I&#39;ve seen plenty of info about how to stream video from the server to an android device, but not much about the other way, ala Qik.  Could someone point me in the right direction here, or give me some advice on how to approach this?

Streaming video from Android camera to server

I&#39;m really stuck trying to understand the best way to stream real time output of ffmpeg to a HTML5 client using node.js, as there are a number of variables at play and I don&#39;t have a lot of experience in this space, having spent many hours trying different combinations.

My use case is:

1) IP video camera RTSP H.264 stream is picked up by FFMPEG and remuxed into a mp4 container using the following FFMPEG settings in node, output to STDOUT. This is only run on the initial client connection, so that partial content requests don&#39;t try to spawn FFMPEG again.

    liveFFMPEG = child_process.spawn(&quot;ffmpeg&quot;, [
                    &quot;-i&quot;, &quot;rtsp://admin:12345@192.168.1.234:554&quot; , &quot;-vcodec&quot;, &quot;copy&quot;, &quot;-f&quot;,
                    &quot;mp4&quot;, &quot;-reset_timestamps&quot;, &quot;1&quot;, &quot;-movflags&quot;, &quot;frag_keyframe+empty_moov&quot;, 
                    &quot;-&quot;   // output to stdout
                    ],  {detached: false});

2) I use the node http server to capture the STDOUT and stream that back to the client upon a client request. When the client first connects I spawn the above FFMPEG command line then pipe the STDOUT stream to the HTTP response.

    liveFFMPEG.stdout.pipe(resp);

I have also used the stream event to write the FFMPEG data to the HTTP response but makes no difference

    xliveFFMPEG.stdout.on(&quot;data&quot;,function(data) {
            resp.write(data);
    }

I use the following HTTP header (which is also used and working when streaming pre-recorded files)

    var total = 999999999         // fake a large file
    var partialstart = 0
    var partialend = total - 1

    if (range !== undefined) {
        var parts = range.replace(/bytes=/, &quot;&quot;).split(&quot;-&quot;); 
        var partialstart = parts[0]; 
        var partialend = parts[1];
    } 

    var start = parseInt(partialstart, 10); 
    var end = partialend ? parseInt(partialend, 10) : total;   // fake a large file if no range reques 

    var chunksize = (end-start)+1; 

    resp.writeHead(206, {
                      &#39;Transfer-Encoding&#39;: &#39;chunked&#39;
                     , &#39;Content-Type&#39;: &#39;video/mp4&#39;
                     , &#39;Content-Length&#39;: chunksize // large size to fake a file
                     , &#39;Accept-Ranges&#39;: &#39;bytes &#39; + start + &quot;-&quot; + end + &quot;/&quot; + total
    });


3) The client has to use HTML5 video tags.

I have no problems with streaming playback (using fs.createReadStream with 206 HTTP partial content) to the HTML5 client a video file previously recorded with the above FFMPEG command line (but saved to a file instead of STDOUT), so I know the FFMPEG stream is correct, and I can even correctly see the video live streaming in VLC when connecting to the HTTP node server.

However trying to stream live from FFMPEG via node HTTP seems to be a lot harder as the client will display one frame then stop. I suspect the problem is that I am not setting up the HTTP connection to be compatible with the HTML5 video client. I have tried a variety of things like using HTTP 206 (partial content) and 200 responses, putting the data into a buffer then streaming with no luck, so I need to go back to first principles to ensure I&#39;m setting this up the right way.

Here is my understanding of how this should work, please correct me if I&#39;m wrong:

1) FFMPEG should be setup to fragment the output and use an empty moov (FFMPEG frag_keyframe and empty_moov mov flags). This means the client does not use the moov atom which is typically at the end of the file which isn&#39;t relevant when streaming (no end of file), but means no seeking possible which is fine for my use case.

2) Even though I use MP4 fragments and empty MOOV, I still have to use HTTP partial content, as the HTML5 player will wait until the entire stream is downloaded before playing, which with a live stream never ends so is unworkable.

