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  • I dont know how or where to add the correct encoding code to this iPhone code...

    - by BC
    Ok, I understand that using strings that have special characters is an encoding issue. However I am not sure how to adjust my code to allow these characters. Below is the code that works great for text that contains no special characters, but can you show me how and where to change the code to allow for the special characters to be used. Right now those characters crash the app. enter code here - (void)alertView:(UIAlertView *)alertView clickedButtonAtIndex:(NSInteger)buttonIndex{ if (buttonIndex == 1) { //iTunes Audio Search NSString *stringURL = [NSString stringWithFormat:@"http://phobos.apple.com/WebObjects/MZSearch.woa/wa/search?WOURLEncoding=ISO8859_1&lang=1&output=lm&term=\"%@\"",currentSong.title]; stringURL = [stringURL stringByAddingPercentEscapesUsingEncoding:NSASCIIStringEncoding]; NSURL *url = [NSURL URLWithString:stringURL]; [[UIApplication sharedApplication] openURL:url]; } } And this: -(IBAction)launchLyricsSearch:(id)sender{ WebViewController * webView = [[WebViewController alloc] initWithNibName:@"WebViewController" bundle:[NSBundle mainBundle]]; webView.webURL = [NSString stringWithFormat:@"http://www.google.com/m/search?hl=es&q=\"%@\"+letras",currentSong.title]; webView.webTitle = @"Letras"; [self.navigationController pushViewController:webView animated:YES]; } Please show me how and where to do this for these two bits of code.

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  • How to eliminate tearing from animation?

    - by MusiGenesis
    I'm running an animation in a WinForms app at 18.66666... frames per second (it's synced with music at 140 BPM, which is why the frame rate is weird). Each cel of the animation is pre-calculated, and the animation is driven by a high-resolution multimedia timer. The animation itself is smooth, but I am seeing a significant amount of "tearing", or artifacts that result from cels being caught partway through a screen refresh. When I take the set of cels rendered by my program and write them out to an AVI file, and then play the AVI file in Windows Media Player, I do not see any tearing at all. I assume that WMP plays the file smoothly because it uses DirectX (or something else) and is able to synchronize the rendering with the screen's refresh activity. It's not changing the frame rate, as the animation stays in sync with the audio. Is this why WMP is able to render the animation without tearing, or am I missing something? Is there any way I can use DirectX (or something else) in order to enable my program to be aware of where the current scan line is, and if so, is there any way I can use that information to eliminate tearing without actually using DirectX for displaying the cels? Or do I have to fully use DirectX for rendering in order to deal with this problem? Update: forgot a detail. My app renders each cell onto a PictureBox using Graphics.DrawImage. Is this significantly slower than using BitBlt, such that I might eliminate at least some of the tearing by using BitBlt?

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  • What can cause my code to run slower when the server JIT is activated?

    - by durandai
    I am doing some optimizations on an MPEG decoder. To ensure my optimizations aren't breaking anything I have a test suite that benchmarks the entire codebase (both optimized and original) as well as verifying that they both produce identical results (basically just feeding a couple of different streams through the decoder and crc32 the outputs). When using the "-server" option with the Sun 1.6.0_18, the test suite runs about 12% slower on the optimized version after warmup (in comparison to the default "-client" setting), while the original codebase gains a good boost running about twice as fast as in client mode. While at first this seemed to be simply a warmup issue to me, I added a loop to repeat the entire test suite multiple times. Then execution times become constant for each pass starting at the 3rd iteration of the test, still the optimized version stays 12% slower than in the client mode. I am also pretty sure its not a garbage collection issue, since the code involves absolutely no object allocations after startup. The code consists mainly of some bit manipulation operations (stream decoding) and lots of basic floating math (generating PCM audio). The only JDK classes involved are ByteArrayInputStream (feeds the stream to the test and excluding disk IO from the tests) and CRC32 (to verify the result). I also observed the same behaviour with Sun JDK 1.7.0_b98 (only that ist 15% instead of 12% there). Oh, and the tests were all done on the same machine (single core) with no other applications running (WinXP). While there is some inevitable variation on the measured execution times (using System.nanoTime btw), the variation between different test runs with the same settings never exceeded 2%, usually less than 1% (after warmup), so I conclude the effect is real and not purely induced by the measuring mechanism/machine. Are there any known coding patterns that perform worse on the server JIT? Failing that, what options are available to "peek" under the hood and observe what the JIT is doing there?

