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• Embed VLC player in GWT

  - by chrisnfoneur

  Hello, I want to embed a VLC player in my webapp build with Google's GWT. First I had a look at this page: http://wiki.videolan.org/GWT, which offers a nice solution but I add to implements all javascript functions calls (play, stop, fullscreen) with JSNI. Then I found gwt-player (hosted by Google code) which does all the job for me but the annoying part is that the project is not widely used (few posts each month on the project's group, not so many talks about it in blogs/forums...) Do you know another option to easly embed & control a VLC player in a GWT app ? My main goal is to play any video/audio file in a webapp and offer the user a fast/forward feature (set rate in VLC), is there any other player I could use ? I already had a look at Quicktime, Windows Media player & Flowplayer, none of them offers as much features as VLC. Thanks in advance & have a nice new year's eve. Chris

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• Javascript: Mediaplayer and its Progress Bar

  - by Geetha

  Hi All, In my asp.net application i am using mediaplayer to paly the audio and video. i am controling volume using javascript code. I want to display a userdefined progress bar. How to create it. Code: <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="100" /> <param name="loop" value="true" /> </object>

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• No Microphone error on iPod Touch

  - by Bob Vork

  I've build an iPhone app that should work on an iPod Touch as well, but I'm getting reports that the app is not working on iPod touches. It's displaying an error message saying there's no mic available on the device. The thing is, the app does nothing whatsoever with audio, and I can't find anything related in the project settings. The other problem is I don't have an iPod Touch available to test this myself. Are some people running an old firmware version? Am I compiling the wrong firmware version? To my surprise I couldn't find anything about this on SO or Google… Any help is appreciated

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• win7 amd64 guest in kvm does not have sound

  - by davidshen84

  hi, my host system is gentoo amd64, guest system is win 7 amd64. the guest system can work, except it does not have sound. i start kvm with -soundhw ac97, QEMU_AUDIO_DRV='alsa', and after i get into the guest system, i can see a 'Multimedia Audio Controller' in the device manager. but win7 cannot find the driver for it. i searched the network for a long time, and i cannot find a driver for intel ac97 for win7 amd64. i also tried -soundhw sb16, es1370, none of them work. please help me fix this.

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• Is there a DRM scheme that works?

  - by Simon

  We help our clients to manage and publish their media online - images, video, audio, whatever. They always ask my boss whether they can stop users from copying their media, and he asks me, and I always tell him the same thing: no. If the users can view the media, then a sufficiently determined user will always be able to make a copy. But am I right? I've been asked again today, and I promised my boss I'd ask about it online. So - is there a DRM scheme that will work? One that will stop users making copies without stopping legitimate viewing of the media? And if there isn't, how do I convince my boss?

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• How to create playable FLV video from part of FLV file using FFMPEG?

  - by Ole Jak

  So we had real FLV video file. we had devided it into 3 parts (more or less equal, not looking into structure orcontext). We have taken second part and forgot about first 2. Video contained audio and video track. mp3 and on vp6. Is it any how possible to play thsat second part after sending to ffmpeg some command? So how to (using any FFMPEG API (in general in any programming language) or using command line) turn bytearray into playable video? (knowing what format video was created in and some other data like used codecs )

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• fourier transform to transpose key of a wav file

  - by tbischel

  I want to write an app to transpose the key a wav file plays in (for fun, I know there are apps that already do this)... my main understanding of how this might be accomplished is to 1) chop the audio file into very small blocks (say 1/10 a second) 2) run an FFT on each block 3) phase shift the frequency space up or down depending on what key I want 4) use an inverse FFT to return each block to the time domain 5) glue all the blocks together But now I'm wondering if the transformed blocks would no longer be continuous when I try to glue them back together. Are there ideas how I should do this to guarantee continuity, or am I just worrying about nothing?