3) I don&#39;t understand why piping the STDOUT stream to the HTTP response doesn&#39;t work when streaming live yet if I save to a file I can stream this file easily to HTML5 clients using similar code. Maybe it&#39;s a timing issue as it takes a second for the FFMPEG spawn to start, connect to the IP camera and send chunks to node, and the node data events are irregular as well. However the bytestream should be exactly the same as saving to a file, and HTTP should be able to cater for delays.

4) When checking the network log from the HTTP client when streaming a MP4 file created by FFMPEG from the camera, I see there are 3 client requests: A general GET request for the video, which the HTTP server returns about 40Kb, then a partial content request with a byte range for the last 10K of the file, then a final request for the bits in the middle not loaded. Maybe the HTML5 client once it receives the first response is asking for the last part of the file to load the MP4 MOOV atom? If this is the case it won&#39;t work for streaming as there is no MOOV file and no end of the file.

5) When checking the network log when trying to stream live, I get an aborted initial request with only about 200 bytes received, then a re-request again aborted with 200 bytes and a third request which is only 2K long. I don&#39;t understand why the HTML5 client would abort the request as the bytestream is exactly the same as I can successfully use when streaming from a recorded file. It also seems node isn&#39;t sending the rest of the FFMPEG stream to the client, yet I can see the FFMPEG data in the .on event routine so it is getting to the FFMPEG node HTTP server.

6) Although I think piping the STDOUT stream to the HTTP response buffer should work, do I have to build an intermediate buffer and stream that will allow the HTTP partial content client requests to properly work like it does when it (successfully) reads a file? I think this is the main reason for my problems however I&#39;m not exactly sure in Node how to best set that up. And I don&#39;t know how to handle a client request for the data at the end of the file as there is no end of file.

7) Am I on the wrong track with trying to handle 206 partial content requests, and should this work with normal 200 HTTP responses? HTTP 200 responses works fine for VLC so I suspect the HTML5 video client will only work with partial content requests?

As I&#39;m still learning this stuff its difficult to work through the various layers of this problem (FFMPEG, node, streaming, HTTP, HTML5 video) so any pointers will be greatly appreciated. I have spent hours researching on this site and the net, and I have not come across anyone who has been able to do real time streaming in node but I can&#39;t be the first, and I think this should be able to work (somehow!).

Best approach to real time http streaming to HTML5 video client

I&#39;m building a web app that should play back an RTSP/RTP stream from a server &lt;strike&gt;http://lscube.org/projects/feng&lt;/strike&gt;.

Does the HTML5 video/audio tag support the rtsp or rtp? If not, what would the easiest solution be? Perhaps drop down to a VLC plugin or something like that.

Streaming via RTSP or RTP in HTML5

I have searched the internet for days now on how to implement a video streaming feature from an android phone to another android phone over a WiFi connection but I can&#39;t seem to find anything useful. I looked on android developers for sample code, stackoverflow, google, android blogs but nothing. All I can find are some sort of phone-to-desktop or desktop-to-phone solutions for streaming, but nothing that I can borrow in my implementation. 

I need to control a robot using an arduino ADK, so I am using 2 phones, one which will be mounted on the robot and another which will receive the video stream from the robot. I am mentioning this because I am trying to achieve the smallest delay between the broadcast time and the viewing time.

I am writing 2 apps, one master app to control the robot(from the handheld phone) which will control the slave app and receive the stream, and the second slave app which will run on the robot-strapped phone, controlling the motors/actuators/streaming to master app. I can not use third party apps unfortunately. I need to integrate the video stream code into my 2 apps.

What options are there for achieving this? Also is it very hard to do because I never worked with videostreaming, tough I am doing pretty good in both Java and Android development. How should I encode/decode the stream, how do I initiate the connection, will I need to work with UDP instead of TCP/IP ? I really don&#39;t know where to start, with no sample code anywhere. I am pretty sure this can be achieved. I just can&#39;t find anything useful to get me started in the right direction.