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  • How can i use a commandlinetool (ie. sox) via subprocess.Popen with mod_wsgi?

    - by marue
    I have a custom django filefield that makes use of sox, a commandline audiotool. This works pretty well as long as i use the django development server. But as soon as i switch to the production server, using apache2 and mod_wsgi, mod_wsgi catches every output to stdout. This makes it impossible to use the commandline tool to evaluate the file, for example use it to check if the uploaded file really is an audio file like this: filetype=subprocess.Popen([sox,'--i','-t','%s'%self.path], shell=False,\ stdout=subprocess.PIPE, stderr=subprocess.PIPE) (filetype,error)=filetype.communicate() if error: raise EnvironmentError((1,'AudioFile error while determining audioformat: %s'%error)) Is there a way to workaround for this? edit the error i get is "missing filename". I am using mod_wsgi 2.5, standard with ubuntu 8.04. edit2 What exactly happens, when i call subprocess.Popen from within django in mod_wsgi? Shouldn't subprocess stdin/stdout be independent from django stdin/stdout? In that case mod_wsgi should not affect programms called via subprocess... I'm really confused right now, because the file i am trying to access is a temporary file, created via a filenamevariable that i pass to the file creation and the subprocess command. That file is being written to /tmp, where the rights are 777, so it can't be a rights issue. And the error message is not "file does not exist", but "missing filename", which suggests i am not passing a filename as parameter to the commandlinetool.

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  • reactivating or binding a hover function in jquery??

    - by mathiregister
    hi guys, with the following three lines: $( ".thumb" ).bind( "mousedown", function() { $('.thumb').not(this).unbind('mouseenter mouseleave'); }); i'm unbinding this hover-function: $(".thumb").hover( function () { $(this).not('.text, .file, .video, .audio').stop().animate({"height": full}, "fast"); $(this).css('z-index', z); z++; }, function () { $(this).stop().animate({"height": small}, "fast"); } ); i wonder how i can re-bind the exact same hover function again on mouseup? the follwoing three lines arent't working! $( ".thumb" ).bind( "mouseup", function() { $('.thumb').bind('mouseenter mouseleave'); }); to get what i wanna do here's a small explanation. I want to kind of deactivate the hover function for ALL .thumbs-elements when i click on one. So all (but not this) should not have the hover function assigned while i'm clicking on an object. If i release the mouse again, the hover function should work again like before. Is that even possible to do? thank you for your help!

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  • How do I post a .wav file from CS5 Flash, AS3 to a Java servlet?

    - by Muostar
    Hi, I am trying to send a byteArray from my .fla to my running tomcat server integrated in Eclipse. From flash I am using the following code: var loader:URLLoader = new URLLoader(); var header:URLRequestHeader = new URLRequestHeader("audio/wav", "application/octet-stream"); var request:URLRequest = new URLRequest("http://localhost:8080/pdp/Server?wav=" + tableID); request.requestHeaders.push(header); request.method = URLRequestMethod.POST; request.data = file;//wav; loader.load(request); And my java servlet looks as follows: try{ int readBytes = -1; int lengthOfBuffer = request.getContentLength(); InputStream input = request.getInputStream(); byte[] buffer = new byte[lengthOfBuffer]; ByteArrayOutputStream output = new ByteArrayOutputStream(lengthOfBuffer); while((readBytes = input.read(buffer, 0, lengthOfBuffer)) != -1) { output.write(buffer, 0, readBytes); } byte[] finalOutput = output.toByteArray(); input.close(); FileOutputStream fos = new FileOutputStream(getServletContext().getRealPath(""+"/temp/"+wav+".wav")); fos.write(finalOutput); fos.close(); When i run the flash .swf file and send the bytearray to the server, I receive following in the server's console window:: (loads of loads of Chinese symbols) May 20, 2010 7:04:57 PM org.apache.tomcat.util.http.Parameters processParameters WARNING: Parameters: Character decoding failed. Parameter '? (loads of loads of Chinese symbols) and then looping this for a long time. It is like I recieve the bytes but not encoding/decoding them correctly. What can I do?

