Search Results

Search found 5304 results on 213 pages for 'audio streaming'.

Page 183/213 | < Previous Page | 179 180 181 182 183 184 185 186 187 188 189 190  | Next Page >

• 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...

  Read the article

• Problems with MediaPlayer, raw resources, stop and start

  - by arakn0

  Hello everybody, I'm new in Android development and I have the next question/problem. I'm playing around with the MediaPlayer class to reproduce some sounds/music. I am playing raw resources (res/raw) and it looks kind of easy. To play a raw resource, the MediaPlayer has to be initialized like this: MediaPlayer mp = MediaPlayer.create(appContext, R.raw.song); mp.start(); Until here there is no problem. The sound is played, and everything works fine. My problem appears when I want to add more options to my application. Specifically when I add the "Stop" button/option. Basically, what I want to do is...when I press "Stop", the music stops. And when I press "Start", the song/sound starts over. (pretty basic!) To stop the media player, you only have to call stop(). But to play the sound again, the media player has to be reseted and prepared. mp.reset(); mp.setDataSource(params); mp.prepare(); The problem is that the method setDataSource() only accepts as params a file path, Content Provider URI, streaming media URL path, or File Descriptor. So, since this method doesn't accept a resource identifier, I don't know how to set the data source in order to call prepare(). In addition, I don't understand why you can't use a Resouce identifier to set the data source, but you can use a resource identifier when initializing the MediaPlayer. I guess that I'm missing something. I wonder if I am mixing concepts, and the method stop() doesn't have to be called in the "Stop" button. Any help? Thanks in advanced!!!

  Read the article

• 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

  Read the article

• 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 :)

  Read the article

• 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?

  Read the article

• Keymap issues with NX from Mac OS X Lion

  - by Andy

  I tried to answer the question from Mark: Keymap issues with NX from Mac OS X Lion to Ubuntu However, it is locked so I figured I would post a new question / answer. I have been trying to answer this for a few days now because I have no issues when connecting through NX Client (technically OpenNX) to FreeNX server from an iMac (with Lion), but if I try to connect with a Macbook Pro I get horrible keyboard binding issues. The fix that is working for me is to go into: ~/.nx/config/HOST.nxs and change: <option key="Current keyboard" value="false"/> <option key="Custom keyboard layout" value="empty"/> <option key="Grab keyboard" value="false"/> I have tried this on three NX Servers and all are fixed. Hope it helps or gets you closer. Always check in the ~/.nx/temp/ for the sshlog and see if --keyboard="empty/empty" instead of "pc105/en" because the Mac is really pc104. 9:05:35: startsession --session="HOST" --type="unix-gnome" --cache="8M" --images="32M" --link="adsl" --geometry="2556\ x1396" --screeninfo="2560x1440x32+render" --keyboard="empty/empty" --backingstore="1" --encryption="1" --composite="1" --\ shmem="1" --shpix="1" --streaming="1" --samba="0" --cups="0" --nodelay="1" --defer="0" --client="macosx" --media="0" --st\ rict="0" --aux="1"

  Read the article

• 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!

  Read the article

• 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.

  Read the article

• Executing bat file and returning the prompt

  - by Lieven Cardoen

  I have a problem with cruisecontrol where an ant scripts executes a bat file that doesn't give me the prompt back. As a result, the project in cruisecontrol keeps on bulding forever until I restart cruisecontrol. How can I resolve this? It's a startup.bat from wowza (Streaming Server) that I'm executing: @echo off call setenv.bat if not %WMSENVOK% == "true" goto end set _WINDOWNAME="Wowza Media Server 2" set _EXESERVER= if "%1"=="newwindow" ( set _EXESERVER=start %_WINDOWNAME% shift ) set CLASSPATH="%WMSAPP_HOME%\bin\wms-bootstrap.jar" rem cacls jmxremote.password /P username:R rem cacls jmxremote.access /P username:R rem NOTE: Here you can configure the JVM's built in JMX interface. rem See the "Server Management Console and Monitoring" chapter rem of the "User's Guide" for more information on how to configure the rem remote JMX interface in the [install-dir]/conf/Server.xml file. set JMXOPTIONS=-Dcom.sun.management.jmxremote=true rem set JMXOPTIONS=%JMXOPTIONS% -Djava.rmi.server.hostname=192.168.1.7 rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.port=1099 rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.authenticate=false rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.ssl=false rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.password.file= "%WMSCONFIG_HOME%/conf/jmxremote.password" rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.access.file= "%WMSCONFIG_HOME%/conf/jmxremote.access" rem log interceptor com.wowza.wms.logging.LogNotify - see Javadocs for ILogNotify %_EXESERVER% "%_EXECJAVA%" %JAVA_OPTS% %JMXOPTIONS% -Dcom.wowza.wms.AppHome="%WMSAPP_HOME%" -Dcom.wowza.wms.ConfigURL="%WMSCONFIG_URL%" -Dcom.wowza.wms.ConfigHome="%WMSCONFIG_HOME%" -cp %CLASSPATH% com.wowza.wms.bootstrap.Bootstrap start :end

  Read the article

• Frame Accurate Browser Launchable Video Player ... ?