I stumbled across [spydroid][1]  but it is using VLC on a desktop so its no good for me.

---
**EDIT:** Check out Cagney Moreau&#39;s [blog](http://cagneymoreau.com/stream-video-android/). He goes into details about implementing this.

  [1]: http://code.google.com/p/spydroid-ipcamera/

Live-stream video from one android phone to another over WiFi

     private void StartReceivingData(string ipAddress, int iPort)
        {
            try
            {
                if (!_bContinueReciving)
                {
                    //initializeMainSocket(ipAddress, iPort);
                    _mSocket = new Socket(AddressFamily.InterNetwork, SocketType.Raw, ProtocolType.IP);//&lt;------HERE IS RAISED THE EXCEPTION
                    _mSocket.Bind(new IPEndPoint(IPAddress.Parse(ipAddress), iPort));
                    //  _mSocket.Bind(new IPEndPoint(IPAddress.Loopback, iPort));
                    _mSocket.SetSocketOption(SocketOptionLevel.IP, SocketOptionName.HeaderIncluded, true);
                    _mSocket.IOControl(IOControlCode.ReceiveAll, new byte[4] { 1, 0, 0, 0 }, new byte[4] { 0, 0, 0, 0 });
                    //var 1
                    _mSocket.BeginReceive(_buffReceivedData, 0, _buffReceivedData.Length, SocketFlags.None,
                                         new AsyncCallback(OnReceive), null);
                    initializeLocalSocket();
                }
                else
                {
                    _bContinueReciving = false;
                    _mSocket.Close();
                }
            }
            catch (Exception exception)
            {
                Debug.WriteLine(exception);
            }
        }

I don&#39;t understand why...it worked and now it doesn&#39;t. could someone help me?
i am streaming with vlc, and i wan&#39;t to receive the packets, do some reports, and then restream local to a player 

An attempt was made to access a socket in a way forbidden by its access permissions. Why?

With a RESTful service you can create, read, update, and delete resources.  This all works well when you&#39;re dealing with something like a database assets - but how does this translate to streaming data?  (Or does it?) For instance, in the case of video, it seems silly to treat each frame as resource that I should query one at a time.  Rather I would set up a socket connection and stream a series of frames.  But does this break the RESTful paradigm?  What if I want to be able to rewind or fast forward the stream?  Is this possible within the RESTful paradigm?  So: **How do streaming resources fit within the RESTful paradigm?**

As a matter of implementation, I am getting ready to create such a streaming data service, and I want to make sure I&#39;m doing it the &quot;best way&quot;.  I&#39;m sure this problem&#39;s been solved before.  Can someone point me to good material?


How do streaming resources fit within the RESTful paradigm?

I&#39;ve got a browser game and I have recently started adding audio to the game. 

Chrome does not load the whole page and gets stuck at `&quot;91 requests | 8.1 MB transferred&quot;` and does not load any more content; and it even breaks the website in all other tabs, saying `Waiting for available socket`. 

After 5 mins (exactly) the data are loaded. 

![enter image description here][1]

![enter image description here][2]

This does not happen on any other browser.

Removing one MP3 file (the latest added one) fixed the problem, so is it perhaps a data limit problem?


  [1]: http://i.stack.imgur.com/ZutCF.png
  [2]: http://i.stack.imgur.com/FEW4q.png

Chrome hangs after certain amount of data transfered - waiting for available socket

I was wondering if there&#39;s a difference between them when it comes to streaming videos.

I know `VideoView` can be used for streaming and what is `Mediaplayer` for? As far as I know, `MediaPlayer` can do the same thing as `VideoView` right?

Can anyone give me the answer?

And if I want to stream video from the server by using RTSP to Android, which one should I start with? `VideoView` or `MediaPlayer`?