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  • Suggestion on UPnP presentation

    - by Microkernel
    Hi all, I am working on an embedded device (bit higher end in terms of system resources but still an embedded one) which has lot of media content in it. I am trying to make it UPnP complaint and want to be able to control this device using a UPnP complaint control point/companion device like ipad. The step towards this is to be able to present the playlist content to the user. We thought of using HTML5 as a format to use. But as I am a noob in web technologies, I am not sure whats the best way to produce and present rich dynamic web pages. The content thats presented are video/audio listing that device can play and want this listing to be generated using the user's input criteria. So, what would be the best way to generate these dynamic pages which are rich and rendered as HTML5 pages. (looked at XML & XSLT, but there seems to be some limitations in how well one can use XSLT from some rewviews I saw). Thanks Microkernel PS: This may be silly or very basic as I am a embedded systems developer and not even a noob in web technologoes...

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  • Leveraging hobby experience to get a job

    - by Bernard
    Like many other's I began programming at an early age. I started when I was 11 and I learned C when I was 14 (now 26). While most of what I did were games just to entertain myself I did everything from low level 2D graphics, and binary I/O, to interfacing with free API's, custom file systems, audio, 3D animations, OpenGL, web sites, etc. I worked on a wide variety of things trying to make various games. Because of this experience I have tested out of every college level C/C++ programming course I have ever been offered. In the classes I took, my classmates would need a week to do what I finished in class with an hour or two of work. I now have my degree now and I have 2 years of experience working full time as a web developer however I would like to get back into C++ and hopefully do simulation programming. Unfortunately I have yet to do C++ as a job, I have only done it for testing out of classes and doing my senior project in college. So most of what I have in C++ is still hobby experience and I don't know how to best convey that so that I don't end up stuck doing something too low level for me. Right now I see a job offer that requires 2 years of C++ experience, but I have at least 9 (I didn't do C++ everyday for the last 14 years). How do I convey my experience? How much is it truly worth? and How do I get it's full value? The best thing that I can think of is a demo and a portfolio, however that only comes into play after an interview has been secured. I used a portfolio to land my current job. All answers and advice are appreciated.

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  • What does "Value does not fall within expected range" mean in runtime error?

