Search Results

Search found 1357 results on 55 pages for 'mp3'.

Page 50/55 | < Previous Page | 46 47 48 49 50 51 52 53 54 55  | Next Page >

  • Javascript force GC collection? / Forcefully free object?

    - by plash
    I have a js function for playing any given sound using the Audio interface (creating a new instance for every call). This works quite well, until about the 32nd call (sometimes less). This issue is directly related to the release of the Audio instance. I know this because I've allowed time for the GC in Chromium to run and it will allow me to play another 32 or so sounds again. Here's an example of what I'm doing: <html><head> <script language="javascript"> function playSound(url) { snd = new Audio(url); snd.play(); delete snd; snd = null; } </script> </head> <body> <a href="#" onclick="playSound('blah.mp3');">Play sound</a> </body></html> I also have this, which works well for pages that have less than 32 playSound calls: var AudioPlayer = { cache: {}, play: function(url) { if (!AudioPlayer.cache[url]) AudioPlayer.cache[url] = new Audio(url); AudioPlayer.cache[url].play(); } }; But this will not work for what I want to do (dynamically replace a div with other content (from separate files), which have even more sounds on them - 1. memory usage would easily skyrocket, 2. many sounds will never play). I need a way to release the sound immediately. Is it possible to do this? I have found no free/close/unload method for the Audio interface. The pages will be viewed locally, so the constant loading of sounds is not a big factor at all (and most sounds are rather short).

    Read the article

  • Audio output from Silverlight

    - by leecarter
    I'm looking to develop a Silverlight application which will take a stream of data (not an audio stream as such) from a web server. The data stream would then be manipulated to give audio of a certain format (G.711 a-Law for example) which would then be converted into PCM so that additional effects can be applied (such as boosting the volume). I'm OK up to this point. I've got my data, converted the G.711 into PCM but my problem is being able to output this PCM audio to the sound card. I basing a solution on some C# code intended for a .Net application but in Silverlight there is a problem with trying to take a copy of a delegate (function pointer) which will be the topic of a separate question once I've produced a simple code sample. So, the question is... How can I output the PCM audio that I have held in a data structure (currently an array) in my Silverlight to the user? (Please don't say write the byte values to a text box) If it were a MP3 or WMA file I would play it using a MediaElement but I don't want to have to make it into a file as this would put a crimp on applying dynamic effects to the audio. I've seen a few posts from people saying low level audio support is poor/non-existant in Silverlight so I'm open to any suggestions/ideas people may have.

    Read the article

  • Playing video and audio in iPhone not working...

    - by Scott
    So we have buttons linked up to display images/videos/audio on click depending on a check we do earlier. That part works fine. It knows which one to play, however, when we click the buttons for video and audio, nothing happens. The image one works fine. The video and audio are being taken for a URL online, they are not local, but everywhere said this was still possible. Here is a little snippet of the code where we play the two files: if ( [fName hasSuffix:@".png"]) { NSLog(@"PICTURE"); NSURL *url = [NSURL URLWithString: fName]; UIImage *image = [UIImage imageWithData: [NSData dataWithContentsOfURL:url]]; self.view = [[UIView alloc] initWithFrame:[[UIScreen mainScreen] applicationFrame]]; // self.view.backgroundColor = [[UIColor alloc] initWithPatternImage:[UIImage imageNamed:@"MainBG.jpg"]]; [self.view addSubview:[[UIImageView alloc] initWithImage:image]]; } if ( [fName hasSuffix:@".mp4"]) { NSLog(@"VIDEO"); //NSString *path = [[NSBundle mainBundle] pathForResource:fName ofType:@"mp4"]; //NSLog(path); NSURL *url = [NSURL fileURLWithPath:fName]; MPMoviePlayerController *player = [[MPMoviePlayerController alloc] initWithContentURL:url]; [player play]; } if ( [fName hasSuffix:@".mp3"]) { NSLog(@"AUDIO"); NSURL *url = [NSURL fileURLWithPath:fName]; NSData *soundData = [NSData dataWithContentsOfURL:url]; AVAudioPlayer *avPlayer = [[AVAudioPlayer alloc] initWithData:soundData error: nil]; [avPlayer play]; } See anything wrong? By the way it compiles and runs, but nothing happens when we hit the button that executes that code.