Any suggestion?

What is the difference between MediaPlayer and VideoView in Android

I&#39;m trying to setup a very basic html5 page that loads a .mp4 video that is 20MB.  It appears that the browser needs to download the entire thing rather than just playing the first part of the video and streaming in the rest.

[This post][1] is the closest thing I&#39;ve found while searching... I tried both Hand Brake and Data Go Round by neither appeared to make a difference: 

Any ideas on how to do this or if it&#39;s possible?

Here is the code I&#39;m using:

&lt;!-- language: lang-html --&gt;

    &lt;video controls=&quot;controls&quot;&gt;
        &lt;source src=&quot;/video.mp4&quot; type=&quot;video/mp4&quot; /&gt;
        Your browser does not support the video tag.
    &lt;/video&gt;


  [1]: https://stackoverflow.com/questions/6683418/inline-video-large-file-with-buffer

HTML5 - How to stream large .mp4 files?

I&#39;m trying to stream some youTube videos using the MPMoviePlayerController but I&#39;m having some problems.  The code I&#39;m using is pretty simple and I can play .m4v videos by passing a URL to initWithContentURL.  When I launch the movie player the player comes up but just goes away after about 20 seconds.  When I try it in the simulator I get an alert view that says the server is not configured correctly.  Is there an argument I need to pass with the URL to get a specific type of video feed from google?

    NSURL *videoURL = [NSURL URLWithString:@&quot;http://www.youtube.com/v/HGd9qAfpZio&amp;hl=en_US&amp;fs=1&amp;&quot;];
    MPMoviePlayerController *moviePlayer;
    moviePlayer = [[MPMoviePlayerController alloc] initWithContentURL:videoURL];
    
    [moviePlayer play];

I&#39;ve also tried the following URL&#39;s
http://www.youtube.com/watch?v=HGd9qAfpZio

I have also seen the argument &amp;format=1 and tried to add that to the end of both of the strings but no luck.

Play YouTube videos with MPMoviePlayerController instead of UIWebView

I need to stream a file which will result in save as prompt in the browser.
The issue is, the directory that the file is located is virtually mapped, so I am unable to use Server.MapPath to determine it&#39;s actual location. The directory is not in the same location (or even phyical server on the live boxes) as the website.


I&#39;d like something like the following, but that will allow me to pass a web URL, and not a server file path.

I may have to end up building my file path from a config base path, and then append on the rest of the path, but hopefully I can do it this way instead.

    var filePath = Server.MapPath(DOCUMENT_PATH);

    if (!File.Exists(filePath))
        return;

    var fileInfo = new System.IO.FileInfo(filePath);
    Response.ContentType = &quot;application/octet-stream&quot;;
    Response.AddHeader(&quot;Content-Disposition&quot;, String.Format(&quot;attachment;filename=\&quot;{0}\&quot;&quot;, filePath));
    Response.AddHeader(&quot;Content-Length&quot;, fileInfo.Length.ToString());
    Response.WriteFile(filePath);
    Response.End();

Download/Stream file from URL - asp.net

I&#39;m thinking about developing a streaming server and I have the following question, do over RTSP (example url: `rtsp://192.168.0.184/myvideo.mpg`) or `RTP` (example url: `rtp://192.168.0.184`).

As I have understood, an `RTSP` server is mainly used for streaming of files that already exist, ie, not live. `RTP` server is used to broadcast.

Somebody correct me if I&#39;m wrong, am I right?.

What I want to develop a server to broadcast live content on the computer screen, that is, which is displayed at the time that is broadcast in streaming.


What is the difference between RTP or RTSP in a streaming server?

Is there a streaming API for JSON?

I am trying to use HTTP Live Streaming (HLS) to stream video to my computers and my iPhone.  After reading through the Apple &#39;HTTP Live Streaming Overview&#39; as well as &#39;Best Practices for Creating and Deploying HTTP Live Streaming Media for the iPhone and iPad&#39;, I am a bit stuck.

