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• Cross-Platform Language + GUI Toolkit for Prototyping Multimedia Applications

  - by msutherl

  I'm looking for a language + GUI toolkit for rapidly prototyping utility applications for multimedia installations. I've been working with Max/MSP/Jitter for many years, but I'd like to add a text-based language to my 'arsenal' for tasks apart from 'content production'. (When it comes to actual media synthesis, my choices are clear [SuperCollider + MSP for audio, Jitter + Quartz + openFrameworks for video]). I'm looking for something that maintains some of the advantages of Max, but is lower-level, faster, more cross-platfrom (Linux support), and text-based. Integration with powerful sound/video libraries is not a requirement. Some requirements: Cross-platform (at least OSX and Linux, Windows is a plus) Fast and easy cross-platform GUIs with no platform-specific modification GUI code separated from backend code as much as possible Good for interfacing with external serial devices (micro-controllers) Good network support (UDP/TCP) Good libraries for multi-media (video, sound, OSC) are a plus Asynchronous synchronous UNIX integration is a plus The options that come to mind: AS3/Flex (not a fan of AS3 or the idea of running in the Flash Player) openFrameworks (C++ framework, perhaps a bit too low level [looking for fast development time] and biased toward video work) Java w/ Processing libraries (like openFrameworks, just slower) Python + Qt (is Qt appropriate for rapid prototyping?) Python + Another GUI toolkit SuperCollider + Swing (yucky GUI development) Java w/ SWT Any other options? What do you recommend?

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• Cocos2d shake/accelerometer issue.

  - by Ryan Poolos

  So I a little backstory. I wanted to implement a particle effect and sound effect that both last about 3 sec or so when the user shakes their iDevice. But first issue arrived when the build in UIEvent for shakes refused to work. So I took the advice of a few Cocos veterans to just use some script to get "violent" accelerometer inputs as shakes. Worked great until now. The problem is that if you keep shaking it just stacks the particle and sounds over and over. Now this wouldn't be that big of a deal except it happens even if you are careful to try and not do so. So what I am hoping to do is disable the accelerometer when the particle effect/sound effect start and then reenable it as soon as they finish. Now I don't know if I should do this by schedule, NStimer, or some other function. I am open to ALL suggestions. here is my current "shake" code. - (void)accelerometer:(UIAccelerometer *)accelerometer didAccelerate:(UIAcceleration *)acceleration { const float violence = 1; static BOOL beenhere; BOOL shake = FALSE; if (beenhere) return; beenhere = TRUE; if (acceleration.x > violence * 1.5 || acceleration.x < (-1.5* violence)) shake = TRUE; if (acceleration.y > violence * 2 || acceleration.y < (-2 * violence)) shake = TRUE; if (acceleration.z > violence * 3 || acceleration.z < (-3 * violence)) shake = TRUE; if (shake) { id particleSystem = [CCParticleSystemQuad particleWithFile:@"particle.plist"]; [self addChild: particleSystem]; // Super simple Audio playback for sound effects! [[SimpleAudioEngine sharedEngine] playEffect:@"Sound.mp3"]; shake = FALSE; } beenhere = FALSE; }

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• regex split and extract multiple parts from a string

  - by nLL

  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 different string objects instead of single

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• .NET Speech recognition plugin Runtime Error: Unhandled Exception. What could possibly cause it?

  - by manuel

  I'm writing a plugin (dll file) for speech recognition, and I'm creating a WinForm as its interface/dialog. When I run the plugin and click the 'Speak' to start the initialization, I get an unhandled exception. Here is a piece of the code: 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() { if (System::Threading::Thread::CurrentThread->GetApartmentState() != System::Threading::ApartmentState::STA) { throw gcnew InvalidOperationException("UI thread required"); } //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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• Toggling between instances of NiftyPlayer on a page - won't stop playing when hidden on IE

