When i start my webcam in WebRTC, sound is going to loop and very tiny voice starting.
navigator.getUserMedia({audio: true, video: true}, function(stream){
window.localStream = stream;
stream.getAudioTracks()[0].enabled = false;
stream.getVideoTracks()[0].enabled = true;
$('#my-video').prop('src', URL.createObjectURL(stream));
}
This solution is disabling all audio. I just want to mute only "stream" audio not window.localSream so send audio remote.
Not quite sure what you mean by "sound is going to loop and very tiny voice starting". Do you mean the voice is too low and you hear echoing?
In your code, the audio track is still in your localStream, but just not rendered by disabling it, so the other end user will not hear any voice from you, you can also turn it on on fly but enable it, and your voice will be heard again.
To not include audio at all, you will have to set your constraints like {audio: false, video: true}
Related
My RTC session was started with text only. And video is added by user when needed (renegotiation)
navigator.getUserMedia({ video: true, audio: false }, function (myStream) {
localVideo[0].srcObject = myStream;
myConn.addStream(myStream);
}, function (error) {
console.log(error);
});
When user do not need the video session anymore, I remove using:
var tracks = localVideo[0].srcObject.getTracks();
tracks.forEach(function (t) {
t.stop();
});
myConn.removeStream(localVideo[0].srcObject);
localVideo[0].srcObject = null;
Everything is working fine, until I try to add the video again I noticed that the createOffer() request size is getting larger and larger.
Seems to me that WebRTC didn't forget about the previous stream, and is adding to the offer again and again. Or maybe my way of removing a video stream / track is wrong?
This is a known issue see this thread on the W3C list.
The best way to get around this is to use replaceTrack and is suggested in the thread.
Note: It is still possible to prevent the list of transceivers from growing
by *manually* recycling them using transceiver.sender.replaceTrack() and
transceiver.direction, but that still wastes resources on transceivers
currently not used, and implies you probably shouldn't use
transceiver.stop() in most cases.
Also see the "Unified Plan" Transition Guide
Safari doesn't support MediaRecorder to listen to the stream from WebCam like the below code.
This works perfectly in Chrome and I'm able to convert the blob to a webm video file.
if(navigator.mediaDevices.getUserMedia)
{
navigator.mediaDevices.getUserMedia({video: true, audio: true}).then (stream => {
videoRef.srcObject = stream
mediaRecorder.value = new MediaRecorder(stream, {mimeType: 'video/webm; codecs=vp8,opus'})
mediaRecorder.value.addEventListener('dataavailable', function(e) {
blobs.push(e.data)
})
})
}
})
I need to save the video streamed from webcam in my server. What should be the approach to achieve the same in Safari?
I researched a lot, saw a similar question. But there was no proper solution given.
Could someone guide to a tutorial on how to achieve this using WebRTC if needed?
I am trying to implement a toggle video feature using webRTC. Refer to the following code:
<video id="remote" autoPlay></video>
<button onclick="toggleVideo()">Toggle video</button>
let localVideo = document.querySelector('#local');
const toggleVideo = () => {
localVideo.srcObject.getVideoTracks()[0].enabled = !localVideo.srcObject.getVideoTracks()[0].enabled
}
This turns off video as well as webcam indicator in firefox but not in chrome. Chrome only turns off the video.
According to MDN docs,
If the MediaStreamTrack represents the video input from a camera, disabling the track by setting enabled to false also updates device activity indicators to show that the camera is not currently recording or streaming. For example, the green "in use" light next to the camera in iMac and MacBook computers turns off while the track is muted in this way.
MDN docs
Is there any other workaround?
I tested an audio and video conversation over the webRTC, using Chrome on 2 computers from two different networks. The quality of the video is perfect and does not crash, but the sound hangs all the time, can hear my own voice and does not have quality.
How can I improve?
Code:
var haveGum = navigator.mediaDevices.getUserMedia({fake:true, video:true})
.then(stream => streams[1] = stream)
.then(() => navigator.mediaDevices.getUserMedia({ video: true, audio: true }))
.then(stream => v1.srcObject = streams[0] = stream)
I found this answer from 2014, the audio was very good. But still with cuts and sometimes the voice becomes "robotic" and it is not possible to understand what is spoken.
Does SimpleWebRTC has this feature to get data(video/audio) without giving permission to browser to use my camera/microphone?
// create our webrtc connection
var webrtc = new SimpleWebRTC({
// the id/element dom element that will hold "our" video
localVideoEl: 'localVideo',
// the id/element dom element that will hold remote videos
remoteVideosEl: '',
// immediately ask for camera access
**autoRequestMedia: true,**
debug: true,
detectSpeakingEvents: true,
autoAdjustMic: false,
media: {
video: false,
**audio: true**
},
});
When I change those parts surrounded by asterisks to true it works, otherwise it doesn't.
Have you tried setting autoRequestMedia to true and while having both video and audio of the media object set to false? You should receive the readyToCall event and can join the room as shown on the simplewebrtc homepage.
First negotiate (accept the call/join the room) with video and audio and then disable the video, somehting like webrtc.videoStreams.disable()