I took my source file (an mkv) and used ffmpeg to encode the file the MPEG-TS format and Apple-recommended settings and a Baseline 3.0 profile:

    ffmpeg -i &quot;example.mkv&quot; -f mpegts -threads:v 4 -sws_flags bicubic -vf &quot;scale=640:352,setdar=16/9,ass=sub.ass&quot; -codec:v libx264 -r 29.970 -b:v 1200k -profile:v baseline -level:v 3.0 -movflags faststart -coder 1 -flags +loop -cmp chroma -partitions +parti8x8+parti4x4+partp8x8+partb8x8 -me_method hex -subq 6 -me_range 16 -g 239 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -b_strategy 1 -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -direct-pred 1 -fast-pskip 1 -af &quot;aresample=48000&quot; -codec:a libvo_aacenc -b:a 96k -ac 2  -y &quot;output.ts&quot;

No worries there.  I used a [pre-compiled segmenting tool][1] to segment the video and build a .m3u8 playlist.  The resultant file looked like this:

    #EXTM3U
    #EXT-X-TARGETDURATION:10
    #EXTINF:10,
    http://localhost/media/stream/stream-1.ts
    #EXTINF:10,
    http://localhost/media/stream/stream-2.ts
    #EXTINF:10,
    http://localhost/media/stream/stream-3.ts
    #EXT-X-ENDLIST

I checked that against some [Example Playlist Files for use with HTTP Live Streaming][2], and I don&#39;t see any issues.  I also tried playing the .m3u8 file in VLC, and it works like a charm.

I created an HTML page to play the file:

    &lt;html lang=&quot;en&quot;&gt;
    	&lt;head&gt;
    		&lt;meta charset=utf-8/&gt;
    	&lt;/head&gt;
    	&lt;body&gt;
    		&lt;div id=&#39;player&#39;&gt;
    			&lt;video width=&quot;352&quot; height=&quot;288&quot; src=&quot;stream.m3u8&quot; controls autoplay&gt;
    			&lt;/video&gt;
    		&lt;/div&gt;
    	&lt;/body&gt;
    &lt;/html&gt;

And this page does not work in Chrome, Safari, on my iPhone.  The html5 video tag examples on w3schools work fine on my computer, and the official Apple overview mentioned above gives an HTML example very similar to my page.  Nevertheless, my video player is completely unresponsive when I visit my own .m3u8 page.



  [1]: https://code.google.com/p/httpsegmenter/
  [2]: https://developer.apple.com/library/ios/technotes/tn2288/_index.html

Playing m3u8 Files with HTML Video Tag

I&#39;m using flask for my application. I&#39;d like to send an image (dynamically generated by PIL) to client without saving on disk.

Any idea how to do this ? 

How to send image generated by PIL to browser?

I am trying to download images from the site. The code which I am using is working fine while the image is available. If the image it not available it is creating a problem. How to validate availability of the image?

Code:

Method 1:

    WebRequest requestPic = WebRequest.Create(imageUrl);

    WebResponse responsePic = requestPic.GetResponse();

    Image webImage = Image.FromStream(responsePic.GetResponseStream()); // Error

    webImage.Save(&quot;D:\\Images\\Book\\&quot; + fileName + &quot;.jpg&quot;);

Method 2:

    WebClient client = new WebClient();
    Stream stream = client.OpenRead(imageUrl);

    bitmap = new Bitmap(stream); // Error : Parameter is not valid.
    stream.Flush();
    stream.Close();
    client.dispose();

    if (bitmap != null)
    {
        bitmap.Save(&quot;D:\\Images\\&quot; + fileName + &quot;.jpg&quot;);
    }


**Edit:**

  Stream has the following statements:

