As I understand correctly we have video adapter that can change resolution.
How to get frame aspect ratio or resolution of outbound stream or inbound?
Browser somehow know resolution for decoder...so should be a solution for get this resolution in native webrtc or js browser side, but I can't find this info
Related
I want to know how to identify webrtc decoding type of safari browser？For example, H264, H265 etc...
Chrome has a webrtc tool "chrome://webrtc-internals/", while it didn't work in Safari. I also add debug menu to Safari, while I still can't run the webrtc log. Is there any solution to know the webrtc decoder of Safari browser on Mac and ios?
chrome's webrtc-internals page uses the RTCPeerConncetion.getStats() API and provides (roughly) the same data. You can get the codecId
https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats-codecid
from the inbound-rtp statistics and then look up the codec statistics in the getStats result.
https://webrtc.github.io/samples/src/content/peerconnection/change-codecs/ shows a sample for the outbound codec.
I have a transfer of a live video stream from a server to the javascript function of a client browser:
server: gstreamer x264enc-hardware ! whatever-I-want ! appsink
=== transfer of data stream with a proprietary protocol ===>
HTML5 browser client: javascript function receives data sent by the appsink
In other words, I'm trying to display a h264 live stream created on a server with a proprietary transfer protocol, the data re-appearing in a javascript function inside an HTML5 client browser.
I was thinking of using MediaSource MSE in the browser to decode h264 and display the image.
Note that the video stream settings (video only, resolution, bandwidth) are fixed and known on both sides. So, everything can be hard-coded and the purpose is not to implement a generic solution.
What could I use on the server side (replacement of the "whatever-I-want" gstreamer plugin) so that the work in the HTML5 browser is not too complicated?
One solution would be to do nothing on the server side and use the broadway.js library to decode NALU h264 in javascript but it obviously doesn't leverage video MediaSource and the decoding capability of the browser.
Could I use Gstreamer avmux_dash and hope that MediaSource can input the transmitted data?
Alternatively, how could I create "MP4 fragments" and could MediaSource read them "easily"?
One approach, that has been used by some major players in the past to translate from one streaming protocol to another, is to receive in your proprietary transfer protocol and then re-package into HLS or DASH on a local server on the device.
You can then stream from that local host to a regular HLS or DASh player on the device.
It sounds inefficient (it is inefficient) but it works, even on mobile devices which their lower processing and power capabilities.
I am sending camera video from android to a janus server, but when I check from website, the video is rotated, Do you know why and How should I fix this?
I had to change something in the room config
https://github.com/meetecho/janus-gateway/issues/1021#issuecomment-335792000
I've seen that kurento-room isn't able of managing a user that enters only with microphone(no webcam).
The user actually appears in the room, with a black screen in the place where the webcam is normally located, but no audio is received from him either.
Why can this be happening?
That's an error in the connection of WebRTC endpoints. The thing is that the endpoint only negotiates audio, but the connection was made with audio and video profiles, and the media server committed seppuku. It should be fixed by now, providing the right media profiles in the connect method.
I want to let WebRTC encoded and play h264(NAL) stream(local file).
In the WebRTC tutorial, getUserMedia is use for get local camera connecting to the system, I don`t know if the getUserMedia function support
capture the local stream file like h264 stream.
If it doesn't work that way, may be I should modify WebRTC source code(I'm studying it).
Here is the question, If i change WebRTC code, how can i integration the new code into browser? Made it a plugin?
Firefox supports an extension to the <video> element that you can use to do this.
First, set the source of a video element:
v1.src = "file:///...";
Then you can call the (currently prefixed) mozCaptureStream or mozCaptureStreamUntilEnded function to get a MediaStream.
stream = v1.mozCaptureStream();
The proposed specification.
Note however that you need to ensure that the file is same origin with respect to the page. The same origin rules for file:/// are probably going to cause issues. Otherwise your MediaStream isn't going to be accessible to you. One way to ensure that is not to set the location directly, but to load the file using an <input type="file"> element.
As noted in other answers, Firefox currently only supports the baseline profile of H.264.
First, you are right getusermedia will not work for you. However, there are a couple of options.
Hack a stream together using RTCDataChannel. Breaking up the media stream and delivering each packet and then handling it on the client side.
Take a look at this demo for precorded media streams. I do not believe that H264 is addressed but it could help you on your way(probably for Firefox only)
Use some sort of webrtc breaker/endpoint that is native to stream the file. I know specifically that others(including myself) have streamed H264 to Firefox through the Janus-Gateway
Couple of asides:
Firefox only supports Baseline profiles in streaming h264 for a webrtc peerconnection
Chrome does not support h264 for webrtc at all
Are you trying to have getUserMedia return a h.264 encoded stream?
In which case, today it will only be possible with Firefox today, under some specific environment (cisco 264 plugin installed) and only for the base profile.
Chrome promised in november to add this capacity, but there is no timeline that I know of Expect at least Q2 2015.
Using our (temasys) commercial plugin you will soon be able to do that in IE and Safari.
Those are the only options on client side I can think of. On server side you can use whatever you want to transcode, including janus, kurento, powermedia, licode/lynkia, ....
Note: using other means like Datachannel or WebSocket are ok to transfer files, but would greatly reduce the user experience as you would not have all the added recovery (and security) mechanisms included in SRTP, DTLS, and would also not have specific mistreated media enhancements that are in webRTC like jitter, buffers, netQ, ect ...