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  • [android] MediaRecorder prepare() causes segfault

    - by dwilde1
    Folks, I have a situation where my MediaRecorder instance causes a segfault. I'm working with a HTC Hero, Android 1.5+APIs. I've tried all variations, including 3gpp and H.263 and reducing the video resolution to 320x240. What am I missing? The state machine causes 4 MediaPlayer beeps and then turns on the video camera. Here's the pertinent source: UPDATE: ADDING SURFACE CREATE INFO I have rebooted the device based on previous answer to similar question. UPDATE 2: I seem to be following the MediaRecorder state machine perfectly, and if I trap out the MR code, the blank surface displays perfectly and everything else functions perfectly. I can record videos manually and play back via MediaPlayer in my code, so there should be nothing wrong with the underlying code. I've copied sample code on the surface and surfaceHolder code. I've looked at the MR instance in the Debug perspective in Eclipse and see that all (known) variables seem to be instantiated correctly. The setter calls are all now implemented in the exaxct order specced in the state diagram. // in activity class definition protected MediaPlayer mPlayer; protected MediaRecorder mRecorder; protected boolean inCapture = false; protected int phaseCapture = 0; protected int durCapturePhase = INF; protected SurfaceView surface; protected SurfaceHolder surfaceHolder; // in onCreate() // panelPreview is an empty LinearLayout surface = new SurfaceView(getApplicationContext()); surfaceHolder = surface.getHolder(); surfaceHolder.setType(SurfaceHolder.SURFACE_TYPE_PUSH_BUFFERS); panelPreview.addView(surface); // in timer handler runnable if (mRecorder == null) mRecorder = new MediaRecorder(); mRecorder.setAudioSource(MediaRecorder.AudioSource.MIC); mRecorder.setVideoSource(MediaRecorder.VideoSource.CAMERA); mRecorder.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP); mRecorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB); mRecorder.setOutputFile(path + "/" + vlip); mRecorder.setVideoSize(320, 240); mRecorder.setVideoFrameRate(15); mRecorder.setPreviewDisplay(surfaceHolder.getSurface()); panelPreview.setVisibility(LinearLayout.VISIBLE); mRecorder.prepare(); mRecorder.start(); Here is a complete log trace for the process run and crash: I/ActivityManager( 80): Start proc com.ejf.convince.jenplus for activity com.ejf.convince.jenplus/.JenPLUS: pid=17738 uid=10075 gids={1006, 3003} I/jdwp (17738): received file descriptor 10 from ADB W/System.err(17738): Can't dispatch DDM chunk 46454154: no handler defined W/System.err(17738): Can't dispatch DDM chunk 4d505251: no handler defined I/WindowManager( 80): Screen status=true, current orientation=-1, SensorEnabled=false I/WindowManager( 80): needSensorRunningLp, mCurrentAppOrientation =-1 I/WindowManager( 80): Enabling listeners W/ActivityThread(17738): Application com.ejf.convince.jenplus is waiting for the debugger on port 8100... I/System.out(17738): Sending WAIT chunk I/dalvikvm(17738): Debugger is active I/AlertDialog( 80): [onCreate] auto launch SIP. I/WindowManager( 80): onOrientationChanged, rotation changed to 0 I/System.out(17738): Debugger has connected I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): debugger has settled (1370) I/ActivityManager( 80): Displayed activity com.ejf.convince.jenplus/.JenPLUS: 5186 ms I/OpenCore( 2696): [Hank debug] LN 289 FN CreateNode I/AudioHardwareMSM72XX( 2696): AUDIO_START: start kernel pcm_out driver. W/AudioFlinger( 2696): write blocked for 96 msecs I/PlayerDriver( 2696): CIQ 1625 sendEvent state=5 I/OpenCore( 2696): [Hank debug] LN 289 FN CreateNode I/PlayerDriver( 2696): CIQ 1625 sendEvent state=5 I/OpenCore( 2696): [Hank debug] LN 289 FN CreateNode I/PlayerDriver( 2696): CIQ 1625 sendEvent state=5 I/OpenCore( 2696): [Hank debug] LN 289 FN CreateNode I/PlayerDriver( 2696): CIQ 1625 sendEvent state=5 W/AuthorDriver( 2696): Intended width(640) exceeds the max allowed width(352). Max width is used instead. W/AuthorDriver( 2696): Intended height(480) exceeds the max allowed height(288). Max height is used instead. I/AudioHardwareMSM72XX( 2696): AudioHardware pcm playback is going to standby. I/DEBUG (16094): *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** I/DEBUG (16094): Build fingerprint: 'sprint/htc_heroc/heroc/heroc: 1.5/CUPCAKE/85027:user/release-keys' I/DEBUG (16094): pid: 17738, tid: 17738 com.ejf.convince.jenplus Thanks in advance! -- Don Wilde http://www.ConvinceProject.com

