mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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ed54734825
Fix audio streaming on Chrome, see https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1524 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6512>
309 lines
11 KiB
C#
309 lines
11 KiB
C#
using System;
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using static System.Diagnostics.Debug;
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using Gst;
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using WebSocketSharp;
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using Gst.WebRTC;
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using Newtonsoft.Json;
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using System.Net.Security;
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using System.Security.Cryptography.X509Certificates;
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using Gst.Sdp;
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using System.Text;
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using GLib;
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namespace GstWebRTCDemo
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{
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class WebRtcClient : IDisposable
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{
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const string SERVER = "wss://127.0.0.1:8443";
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const string PIPELINE_DESC = @"webrtcbin name=sendrecv bundle-policy=max-bundle
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videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay !
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queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 ! sendrecv.
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audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc perfect-timestamp=true ! rtpopuspay !
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queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! sendrecv.";
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readonly int _id;
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readonly int _peerId;
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readonly string _server;
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readonly WebSocket _conn;
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Pipeline pipe;
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Element webrtc;
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bool terminate;
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public WebRtcClient(int id, int peerId, string server = SERVER)
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{
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_id = id;
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_peerId = peerId;
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_server = server;
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_conn = new WebSocket(_server);
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_conn.SslConfiguration.ServerCertificateValidationCallback = validatCert;
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_conn.OnOpen += OnOpen;
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_conn.OnError += OnError;
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_conn.OnMessage += OnMessage;
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_conn.OnClose += OnClose;
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pipe = (Pipeline)Parse.Launch(PIPELINE_DESC);
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}
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bool validatCert(object sender, X509Certificate certificate, X509Chain chain, SslPolicyErrors sslPolicyErrors)
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{
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return true;
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}
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public void Connect()
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{
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_conn.ConnectAsync();
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}
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void SetupCall()
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{
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_conn.Send($"SESSION {_peerId}");
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}
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void OnClose(object sender, CloseEventArgs e)
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{
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Console.WriteLine("Closed: " + e.Reason);
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terminate = true;
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}
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void OnError(object sender, ErrorEventArgs e)
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{
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Console.WriteLine("Error " + e.Message);
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terminate = true;
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}
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void OnOpen(object sender, System.EventArgs e)
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{
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var ws = sender as WebSocket;
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ws.SendAsync($"HELLO {_id}", (b) => Console.WriteLine($"Opened {b}"));
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}
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void OnMessage(object sender, MessageEventArgs args)
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{
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var msg = args.Data;
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switch (msg)
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{
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case "HELLO":
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SetupCall();
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break;
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case "SESSION_OK":
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StartPipeline();
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break;
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default:
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if (msg.StartsWith("ERROR")) {
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Console.WriteLine(msg);
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terminate = true;
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} else {
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HandleSdp(msg);
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}
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break;
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}
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}
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void StartPipeline()
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{
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webrtc = pipe.GetByName("sendrecv");
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Assert(webrtc != null);
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webrtc.Connect("on-negotiation-needed", OnNegotiationNeeded);
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webrtc.Connect("on-ice-candidate", OnIceCandidate);
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webrtc.Connect("pad-added", OnIncomingStream);
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pipe.SetState(State.Playing);
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Console.WriteLine("Playing");
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}
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#region Webrtc signal handlers
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#region Incoming stream
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void OnIncomingStream(object o, GLib.SignalArgs args)
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{
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var pad = args.Args[0] as Pad;
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if (pad.Direction != PadDirection.Src)
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return;
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var decodebin = ElementFactory.Make("decodebin");
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decodebin.Connect("pad-added", OnIncomingDecodebinStream);
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pipe.Add(decodebin);
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decodebin.SyncStateWithParent();
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webrtc.Link(decodebin);
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}
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void OnIncomingDecodebinStream(object o, SignalArgs args)
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{
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var pad = (Pad)args.Args[0];
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if (!pad.HasCurrentCaps)
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{
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Console.WriteLine($"{pad.Name} has no caps, ignoring");
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return;
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}
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var caps = pad.CurrentCaps;
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Assert(!caps.IsEmpty);
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Structure s = caps[0];
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var name = s.Name;
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if (name.StartsWith("video"))
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{
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var q = ElementFactory.Make("queue");
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var conv = ElementFactory.Make("videoconvert");
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var sink = ElementFactory.Make("autovideosink");
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pipe.Add(q, conv, sink);
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pipe.SyncChildrenStates();
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pad.Link(q.GetStaticPad("sink"));
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Element.Link(q, conv, sink);
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}
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else if (name.StartsWith("audio"))
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{
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var q = ElementFactory.Make("queue");
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var conv = ElementFactory.Make("audioconvert");
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var resample = ElementFactory.Make("audioresample");
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var sink = ElementFactory.Make("autoaudiosink");