    Read the article

  • About redirected stdout in System.Diagnostics.Process

    - by sforester
    I've been recently working on a program that convert flac files to mp3 in C# using flac.exe and lame.exe, here are the code that do the job: ProcessStartInfo piFlac = new ProcessStartInfo( "flac.exe" ); piFlac.CreateNoWindow = true; piFlac.UseShellExecute = false; piFlac.RedirectStandardOutput = true; piFlac.Arguments = string.Format( flacParam, SourceFile ); ProcessStartInfo piLame = new ProcessStartInfo( "lame.exe" ); piLame.CreateNoWindow = true; piLame.UseShellExecute = false; piLame.RedirectStandardInput = true; piLame.RedirectStandardOutput = true; piLame.Arguments = string.Format( lameParam, QualitySetting, ExtractTag( SourceFile ) ); Process flacp = null, lamep = null; byte[] buffer = BufferPool.RequestBuffer(); flacp = Process.Start( piFlac ); lamep = new Process(); lamep.StartInfo = piLame; lamep.OutputDataReceived += new DataReceivedEventHandler( this.ReadStdout ); lamep.Start(); lamep.BeginOutputReadLine(); int count = flacp.StandardOutput.BaseStream.Read( buffer, 0, buffer.Length ); while ( count != 0 ) { lamep.StandardInput.BaseStream.Write( buffer, 0, count ); count = flacp.StandardOutput.BaseStream.Read( buffer, 0, buffer.Length ); } Here I set the command line parameters to tell lame.exe to write its output to stdout, and make use of the Process.OutPutDataRecerved event to gather the output data, which is mostly binary data, but the DataReceivedEventArgs.Data is of type "string" and I have to convert it to byte[] before put it to cache, I think this is ugly and I tried this approach but the result is incorrect. Is there any way that I can read the raw redirected stdout stream, either synchronously or asynchronously, bypassing the OutputDataReceived event? PS: the reason why I don't use lame to write to disk directly is that I'm trying to convert several files in parallel, and direct writing to disk will cause severe fragmentation. Thanks a lot!

    Read the article

  • how to play an audio soundclip when a nib is loaded - welcome screen - xcode

    - by Pavan
    I would like to do the following two things: 1) I would like to play an audio file qhen a nib is loaded 2) After that i would like to switch views when the audio file has finished playing. This will be easy as i just need to call the event that initiates the change of view by using the delegate method -(void) audioPlayerDidFinishPlaying{ //code to change view } I dont know how to to play the audio file when a nib is loaded. Using the AVFoundation framework, I tried doing the following after setting up the audio player and the variables associated with it in the appropriate places i wrote the following: - (void)viewDidLoad { [super viewDidLoad]; NSString *soundFilePath = [[NSBundle mainBundle] pathForResource: @"sound" ofType: @"mp3"]; NSURL *fileURL = [[NSURL alloc] initFileURLWithPath: soundFilePath]; AVAudioPlayer *newPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL: fileURL error:nil]; [fileURL release]; self.player = newPlayer; [newPlayer release]; [player prepareToPlay]; [player setDelegate: self]; [player play]; } Although this does not play the file as the viewdidload method gets called before the nib is shown so the audio is never played or heard. What do i need to do so that i can play the audio file AFTER the nib has loaded and is shown on the screen? can someone please help me as ive been working on this for 3 hours now. Thanks in advance.

    Read the article

  • php.ini on goDaddy

    - by Afrosimon
    Hey all, I've got a little problem on a goDaddy server. I have a php script (ajaxCRUD) in which there's an upload field, and I can't get it to accept file over the default limit. I always get this (when I output the $_FILE[$fieldname]) : array(5) { ["name"]=> string(13) "children2.mp3" ["type"]=> string(0) "" ["tmp_name"]=> string(0) "" ["error"]=> int(1) ["size"]=> int(0) } Things I tried : Added a parameter in the HTML form ([...]name="MAX_FILE_SIZE" value="10000000"[...]) Changed the php5.ini at the root of the server, to no avail. After a phpinfo(), no differences are seen, even though the phpinfo clearly indicate it is reading the same php5.ini : [...]/html/php5.ini. Here is what I added in this file : upload_tmp_dir = ./temp upload_max_filesize = 20M Anything under 2M (the default value) is okay, so there's no problem with the upload path or file permission. I don't have any more idea for the moment, do any of you has one?