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• UIWebView/MPMoviePlayerController and the "Done" button

  - by David Sowsy

  I am using the UIWebView to load both streaming audio and video. I have properly set up the UIWebView delegate and I am receiving webViewDidStartLoading and webViewFinishedLoading events perfectly. The webview launches a full screen window (likely a MPMoviePlayerController) Apple's MoviePlayer example gets the array of Windows to determine which window the moviePlayerWindow is for adding custom drawing/getting at the GUI components. I believe this to be a bad practice/hack. My expectation is that I should be able to figure out when that button was clicked by either a delegate method or an NSNotification. It may also be the case that I have to poke around subviews or controllers with isKindOf calls, but I don't think those are correct approaches. Are my expectations incorrect, and if so, why? What is the correct way to bind an action to that "Done" button?

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• ActionController::MethodNotAllowed

  - by Lowgain

  I have a rails model called 'audioclip'. I orginally created a scaffold with a 'new' action, which I replaced with 'new_record' and 'new_upload', becasue there are two ways to attach audio to this model. Going to /audioclips/new_record doesn't work because it takes 'new_record' as if it was an id. Instead of changing this, I was trying to just create '/record_clip' and '/upload_clip' paths. So in my routes.db I have: map.record_clip '/record_clip', :controller => 'audioclips', :action => 'new_record' map.upload_clip '/upload_clip', :controller => 'audioclips', :action => 'new_upload' When I navigate to /record_clip, I get ActionController::MethodNotAllowed Only get, head, post, put, and delete requests are allowed. I'm not extremely familiar with the inner-workings of routing yet. What is the problem here? (If it helps, I have these two statements above map.resources = :audioclips

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• How to make QT support HTML 5 database?

  - by Mickey Shine

  I am using Qt 4.7.1 and embedded a webview in my app. But I got the following error when trying to visit http://webkit.org/demos/sticky-notes/ to test the HTML 5 database feature Failed to open the database on disk. This is probably because the version was bad or there is not enough space left in this domain's quota I compiled my static Qt library with the following command: configure --prefix=/usr/local/qt-static-release-db --accessibility --multimedia --audio-backend --svg --webkit --javascript-jit --script --scripttools --declarative --release -nomake examples -nomake demos --static --openssl -I /usr/local/ssl/include -L /usr/local/ssl/lib -confirm-license -sql-qsqlite -sql-qmysql -sql-qodbc

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• DirectSound affects system volume on WinXP

  - by Anton

  Hello guys, I'm currently developing an audio engine that is used in voice network chat software. Everything is working fine - capture/playback/mixing channels. The problem is in using it under Windows XP. I've been getting user reports with information that their global system volume is set to zero after launching the application. I'm assuming that happens because of WaveOut/DSound conflict. How can I force DSound not to affect system volume? Playback device is initialized: DirectSoundCreate8(&GUID, &pAudio, NULL); and: pAudio-SetCooperativeLevel(parentWnd, DSSCL_PRIORITY); I'm currently not able to debug the application, cause I'm using Vista and everything is OK. Hope you can help me with this issue! Thanks a Lot! Regards, Anton.

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• How do I force one method to be executed before another method?

  - by RexOnRoids

  I've got 2 methods. One method starts playing an audio file (.mp3), the other method updates a UIToolBar to show a button (PLAY or PAUSE). These two methods are called in the following order: //Adds some UIBarButtonItems to a UIToolBar [self togglePlayer]; //Uses AVAudioPlayer [audioPlayer play]; I call the methods in the above order so that the (pause) button will be shown at the time the song starts playing. But, the problem is that the song starts playing first, and the UIToolBar remains unchanged for quite a while (from 2 to 5 secs) until the button is added and shown. What I want is for the button to be shown at the same time the song starts playing (i.e. NO DELAY). Is there any way to do this?

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• Alternative to Rtmp and red5 for Iphone application

  - by IeN

  I am using red5 + rtmp in client-server flash application. There isnt audio/video streams in my applications, rtmp used for transfering messages from app to server and back. Now i need to develop application for Iphone and need help: 1) is there any rtmp implementation on Iphone ?? 2) If not, how could i solve this problem? Is there is any alternative to rtmp on iphone? And most important question : could it be solved without rewriting whole server part of application? (red5+ rtmp) Thanks

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• super dealloc error using multiple table view calsses

  - by padatronic

  I am new to Iphone apps and I am trying to build a tab based app. I am attempting to have a table ontop of an image in both tabs. On tab with a table of audio links and the other tab with a table of video links. This has all gone swimmingly, I have created two viewControllers for the two tables. All the code works great apart from to get it to work I have to comment out the super dealloc in the - (void)dealloc {} in the videoTableViewController for the second tab. If I don't I get the error message: FREED(id): message numberOfSectionsInTableView: sent to freed object please help, i have no idea why it is doing this...