  - by cliftonc

  I have a requirement where I need to enable playback (full screen) of a h.264 MPEG4 (thanks for the correction!) video from a local network, launchable from a browser link on a Windows workstation, and be frame accurate. By frame accurate I mean that I need to be able to scrub through the video in the same way you would with a vtr, stop at a frame, and then move backwards and forwards frame by frame (it is for a very specific compliance requirement where have to be able to check every frame if there is something that is potentially against broadcasting guidelines). The application itself is used to capture notes while viewing the material, so the end model is for a dual monitor workstation, with a web form in one, the video playing full screen in the second (no issue launching the video and manually having to move it to the second screen), and then the user controls the video via keyboard shortcuts or a jog shuttle. I have looked at QT, but the java bindings seem to be dead or nearly so, flash isn't frame accurate, VLC given its streaming heritage seems to be only able to move forward by a frame and not backwards, and all I have left are commercial offerings that in my experience are difficult and expensive to change. Any ideas of where I should look or alternative options? Any advice appreciated!

  Read the article

• 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.

  Read the article

• Webrick transparent proxy

  - by zzeroo

  Hi there, I've a absolute simple proxy running. require 'webrick' require 'webrick/httpproxy' s = WEBrick::HTTPProxyServer.new(:Port => 8080, :RequestCallback => Proc.new{|req,res| puts req.request_line, req.raw_header}) # Shutdown functionality trap("INT"){s.shutdown} # run the beast s.start This should in my mind not influence the communication in any way. But some sites doesn't work any more. Specially http://lastfm.de 's embedded flash players doesn't work. The header looks link: - -> http://ext.last.fm/2.0/?api%5Fsig=aa3e9ac9edf46ceb9a673cb76e61fef4&flashresponse=true&y=1269686332&streaming=true&playlistURL=lastfm%3A%2F%2Fplaylist%2Ftrack%2F42620245&fod=true&sk=ee93ae4f438767bf0183d26478610732&lang=de&api%5Fkey=da6ae1e99462ee22e81ac91ed39b43a4&method=playlist%2Efetch GET http://play.last.fm/preview/118270350.mp3 HTTP/1.1 Host: play.last.fm User-Agent: Mozilla/5.0 (X11; U; Linux i686; de; rv:1.9.2) Gecko/20100308 Ubuntu/10.04 (lucid) Firefox/3.6 Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8 Accept-Language: de,en-us;q=0.7,en;q=0.3 Accept-Encoding: gzip,deflate Accept-Charset: ISO-8859-1,utf-8;q=0.7,*;q=0.7 Keep-Alive: 115 Proxy-Connection: keep-alive Cookie: AnonWSSession=ee93ae4f438767bf0183d26478610732; AnonSession=cb8096e3b0d8ec9f4ffd6497a6d052d9-12bb36d49132e492bb309324d8a4100fc422b3be9c3add15ee90eae3190db5fc localhost - - [27/Mar/2010:11:38:52 CET] "GET http://www.lastfm.de/log/flashclient/minor/Track_Loading_Fail/Buffering_Timeout HTTP/1.1" 404 7593 - -> http://www.lastfm.de/log/flashclient/minor/Track_Loading_Fail/Buffering_Timeout localhost - - [27/Mar/2010:11:38:52 CET] "GET http://play.last.fm/preview/118270350.mp3 HTTP/1.1" 302 0 I nead some hints why or what the communication disturb.

  Read the article

• 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.

  Read the article

• 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

  Read the article

• 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?

  Read the article

• 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?

  Read the article

• Internet Explorer Warning when embedding Youtube on HTTPS site?

  - by pellepim

  Our application is run over HTTPS which rarely presents any problems for us. When it comes to youtube however, the fact that they do not present any content over SSL connections is giving us some head ache when trying to embed clips. Mostly because of Internet Explorers famous little warning message: "Do you want to view only the webpage content that was delivered securely? This page contains content that will not be delivered using a secure HTTPS ... etc" I've tried to solve this in several ways. The most promising one was to use the ProxyPass functionality in Apache to map to YouTube. Like this: ProxyPass: /youtube/ http://www.youtube.com ProxyPassReverse: /youtube/ http://www.youtube.com This gets rid of the annoying warning. However, the youtube SWF fails to start streaming The SWF i manage to load into the browser simply states : "An error occurred, please try again later". Potential solutions are perhaps: Download youtube FLV:s and serve them out of own domain (gah) Use custom FLV-player and stream only FLV:s from youtube over a https proxy? Update 10 March: I've tried to use Googles Youtube API for ActionScript to load a player. It looked promising at first and I was able to load a player through my https:// proxy. However, the SWF that is loaded contains loads of explicit calls to different non-ssl urls to create authentication links for the FLV-stream and for loading different crossdomain policies. It really seems like we're not supposed to access flv-streams directly. This makes it very hard to bypass the Internet Explorer warning, short of ripping out the FLV:s from youtube and serving them out of your own domain. There are solutions out there for downloading youtubes FLV:s. But that is not compliant with the Youtube terms of use and is really not an option for us.