    Read the article

  • How to play simultaneous multiply audio sources in Silverlight

    - by Shurup
    I want to play simultaneous multiply audio sources in Silverlight. So I've created a prototype in Silverlight 4 that should play a two mp3 files containing the same ticks sound with an intervall 1 second. So these files must be sounded as one sound if they will be played together with any whole second offsets (0 and 1, 0 and 2, 1 and 1 seconds, etc.) I my prototype I use two MediaElement (me and me2) objects. DateTime startTime; private void Play_Clicked(object sender, RoutedEventArgs e) { me.SetSource(new FileStream(file1), FileMode.Open))); me2.SetSource(new FileStream(file2), FileMode.Open))); var timer = new DispatcherTimer { Interval = TimeSpan.FromMilliseconds(1) }; timer.Tick += RefreshData; timer.Start(); } First file should be played at 00:00 sec. and the second in 00:02 second. void RefreshData(object sender, EventArgs e) { if(me.CurrentState != MediaElementState.Playing) { startTime = DateTime.Now; me.Play(); return; } var elapsed = DateTime.Now - startTime; if(me2.CurrentState != MediaElementState.Playing && elapsed >= TimeSpan.FromSeconds(2)) { me2.Play(); ((DispatcherTimer)sender).Stop(); } } The tracks played every time different and not simultaneous as they should (as one sound).

    Read the article

  • OSMF seek with Amazon Cloudfront

    - by giorrrgio
    I've written a little OSMF player that streams via RTMP from Amazon Cloudfront. There's a known issue, the mp3 duration is not correctly readed from metadata and thus the seek function is not working. I know there's a workaround implying the use of getStreamLength function of NetConnection, which I successfully implemented in a previous non-OSMF player, but now I don't know how and when to call it, in terms of OSMF Events and Traits. This code is not working: protected function initApp():void { //the pointer to the media var resource:URLResource = new URLResource( STREAMING_PATH ); // Create a mediafactory instance mediaFactory = new DefaultMediaFactory(); //creates and sets the MediaElement (generic) with a resource and path element = mediaFactory.createMediaElement( resource ); var loadTrait:NetStreamLoadTrait = element.getTrait(MediaTraitType.LOAD) as NetStreamLoadTrait; loadTrait.addEventListener(LoaderEvent.LOAD_STATE_CHANGE, _onLoaded); player = new MediaPlayer( element ); //Marker 5: Add MediaPlayer listeners for media size and current time change player.addEventListener( DisplayObjectEvent.MEDIA_SIZE_CHANGE, _onSizeChange ); player.addEventListener( TimeEvent.CURRENT_TIME_CHANGE, _onProgress ); initControlBar(); } private function onGetStreamLength(result:Object):void { Alert.show("The stream length is " + result + " seconds"); duration = Number(result); } private function _onLoaded(e:LoaderEvent):void { if (e.newState == LoadState.READY) { var loadTrait:NetStreamLoadTrait = player.media.getTrait(MediaTraitType.LOAD) as NetStreamLoadTrait; if (loadTrait && loadTrait.netStream) { var responder:Responder = new Responder(onGetStreamLength); loadTrait.connection.call("getStreamLength", responder, STREAMING_PATH); } } }

    Read the article

  • Jquery number of clicks record

    - by Maxime
    Hi, I am working on this page where I'm using a Jquery Mp3 Player (Jplayer) with its playlist. What I want to do is very simple in theory: I want to record the number of clicks for every playlist element. When someone enters the page, his number of clicks are at 0 for every element. The visitor can click a few times on element #1, then go to element #2 (which will be at 0), then come back to element #1 and the number of clicks must be saved. We don't need to save it for next visits. Jplayer has this function that loads each time a new playlist element is being loaded: function playListChange( index ) In which every element has its own ID dynamically updated: myPlayList[index].song_id Here's my code: function playListChange( index ) { var id = myPlayList[index].song_id; if(!o) { var o = {}; } if(!o[id]) { o[id] = 0; } alert(o[id]); … $("#mydiv").click { o[id] = o[id]+1; } … But o[id] is reset every time and the alert always show 0. Why? Thanks for any reply.

