From 3fd0e265e42c62371cb7524a5c702ca3405d9944 Mon Sep 17 00:00:00 2001 From: mekya Date: Sat, 24 Aug 2024 03:06:20 +0300 Subject: [PATCH 01/34] Reset websocket connections flags to re-connect faster --- src/main/js/websocket_adaptor.js | 6 ++++++ src/main/webapp/conference.html | 4 ++-- src/main/webapp/samples/publish_webrtc.html | 13 +++++++++---- 3 files changed, 17 insertions(+), 6 deletions(-) diff --git a/src/main/js/websocket_adaptor.js b/src/main/js/websocket_adaptor.js index 6f230125..a34ecb60 100644 --- a/src/main/js/websocket_adaptor.js +++ b/src/main/js/websocket_adaptor.js @@ -19,6 +19,12 @@ export class WebSocketAdaptor { } this.initWebSocketConnection(); + + addEventListener("offline", (event) => { + this.connected = false; + this.connecting = false; + Logger.info("Network status has changed to offline. Resetting flags to reconnect faster"); + }); } /** diff --git a/src/main/webapp/conference.html b/src/main/webapp/conference.html index 3f224920..e1379fae 100644 --- a/src/main/webapp/conference.html +++ b/src/main/webapp/conference.html @@ -791,7 +791,7 @@

WebRTC Multitrack Conference

else if (info == "reconnection_attempt_for_player") { console.log("Reconnection attempt for player") if (playOnly && isNoSreamExist) { - console.log("Reconnection attempt for player with no stream existmfor play only mode.") + console.log("Reconnection attempt for player with no stream exist for play only mode.") } else { playReconnected = false; if (!reconnecting) { @@ -801,7 +801,7 @@

WebRTC Multitrack Conference

} else if (info == "reconnection_attempt_for_publisher") { console.log("Reconnection attempt for publisher") - publishReconnected = isPlayOnly; + publishReconnected = playOnly ? true : false; if (!reconnecting) { reconnectionInProgress(); } diff --git a/src/main/webapp/samples/publish_webrtc.html b/src/main/webapp/samples/publish_webrtc.html index 0a01fdeb..dad95560 100644 --- a/src/main/webapp/samples/publish_webrtc.html +++ b/src/main/webapp/samples/publish_webrtc.html @@ -304,16 +304,21 @@ // instantMeter.value = instantValueDisplay.innerText = value; //}, 200); - audioLevelTimerId = setInterval(() => { - webRTCAdaptor.remotePeerConnection[streamId].getStats(null).then(stats => { + audioLevelTimerId = setInterval(() => + { + if (webRTCAdaptor.remotePeerConnection[streamId]) { + + webRTCAdaptor.remotePeerConnection[streamId].getStats(null).then(stats => { for (const stat of stats.values()) { - if (stat.type === 'media-source' && stat.kind === 'audio') { + if (stat.type === 'media-source' && stat.kind === 'audio' && stat.audioLevel) { instantMeter.value = instantValueDisplay.innerText = stat.audioLevel.toFixed(2); return; } } - }) + }); + + } }, 250); From e39713044f131215dc17efa157f13859fbe87a63 Mon Sep 17 00:00:00 2001 From: mekya Date: Sat, 24 Aug 2024 03:24:04 +0300 Subject: [PATCH 02/34] Increase test coverage --- src/test/js/webrtc_adaptor.test.js | 2722 ++++++++++++++-------------- 1 file changed, 1375 insertions(+), 1347 deletions(-) diff --git a/src/test/js/webrtc_adaptor.test.js b/src/test/js/webrtc_adaptor.test.js index eea4b4fa..1d19038e 100644 --- a/src/test/js/webrtc_adaptor.test.js +++ b/src/test/js/webrtc_adaptor.test.js @@ -1,1689 +1,1717 @@ -import {WebRTCAdaptor} from '../../main/js/webrtc_adaptor.js'; -import {MediaManager} from "../../main/js/media_manager.js"; -import {PeerStats} from "../../main/js/peer_stats.js"; +import { WebRTCAdaptor } from '../../main/js/webrtc_adaptor.js'; +import { MediaManager } from "../../main/js/media_manager.js"; +import { PeerStats } from "../../main/js/peer_stats.js"; -describe("WebRTCAdaptor", function () { +describe("WebRTCAdaptor", function() { - var clock; + var clock; - var sandbox; + var sandbox; - var initialized = false; + var initialized = false; - var currentTest; + var currentTest; - beforeEach(function () { - clock = sinon.useFakeTimers(); - sandbox = sinon.createSandbox(); + beforeEach(function() { + clock = sinon.useFakeTimers(); + sandbox = sinon.createSandbox(); - currentTest = this.currentTest; - console.log("**** starting test: ****", currentTest.title); - }); + currentTest = this.currentTest; + console.log("**** starting test: ****", currentTest.title); + }); - afterEach(() => { - console.log("**** ending test: ****", currentTest.title); - // Restore the default sandbox here - sinon.restore(); - clock.restore(); - sandbox.restore(); + afterEach(() => { + console.log("**** ending test: ****", currentTest.title); + // Restore the default sandbox here + sinon.restore(); + clock.restore(); + sandbox.restore(); - }); + }); - it("Initialize", async function () { + it("Initialize", async function() { - try { - var adaptor = new WebRTCAdaptor({}); - expect.fail("It should throw exception because websocket url is mandatory"); - } catch (err) { + try { + var adaptor = new WebRTCAdaptor({}); + expect.fail("It should throw exception because websocket url is mandatory"); + } catch (err) { - } + } - try { - var websocketURL = "ws://localhost"; - var adaptor = new WebRTCAdaptor({ - websocketURL: websocketURL - }); + try { + var websocketURL = "ws://localhost"; + var adaptor = new WebRTCAdaptor({ + websocketURL: websocketURL + }); - expect(adaptor.websocketURL).to.be.equal(websocketURL); - } catch (err) { - expect.fail(err); - } + expect(adaptor.websocketURL).to.be.equal(websocketURL); + } catch (err) { + expect.fail(err); + } - }); + }); - it("Auto reconnect play", async function () { + it("Auto reconnect play", async function() { - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); - var webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - var stopCall = sinon.replace(adaptor, "stop", sinon.fake()); + var webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + var stopCall = sinon.replace(adaptor, "stop", sinon.fake()); - var sendExpectation = webSocketAdaptor.expects("send"); - //sendExpectation first one is direct, second one through tryAgain - sendExpectation.exactly(2); + var sendExpectation = webSocketAdaptor.expects("send"); + //sendExpectation first one is direct, second one through tryAgain + sendExpectation.exactly(2); - var streamId = "stream123"; - expect(adaptor.remotePeerConnection[streamId]).to.be.undefined; + var streamId = "stream123"; + expect(adaptor.remotePeerConnection[streamId]).to.be.undefined; - //first call for sendExpectation direct - adaptor.play("stream123"); + //first call for sendExpectation direct + adaptor.play("stream123"); - expect(adaptor.remotePeerConnection[streamId]).to.not.be.undefined; + expect(adaptor.remotePeerConnection[streamId]).to.not.be.undefined; - clock.tick(2000); - expect(stopCall.called).to.be.false; - clock.tick(1000); - expect(stopCall.called).to.be.true; + clock.tick(2000); + expect(stopCall.called).to.be.false; + clock.tick(1000); + expect(stopCall.called).to.be.true; - expect(stopCall.calledWithMatch("stream123")).to.be.true; + expect(stopCall.calledWithMatch("stream123")).to.be.true; - adaptor.stop(streamId); + adaptor.stop(streamId); - expect(adaptor.remotePeerConnection[streamId]).to.not.be.undefined; + expect(adaptor.remotePeerConnection[streamId]).to.not.be.undefined; - //Add extra delay because play is called a few seconds later then the stop in tryAgain - clock.tick(1500); + //Add extra delay because play is called a few seconds later then the stop in tryAgain + clock.tick(1500); - sendExpectation.verify(); + sendExpectation.verify(); - }); + }); - it("Auto reconnect publish", async function () { + it("Auto reconnect publish", async function() { - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - onlyDataChannel: true - }); + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + onlyDataChannel: true + }); - var webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - var stopCall = sinon.replace(adaptor, "stop", sinon.fake()); + var webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + var stopCall = sinon.replace(adaptor, "stop", sinon.fake()); - var sendExpectation = webSocketAdaptor.expects("send"); - //sendExpectation first one is direct, second one through tryAgain - sendExpectation.exactly(2); + var sendExpectation = webSocketAdaptor.expects("send"); + //sendExpectation first one is direct, second one through tryAgain + sendExpectation.exactly(2); - var streamId = "stream1234"; - expect(adaptor.remotePeerConnection[streamId]).to.be.undefined; + var streamId = "stream1234"; + expect(adaptor.remotePeerConnection[streamId]).to.be.undefined; - //first call for sendExpectation direct - adaptor.publish(streamId); + //first call for sendExpectation direct + adaptor.publish(streamId); - expect(adaptor.remotePeerConnection[streamId]).to.not.be.undefined; + expect(adaptor.remotePeerConnection[streamId]).to.not.be.undefined; - clock.tick(2000); - expect(stopCall.called).to.be.false; - clock.tick(1000); - expect(stopCall.called).to.be.true; + clock.tick(2000); + expect(stopCall.called).to.be.false; + clock.tick(1000); + expect(stopCall.called).to.be.true; - expect(stopCall.calledWithMatch(streamId)).to.be.true; + expect(stopCall.calledWithMatch(streamId)).to.be.true; - adaptor.enableStats(streamId); - expect(adaptor.remotePeerConnectionStats[streamId]).to.not.be.undefined + adaptor.enableStats(streamId); + expect(adaptor.remotePeerConnectionStats[streamId]).to.not.be.undefined - expect(await adaptor.getStats(streamId)).to.be.true; + expect(await adaptor.getStats(streamId)).to.be.true; - console.log(adaptor.remotePeerConnectionStats[streamId]) + console.log(adaptor.remotePeerConnectionStats[streamId]) - adaptor.stop(streamId); + adaptor.stop(streamId); - expect(adaptor.remotePeerConnection[streamId]).to.not.be.undefined; - //Add extra delay because publish is called a few seconds later the stop in tryAgain method + expect(adaptor.remotePeerConnection[streamId]).to.not.be.undefined; + //Add extra delay because publish is called a few seconds later the stop in tryAgain method - clock.tick(1500); + clock.tick(1500); - sendExpectation.verify(); + sendExpectation.verify(); - }); + }); - it("toggleVideo", async function () { - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); + it("toggleVideo", async function() { + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); - let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - let streamId = "stream1"; - let trackId = "trackId"; - let enabled = true; + let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + let streamId = "stream1"; + let trackId = "trackId"; + let enabled = true; - let jsCmd = { - command: "toggleVideo", - streamId: streamId, - trackId: trackId, - enabled: enabled, - }; + let jsCmd = { + command: "toggleVideo", + streamId: streamId, + trackId: trackId, + enabled: enabled, + }; - let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); + let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); - adaptor.toggleVideo(streamId, trackId, enabled); + adaptor.toggleVideo(streamId, trackId, enabled); - sendExpectation.verify() - }) + sendExpectation.verify() + }) - it("Close websocket", async function () { - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); - let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - let closeExpectation = webSocketAdaptor.expects("close"); + it("Close websocket", async function() { + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); + let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + let closeExpectation = webSocketAdaptor.expects("close"); - let closePeerConnection = sinon.replace(adaptor, "closePeerConnection", sinon.fake()); + let closePeerConnection = sinon.replace(adaptor, "closePeerConnection", sinon.fake()); - let streamId = "stream123"; - expect(adaptor.remotePeerConnection[streamId]).to.be.undefined; - adaptor.initPeerConnection(streamId, "play"); - expect(adaptor.remotePeerConnection[streamId]).to.not.be.undefined; + let streamId = "stream123"; + expect(adaptor.remotePeerConnection[streamId]).to.be.undefined; + adaptor.initPeerConnection(streamId, "play"); + expect(adaptor.remotePeerConnection[streamId]).to.not.be.undefined; - adaptor.closeWebSocket(); + adaptor.closeWebSocket(); - expect(closePeerConnection.calledWithMatch(streamId)).to.be.true; + expect(closePeerConnection.calledWithMatch(streamId)).to.be.true; - closeExpectation.verify(); + closeExpectation.verify(); - }); + }); - it("Frequent try again call", async function () { - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); - let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - let closeExpectation = webSocketAdaptor.expects("close"); - let closePeerConnection = sinon.replace(adaptor, "closePeerConnection", sinon.fake()); + it.only("should set connected and connecting to false and log the correct message", function() { - const now = Date.now(); - adaptor.tryAgain(); + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); + let webSocketAdaptor = adaptor.webSocketAdaptor; - expect(adaptor.lastReconnectiontionTrialTime - now).to.be.at.most(100); + + webSocketAdaptor.connected = true; + webSocketAdaptor.connecting = true; + + expect(webSocketAdaptor.connected).to.be.true; + expect(webSocketAdaptor.connecting).to.be.true; + // Simulate offline event + const event = new Event("offline"); + window.dispatchEvent(event); - const lrt = adaptor.lastReconnectiontionTrialTime; + // Assertions + expect(webSocketAdaptor.connected).to.be.false; + expect(webSocketAdaptor.connecting).to.be.false; - for (let i = 0; i < 100; i++) { - adaptor.tryAgain(); - expect(adaptor.lastReconnectiontionTrialTime).to.be.equal(lrt); + }); - } - clock.tick(3000); - adaptor.tryAgain(); - expect(adaptor.lastReconnectiontionTrialTime).not.to.be.equal(lrt); - }); + it("Frequent try again call", async function() { + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); + let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + let closeExpectation = webSocketAdaptor.expects("close"); - it("Test reconnection process started callback", async function () { - var isReconnectionProcessStartedForPublisher = false; - var isReconnectionProcessStartedForPlayer = false; + let closePeerConnection = sinon.replace(adaptor, "closePeerConnection", sinon.fake()); - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true, - callback: (info, obj) => { - if (info === "reconnection_attempt_for_publisher") { - isReconnectionProcessStartedForPublisher = true; - } else if (info === "reconnection_attempt_for_player") { - isReconnectionProcessStartedForPlayer = true; - } - } - }); - var webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + const now = Date.now(); + adaptor.tryAgain(); - var closeExpectation = webSocketAdaptor.expects("close"); + expect(adaptor.lastReconnectiontionTrialTime - now).to.be.at.most(100); - var closePeerConnection = sinon.replace(adaptor, "closePeerConnection", sinon.fake()); + const lrt = adaptor.lastReconnectiontionTrialTime; - // some times Data.now() returns 0 and it is blocking the test - // so we set lastReconnectiontionTrialTime to -3000 to avoid this - adaptor.lastReconnectiontionTrialTime = -3000; + for (let i = 0; i < 100; i++) { + adaptor.tryAgain(); + expect(adaptor.lastReconnectiontionTrialTime).to.be.equal(lrt); - adaptor.publishStreamId = "testPublisher"; - adaptor.remotePeerConnection["testPublisher"] = sinon.mock(RTCPeerConnection); - adaptor.remotePeerConnection["testPublisher"].iceConnectionState = "disconnected"; + } - adaptor.playStreamId.push("testPlayer"); - adaptor.remotePeerConnection["testPlayer"] = sinon.mock(RTCPeerConnection); - adaptor.remotePeerConnection["testPlayer"].iceConnectionState = "disconnected"; + clock.tick(3000); + adaptor.tryAgain(); + expect(adaptor.lastReconnectiontionTrialTime).not.to.be.equal(lrt); + }); - adaptor.tryAgain(); + it("Test reconnection process started callback", async function() { + var isReconnectionProcessStartedForPublisher = false; + var isReconnectionProcessStartedForPlayer = false; - clock.tick(3000); + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true, + callback: (info, obj) => { + if (info === "reconnection_attempt_for_publisher") { + isReconnectionProcessStartedForPublisher = true; + } else if (info === "reconnection_attempt_for_player") { + isReconnectionProcessStartedForPlayer = true; + } + } + }); + var webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - expect(isReconnectionProcessStartedForPublisher).equal(true); - expect(isReconnectionProcessStartedForPlayer).equal(true); - }); + var closeExpectation = webSocketAdaptor.expects("close"); - it("Reconnection for play", async function () { - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); - var fakeSend = sinon.replace(adaptor.webSocketAdaptor, "send", sinon.fake()); + var closePeerConnection = sinon.replace(adaptor, "closePeerConnection", sinon.fake()); - const streamId = "test" + Math.floor(Math.random() * 100); - adaptor.playStreamId.push(streamId); - var mockPC = sinon.mock(RTCPeerConnection); - adaptor.remotePeerConnection[streamId] = mockPC - mockPC.iceConnectionState = "disconnected"; - mockPC.close = sinon.fake(); + // some times Data.now() returns 0 and it is blocking the test + // so we set lastReconnectiontionTrialTime to -3000 to avoid this + adaptor.lastReconnectiontionTrialTime = -3000; + adaptor.publishStreamId = "testPublisher"; + adaptor.remotePeerConnection["testPublisher"] = sinon.mock(RTCPeerConnection); + adaptor.remotePeerConnection["testPublisher"].iceConnectionState = "disconnected"; - clock.tick(3000); - adaptor.tryAgain(); - //Add extra delay because publish is called a few seconds later the stop in tryAgain method + adaptor.playStreamId.push("testPlayer"); + adaptor.remotePeerConnection["testPlayer"] = sinon.mock(RTCPeerConnection); + adaptor.remotePeerConnection["testPlayer"].iceConnectionState = "disconnected"; - clock.tick(1500); - assert(fakeSend.calledOnce); - clock.tick(2500); - assert(fakeSend.calledTwice); + adaptor.tryAgain(); + clock.tick(3000); - }); + expect(isReconnectionProcessStartedForPublisher).equal(true); + expect(isReconnectionProcessStartedForPlayer).equal(true); + }); - it("sanitize HTML", async function () { - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); - var scriptMsg = ""; //message with script - var sanitizeMsg = adaptor.sanitizeHTML(scriptMsg); - assert.notEqual(scriptMsg, sanitizeMsg) + it("Reconnection for play", async function() { + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); + var fakeSend = sinon.replace(adaptor.webSocketAdaptor, "send", sinon.fake()); - var text = "hi how are you"; //message without script - var message = adaptor.sanitizeHTML(text) - assert.strictEqual(text, message) - }) + const streamId = "test" + Math.floor(Math.random() * 100); + adaptor.playStreamId.push(streamId); + var mockPC = sinon.mock(RTCPeerConnection); + adaptor.remotePeerConnection[streamId] = mockPC + mockPC.iceConnectionState = "disconnected"; + mockPC.close = sinon.fake(); - it("Reconnection for publish", async function () { - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); - var fakeSendPublish = sinon.replace(adaptor, "sendPublishCommand", sinon.fake()); - var fakeStop = sinon.replace(adaptor, "stop", sinon.fake()); - - const streamId = "test" + Math.floor(Math.random() * 100); - adaptor.publishStreamId = streamId; - var mockPC = sinon.mock(RTCPeerConnection); - adaptor.remotePeerConnection[streamId] = mockPC - mockPC.iceConnectionState = "disconnected"; - mockPC.close = sinon.fake(); + clock.tick(3000); + adaptor.tryAgain(); + //Add extra delay because publish is called a few seconds later the stop in tryAgain method - adaptor.mediaManager.localStream = sinon.mock(); - var callback = sinon.stub(); - callback.returns([sinon.mock()]); - adaptor.mediaManager.localStream.getVideoTracks = callback; - adaptor.mediaManager.localStream.getAudioTracks = callback; - adaptor.mediaManager.localStream.getTracks = sinon.stub().returns([]); + clock.tick(1500); + assert(fakeSend.calledOnce); + clock.tick(2500); + assert(fakeSend.calledTwice); - clock.tick(3000); - adaptor.tryAgain(); - //Add extra delay because publish is called a few seconds later the stop in tryAgain method - clock.tick(1500); - assert(fakeSendPublish.calledOnce); - assert(fakeStop.calledOnce); + }); - clock.tick(2500); - assert(fakeSendPublish.calledTwice); + it("sanitize HTML", async function() { + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); + var scriptMsg = ""; //message with script + var sanitizeMsg = adaptor.sanitizeHTML(scriptMsg); + assert.notEqual(scriptMsg, sanitizeMsg) + var text = "hi how are you"; //message without script + var message = adaptor.sanitizeHTML(text) + assert.strictEqual(text, message) + }) - }); + it("Reconnection for publish", async function() { + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); + var fakeSendPublish = sinon.replace(adaptor, "sendPublishCommand", sinon.fake()); + var fakeStop = sinon.replace(adaptor, "stop", sinon.fake()); + + const streamId = "test" + Math.floor(Math.random() * 100); + adaptor.publishStreamId = streamId; + var mockPC = sinon.mock(RTCPeerConnection); + adaptor.remotePeerConnection[streamId] = mockPC + mockPC.iceConnectionState = "disconnected"; + mockPC.close = sinon.fake(); - it("EnableStats - DisableStats", async function () { - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); + adaptor.mediaManager.localStream = sinon.mock(); + var callback = sinon.stub(); + callback.returns([sinon.mock()]); + adaptor.mediaManager.localStream.getVideoTracks = callback; + adaptor.mediaManager.localStream.getAudioTracks = callback; + adaptor.mediaManager.localStream.getTracks = sinon.stub().returns([]); - const streamId = "test" + Math.floor(Math.random() * 100); - adaptor.publishStreamId = streamId; - var mockPC = sinon.mock(RTCPeerConnection); - adaptor.remotePeerConnection[streamId] = mockPC + clock.tick(3000); + adaptor.tryAgain(); - expect(adaptor.remotePeerConnectionStats[streamId]).to.be.undefined; + //Add extra delay because publish is called a few seconds later the stop in tryAgain method + clock.tick(1500); + assert(fakeSendPublish.calledOnce); + assert(fakeStop.calledOnce); - adaptor.enableStats(streamId); - expect(adaptor.remotePeerConnectionStats[streamId].timerId).to.be.not.undefined; + clock.tick(2500); + assert(fakeSendPublish.calledTwice); - adaptor.disableStats(streamId); - expect(adaptor.remotePeerConnectionStats[streamId]).to.be.undefined; + }); - adaptor.enableStats(streamId); - expect(adaptor.remotePeerConnectionStats[streamId].timerId).to.be.not.undefined; + it("EnableStats - DisableStats", async function() { + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); - adaptor.disableStats(streamId); - expect(adaptor.remotePeerConnectionStats[streamId]).to.be.undefined; + const streamId = "test" + Math.floor(Math.random() * 100); + adaptor.publishStreamId = streamId; + var mockPC = sinon.mock(RTCPeerConnection); + adaptor.remotePeerConnection[streamId] = mockPC + expect(adaptor.remotePeerConnectionStats[streamId]).to.be.undefined; - }); + adaptor.enableStats(streamId); + expect(adaptor.remotePeerConnectionStats[streamId].timerId).to.be.not.undefined; - it("Websocket send try catch", async function () { - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); + adaptor.disableStats(streamId); + expect(adaptor.remotePeerConnectionStats[streamId]).to.be.undefined; - adaptor.webSocketAdaptor.send("test"); - adaptor.webSocketAdaptor.close(); - adaptor.webSocketAdaptor.send("test"); - adaptor.webSocketAdaptor.connected = true; - var spySend = sinon.spy(adaptor.webSocketAdaptor.send); - try { - spySend(); - } catch (e) { - // pass - } - adaptor.webSocketAdaptor.send("test"); - assert(spySend.threw()); - }); + adaptor.enableStats(streamId); + expect(adaptor.remotePeerConnectionStats[streamId].timerId).to.be.not.undefined; - //there was a bug and this method is not initialized - it("enableAudioLevelForLocalStream", async function () { - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://localhost", - initializeComponents: false, - volumeMeterUrl: 'base/src/main/js/volume-meter-processor.js', - }); + adaptor.disableStats(streamId); + expect(adaptor.remotePeerConnectionStats[streamId]).to.be.undefined; - initialized = false; - await adaptor.initialize().then(() => { - initialized = true; - }) - expect(initialized).to.be.true; + }); - expect(adaptor.mediaManager.localStream).to.be.not.null; + it("Websocket send try catch", async function() { + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); - initialized = false; - await adaptor.enableAudioLevelForLocalStream((event) => { - console.log("audio level: " + event.data); - }).then(() => { - initialized = true; - }).catch((err) => { - console.error("audiolevel error " + err); - }); + adaptor.webSocketAdaptor.send("test"); + adaptor.webSocketAdaptor.close(); + adaptor.webSocketAdaptor.send("test"); + adaptor.webSocketAdaptor.connected = true; + var spySend = sinon.spy(adaptor.webSocketAdaptor.send); + try { + spySend(); + } catch (e) { + // pass + } + adaptor.webSocketAdaptor.send("test"); + assert(spySend.threw()); - expect(initialized).to.be.true; + }); - adaptor.disableAudioLevelForLocalStream(); - expect(adaptor.mediaManager.localStreamSoundMeter).to.be.null; + //there was a bug and this method is not initialized + it("enableAudioLevelForLocalStream", async function() { + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://localhost", + initializeComponents: false, + volumeMeterUrl: 'base/src/main/js/volume-meter-processor.js', + }); - }); + initialized = false; + await adaptor.initialize().then(() => { + initialized = true; + }) - it("sendData", async function () { - try { - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://localhost", - initializeComponents: false - }); + expect(initialized).to.be.true; - let streamId = "test"; - var webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + expect(adaptor.mediaManager.localStream).to.be.not.null; - adaptor.remotePeerConnection[streamId] = sinon.mock(RTCPeerConnection); + initialized = false; + await adaptor.enableAudioLevelForLocalStream((event) => { + console.log("audio level: " + event.data); + }).then(() => { + initialized = true; + }).catch((err) => { + console.error("audiolevel error " + err); + }); - adaptor.remotePeerConnection[streamId].dataChannel = sinon.fake.returns({ - readyState: "open", - send: sinon.fake() - }); - adaptor.sendData(streamId, "test"); + expect(initialized).to.be.true; - adaptor.remotePeerConnection[streamId].dataChannel = undefined - adaptor.sendData(streamId, "test"); + adaptor.disableAudioLevelForLocalStream(); - adaptor.remotePeerConnection[streamId].dataChannel = null - adaptor.sendData(streamId, "test"); - } catch (e) { - console.error(e); - assert(false); - } - }); + expect(adaptor.mediaManager.localStreamSoundMeter).to.be.null; - it("dummyStreamAndSwitch", async