I am playing a remote webrtc stream.
When I run that:
remoteStream.getVideoTracks()[0].stop();
remoteStream.getAudioTracks()[0].stop();
Video stops but audio continues: any idea ?
ID.png
Working fine in Firefox, may be a bug in chrome.
After calling remoteStream.getAudioTracks()[0].stop();
Track readyState is changing from live to ended.
You can report # https://bugs.chromium.org/p/webrtc/issues/list
For time being you can set audio/video tag volume as follows
document.getElementById("remotevideo").volume = 0;
Related
I create a WebRTC with simple peer. I need to capture stream from video element and send it to peers. I use vidEl.captureStream() to get Media Stream from video element and send it to other peer. It's working but only audio and video is blackscreen(not something is display).
[Testing] I create another video element on same origin of first video element and use captureStream() and test setObject with Media Stream captured and it's working.
How to send MediaStream from video.captureStream() to new peer through WebRTC and display video from remote peer.
Sorry for bad english.
https://webrtc.github.io/samples/src/content/capture/video-pc/
This sample does'nt work on chrome, edge but fire is worked. I don't know why.
There is currently an open bug in Chrome which breaks this:
https://bugs.chromium.org/p/chromium/issues/detail?id=1156408
I'm running a VueJS application that displays a full screen story of videos. I don't create as many tag as number of media in my story : I'm just changing component video sources each time I play a new video.
But it looks like Safari (Desktop & mobile) still does not cache HTML video once loaded : when I'm playing again a previous media, Safari is downloading again the asset. Instead of getting from cache like Chrome does.
The same issue has already been reported here but sill no correct answer.
Safari even stops downloading the final bytes video (producing a sort of timeout) when we go back and forth quicky in the story, so the story looks stuck.
Here's an example link.
Does anyone know a good alternative that avoids re-downloading video data at each play on Safari ?
Partial solution
Found myself a workaround that works pretty well if video is small size - all video are less than 3Mb in my case.
The trick is to use js fetch API to download full video, then stream it into video tag.
const videoRequest = fetch("/path/to/video.mp4")
.then(response => response.blob());
videoRequest.then(blob => {
video.src = window.URL.createObjectURL(blob);
});
Contrary to video src attribute, fetch API will get video data from cache if the same video was already fetched before.
Here a codepen demo that can be tested in Safari desktop/mobile (when NOT in private mode).
Pro : Video are now pulled from cache in Safari !
Con : You can't start the video until full data has been downloaded. That's why this solution can be used only for small video (like < 5Mb), else your users may wait a while before being able to play the video.
I have created webrtc based video chat suing peerjs.
The local and remote video element is created using control:
local:
'video id= [local_peer_id] autoplay="true" controls="true">'
remote:
and
'video id= [remote_peer_id] autoplay="true" controls="true">'
Now during the video chat if local user mute auido remote user can not hear anything and its working perfect.
Problem is with the video. If local user pause his own video he can see the video is paused but remote user still can see his video live.
on the other hand if remote user pause his video, local user still can see his video live.
Any one tell what need to do to deploy the feathure
"Pause" and "resume" video that works real time for both peer?
You need to know the difference between the HTML tags and the WebRTC streams...
You can have streams running, without having them attached to any HTML tag, and the media can still be sent and received by each peer. So, each peer can attach the stream to a audio/video tag, and the tag will only act as a player that you use to play a stream that is already running.
So, if you mute the HTML tag, you will only be muting the player, and not the stream. If you want to make anything to have effect on the other peer, you need to do stuff on the stream or in the peer connection.
In particular, to mute and resume audio or video, you need to toggle the media tracks in the media stream
// create a button to toggle video
var button = document.createElement("button");
button.appendChild(document.createTextNode("Toggle Hold"));
button.onclick = function(){
mediaStream.getVideoTracks()[0].enabled =
!(mediaStream.getVideoTracks()[0].enabled);
}
To pause/resume audio, use getAudioTracks() instead.
calling mediaStream.stop() will stop the camera
where mediaStream is the stream that you got when calling getUserMedia
Here is a working example
mediaStream.getAudioTracks()[0].stop();
mediaStream.getVideoTracks()[0].stop();
Hope this will work with new standards. Its working fine in my app.