    - by manuel
    Hi, I'm writing a plugin (dll file) using /clr and trying to implement speech recognition using .NET. But when I run it, I got a runtime error saying "Value does not fall within expected range", what does the message mean? public ref class Dialog : public System::Windows::Forms::Form { public: SpeechRecognitionEngine^ sre; private: System::Void btnSpeak_Click(System::Object^ sender, System::EventArgs^ e) { Initialize(); } protected: void Initialize() { //create the recognition engine sre = gcnew SpeechRecognitionEngine(); //set our recognition engine to use the default audio device sre->SetInputToDefaultAudioDevice(); //create a new GrammarBuilder to specify which commands we want to use GrammarBuilder^ grammarBuilder = gcnew GrammarBuilder(); //append all the choices we want for commands. //we want to be able to move, stop, quit the game, and check for the cake. grammarBuilder->Append(gcnew Choices("play", "stop")); //create the Grammar from th GrammarBuilder Grammar^ customGrammar = gcnew Grammar(grammarBuilder); //unload any grammars from the recognition engine sre->UnloadAllGrammars(); //load our new Grammar sre->LoadGrammar(customGrammar); //add an event handler so we get events whenever the engine recognizes spoken commands sre->SpeechRecognized += gcnew EventHandler<SpeechRecognizedEventArgs^> (this, &Dialog::sre_SpeechRecognized); //set the recognition engine to keep running after recognizing a command. //if we had used RecognizeMode.Single, the engine would quite listening after //the first recognized command. sre->RecognizeAsync(RecognizeMode::Multiple); //this->init(); } void sre_SpeechRecognized(Object^ sender, SpeechRecognizedEventArgs^ e) { //simple check to see what the result of the recognition was if (e->Result->Text == "play") { MessageBox(plugin.hwndParent, L"play", 0, 0); } if (e->Result->Text == "stop") { MessageBox(plugin.hwndParent, L"stop", 0, 0); } } };

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  • calculate camera up vector after glulookat()?

    - by carrots
    I'm just starting out teaching myself openGL and now adding openAL to the mix. I have some planets scattered around in 3D space and when I touch the screen, I'm assigning a sound to a random planet and then slowly and smoothly flying the "camera" over to look at it and listen to it. The animation/tweening part is working perfectly, but the openAL piece isn't quiet right. I move the camera around by doing a tiny translate() and gluLookAt() for every frame to keep things smooth (tweening the camera position and lookAt coords). The trouble seems to be with the stereo image I'm getting out of the headphones.. it seems like the left/right/up/down is mixed up sometimes after the camera rolls or spins. I am pretty sure the trouble is here: ALfloat listenerPos[]={camera->currentX,camera->currentY,camera->currentZ}; ALfloat listenerOri[]={camera->currentLookX, camera->currentLookY, camera->currentLookZ, 0.0,//Camera Up X <--- here 0.1,//Camera Up Y <--- here 0.0}//Camera Up Z <--- and here alListenerfv(AL_POSITION,listenerPos); alListenerfv(AL_ORIENTATION,listenerOri); I'm thinking I need to recompute the UP vector for the camera after each gluLookAt() to straighten out the audio positioning problem.. but after hours of googling and experimenting I'm stuck in math that suddenly got way over my head. 1) Am I right that I need to recalculate the up vector after each gluLookAt() i do? 2) If so, can someone please walk me though figuring out how to do that?

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  • Django: Serving a Download in a Generic View

    - by TheLizardKing
    So I want to serve a couple of mp3s from a folder in /home/username/music. I didn't think this would be such a big deal but I am a bit confused on how to do it using generic views and my own url. urls.py url(r'^song/(?P<song_id>\d+)/download/$', song_download, name='song_download'), The example I am following is found in the generic view section of the Django documentations: http://docs.djangoproject.com/en/dev/topics/generic-views/ (It's all the way at the bottom) I am not 100% sure on how to tailor this to my needs. Here is my views.py def song_download(request, song_id): song = Song.objects.get(id=song_id) response = object_detail( request, object_id = song_id, mimetype = "audio/mpeg", ) response['Content-Disposition'= "attachment; filename=%s - %s.mp3" % (song.artist, song.title) return response I am actually at a loss of how to convey that I want it to spit out my mp3 instead of what it does now which is to output a .mp3 with all of the current pages html contained. Should my template be my mp3? Do I need to setup apache to serve the files or is Django able to retrieve the mp3 from the filesystem(proper permissions of course) and serve that? If it do need to configure Apache how do I tell Django that? Thanks in advanced. These files are all on the HD so I don't need to "generate" anything on the spot and I'd like to prevent revealing the location of these files if at all possible. A simple /song/1234/download would be fantastic.

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  • Creating C++ client app for some abstract windows server - how to manage TCP connection to server speed?

