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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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  • 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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  • Python Imaging: YCbCr problems

    - by daver
    Hi, I'm doing some image processing in Python using PIL, I need to extract the luminance layer from a series of images, and do some processing on that using numpy, then put the edited luminance layer back into the image and save it. The problem is, I can't seem to get any meaningful representation of my Image in a YCbCr format, or at least I don't understand what PIL is giving me in YCbCr. PIL documentation claims YCbCr format gives three channels, but when I grab the data out of the image using np.asarray, I get 4 channels. Ok, so I figure one must be alpha. Here is some code I'm using to test this process: import Image as im import numpy as np pengIm = im.open("Data\\Test\\Penguins.bmp") yIm = pengIm.convert("YCbCr") testIm = np.asarray(yIm) grey = testIm[:,:,0] grey = grey.astype('uint8') greyIm = im.fromarray(grey, "L") greyIm.save("Data\\Test\\grey.bmp") I'm expecting a greyscale version of my image, but what I get is this jumbled up mess: http://i.imgur.com/zlhIh.png Can anybody explain to me where I'm going wrong? The same code in matlab works exactly as I expect.

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  • Detect aborted connection during ASIO request

    - by Tim Sylvester
    Is there an established way to determine whether the other end of a TCP connection is closed in the asio framework without sending any data? Using Boost.asio for a server process, if the client times out or otherwise disconnects before the server has responded to a request, the server doesn't find this out until it has finished the request and generated a response to send, when the send immediately generates a connection-aborted error. For some long-running requests, this can lead to clients canceling and retrying over and over, piling up many instances of the same request running in parallel, making them take even longer and "snowballing" into an avalanche that makes the server unusable. Essentially hitting F5 over and over is a denial-of-service attack. Unfortunately I can't start sending a response until the request is complete, so "streaming" the result out is not an option, I need to be able to check at key points during the request processing and stop that processing if the client has given up.

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  • What is the best way for communication between cluster nodes

    - by Tom
    I have an application written in a combination of ASP/VB6/VBScript and ASP.NET/C# that consists of a website part, SOAP-like webservice part and a queue processing part processing incoming files in a hotfolder. We are used to running under load balancers (Microsoft or other make). Often we need to communicate between the different load balanced servers. Currently we do this through the SQL Server database that is common for all nodes, however, this comes with a performance penalty as each message requires a transaction and continual polling from the other nodes. What would be better ways to achieve this? Tom, Appelby

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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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  • Choosing a distributed shared memory solution

    - by mindas
    I have a task to build a prototype for a massively scalable distributed shared memory (DSM) app. The prototype would only serve as a proof-of-concept, but I want to spend my time most effectively by picking the components which would be used in the real solution later on. The aim of this solution is to take data input from an external source, churn it and make the result available for a number of frontends. Those "frontends" would just take the data from the cache and serve it without extra processing. The amount of frontend hits on this data can literally be millions per second. The data itself is very volatile; it can (and does) change quite rapidly. However the frontends should see "old" data until the newest has been processed and cached. The processing and writing is done by a single (redundant) node while other nodes only read the data. In other words: no read-through behaviour. I was looking into solutions like memcached however this particular one doesn't fulfil all our requirements which are listed below: The solution must at least have Java client API which is reasonably well maintained as the rest of app is written in Java and we are seasoned Java developers; The solution must be totally elastic: it should be possible to add new nodes without restarting other nodes in the cluster; The solution must be able to handle failover. Yes, I realize this means some overhead, but the overall served data size isn't big (1G max) so this shouldn't be a problem. By "failover" I mean seamless execution without hardcoding/changing server IP address(es) like in memcached clients when a node goes down; Ideally it should be possible to specify the degree of data overlapping (e.g. how many copies of the same data should be stored in the DSM cluster); There is no need to permanently store all the data but there might be a need of post-processing of some of the data (e.g. serialization to the DB). Price. Obviously we prefer free/open source but we're happy to pay a reasonable amount if a solution is worth it. In any way, paid 24hr/day support contract is a must. The whole thing has to be hosted in our data centers so SaaS offerings like Amazon SimpleDB are out of scope. We would only consider this if no other options would be available. Ideally the solution would be strictly consistent (as in CAP); however, eventual consistence can be considered as an option. Thanks in advance for any ideas.