  - by Ashley

  Hi, i've got a page with links to MP3s, when the link is clicked I use javascript to show a small Flash player (NiftyPlayer) under the link. When a different link is clicked, the old player is hidden and the new player is revealed. The player auto-starts when the element is shown, and auto-stops when hidden - in Firefox. In IE it will only auto-start and NOT auto-stop. This is what I would like to solve. This is an example HTML with link and player <a href="Beat The Radar - Misunderstood What You Said.mp3" onclick="toggle_visibility('player662431');return false;" class="mp3caption">Misunderstood What You Said</a> <div id="player662431" class="playerhide"><embed src="http://www.xxx.com/shop/flash/player.swf?file=/mp3/Beat The Radar - Misunderstood What You Said.mp3&as=1" quality="high" bgcolor="#000000" width="161" height="13" name="niftyPlayer662431" align="" type="application/x-shockwave-flash" swLiveConnect="true" pluginspage="http://www.macromedia.com/go/getflashplayer"></embed> Here is the javascript (i've got jquery installed to let me hide all the open players on this page apart from the new one) function toggle_visibility(id) { $('.playerhide').hide(); var e = document.getElementById(id); e.style.display = 'block'; } I think what I need to do is start the player manually with javascript (rather than using the autostart as=1 function in the URL string) There is some javascript that comes with NiftyPlayer to allow this EG niftyplayer('niftyPlayer1').play() there is also a stop method. I need some help with javascript - how do I add this call to play into my toggle_visibility function (it has the same unique ID number added to the name of the player as the ID of the div that's being shown, but I don't know how to pull this ID number out of one thing and put it in another) I also would like to be able to do niftyplayer('niftyPlayer1').stop() to stop the audio of the previously running player. Is it possible to store the current ID number somewhere and call it back when needed? Thanks for the help, i'm a PHP programmer who needs some support with Javascript - I know what I want to achieve, just don't know the commands to do it! Thanks

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• .net real time stream processing - needed huge and fast RAM buffer

  - by mack369

  The application I'm developing communicates with an digital audio device, which is capable of sending 24 different voice streams at the same time. The device is connected via USB, using FTDI device (serial port emulator) and D2XX Drivers (basic COM driver is to slow to handle transfer of 4.5Mbit). Basically the application consist of 3 threads: Main thread - GUI, control, ect. Bus reader - in this thread data is continuously read from the device and saved to a file buffer (there is no logic in this thread) Data interpreter - this thread reads the data from file buffer, converts to samples, does simple sample processing and saves the samples to separate wav files. The reason why I used file buffer is that I wanted to be sure that I won't loose any samples. The application doesn't use recording all the time, so I've chosen this solution because it was safe. The application works fine, except that buffered wave file generator is pretty slow. For 24 parallel records of 1 minute, it takes about 4 minutes to complete the recording. I'm pretty sure that eliminating the use of hard drive in this process will increase the speed much. The second problem is that the file buffer is really heavy for long records and I can't clean this up until the end of data processing (it would slow down the process even more). For RAM buffer I need at lest 1GB to make it work properly. What is the best way to allocate such a big amount of memory in .NET? I'm going to use this memory in 2 threads so a fast synchronization mechanism needed. I'm thinking about a cycle buffer: one big array, the Bus Reader saves the data, the Data Interpreter reads it. What do you think about it? [edit] Now for buffering I'm using classes BinaryReader and BinaryWriter based on a file.

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• How to control the system volume using javascript

  - by Geetha

  Hi, I am using media player to play audio and video. I am creating own button to increase and decrease the volume of the media player. working fine too. Problem: Even after reaches 0% volume its audible. If the player volume increase the system volume also be increased. Is it possible. How to achieve this task. Control: <object id="mediaPlayer" classid="clsid:22D6F312-B0F6-11D0-94AB-0080C74C7E95" codebase="http://activex.microsoft.com/activex/controls/mplayer/en/nsmp2inf.cab#Version=5,1,52,701" height="1" standby="Loading Microsoft Windows Media Player components..." type="application/x-oleobject" width="1"> <param name="fileName" value="" /> <param name="animationatStart" value="true" /> <param name="transparentatStart" value="true" /> <param name="autoStart" value="true" /> <param name="showControls" value="true" /> <param name="volume" value="70" /> </object> Code: function decAudio() { if (document.mediaPlayer.Volume >= -1000) { var newVolume = document.mediaPlayer.Volume - 100; if (newVolume >= -1000) { document.mediaPlayer.Volume = document.mediaPlayer.Volume - 100; } else { document.mediaPlayer.Volume = -1000; } } }

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• LGPL library with plugins of varied licenses

  - by Chris

  Note: "Plugins" here refers to shared objects that are accessed via dlopen() and friends. I'm writing a library that I'm planning on releasing under the LGPL. Its functionality can be extended (supporting new audio file formats, specifically) through plugins. I'm planning on creating an exception to the LGPL for this library so that plugins can be released under any license. So far so good. I've written a number of plugins already, some of which use LGPL and some of which use GPL libraries. I'm wary of releasing them with the main library, however, due to licensing issues. The LGPL-based ones would generally be fine, but for my "any license" clause. Would distributing these LGPL-based plugins with the library require the consent of the other license holders to create this exception? Along the same lines, would the inclusion of GPL-based plugins with my library force the whole thing to go GPL? I could also release the plugins separately. The advantage, I presume, is that the plugins an d library will now not be distributed together, creating more separation. But this seems to be no different, really, in the end. Boiled down: Can I include, with my LGPL library, plugins of varied licenses? If not, is it really any different releasing them separately? And if so, there's no real need to create an exception for non-LGPL plugins, is there? It's LGPL or nothing. I'd prefer asking a lawyer, of course, but this is just a hobby and I can't afford to hire a lawyer when I don't expect or want monetary compensation. I'm just hoping others have been in similar situations and have insight.