    - by Kabumbus
    So we have some server with some address port and ip. we are developing that server so we can implement on it what ever we need for help. What are standard/best practices for data transfer speed management between C++ windows client app and server (C++)? My main point is in how to get how much data can be uploaded/downloaded from/to client via his low speed network to my relatively super fast server. (I need it for set up of his live stream Audio/Video bit rate) My try on explaining number 3. We do not care how fast is our server. It is always faster than needed. We care about client tyring to stream out to our server his media. he streams encoded (via ffmpeg) live video data to our server. But he has say ADSL with 500kb/s of outgoing traffic. Also he uses some ICQ or what so ever so he has less than 500 kb/s per second. And he wants to stream live video! So we need to set up our ffmpeg to encode video with respect to the bit rate user can provide. We develop server side and client side. We need a way of finding out how much user can upload per second currently (so value can change dynamically over time)

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  • 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; } }

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  • Multi-Part HTTP Request through xcode

    - by devsri
    Hello Everyone, i want to upload image,video and audio files to a server. I have read this thread on the similar topic but wasn't able to understand completely the flow of the code. It would be great if you can suggest me some sample code or tutorial to start with. I am using the following code to connect without any media to the server [UIApplication sharedApplication].networkActivityIndicatorVisible = YES; NSString *url =[[NSString alloc]initWithFormat:@"%@",[NetworkConstants getURL]]; NSURL *theURL =[NSURL URLWithString:url]; [url release]; NSMutableURLRequest *theRequest =[NSMutableURLRequest requestWithURL:theURL cachePolicy:NSURLRequestReloadIgnoringCacheData timeoutInterval:0.0f]; [theRequest setHTTPMethod:@"POST"]; NSString *theBodyString = [NSString stringWithFormat:@"json1=%@&userID=%@",jsonObject,[GlobalConstants getUID]]; NSData *theBodyData = [theBodyString dataUsingEncoding:NSUTF8StringEncoding]; [theRequest setHTTPBody:theBodyData]; NSURLConnection *conn = [[NSURLConnection alloc] initWithRequest:theRequest delegate:self]; if (conn) { NSLog(@"Successful in sending sync"); } else { NSLog(@"Failed Connection in sending sync"); } [conn release]; It would be really convenient for me if anything could be done editing this part of code. Any form of help would be highly appreciated. Thanks in advance!!

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  • How to use libavformat for a separate encoder?

    - by Brendon Tsai
    I've build a encoder based on the sample of QUALCOMM, which captures the video and compresses it into h264 file. I am using Android 4.2.2. Now I want to add a mp4 muxer into this encoder(actually, just video will be fine, I don't need audio). I want to use FFMpeg. But after I read the example, I found out that the muxer was using the encoder of FFMpeg. I don't know how to use the muxer part for another encoder. I've read this post, but I don't understand how the code provide video stream to the muxer. I think that mainly because I don't understand these code: AVCodecContext * strmCodec = oFmtCtx->streams[0]->codec; // Fill the required properties for codec context. // *from the documentation: // *The user sets codec information, the muxer writes it to the output. // *Mandatory fields as specified in AVCodecContext // *documentation must be set even if this AVCodecContext is // *not actually used for encoding. my_tune_codec(strmCodec); Can anyone give me a hint? Thank you!

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  • Custom Header in ASIHTTPREQUEST

