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  • Web P2P video confrence solution

    - by dtroy
    I'm looking for the best possible solution which will allow me to incorporate live video/audio conference between 2 users(only 2 at this point) into a flash gaming platform. The video chat is not just an extra feature, it's the main one. I'm mainly looking at open source implementations or something I'll be able to implement myself, but will consider commercial products if they are exactly what I need. Here are a few things I've looked at, but so far, I didn't find any of them good enough: Flash player 10's P2P capabilities sound promising, but I am aware of the fact that Adobe has not release any information on the RTMFP protocol and that there is no commercial server which supports it at this point. Stream all the video/audio live through a flash server (not p2p), but from my personal experience you don't get a smooth conversation. I think TokBox uses this method Java applets are a possible solution too (to perform p2p), but I don't think it will be a nice and elegant solution to combine them in the game at this point (and requires the user to authorize them). BTW, I couldn't find any useful implementations. So, If you know of any, i'll look into them. Google Gmail Video Chat uses a custom (and proprietary) browser plug-in which does the p2p and streams the video/audio into the flash player. This is a possible solution, but I rather not implement the entire p2p protocol stack + browser plug-in at this stage and concentrate on other aspect of the game itself. I think they are using XMPP based protocol similar to Jingle and they've release a Jingle librarby but without the video confrencing implementation. EDIT: In response to Branden: I am aware of Adobe Stratus. Stratus is a beta, hosted rendezvous service that aids establishing communications between Flash Player endpoints (RTMFP server). This current release of the Stratus is prerelease and is designed for evaluation purposes only. The service is not final. There is no guarantee that the service will continue to exist in the future or any information about the future cost. That's why I don't think it can be used as a commercial solution. At least not yet. I'd appreciate your suggestions and advice. thanks!

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  • How to get mp3 files to play in iPhone Safari web browser?

    - by grrussel
    How can I get an MP3 audio file to play in iPhone Safari (OS 3.1)? Currently, I am generating HTML e.g. <a href="file.mp3"><img src="sound.png" alt="Play audio"/></a> to play the file on clicking on the nested image. This works on Safari on OSX, but not on the iPhone. There, the content of the file is shown as text, but it does not appear to be a mime-type problem when checked with Live HTTP Headers from Firefox. I have found approaches referenced here. These require the Safari Plugins setting to be on in the preferences, which is why it did not previously work for me.

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  • Can the Windows Media Player COM control play AVI files from memory (instead of from a file)?

    - by MusiGenesis
    I have a C# app, and I'm looking at using the Windows Media Player COM control to play animation and audio. So far, the only way I see of programatically controlling what the control is playing is to set its URL property to point to some file (I assume there's some way to pass in a playlist). Is there any way the WMP can render an AVI that is entirely in-memory, like a MemoryStream or something? If so, can WMP skip from one AVI to the next seamlessly (i.e. no glitch in either the audio or the video as it transitions from one to the next)? If WMP only plays files, is there some way to cue up a list of the files in advance of play start? If so, can WMP be made to skip from one file to the next without a brief interruption? Any knowledge or links to knowledge would be much appreciated.

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  • "Cannot find executable for CFBundle/CFPlugIn" error

    - by Emil
    Cannot find executable for CFBundle/CFPlugIn 0x432bfa0 </Library/Audio/Plug-Ins/HAL/DVCPROHDAudio.plugin> (not loaded) Cannot find function pointer NewPlugIn for factory C5A4CE5B-0BB8-11D8-9D75-0003939615B6 in CFBundle/CFPlugIn 0x432bfa0 </Library/Audio/Plug-Ins/HAL/DVCPROHDAudio.plugin> (not loaded) That's the error I get when I try to run this code: NSString *path = [[NSBundle mainBundle] pathForResource:[arraySubFarts objectAtIndex:indexPath.row] ofType:@"mp3"]; NSURL *file = [[NSURL alloc] initFileURLWithPath:path]; AVAudioPlayer *player = [[AVAudioPlayer alloc] initWithContentsOfURL:file error:nil]; self.player = player; [player prepareToPlay]; [player setDelegate:self]; [self.player play]; Any idea why? :S I have included the needed frameworks, and the code works great, the only thing is this odd Console-message..