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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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• Android SDK: hello world does not run

  - by Alex

  I have installed Java x64, Eclipse Classic Judo x64 + ADT Pluggin. OS win 7 x64. I did installation everything according to the manual. Then created first application and launched it. Emulator was launched but hello world was not. I have not idea what doing wrong. Do anyone knows of such error and my problem as a whole? thx Console log: [2012-10-06 13:35:42 - test] ------------------------------ [2012-10-06 13:35:42 - test] Android Launch! [2012-10-06 13:35:42 - test] adb is running normally. [2012-10-06 13:35:42 - test] Performing com.example.test.MainActivity activity launch [2012-10-06 13:35:42 - test] Automatic Target Mode: launching new emulator with compatible AVD 'AVD_41' [2012-10-06 13:35:42 - test] Launching a new emulator with Virtual Device 'AVD_41' [2012-10-06 13:35:42 - Emulator] Failed to create Context 0x3005 [2012-10-06 13:35:42 - Emulator] emulator: WARNING: Could not initialize OpenglES emulation, using software renderer. [2012-10-06 13:35:42 - Emulator] WARNING: Data partition already in use. Changes will not persist! [2012-10-06 13:35:42 - Emulator] WARNING: SD Card image already in use: C:\Users\Zewisa\.android\avd\AVD_41.avd/sdcard.img [2012-10-06 13:35:42 - Emulator] WARNING: Cache partition already in use. Changes will not persist! [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] emulator: warning: opening audio input failed [2012-10-06 13:35:42 - Emulator]

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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 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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• 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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• 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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• 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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• 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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• 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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• 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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• 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 extract part of the path and the ending file name with Regex?

  - by brasofilo

  I need to build an associative array with the plugin name and the language file it uses in the following sequence: /whatever/path/length/public_html/wp-content/plugins/adminimize/languages/adminimize-en_US.mo /whatever/path/length/public_html/wp-content/plugins/audio-tube/lang/atp-en_US.mo /whatever/path/length/public_html/wp-content/languages/en_US.mo /whatever/path/length/public_html/wp-content/themes/twentyeleven/languages/en_US.mo Those are the language files WordPress is loading. They are all inside /wp-content/, but with variable server paths. I'm looking only for those inside the plugins folder, grab the plugin folder name and the filename. Hipothetical case in PHP, where reg_extract_* functions are the parts I'm missing: $plugins = array(); foreach( $big_array as $item ) { $folder = reg_extract_folder( $item ); if( 'plugin' == $folder ) { // "folder-name-after-plugins-folder" $plugin_name = reg_extract_pname( $item ); // "ending-mo-file.mo" $file_name = reg_extract_fname( $item ); $plugins[] = array( 'name' => $plugin_name, 'file' => $file_name ); } } [update] Ok, so I was missing quite a basic function, pathinfo... :/ No problem to detect if /plugins/ is contained in the array. But what about the plugin folder name?

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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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• 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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• 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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• 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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• Creating a web application that can be extended by plugins/modules

  - by Adam Pope

  I'm currently involved with developing a C# CMS-like web application which will be used to standardise our development of websites. From the outset, the idea has been to keep the core as simple as possible to avoid the complexity and menu/option overload that blights many CMS systems. This simple core is now complete and working very well. We envisisaged that the system would be able to accept plugins or modules which would extend the core functionality to suit a given projects needs. These would also be re-usable across projects. For example, a basic catalogue and shopping basket might be needed. All the code for such extensions should be in seperate assemblies. They should be able to provide their own admin interfaces and front-end code from this library. The system should search for available plugins and give the admin user the option to enable/disable the feature. (This is all very much like WordPress plugins) It is crucial that we attack this problem in the correct way, so I'm trying to perform as much due dilligence as possible before jumping in. I am aware of the Plugin Pattern (http://msdn.microsoft.com/en-us/library/ms972962.aspx) and have read some articles on it's use. It seems reasonable but I'm not convinced it's necessarily the correct/best technique for this situation. It seems more suited to processing applications (image/audio manipulation, maths etc). Are there any other options for achieving this kind of UI extensibility functionality? Or is the plugin pattern the way to go? I'd also be interested if anybody has links to articles that explain using the plugin pattern for this purpose?

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