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  • How to install FFMpeg in WampServer 2.0 (Windows XP)

    - by Richard Knop
    I need to install the ffmpeg PHP extension on my localhost so I can test few of my scripts but I am having troubles figuring out how to do that. I have WampServer 2.0 with PHP 5.2.9-2, my OS is Windows XP. Please somebody give me step by step instructions. I have found some Windows builds here: http://sourceforge.net/projects/ffmpeg-php/files/ But I don't know which one to download and what to do with files. EDITED: What I have done so far: Download ffmpeg_new Copy php_ffmpeg.dll from the php5 folder to the C:\wamp\bin\php\php5.2.9-2\ext Copy files from common to the windows/system32 folder Add extension=php_ffmpeg.dll to php.ini file Restarted all services (Apache, PHP...) I am gettings an error after using this code: $extension = 'ffmpeg'; $extension_soname = 'php_ffmpeg.dll'; $extension_fullname = PHP_EXTENSION_DIR . "/" . $extension_soname; // load extension if(false === extension_loaded($extension)) { if (false === dl($extension_soname)) throw new Exception("Can't load extension $extension_fullname\n"); } The error: Warning: dl() [function.dl]: Not supported in multithreaded Web servers - use extension=ffmpeg.dll in your php.ini in C:\wamp\www\hunnyhive\application\modules\default\controllers\MyAccountController.php on line 314 Plus I also get the exception from above.

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  • Why is WCF Stream response getting corrupted on write to disk?

    - by Alvin S
    I am wanting to write a WCF web service that can send files over the wire to the client. So I have one setup that sends a Stream response. Here is my code on the client: private void button1_Click(object sender, EventArgs e) { string filename = System.Environment.CurrentDirectory + "\\Picture.jpg"; if (File.Exists(filename)) File.Delete(filename); StreamServiceClient client = new StreamServiceClient(); int length = 256; byte[] buffer = new byte[length]; FileStream sink = new FileStream(filename, FileMode.CreateNew, FileAccess.Write); Stream source = client.GetData(); int bytesRead; while ((bytesRead = source.Read(buffer,0,length))> 0) { sink.Write(buffer,0,length); } source.Close(); sink.Close(); MessageBox.Show("All done"); } Everything processes fine with no errors or exceptions. The problem is that the .jpg file that is getting transferred is reported as being "corrupted or too large" when I open it. What am I doing wrong? On the server side, here is the method that is sending the file. public Stream GetData() { string filename = Environment.CurrentDirectory+"\\Chrysanthemum.jpg"; FileStream myfile = File.OpenRead(filename); return myfile; } I have the server configured with basicHttp binding with Transfermode.StreamedResponse.

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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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  • MediaRecorder prepare() causes segfault