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• AVAudioPlayer Output to Speaker Problem

  - by Max

  After searching around for how to send AVAudioPlayer output to the iPhone's speaker, I found this: http://stackoverflow.com/questions/1064846/iphone-audio-playback-force-through-internal-speaker Despite setting the category correctly to AVAudioSessionCategoryPlayAndRecord, this solution doesn't seem to be working for me and won't even let the build compile, giving me this error: "_AudioSessionSetProperty", referenced from: ... ... ld: symbol(s) not found collect2: ld returned 1 exit status Am I not including something? I'm importing AudioToolbox, AVFoundation, and CoreAudio. My class implements AVAudioSessionDelegate, AVAudioRecorderDelegate, AVAudioPlayerDelegate, and UITextFieldDelegate. Any help would be greatly appreciated!

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• Why does ffmpeg stop randomly in the middle of a process?

  - by acidzombie24

  ffmpeg feels like its taking a long time. I then look at my output file and i see it stops between 6 and 8mbs. A fully encoded file is about 14mb. Why does ffmpeg stop? My code locks up on StandardOutput.ReadToEnd();. I had to kill the process (after seeing it not move for more then 10 seconds when i see it update every second previously) then i get the results of stdout and err. stdout is "" stderr is below. The output msg shows the filesize ended. I also see a drop in my CPU usage when it stops. I copyed the argument from visual studios. CD to the same working directory and ran the cmd (bin/ffmpeg) and pasted the argument. It was able to complete. int soundProcess(string infn, string outfn) { string aa, aa2; aa = aa2 = "DEAD"; var app = new Process(); app.StartInfo.UseShellExecute = false; app.StartInfo.RedirectStandardOutput = true; app.StartInfo.RedirectStandardError = true; //*/ app.StartInfo.FileName = @"bin\ffmpeg.exe"; app.StartInfo.Arguments = string.Format(@"-i ""{0}"" -ab 192k -y {2} ""{1}""", infn, outfn, param); app.Start(); try { app.PriorityClass = ProcessPriorityClass.BelowNormal; } catch (Exception ex) { if (!Regex.IsMatch(ex.Message, @"Cannot process request because the process .*has exited")) throw ex; } aa = app.StandardOutput.ReadToEnd(); aa2 = app.StandardError.ReadToEnd(); app.WaitForExit(); if (aa2.IndexOf("could not find codec parameters") != -1) return 1; else if (aa == "DEAD" || aa2 == "DEAD") return -1; else if (aa2.Length != 0) return -2; else return 0; } The output of stderr. stdout is empty. FFmpeg version SVN-r15815, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --enable-memalign-hack --enable-postproc --enable-swscale --enable-gpl --enable-libfaac --enable-libfaad --enable-libgsm --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libx264 --enable-libxvid --disable-ffserver --disable-vhook --enable-avisynth --enable-pthreads libavutil 49.12. 0 / 49.12. 0 libavcodec 52. 3. 0 / 52. 3. 0 libavformat 52.23. 1 / 52.23. 1 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 6. 1 / 0. 6. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 13 2008 10:28:29, gcc: 4.2.4 (TDM-1 for MinGW) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\dev\src\trunk\prjname\prjname\App_Data/temp/m/o/6304266424778814852': Duration: 00:12:53.36, start: 0.000000, bitrate: 154 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16 Output #0, ipod, to 'C:\dev\src\trunk\prjname\prjname\App_Data\temp\m\o\2.m4a': Stream #0.0(und): Audio: libfaac, 44100 Hz, stereo, s16, 192 kb/s Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding size= 87kB time=4.74 bitrate= 150.7kbits/s size= 168kB time=9.06 bitrate= 151.9kbits/s size= 265kB time=14.28 bitrate= 151.8kbits/s size= 377kB time=20.29 bitrate= 152.1kbits/s size= 487kB time=26.22 bitrate= 152.1kbits/s size= 594kB time=32.02 bitrate= 152.1kbits/s size= 699kB time=37.64 bitrate= 152.1kbits/s size= 808kB time=43.54 bitrate= 152.0kbits/s size= 930kB time=50.09 bitrate= 152.2kbits/s size= 1058kB time=57.05 bitrate= 152.0kbits/s size= 1193kB time=64.23 bitrate= 152.1kbits/s size= 1329kB time=71.63 bitrate= 152.0kbits/s size= 1450kB time=78.16 bitrate= 152.0kbits/s size= 1578kB time=85.05 bitrate= 152.0kbits/s size= 1706kB time=92.00 bitrate= 152.0kbits/s size= 1836kB time=98.94 bitrate= 152.0kbits/s size= 1971kB time=106.25 bitrate= 151.9kbits/s size= 2107kB time=113.57 bitrate= 152.0kbits/s size= 2214kB time=119.33 bitrate= 152.0kbits/s size= 2345kB time=126.39 bitrate= 152.0kbits/s size= 2479kB time=133.56 bitrate= 152.0kbits/s size= 2611kB time=140.76 bitrate= 152.0kbits/s size= 2745kB time=147.91 bitrate= 152.1kbits/s size= 2880kB time=155.20 bitrate= 152.0kbits/s size= 3013kB time=162.40 bitrate= 152.0kbits/s size= 3146kB time=169.58 bitrate= 152.0kbits/s size= 3277kB time=176.61 bitrate= 152.0kbits/s size= 3412kB time=183.90 bitrate= 152.0kbits/s size= 3540kB time=190.80 bitrate= 152.0kbits/s size= 3670kB time=197.81 bitrate= 152.0kbits/s size= 3805kB time=205.08 bitrate= 152.0kbits/s size= 3932kB time=211.93 bitrate= 152.0kbits/s size= 4052kB time=218.38 bitrate= 152.0kbits/s size= 4171kB time=224.82 bitrate= 152.0kbits/s size= 4277kB time=230.55 bitrate= 152.0kbits/s size= 4378kB time=235.96 bitrate= 152.0kbits/s size= 4486kB time=241.79 bitrate= 152.0kbits/s size= 4592kB time=247.50 bitrate= 152.0kbits/s size= 4698kB time=253.21 bitrate= 152.0kbits/s size= 4804kB time=258.95 bitrate= 152.0kbits/s size= 4906kB time=264.41 bitrate= 152.0kbits/s size= 5012kB time=270.09 bitrate= 152.0kbits/s size= 5118kB time=275.85 bitrate= 152.0kbits/s size= 5234kB time=282.10 bitrate= 152.0kbits/s size= 5331kB time=287.39 bitrate= 151.9kbits/s size= 5445kB time=293.55 bitrate= 152.0kbits/s size= 5555kB time=299.40 bitrate= 152.0kbits/s size= 5665kB time=305.37 bitrate= 152.0kbits/s size= 5766kB time=310.80 bitrate= 152.0kbits/s size= 5876kB time=316.70 bitrate= 152.0kbits/s size= 5984kB time=322.50 bitrate= 152.0kbits/s size= 6094kB time=328.49 bitrate= 152.0kbits/s size= 6212kB time=334.76 bitrate= 152.0kbits/s size= 6327kB time=340.99 bitrate= 152.0kbits/s

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• Fast block placement algorithm, advice needed?