          Length  &#39;((System.Net.ConnectStream)(str)).Length&#39; threw an exception of type  &#39;System.NotSupportedException&#39;    long {System.NotSupportedException}
        Position  &#39;((System.Net.ConnectStream)(str)).Position&#39; threw an exception of type &#39;System.NotSupportedException&#39;    long {System.NotSupportedException}
     ReadTimeout  300000    int
    WriteTimeout  300000    int


Download image from the site in .NET/C#

I&#39;m trying to get the &#39;hello world&#39; of streaming responses working for Django (1.2).  I figured out how to use a generator and the `yield` function.  But the response still not streaming.  I suspect there&#39;s a middleware that&#39;s mucking with it -- maybe ETAG calculator?  But I&#39;m not sure how to disable it.  Can somebody please help?

Here&#39;s the &quot;hello world&quot; of streaming that I have so far:

    def stream_response(request):
        resp = HttpResponse( stream_response_generator())
        return resp
    
    def stream_response_generator():
        for x in range(1,11):
            yield &quot;%s\n&quot; % x  # Returns a chunk of the response to the browser
            time.sleep(1)



How to stream an HttpResponse with Django

For an application I&#39;m working on, I need to allow the user to upload very large files--i.e., potentially many gigabytes--via our website.  Unfortunately, ASP.NET MVC appears to load the entire request into RAM before beginning to service it--not exactly ideal for such an application.  Notably, trying to circumvent the issue via code such as the following:

    if (request.Method == &quot;POST&quot;)
    {
        request.ContentLength = clientRequest.InputStream.Length;
        var rgbBody = new byte[32768];

        using (var requestStream = request.GetRequestStream())
        {
            int cbRead;
            while ((cbRead = clientRequest.InputStream.Read(rgbBody, 0, rgbBody.Length)) &gt; 0)
            {
                fileStream.Write(rgbBody, 0, cbRead);
            }
        }
    }

fails to circumvent the buffer-the-request-into-RAM mentality.  Is there an easy way to work around this behavior?

Streaming large file uploads to ASP.NET MVC

I&#39;ve been experimenting with binary streams in Node.js, and much to my amazement do actually have a working demo of taking a Shoutcast stream using node-radio-stream and pushing it into a HTML5 &lt;audio&gt; element using chunked encoding. But it only works in Safari!

Here is my server code:

    var radio = require(&quot;radio-stream&quot;);
    var http = require(&#39;http&#39;);
    var url = &quot;http://67.205.85.183:7714&quot;;
    var stream = radio.createReadStream(url);

    var clients = [];

    stream.on(&quot;connect&quot;, function() {
      console.error(&quot;Radio Stream connected!&quot;);
      console.error(stream.headers);
    });


    // When a chunk of data is received on the stream, push it to all connected clients
    stream.on(&quot;data&quot;, function (chunk) {
        if (clients.length &gt; 0){
            for (client in clients){
                clients[client].write(chunk);
            };
        }
    });

    // When a &#39;metadata&#39; event happens, usually a new song is starting.
    stream.on(&quot;metadata&quot;, function(title) {
      console.error(title);
    });

    // Listen on a web port and respond with a chunked response header. 
    var server = http.createServer(function(req, res){ 
        res.writeHead(200,{
            &quot;Content-Type&quot;: &quot;audio/mpeg&quot;,
            &#39;Transfer-Encoding&#39;: &#39;chunked&#39;
        });
        // Add the response to the clients array to receive streaming
        clients.push(res);
        console.log(&#39;Client connected; streaming&#39;); 
    });
    server.listen(&quot;8000&quot;, &quot;127.0.0.1&quot;);

    console.log(&#39;Server running at http://127.0.0.1:8000&#39;); 

My client code is simply:

    &lt;audio controls src=&quot;http://localhost:8000/&quot;&gt;&lt;/audio&gt;

This works fine in Safari 5 on the Mac, but doesn&#39;t seem to do anything in Chrome or Firefox. Any ideas? 

Possible candidates including encoding issues, or just partially-implemented HTML5 features...