    Read the article

  • Java Trying to get a line of source from a website

    - by dsta
    I'm trying to get one line of source from a website and then I'm returning that line back to main. I keep on getting an error at the line where I define InputStream in. Why am I getting an error at this line? public class MP3LinkRetriever { private static String line; public static void main(String[] args) { String link = "www.google.com"; String line = ""; while (link != "") { link = JOptionPane.showInputDialog("Please enter the link"); try { line = Connect(link); } catch(Exception e) { } JOptionPane.showMessageDialog(null, "MP3 Link: " + parseLine(line)); String text = line; Toolkit.getDefaultToolkit( ).getSystemClipboard() .setContents(new StringSelection(text), new ClipboardOwner() { public void lostOwnership(Clipboard c, Transferable t) { } }); JOptionPane.showMessageDialog(null, "Link copied to your clipboard"); } } public static String Connect(String link) throws Exception { String strLine = null; InputStream in = null; try { URL url = new URL(link); HttpURLConnection uc = (HttpURLConnection) url.openConnection(); in = new BufferedInputStream(uc.getInputStream()); Reader re = new InputStreamReader(in); BufferedReader r = new BufferedReader(re); int index = -1; while ((strLine = r.readLine()) != null && index == -1) { index = strLine.indexOf("<source src"); } } finally { try { in.close(); } catch (Exception e) { } } return strLine; } public static String parseLine(String line) { line = line.replace("<source", ""); line = line.replace(" src=", ""); line = line.replace("\"", ""); line = line.replace("type=", ""); line = line.replace("audio/mpeg", ""); line = line.replace(">", ""); return line; } }

    Read the article

  • FMOD Compiling trouble

    - by CptAJ
    I'm trying to get started with FMOD but I'm having some issues compiling the example code in this tutorial: http://www.gamedev.net/reference/articles/article2098.asp I'm using MinGW, I placed the libfmodex.a file in MinGW's include folder (I also tried linking directly to the filename) but it doesn't work. Here's the output. C:\>g++ -o test1.exe test1.cpp -lfmodex test1.cpp:4:1: error: 'FSOUND_SAMPLE' does not name a type test1.cpp: In function 'int main()': test1.cpp:9:29: error: 'FSOUND_Init' was not declared in this scope test1.cpp:12:4: error: 'handle' was not declared in this scope test1.cpp:12:53: error: 'FSOUND_Sample_Load' was not declared in this scope test1.cpp:13:30: error: 'FSOUND_PlaySound' was not declared in this scope test1.cpp:21:30: error: 'FSOUND_Sample_Free' was not declared in this scope test1.cpp:22:17: error: 'FSOUND_Close' was not declared in this scope This is the particular example I'm using: #include <conio.h> #include "inc/fmod.h" FSOUND_SAMPLE* handle; int main () { // init FMOD sound system FSOUND_Init (44100, 32, 0); // load and play sample handle=FSOUND_Sample_Load (0,"sample.mp3",0, 0, 0); FSOUND_PlaySound (0,handle); // wait until the users hits a key to end the app while (!_kbhit()) { } // clean up FSOUND_Sample_Free (handle); FSOUND_Close(); } I have the header files in the "inc" path where my code is. Any ideas as to what I'm doing wrong?

    Read the article

  • Embedded CSS Media Queries Not Working

    - by Greg
    I am new to CSS media queries, and I was first trying to get pdf/mp3/mp4 buttons to get centered on this page whenever a mobile device is using it (http://www.mannachurch.org/portfolio-type/recycled-junk/). Keep in mind for that I am using a highly modified wordpress theme. So I tried experimenting to isolate this issue. However, I don't seem to have any control over using media queries and I can't even perform anything even on this simple HTML file: <!DOCTYPE html> <html> <head> <title>Title of the document</title> <style type="text/css"> body{background-color: blue;} @media only screen and (min-device-width : 599px) and (max-device-width : 600px) { body {background-color:black; } } </style> </head> <body> <p>This is an experiment<p/> </body> </html> What am I doing wrong?

    Read the article

  • Which way to go in Linux 3D programming?

    - by Tek
    I'm looking for some answers for a project I'm thinking of. I've searched and from what I understand (correct me if I'm wrong) the only way the program I want to make will work is through 3D application. Let me explain. I plan to make a studio production program but it's unique in the fact that I want to be able to make it fluid. Let me explain. Imagine Microsoft's Surface program where you're able to touch and drag pictures across the screen. Instead of pictures I want them to be sound samples (wavs,mp3,etc). Of course instead the input will be with the mouse but if I ever do finish the project I would totally add touch screen input compatibility! Anyway, I'm guessing there's "physics" to do with it which is why I'm thinking that even though it'll be a 2D application I'll need to code it in a 3D environment. Assuming that I'm correct in how I want to approach my project, where can I start learning about 3D programming? I actually come from PHP programming which will make C++ easier for me to learn. But I don't even know where to start. If I'm not wrong OpenGL is the most up to date API as far as I know. Anyway, please give me your insights guys. I could really use some guidance here since I could totally be wrong in everything that I wrote :)