  - by James Morris

  I need to emulate the window placement strategy of the Fluxbox window manager. As a rough guide, visualize randomly sized windows filling up the screen one at a time, where the rough size of each results in an average of 80 windows on screen without any window overlapping another. It is important to note that windows will close and the space that closed windows previously occupied becomes available once more for the placement of new windows. The window placement strategy has three binary options: Windows build horizontal rows or vertical columns (potentially) Windows are placed from left to right or right to left Windows are placed from top to bottom or bottom to top Why is the algorithm a problem? It needs to operate to the deadlines of a real time thread in an audio application. At this moment I am only concerned with getting a fast algorithm, don't concern yourself over the implications of real time threads and all the hurdles in programming that that brings. So far I have two choices which I have built loose prototypes for: 1) A port of the Fluxbox placement algorithm into my code. The problem with this is, the client (my program) gets kicked out of the audio server (JACK) when I try placing the worst case scenario of 256 blocks using the algorithm. This algorithm performs over 14000 full (linear) scans of the list of blocks already placed when placing the 256th window. 2) My alternative approach. Only partially implemented, this approach uses a data structure for each area of rectangular free unused space (the list of windows can be entirely separate, and is not required for testing of this algorithm). The data structure acts as a node in a doubly linked list (with sorted insertion), as well as containing the coordinates of the top-left corner, and the width and height. Furthermore, each block data structure also contains four links which connect to each immediately adjacent (touching) block on each of the four sides. IMPORTANT RULE: Each block may only touch with one block per side. The problem with this approach is, it's very complex. I have implemented the straightforward cases where 1) space is removed from one corner of a block, 2) splitting neighbouring blocks so that the IMPORTANT RULE is adhered to. The less straightforward case, where the space to be removed can only be found within a column or row of boxes, is only partially implemented - if one of the blocks to be removed is an exact fit for width (ie column) or height (ie row) then problems occur. And don't even mention the fact this only checks columns one box wide, and rows one box tall. I've implemented this algorithm in C - the language I am using for this project (I've not used C++ for a few years and am uncomfortable using it after having focused all my attention to C development, it's a hobby). The implementation is 700+ lines of code (including plenty of blank lines, brace lines, comments etc). The implementation only works for the horizontal-rows + left-right + top-bottom placement strategy. So I've either got to add some way of making this +700 lines of code work for the other 7 placement strategy options, or I'm going to have to duplicate those +700 lines of code for the other seven options. Neither of these is attractive, the first, because the existing code is complex enough, the second, because of bloat. The algorithm is not even at a stage where I can use it in the real time worst case scenario, because of missing functionality, so I still don't know if it actually performs better or worse than the first approach. What else is there? I've skimmed over and discounted: Bin Packing algorithms: their emphasis on optimal fit does not match the requirements of this algorithm. Recursive Bisection Placement algorithms: sounds promising, but these are for circuit design. Their emphasis is optimal wire length. Both of these, especially the latter, all elements to be placed/packs are known before the algorithm begins. I need an algorithm which works accumulatively with what it is given to do when it is told to do it. What are your thoughts on this? How would you approach it? What other algorithms should I look at? Or even what concepts should I research seeing as I've not studied computer science/software engineering? Please ask questions in comments if further information is needed. [edit] If it makes any difference, the units for the coordinates will not be pixels. The units are unimportant, but the grid where windows/blocks/whatever can be placed will be 127 x 127 units.

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• Publishing SWF using Adobe Flash

  - by Kim

  Hello everyone, I have a SWF file which contains of an image (1keyframe) and also, it contains an AS3 file with the following codes: var loader:Loader=new Loader(); var ur:URLRequest=new URLRequest("1.swf"); loader.load(ur); addChild(loader); so basically, i am trying to play the swf file (1.swf - an audio) while the image is being displayed. What I want to know is how will I be able to publish this project into an SWF file which can still play as expected even without the raw 1.swf file. I can publish SWF right now but when I delete the 1.swf file, my generated swf can only display the image. Help me please. Thanks in advance :)

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• Streaming Media Server and Hosting

  - by Ryan Max

  My partner and I have a webcam site that basically runs the old-school method....Every 0.5 seconds the javascript reloads the image in the browser from the webcam. However we are wanting to upgrade to a streaming media server to get higher quality video, and possibly audio. We aren't tied to any one specific file format or server type, as of right now we are leaning towards slicehost (as scalability is important), and installing darwin streaming server or wowza. This is meant to be a live stream. Does anyone have any suggestions for hosts/server software?