  Read the article

• Should I move big data blobs in JSON or in separate binary connection?

  - by Amagrammer

  QUESTION: Is it better to send large data blobs in JSON for simplicity, or send them as binary data over a separate connection? If the former, can you offer tips on how to optimize the JSON to minimize size? If the latter, is it worth it to logically connect the JSON data to the binary data using an identifier that appears in both, e.g., as "data" : "< unique identifier " in the JSON and with the first bytes of the data blob being < unique identifier ? CONTEXT: My iPhone application needs to receive JSON data over the 3G network. This means that I need to think seriously about efficiency of data transfer, as well as the load on the CPU. Most of the data transfers will be relatively small packets of text data for which JSON is a natural format and for which there is no point in worrying much about efficiency. However, some of the most critical transfers will be big blobs of binary data -- definitely at least 100 kilobytes of data, and possibly closer to 1 megabyte as customers accumulate a longer history with the product. (Note: I will be caching what I can on the iPhone itself, but the data still has to be transferred at least once.) It is NOT streaming data. I will probably use a third-party JSON SDK -- the one I am using during development is here. Thanks

  Read the article

• Playing a .TS file on iOS

  - by Jonathan Grynspan

  We're working with some hardware that produces files in the .TS format, and we'd like to play them on an iOS device. (The files are internally consistent with what iOS supports--MPEG-4 video, AAC audio.) We've been investigating three options so far: Roll our own integrated HTTP Live Streaming server and serve up a faux M3U8 playlist from within the app. This... doesn't seem to want to play nice, and we've had mixed luck actually getting the .TS files to play on devices. Unwrap the MPEG-4 and AAC data from the TS file and re-wrap it as MP4. This, I'm told, is exceedingly difficult to do, and I haven't found anything useful online that could shed light on how to do it. We've got code in the pipeline to do it but it won't be ready until long after we need it. If we could do it, I could easily subclass NSURLProtocol and have it working within a matter of hours minutes. Use FFmpeg to implement option #2. FFmpeg seems like a possible solution but it isn't configured to build for iOS and I don't have the background to get it working (whereas the rest of our engineers don't have the Apple background needed.) I think #2 is our best bet, but as I don't know the ins and outs of MPEG-2 TS and MPEG-4, I don't have the ability to put it together myself. Does anybody have any insight into this problem? Perhaps some experience playing local TS files on iOS, or some tips on converting from TS to MP4?

  Read the article

• Best solution for a comment table for multiple content types

  - by KRTac

  I'm currently designing a comments table for a site I'm building. Users will be able to upload images, link videos and add audio files to the profile. Each of these types of content must be commentable. Now I'm wondering what's the best approach to this. My current options are: 1. to have one big comments table and a link tables for every content type (comments_videos, ...) with comment_id and _id. 2. to have comments separated by the type of content their for. So each type of content would have his own comments table with the comments for that type.

  Read the article

• 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?

  Read the article

• 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

  Read the article

• Internet Radio Station for University

  - by ryan

  I am trying to help my University Student Radio station rethink the setup of the way they stream music, but I have some questions regarding the use of Ubuntu to stream music. Currently, the radio station uses two windows machines: one of which is used to stream the radio station and serve the website, and the other is used by rotating djs to select songs and create playlists. The computer used by djs feeds mono into the sound card of the server and the server streams the feed online. -Ideally I would like to maintain a two-computer setup: One computer as server, and another that is used to select and play music by rotating djs. -I would like to use Ubuntu for the server. -I would like to use Windows for the other machine. -The server should be able to stream song information. First, is there a way to somehow get the song information from an analog feed? Second, what is the best streaming server for radio? I have encountered shoutcast, icecast, and darwin, but I don't know where to begin in attempting to gauge them. Finally, if anyone has any tips or pointers about small internet radio station management/ setup they would be appreciated as this is my first radio station, and I am eager to hear of past experiences.

  Read the article

• 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?

  Read the article

• 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.

  Read the article

< Previous Page | 179 180 181 182 183 184 185 186 187 188 189 190  | Next Page >