    Read the article

  • Save as Ringtone from ContextMenu

    - by kostas_menu
    I have created a button that onClick plays a mp3 file.I have also create a context menu that when you press the button for 2 secs it prompts you to save it as ringtone.How can i save it somewhere in my sd?this is my code: public void onCreate(Bundle icicle) { super.onCreate(icicle); setContentView(R.layout.main); Toast.makeText(a.this, "Touch and listen", Toast.LENGTH_SHORT).show(); Button button = (Button) findViewById(R.id.btn1); registerForContextMenu(button); button.setOnClickListener(new View.OnClickListener() { public void onClick(View v){ MediaPlayer mp = MediaPlayer.create(a.this, R.raw.myalo); mp.start(); Toast.makeText(a.this, "Eisai sto myalo", Toast.LENGTH_SHORT).show(); } }); @Override public void onCreateContextMenu(ContextMenu menu, View v,ContextMenuInfo menuInfo) { super.onCreateContextMenu(menu, v, menuInfo); menu.setHeaderTitle("Save As:"); menu.add(0, v.getId(), 0, "Ringtone"); } @Override public boolean onContextItemSelected(MenuItem item) { if(item.getTitle()=="Ringtone"){function1(item.getItemId());} else {return false;} return true; } public void function1(int id){ Toast.makeText(this, "Ringtone Saved", Toast.LENGTH_SHORT).show(); } }

    Read the article

  • Setting Image.Source doesn't update when loading from a resource.

    - by ChrisF
    I've got this definition in my XAML: <Image Name="AlbumArt" Source="/AlbumChooser2;component/Resources/help.png" /> The image is display OK on startup. In my code I'm looking for mp3's to play and I display the associated album art in this Image. Now if there's no associated image I want to display a "no image" image. So I've got one defined and I load it using: BitmapImage noImage = new BitmapImage( new Uri("/AlbumChooser2;component/Resources/no_image.png", UriKind.Relative)); I've got a helper class that finds the image if there is one (returning it as a BitmapImage), or returns null if there isn't one: if (findImage.Image != null) { this.AlbumArt.Source = findImage.Image; // This works } else { this.AlbumArt.Source = noImage; // This doesn't work } In the case where an image is found the source is updated and the album art gets displayed. In the case where an image isn't found I don't get anything displayed - just a blank. I don't think that it's the setting of AlbumArt.Source that's wrong, but the loading of the BitmapImage. If I use a different image it works, but if I recreate the image it doesn't work. What could be wrong with the image?

    Read the article

  • Saving an ActiveRecord non-transactionally.

    - by theFunkyEngineer
    My application accepts file uploads, with some metadata being stored in the DB, and the file itself on the file system. I am trying to make the metadata visible in the application before the file upload and post-processing are finished, but because saves are transactional, I have had no success. I have tried the callbacks and calling create_or_update() instead of save(), all to no avail. Is there a way to do this without re-writing the guts of ActiveRecord::Base? I've even attempted naming the method make() instead of save(), but perplexingly that had no effect. The code below "works" fine, but the database is not modified until everything else is finished. def save(upload) uploadFile = upload['datafile'] originalName = uploadFile.original_filename self.fileType = File.extname(originalName) create_or_update() # write the file File.open(self.filePath, "wb") { |f| f.write(uploadFile.read) } begin musicFile = TagLib::File.new(self.filePath()) self.id3Title = musicFile.title self.id3Artist = musicFile.artist self.id3Length = musicFile.length rescue TagLib::BadFile => exc logger.error("Failed to id track: \n #{exc}") end if(self.fileType == '.mp3') convertToOGG(); end create_or_update() end Any ideas would be quite welcome, thanks.