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  • My headset doesn't work [closed]

    - by Kristian Flatheim Jensen
    Hello! I am not sure if this question fits on this site :S if it doesn't please let me know and i remove it :) well... here is my problem I just got a steel series headset for christmas(yay!) and i quickly plugged it into my MacBook Pro. The audio output works fine, but i am having issues with the audio input :( i had these kinds of problems before and i think my mac may be broken :( but then i saw this post: http://www.biloca.com/blog/?p=25 and they talked about some power issues to the microphone on the Mac Mini... I did not quite get what they were discussing so i have a simple question :) Do any of you have an idea why my microphone doesn't work? Please help! :) Best Regards Kristian

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  • Saving generated .wav file to server with PHP

    - by bionicOnion
    I am developing a website where users can compose their own music, and the site will generate a .wav file for their creation. This is working correctly (inasmuch as I can play it on the page). However, I would like to save this file to the server to be listened to/downloaded at a later time, and the saved version of the file can no longer be opened and played by the HTML audio tag. What, if anything, must I put into the file besides the file besides the raw data? Instead of setting the src attribute of the audio tag to the location of the file, will I actually need to open it and generate a URI?

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  • Stop method not working

    - by avoq
    Hi everyone , can anybody tell me why the following code doesn't work properly? I want to play and stop an audio file. I can do the playback but whenever I click the stop button nothing happens. Here's the code : Thank you. .................. import java.io.*; import javax.sound.sampled.*; import javax.swing.*; import java.awt.event.*; public class SoundClipTest extends JFrame { final JButton button1 = new JButton("Play"); final JButton button2 = new JButton("Stop"); int stopPlayback = 0; // Constructor public SoundClipTest() { button1.setEnabled(true); button2.setEnabled(false); // button play button1.addActionListener( new ActionListener(){ public void actionPerformed(ActionEvent e){ button1.setEnabled(false); button2.setEnabled(true); play(); }// end actionPerformed }// end ActionListener );// end addActionListener() // button stop button2.addActionListener( new ActionListener(){ public void actionPerformed( ActionEvent e){ //Terminate playback before EOF stopPlayback = 1; }//end actionPerformed }//end ActionListener );//end addActionListener() this.setDefaultCloseOperation(JFrame.EXIT_ON_CLOSE); this.setTitle("Test Sound Clip"); this.setSize(300, 200); JToolBar bar = new JToolBar(); bar.add(button1); bar.add(button2); bar.setOrientation(JToolBar.VERTICAL); add("North", bar); add("West", bar); setVisible(true); } void play() { try { final File inputAudio = new File("first.wav"); // First, we get the format of the input file final AudioFileFormat.Type fileType = AudioSystem.getAudioFileFormat(inputAudio).getType(); // Then, we get a clip for playing the audio. final Clip c = AudioSystem.getClip(); // We get a stream for playing the input file. AudioInputStream ais = AudioSystem.getAudioInputStream(inputAudio); // We use the clip to open (but not start) the input stream c.open(ais); // We get the format of the audio codec (not the file format we got above) final AudioFormat audioFormat = ais.getFormat(); c.start(); if (stopPlayback == 1 ) {c.stop();} } catch (UnsupportedAudioFileException e) { e.printStackTrace(); } catch (IOException e) { e.printStackTrace(); } catch (LineUnavailableException e) { e.printStackTrace(); } }// end play public static void main(String[] args) { //new SoundClipTest().play(); new SoundClipTest(); } }

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  • Open source VideoPlayer / AudioPlayer / MediaPlayer GUI / UI resource available?

    - by steff
    Hi, I'm looking for a user interface for a MediaPlayer which should be able to play video as well as audio files. Furthermore it needs the following things (nothing fancy): TextView for playing time Progress Bar for progress visulization Play/Pause/Stop buttons NO playlist functionality required, the player will only play a single item (that's why I don't need next/previous buttons). It sounds pretty much like the standard audio-player of Android = 2.0. Sure, I could try to find its source code but that would require to tediously check out the entire source. I'm just asking for a more efficient way. Thanks in advance, steff

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  • Found your wavplayer but can't make it work...

