Search Results

Search found 3993 results on 160 pages for 'audio'.

Page 91/160 | < Previous Page | 87 88 89 90 91 92 93 94 95 96 97 98  | Next Page >

• 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?

  Read the article

• 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(); } }

  Read the article

• 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

  Read the article

• 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

  Read the article

• 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

  Read the article

• Android Application

  - by devang

  Hey i want to develop a player on Android, which can play streaming audio and video.how should i start about?

  Read the article

• how code a Image button in PyQt?

  - by Alquimista

  Im trying to do simple audio player, but y want use a image(icon) as a pushbutton.

  Read the article

• 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!

  Read the article

• 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.

  Read the article

• 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!

  Read the article

• Where to get live video streaming examples ( GStreamer )?

  - by Ole Jak

  Where to get live video + audio streaming examples ( GStreamer )? So for example streaming from File or Web camera to some web address

  Read the article

• 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.

  Read the article

• 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

  Read the article

• 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?

  Read the article

• 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?

  Read the article

• 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.

  Read the article

• 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...

  Read the article

• 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.

  Read the article

• 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?

  Read the article

• 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!

  Read the article

• Multimedia content in REST responce(XML/JSON)

  - by Koushik

  In my thesis I need to test different architectures. A request to a REST web service developed using Apache CXF and Spring MVC with MySQL as back end serving references(a field in database) to images,audio and video files stored in file system. In the response message, what is the best method to send the content to the client(another application using the service which I developed). URI: http://www.filmservices.com/film/{id} A client here is not the end user. Send the encoded hyperlink's(where the content is stored in the file system) to the client, so that the client renders the response and displays it to the browser. Use Base64 to encode the message(image,audio,video) and send it to the client. Main concern is performance.

  Read the article

• VLC helper protocol on Mac OS X

  - by Preben

  Hey everybody, I am trying to add a vlc:// helper protocol on Mac OS X. To register the protocol, I have unsuccessfully been playing around with the MoreInternet PrefPane. What I want to have in my browser is a vlc://someressource.com/audio.mp3, which should launch VLC and add http://someressource.com/audio.mp3 to the playlist (this works fine on Windows and also Linux if I remember correctly). Maybe even just have vlc://http:// so that https would also be supported. I have no idea how to achieve this. I tried making a bash script, which MoreInternet would not accept. Then I tried making an application through Automator with my Bash script embedded. That did not work either, as the Automator application has no "creator code" - whatever that is?! Can any of you guys point me in the right direction? Thanks in advance!

  Read the article

• What does this error mean when using OpenAL in the iPhone Simulator?

  - by mystify

  I'm getting this in the console, when creating my OpenAL Sources and Buffers: Cannot find executable for CFBundle/CFPlugIn 0xf530d0 </Library/Audio/Plug-Ins/HAL/Hear.plugin> (not loaded) 2010-05-05 17:11:13.934 Testproj[43173:207] Cannot find function pointer HearCFPlugInFactory for factory 5268FAAB-0147-4272-93FD-4D60A2433C1C in CFBundle/CFPlugIn 0xf430d0 </Library/Audio/Plug-Ins/HAL/Hear.plugin> (not loaded) However, the sounds play nicely. I think HAL is not available on the iPhone, it's just on Mac OS X, right? Do you guys also get that error in the simulator when using OpenAL?

  Read the article

• Ripping a CD to mp3 in C# - third party component or api out there?

  - by Jonathan Williamson

  We're working on a project that requires the ripping of audio tracks from CDs to MP3s (ideally also retrieving the track information from CDDB or similar). More background information: Various music labels send us CDs of music which we then deliver to people via an online delivery system. We're looking at automating the process of converting those CDs into MP3s with full track information where possible. We want to produce a simple desktop application that allows a member of editorial staff to setup the information about the new music we receive. To streamline the process we'd like to include the ripping of the audio and retrieval of the track information.

  Read the article

• Controlling Windows Media Player through PHP / batch files

  - by Duroth

  I'm currently writing a tiny webapp for my HTPC (actually, a PC serving as both a media player and web- / fileserver) that will allow me to remotely control the playing of audio, without having to turn on my TV just to switch songs. I'm using Windows Media Player as my audio player of choice, and I thought I could control it through PHP's COM Class. Unfortunately, I've not been able to find any documentation or examples on controlling WMP through this interface. Can anyone point me in the right direction here? A second (and much less preferable) solution would be to use PHP's exec() call to start batch files that, in turn, control WMP.

  Read the article

< Previous Page | 87 88 89 90 91 92 93 94 95 96 97 98  | Next Page >