    Read the article

  • JavaScript: Changing src-attribute of a embed-tag

    - by Mike
    I have the following scenario. I show the user some audio files from the server. The user clicks on one, then onFileSelected is eventually executed with both the selected folder and file. What the function does is change the source from the embedded object. So in a way, it is a preview of the selected file before accepting it and save the user's choice. A visual aid. HTML <embed src="/resources/audio/_webbook_0001/embed_test.mp3" type="audio/mpeg" id="audio_file"> JavaScript function onFileSelected(file, directory) { jQuery('embed#audio_file').attr('src', '/resources/audio/'+directory+'/'+file); }; Now, this works fine in Firefox, but Safari and Chrome simply refuse to change the source, regardless of Operating System. jQuery finds the object (jQuery.size() returns 1), it executes the code, but no change in the HTML Code. Why does Safari prevent me from changing the <embed> source and how can I circumvent this?

    Read the article

  • How can flash call jquery function in its event

    - by user2955639
    I want jquery to do something during some of the events when an audio is playing. So I'm coding a function like this <script> $.fn.playMedia = function(options){ var opts = $.extend({}, { swfSrc: '' timeUpdated: function(currentTime){}, startPlay: function(){}, endPlay: function(){} }, options); return $(this).each(function(){ // call flash to play the media whose src is opts.swfSrc // Is it possible that flash can call the js functions(opts.timeUpdate, opts.startPlay and opts.endPlay) at each time of the event is triggered? }); }}; </script> // Usage <div id="player"></div> <script> $('#player').playMedia({ swfSrc: '/path/song.mp3', timeUpdated: function(currentTime){ comsole.log(currentTime); } }); </script> I'm a totally layman of flash, I just guess this works. Hope someone could tell me how to make up a swf file for this jquery function. Or is there any existing jquery plugin which does this thing but can re-design apperance flexibly. Thank you very much!

    Read the article

  • HTML5 Local Storage of audio element source - is it possible?

    - by andrewdotcom
    Hi stackoverflow experts I've been experimenting with the audio and local storage features of html5 of late and have run into something that has me stumped. I'd like to be able to cache or store the source of the audio element locally to enable speedier and offline playback. The problem is I can't see how this is possible with the current implementation. I have tried the following using webkit: Creating a manifest file to set up local caching but the audio file appears not to be a cacheable item maybe due to the way it is stream or something I have also attempted to use javascript to put an audio object into local storage but the size of the mp3 makes this impossible due to memory issues (i think). I have tried to use the data uri and base64 to use the html as a audio transport that can be cached but again the filesize makes this prohibitive. Also the audio element does not seem to like this in webkit (works fine in mozilla) I have tried several methods of putting the data into the local database store. Again suffering the same issues as the other cases. I'd love to hear any other ideas anyone may have as to how I could achieve my goal of offline playback using caching/local storage in webkit.

    Read the article

  • How to do something when AVQueuePlayer finishes the last playeritem

    - by user1634529
    I've got an AVQueuePlayer which I'm creating from an array of 4 AVPlayerItems, and it all plays fine. I want to do something when the last item in the queue finishes playing, I've looked a load of answers on here and this is the one that looks most promising for what I want: The best way to execute code AFTER a sound has finished playing In my button handler i have this code: static const NSString *ItemStatusContext; [thePlayerItemA addObserver:self forKeyPath:@"status" options:0 context:&ItemStatusContext]; [[NSNotificationCenter defaultCenter] addObserver:self selector:@selector(playerItemDidReachEnd) name:AVPlayerItemDidPlayToEndTimeNotification object:thePlayerItemA]; theQueuePlayer = [AVQueuePlayer playerWithPlayerItem:thePlayerItemA]; [theQueuePlayer play]; and then I have a function to handle playerItemDidReachEnd: - (void)playerItemDidReachEnd:(NSNotification *)notification { // Do stuff here NSLog(@"IT REACHED THE END"); } But when I run this I get an Internal Inconsistency Exception: An -observeValueForKeyPath:ofObject:change:context: message was received but not handled. Key path: status Observed object: <AVPlayerItem: 0x735a130, asset = <AVURLAsset: 0x73559c0, URL = file://localhost/Users/mike/Library/Application%20Support/iPhone%20Simulator/5.0/Applications/A0DBEC13-2DA6-4887-B29D-B43A78E173B8/Phonics%2001.app/yes.mp3>> Change: { kind = 1; } What am I doing wrong?

    Read the article

  • How can I get the contents of my table with dynamic row adding?

