I'm using Hls for streaming audio playlists with m3u8 format.
When I stop the music I run:
hls.stopLoad();
hls.detachMedia();
But I still see requests on network for the playlist/chunklist.
Is there a way to stop the requests other than refreshing/closing the window?
The player will continue to refresh until it encounters a ‘#EXT-X-ENDTAG’ in the playlist
Related
here is my question:
I use Android Apis "MediaMuxer" to combine h264 stream and aac stream to mp4 file,when I want to stop record,so I call this:mMediaMuxer.stop(),and the mp4 file can play well.
but sometimes,there is an unexpected happen,like kill power or power is gone suddendly,so there is no time to call "mMediaMuxer.stop()",finally this file can not play anyway....
Is anybody know now to fix this problem? I want to play this video event didn't call "mMediaMuxer.stop()" this method... or there is other Apis or sdk can combine h264+aac stream well?
I am building a YouTube video player in Flash / ActionScript 3.0, using their API.
However I am struggling to find a simple way of getting a play progress indicator. All I am looking for is a numerical display that shows time elapsed / total duration. It doesnt need to be a bar or anything.
After going through the YouTube API, I cant see any kind of event or method from which to get this kind of information.
Without any kind of event the best thing I can think of doing is creating a timer to mimick the time progress, pausing the timer when the video buffer event is triggered, but this seems a bit hacky and wrong.
What is the best way of getting the time elapsed data from the YouTube API?
Yes in the Youtube API there are such informations:
player.getDuration():Number
Returns the duration in seconds of the currently playing video. Note that getDuration() will return 0 until the video's metadata is loaded, which normally happens just after the video starts playing.
If the currently playing video is a live event, the getDuration() function will return the elapsed time since the live video stream began. Specifically, this is the amount of time that the video has streamed without being reset or interrupted. In addition, this duration is commonly longer than the actual event time since streaming may begin before the event's start time.
player.getCurrentTime():Number
Returns the elapsed time in seconds since the video started playing.
I've been playing around with node and websockets and built a small test app that streams audio using websockets. The server breaks apart the mp3 using createReadStream, throttles the stream using node-throttle and sens the binary data using the "ws" module.
On the client side I pick up the chunks on the websocket and use decodeAudioData (http://www.html5rocks.com/en/tutorials/webaudio/intro/) to decode and play the chunk. It all works relatively ok.
What I was curious to do next was to stream video in the same manner to the HTML5 video tag. But I can't really find any reference material on the web to achieve this in the same manner as my audio test above.
Is there a video equivalent for "decodeAudioData"?
Can I feed chunks of data into a video tag?
I've got a similar sample running that I picked up from...
https://gist.github.com/paolorossi/1993068
But this isn't really what I am looking for. First of all it doesn't really seem to be streaming to me. The client buffers it all before playing it.
Also, similar to my audio test I want the stream to be throttled on the server side so that when a new client connects they join the video at whatever point it is currently at. i.e. 30 minutes in or whatever.
Thanks
OK,
I found a solution to this after much searching.
The MediaSource API is what I was looking for...
var mediaSource = new MediaSource();
var sourceBuffer = mediaSource.addSourceBuffer('video/webm; codecs="vorbis,vp8"');
sourceBuffer.append(new Uint8Array(data));
This link provided the solution...
http://html5-demos.appspot.com/static/media-source.html
I'm trying to get a video stream RTP/RTCP using HTML5, the stream was generated by gstreamer. I used examples of gstreamer, so I can pass through RTP ports:5000, and RTCP:5001, and can receive streams using gstreamer. But using HTML5 could not receive them. So I tried to read a bit about HTML5 and saw that it can receive theora/ogg, webm/vp8, mp4/avc, and protocols may be, HTTP, RTP, RTCP, UDP, and others, but I could not use RTP, RTCP or UDP, HTTP only managed to receive. But I had a very satisfactory result using the VLC plugin for Mozilla Firefox, using the UDP protocol. I wonder if anyone has any tips, I do not want to use source files as src="/tmp/test.avi" needs to be a video stream that can be udp, RTP, RTCP. Thank you!
If you don't need to stream at low fps, you can use GStreamer to transcode your stream in MJPEG and stream it in TCP, and then use VLC to get this TCP stream and stream it to HTTP. It works very well (0.5 sec of latency), but if you decrease the fps (1 fps) VLC introduces a latency of around 11 sec.
Here are some test commands that should work out of the box, using the GStreamer videotestsrc :
GStreamer :
gst-launch -v videotestsrc horizontal-speed=1 ! deinterlace ! videorate ! videoscale ! video/x-raw-yuv, framerate=15/1, width=256,
height=144 ! jpegenc quality=20 ! multipartmux
boundary="--videoboundary" ! tcpserversink host=localhost port=3000
VLC :
vlc -vvv -I rc tcp://localhost:3000 --sout
'#standard{access=http{mime=multipart/x-mixed-replace;boundary=--7b3cc56e5f51db803f790dad720ed50a},mux=mpjpeg,dst=localhost:8081}'
then open a browser to http://localhost:8081 (or create an HTML page with an img tag whose "src" attribute is http://localhost:8081)
I'm trying to serve up a live stream (ie. completely buffered in memory, cannot access the past) and am having trouble with Expression Encoder 4.
Ideally, I'd like to just stream a bare H.264 byte stream to the client consumed by:
<video id="mainVideoWindow">
<source src='http://localhost/path/to/my/stream.mp4' type='video/mp4' />
</video>
I figured I could stream it to the client just like any other byte stream over HTTP. However, I'm having trouble figuring out the appropriate code required to do (first day with Expression Encoder, not sure how to go about getting the raw byte stream) so nor do I know if it would work in the first place.
An alternate was to use IIS Live Streaming server:
var source = job.AddDeviceSource(device, null);
job.ActivateSource(source);
job.ApplyPreset(LivePresets.VC1IISSmoothStreaming720pWidescreen);
var format = new PushBroadcastPublishFormat();
format.PublishingPoint = new Uri("http://localhost/test.isml");
job.PublishFormats.Add(format);
job.StartEncoding();
// Let's listen for a keypress or error message to know when to stop encoding
while (Console.ReadKey(true).Key != ConsoleKey.X) ;
// Stop our encoding
Console.WriteLine("Encoding stopped.");
job.StopEncoding();
However, I'm having trouble getting the client side markup to want to display the video on Chrome and I haven't seen anything to indicate that it'd work on Chrome (though http://learn.iis.net/page.aspx/854/apple-http-live-streaming-with-iis-media-services indicates how it would work with an iOS device).
Anyone have any insights?
You are trying to consume (with your sencond example) a Smooth Streaming feed (HTTP-Adaptive Streaming by Microsoft) through HTML5, which is not supported.
This could work on iOS devices if you enable the Apple HTTP Live Streaming to transmux the fragments into MPEG-2 Transport Stream. This will also generate an Apple HTTP Live Streaming manifest which than can be called though the video tag.
...I saw that you have the IIS link. The Apple HTTP Live Streaming needs to be enabled on the IIS Server (IIS Media Services). This will work for iOS devices. Quicktime will get into play...