    - by dwilde1
    Folks, I have a situation where my MediaRecorder instance causes a segfault. I'm working with a HTC Hero, Android 1.5+APIs. I've tried all variations, including 3gpp and H.263 and reducing the video resolution to 320x240. What am I missing? The state machine causes 4 MediaPlayer beeps and then turns on the video camera. Here's the pertinent source: UPDATE: ADDING SURFACE CREATE INFO I have rebooted the device based on previous answer to similar question. UPDATE 2: I seem to be following the MediaRecorder state machine perfectly, and if I trap out the MR code, the blank surface displays perfectly and everything else functions perfectly. I can record videos manually and play back via MediaPlayer in my code, so there should be nothing wrong with the underlying code. I've copied sample code on the surface and surfaceHolder code. I've looked at the MR instance in the Debug perspective in Eclipse and see that all (known) variables seem to be instantiated correctly. The setter calls are all now implemented in the exaxct order specced in the state diagram. UPDATE 3: I've tried all permission combinations: CAMERA + RECORD_AUDIO+RECORD_VIDEO, CAMERA only, RECORD_AUDIO+RECORD_VIDEO This is driving me bats! :))) // in activity class definition protected MediaPlayer mPlayer; protected MediaRecorder mRecorder; protected boolean inCapture = false; protected int phaseCapture = 0; protected int durCapturePhase = INF; protected SurfaceView surface; protected SurfaceHolder surfaceHolder; // in onCreate() // panelPreview is an empty LinearLayout surface = new SurfaceView(getApplicationContext()); surfaceHolder = surface.getHolder(); surfaceHolder.setType(SurfaceHolder.SURFACE_TYPE_PUSH_BUFFERS); panelPreview.addView(surface); // in timer handler runnable if (mRecorder == null) mRecorder = new MediaRecorder(); mRecorder.setAudioSource(MediaRecorder.AudioSource.MIC); mRecorder.setVideoSource(MediaRecorder.VideoSource.CAMERA); mRecorder.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP); mRecorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB); mRecorder.setOutputFile(path + "/" + vlip); mRecorder.setVideoSize(320, 240); mRecorder.setVideoFrameRate(15); mRecorder.setPreviewDisplay(surfaceHolder.getSurface()); panelPreview.setVisibility(LinearLayout.VISIBLE); mRecorder.prepare(); mRecorder.start(); Here is a complete log trace for the process run and crash: I/ActivityManager( 80): Start proc com.ejf.convince.jenplus for activity com.ejf.convince.jenplus/.JenPLUS: pid=17738 uid=10075 gids={1006, 3003} I/jdwp (17738): received file descriptor 10 from ADB W/System.err(17738): Can't dispatch DDM chunk 46454154: no handler defined W/System.err(17738): Can't dispatch DDM chunk 4d505251: no handler defined I/WindowManager( 80): Screen status=true, current orientation=-1, SensorEnabled=false I/WindowManager( 80): needSensorRunningLp, mCurrentAppOrientation =-1 I/WindowManager( 80): Enabling listeners W/ActivityThread(17738): Application com.ejf.convince.jenplus is waiting for the debugger on port 8100... I/System.out(17738): Sending WAIT chunk I/dalvikvm(17738): Debugger is active I/AlertDialog( 80): [onCreate] auto launch SIP. I/WindowManager( 80): onOrientationChanged, rotation changed to 0 I/System.out(17738): Debugger has connected I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): waiting for debugger to settle... I/System.out(17738): debugger has settled (1370) I/ActivityManager( 80): Displayed activity com.ejf.convince.jenplus/.JenPLUS: 5186 ms I/OpenCore( 2696): [Hank debug] LN 289 FN CreateNode I/AudioHardwareMSM72XX( 2696): AUDIO_START: start kernel pcm_out driver. W/AudioFlinger( 2696): write blocked for 96 msecs I/PlayerDriver( 2696): CIQ 1625 sendEvent state=5 I/OpenCore( 2696): [Hank debug] LN 289 FN CreateNode I/PlayerDriver( 2696): CIQ 1625 sendEvent state=5 I/OpenCore( 2696): [Hank debug] LN 289 FN CreateNode I/PlayerDriver( 2696): CIQ 1625 sendEvent state=5 I/OpenCore( 2696): [Hank debug] LN 289 FN CreateNode I/PlayerDriver( 2696): CIQ 1625 sendEvent state=5 W/AuthorDriver( 2696): Intended width(640) exceeds the max allowed width(352). Max width is used instead. W/AuthorDriver( 2696): Intended height(480) exceeds the max allowed height(288). Max height is used instead. I/AudioHardwareMSM72XX( 2696): AudioHardware pcm playback is going to standby. I/DEBUG (16094): *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** I/DEBUG (16094): Build fingerprint: 'sprint/htc_heroc/heroc/heroc: 1.5/CUPCAKE/85027:user/release-keys' I/DEBUG (16094): pid: 17738, tid: 17738 com.ejf.convince.jenplus Thanks in advance! -- Don Wilde http://www.ConvinceProject.com

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  • How do continuously update data to an asp page?