Streaming audio from a Node.js server to HTML5 &lt;audio&gt; tag

I want to make a simple server application where people can connect using a browser-based client (which I will make later) to watch streaming video. And I want to use C#.

What do I need to capture video or rapid images through a webcam and send them over the network?


How can I stream webcam video with C#?

I use the following code to stream large files from the Internet into a local file:

    fp = open(file, &#39;wb&#39;)
    req = urllib2.urlopen(url)
    for line in req:
        fp.write(line)
    fp.close()

This works but it downloads quite slowly. Is there a faster way? (The files are large so I don&#39;t want to keep them in memory.)


Stream large binary files with urllib2 to file

I am finding big differences in the time it takes the Android MediaPlayer to prepare for live stream playback with different streams.

**The hard data**

I added logging between prepareAsync() and the onPrepared(MediaPlayer mp) callback and tested several streams a few times each.  The times for each stream were very consistent (+/- one second), and here are the results:

1. MPR news stream: 27 seconds (http://newsstream1.publicradio.org:80/)
2. MPR classical music stream: 15 seconds (http://classicalstream1.publicradio.org:80/)
3. MPR The Current stream: 7 seconds (http://currentstream1.publicradio.org:80/)
4. PRI stream: 52 seconds (http://pri-ice.streamguys.biz/pri1)

The tests were performed on a Nexus S with Android 2.3.4 on a 3G connection (~1100 Kbps).

Playing non-streaming MP3 audio files is not an issue.

Here are snippets of how I am playing the streams:

Prepare MediaPlayer:

    ...
    mediaPlayer.setDataSource(playUrl);
    mediaPlayer.setAudioStreamType(AudioManager.STREAM_MUSIC);
    mediaPlayer.prepareAsync();
    ...

Then in onPrepared(MediaPlayer mp):

    mediaPlayer.start();

Why does it take so long to prepare some streams but not others?  The above data seems to suggest that it *might* be based on the *amount* of data that has been buffered and not the *duration* of the buffered audio content.  Could this really be?

**Update:** I have tested live streaming on physical devices with Android 1.6, 2.2 and 2.3.4 and emulators with 1.6, 2.1, 2.2, 2.3.1 and 2.3.3.  I am only seeing the long delay on 2.3.3 and 2.3.4.  The older versions start playback within 5 seconds.

Why does it take so long for Android&#39;s MediaPlayer to prepare some live streams for playback?

The [documentation for AVPlayer][1] states the following:

&gt; [The] player works equally well with local and remote media files

However, the [documentation for AVAudioPlayer][2] states the following:

&gt; Apple recommends that you use this class for audio playback unless you are playing audio captured from a network stream

For the work I am doing I need some of the capabilities of AVAudioPlayer, but all my audio is being streamed. The main thing I need from AVAudioPlayer that AVPlayer does not have is the &quot;playing&quot; property. It is difficult to build a player UI without that property, among others.

So what is the difference between AVPlayer and AVAudioPlayer that makes the latter unsuitable for network streaming? Is there a way to get some of the info from AVPlayer that AVAudioPlayer provides such as the &quot;playing&quot; property?

  [1]: https://developer.apple.com/reference/avfoundation/avplayer
  [2]: https://developer.apple.com/reference/avfoundation/avaudioplayer

AVPlayer vs. AVAudioPlayer

I am trying to convert a live rtmp stream to hls stream on real time.

I got some idea after reading

http://sonnati.wordpress.com/2011/08/30/ffmpeg-%E2%80%93-the-swiss-army-knife-of-internet-streaming-%E2%80%93-part-iv/

i am able to convert the live rtmp stream to hls but not at run time. when i run the command and test for any hsl files (.m3u8 and .ts) i am not able to see but when i interrupt the command and check there i get the hls files as required.

I searched on google for solution but not able to get proper answer.

Can any body help me? 

Thanks in Advance...

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