    - by ifoks
    Hello man, I was looking for weeks an audio wav flash player and I found your blog where you post your WavPlayer, I download it and place it on my web site. I tried to read a wav file but it can't, I check it with the debug player and i found the problem, it come from that line : FileWav.hx:58 : Wrong RIFF magic! got 1974609456 instead of 0x46464952 But my audio files really are WAV files ! You're the only one who create a wav player you're my only hope ! Please if you see that message write me at [email protected] (e-mail adress), I really need your help on this man ! Thank's

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  • Youtube video embed in WebView and MoviePlayer control

    - by Tronic
    hi, i embed a youtube video into a webview. the problem is: when i pop the current view (which includes the webview) from navigation controller, the the movie itself stops, but the audio is continues running. when i push the view controller on the navigation controller again, i can play the movie newly, but the old audio is still there. my webview code ids_ = [NSArray arrayWithObjects:@"2b84g38Z_60",@"3URx0tM-rMc",@"HZpi-2HVhq0",@"Hhns0DRPI44",@"hRuoxRQ4Q3k",@"lkMXwNBGRA8",@"tXGc6wWIFJo",@"uzGdEn8aW-Q",@"ZAoEBdt8C5M",@"vn8EJqt2BvQ",@"7Z_qRbjG6Ck",@"JspRcxGUijs"@"lM2lcVOh5YU",@"2b84g38Z_60",nil]; int numIds_ = [ids_ count]; NSLog(@"%d", arc4random()%numIds_); NSString *youTubeVideoHTML = [[NSString alloc] initWithFormat:@"<html><head></head><body style=\"margin:0;background-color:#000;\"><iframe class=\"youtube-player\" type=\"text/html\" width=\"640\" height=\"365\" src=\"http://www.youtube.com/embed/%@\" frameborder=\"0\"></iframe></body></html>", [ids_ objectAtIndex:arc4random()%numIds_]]; NSLog(youTubeVideoHTML); [youtubeView loadHTMLString:youTubeVideoHTML baseURL:nil]; thanks in advance

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  • Properly trimming PCM data from a ByteArray

    - by Lowgain
    I have a situation where I need to trim a small amount of audio from the beginning of a recorded clip (generally somewhere between 110-150ms, it is an inconsistent amount). I'm recording in 44100 frequency and 16 bitrate. This is the code I'm using: public function get trimmedData():ByteArray { var ba:ByteArray = new ByteArray(); var bitPosition:uint = 44100 * 16 * (recordGap / 1000); bitPosition -= int(bitPosition % 16); //should keep snapped to nearest sample, I hope ba.writeBytes(_rawData, (bitPosition / 8)); return ba; } This seems to work time-wise, but all the recorded audio gets staticy and gross. Is something off about my rounding? This is the first time I've needed to alter raw PCM data so I'm not sure about the finer details of it. Thanks!

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  • Why does use of H264 in sender/receiver pipelines introduce just HUGE delay?

    - by Serguey Zefirov
    When I try to create pipeline that uses H264 to transmit video, I get some enormous delay, up to 10 seconds to transmit video from my machine to... my machine! This is unacceptable for my goals and I'd like to consult StackOverflow over what I (or someone else) do wrong. I took pipelines from gstrtpbin documentation page and slightly modified them to use Speex: This is sender pipeline: #!/bin/sh gst-launch -v gstrtpbin name=rtpbin \ v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5001 sync=false async=false \ udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ pulsesrc ! audioconvert ! audioresample ! audio/x-raw-int,rate=16000 ! \ speexenc bitrate=16000 ! rtpspeexpay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false async=false \ udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 Receiver pipeline: !/bin/sh gst-launch -v\ gstrtpbin name=rtpbin \ udpsrc caps="application/x-rtp,media=(string)video, clock-rate=(int)90000, encoding-name=(string)H263-1998" \ port=5000 ! rtpbin.recv_rtp_sink_0 \ rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \ udpsrc caps="application/x-rtp,media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" \ port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! rtpspeexdepay ! speexdec ! audioresample ! audioconvert ! alsasink \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5007 sync=false async=false Those pipelines, a combination of H263 and Speex, work fine enough. I snap my fingers near camera and micropohne and then I see movement and hear sound at the same time. Then I changed pipelines to use H264 along the video path. The sender becomes: #!/bin/sh gst-launch -v gstrtpbin name=rtpbin \ v4l2src ! ffmpegcolorspace ! x264enc bitrate=300 ! rtph264pay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5001 sync=false async=false \ udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ pulsesrc ! audioconvert ! audioresample ! audio/x-raw-int,rate=16000 ! \ speexenc bitrate=16000 ! rtpspeexpay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false async=false \ udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 And receiver becomes: #!/bin/sh gst-launch -v\ gstrtpbin name=rtpbin \ udpsrc caps="application/x-rtp,media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264" \ port=5000 ! rtpbin.recv_rtp_sink_0 \ rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \ udpsrc caps="application/x-rtp,media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" \ port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! rtpspeexdepay ! speexdec ! audioresample ! audioconvert ! alsasink \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5007 sync=false async=false This is what happen under Ubuntu 10.04. I didn't noticed such huge delays on Ubuntu 9.04 - the delays there was in range 2-3 seconds, AFAIR.