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• How to use Speech 2 Text in Microsoft Surface

  - by Roflcoptr

  I'd like to use some speech 2 text in my microsoft surface application. I saw that it is possible, but I don't really know where to start. Is there any framework/library available, or a code snippet, or a tutorial?? I don't even know exactly what i should google for ;) ===EDIT=== I read that it is necessary to use a grammar to recognize words. So if I want to proceed free text, is there a predefined grammar for the english language? Or is it a better choice to don't use speech2text but just audio files instead?

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• Testing MPMoviePlayerViewController in iPad simulator

  - by hgpc

  I have a view that shows a MPMoviePlayerViewController modally. When testing it in the iPad simulator it works well on the first try. If I dismiss the video and then show the view again, the player only plays the audio, but not the video. Is this a simulator quirk or am I doing something wrong? Here's my code: - (void)viewWillAppear:(BOOL)animated { [super viewWillAppear:animated]; MPMoviePlayerViewController* v = [[MPMoviePlayerViewController alloc] initWithContentURL:url]; [[NSNotificationCenter defaultCenter] addObserver:self selector: @selector(playbackDidFinish:) name:MPMoviePlayerPlaybackDidFinishNotification object:v.moviePlayer]; [self presentMoviePlayerViewControllerAnimated:v]; [v release]; } -(void) playbackDidFinish:(NSNotification*)aNotification { MPMoviePlayerController *player = [aNotification object]; [[NSNotificationCenter defaultCenter] removeObserver:self name:MPMoviePlayerPlaybackDidFinishNotification object:player]; [player stop]; [self dismissMoviePlayerViewControllerAnimated]; }

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• DSP - How are frequency amplitudes modified using DFT?

  - by Trap

  I'm trying to implement a DFT-based equalizer (not FFT) for the sole purpose of learning. To check if it works I took an audio signal, analyzed it and then resynthesized it again with no modifications made to the frequency spectrum. So far so good. Now I tried to silence some frequency bands, just by setting their amplitudes to zero before resynthesis, but definitely it's not the way to go. What I get is a rather distorted signal. I'm using the so-called 'standard way of calculating the DFT' which is by correlation. I first tried to modify the real part amplitudes only, then modifying both the real and imaginary part amplitudes. I also tried to convert the DFT output to polar notation, then modifying the magnitude and convert back to rectangular notation, but none of this is working. Can someone show me what I'm doing wrong? I tried to find info on this subject in the internet but couldn't find any. Thanks in advance.

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• C# How to Present Such Question?

  - by ikurtz

  greetings! i have a C# game program that im developing. it uses sound samples and winsock. when i test run the game most of the audio works fine but from time to time if it is multiple samples being played sequentially the application form shakes a little bit and then goes back to its old position. how do i go about debugging this or present it to you folks in a manageable manner? im sure no one is going to want the whole app code in fear of virus attacks. please guide me.. thanking you. EDIT: i have not been able to pin down any code section that produces this result. it just does and i cannot explain it. EDIT: no the x/y position are not changing. the window like shakes around a few pixels and then goes back to the position were it was before the shake.

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• Videoconference using Flash and SIP

  - by Júlio Santos

  The front-end will be Flash, to run in a browser and have access to the camera. I must use SIP to control the sessions. How could I do this? Will a Red5 server and a MjSip sever do the trick? As in i'd use MjSip to setup the session and warn users about calls, and Red5 to stream the video and audio? Any suggestions? Note: only 1-on-1 conference is required.

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• How can HTML5 "replace" Flash?

  - by Kassini

  A topic of debate that's seen a resurgence since the unveiling of the iPad is the issue of Flash versus HTML5. There are those that suggest that HTML5 will one day supplant/replace Adobe Flash. I do not develop software that runs in a browser, so my (limited) understanding is: HTML is a pure-text markup language that is delivered over HTTP to a client browser. The client browser interprets the markup and renders (with varying degrees of success) the page according to an standard specification. Adobe Flash is a propriety framework for working with audio, video, sound and raster/vector graphics. It requires special authoring tools (a compiler perhaps?) and a custom player that's available as a plug-in to most common browsers. Could someone please explain (to this C/C++ developer) how it is possible from a technical/coding point-of-view that a text-based markup language (HTML5) could be considered a replacement to a multimedia framework (Flash)? Please no opinionated arguments - just technical facts.

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