    - by Sharon Nathaniel
    Here is my problem. I am using ASIHTTPREQUEST to send request and get response. Now my need of the time is to upload a image,audio , video or docs via PUT method with a custom header . The header takes filename , filesize,sha256sum,username,mediatype,resume as parameters. Here is the code how I am doing that . -(void)uploadFile { NSString *resume =@"0"; NSUserDefaults *userCredentials =[NSUserDefaults standardUserDefaults]; NSString *userName =[userCredentials objectForKey:@"userName"]; NSArray *objects =[NSArray arrayWithObjects:selectedMediaType,delegate.fileName,userName,fileSize,fileSHA256Sum,resume, nil]; NSArray *keys =[NSArray arrayWithObjects:@"mediatype",@"file_name",@"usrname",@"filesize",@"sha256sum",@"resume", nil]; discSentValue =[NSDictionary dictionaryWithObjects:objects forKeys:keys]; NSURL *url = [NSURL URLWithString:@"http://briareos.code20.com/putmedia.php"]; uploadMedia = [ASIFormDataRequest requestWithURL:url]; [uploadMedia addRequestHeader:@"MEDIA_UPLOAD_CUSTOM_HEADER" value:[discSentValue JSONRepresentation]]; [uploadMedia setData:dataToUpload forKey:@"File"]; [uploadMedia setDelegate:self]; [uploadMedia setRequestMethod:@"PUT"]; [uploadMedia startAsynchronous]; } But the server is unable to recognize the header , it always returns "invalid header"

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  • how do i call mpmovieplayer from applicationwillresignactive in ppdelegate

    - by ss30
    i'm using mpmovieplayer to play audio stream, one problem i'm having trouble with handling intruptions e.g when call is received. My player is declared in viewcontroller and i beleive i need to do something in applicationdidresignactive in my appdelegate right? how do i do that if my appdelegate isn't aware of my moviePlayer? i'm new to iphone dev so i'm learning as i go and enjoying it :) here is what i'm doing in viewcontroller -(IBAction)play1MButton:(id)sender{ [[NSNotificationCenter defaultCenter] addObserver:self selector:@selector(moviePlayerStatus:) name:MPMoviePlayerPlaybackStateDidChangeNotification object:nil]; NSString *url = @"http://22.22.22.22:8000/listen.pls"; [self.activityIndicator startAnimating]; moviePlayer = [[MPMoviePlayerController alloc] initWithContentURL:[NSURL URLWithString:url]]; [moviePlayer prepareToPlay]; [[NSNotificationCenter defaultCenter] addObserver:self selector:@selector(moviePlayerLoadStateChanged:) name:MPMoviePlayerLoadStateDidChangeNotification object:nil]; } } -(void) moviePlayerStatus:(NSNotification*)notification { //MPMoviePlayerController *moviePlayer = notification.object; MPMoviePlaybackState playbackState = moviePlayer.playbackState; if(playbackState == MPMoviePlaybackStateStopped) { NSLog(@"MPMoviePlaybackStateStopped"); } else if(playbackState == MPMoviePlaybackStatePlaying) { NSLog(@"MPMoviePlaybackStatePlaying"); } else if(playbackState == MPMoviePlaybackStatePaused) { NSLog(@"MPMoviePlaybackStatePaused"); } else if(playbackState == MPMoviePlaybackStateInterrupted) { NSLog(@"MPMoviePlaybackStateInterrupted"); } else if(playbackState == MPMoviePlaybackStateSeekingForward) { NSLog(@"MPMoviePlaybackStateSeekingForward"); } else if(playbackState == MPMoviePlaybackStateSeekingBackward) { NSLog(@"MPMoviePlaybackStateSeekingBackward"); } } - (void) moviePlayerLoadStateChanged:(NSNotification*)notification { if ([moviePlayer loadState] != MPMovieLoadStateUnknown) { [[NSNotificationCenter defaultCenter] removeObserver:self name:MPMoviePlayerLoadStateDidChangeNotification object:moviePlayer]; [moviePlayer play]; [self.activityIndicator stopAnimating]; [moviePlayer setFullscreen:NO animated:NO]; } } and in appdelegate - (BOOL)application:(UIApplication *)application didFinishLaunchingWithOptions:(NSDictionary *)launchOptions { // Override point for customization after application launch. self.window.rootViewController = self.viewController; [self.window makeKeyAndVisible]; AVAudioSession *audioSession = [AVAudioSession sharedInstance]; NSError *setCategoryError = nil; [audioSession setCategory:AVAudioSessionCategoryPlayback error:&setCategoryError]; if (setCategoryError) { } NSError *activationError = nil; [audioSession setActive:YES error:&activationError]; if (activationError) { } I can catch the errors but how do i use the player from appdelegate?!! thanks in advance