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pipe.Add(q, conv, resample, sink);
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pipe.SyncChildrenStates();
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pad.Link(q.GetStaticPad("sink"));
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Element.Link(q, conv, resample, sink);
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}
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}
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#endregion
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void OnIceCandidate(object o, GLib.SignalArgs args)
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{
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var index = (uint)args.Args[0];
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var cand = (string)args.Args[1];
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var obj = new { ice = new { sdpMLineIndex = index, candidate = cand } };
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var iceMsg = JsonConvert.SerializeObject(obj);
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_conn.SendAsync(iceMsg, (b) => { } );
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}
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void OnNegotiationNeeded(object o, GLib.SignalArgs args)
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{
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var webRtc = o as Element;
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Assert(webRtc != null, "not a webrtc object");
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Promise promise = new Promise(OnOfferCreated, webrtc.Handle, null); // webRtc.Handle, null);
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Structure structure = new Structure("struct");
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webrtc.Emit("create-offer", structure, promise);
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}
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void OnOfferCreated(Promise promise)
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{
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promise.Wait();
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var reply = promise.RetrieveReply();
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var gval = reply.GetValue("offer");
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WebRTCSessionDescription offer = (WebRTCSessionDescription)gval.Val;
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promise = new Promise();
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webrtc.Emit("set-local-description", offer, promise);
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promise.Interrupt();
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SendSdpOffer(offer) ;
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}
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#endregion
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void SendSdpOffer(WebRTCSessionDescription offer)
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{
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var text = offer.Sdp.AsText();
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var obj = new { sdp = new { type = "offer", sdp = text } };
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var json = JsonConvert.SerializeObject(obj);
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Console.Write(json);
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_conn.SendAsync(json, (b) => Console.WriteLine($"Send offer completed {b}"));
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}
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void HandleSdp(string message)
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{
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var msg = JsonConvert.DeserializeObject<dynamic>(message);
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if (msg.sdp != null)
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{
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var sdp = msg.sdp;
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if (sdp.type != null && sdp.type != "answer")
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{
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throw new Exception("Not an answer");
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}
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string sdpAns = sdp.sdp;
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Console.WriteLine($"received answer:\n{sdpAns}");
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SDPMessage.New(out SDPMessage sdpMsg);
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SDPMessage.ParseBuffer(ASCIIEncoding.Default.GetBytes(sdpAns), (uint)sdpAns.Length, sdpMsg);
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var answer = WebRTCSessionDescription.New(WebRTCSDPType.Answer, sdpMsg);
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var promise = new Promise();
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webrtc.Emit("set-remote-description", answer, promise);
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}
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else if (msg.ice != null)
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{
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var ice = msg.ice;
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string candidate = ice.candidate;
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uint sdpMLineIndex = ice.sdpMLineIndex;
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webrtc.Emit("add-ice-candidate", sdpMLineIndex, candidate);
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}
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}
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public void Run()
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{
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// Wait until error, EOS or State Change
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var bus = pipe.Bus;
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do {
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var msg = bus.TimedPopFiltered (Gst.Constants.SECOND, MessageType.Error | MessageType.Eos | MessageType.StateChanged);
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// Parse message
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if (msg != null) {
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switch (msg.Type) {
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case MessageType.Error:
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string debug;
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GLib.GException exc;
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msg.ParseError (out exc, out debug);
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Console.WriteLine ("Error received from element {0}: {1}", msg.Src.Name, exc.Message);
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Console.WriteLine ("Debugging information: {0}", debug != null ? debug : "none");
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terminate = true;
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break;
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case MessageType.Eos:
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Console.WriteLine ("End-Of-Stream reached.\n");
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terminate = true;
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break;
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case MessageType.StateChanged:
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// We are only interested in state-changed messages from the pipeline
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if (msg.Src == pipe) {
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State oldState, newState, pendingState;
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msg.ParseStateChanged (out oldState, out newState, out pendingState);
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Console.WriteLine ("Pipeline state changed from {0} to {1}:",
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Element.StateGetName (oldState), Element.StateGetName (newState));
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}
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break;
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default:
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// We should not reach here because we only asked for ERRORs, EOS and STATE_CHANGED
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Console.WriteLine ("Unexpected message received.");
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break;
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}
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}
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} while (!terminate);
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}
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public void Dispose()
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{
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((IDisposable)_conn).Dispose();
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pipe.SetState(State.Null);
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pipe.Dispose();
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}
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}
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static class WebRtcSendRcv
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{
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const string SERVER = "wss://webrtc.gstreamer.net:8443";
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static Random random = new Random();
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public static void Main(string[] args)
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{
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// Initialize GStreamer
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Gst.Application.Init (ref args);
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if (args.Length == 0)
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throw new Exception("need peerId");
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int peerId = Int32.Parse(args[0]);
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var server = (args.Length > 1) ? args[1] : SERVER;
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var ourId = random.Next(100, 10000);
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Console.WriteLine($"PeerId:{peerId} OurId:{ourId} ");
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var c = new WebRtcClient(ourId, peerId, server);
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c.Connect();
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c.Run();
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c.Dispose();
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}
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}
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}
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