    - by Lori
    Hi, I have an asp page based on a very simple database. It references a single table of probably 30 records and maybe 12 data fields and everything works great as I am only uploading a new database every week or so. I have a special circumstance where I would like upload new data to the database and display automatically on the page every 20 to 30 seconds without the user having to refresh their screen. I would expect up to 1000 concurrent users accessing the data. I have been manually uploading the database via ftp, which will obviously not work on this timeline and would also run the risk of error pages as the database is being replaced. So, can anyone point me the right direction to setup this scenario? Other details that might be helpful: The database is an Access database (but I could change to another format if needed) Running on Windows platform hosted by an ISP, not my own server Thanks in advance for any help on this! Lori

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  • Use a "x-dom-event-stream" stream in javascript ?

    - by rnaud
    Hello, HTML5 draft contains an API called EventSource to stream data (notifications) trough javascript using only one server call. Looking it up, I found an exemple on Opera Labs of the javascript part : document.getElementsByTagName("event-source")[0] .addEventListener("server-time", eventHandler, false); function eventHandler(event) { // Alert time sent by the server alert(event.data); } and the server side part : <?php header("Content-Type: application/x-dom-event-stream"); while(true) { echo "Event: server-time\n"; $time = time(); echo "data: $time\n"; echo "\n"; flush(); sleep(3); } ?> But as of today, it seems only Opera has implemented the API, neither Chrome nor Safari have a working version (Am I wrong here ?) So my question is, is there any other way in javascript, maybe more complex, to use this one stream to get data ?

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  • Is there a way to make PHP progressively output as the script executes?

    - by Iain Fraser
    So I'm writing a disposable script for my own personal single use and I want to be able see how the process is going. Basically I'm processing a couple of thousand media releases and sending them to our new CMS. So I don't hammer the CMS, I'm making the script sleep for a couple of seconds after every 5 requests. I would like - as the script is executing - to be able to see my echos telling me the script is going to sleep or that the last transaction with the webservice was successful. Is this possible in PHP? Thanks for your help! Iain

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  • cancel stream request from WCF server to client

    - by ArsenMkrt
    Hi, I posted about stream request here [wcf-chunk-data-with-stream]:http://stackoverflow.com/questions/853448/wcf-chunk-data-with-stream I solved that task but now when i close request in client part server continue to send data. is it possible to cancel stream request from WCF server to client?

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  • batch file that detects keystrokes. how?

    - by daniel11
    im designing a collection of video games for the command line (such as deal or no deal, tic tac toe, racing, maze puzzle, connect four, wack a mole, etc.) however it would really make things easier for me if i could make it so that when the user makes a selection (such as what direction to move in the game) , that as soon as they press the arrow keys then it carries out the IF statements that follow. Instead of having to press enter after each selection. something like... :one1 set /p direction1= : IF %direction1%== {ARROW KEY LEFT} goto two2 IF %direction1%== {ARROW KEY RIGHT} goto three3 IF %direction1%== {ARROW KEY UP} goto four4 IF %direction1%== {ARROW KEY DOWN} goto five5 goto one1 Any Ideas?

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  • How to use RTPSocket to send RTP packets

    - by Afro Genius
    Hi there, am relatively new to JMF but have gone through the documents and have a sufficient understanding of how it works. That been said am having some trouble implementing a the server side for RTPSockets. After looking at their illustrations and example. I am still abit confused. Am I to develop a datasource and also datasink classes to handle the transfer? What am trying to do is stream data from my application to the underlying network and receive it back through another application. I have and understand receiving but just can't get my head around the steps involved for sending. Any help would be most appreciated.

