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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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • Python File Meta Tag reading

    - by Jeff
    Anyone know of a Python module that can pull Tag data from multiple media formats? Trying to build an app that allows for manipulation of ASF (Windows Media Player files, ie WMA, WMV, etc), ID3, including both ID3v1 and ID3v2 (MPEG files, ie MP3), MPEG Audio Bit Stream (ie ABS, MP1, MP2, MP3), MPEG Program Stream (MPEG movies, and DVD and HD DVD video discs, ie MPG, MPEG, VOB, EVO), and ISO Base Media File Format (eg QuickTime, MPEG-4 and iTunes AAC files, ie QT, MOV, MP4, M4A, M4B, M4P, M4V, etc). Don't need ALL of that but just most standard consumer formats like mov and mpeg. I can't seem to find a good module to support that or a library. Any recommendations?

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  • Reading iTunesMovies file in iPhone?

    - by raziiq
    Hi there. In iPhone, the iPod app saves the media files (audio, video) with strange names and in weird folders (F00,F01 etc). There is a file named iTunesMovies in iPhone, which contains all the information about the metadata of those video files and how they are to be displayed in iPod app. I copied that to my Mac also, and when i tried to open that file in textEdit, it showed some alien characters which made me believe that it is encrypted may be(Thats just a wild guess). I want to read/change the contents of that iTunesMovies file. Can i do that? Is there any Framework which deals with that iTunesMovies file? Thanks in advance

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  • How can I find the program making a harmonica sound?

    - by Josh
    A friend has a Windows XP SP3 machine that plays a harmonica sound for about 5 seconds throughout the day at what seems to be random intervals (every couple hours). My question is how can I find the program making this sound? Is there a Windows API hook for monitoring audio access? I've gone through and checked all the standard Windows sounds in the Control Panel and right now the theme is set to no sounds and I personally checked to make sure none of the events have a sound specified. I also checked the Task Scheduler to make sure there wasn't something scheduled to go off every couple hours. Any ideas on how to go about finding the bugger?

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  • Set a OGG in raw folder as Ringtone/Notification?

    - by YaW
    Hi, I have some ogg audios in my raw folder and I'm trying to set one of them as a Ringtone (or Notification, Alarm... whatever). I've been looking at the source code of RingDroid and I can see how is this done using the ContentValues and MediaStore, but in all the examples I've seen, the audio files is in the SDCard. Is it possible to set the ringtone directly from the raw folder? If not, how can I make a copy of the raw file to a folder in the SD? Thanks in advance.

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  • How to send Sound Stream of a file from disk over network using FMOD?

    - by chris
    Hey everyone, i'm currently working on a project in college. my application should do some things with audio files from my computer. i'm using FMOD as sound library. the problem i have is, that i dont know how to access the data of a soundfile (wich was opened and startet using the FMOD methods) to stream it over network for playback on another pc in the net. does anyone has a similar problem?! any help is apreciated. thanks in advance. chris

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  • Can't get max_post_size php variable set in lunar pages.

    - by Behrooz Karjooravary
    I need to increase max post size and upload size for php to use the audio module of drupal. I read this has to be set in php.ini. However I don't think I have access to that file in lunar pages. I also read it can also be set in .htaccess. However it doesn't change anything. I tried: php_value post_max_size "40M" php_value upload_max_filesize "40M" i also tried: php_value post_max_size 40M php_value upload_max_filesize 40M On localhost it says restart webserver. But this is not possible on shared host. Could that be the problem?