    - by user359706
    how to retrieve from the server-side contained a table html constructed this way: <table id="myTable"> <tr> <th> <input type="text"> name </th> <th> <input type="text"> quantity </th> </tr> <tr> <th> <input id="name_1"> phone </th> <th> <input id="quantity_1"> 15 </th> </tr> <tr> <th> <input ="name_2"> id mp3 </th> <th> <input id="quantity_2"> 26 <</th> </tr> ... I can not make use of <asp:Table> ... because for technical reasons I did not find a solution following this post: http://stackoverflow.com/questions/3003912/how-to-dynamic-adding-rows-into-asp-net-table How can retrieve the contents values of my table (dynamic) for each row. Rows will be added in client-side js Thank you.

    Read the article

  • Dealing with large directories in a checkout

    - by Eric
    I am trying to come up with a version control process for a web app that I work on. Currently, my major stumbling blocks are two directories that are huge (both over 4GB). Only a few people need to work on things within the huge directories; most people don't even need to see what's in them. Our directory structure looks something like: / --file.aspx --anotherFile.aspx --/coolThings ----coolThing.aspx --/bigFolder ----someHugeMovie.mov ----someHugeSound.mp3 --/anotherBigFolder ----... I'm sure you get the picture. It's hard to justify a checkout that has to pull down 8GB of data that's likely useless to a developer. I know, it's only once, but even once could be really frustrating for someone (and will make it harder for me to convince everyone to use source control). (Plus, clean checkouts will be painfully slow.) These folders do have to be available in the web application. What can I do? I've thought about separate repositories for the big folders. That way, you only download if you need it; but then how do I manage checking these out onto our development server? I've also thought about not trying to version control those folders: just update them directly on the web server... but I am not enamored of this idea. Is there some magic way to simply exclude directories from a checkout that I haven't found? (Pretty sure there is not.) Of course, there's always the option to just give up, bite the bullet, and accept downloading 8 useless GB. What say you? Have you encountered this problem before? How did you solve it?

    Read the article

  • How can you know what is w3wp.exe doing? (or how to diagnose a performance problem)

    - by Daniel Magliola
    I'm having a performance problem in a site we've made, and I'm not exactly sure how to start diagnosing it. The short description is: We have a very small site (http://hearablog.com) with very little traffic, in a crappy dedicated server, CPU is always very high, sometimes it stays at 100% for minutes, and w3wp.exe is taking most of it. A typical scenario is w3wp.exe takes 60%, and SQL Server takes about 30%. Our DB is pretty small too. Long description and more details: The site is hosted in a very crappy server by Cari.Net. From the beginning we had the feeling that the server didn't quite behave correctly, like some things would take just too long, so this could be a configuration problem from the get go. It may also be that we are getting a virtual server while we're supposed to have a dedicated one, although we have no evidence that'd indicate this, except for the fact that the server tends to be quite slow. The server is Windows 2008 Standard 64-bit, with SQL 2008 Express Hardware is a Celeron 2.80 GHz, 1Gb RAM The website is developed in ASP.Net MVC, using Entity Framework for data access. Now, this is pretty crappy hardware, but i've had other servers with these guys, with equivalent (or worse) HW, and performance is much better than this one. That said, the other servers have W2003 and SQL2005, and I'm using ASP.Net "WebForms" 2.0, no MVC, no LINQ, no EF; so I'm not sure whether going to 2008 / the other stuff means a big performance penalty is expected. I'm serving MP3 files (5-20 Mb) regularly, which is a slightly unusual load, maybe that is causing some kind of problems? Would that cause w3wp to use a lot of CPU? Disk usage seems very low. Memory is usually around 90%, but disk usage seems to indicate it's not paging much. I get tons of e-mails every day about SQL timeouts, for queries taking over 30 seconds, although all our queries are pretty straightforward (or should be, but EF may be screwing it up). This is what resource monitor looks like in one of these "sprints" of 100% CPU, in case there's anything useful there. And a snapshot of some performance counters: Now, what confuses me very much is that CPU usage of w3wp is just so high. It shouldn't be doing much really... So my questions are... Is there any way of finding out "what" it is doing? Maybe even profile it? Any performance counters I should be looking at? Is this to be expected given this hardware/software configuration? Is this could be cause by some kind of configuration failure, where would you start looking? Thank you VERY much. Daniel Magliola

    Read the article

  • How can I simulate blocking RTMP over port 80 on Windows?