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  • Enhanced Podcasts and MPMoviePlayerViewController

    - by Ben Robinson
    Hi, This is a bit of an odd/specific one - possibly a bug? I'm using MPMoviePlayerViewController to play a variety of files, including Enhanced Podcasts - these are audio files, but with a slideshow of images, often created using GarageBand. Until (i think) iOS 3.2 they weren't supported at all, now they are and play fine in the iPod app, but in my app the slideshow doesn't start, the full screen movie player opens, and the audio begins, but all I see is the QuickTime logo. If I scrub the track the pictures appear - and will continue to play correctly - but I see nothing if I don't scrub! Any ideas?? On a related note, these files also include a small rectangluar button containing an (i) button on the right hand side - anybody know what it is or should do?! It does nothing for me!

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  • How do I stop Safari from caching my Servlet response?

    - by Cliff
    I'm having trouble testing a web app with Safari. My app returns wave audio data. The problem happens when I change the application and hit it again from Safari. Safari caches the original response so no matter how many times I hit refresh it seems like I've not updated anything. I can almost get around this using force refresh with Firefox but because I'm having trouble generating the wave headers using the javax.sound API Firefox only plays the first second of audio returned. A few weeks ago I tried setting the HTTP header in my servlet to prevent caching but I don't think I was setting it correctly. (What is the header for browser cache control?) This is becoming a real pain and I'm looking for any ideas, comments, or alternative approaches. I'm getting ready to try again but I figured I'd ask here in the interim to see if someone can provide help.

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  • iSightAudio.plugin error when playing video using MediaPlayer

    - by Elisabeth
    I am working on creating a simple iPhone app that plays a movie via URL. When I Build&Run to test in the simulator, it works fine; as soon as I start playing the movie, I get the following message in the console: [1757:4b03] Cannot find executable for CFBundle/CFPlugIn 0x820ffe0 </Library/Audio/Plug-Ins/HAL/iSightAudio.plugin> (not loaded) [1757:4b03] Cannot find function pointer iSightAudioNewPlugIn for factory 9BE7661E-8AEF-11D7-8692-000A959F49B0 in CFBundle/CFPlugIn 0x820ffe0 </Library/Audio/Plug-Ins/HAL/iSightAudio.plugin> (not loaded) I don't get this error on other programs, so I assume it has something to do with this specific program, which uses the MediaPlayer.framework. Does anyone know what is causing this problem and how to fix it? Thank you

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  • How do I get callgrind to dump source line information?

    - by Jeremybub
    I'm trying to profile a shared library on GNU/Linux which does real-time audio processing, so performance is important. I run another program which hooks it up to the audio input and output of my system, and profile that with callgrind. Looking at the results in KCacheGrind, I get great information about what functions are taking up most of my time. However, it won't let me look at the line by line information, and instead says I need to compile it with debugging symbols and run the profiling again. The program which I am profiling is not compiled with debug symbols, but the library is. And I know this, because interestingly, source code annotations for cachegrind work fine. When I run callgrind, it says the default is to dump source line information, but it just isn't doing that. Is there some way I could force it to, or figure out what's stopping it?