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  • record output sound in python

    - by aaronstacy
    i want to programatically record sound coming out of my laptop in python. i found PyAudio and came up with the following program that accomplishes the task: import pyaudio, wave, sys chunk = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 44100 RECORD_SECONDS = 5 WAVE_OUTPUT_FILENAME = sys.argv[1] p = pyaudio.PyAudio() channel_map = (0, 1) stream_info = pyaudio.PaMacCoreStreamInfo( flags = pyaudio.PaMacCoreStreamInfo.paMacCorePlayNice, channel_map = channel_map) stream = p.open(format = FORMAT, rate = RATE, input = True, input_host_api_specific_stream_info = stream_info, channels = CHANNELS) all = [] for i in range(0, RATE / chunk * RECORD_SECONDS): data = stream.read(chunk) all.append(data) stream.close() p.terminate() data = ''.join(all) wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb') wf.setnchannels(CHANNELS) wf.setsampwidth(p.get_sample_size(FORMAT)) wf.setframerate(RATE) wf.writeframes(data) wf.close() the problem is i have to connect the headphone jack to the microphone jack. i tried replacing these lines: input = True, input_host_api_specific_stream_info = stream_info, with these: output = True, output_host_api_specific_stream_info = stream_info, but then i get this error: Traceback (most recent call last): File "./test.py", line 25, in data = stream.read(chunk) File "/Library/Python/2.5/site-packages/pyaudio.py", line 562, in read paCanNotReadFromAnOutputOnlyStream) IOError: [Errno Not input stream] -9975 is there a way to instantiate the PyAudio stream so that it inputs from the computer's output and i don't have to connect the headphone jack to the microphone? is there a better way to go about this? i'd prefer to stick w/ a python app and avoid cocoa.

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  • getAudioInputStream can not convert [stereo, 4 bytes/frame] stream to [mono, 2 bytes/frame]

    - by brian_d
    Hello. I am using javasound and have an AudioInputStream of format PCM_SIGNED 8000.0 Hz, 16 bit, stereo, 4 bytes/frame, little-endian Using AudioSystem.getAudioInputStream(target_format, original_stream) produces an 'IllegalArgumentException: Unsupported Conversion' when the target_format is PCM_SIGNED 8000.0 Hz, 16 bit, mono, 2 bytes/frame, little-endian Is it possible to convert this stream manually after every read() call? And if yes, how? In general, how can you compare two formats and tell if a conversion is possible?

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  • ShoutCast over SSL

    - by Honus Wagner
    So I've gone ahead and set up my ShoutCast server DNAS and set my DSP in Winamp on my host computer. The server listens on port 8000, so per some instructions I installed an output plugin for winamp (Shoutcast DSP) and used 8000 and the password to connect. Server accepts the connection. Now, what the heck do I do now? My host computer is SSL secured and the DNAS server is installed within the secure web directory (if that matters). My desired end result is that I want to listen to my ShoutCast setup at home (host computer) from any computer. I try browsing to my ip address and port 8000 (without using HTTPS) and it comes back with nothing. If I browse with HTTPS://my.server.com:8000, I get Error code: ssl_error_rx_record_too_long) Have I completely missed something, or am I just a total moron? Thanks.

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  • Dimdim Change name

    - by islam
    i build dimdim v4.5 on my pc and its work fine with me. each time i want to start meeting i type my pc IP address like this : http://<my-ip-address>/dimdim i want to change the word dimdim to be anything else like : http://<my-ip-address>/meeting regards

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  • Steaming a non-PCM WAV file to a SilverLight application

    - by Satumba
    Hi, I would like to allow users to play recorded WAV files that stored on a server back to a Silverlight application as a client to play them. I saw that there is a way to play a WAV file on Silverlight (here), but when i tried to impliment it, i got an error playing the file because it is not in PCM format but encoded. The files that i'm trying to play are encoded with a special encoder, so i thought that the only way is to decode the WAV file on the server and stream it back to the client. The limitation is that the decode process should occur in real time because it is not reasonable to convert all the WAV files that exists. Is it possible to do it? Which streamer can i use? (Windows Media Service can help here?) Does somebody has any experience with such a scenario? Appreciate your help.

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  • Blackberry buffered playback demo??