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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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  • 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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  • HSM - cryptoki - opening sessions overhead

    - by Raj
    I am having a query regarding sessions with HSM. I am aware that there is an overhead if you initialise and finalise the cryptoki api for every file you want to encrypt/decrypt. My queries are, Is there an overhead in opening and closing individual sessions for every file, you want to encrypt/decrypt.(C_Initialize/C_Finalize) How many maximum number of sessions can i have for a HSM simultaneously, with out affecting the performance? Is opening and closing the session for processing individual files the best approach or opening a session and processing multiple files and then closing the session the best approach? Thanks

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  • php variable scope

    - by Illes Peter
    I have two files: index.php /lib/user.php Index contains the form: <div class="<? echo $msgclass; ?>"> <? echo $msg; ?> </div> <form id="signin" action="/lib/user.php" method="post"> ... </form> User.php makes all the processing. It sets $msg to 'some error message' and $msgalert to 'error' in case of any error. At the end of processing it uses header() to redirect to index.php But after redirection $msg and $msgalert get out of scope and index only gets empty vars. How can i fix this?

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  • Is there a way to change the maximum width of a window without using the WM_GETMINMAXINFO message?

    - by David
    I want to change the imposed Windows maximum width that a window can be resized to, for an external application's window (not my C#/WinForms program's window). The documentation of GetSystemMetrics for SM_CXMAXTRACK says: "The default maximum width of a window that has a caption and sizing borders, in pixels. This metric refers to the entire desktop. The user cannot drag the window frame to a size larger than these dimensions. A window can override this value by processing the WM_GETMINMAXINFO message." Is there a way to modify this SM_CXMAXTRACK value (either system wide or for one particular window), without processing the WM_GETMINMAXINFO message? Maybe an undocumented function, a registry setting, etc.? (Or: The documentation for MINMAXINFO.ptMaxTrackSize says: "This value is based on the size of the virtual screen and can be obtained programmatically from the system metrics SM_CXMAXTRACK and SM_CYMAXTRACK." Maybe there is a way to change the size of the virtual screen?) Thank you

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  • Perl, efficient parsing of csv file

    - by Mike
    I'm working on a project that involves parsing a large csv formatted file in Perl and am looking to make things more efficient. My approach has been to split() the file by lines first, and then split() each line again by commas to get the fields. But this suboptimal since at least two passes on the data are required. (once to split by lines, then once again for each line). This is a very large file, so cutting processing in half would be a significant improvement to the entire application. My question is, what is the most time efficient means of parsing a large CSV file using only built in tools? note: Each line has a varying number of tokens, so we can't just ignore lines and split by commas only. Also we can assume fields will contain only alphanumeric ascii data (no special characters or other tricks). Also, i don't want to get into parallel processing, although that might work effectively.

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  • How to connect to a network of activemq brokers from a client application?

    - by subh
    I have setup a network of brokers in activemq, how do i connect to that from my client application I tried with network:static:(tcp://master1.IP:61616,tcp://master2.IP:61617) and but I get the following exception javax.jms.JMSException: Uncategorized exception occured during JMS processing; nested exception is javax.jms.JMSException: Could not create Transport. Reason: java.io.IOException: Transport scheme NOT recognized: [network]; With static:(tcp://master1.IP:61616,tcp://master2.IP:61617) I get exception javax.jms.JMSException: Uncategorized exception occured during JMS processing; nested exception is javax.jms.JMSException: Could not create Transport. Reason: java.io.IOException: Transport scheme NOT recognized: [static]; Thanks

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  • Writing my own iostream utility class: Is this a good idea?

    - by Alex
    I have an application that wants to read word by word, delimited by whitespace, from a file. I am using code along these lines: std::istream in; string word; while (in.good()) { in>>word; // Processing, etc. ... } My issue is that the processing on the words themselves is actually rather light. The major time consumer is a set of mySQL queries I run. What I was thinking is writing a buffered class that reads something like a kilobyte from the file, initializes a stringstream as a buffer, and performs extraction from that transparently to avoid a great many IO operations. Thoughts and advice?

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  • Why does addSubview load the view asynchronously

    - by moshe
    I have a UIView that I want to load when the user clicks a button. There happens to be some data processing that happens as well after I call addSubview that involves parsing an XML file retrieved from the web. The problem is the view doesn't show up until after the data processing even if addSuview is called first. I think I'm missing something here, can anyone help? Code: I have a "Loading..." view I'm adding as a custom modal (meaning I'm not using the modalViewController). This action is linked to a button in the navigationController. - (IBAction)parseXml:(id)sender { LoadingModalViewController *loadingModal = [[LoadingModalViewController alloc] initWithNibName:@"LoadingModalViewController" bundle:nil]; [navigationController.view addSubview:loadingModal.view]; [xmlParser parse]; }

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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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  • 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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  • 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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  • 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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