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  • How to disable UI control based on domain object's state?

    - by Subb
    Here's my problem. I have a somewhat complex domain object, which, depending on its state, responds to certain actions. I think the state pattern is pretty much the solution for that. However, I need to display which actions are possible at any moment in the UI. Ex: The domain object is an audio player. Some songs can't be skipped (like ads), so I need to disable the "next" and "previous" buttons in the GUI so the user have some kind of feedback of which action he can execute. I've looked at Swing's Action class (note: this is not a Java project), but I think I would need to keep every Actions in my domain object class (audio player), so it can enable or disable them depending on its own state (thus, affecting the UI). Is it the way to do it?

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  • Signal amplitude against time (java)

    - by wsr74ws84
    Hi everyone , I'm racking my brain in order to solve a knotty problem (at least for me) While playing an audio file (using java) I want the signal amplitude to be displayed against time. I mean I'd like to implement a small panel showing a sort of oscilloscope .(SPECTRUM ANALYZER) The audio signal should be viewed in the time domain (vertical axis is amplitude and the horizontal axis is time) Does anyone know how to do it? Is there a good tutorial I can rely on? Since I know vwry little about java , I wish someone could help me . Thanks in advance.

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  • Onclick starts gif animation and .mp3, how to sync across browsers

    - by user2958163
    So I am using a text-based jplayer (http://jplayer.org/latest/demo-04/) that I want to sync with a gif animation. Onclick the text link does two things -1. feed the jplayer an mp3 and 2. trigger an animation (via SwapImage). It is important for these two to start at the same time. Right now, this works perfectly in chrome/firefox but in IE and mobile browsers the audio lags considerably. I have tried with the audio preloaded (it is a small 40K mp3) and it makes no difference. I dont think its a bandwidth problem because the problem is the same on repeat clicks. Any pointers on how I can resolve this...

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  • getAssetFileDescriptor from ZipResourceFile merges all mp3 in mediaplayer SOLVED

    - by Jordi
    I've a program with an Expansion file that stores 4 mp3 in a obb file (zip without compression). I can retrieve the data, but instead of taking the audio file i asked for, it merges ALL audio files in the same AssetFileDescriptor. ---SOLVED--- with the fixes Support class public AssetFileDescriptor getaudio(){ ZipResourceFile expansionFile = APKExpansionSupport.getAPKExpansionZipFile(c,21,21); AssetFileDescriptor afd=null; if(take==1) { afd = expansionFile.getAssetFileDescriptor("file01.mp3"); }else if(take==2 { afd = expansionFile.getAssetFileDescriptor("file02.mp3"); } //more els eif ............ return afd; } In the MediaPlayer class AssetFileDescriptor fd = Llistat.getInstance().getAudio(); mPlayer.setDataSource(fd.getFileDescriptor(), fd.getStartOffset(),fd.getLength()); mPlayer.prepare(); fd.close(); My problem was that i directly was returning and using a FileDescriptor, while i was needing the AssetFileDescriptor to take its StartOffset and Length.

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  • avmutablecomposition insertEmptyTimeRange

    - by smartfaceweb
    I have created an avmutablecomposition and tried to use insertEmptyTimeRange to generate 1 minute of silence. This doesn't appear to be working. I have also tried creating an avmutablecompositiontrack using addMutableTrackWithMediaType:preferredTrackID: and then insertEmptyTimeRange on the track and still no success. To give some background on my app, I allow users to add audio samples a timeline and then playback or export and this is working really well using the av classes. The problem is that I need to make sure that the audio is exactly 1 min (for example). Regardless of the info about my specific app above, is it possible to insert an empty time range into a comp or comptrack?

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  • html5 video player with simplest controls (only play and paus)

    - by mathiregister
    hi guys, somehow there are really little tutorials out there for html5 video and audio playback. I simply want to embed video and audio files with customized controls. However the controls should be farely simple. I only need a play-button. If clicked, play gets replaced by pause. that's all! however i even don't know how to embed/display a video without "preload controls". Somehow if i only set (without preload controls) Firefox even don't shows anything. Chrome does show a black window. I would love to be able to use jquery to control the video play and pause button. Maybe you have some little start-approach for me! thank you very much!

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