I am hearing my own voice come back from remote client during a webRTC call. What comes out of the speaker is picked up by the mic. I have applied echoCancellation:true and also tried it for false but there seems to be no effect at all.
After failing to make it work, I want to try custom echoCancellation using webAudio. I tried to find some code example on the internet but I failed. Can anyone help with an echo cancellation implementation using webAudio?
Thank you
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I am currently looking through the options of creating an offline indication for the nuxt/pwa project. Since this moment, app is running perfectly offline, but what I want to do is to push a small notification when there is no connection saying something simple such as "you are currently offline".
I can see that there are multiple ways of doing this such as writing the event listener directly in the default layout, but my question is which one is the most suitable and reliable for the nuxt setup.
I think you don't need to write your own event listener, as this seems to be taken care of by the nuxt already. The network status seems to be accessible via $nuxt helper's isOnline and isOffline properties. Check out this example:
https://nuxtjs.org/api/$nuxt/
I have not worked with this yet, but I think it might be what you are looking for.
Note: Make sure to copy the whole link, as stackoverflow cuts it off at /$nuxt.
How do I make an actual remote call?
I've followed the guide: https://codelabs.developers.google.com/codelabs/webrtc-web/#4
And gotten their example fully integrated in my application (Angular, TypeScript, multi webcam &etc).
How do I make the remote call actually remote? - I get the idea of a signalling server, but maybe someone can show with basic strings?
I found this, but it's not been updated in a while so I'm not sure what's still valid:
Found some nice sequence diagrams https://webrtc.org/native-code/native-apis/
Setup call
(source: webrtc.org)
Receive a Call
(source: webrtc.org)
Close Down a Call
(source: webrtc.org)
I am trying to write a small application usingwebrtc that can be used as a messaging/Chat application between 2 computers.
I see this:
http://simpl.info/rtcdatachannel/
and it is not working. any suggestions?
I wrote the simpl.info/rtcdatachannel example. It's only designed to show off data channels working within one page.
For a complete peer-to-peer messaging application, I suggest adding RTCDataChannel functionality to something like apprtc.appspot.com. You could also consider a readymade abstraction library like PeerJS or EasyRTC.
You might also want to take a look at the RTCPeerConnection/RTCDataChannel/signaling codelab I built.
In above example, from the trace log, the ice-candidates are generated, but they are either not exchanged between each other because of there may be problem in sending 'offer' or responding the 'answer'. Also above example works only in chrome( because of only webkitRTCPeerConnection is used, with mozRTCPeerConnection this can work on firefox also.
If you want to develope chat application for only text and not for the video chat, then you can use node-js & socket.io or websockets for this.
You may like :) following two libraries:
DataChannel.js / for webrtc data/text/file sharing (among multi-users)
RTCMultiConnection.js / for data as well as media (screen/audio/video/etc) sharing
Firebase.com is a "suggested" starting point for newcomers; that can be used for signaling. You just need to override "openSignalingChannel" and done!
You should use peer.js (https://github.com/peers/peerjs) or use peer chat (https://github.com/Hironate/PeerChat) if you want to do with node js.
Now since device enumeration is present in chrome, i know i can select a device during "getUserMedia" negotiation. I was also wondering whether i could switch devices during the middle of a call (queue up a local track and switch tracks or do i have to renegotiate the stream)? I am not sure if this is something that is still blocked or now is "allowable"
I have tried to make a new track, but i can't figure out how to switch the track on the go. I know this was previously impossible, but was wondering now if it is possible?
Even i have the same requirement. I have to record the video using MediaRecorder. For this I am using navigator.getUserMedia with constraints of audio and video. You can pass the video or audio tracks dynamically by getting the available devices from navigator.mediaDevices.enumerateDevices() and attaching the respective device to constraints and calling navigator.getUserMedia with new constraints again. The point to be noted when doing this is, you have to kill the existing tracks using track.stop() method.
You can see my example here.
StreamTrack's readyState is getting changed to ended, just before playing the stream (MediaStream - MediaStreamTrack - WebRTC)
In Firefox, you can use the RTPSender object to call replaceTrack() to replace a track on the fly (with no renegotiation). This should eventually be supported by other browsers as part of the spec.
Without replaceTrack(), you can remove the old stream, add a new one, deal with onnegotiationnedded, and let the client process the change in streams.
See the replaceTrack() test in the Mozilla source: https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpSender/replaceTrack
Have you tried calling getUserMedia() when you want to change to a different device?
There's an applyConstraints() method in the Media Capture and Streams spec that makes it possible to change constraints on the fly, but it hasn't been implemented yet:
dev.w3.org/2011/webrtc/editor/getusermedia.html#the-model-sources-sinks-constraints-and-states
dev.w3.org/2011/webrtc/editor/getusermedia.html#methods-1
Would anyone have an idea as to why an app would work on almost every phone that has 2.1 but not the Desire?
One of my apps uses voice input and the Desire is the only phone that force closes when the voice prompt comes up.
The worst part is that I don't know how to test this, I don't have one or know anyone who does.
Any ideas?
EDIT:
I finally found out that HTC disabled voice in the Desire and you have to do a work around to install it.
So if you are relying on voice input make sure you use the code in the google example to catch the error:
PackageManager pm = getPackageManager();
List<ResolveInfo> activities = pm.queryIntentActivities(
new Intent(RecognizerIntent.ACTION_RECOGNIZE_SPEECH), 0);
if (activities.size() == 0) {
noResults.setText("Voice input not found on this phone.");
}else{
//If voice is enabled
}
I think the most important thing to do first is to get the exception report. Since you can't test it by yourself, I would use a tool to get the exception report from your customers. In Android 2.2 the built-in tool can be used. If you have other targeting SDKs I would recommend this services: http://code.google.com/p/android-remote-stacktrace/ to get a remote stacktrace.
Then if you post the stacktrace here, I think somebody will be able to help you!