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  • Crashes when using AVAudioPlayer on iPhone

    - by mindthief
    Hi all, I am trying to use AVAudioPlayer to play some sounds in quick succession. When I invoke the sound-playing function less frequently so that the sounds play fully before the function is invoked again, the application runs fine. But if I invoke the function quickly in rapid succession (so that sounds are played while the previous sounds are still being played), the app eventually crashes after ~20 calls to the function, with the message "EXC_BAD_ACCESS". Here is code from the function: NSString *nsWavPath = [[[NSBundle mainBundle] resourcePath] stringByAppendingPathComponent:wavFileName]; AVAudioPlayer* theAudio = [[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:nsWavPath] error:NULL]; theAudio.delegate = self; [theAudio play]; As mentioned in another thread, I implemented the following delegate function: - (void) audioPlayerDidFinishPlaying:(AVAudioPlayer *)player successfully:(BOOL)flag { if(!flag) NSLog(@"audio did NOT finish successfully\n"); [player release]; } But the app still crashes after around ~20 rapid calls to the function. Any idea what I'm doing wrong?

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  • showSettings callback in Flex?

    - by Jim Robert
    I am pretty new to flex, so forgive me if this is an obvious question. Is there a way to open the Security.showSettings (flash.system.Security) with a callback? or at least to detect if it is currently open or not? My flex application is used for streaming audio, and is normally controlled by javascript, so I keep it hidden for normal use (via absolute positioning it off the page). When I need microphone access I need to make the flash settings dialog visible, which works fine, I move it into view and open the dialog. When the user closes it, I need to move it back off the screen so they don't see an empty flex app sitting there after they change their settings. thanks :)

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  • Do you know a good and efficient FFT?

    - by yan bellavance
    Hi, I am trying to find a very fast and efficient Fourier transform (FFT). Does anyone know of any good ones. I need to run it on the iPhone so it must not be intensive. Instead, maybe you know of one that is wavelet like, i need frequency resolution but only a narrow band (vocal audio range up to 10khz max...even 10Khz might be too high). Im thinking also of truncating this FFT to keep the frequency resolution while eliminating the unwanted frequency band. This is for an iphone ...I have taken a look at the FFT in Aurio touch but it seems this is an int FFT but my app uses floats.....would it give a big performance increase to try and adapt program to an int FFT or not(which i really dont feel like doing...plus aurio touch uses a radix 2 FFT which is not that great).

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  • Make two servers talk to each other

    - by Maksim
    I have application written in GWT and hosted on Google AppEngine/Java. In this application user will have an option to upload video/audio/text file to the server. Those files could be big, up to 1gb or so and because GAE/J does not support large file I have to use another server to store those files. This would be easy to implement if there was no cross-domain security feature in browsers. So, what I'm thinking is to make GAE Server talk to my server (Glassfish or any other java servers if needed) to tell url to the file and if possible send status of uploaded file (how many percent was uploaded) so I can show status on clients screen. Here is what I'm thinking to do. When user loads GWT page that is stored on GAE/J he/she will upload file to my server, then my server will send response back to GAE and GAE will send response to the client. If this scenario is possible what would be the best way to implement GAE to Glassfish conversation?

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  • Best Design pattern for social media file transfer

    - by Onema
    Our system would like our clients to link their accounts with different social media sites like youtube, vimeo, facebook, myspace and so on. One of the benefits we would like to give to the user is to transfer, update and delete files they have uploaded to our sites and transfer them to the social media sites mentioned above. this files could be videos, images or audio. We started thinking about using a strategy pattern, as all of these sites share a common process ( authentication, connection, use the API to transfer/edit/delete the file ), but we soon realized that it may not work as me may want to use some of the extended functionality that is specific to each service (eg: associate a youtube video with a channel, or upload images to a specific album on facebook, and much, much more...) My question is, what would be the best Structural Design Patter to use for this scenario?

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  • Looking for a component (.NET or COM/ActiveX) that can play AVI files in a WinForms app

    - by MusiGenesis
    I'm looking for something like the Windows Media Player control that can be hosted on a form. The WMP doesn't work for me because I need a control that can play a continuously-appended playlist of AVI files in sequence, so that the transition from one file to the next happens seamlessly (i.e. without any glitches or pauses in the video and audio). With WMP, there's always a delay between files of half a second or so. Does anyone know of a control (it can be either commercial or open-source) that can do this? I assume anything like this wraps DirectX, and that's OK too.

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