I am Currently using a Voxl from the Company Modal AI.But it's documtation does not provide voxl implementation through rtmp or dash streaming protocol.Is there any way to implement rtmp or dash protocol through voxl?
I am Currently using a Voxl from the Company Modal AI.But it's documtation does not provide voxl implementation through rtmp or dash streaming protocol.Is there any way to implement rtmp or dash protocol through voxl?
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My setup is as follows:
OBS Studio to create the video feed
Ant Media Server to distribute the stream
Now I'm building an app that will display this stream and I'm currently using ExoPlayer, however I'm having a hard time getting it to work for both RTMP and HLS, I read some where that I could embed a webplayer in my app would that be easier? Here is my code for ExoPlayer:
//RTMP Url
String url = "rtmp://192.168.1.244/WebRTCApp/379358104902020985845622";
BandwidthMeter bandwidthMeter = new DefaultBandwidthMeter();
TrackSelection.Factory videoTrackSelectionFactory =
new AdaptiveTrackSelection.Factory();
TrackSelector trackSelector =
new DefaultTrackSelector(videoTrackSelectionFactory);
SimpleExoPlayer player = ExoPlayerFactory.newSimpleInstance(this, trackSelector);
PlayerView playerView = findViewById(R.id.simple_player);
playerView.setPlayer(player);
/*
Create RTMP Data Source
*/
RtmpDataSourceFactory rtmpDataSourceFactory = new RtmpDataSourceFactory();
MediaSource videoSource = new ExtractorMediaSource.Factory(rtmpDataSourceFactory)
.createMediaSource(Uri.parse(url));
player.prepare(videoSource);
player.setPlayWhenReady(true);
Any help on this would be much appreciated.
Most online video streaming use Adaptive Bit Rate streaming (ABR) protocols to deliver the video, mainly HSL and DASH this days.
Most Media players, like ExoPlayer, support these protocols well, although they are complex and evolving protocols so there are always edge cases.
Many video conferencing applications use WebRTC which is a real time optimised protocol - the usual approach is to use a WebRTC client for this type of stream.
The difference between the two approaches from a streaming latency point of view, at a very high level, is:
ABR protocols prioritise quality and avoiding interruptions and buffer enough of the video to try to gaurantee uninterrupted playback. They are usually aimed at movie and live video streaming services. Even for low latency implementation the latency is measured in multiple seconds and more.
WebRTC prioritises latency and sacrifices quality if necessary. It is aimed typically at real time sensitive applications like video conferencing where it is important not to fall behind the discussion even if it means a temporary video glitch or even brief interruption in video. Latency is usually measured in sub seconds.
Any Media Server comes from the WebRTC side although recent versions support HLS /CMAF and Low Latency DASH (these are still higher latency than WebRTC generally as noted above).
For your service, if you are able to use a DASH or HLS stream you may find that it is an easier path with ExoPlayer. If you look at the demo app for example you will see DASH and HLS streams but no RTMP ones. You can easily extend or modify the demo app to play your own HLS or DASH stream and this is often an easy way to start - look at the sample material in the assets/media.exolist.json and add your own URL:
https://github.com/google/ExoPlayer/blob/aeb306a164911aa1491b46c2db4da0d329c83c65/docs/demo-application.md
However, ExoPlayer should also support RTMP via an extension if this is your preferred route - there is a specific extension that allows this:
https://github.com/google/ExoPlayer/blob/0ba317b1337eaa789f05dd6c5241246478a3d1e5/extensions/rtmp/README.md
In theory you simply need to add this dependency to your application:
if your application is using DefaultDataSource or DefaultDataSourceFactory, adding support for RTMP streams is as simple as adding a dependency to the RTMP extension
It would be worth checking the issues list in this repository for any recent issues and/or workarounds.
as the title shows, is there any methods I can use to play multiple videos continuously using simple rtmp client(my rtmp server is wowza)? Here is the way I think:
when the first video is about to be finished，open a new thread to send a new createStream command and a new play command and get the video rtmp packet and put them into a buffer list, when the first video is finished, then play the video rtmp in the buffer list..
Can this way be available or are there any other recommended methods to achieve it? Any suggestion will be appreciated, thanks!
While the functionality is not built-in, Wowza does have a module called StreamPublisher to allow you to implement a server-side type of playlist. The source code for the module is available on GitHub. A scheduled playlist of multiple VOD files is streamed as a live stream, similar to a TV broadcast.
I am able to stream video with Kurento using WebRTC, I need to implement multi party audio conference using MCU feature of Kurento Media server. So audio coming from all clients are merged and send back that combined audio to all clients in efficient manner using WebRTC.
if it will works then we need only two connection(one for send and one for receive) other wise we need peer connection to all clients using WebRTC. It is not feasible to establish peer connection to all all clients.
Please suggest me any sample code which have implemented MCU for audio using Kurento Media Server or guide me to implement same using Kurento Media Server.
I'm afraid there's no code that allows that un Kurento. There is the Composite media element, but that is usually for audio AND video. It combines streams into a single stream matrix of the required size, usually more than 9 streams may have performance problems. If you only want to process audio, surely it could handle much more than 9 streams. To use only audio just connect AUDIO stream to the HubPort.
EDIT 1
The code to generate the media elements needed, and the correct way establish an audio-only connection is as follows.
WebRtcEndpoint webrtc = new WebRtcEndpoint.Builder(pipeline).build();
Composite composite = new Composite.Builder(pipeline).build();
HubPort hubport = new HubPort.Builder(composite).build();
webrtc.connect(hubport, MediaType.AUDIO);
Please note that the connection is from the WebRtcEndpoint to the HubPort. If you need it to be bidirectional, you'll need to connect that way also.
hubport.connect(webrtc, MediaType.AUDIO);
how to interrupt rtmp flv broadcast and publish live rtmp broadcast on red5?
I am using osmf strobe player. I have my flv playlist working but when I broadcast live from my webcam what is the formula to stop the flv streams, then play flv countdown video then connect live broadcast from web cam?
Here is how I would do it, from a high-level since I don't have the code for what you're asking and "easy" transitions between streams isn't built-in to the server.
First, create a signaling or event system within your app to accept actions triggered by your broadcaster. Using the signaling system, transition your subscribers / viewers by sending triggered events telling their players to stop playing a current video and start a new one. I suggest using Shared Objects for this when passing signals around. Use server-side methods called by your broadcaster to send the signals on the Share Object. The "play" functionality is the easy part since you simply provide the stream name in your signal / event.
I am working on a project (a digital camera) that should be able to take still images and short video clips and make those available to the host. As well as being able to stream live video.
Which USB class[es] should I use?
Should I use PTP (for still images and video) and USB Video Class for streaming?
Does PTP support transfer of video?
Does PTP support video streaming?
For static objects PTP is better since it does have a definition for an object. Current Canon DSLRs are also capable of "streaming" live video via standard PTP transaction mechanism. However, real streaming would benefit from isochronous transfer so UVC is better for streaming.
You can also have both classes implemented in a single device and have objects transferred bia PTP set of endpoints and stream via UVC endpoint.