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  • Problems with video conversions through the web (local host)

    - by ron-d
    Hello, I get the following errors when I attempt video format conversions called from the local host: “An invalid media type was specified” for M4V to WMV conversions. “One or more arguments are invalid” for MP4 to WMV conversions. Here are the details of the problems: I’ve written a dll in C# that accepts videos in the formats AVI, WMV, M4V and MP4 and performs the following actions: Creates a copy of the input video in WMV format . Creates a WAV file of the input video audio portion. Creates a JPG image from a frame of the input video. I attached the dll to an ASP.NET web project that performs the dll actions. When tested through the developer studio, the actions are performed as intended for all formats. When I place the web project in place to be read when the local host is called through the web browser, the following behavior takes place: WMV format: All actions performed as intended. AVI format: Creates WMV file – OK Creates JPG image – OK Creates empty WAV file – problem. M4V format: Creates empty WAV file – problem. Does not create WMV file -problem Does not create JPG file –problem Throws me the error “An invalid media type was specified” MP4 format: Creates empty WAV file – problem. Does not create WMV file -problem Does not create JPG file –problem Throws me the error “One or more arguments are invalid” When I check their security property, all the files have the same permission access parameters (when I check their security property. Can anyone guide me as to how to solve these problems when the web project is called from the local host? Thank you.

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  • Real Time Sound Captureing J2ME

    - by Abdul jalil
    i am capturing sound in J2me and send these bytes to remote system, i then play these bytes on remote system.five second voice is capture and send to remote system. i get the repeated sound again .i am making a sound messenger please help me where i am doing wrong i am using the follown code . String remoteTimeServerAddress="192.168.137.179"; sc = (SocketConnection) Connector.open("socket://"+remoteTimeServerAddress+":13"); p = Manager.createPlayer("capture://audio?encoding=pcm&rate=11025&bits=16&channels=1"); p.realize(); RecordControl rc = (RecordControl)p.getControl("RecordControl"); ByteArrayOutputStream output = new ByteArrayOutputStream(); OutputStream outstream =sc.openOutputStream(); rc.setRecordStream(output); rc.startRecord(); p.start(); int size=output.size(); int offset=0; while(true) { Thread.currentThread().sleep(5000); rc.commit(); output.flush(); size=output.size(); if(size0) { recordedSoundArray=output.toByteArray(); outstream.write(recordedSoundArray,0,size); } output.reset(); rc.reset(); rc.setRecordStream(output); rc.startRecord(); }

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  • how to convert video from one format to another using php

    - by Meena
    hi i want to include the vedio download option in my webpage. I am using ffmpeg, but it seems to work very slow. Is there is any other way to do this or how to spead up the ffmpeg. i am using this code to get the frames from the vedio. to convert the vedio $call="ffmpeg -i ".$_SESSION['video_to_convert']." -vcodec libvpx -r 30 -b ".$quality." -acodec libvorbis -ab 128000 -ar ".$audio." -ac 2 -s ".$size." ".$converted_vids.$name.".".$type." -y 2> log/".$name.".txt"; $convert = (popen("start /b ".$call, "r")); pclose($convert); to get the frame from the vedio exec("ffmpeg -vframes 1 -ss ".$time_in_seconds." -i $converted_vids video_images.jpg -y 2>); but this code does not generate any error its loading continously.