    - by Christian Nunciato
    It seems like this should be so simple, but since this isn't my area of expertise, I'm having a hell of a time figuring out how to do it. Basically, I have a Flash app and I'm connecting to a Flash Media Server to stream some content. The URL I'm using to do this, for example, looks like this: rtmp://someserver.com/some/path/mp3:somefile Everything works -- but that's sort of the problem. When I'm trying to do is simulate my users attempting to play back my media under more restrictive conditions than the ones I have here (i.e., none) -- namely being stuck behind firewalls or proxy servers that block access to RTMP streams. Flash, according to Adobe, is equipped to handle proxy servers and firewalls automatically, like so (from the docs): When you do not specify a port number in an RTMP address, Flash will attempt to connect to port 1935. If it fails it will then try to connect to port 443; if that fails, it will try port 80. [And if that fails, it will attempt to connect via RTMPT (i.e., HTTP tunneling) on port 80.] So no coding is required to access ports 1935, 443, or port 80 if you do not specify a port in the RTMP address. The problem I'm having is setting up a reliable environment in which to test that this behavior actually happens. I'm on a Windows machine, for example, so with Windows Firewall, I can block certain ports and protocols (1935, 443), but I don't want to block port 80, because the final fallback protocol (RTMPT) is supposed to run on port 80, and Windows Firewall only gives me enough granularity (as far as I know, anyway) to block "all outbound TCP traffic to remote port 80" -- that is, I can't, apparently, block "all outbound RTMP traffic to port 80" while leaving RTMPT traffic to port 80 unaffected. My understanding thus far is that I'll probably need to set up a proxy server to do this. Is this correct? Or is there a simpler way (on Win 7, at least) to filter out RTMP to 1935, RTMP to 443, RTMP to 80, but still allow RTMPT to 80 (where all four hostnames are identical)? And if I do have to set up a proxy server, what's the simplest way to go on Windows? I've set up WinProxy, which seems a bit janky but apparently works -- but then what I can't figure out is how to tell Windows to force all TCP traffic (including RTMP, RTMPT and HTTO) through this proxy server so I can turn around and reject the requests for RTMP. Any help would be hugely appreciated. This isn't my realm of expertise and I've alreasdy spent more time on it than I probably should. :)

    Read the article

  • DVD playback with Windows Media Player 11 works fine, but when copied to HDD and then played back, t

    - by stakx
    I have several DVDs with short documentaries on it. Since the notebook I'm using (a Dell Latitude E6400) has only one DVD drive, and I might play back those short movies very often, I thought of copying them to the HDD and playing them back from there. However, I've run into a problem, namely stuttering audio. Problem description: When I play back these movies directly from DVD (with Windows Media Player 11 under Windows Vista), everything works fine. Smooth video, no significant audio problems (only the occasional click). But as soon as I copy any of these DVDs to the HDD and try to play them back from there (e.g. using the wmpdvd://drive/title/chapter?contentdir=path protocol, I get stuttering audio — audio playback sounds like a machine gun for a third of a second or so, approx. every 8 seconds. I have tried converting the VOB files from the DVD to another format (ie. ripping), but that resulted in a noticeable downgrade of picture quality. Therefore I thought it best to keep the files in their original format, if possible. Still, I suspect that the stuttering audio is due to some (de-)muxing problem, and that changing the file format might help. (After all, video playback is fine; therefore I don't think that the hardware is too slow for playback.) Only thing is, I don't know how to convert the VOB files to another Windows Media Player-compatible format without quality loss. I hope someone can help me, or give me further pointers on things I could try out to get HDD playback to work without the problem described. Some things I've tried so far, without any success: VOB2MPG, in order to convert the .vob file to a .mpg file. But that changes only the A/V container, not the content. No re-encoding takes place at all. Re-encoding with MPlayer/MEncoder. Lots of quality loss there, and I frankly haven't got the time to test all possible settings combinations available. Disabling all plug-ins, equalizers, etc. in Windows Media Player. Disabling all hardware acceleration on the audio playback device. Further info on the VOB files I'm trying to playback: The video format is MPEG ES, PAL 720x576 pixels @ 24/25 frames per second. The sound stream is uncompressed PCM, 16-bit stereo @ 48kHz. (Might it help if I somehow re-encoded the sound stream at a lower resolution, or as an MP3? If so, how would I do this without changing the video stream?) P.S.: I am limited to using Windows Media Player (11). (I previously tried MPlayer btw., but the video playback quality was surprisingly bad.)

    Read the article

< Previous Page | 46 47 48 49 50 51 52 53 54 55  | Next Page >