    - by Bohemian
    Can someone help me to buffer a mp3 file on a server using the Blackberry buffered pllayback demo app provided with the jde? I hav loaded it in the simulator. And my mds is started but I m unable to play the audio. There is no error but it doesnt play/load. The code looks all fine. Thanks

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  • Wireshark doesnt' recognises RTMP streams

    - by Andrew
    Hello! I found on the web few samples on tracking RTMP (Real Time Messaging Protocol) with Wireshark, but it doesn't work for me. All RTMPT packets rendered as basic TCP packet like this: 149 14.324999 85.115.xxx.xxx 192.168.1.20 TCP macromedia-fcs > 54557 [ACK] Seq=1 Ack=1452 Win=69 Len=0 I'm using Wireshark 1.2.8 with all protocols installed on Windows Vista. What can i do to fix it? Thx!

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  • Trying to build automatic audio-conferencing capability into a WebApp

    - by Keller
    Hey all, I'm working with a team of relatively novice programmers, and we are trying to create a site that will have audio-conferencing capabilities such that whenever someone visits the page, they will immediately have audio-conferencing capabilities with everyone else on the page (5 people max). Can anyone point us in a general direction? Should we be looking into building a custom app, leveraging audio conferencing software, or trying to mimic a webex program? Would Adobe Stratus be useful in getting this kind of functionality? Does anyone have any ideas about how we would design something like this on a macro level? Sorry for the noobish question, but any guidance would be deeply appreciated. Thanks, Keller

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  • Stream (.NET) handling best-practices

    - by Jader Dias
    The question is entitled with the word "Stream" because the question below is a concrete example of a more generic doubt I have about Streams: I have a problem that accepts two solutions and I want to know the best one: I download a file, save it to disk (2 min), read it and write the contents to the DB (+ 2 min). I download a file and write the contents directly to the DB (3 min). If the write to DB fails I'll have to download again in the second case, but not in the first case. Which is best? Which would you use?

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  • Recording Audio through RTMP/Rails

    - by Lowgain
    I am in the process of building a rails/flex application which requires audio to be recorded and then stored in our amazon s3 account. I have found no alternative to using some form of RTMP server for recording audio through flash, but our hosting environment will not allow us to install anything like FMS, Red5, etc. Is there any existing Ruby/Rails RTMP solution that will allow audio recording? If not, is it possible for Rails to at least intercept the RTMP stream and then I can hope to reference red5 or something for parsing the data (long shot, I know)? The other alternative I can think of is hosting a red5 server on another host and communicating with our rails app once the saving/uploading is done, which is not preferred. Am I going to have any luck here?

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  • XMLStreamReader and a real stream

    - by Yuri Ushakov
    Update There is no ready XML parser in Java community which can do NIO and XML parsing. This is the closest I found, and it's incomplete: http://wiki.fasterxml.com/AaltoHome I have the following code: InputStream input = ...; XMLInputFactory xmlInputFactory = XMLInputFactory.newInstance(); XMLStreamReader streamReader = xmlInputFactory.createXMLStreamReader(input, "UTF-8"); Question is, why does the method #createXMLStreamReader() expects to have an entire XML document in the input stream? Why is it called a "stream reader", if it can't seem to process a portion of XML data? For example, if I feed: <root> <child> to it, it would tell me I'm missing the closing tags. Even before I begin iterating the stream reader itself. I suspect that I just don't know how to use a XMLStreamReader properly. I should be able to supply it with data by pieces, right? I need it because I'm processing a XML stream coming in from network socket, and don't want to load the whole source text into memory. Thank you for help, Yuri.

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  • Flash FLVPlayback states

    - by rob5408
    I'm writing my own class to manage a skin for an FLVPlayback component. It works 90% of the time, but sometimes the state get really messed up. Specifically, the video is playing, meaning I can see it play and the VideoEvent.PLAYHEAD_UPDATE event is firing, but when I poll the FLVPlayback component about its playing property, it returns false. I assume this may be because 'buffering' is kind of a subset of 'playing', but I cannot confirm this in the documentation. I guess another way to ask this question is, "Does the FLVPlayback component ever buffer while it is in a stopped state?"

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