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  • Turning a series of raw images into movie frames in Android

    - by Nicholas Killewald
    I've got an Android project I'm working on that, ultimately, will require me to create a movie file out of a series of still images taken with a phone's camera. That is to say, I want to be able to take raw image frames and string them together, one by one, into a movie. Audio is not a concern at this stage. Looking over the Android API, it looks like there are calls in it to create movie files, but it seems those are entirely geared around making a live recording from the camera on an immediate basis. While nice, I can't use that for my purposes, as I need to put annotations and other post-production things on the images as they come in before they get fed into a movie (plus, the images come way too slowly to do a live recording). Worse, looking over the Android source, it looks like a non-trivial task to rewire that to do what I want it to do (at least without touching the NDK). Is there any way I can use the API to do something like this? Or alternatively, what would be the best way to go about this, if it's even feasible on cell phone hardware (which seems to keep getting more and more powerful, strangely...)?

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  • AudioRecord - empty buffer

    - by Arxas
    I' m trying to record some audio using AudioRecord class. Here is my code: int audioSource = AudioSource.MIC; int sampleRateInHz = 44100; int channelConfig = AudioFormat.CHANNEL_IN_MONO; int audioFormat = AudioFormat.ENCODING_PCM_16BIT; int bufferSizeInShorts = 44100; int bufferSizeInBytes = 2*bufferSizeInShorts; short Data[] = new short[bufferSizeInShorts]; Thread recordingThread; AudioRecord audioRecorder = new AudioRecord(audioSource, sampleRateInHz, channelConfig, audioFormat, bufferSizeInBytes); @Override protected void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.activity_main); } @Override public boolean onCreateOptionsMenu(Menu menu) { getMenuInflater().inflate(R.menu.activity_main, menu); return true; } public void startRecording(View arg0) { audioRecorder.startRecording(); recordingThread = new Thread(new Runnable() { public void run() { while (Data[bufferSizeInShorts-1] == 0) audioRecorder.read(Data, 0, bufferSizeInShorts); } }); audioRecorder.stop(); } Unfortunately my short array is empty after the recording is over. May I kindly ask you to help me figure out what's wrong?

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  • How to control access to third party HTML pages

    - by Wylie
    Hello, We have a Learning Management System (LMS) that runs on its own server (IIS/Server 2003). Students must login with Forms authentication to gain access to the content. We want to offer access to third party flash and audio that is embedded in HTML pages hosted on the third party server (IIS/Server 2003). Currently we use a frame in a pop-up window that is populated via a simple URL to the third party HTML pages. How can the third party control access to their content, so that only students who launch the pop-up windows from our site can access their content? Since the content is mostly video and flash, we would prefer not to stream all of their content through our server to the Student. We have a programming staff, so we could maybe... - either post or get for our HTTP request to the third party server - we could use SSL - we could programmatically assign a global NT user account to all of our users and then do some kind of Active Directory login from the LMS server to the third party server - could the third party content be hosted at Amazon S3? Would this allow for secure access/download? These are just ideas. We really have no idea. Any suggestions would be greatly appreciated. TIA, Wylie

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  • download large files using servlet

    - by niks
    I am using Apache Tomcat Server 6 and Java 1.6 and am trying to write large mp3 files to the ServletOutputStream for a user to download. Files are ranging from a 50-750MB at the moment. The smaller files aren't causing too much of a problem but with the larger files it and getting socket exception broken pipe. File fileMp3 = new File(objDownloadSong.getStrSongFolder() + "/" + strSongIdName); FileInputStream fis = new FileInputStream(fileMp3); response.setContentType("audio/mpeg"); response.setHeader("Content-Disposition", "attachment; filename=\"" + strSongName + ".mp3\";"); response.setContentLength((int) fileMp3.length()); OutputStream os = response.getOutputStream(); try { int byteRead = 0; while ((byteRead = fis.read()) != -1) { os.write(byteRead); } os.flush(); } catch (Exception excp) { downloadComplete = "-1"; excp.printStackTrace(); } finally { os.close(); fis.close(); }

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  • 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!

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