nsec = d%1000000000; return posix_time::from_time_t(0)+ posix_time::seconds(static_cast(sec))+ posix_time::nanoseconds(nsec); } And use the conversion as any other conversion. So the question is: What is the rationale to non allow overloading of C++ conversions operator with non-member functions?

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