I am using Opentok JavaScript WebRTC library to host a 1-to-1 video chat (peer-to-peer).
I can see my peer's video and hear the audio flawlessly.
My wish is to record audio / video of other chat party (remote). For this purpose, I'm using RecordRTC.
I was able to record the video of other chat participant (video is outputted to HTML video element), but, so far, I have not succeeded in recording audio (a dead-silence .wav file is as far as I could get). Using Chrome Canary (30.0.1554.0). This is my method:
var clientVideo = $('#peerdiv video')[0];//peer's video (html element)
var serverVideo = $('#myselfdiv video')[0];//my video (html element)
var context = new webkitAudioContext();
var clientStream = context.createMediaStreamSource(clientVideo.webRTCStream);
var serverStream = context.createMediaStreamSource(serverVideo.webRTCStream);
webRTCStream is a custom property i assigned to HTMLVideoElement object by modifying source of opentok js library. It contains MediaStream object linked to respective < video > element.
var recorder = RecordRTC({
video: clientVideo,
stream: clientStream
});
recorder.recordAudio();
recorder.recordVideo();
Video is recorded. Audio file is also created, it has a length that is close to video's length, however, it's completely silent (and yes, there was a lot of noise making on the other side during recording)
I've tested this with video element which displays my webcam's video stream (and audio), and it worked: both audio and video were recorded:
...
var recorder = RecordRTC({
video: serverVideo,
stream: serverStream
});
...
Is there something special about streams originating from a remote location? Any guidance on this issue would be very helpful.
This is the same issue occurs in following situations...
If not a stereo audio (dual channel audio)...i.e. it is mono audio
If audio input channels are not equal to audio output channels
If audio input device is not the default device selected on chrome
I'm still trying to find the actual issue.
I added this experiment for testing purpose... see console...
https://webrtc-experiment.appspot.com/demos/remote-stream-recording.html
Updated at: Saturday, 1 February 2014, 09:22:04 PKT
Remote audio recording is not supported; and this issue is considered as low-priority wontfix:
Support feeding remote WebRTC MediaStreamTrack output to WebAudio
Connect WebRTC MediaStreamTrack output to Web Audio API
Updated at March 28, 2016
Remote audio+video recording is now supported in RecordRTC, since Chrome version 49+.
Firefox, on the other hand, can merely record remote-audio.
If Chrome/WebRTC/Opus outputs mono audio by default and if that is the problem here, I see two options in that case:
By making opus output stereo - not sure how.
By making the RecordRTC/Recorderjs code work with mono
Or does anyone know any other recording library that works?
This actually now works fine in Firefox. I am using FireFox 29.0.1 and the AudioAPI can now work with audio streams sources grabbed from remote parties from a peer connection.
To test go to Muaz Khan's experiment page. I am not sure with what version of Firefox this rolled out but I would like to thank the team for cranking it out!
The chrome bug was moved to the AudioAPI team cr bug to track progress
Using IE everything goes well (i got a mp4 video with priority execution setted on flash). when i try to view my video on chrome all seem to be good too, but in few seconds (not always but often) videojs crashes.
This is the log:
error:MediaError
code:3
What should i do?
According to HTML5 spec error code MEDIA_ERR_DECODE = 3 means that video was failed to be decoded. Chrome could reach broken frame e.g. video was encoded incorrectly.
In my experience HTML5 video decoding has poor support when it comes to error handling. Flash is more bullet-proof here. I got an example when Chrome (HTML5) failed to play video but it was playing fine in FF (flash). Converting to different pixel map (yuvj420p -> yuv420p) resolved my issue.
It means that first you should try encoding video with different options. As an option you can use ffmpeg