apprtcdemo

How to get frame data in AppRTC iOS app for video modifications?

由 眉间皱痕 提交于 2021-02-07 20:31:52
问题 I am currently trying to make some modifications to the incoming WebRTC video stream in the AppRTC app for iOS in Swift (which in turn is based on this Objective-C version). To do so, I need access to the data which is stored in the frame objects of class RTCI420Frame (which is a basic class for the Objective-C implementation of libWebRTC). In particular, I need an array of bytes: [UInt8] and Size of the frames. This data is to be used for further processing & addition of some filters. The

Not able to compile WebRtc AppRTCDemo android application

由 主宰稳场 提交于 2019-12-25 07:42:20
问题 Not able to compile WebRtc AppRTCDemo Android application using Ninja tool. I am following instructions as given in this README file. I run this command sudo ninja -C out/Debug AppRTCDemo But it gets stuck on following lines: log: ninja version 0.1.3 initializing log: magic group: gid=0 (root) log: entering main loop log: generating initial pid array.. log: now monitoring process activity It never returns from this and I am not able to build the demo application. 回答1: Follow this example, i

Websocket open error, websocket register error

由 痴心易碎 提交于 2019-12-25 05:19:11
问题 I am facing web socket issue with https://appr.tc and unable to create or join the room. Is there any temporary break down at the server end? 回答1: its Apprtc signalling server issue. It will be back soon. Permament solution would be to set up your own Collider server Refer 来源： https://stackoverflow.com/questions/43752796/websocket-open-error-websocket-register-error

how to customise AppRTC to receive call

由 限于喜欢 提交于 2019-12-23 03:23:14
问题 I have integrated AppRTC in my project and I am able to join the WebRTC call in the room of https://apprtc.appspot.com using RoomName appClient = ARDAppClient(delegate: self) appClient?.createLocalMediaStream() appClient?.connectToRoomWithId(String(roomId), options: nil) I am able to hardcode a RoomName in my app and install the app in two devices A & B and if I make a call from both the devices at same time in A & B then I am able to successfully have a WebRTC call between them . Now I'd

ApprtcDemo with local server works between browsers but not Android native to browser

由 拟墨画扇 提交于 2019-12-17 22:19:39
问题 I am developing a chat application and done with it. Now I want to implement video chat also. After research a lot I decided to go with "WebRTC" library. What I have done? 1) Able to run AppRtcDemo at local server and Its working fine between browsers. Reference : http://www.webrtc.org/reference/getting-started 2) Able to build Android AppRtcDemo.But when I run it say "Cross origin does not support". After research I found in webrtc discussion that to resolve this issue I need to set-up own

Add dial/receive signalling mechanism for AppRTCDemo at client side

由 二次信任 提交于 2019-12-13 09:45:50
问题 Hi I am using AppRTCDemo and its working on their server. How ever the current mechanism is for exchanging chat-rooms name and entering the same room connects the peers. But I want to dial a call from one device to receive a call from other device and then peers should enter a room for video session , I have searched a lot , I have come up with for that I need signalling-server which I don't have and don't want to put hands on it , Now in this situation how can the other device know that

Apprtc with coturn STUN/TURN server

由 拈花ヽ惹草 提交于 2019-12-12 09:43:15
问题 Simply, I am going run locally popular example of WEBRTC app: github.com/webrtc/apprtc The apprtc installed, and even works locally without turn server ( "Same origin policy" don't allow use Google TURN server, which works only from apprtc.appspot.com: access-control-allow-origin:"https://apprtc.appspot.com"). But I know that in real internet world (nats and firewalls) I need turn server. So I have decided to use own STUN/TURN server: code.google.com/p/coturn/ I am trying integrate my apprtc

How to deploy Apprtc's collider into Google App Engine?

由 扶醉桌前 提交于 2019-12-11 12:25:51
问题 I have deployed the apprtc's google app engine code on my own GAE server. I don't find instructions on how to deploy the Go based collider for websocket signaling. Here is the collider project: https://github.com/webrtc/apprtc/tree/master/src/collider Any insights would help. 回答1: Collider needs to be deployed in Google Compute Engine or equivalent services by Amazon. Here is the list of steps that I had go through: 1) Install go from: https://storage.googleapis.com/golang/go1.5.1.linux-amd64

how to customise AppRTC to receive call

由 荒凉一梦 提交于 2019-12-09 08:13:26
I have integrated AppRTC in my project and I am able to join the WebRTC call in the room of https://apprtc.appspot.com using RoomName appClient = ARDAppClient(delegate: self) appClient?.createLocalMediaStream() appClient?.connectToRoomWithId(String(roomId), options: nil) I am able to hardcode a RoomName in my app and install the app in two devices A & B and if I make a call from both the devices at same time in A & B then I am able to successfully have a WebRTC call between them . Now I'd like to do a real VOIP call that is make a call from my app in device A and receive the call at my app in

How to modify (add filters to) the camera stream that WebRTC is sending to other peers/server

由 拥有回忆 提交于 2019-12-05 02:18:09
问题 Scope I am using RTCCameraPreviewView to show the local camera stream let videoSource = self.pcFactory.avFoundationVideoSource(with: nil) let videoTrack = self.pcFactory.videoTrack(with: sVideoSource, trackId: "video0") //setting the capture session to my RTCCameraPreviewView: (self.previewView as! RTCCameraPreviewView).captureSession = (videoTrack.source as! RTCAVFoundationVideoSource).captureSession stream = self.pcFactory.mediaStream(withStreamId: "unique_label") audioTrack = self
订阅 apprtcdemo