function () { + }); - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://localhost", - mediaConstraints: { - video: "dummy", - audio: "dummy" - }, - initializeComponents: false - }); + it("sendData", async function() { + try { + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://localhost", + initializeComponents: false + }); + let streamId = "test"; + var webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - expect(adaptor.mediaManager.blackVideoTrack).to.be.null - expect(adaptor.mediaManager.silentAudioTrack).to.be.null - expect(adaptor.mediaManager.oscillator).to.be.null - - await adaptor.initialize(); - - - expect(adaptor.mediaManager.mediaConstraints).to.deep.equal({video: "dummy", audio: "dummy"}); + adaptor.remotePeerConnection[streamId] = sinon.mock(RTCPeerConnection); - expect(adaptor.mediaManager.blackVideoTrack).to.not.be.null - expect(adaptor.mediaManager.silentAudioTrack).to.not.be.null - expect(adaptor.mediaManager.oscillator).to.not.be.null - expect(adaptor.mediaManager.localStream.getVideoTracks().length).to.be.equal(1) - expect(adaptor.mediaManager.localStream.getAudioTracks().length).to.be.equal(1) + adaptor.remotePeerConnection[streamId].dataChannel = sinon.fake.returns({ + readyState: "open", + send: sinon.fake() + }); + adaptor.sendData(streamId, "test"); + adaptor.remotePeerConnection[streamId].dataChannel = undefined + adaptor.sendData(streamId, "test"); - await adaptor.openStream({video: true, audio: true}); + adaptor.remotePeerConnection[streamId].dataChannel = null + adaptor.sendData(streamId, "test"); + } catch (e) { + console.error(e); + assert(false); + } + }); - expect(adaptor.mediaManager.blackVideoTrack).to.be.null - expect(adaptor.mediaManager.silentAudioTrack).to.be.null - expect(adaptor.mediaManager.oscillator).to.be.null + it("dummyStreamAndSwitch", async function() { - expect(adaptor.mediaManager.mediaConstraints).to.deep.equal({video: true, audio: true}); - expect(adaptor.mediaManager.localStream.getVideoTracks().length).to.be.equal(1) - expect(adaptor.mediaManager.localStream.getAudioTracks().length).to.be.equal(1) + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://localhost", + mediaConstraints: { + video: "dummy", + audio: "dummy" + }, + initializeComponents: false + }); - }); - it("updateAudioTrack", async function () { - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://localhost", - mediaConstraints: { - video: "dummy", - audio: "dummy" - }, - initializeComponents: false - }); + expect(adaptor.mediaManager.blackVideoTrack).to.be.null + expect(adaptor.mediaManager.silentAudioTrack).to.be.null + expect(adaptor.mediaManager.oscillator).to.be.null + + await adaptor.initialize(); + + + expect(adaptor.mediaManager.mediaConstraints).to.deep.equal({ video: "dummy", audio: "dummy" }); - await adaptor.initialize(); + expect(adaptor.mediaManager.blackVideoTrack).to.not.be.null + expect(adaptor.mediaManager.silentAudioTrack).to.not.be.null + expect(adaptor.mediaManager.oscillator).to.not.be.null + expect(adaptor.mediaManager.localStream.getVideoTracks().length).to.be.equal(1) + expect(adaptor.mediaManager.localStream.getAudioTracks().length).to.be.equal(1) - expect(adaptor.mediaManager.localStreamSoundMeter).to.be.null; - adaptor.enableAudioLevelForLocalStream((value) => { + await adaptor.openStream({ video: true, audio: true }); - }, 200); + expect(adaptor.mediaManager.blackVideoTrack).to.be.null + expect(adaptor.mediaManager.silentAudioTrack).to.be.null + expect(adaptor.mediaManager.oscillator).to.be.null - expect(adaptor.mediaManager.localStreamSoundMeter).to.not.be.null; + expect(adaptor.mediaManager.mediaConstraints).to.deep.equal({ video: true, audio: true }); + expect(adaptor.mediaManager.localStream.getVideoTracks().length).to.be.equal(1) + expect(adaptor.mediaManager.localStream.getAudioTracks().length).to.be.equal(1) - var audioTrack = adaptor.mediaManager.getSilentAudioTrack(); + }); - var stream = new MediaStream(); - stream.addTrack(audioTrack); + it("updateAudioTrack", async function() { + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://localhost", + mediaConstraints: { + video: "dummy", + audio: "dummy" + }, + initializeComponents: false + }); - await adaptor.updateAudioTrack(stream, null, null); - }); + await adaptor.initialize(); - it("testSoundMeter", function (done) { - this.timeout(5000); - console.log("Starting testSoundMeter"); + expect(adaptor.mediaManager.localStreamSoundMeter).to.be.null; - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://localhost", - mediaConstraints: { - video: true, - audio: true - }, - initializeComponents: false, - volumeMeterUrl: 'base/src/main/js/volume-meter-processor.js', - }); - - //fake stream in te browser is a period audio and silence, so getting sound level more than 0 requires - - adaptor.initialize().then(() => { - var audioContext = new (window.AudioContext || window.webkitAudioContext)(); - var oscillator = audioContext.createOscillator(); - oscillator.type = "sine"; - oscillator.frequency.value = 800; - var mediaStreamSource = audioContext.createMediaStreamDestination(); - oscillator.connect(mediaStreamSource); - var mediaStreamTrack = mediaStreamSource.stream.getAudioTracks()[0]; - oscillator.start(); - - adaptor.mediaManager.localStream = new MediaStream([mediaStreamTrack]) - adaptor.mediaManager.audioContext = audioContext; - adaptor.enableAudioLevelForLocalStream((level) => { - console.log("sound level -> " + level); - if (level > 0) { - done(); - } - }); - - expect(adaptor.mediaManager.localStreamSoundMeter).to.not.be.null; - }) - }) - - - it("takeConfiguration", async function () { - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://localhost", - mediaConstraints: { - video: true, - audio: true - }, - initializeComponents: false - }); - - await adaptor.initialize(); - expect(adaptor.remotePeerConnection["stream1"]).to.be.undefined; - - adaptor.takeConfiguration("stream1", "conf", "offer", "track1"); - - expect(adaptor.remotePeerConnection["stream1"]).to.not.be.undefined; - - }); - - it("takeCandidate", async function () { - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://localhost", - mediaConstraints: { - video: true, - audio: true - }, - initializeComponents: false - }); - - await adaptor.initialize(); - - expect(adaptor.remotePeerConnection["stream1"]).to.be.undefined; - expect(adaptor.iceCandidateList["stream1"]).to.be.undefined; - - - adaptor.takeCandidate("stream1", "label", "candidate"); - - expect(adaptor.remotePeerConnection["stream1"]).to.not.be.undefined; - - expect(adaptor.iceCandidateList["stream1"].length).to.be.equal(1); - - }); - it("mutedButSpeaking", async () => { - this.timeout(10000); - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://localhost", - mediaConstraints: { - video: true, - audio: true - }, - initializeComponents: false, - volumeMeterUrl: 'base/src/main/js/volume-meter-processor.js', - }); - - var audioContext = new (window.AudioContext || window.webkitAudioContext)(); - var oscillator = audioContext.createOscillator(); - oscillator.type = "sine"; - oscillator.frequency.value = 800; - var mediaStreamSource = audioContext.createMediaStreamDestination(); - oscillator.connect(mediaStreamSource); - var mediaStreamTrack = mediaStreamSource.stream.getAudioTracks()[0]; - oscillator.start(); - - - adaptor.mediaManager.mutedAudioStream = new MediaStream([mediaStreamTrack]) - adaptor.mediaManager.localStream = new MediaStream([mediaStreamTrack]) - adaptor.mediaManager.audioContext = audioContext; - - var getUserMediaFailed = new Promise(function (resolve, reject) { - navigator.mediaDevices.getUserMedia = async () => { - return Promise.reject(); - }; - adaptor.initialize().then(async () => { - try { - await adaptor.enableAudioLevelWhenMuted(); - } catch (e) { - console.log("get user media failed test") - resolve(); - } - }); - }); - var speakingButMuted = getUserMediaFailed.then(() => { - return new Promise(function (resolve, reject) { - navigator.mediaDevices.getUserMedia = async () => { - return Promise.resolve(new MediaStream([mediaStreamTrack])); - }; - - adaptor.initialize().then(async () => { - adaptor.mediaManager.callback = (info) => { - console.log("callback ", info); - if (info === "speaking_but_muted") { - console.log("speaking_but_muted1"); - resolve(); - } - }; - await adaptor.enableAudioLevelWhenMuted(); - }); - }); - }); - - var soundMeteraddModuleFailed = speakingButMuted.then(() => { - adaptor.mediaManager.mutedSoundMeter.context.audioWorklet.addModule = async () => { - return Promise.reject("error"); - }; - return new Promise(async function (resolve, reject) { - adaptor.enableAudioLevelWhenMuted().catch((e) => { - resolve() - }) - }); - }); - - console.assert(soundMeteraddModuleFailed, "soundMeteraddModuleFailed"); + adaptor.enableAudioLevelForLocalStream((value) => { - }); - - - it("startPublishing", async function () { - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); - - let peerConnection = new RTCPeerConnection(); - let initPeerConnection = sinon.replace(adaptor, "initPeerConnection", sinon.fake.returns(peerConnection)); - - let createOfferFake = sinon.replace(peerConnection, "createOffer", sinon.fake.returns(Promise.reject("this is on purpose"))); - - adaptor.startPublishing("stream123"); + }, 200); - expect(initPeerConnection.calledWithExactly("stream123", "publish")).to.be.true; - }); + expect(adaptor.mediaManager.localStreamSoundMeter).to.not.be.null; - it("join", async function () { + var audioTrack = adaptor.mediaManager.getSilentAudioTrack(); - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); + var stream = new MediaStream(); + stream.addTrack(audioTrack); - let streamId = "stream123"; - let jsCmd = { - command: "join", - streamId: streamId, - multiPeer: false, - mode: "play" - }; - - let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + await adaptor.updateAudioTrack(stream, null, null); + }); - let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); + it("testSoundMeter", function(done) { + this.timeout(5000); + console.log("Starting testSoundMeter"); - adaptor.join(streamId); + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://localhost", + mediaConstraints: { + video: true, + audio: true + }, + initializeComponents: false, + volumeMeterUrl: 'base/src/main/js/volume-meter-processor.js', + }); + + //fake stream in te browser is a period audio and silence, so getting sound level more than 0 requires + + adaptor.initialize().then(() => { + var audioContext = new (window.AudioContext || window.webkitAudioContext)(); + var oscillator = audioContext.createOscillator(); + oscillator.type = "sine"; + oscillator.frequency.value = 800; + var mediaStreamSource = audioContext.createMediaStreamDestination(); + oscillator.connect(mediaStreamSource); + var mediaStreamTrack = mediaStreamSource.stream.getAudioTracks()[0]; + oscillator.start(); + + adaptor.mediaManager.localStream = new MediaStream([mediaStreamTrack]) + adaptor.mediaManager.audioContext = audioContext; + adaptor.enableAudioLevelForLocalStream((level) => { + console.log("sound level -> " + level); + if (level > 0) { + done(); + } + }); + + expect(adaptor.mediaManager.localStreamSoundMeter).to.not.be.null; + }) + }) + + + it("takeConfiguration", async function() { + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://localhost", + mediaConstraints: { + video: true, + audio: true + }, + initializeComponents: false + }); + + await adaptor.initialize(); + expect(adaptor.remotePeerConnection["stream1"]).to.be.undefined; + + adaptor.takeConfiguration("stream1", "conf", "offer", "track1"); + + expect(adaptor.remotePeerConnection["stream1"]).to.not.be.undefined; + + }); + + it("takeCandidate", async function() { + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://localhost", + mediaConstraints: { + video: true, + audio: true + }, + initializeComponents: false + }); + + await adaptor.initialize(); + + expect(adaptor.remotePeerConnection["stream1"]).to.be.undefined; + expect(adaptor.iceCandidateList["stream1"]).to.be.undefined; + + + adaptor.takeCandidate("stream1", "label", "candidate"); + + expect(adaptor.remotePeerConnection["stream1"]).to.not.be.undefined; + + expect(adaptor.iceCandidateList["stream1"].length).to.be.equal(1); + + }); + it("mutedButSpeaking", async () => { + this.timeout(10000); + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://localhost", + mediaConstraints: { + video: true, + audio: true + }, + initializeComponents: false, + volumeMeterUrl: 'base/src/main/js/volume-meter-processor.js', + }); + + var audioContext = new (window.AudioContext || window.webkitAudioContext)(); + var oscillator = audioContext.createOscillator(); + oscillator.type = "sine"; + oscillator.frequency.value = 800; + var mediaStreamSource = audioContext.createMediaStreamDestination(); + oscillator.connect(mediaStreamSource); + var mediaStreamTrack = mediaStreamSource.stream.getAudioTracks()[0]; + oscillator.start(); + + + adaptor.mediaManager.mutedAudioStream = new MediaStream([mediaStreamTrack]) + adaptor.mediaManager.localStream = new MediaStream([mediaStreamTrack]) + adaptor.mediaManager.audioContext = audioContext; + + var getUserMediaFailed = new Promise(function(resolve, reject) { + navigator.mediaDevices.getUserMedia = async () => { + return Promise.reject(); + }; + adaptor.initialize().then(async () => { + try { + await adaptor.enableAudioLevelWhenMuted(); + } catch (e) { + console.log("get user media failed test") + resolve(); + } + }); + }); + var speakingButMuted = getUserMediaFailed.then(() => { + return new Promise(function(resolve, reject) { + navigator.mediaDevices.getUserMedia = async () => { + return Promise.resolve(new MediaStream([mediaStreamTrack])); + }; + + adaptor.initialize().then(async () => { + adaptor.mediaManager.callback = (info) => { + console.log("callback ", info); + if (info === "speaking_but_muted") { + console.log("speaking_but_muted1"); + resolve(); + } + }; + await adaptor.enableAudioLevelWhenMuted(); + }); + }); + }); + + var soundMeteraddModuleFailed = speakingButMuted.then(() => { + adaptor.mediaManager.mutedSoundMeter.context.audioWorklet.addModule = async () => { + return Promise.reject("error"); + }; + return new Promise(async function(resolve, reject) { + adaptor.enableAudioLevelWhenMuted().catch((e) => { + resolve() + }) + }); + }); + + console.assert(soundMeteraddModuleFailed, "soundMeteraddModuleFailed"); - sendExpectation.verify() - }) + }); + + + it("startPublishing", async function() { + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); + + let peerConnection = new RTCPeerConnection(); + let initPeerConnection = sinon.replace(adaptor, "initPeerConnection", sinon.fake.returns(peerConnection)); + + let createOfferFake = sinon.replace(peerConnection, "createOffer", sinon.fake.returns(Promise.reject("this is on purpose"))); + + adaptor.startPublishing("stream123"); + expect(initPeerConnection.calledWithExactly("stream123", "publish")).to.be.true; + }); - it("getSubtracks", async function () { + it("join", async function() { - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); - let streamId = "test" + Math.floor(Math.random() * 100); - let offset = Math.floor(Math.random() * 100); - let size = Math.floor(Math.random() * 100); - let role = "role1"; - let jsCmd = { - command: "getSubtracks", - streamId: streamId, - role: role, - offset: offset, - size: size, - }; + let streamId = "stream123"; + let jsCmd = { + command: "join", + streamId: streamId, + multiPeer: false, + mode: "play" + }; + + let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); - let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); + adaptor.join(streamId); - adaptor.getSubtracks(streamId, role, offset, size); + sendExpectation.verify() + }) - sendExpectation.verify() - }) - it("joinRoom", async function () { + it("getSubtracks", async function() { - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); - let streamId = "stream123"; - let roomId = "roomId"; + let streamId = "test" + Math.floor(Math.random() * 100); + let offset = Math.floor(Math.random() * 100); + let size = Math.floor(Math.random() * 100); + let role = "role1"; + let jsCmd = { + command: "getSubtracks", + streamId: streamId, + role: role, + offset: offset, + size: size, + }; - let jsCmd = { - command: "joinRoom", - room: roomId, - streamId: streamId, - mode: "multitrack", - } + let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); - let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); + adaptor.getSubtracks(streamId, role, offset, size); - adaptor.joinRoom(roomId, streamId, "multitrack"); + sendExpectation.verify() + }) - sendExpectation.verify() + it("joinRoom", async function() { - }); + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); - it("eventListeners", async function () { - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); + let streamId = "stream123"; + let roomId = "roomId"; + let jsCmd = { + command: "joinRoom", + room: roomId, + streamId: streamId, + mode: "multitrack", + } - var eventListenerCalled = false; - adaptor.addEventListener((info, obj) => { - eventListenerCalled = true; - }); + let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - var errorListenerCalled = false; - adaptor.addErrorEventListener((error, message) => { - errorListenerCalled = true; - }); + let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); + adaptor.joinRoom(roomId, streamId, "multitrack"); - adaptor.mediaManager.callback("info", "obj"); + sendExpectation.verify() - adaptor.mediaManager.callbackError("info", "obj"); + }); - expect(eventListenerCalled).to.be.true; - expect(errorListenerCalled).to.be.true; + it("eventListeners", async function() { + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); - }); - it("onTrack", async function () { + var eventListenerCalled = false; + adaptor.addEventListener((info, obj) => { + eventListenerCalled = true; + }); - { - var videoElement = document.createElement("video"); - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true, - remoteVideoElement: videoElement - }); + var errorListenerCalled = false; + adaptor.addErrorEventListener((error, message) => { + errorListenerCalled = true; + }); - var mediaStream = new MediaStream(); - var event = { - streams: [mediaStream] - } - expect(videoElement.srcObject).to.be.null; + adaptor.mediaManager.callback("info", "obj"); - adaptor.onTrack(event, "stream1"); + adaptor.mediaManager.callbackError("info", "obj"); - expect(videoElement.srcObject).to.not.be.null; - } + expect(eventListenerCalled).to.be.true; + expect(errorListenerCalled).to.be.true; + }); - { - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true, - }); + it("onTrack", async function() { - var eventListenerCalled = false; - adaptor.addEventListener((info, obj) => { - if (info == "newTrackAvailable") { - eventListenerCalled = true; - } - }) + { + var videoElement = document.createElement("video"); + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true, + remoteVideoElement: videoElement + }); - var mediaStream = new MediaStream(); - var event = { - streams: [mediaStream], - transceiver: { - id: "anyid" - } - } + var mediaStream = new MediaStream(); + var event = { + streams: [mediaStream] + } - adaptor.idMapping["stream1"] = "anything"; + expect(videoElement.srcObject).to.be.null; - adaptor.onTrack(event, "stream1"); + adaptor.onTrack(event, "stream1"); - expect(eventListenerCalled).to.be.true; + expect(videoElement.srcObject).to.not.be.null; + } - } - }); + { + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true, + }); - it("getStreamInfo", async function () { - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); + var eventListenerCalled = false; + adaptor.addEventListener((info, obj) => { + if (info == "newTrackAvailable") { + eventListenerCalled = true; + } + }) - let streamId = "stream123"; - let jsCmd = { - command: "getStreamInfo", - streamId: streamId, - }; + var mediaStream = new MediaStream(); + var event = { + streams: [mediaStream], + transceiver: { + id: "anyid" + } + } - let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + adaptor.idMapping["stream1"] = "anything"; - let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); + adaptor.onTrack(event, "stream1"); - adaptor.getStreamInfo(streamId); + expect(eventListenerCalled).to.be.true; - sendExpectation.verify() - }); + } + }); - it("getBroadcastObject", async function () { - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); + it("getStreamInfo", async function() { + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); - let streamId = "stream123"; - let jsCmd = { - command: "getBroadcastObject", - streamId: streamId, - }; + let streamId = "stream123"; + let jsCmd = { + command: "getStreamInfo", + streamId: streamId, + }; - let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); + let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); - adaptor.getBroadcastObject(streamId); + adaptor.getStreamInfo(streamId); - sendExpectation.verify() - }); + sendExpectation.verify() + }); - it("requestVideoTrackAssignments", async function () { - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); + it("getBroadcastObject", async function() { + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); - let streamId = "stream123"; + let streamId = "stream123"; + let jsCmd = { + command: "getBroadcastObject", + streamId: streamId, + }; - let jsCmd = { - command: "getVideoTrackAssignmentsCommand", - streamId: streamId, - }; + let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); - let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); + adaptor.getBroadcastObject(streamId); - adaptor.requestVideoTrackAssignments(streamId); + sendExpectation.verify() + }); - sendExpectation.verify() + it("requestVideoTrackAssignments", async function() { - }) + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); + let streamId = "stream123"; - it("registerPushNotificationToken", async function () { - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); + let jsCmd = { + command: "getVideoTrackAssignmentsCommand", + streamId: streamId, + }; - let subscriberId = "subscriberId"; - let authToken = "autotokenkdnkf"; - let pnsRegistrationToken = "pnsRegistrationTokenpnsRegistrationTokenpnsRegistrationTokenpnsRegistrationToken"; - let pnstype = "fcm"; + let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - let jsCmd = { - command: "registerPushNotificationToken", - subscriberId: subscriberId, - token: authToken, - pnsRegistrationToken: pnsRegistrationToken, - pnsType: pnstype - }; + let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); - let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + adaptor.requestVideoTrackAssignments(streamId); - let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); + sendExpectation.verify() - adaptor.registerPushNotificationToken(subscriberId, authToken, pnsRegistrationToken, pnstype); + }) - sendExpectation.verify() - }); + it("registerPushNotificationToken", async function() { + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); + let subscriberId = "subscriberId"; + let authToken = "autotokenkdnkf"; + let pnsRegistrationToken = "pnsRegistrationTokenpnsRegistrationTokenpnsRegistrationTokenpnsRegistrationToken"; + let pnstype = "fcm"; - it("sendPushNotification", async function () { - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); + let jsCmd = { + command: "registerPushNotificationToken", + subscriberId: subscriberId, + token: authToken, + pnsRegistrationToken: pnsRegistrationToken, + pnsType: pnstype + }; - let subscriberId = "subscriberId"; - let authToken = "autotokenkdnkf"; - let pushNotificationContent = "pnsRegistrationTokenpnsRegistrationTokenpnsRegistrationTokenpnsRegistrationToken"; - let subscriberIdsToNotify = "string1"; + let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - try { - adaptor.sendPushNotification(subscriberId, authToken, pushNotificationContent, subscriberIdsToNotify); - assert.fail("It should throw exception because pushNotificationContent is not json"); - } catch (e) { - //pass - } + let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); + adaptor.registerPushNotificationToken(subscriberId, authToken, pnsRegistrationToken, pnstype); - pushNotificationContent = {title: "title", body: "body"}; - let jsCmd = { - command: "sendPushNotification", - subscriberId: subscriberId, - token: authToken, - pushNotificationContent: pushNotificationContent, - subscriberIdsToNotify: subscriberIdsToNotify - }; + sendExpectation.verify() - try { - adaptor.sendPushNotification(subscriberId, authToken, pushNotificationContent, subscriberIdsToNotify); - assert.fail("It should throw exception because subscriberIdsToNotify is not array"); - } catch (e) { - //pass - } + }); - jsCmd = { - command: "sendPushNotification", - subscriberId: subscriberId, - token: authToken, - pushNotificationContent: pushNotificationContent, - subscriberIdsToNotify: subscriberIdsToNotify - }; - subscriberIdsToNotify = ["string1"]; + it("sendPushNotification", async function() { + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); - jsCmd = { - command: "sendPushNotification", - subscriberId: subscriberId, - token: authToken, - pushNotificationContent: pushNotificationContent, - subscriberIdsToNotify: subscriberIdsToNotify - }; - let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + let subscriberId = "subscriberId"; + let authToken = "autotokenkdnkf"; + let pushNotificationContent = "pnsRegistrationTokenpnsRegistrationTokenpnsRegistrationTokenpnsRegistrationToken"; + let subscriberIdsToNotify = "string1"; - let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); + try { + adaptor.sendPushNotification(subscriberId, authToken, pushNotificationContent, subscriberIdsToNotify); + assert.fail("It should throw exception because pushNotificationContent is not json"); + } catch (e) { + //pass + } - adaptor.sendPushNotification(subscriberId, authToken, pushNotificationContent, subscriberIdsToNotify); - sendExpectation.verify() + pushNotificationContent = { title: "title", body: "body" }; + let jsCmd = { + command: "sendPushNotification", + subscriberId: subscriberId, + token: authToken, + pushNotificationContent: pushNotificationContent, + subscriberIdsToNotify: subscriberIdsToNotify + }; - }); + try { + adaptor.sendPushNotification(subscriberId, authToken, pushNotificationContent, subscriberIdsToNotify); + assert.fail("It should throw exception because subscriberIdsToNotify is not array"); + } catch (e) { + //pass + } + jsCmd = { + command: "sendPushNotification", + subscriberId: subscriberId, + token: authToken, + pushNotificationContent: pushNotificationContent, + subscriberIdsToNotify: subscriberIdsToNotify + }; - it("sendPushNotificationToTopic", async function () { + subscriberIdsToNotify = ["string1"]; - let adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true - }); + jsCmd = { + command: "sendPushNotification", + subscriberId: subscriberId, + token: authToken, + pushNotificationContent: pushNotificationContent, + subscriberIdsToNotify: subscriberIdsToNotify + }; + let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - let subscriberId = "subscriberId"; - let authToken = "autotokenkdnkf"; - let pushNotificationContent = "text"; - let topic = "topic"; + let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); - let jsCmd = { - command: "sendPushNotification", - subscriberId: subscriberId, - token: authToken, - pushNotificationContent: pushNotificationContent, - topic: topic - }; + adaptor.sendPushNotification(subscriberId, authToken, pushNotificationContent, subscriberIdsToNotify); - try { - adaptor.sendPushNotificationToTopic(subscriberId, authToken, pushNotificationContent, topic); - assert.fail("It should throw exception because pushNotificationContent is not json"); - } catch (error) { - //pass - } + sendExpectation.verify() - pushNotificationContent = {title: "title", body: "body"}; - jsCmd = { - command: "sendPushNotification", - subscriberId: subscriberId, - token: authToken, - pushNotificationContent: pushNotificationContent, - topic: topic - }; - - let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); - - let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); - - adaptor.sendPushNotificationToTopic(subscriberId, authToken, pushNotificationContent, topic); - - sendExpectation.verify() - - }); - - describe("checkAndStopLocalVideoTrackOnAndroid", function () { - - let mediaManager; - let mockLocalStream; - - beforeEach(function () { - window.isAndroid = () => { - }; - - mockLocalStream = { - getVideoTracks: sinon.stub() - }; - - mediaManager = new MediaManager({ - websocketURL: "ws://example.com", - initializeComponents: false, - localStream: mockLocalStream - }); - }); - - it("should not stop video track if local stream exists and is not Android", function () { - const mockVideoTrack = {stop: sinon.fake()}; - mockLocalStream.getVideoTracks.returns([mockVideoTrack]); - sinon.stub(window, 'isAndroid').returns(false); - - mediaManager.checkAndStopLocalVideoTrackOnAndroid(); - - sinon.assert.notCalled(mockVideoTrack.stop); - }); - - it("should not stop video track if local stream does not exist", function () { - mediaManager.localStream = null; - - mediaManager.checkAndStopLocalVideoTrackOnAndroid(); - - sinon.assert.notCalled(mockLocalStream.getVideoTracks); - }); - - }); - - describe("turnOffLocalCamera", () => { - let adaptor; - let mockMediaManager; - - beforeEach(function () { - mockMediaManager = { - turnOffLocalCamera: sinon.fake() - }; - - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - isPlayMode: true, - mediaManager: mockMediaManager, - initializeComponents: false - }); - }); - - it("should call turnOffLocalCamera on mediaManager with correct streamId", function () { - const streamId = "testStreamId"; - let result = adaptor.turnOffLocalCamera(streamId); - assert.notEqual(result, undefined); - }); - - it("should handle undefined streamId", function () { - let result = adaptor.turnOffLocalCamera(undefined); - assert.notEqual(result, undefined); - }); - - it("should handle null streamId", function () { - let result = adaptor.turnOffLocalCamera(null); - assert.notEqual(result, undefined); - }); - - it("should handle empty string streamId", function () { - let result = adaptor.turnOffLocalCamera(""); - assert.notEqual(result, undefined); - }); - }); - - describe("getStats", function () { - let adaptor; - let mockPeerConnection; - let mockStats; - - beforeEach(function () { - mockPeerConnection = { - getStats: sinon.stub() - }; - mockStats = new Map(); - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - initializeComponents: false, - checkAndInitializePeerStats: sinon.fake() - }); - adaptor.remotePeerConnection = {"stream1": mockPeerConnection}; - adaptor.remotePeerConnectionStats = {"stream1": {}}; - adaptor.addEventListener((info, obj) => { - if (info === "updated_stats") { - console.log(JSON.stringify(obj)); - } - }); - }); - - it("should resolve with true when getStats is successful", async function () { - mockPeerConnection.getStats.resolves(mockStats); - const result = await adaptor.getStats("stream1"); - expect(result).to.be.true; - }); - - it("should correctly process inbound RTP with audio kind", async function () { - const consoleSpy = sinon.stub(console, 'log'); - - let localMockStats = { - type: "inbound-rtp", - kind: "audio", - trackIdentifier: "audioTrack1", - bytesReceived: 1000, - packetsLost: 10, - jitterBufferDelay: 5, - lastPacketReceivedTimestamp: 160000, - fractionLost: 0.1, - timestamp: Date.now() - }; - mockPeerConnection.getStats.resolves([localMockStats]); - const result = await adaptor.getStats("stream1"); - - let localMockStatsProcessed = { - "totalBytesReceived": 999, - "videoPacketsLost": -1, - "audioPacketsLost": 10, - "fractionLost": -0.9, - "currentTime": 0, - "totalBytesSent": -1, - "totalVideoPacketsSent": -1, - "totalAudioPacketsSent": -1, - "audioLevel": -1, - "qualityLimitationReason": "", - "totalFramesEncoded": -1, - "resWidth": -1, - "resHeight": -1, - "srcFps": -1, - "frameWidth": -1, - "frameHeight": -1, - "videoRoundTripTime": -1, - "videoJitter": -1, - "audioRoundTripTime": -1, - "audioJitter": -1, - "framesDecoded": -1, - "framesDropped": -1, - "framesReceived": -1, - "videoJitterAverageDelay": -1, - "audioJitterAverageDelay": -1, - "availableOutgoingBitrate": null, - "inboundRtpList": [ - { - "trackIdentifier": "audioTrack1", - "audioPacketsLost": 10, - "bytesReceived": 1000, - "jitterBufferDelay": 5, - "lastPacketReceivedTimestamp": 160000, - "fractionLost": 0.1, - "currentTime": 0 - } - ] - }; - - assert(consoleSpy.calledWith(JSON.stringify(localMockStatsProcessed)), 'console.log was not called with the expected arguments'); - - expect(result).to.be.true; - consoleSpy.restore(); - }); - - it("should correctly process inbound RTP with video kind", async function () { - const consoleSpy = sinon.stub(console, 'log'); - - let localMockStats = { - type: "inbound-rtp", - kind: "video", - trackIdentifier: "videoTrack2", - bytesReceived: 2000, - packetsLost: 5, - framesDropped: 2, - framesDecoded: 50, - framesPerSecond: 25, - jitterBufferDelay: 10, - lastPacketReceivedTimestamp: 160000, - fractionLost: 0.05, - timestamp: Date.now(), - frameWidth: 1920, - frameHeight: 1080 - }; - mockPeerConnection.getStats.resolves([localMockStats]); - const result = await adaptor.getStats("stream1"); - - let localMockStatsProcessed = { - "totalBytesReceived": 1999, - "videoPacketsLost": 5, - "audioPacketsLost": -1, - "fractionLost": -0.95, - "currentTime": 0, - "totalBytesSent": -1, - "totalVideoPacketsSent": -1, - "totalAudioPacketsSent": -1, - "audioLevel": -1, - "qualityLimitationReason": "", - "totalFramesEncoded": -1, - "resWidth": -1, - "resHeight": -1, - "srcFps": -1, - "frameWidth": 1920, - "frameHeight": 1080, - "videoRoundTripTime": -1, - "videoJitter": -1, - "audioRoundTripTime": -1, - "audioJitter": -1, - "framesDecoded": 50, - "framesDropped": 2, - "framesReceived": -1, - "videoJitterAverageDelay": -1, - "audioJitterAverageDelay": -1, - "availableOutgoingBitrate": null, - "inboundRtpList": [ - { - "trackIdentifier": "videoTrack2", - "videoPacketsLost": 5, - "framesDropped": 2, - "framesDecoded": 50, - "framesPerSecond": 25, - "bytesReceived": 2000, - "jitterBufferDelay": 10, - "lastPacketReceivedTimestamp": 160000, - "fractionLost": 0.05, - "currentTime": 0, - "frameWidth": 1920, - "frameHeight": 1080 - } - ] - }; - - assert(consoleSpy.calledWith(JSON.stringify(localMockStatsProcessed)), 'console.log was not called with the expected arguments'); - - expect(result).to.be.true; - consoleSpy.restore(); - }); - - it("should resolve with false when getStats fails", async function () { - mockPeerConnection.getStats.rejects(new Error("getStats error")); - const result = await adaptor.getStats("stream1"); - expect(result).to.be.false; - }); - - it("should not reinitialize remotePeerConnectionStats for an existing streamId", function () { - const streamId = "existingStream"; - adaptor.remotePeerConnectionStats[streamId] = new PeerStats(streamId); - const initialStats = adaptor.remotePeerConnectionStats[streamId]; - adaptor.checkAndInitializePeerStats(streamId); - expect(adaptor.remotePeerConnectionStats[streamId]).to.equal(initialStats); - }); - - it("should handle null streamId gracefully", function () { - const streamId = null; - expect(() => adaptor.checkAndInitializePeerStats(streamId)).not.to.throw(); - expect(adaptor.remotePeerConnectionStats[streamId]).to.be.undefined; - }); - - it("should handle undefined streamId gracefully", function () { - const streamId = undefined; - expect(() => adaptor.checkAndInitializePeerStats(streamId)).not.to.throw(); - expect(adaptor.remotePeerConnectionStats[streamId]).to.be.undefined; - }); - - }); - - describe("changeBandwidth", function () { - - let adaptor; - - beforeEach(function () { - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - initializeComponents: false, - }); - adaptor.mediaManager = { - changeBandwidth: sinon.fake() - }; - }); - - it("should call mediaManager's changeBandwidth with correct parameters", function () { - const bandwidth = 500; - const streamId = "stream1"; - - adaptor.changeBandwidth(bandwidth, streamId); - - expect(adaptor.mediaManager.changeBandwidth.calledWithMatch(bandwidth, streamId)).to.be.true; - }); - - it("should handle zero bandwidth", function () { - const bandwidth = 0; - const streamId = "stream1"; - - adaptor.changeBandwidth(bandwidth, streamId); - - expect(adaptor.mediaManager.changeBandwidth.calledWithMatch(bandwidth, streamId)).to.be.true; - }); - - it("should handle null streamId", function () { - const bandwidth = 500; - const streamId = null; - - adaptor.changeBandwidth(bandwidth, streamId); - - expect(adaptor.mediaManager.changeBandwidth.calledWithMatch(bandwidth, streamId)).to.be.true; - }); - }); - - describe("enableAudioLevelWhenMuted", function () { - - let adaptor; - - beforeEach(function () { - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - initializeComponents: false, - }); - adaptor.mediaManager = { - enableAudioLevelWhenMuted: sinon.fake() - }; - }); + }); - it("should call mediaManager's enableAudioLevelWhenMuted", function () { - adaptor.enableAudioLevelWhenMuted(); - expect(adaptor.mediaManager.enableAudioLevelWhenMuted.called).to.be.true; - }); + it("sendPushNotificationToTopic", async function() { - }); + let adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true + }); - describe("disableAudioLevelWhenMuted", function () { + let subscriberId = "subscriberId"; + let authToken = "autotokenkdnkf"; + let pushNotificationContent = "text"; + let topic = "topic"; - let adaptor; + let jsCmd = { + command: "sendPushNotification", + subscriberId: subscriberId, + token: authToken, + pushNotificationContent: pushNotificationContent, + topic: topic + }; - beforeEach(function () { - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - initializeComponents: false, - }); - adaptor.mediaManager = { - disableAudioLevelWhenMuted: sinon.fake() - }; - }); + try { + adaptor.sendPushNotificationToTopic(subscriberId, authToken, pushNotificationContent, topic); + assert.fail("It should throw exception because pushNotificationContent is not json"); + } catch (error) { + //pass + } - it("should call mediaManager's disableAudioLevelWhenMuted", function () { - adaptor.disableAudioLevelWhenMuted(); + pushNotificationContent = { title: "title", body: "body" }; + jsCmd = { + command: "sendPushNotification", + subscriberId: subscriberId, + token: authToken, + pushNotificationContent: pushNotificationContent, + topic: topic + }; + + let webSocketAdaptor = sinon.mock(adaptor.webSocketAdaptor); + + let sendExpectation = webSocketAdaptor.expects("send").once().withArgs(JSON.stringify(jsCmd)); + + adaptor.sendPushNotificationToTopic(subscriberId, authToken, pushNotificationContent, topic); + + sendExpectation.verify() + + }); + + describe("checkAndStopLocalVideoTrackOnAndroid", function() { + + let mediaManager; + let mockLocalStream; + + beforeEach(function() { + window.isAndroid = () => { + }; + + mockLocalStream = { + getVideoTracks: sinon.stub() + }; + + mediaManager = new MediaManager({ + websocketURL: "ws://example.com", + initializeComponents: false, + localStream: mockLocalStream + }); + }); + + it("should not stop video track if local stream exists and is not Android", function() { + const mockVideoTrack = { stop: sinon.fake() }; + mockLocalStream.getVideoTracks.returns([mockVideoTrack]); + sinon.stub(window, 'isAndroid').returns(false); + + mediaManager.checkAndStopLocalVideoTrackOnAndroid(); + + sinon.assert.notCalled(mockVideoTrack.stop); + }); + + it("should not stop video track if local stream does not exist", function() { + mediaManager.localStream = null; + + mediaManager.checkAndStopLocalVideoTrackOnAndroid(); + + sinon.assert.notCalled(mockLocalStream.getVideoTracks); + }); + + }); + + describe("turnOffLocalCamera", () => { + let adaptor; + let mockMediaManager; + + beforeEach(function() { + mockMediaManager = { + turnOffLocalCamera: sinon.fake() + }; + + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + isPlayMode: true, + mediaManager: mockMediaManager, + initializeComponents: false + }); + }); + + it("should call turnOffLocalCamera on mediaManager with correct streamId", function() { + const streamId = "testStreamId"; + let result = adaptor.turnOffLocalCamera(streamId); + assert.notEqual(result, undefined); + }); + + it("should handle undefined streamId", function() { + let result = adaptor.turnOffLocalCamera(undefined); + assert.notEqual(result, undefined); + }); + + it("should handle null streamId", function() { + let result = adaptor.turnOffLocalCamera(null); + assert.notEqual(result, undefined); + }); + + it("should handle empty string streamId", function() { + let result = adaptor.turnOffLocalCamera(""); + assert.notEqual(result, undefined); + }); + }); + + describe("getStats", function() { + let adaptor; + let mockPeerConnection; + let mockStats; + + beforeEach(function() { + mockPeerConnection = { + getStats: sinon.stub() + }; + mockStats = new Map(); + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + initializeComponents: false, + checkAndInitializePeerStats: sinon.fake() + }); + adaptor.remotePeerConnection = { "stream1": mockPeerConnection }; + adaptor.remotePeerConnectionStats = { "stream1": {} }; + adaptor.addEventListener((info, obj) => { + if (info === "updated_stats") { + console.log(JSON.stringify(obj)); + } + }); + }); + + it("should resolve with true when getStats is successful", async function() { + mockPeerConnection.getStats.resolves(mockStats); + const result = await adaptor.getStats("stream1"); + expect(result).to.be.true; + }); + + it("should correctly process inbound RTP with audio kind", async function() { + const consoleSpy = sinon.stub(console, 'log'); + + let localMockStats = { + type: "inbound-rtp", + kind: "audio", + trackIdentifier: "audioTrack1", + bytesReceived: 1000, + packetsLost: 10, + jitterBufferDelay: 5, + lastPacketReceivedTimestamp: 160000, + fractionLost: 0.1, + timestamp: Date.now() + }; + mockPeerConnection.getStats.resolves([localMockStats]); + const result = await adaptor.getStats("stream1"); + + let localMockStatsProcessed = { + "totalBytesReceived": 999, + "videoPacketsLost": -1, + "audioPacketsLost": 10, + "fractionLost": -0.9, + "currentTime": 0, + "totalBytesSent": -1, + "totalVideoPacketsSent": -1, + "totalAudioPacketsSent": -1, + "audioLevel": -1, + "qualityLimitationReason": "", + "totalFramesEncoded": -1, + "resWidth": -1, + "resHeight": -1, + "srcFps": -1, + "frameWidth": -1, + "frameHeight": -1, + "videoRoundTripTime": -1, + "videoJitter": -1, + "audioRoundTripTime": -1, + "audioJitter": -1, + "framesDecoded": -1, + "framesDropped": -1, + "framesReceived": -1, + "videoJitterAverageDelay": -1, + "audioJitterAverageDelay": -1, + "availableOutgoingBitrate": null, + "inboundRtpList": [ + { + "trackIdentifier": "audioTrack1", + "audioPacketsLost": 10, + "bytesReceived": 1000, + "jitterBufferDelay": 5, + "lastPacketReceivedTimestamp": 160000, + "fractionLost": 0.1, + "currentTime": 0 + } + ] + }; + + assert(consoleSpy.calledWith(JSON.stringify(localMockStatsProcessed)), 'console.log was not called with the expected arguments'); + + expect(result).to.be.true; + consoleSpy.restore(); + }); + + it("should correctly process inbound RTP with video kind", async function() { + const consoleSpy = sinon.stub(console, 'log'); + + let localMockStats = { + type: "inbound-rtp", + kind: "video", + trackIdentifier: "videoTrack2", + bytesReceived: 2000, + packetsLost: 5, + framesDropped: 2, + framesDecoded: 50, + framesPerSecond: 25, + jitterBufferDelay: 10, + lastPacketReceivedTimestamp: 160000, + fractionLost: 0.05, + timestamp: Date.now(), + frameWidth: 1920, + frameHeight: 1080 + }; + mockPeerConnection.getStats.resolves([localMockStats]); + const result = await adaptor.getStats("stream1"); + + let localMockStatsProcessed = { + "totalBytesReceived": 1999, + "videoPacketsLost": 5, + "audioPacketsLost": -1, + "fractionLost": -0.95, + "currentTime": 0, + "totalBytesSent": -1, + "totalVideoPacketsSent": -1, + "totalAudioPacketsSent": -1, + "audioLevel": -1, + "qualityLimitationReason": "", + "totalFramesEncoded": -1, + "resWidth": -1, + "resHeight": -1, + "srcFps": -1, + "frameWidth": 1920, + "frameHeight": 1080, + "videoRoundTripTime": -1, + "videoJitter": -1, + "audioRoundTripTime": -1, + "audioJitter": -1, + "framesDecoded": 50, + "framesDropped": 2, + "framesReceived": -1, + "videoJitterAverageDelay": -1, + "audioJitterAverageDelay": -1, + "availableOutgoingBitrate": null, + "inboundRtpList": [ + { + "trackIdentifier": "videoTrack2", + "videoPacketsLost": 5, + "framesDropped": 2, + "framesDecoded": 50, + "framesPerSecond": 25, + "bytesReceived": 2000, + "jitterBufferDelay": 10, + "lastPacketReceivedTimestamp": 160000, + "fractionLost": 0.05, + "currentTime": 0, + "frameWidth": 1920, + "frameHeight": 1080 + } + ] + }; + + assert(consoleSpy.calledWith(JSON.stringify(localMockStatsProcessed)), 'console.log was not called with the expected arguments'); + + expect(result).to.be.true; + consoleSpy.restore(); + }); + + it("should resolve with false when getStats fails", async function() { + mockPeerConnection.getStats.rejects(new Error("getStats error")); + const result = await adaptor.getStats("stream1"); + expect(result).to.be.false; + }); + + it("should not reinitialize remotePeerConnectionStats for an existing streamId", function() { + const streamId = "existingStream"; + adaptor.remotePeerConnectionStats[streamId] = new PeerStats(streamId); + const initialStats = adaptor.remotePeerConnectionStats[streamId]; + adaptor.checkAndInitializePeerStats(streamId); + expect(adaptor.remotePeerConnectionStats[streamId]).to.equal(initialStats); + }); + + it("should handle null streamId gracefully", function() { + const streamId = null; + expect(() => adaptor.checkAndInitializePeerStats(streamId)).not.to.throw(); + expect(adaptor.remotePeerConnectionStats[streamId]).to.be.undefined; + }); + + it("should handle undefined streamId gracefully", function() { + const streamId = undefined; + expect(() => adaptor.checkAndInitializePeerStats(streamId)).not.to.throw(); + expect(adaptor.remotePeerConnectionStats[streamId]).to.be.undefined; + }); + + }); + + describe("changeBandwidth", function() { + + let adaptor; + + beforeEach(function() { + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + initializeComponents: false, + }); + adaptor.mediaManager = { + changeBandwidth: sinon.fake() + }; + }); + + it("should call mediaManager's changeBandwidth with correct parameters", function() { + const bandwidth = 500; + const streamId = "stream1"; + + adaptor.changeBandwidth(bandwidth, streamId); + + expect(adaptor.mediaManager.changeBandwidth.calledWithMatch(bandwidth, streamId)).to.be.true; + }); + + it("should handle zero bandwidth", function() { + const bandwidth = 0; + const streamId = "stream1"; + + adaptor.changeBandwidth(bandwidth, streamId); + + expect(adaptor.mediaManager.changeBandwidth.calledWithMatch(bandwidth, streamId)).to.be.true; + }); + + it("should handle null streamId", function() { + const bandwidth = 500; + const streamId = null; + + adaptor.changeBandwidth(bandwidth, streamId); + + expect(adaptor.mediaManager.changeBandwidth.calledWithMatch(bandwidth, streamId)).to.be.true; + }); + }); + + describe("enableAudioLevelWhenMuted", function() { + + let adaptor; + + beforeEach(function() { + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + initializeComponents: false, + }); + adaptor.mediaManager = { + enableAudioLevelWhenMuted: sinon.fake() + }; + }); - expect(adaptor.mediaManager.disableAudioLevelWhenMuted.called).to.be.true; - }); + it("should call mediaManager's enableAudioLevelWhenMuted", function() { + adaptor.enableAudioLevelWhenMuted(); - }); + expect(adaptor.mediaManager.enableAudioLevelWhenMuted.called).to.be.true; + }); - describe("getVideoSender", function () { + }); - let adaptor; + describe("disableAudioLevelWhenMuted", function() { - beforeEach(function () { - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - initializeComponents: false, - }); - adaptor.mediaManager = { - getVideoSender: sinon.fake() - }; - }); + let adaptor; - it("should call mediaManager's getVideoSender with correct parameters", function () { - const streamId = "stream1"; + beforeEach(function() { + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + initializeComponents: false, + }); + adaptor.mediaManager = { + disableAudioLevelWhenMuted: sinon.fake() + }; + }); - adaptor.getVideoSender(streamId); + it("should call mediaManager's disableAudioLevelWhenMuted", function() { + adaptor.disableAudioLevelWhenMuted(); - expect(adaptor.mediaManager.getVideoSender.calledWithMatch(streamId)).to.be.true; - }); + expect(adaptor.mediaManager.disableAudioLevelWhenMuted.called).to.be.true; + }); - it("should handle null streamId", function () { - const streamId = null; + }); - adaptor.getVideoSender(streamId); + describe("getVideoSender", function() { - expect(adaptor.mediaManager.getVideoSender.calledWithMatch(streamId)).to.be.true; - }); - }); + let adaptor; - describe("openStream", function () { + beforeEach(function() { + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + initializeComponents: false, + }); + adaptor.mediaManager = { + getVideoSender: sinon.fake() + }; + }); - let adaptor; + it("should call mediaManager's getVideoSender with correct parameters", function() { + const streamId = "stream1"; - beforeEach(function () { - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - initializeComponents: false, - }); - adaptor.mediaManager = { - openStream: sinon.fake() - }; - }); + adaptor.getVideoSender(streamId); - it("should call mediaManager's openStream with correct parameters", function () { - const mediaConstraints = {video: true, audio: true}; - const streamId = "stream1"; + expect(adaptor.mediaManager.getVideoSender.calledWithMatch(streamId)).to.be.true; + }); - adaptor.openStream(mediaConstraints, streamId); + it("should handle null streamId", function() { + const streamId = null; - expect(adaptor.mediaManager.openStream.calledWithMatch(mediaConstraints, streamId)).to.be.true; - }); + adaptor.getVideoSender(streamId); - it("should handle null streamId", function () { - const mediaConstraints = {video: true, audio: true}; - const streamId = null; + expect(adaptor.mediaManager.getVideoSender.calledWithMatch(streamId)).to.be.true; + }); + }); - adaptor.openStream(mediaConstraints, streamId); + describe("openStream", function() { - expect(adaptor.mediaManager.openStream.calledWithMatch(mediaConstraints, streamId)).to.be.true; - }); - }); + let adaptor; - describe("closeStream", function () { + beforeEach(function() { + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + initializeComponents: false, + }); + adaptor.mediaManager = { + openStream: sinon.fake() + }; + }); - let adaptor; + it("should call mediaManager's openStream with correct parameters", function() { + const mediaConstraints = { video: true, audio: true }; + const streamId = "stream1"; - beforeEach(function () { - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - initializeComponents: false, - }); - adaptor.mediaManager = { - closeStream: sinon.fake() - }; - }); + adaptor.openStream(mediaConstraints, streamId); - it("should call mediaManager's closeStream", function () { - adaptor.closeStream(); + expect(adaptor.mediaManager.openStream.calledWithMatch(mediaConstraints, streamId)).to.be.true; + }); - expect(adaptor.mediaManager.closeStream.called).to.be.true; - }); + it("should handle null streamId", function() { + const mediaConstraints = { video: true, audio: true }; + const streamId = null; - }); + adaptor.openStream(mediaConstraints, streamId); - describe("applyConstraints", function () { + expect(adaptor.mediaManager.openStream.calledWithMatch(mediaConstraints, streamId)).to.be.true; + }); + }); - let adaptor; + describe("closeStream", function() { - beforeEach(function () { - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - initializeComponents: false, - }); - adaptor.mediaManager = { - applyConstraints: sinon.fake() - }; - }); + let adaptor; - it("should call mediaManager's applyConstraints", function () { - let constraints = {video: true, audio: true}; + beforeEach(function() { + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + initializeComponents: false, + }); + adaptor.mediaManager = { + closeStream: sinon.fake() + }; + }); - adaptor.applyConstraints(constraints); + it("should call mediaManager's closeStream", function() { + adaptor.closeStream(); - expect(adaptor.mediaManager.applyConstraints.calledWithMatch(constraints)).to.be.true; - }); + expect(adaptor.mediaManager.closeStream.called).to.be.true; + }); - }); + }); - describe("switchVideoCameraFacingMode", function () { + describe("applyConstraints", function() { - let adaptor; + let adaptor; - beforeEach(function () { - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - initializeComponents: false, - }); - adaptor.mediaManager = { - switchVideoCameraFacingMode: sinon.fake() - }; - }); + beforeEach(function() { + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + initializeComponents: false, + }); + adaptor.mediaManager = { + applyConstraints: sinon.fake() + }; + }); - it("should call mediaManager's switchVideoCameraFacingMode", function () { - let streamId = "stream1"; - let facingMode = "user"; + it("should call mediaManager's applyConstraints", function() { + let constraints = { video: true, audio: true }; - adaptor.switchVideoCameraFacingMode(streamId, facingMode); + adaptor.applyConstraints(constraints); - expect(adaptor.mediaManager.switchVideoCameraFacingMode.calledWithMatch(streamId, facingMode)).to.be.true; - }); + expect(adaptor.mediaManager.applyConstraints.calledWithMatch(constraints)).to.be.true; + }); - }); + }); - describe("switchDesktopCaptureWithCamera", function () { + describe("switchVideoCameraFacingMode", function() { - let adaptor; + let adaptor; - beforeEach(function () { - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - initializeComponents: false, - }); - adaptor.mediaManager = { - switchDesktopCaptureWithCamera: sinon.fake() - }; - }); + beforeEach(function() { + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + initializeComponents: false, + }); + adaptor.mediaManager = { + switchVideoCameraFacingMode: sinon.fake() + }; + }); - it("should call mediaManager's switchDesktopCaptureWithCamera", function () { - let streamId = "stream1"; + it("should call mediaManager's switchVideoCameraFacingMode", function() { + let streamId = "stream1"; + let facingMode = "user"; - adaptor.switchDesktopCaptureWithCamera(streamId); + adaptor.switchVideoCameraFacingMode(streamId, facingMode); - expect(adaptor.mediaManager.switchDesktopCaptureWithCamera.calledWithMatch(streamId)).to.be.true; - }); + expect(adaptor.mediaManager.switchVideoCameraFacingMode.calledWithMatch(streamId, facingMode)).to.be.true; + }); - }); + }); - describe("switchAudioInputSource", function () { + describe("switchDesktopCaptureWithCamera", function() { - let adaptor; + let adaptor; - beforeEach(function () { - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - initializeComponents: false, - }); - adaptor.mediaManager = { - switchAudioInputSource: sinon.fake() - }; - }); + beforeEach(function() { + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + initializeComponents: false, + }); + adaptor.mediaManager = { + switchDesktopCaptureWithCamera: sinon.fake() + }; + }); - it("should call mediaManager's switchAudioInputSource", function () { - let streamId = "stream1"; - let deviceId = "deviceId1"; + it("should call mediaManager's switchDesktopCaptureWithCamera", function() { + let streamId = "stream1"; - adaptor.switchAudioInputSource(streamId, deviceId); + adaptor.switchDesktopCaptureWithCamera(streamId); - expect(adaptor.mediaManager.switchAudioInputSource.calledWithMatch(streamId, deviceId)).to.be.true; - }); + expect(adaptor.mediaManager.switchDesktopCaptureWithCamera.calledWithMatch(streamId)).to.be.true; + }); - }); + }); - describe("setVolumeLevel", function () { + describe("switchAudioInputSource", function() { - let adaptor; + let adaptor; - beforeEach(function () { - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - initializeComponents: false, - }); - adaptor.mediaManager = { - setVolumeLevel: sinon.fake() - }; - }); + beforeEach(function() { + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + initializeComponents: false, + }); + adaptor.mediaManager = { + switchAudioInputSource: sinon.fake() + }; + }); - it("should call mediaManager's setVolumeLevel", function () { - let volumeLevel = 50; + it("should call mediaManager's switchAudioInputSource", function() { + let streamId = "stream1"; + let deviceId = "deviceId1"; - adaptor.setVolumeLevel(volumeLevel); + adaptor.switchAudioInputSource(streamId, deviceId); - expect(adaptor.mediaManager.setVolumeLevel.calledWithMatch(volumeLevel)).to.be.true; - }); + expect(adaptor.mediaManager.switchAudioInputSource.calledWithMatch(streamId, deviceId)).to.be.true; + }); - }); + }); - describe("switchDesktopCapture", function () { + describe("setVolumeLevel", function() { - let adaptor; + let adaptor; - beforeEach(function () { - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - initializeComponents: false, - }); - adaptor.mediaManager = { - switchDesktopCapture: sinon.fake() - }; - }); + beforeEach(function() { + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + initializeComponents: false, + }); + adaptor.mediaManager = { + setVolumeLevel: sinon.fake() + }; + }); - it("should call mediaManager's switchDesktopCapture", function () { - let streamId = "stream1"; + it("should call mediaManager's setVolumeLevel", function() { + let volumeLevel = 50; - adaptor.switchDesktopCapture(streamId); + adaptor.setVolumeLevel(volumeLevel); - expect(adaptor.mediaManager.switchDesktopCapture.calledWithMatch(streamId)).to.be.true; - }); + expect(adaptor.mediaManager.setVolumeLevel.calledWithMatch(volumeLevel)).to.be.true; + }); - }); + }); - describe("updateVideoTrack", function () { + describe("switchDesktopCapture", function() { - let adaptor; + let adaptor; - beforeEach(function () { - adaptor = new WebRTCAdaptor({ - websocketURL: "ws://example.com", - initializeComponents: false, - }); - adaptor.mediaManager = { - updateVideoTrack: sinon.fake() - }; - }); + beforeEach(function() { + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + initializeComponents: false, + }); + adaptor.mediaManager = { + switchDesktopCapture: sinon.fake() + }; + }); - it("should call mediaManager's updateVideoTrack", function () { - let stream = "stream0"; - let streamId = "stream1"; - let onEndedCallback = null; - let stopDesktop = false; + it("should call mediaManager's switchDesktopCapture", function() { + let streamId = "stream1"; - adaptor.updateVideoTrack(stream, streamId, onEndedCallback, stopDesktop); + adaptor.switchDesktopCapture(streamId); - expect(adaptor.mediaManager.updateVideoTrack.calledWithMatch(stream, streamId, onEndedCallback, stopDesktop)).to.be.true; - }); + expect(adaptor.mediaManager.switchDesktopCapture.calledWithMatch(streamId)).to.be.true; + }); + + }); + + describe("updateVideoTrack", function() { + + let adaptor; + + beforeEach(function() { + adaptor = new WebRTCAdaptor({ + websocketURL: "ws://example.com", + initializeComponents: false, + }); + adaptor.mediaManager = { + updateVideoTrack: sinon.fake() + }; + }); + + it("should call mediaManager's updateVideoTrack", function() { + let stream = "stream0"; + let streamId = "stream1"; + let onEndedCallback = null; + let stopDesktop = false; + + adaptor.updateVideoTrack(stream, streamId, onEndedCallback, stopDesktop); + + expect(adaptor.mediaManager.updateVideoTrack.calledWithMatch(stream, streamId, onEndedCallback, stopDesktop)).to.be.true; + }); + + + }); - }); }); From 9e92952d7fbba5f51932d0bdcbfe88aac8f5b681 Mon Sep 17 00:00:00 2001 From: burak-58 Date: Sun, 25 Aug 2024 10:18:26 +0300 Subject: [PATCH 03/34] show the assignment in conference sample page --- src/main/webapp/conference.html | 31 ++++++++++++++++++++++--------- 1 file changed, 22 insertions(+), 9 deletions(-) diff --git a/src/main/webapp/conference.html b/src/main/webapp/conference.html index 3f224920..d519f976 100644 --- a/src/main/webapp/conference.html +++ b/src/main/webapp/conference.html @@ -228,8 +228,6 @@

WebRTC Multitrack Conference

- - + + \ No newline at end of file From 9c7fdbd56108c23c2c8ebad917045cb2b321d6aa Mon Sep 17 00:00:00 2001 From: mekya Date: Sat, 31 Aug 2024 14:29:17 +0300 Subject: [PATCH 04/34] Update version to 2.11.1 --- pom.xml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/pom.xml b/pom.xml index 7e670d37..2d3182e1 100644 --- a/pom.xml +++ b/pom.xml @@ -3,7 +3,7 @@ io.antmedia parent - 2.12.0-SNAPSHOT + 2.11.1 4.0.0 io.antmedia.webrtc From dac45d805b3f42b1bbd5ef7e610c03fcd9a83568 Mon Sep 17 00:00:00 2001 From: mekya Date: Sat, 31 Aug 2024 22:58:37 +0300 Subject: [PATCH 05/34] Update version to 2.12.0-SNAPSHOT --- pom.xml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/pom.xml b/pom.xml index 2d3182e1..7e670d37 100644 --- a/pom.xml +++ b/pom.xml @@ -3,7 +3,7 @@ io.antmedia parent - 2.11.1 + 2.12.0-SNAPSHOT 4.0.0 io.antmedia.webrtc From fe9f2eca58c0b1679d9176327ce90b764eaf67f4 Mon Sep 17 00:00:00 2001 From: Mustafa BOLEKEN Date: Mon, 9 Sep 2024 11:08:14 +0300 Subject: [PATCH 06/34] Stop timer and effect --- src/main/js/video-effect.js | 3 +++ 1 file changed, 3 insertions(+) diff --git a/src/main/js/video-effect.js b/src/main/js/video-effect.js index 5d6a2b3c..f538eff4 100644 --- a/src/main/js/video-effect.js +++ b/src/main/js/video-effect.js @@ -209,6 +209,9 @@ export class VideoEffect { case VideoEffect.VIRTUAL_BACKGROUND: case VideoEffect.BLUR_BACKGROUND: case VideoEffect.NO_EFFECT: + //Stop timer + this.stopFpsCalculation(); + await this.#noEffect(); break; default: Logger.warn("Unknown effect name please use the constants VideoEffect.VIRTUAL_BACKGROUND,VideoEffect.BLUR_BACKGROUND or VideoEffect.NO_EFFECT "); From 75347b4c8b480e51d39e65031e4570b445270ad6 Mon Sep 17 00:00:00 2001 From: Mustafa BOLEKEN Date: Mon, 9 Sep 2024 12:10:25 +0300 Subject: [PATCH 07/34] Increate the wait time of resolution change process --- src/main/webapp/player.html | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/main/webapp/player.html b/src/main/webapp/player.html index 2489494c..edbfad4e 100644 --- a/src/main/webapp/player.html +++ b/src/main/webapp/player.html @@ -735,7 +735,7 @@

let overlay = document.getElementById('video-overlay'); getVideo.pause(); overlay.style.display = "block"; - setTimeout(function(){ getVideo.play();overlay.style.display = "none";}, 1000); + setTimeout(function(){ getVideo.play();overlay.style.display = "none";}, 2000); } else { console.log( info + " notification received"); From a8c9ab453325e520d287291470e87f623b6936cc Mon Sep 17 00:00:00 2001 From: Mustafa BOLEKEN Date: Sun, 29 Sep 2024 18:35:59 +0300 Subject: [PATCH 08/34] Update web player version --- embedded-player/package-lock.json | 931 ++++++++++++------------------ embedded-player/package.json | 2 +- 2 files changed, 357 insertions(+), 576 deletions(-) diff --git a/embedded-player/package-lock.json b/embedded-player/package-lock.json index 849ecd73..52c0bde8 100644 --- a/embedded-player/package-lock.json +++ b/embedded-player/package-lock.json @@ -9,7 +9,7 @@ "version": "1.0.0", "license": "ISC", "dependencies": { - "@antmedia/web_player": "2.9.1-SNAPSHOT-2024-May-26-04-53" + "@antmedia/web_player": "2.9.1-SNAPSHOT-2024-Sep-22-05-03" }, "devDependencies": { "@rollup/plugin-babel": "^6.0.4", @@ -35,13 +35,14 @@ } }, "node_modules/@antmedia/videojs-webrtc-plugin": { - "version": "1.2.1", - "resolved": "https://registry.npmjs.org/@antmedia/videojs-webrtc-plugin/-/videojs-webrtc-plugin-1.2.1.tgz", - "integrity": "sha512-C8BdTn7LSkf8vx2OaMIgIjVD9ZSvQp/sgEZTPOWxQqs3DGdNPVoKibJIXQqlTyrT3O+8MmCdZYkA8ievwXlwaQ==", + "version": "1.2.3", + "resolved": "https://registry.npmjs.org/@antmedia/videojs-webrtc-plugin/-/videojs-webrtc-plugin-1.2.3.tgz", + "integrity": "sha512-vQPNaY0W1ZcADNzclAVWUA+WjHw+qcKhn6nf0jLgcsgyZ6mOkze0mMg0IwV492S8ke5ONI34tO5W6dURaPiErA==", + "license": "MIT", "dependencies": { - "@antmedia/webrtc_adaptor": "^2.8.0-SNAPSHOT", + "@antmedia/webrtc_adaptor": "2.9.1-SNAPSHOT-2024-May-17-12-21", "global": "^4.4.0", - "video.js": "^6 || ^7" + "video.js": "^8" }, "engines": { "node": ">=14", @@ -49,33 +50,28 @@ } }, "node_modules/@antmedia/web_player": { - "version": "2.9.1-SNAPSHOT-2024-May-26-04-53", - "resolved": "https://registry.npmjs.org/@antmedia/web_player/-/web_player-2.9.1-SNAPSHOT-2024-May-26-04-53.tgz", - "integrity": "sha512-8pQxkF7cq3mlvR9sRrSLAADv6cwLLjVXqWP77BfdvEhnJ0I+fq7Q+IC1k36SWrD/mz3hJA/A9zrc6/hIQkbCxw==", + "version": "2.9.1-SNAPSHOT-2024-Sep-22-05-03", + "resolved": "https://registry.npmjs.org/@antmedia/web_player/-/web_player-2.9.1-SNAPSHOT-2024-Sep-22-05-03.tgz", + "integrity": "sha512-BEfA9GUORHz2PWE09hAr26zNj/V0IWyKtNNZ/Z3tvBWYPX9uR7A8KCLpACEPhqEx8+Ys11Fbt049pZTNYKsnpQ==", + "license": "ISC", "dependencies": { - "@antmedia/videojs-webrtc-plugin": "^1.2.1", + "@antmedia/videojs-webrtc-plugin": "^1.2.3", "@antmedia/webrtc_adaptor": "2.9.1-SNAPSHOT-2024-May-17-12-21", + "@rollup/plugin-json": "6.1.0", + "@types/videojs-contrib-quality-levels": "^2.0.4", "aframe": "1.5.0", "dashjs": "^4.7.4", "rimraf": "^5.0.5", - "video.js": "^7.18.0", - "videojs-contrib-quality-levels": "^2.2.1", - "videojs-hls-quality-selector": "^1.1.4" + "video.js": "^8.17.3", + "videojs-contrib-quality-levels": "^4.1.0", + "videojs-hls-quality-selector": "^2.0.0" } }, - "node_modules/@antmedia/web_player/node_modules/@antmedia/webrtc_adaptor": { + "node_modules/@antmedia/webrtc_adaptor": { "version": "2.9.1-SNAPSHOT-2024-May-17-12-21", "resolved": "https://registry.npmjs.org/@antmedia/webrtc_adaptor/-/webrtc_adaptor-2.9.1-SNAPSHOT-2024-May-17-12-21.tgz", "integrity": "sha512-fu212FJ+oWnD6qx3vDGpzjy0P2t0YMzYs25ocupGXT3JylCTJMDDJMNc2XkuW4fk2eQoTOdntw+SIDewbqgVNQ==", - "dependencies": { - "@mediapipe/selfie_segmentation": "^0.1.1675465747", - "url": "^0.11.1" - } - }, - "node_modules/@antmedia/webrtc_adaptor": { - "version": "2.9.0", - "resolved": "https://registry.npmjs.org/@antmedia/webrtc_adaptor/-/webrtc_adaptor-2.9.0.tgz", - "integrity": "sha512-Gi38pn2kSEULIE6m2kQ5loHn83MWjLcwmEMkmQZWJmnSPB16Uzzh7gqTRgrmxWkEf0NOd3VHORmpGWeObyCAcg==", + "license": "ISC", "dependencies": { "@mediapipe/selfie_segmentation": "^0.1.1675465747", "url": "^0.11.1" @@ -500,9 +496,10 @@ } }, "node_modules/@babel/runtime": { - "version": "7.23.8", - "resolved": "https://registry.npmjs.org/@babel/runtime/-/runtime-7.23.8.tgz", - "integrity": "sha512-Y7KbAP984rn1VGMbGqKmBLio9V7y5Je9GvU4rQPCPinCyNfUcToxIXl06d59URp/F3LwinvODxab5N/G6qggkw==", + "version": "7.25.6", + "resolved": "https://registry.npmjs.org/@babel/runtime/-/runtime-7.25.6.tgz", + "integrity": "sha512-VBj9MYyDb9tuLq7yzqjgzt6Q+IBQLrGZfdjOekyEirZPHxXWoTSGUTMrpsfi58Up73d13NfYLv8HT9vmznjzhQ==", + "license": "MIT", "dependencies": { "regenerator-runtime": "^0.14.0" }, @@ -783,6 +780,26 @@ "node": ">=10" } }, + "node_modules/@rollup/plugin-json": { + "version": "6.1.0", + "resolved": "https://registry.npmjs.org/@rollup/plugin-json/-/plugin-json-6.1.0.tgz", + "integrity": "sha512-EGI2te5ENk1coGeADSIwZ7G2Q8CJS2sF120T7jLw4xFw9n7wIOXHo+kIYRAoVpJAN+kmqZSoO3Fp4JtoNF4ReA==", + "license": "MIT", + "dependencies": { + "@rollup/pluginutils": "^5.1.0" + }, + "engines": { + "node": ">=14.0.0" + }, + "peerDependencies": { + "rollup": "^1.20.0||^2.0.0||^3.0.0||^4.0.0" + }, + "peerDependenciesMeta": { + "rollup": { + "optional": true + } + } + }, "node_modules/@rollup/plugin-node-resolve": { "version": "15.2.3", "resolved": "https://registry.npmjs.org/@rollup/plugin-node-resolve/-/plugin-node-resolve-15.2.3.tgz", @@ -834,7 +851,6 @@ "version": "5.1.0", "resolved": "https://registry.npmjs.org/@rollup/pluginutils/-/pluginutils-5.1.0.tgz", "integrity": "sha512-XTIWOPPcpvyKI6L1NHo0lFlCyznUEyPmPY1mc3KpPVDYulHSTvyeLNVW00QTLIAFNhR3kYnJTQHeGqU4M3n09g==", - 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"global": "^4.3.2", - "karma-safaritechpreview-launcher": "0.0.6", - "video.js": "^7.5.5", - "videojs-contrib-quality-levels": "^2.0.9" + "global": "^4.4.0", + "video.js": "^8" } }, "videojs-vtt.js": { diff --git a/embedded-player/package.json b/embedded-player/package.json index 42216688..79d39313 100644 --- a/embedded-player/package.json +++ b/embedded-player/package.json @@ -20,7 +20,7 @@ "author": "", "license": "ISC", "dependencies": { - "@antmedia/web_player": "2.9.1-SNAPSHOT-2024-Jun-21-09-02" + "@antmedia/web_player": "2.9.1-SNAPSHOT-2024-Sep-22-05-03" }, "devDependencies": { From 36e6b3325b4ca184ce16a1331b3123ea0c730ba4 Mon Sep 17 00:00:00 2001 From: Last Peony Date: Fri, 4 Oct 2024 14:40:33 +0300 Subject: [PATCH 09/34] Create update-embedded-player.yml update embedded player version via webhook --- .github/workflows/update-embedded-player.yml | 46 ++++++++++++++++++++ 1 file changed, 46 insertions(+) create mode 100644 .github/workflows/update-embedded-player.yml diff --git a/.github/workflows/update-embedded-player.yml b/.github/workflows/update-embedded-player.yml new file mode 100644 index 00000000..d39efdc6 --- /dev/null +++ b/.github/workflows/update-embedded-player.yml @@ -0,0 +1,46 @@ +name: Update @antmedia/web_player Dependency + +on: + repository_dispatch: #triggered via webhook. + types: [update-embedded-player] + +jobs: + update-dependency: + runs-on: ubuntu-latest + steps: + - uses: actions/checkout@v3 + + - name: Use Node.js + uses: actions/setup-node@v3 + with: + node-version: '18' + + - name: Update @antmedia/web_player + working-directory: ./embedded-player + run: | + OLD_VERSION=$(node -p "require('./package.json').dependencies['@antmedia/web_player']") + npm install --save @antmedia/web_player@latest + NEW_VERSION=$(node -p "require('./package.json').dependencies['@antmedia/web_player']") + echo "OLD_EMBEDDED_PLAYER_VERSION=$OLD_EMBEDDED_PLAYER_VERSION" >> $GITHUB_ENV + echo "NEW_EMBEDDED_PLAYER_VERSION=$NEW_EMBEDDED_PLAYER_VERSION" >> $GITHUB_ENV + + - name: Commit changes if version updated + if: env.OLD_VERSION != env.NEW_VERSION + working-directory: ./embedded-player + run: | + git config user.name github-actions + git config user.email github-actions@github.com + git add package.json package-lock.json + git commit -m "Update @antmedia/web_player from ${{ env.OLD_EMBEDDED_PLAYER_VERSION }} to ${{ env.NEW_EMBEDDED_PLAYER_VERSION }}" + + - name: Create Pull Request + if: env.OLD_VERSION != env.NEW_VERSION + uses: peter-evans/create-pull-request@v5 + with: + title: "Update @antmedia/web_player to ${{ env.NEW_EMBEDDED_PLAYER_VERSION }}" + body: | + This PR updates the @antmedia/web_player dependency from ${{ env.OLD_EMBEDDED_PLAYER_VERSION }} to ${{ env.NEW_EMBEDDED_PLAYER_VERSION }}. + + Triggered by webhook from npm package update. + branch: update-antmedia-web-player-${{ env.NEW_EMBEDDED_PLAYER_VERSION }} + delete-branch: true From e8249e2354c5aad1c6307a8142d5d4f97367bef4 Mon Sep 17 00:00:00 2001 From: Last Peony Date: Fri, 4 Oct 2024 17:36:09 +0300 Subject: [PATCH 10/34] Update update-embedded-player.yml --- .github/workflows/update-embedded-player.yml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/.github/workflows/update-embedded-player.yml b/.github/workflows/update-embedded-player.yml index d39efdc6..3e273574 100644 --- a/.github/workflows/update-embedded-player.yml +++ b/.github/workflows/update-embedded-player.yml @@ -1,7 +1,7 @@ name: Update @antmedia/web_player Dependency on: - repository_dispatch: #triggered via webhook. + repository_dispatch: types: [update-embedded-player] jobs: From c9762ecc94b8c41b89572d4fed92198c3094ff45 Mon Sep 17 00:00:00 2001 From: Last Peony Date: Fri, 4 Oct 2024 17:40:03 +0300 Subject: [PATCH 11/34] Update update-embedded-player.yml --- .github/workflows/update-embedded-player.yml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/.github/workflows/update-embedded-player.yml b/.github/workflows/update-embedded-player.yml index 3e273574..043c8b6f 100644 --- a/.github/workflows/update-embedded-player.yml +++ b/.github/workflows/update-embedded-player.yml @@ -35,7 +35,7 @@ jobs: - name: Create Pull Request if: env.OLD_VERSION != env.NEW_VERSION - uses: peter-evans/create-pull-request@v5 + uses: peter-evans/create-pull-request@v7.0.5 with: title: "Update @antmedia/web_player to ${{ env.NEW_EMBEDDED_PLAYER_VERSION }}" body: | From 2e64a729b9992a05d3e393f7c5b40d6d0ffeb385 Mon Sep 17 00:00:00 2001 From: Last Peony Date: Fri, 4 Oct 2024 17:41:29 +0300 Subject: [PATCH 12/34] Update update-embedded-player.yml --- .github/workflows/update-embedded-player.yml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/.github/workflows/update-embedded-player.yml b/.github/workflows/update-embedded-player.yml index 043c8b6f..bbfe6849 100644 --- a/.github/workflows/update-embedded-player.yml +++ b/.github/workflows/update-embedded-player.yml @@ -25,7 +25,7 @@ jobs: echo "NEW_EMBEDDED_PLAYER_VERSION=$NEW_EMBEDDED_PLAYER_VERSION" >> $GITHUB_ENV - name: Commit changes if version updated - if: env.OLD_VERSION != env.NEW_VERSION + if: env.OLD_EMBEDDED_PLAYER_VERSION != env.NEW_EMBEDDED_PLAYER_VERSION working-directory: ./embedded-player run: | git config user.name github-actions @@ -34,7 +34,7 @@ jobs: git commit -m "Update @antmedia/web_player from ${{ env.OLD_EMBEDDED_PLAYER_VERSION }} to ${{ env.NEW_EMBEDDED_PLAYER_VERSION }}" - name: Create Pull Request - if: env.OLD_VERSION != env.NEW_VERSION + if: env.OLD_EMBEDDED_PLAYER_VERSION != env.NEW_EMBEDDED_PLAYER_VERSION uses: peter-evans/create-pull-request@v7.0.5 with: title: "Update @antmedia/web_player to ${{ env.NEW_EMBEDDED_PLAYER_VERSION }}" From 8116b338d0e845a1392ac832ef818ed649875c18 Mon Sep 17 00:00:00 2001 From: mekya Date: Sun, 6 Oct 2024 08:41:06 +0300 Subject: [PATCH 13/34] Update web player version --- embedded-player/package.json | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/embedded-player/package.json b/embedded-player/package.json index 79d39313..04413903 100644 --- a/embedded-player/package.json +++ b/embedded-player/package.json @@ -20,7 +20,7 @@ "author": "", "license": "ISC", "dependencies": { - "@antmedia/web_player": "2.9.1-SNAPSHOT-2024-Sep-22-05-03" + "@antmedia/web_player": "2.12.0-SNAPSHOT-2024-Oct-05-02-55" }, "devDependencies": { From 4c4365dc7356ece6ff4724ce8c4e186642e6de3b Mon Sep 17 00:00:00 2001 From: mekya Date: Sun, 6 Oct 2024 10:47:16 +0300 Subject: [PATCH 14/34] Update web player version --- embedded-player/package.json | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/embedded-player/package.json b/embedded-player/package.json index 04413903..3c2b4101 100644 --- a/embedded-player/package.json +++ b/embedded-player/package.json @@ -20,7 +20,7 @@ "author": "", "license": "ISC", "dependencies": { - "@antmedia/web_player": "2.12.0-SNAPSHOT-2024-Oct-05-02-55" + "@antmedia/web_player": "^2.11.1" }, "devDependencies": { From 21b828f9c4a6d863903ff1d37915b71c99576fd6 Mon Sep 17 00:00:00 2001 From: Last Peony Date: Sun, 6 Oct 2024 22:21:40 +0300 Subject: [PATCH 15/34] Update update-embedded-player.yml small fix --- .github/workflows/update-embedded-player.yml | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/.github/workflows/update-embedded-player.yml b/.github/workflows/update-embedded-player.yml index bbfe6849..9bfddd8e 100644 --- a/.github/workflows/update-embedded-player.yml +++ b/.github/workflows/update-embedded-player.yml @@ -18,9 +18,9 @@ jobs: - name: Update @antmedia/web_player working-directory: ./embedded-player run: | - OLD_VERSION=$(node -p "require('./package.json').dependencies['@antmedia/web_player']") + OLD_EMBEDDED_PLAYER_VERSION=$(node -p "require('./package.json').dependencies['@antmedia/web_player']") npm install --save @antmedia/web_player@latest - NEW_VERSION=$(node -p "require('./package.json').dependencies['@antmedia/web_player']") + NEW_EMBEDDED_PLAYER_VERSION=$(node -p "require('./package.json').dependencies['@antmedia/web_player']") echo "OLD_EMBEDDED_PLAYER_VERSION=$OLD_EMBEDDED_PLAYER_VERSION" >> $GITHUB_ENV echo "NEW_EMBEDDED_PLAYER_VERSION=$NEW_EMBEDDED_PLAYER_VERSION" >> $GITHUB_ENV @@ -41,6 +41,6 @@ jobs: body: | This PR updates the @antmedia/web_player dependency from ${{ env.OLD_EMBEDDED_PLAYER_VERSION }} to ${{ env.NEW_EMBEDDED_PLAYER_VERSION }}. - Triggered by webhook from npm package update. + Triggered by publish-release.yml from web player repo. branch: update-antmedia-web-player-${{ env.NEW_EMBEDDED_PLAYER_VERSION }} delete-branch: true From 6b0828e8a9c0adef8e40ba894fe5c2e8b4249777 Mon Sep 17 00:00:00 2001 From: Last Peony Date: Sun, 6 Oct 2024 22:26:17 +0300 Subject: [PATCH 16/34] Update update-embedded-player.yml --- .github/workflows/update-embedded-player.yml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/.github/workflows/update-embedded-player.yml b/.github/workflows/update-embedded-player.yml index 9bfddd8e..c10ea8f5 100644 --- a/.github/workflows/update-embedded-player.yml +++ b/.github/workflows/update-embedded-player.yml @@ -19,7 +19,7 @@ jobs: working-directory: ./embedded-player run: | OLD_EMBEDDED_PLAYER_VERSION=$(node -p "require('./package.json').dependencies['@antmedia/web_player']") - npm install --save @antmedia/web_player@latest + npm install --save @antmedia/web_player@latest || { echo 'npm install failed' ; exit 1; } NEW_EMBEDDED_PLAYER_VERSION=$(node -p "require('./package.json').dependencies['@antmedia/web_player']") echo "OLD_EMBEDDED_PLAYER_VERSION=$OLD_EMBEDDED_PLAYER_VERSION" >> $GITHUB_ENV echo "NEW_EMBEDDED_PLAYER_VERSION=$NEW_EMBEDDED_PLAYER_VERSION" >> $GITHUB_ENV From 87ab1b8472abaea0b476d58995f29d3f28979b1e Mon Sep 17 00:00:00 2001 From: mekya Date: Mon, 7 Oct 2024 23:07:28 +0300 Subject: [PATCH 17/34] Update web player --- embedded-player/package.json | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/embedded-player/package.json b/embedded-player/package.json index 3c2b4101..187e7502 100644 --- a/embedded-player/package.json +++ b/embedded-player/package.json @@ -20,7 +20,7 @@ "author": "", "license": "ISC", "dependencies": { - "@antmedia/web_player": "^2.11.1" + "@antmedia/web_player": "^2.11.2" }, "devDependencies": { From a35c14b387932bb3886fb987373979321b90614f Mon Sep 17 00:00:00 2001 From: Last Peony Date: Tue, 8 Oct 2024 14:29:25 +0300 Subject: [PATCH 18/34] Update update-embedded-player.yml automaticly create PR on new web player --- .github/workflows/update-embedded-player.yml | 1 + 1 file changed, 1 insertion(+) diff --git a/.github/workflows/update-embedded-player.yml b/.github/workflows/update-embedded-player.yml index c10ea8f5..56af441f 100644 --- a/.github/workflows/update-embedded-player.yml +++ b/.github/workflows/update-embedded-player.yml @@ -1,6 +1,7 @@ name: Update @antmedia/web_player Dependency on: + workflow_dispatch: repository_dispatch: types: [update-embedded-player] From f10d76724fb4bcc84fb6a7fa3e9799f1776d1761 Mon Sep 17 00:00:00 2001 From: Last Peony Date: Tue, 8 Oct 2024 14:43:36 +0300 Subject: [PATCH 19/34] Update update-embedded-player.yml update embedded player version auto --- .github/workflows/update-embedded-player.yml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/.github/workflows/update-embedded-player.yml b/.github/workflows/update-embedded-player.yml index 56af441f..fabfe2de 100644 --- a/.github/workflows/update-embedded-player.yml +++ b/.github/workflows/update-embedded-player.yml @@ -19,9 +19,9 @@ jobs: - name: Update @antmedia/web_player working-directory: ./embedded-player run: | - OLD_EMBEDDED_PLAYER_VERSION=$(node -p "require('./package.json').dependencies['@antmedia/web_player']") + OLD_EMBEDDED_PLAYER_VERSION=$(node -p "require('./package.json').dependencies['@antmedia/web_player'].replace(/\\^/, '')") npm install --save @antmedia/web_player@latest || { echo 'npm install failed' ; exit 1; } - NEW_EMBEDDED_PLAYER_VERSION=$(node -p "require('./package.json').dependencies['@antmedia/web_player']") + NEW_EMBEDDED_PLAYER_VERSION=$(node -p "require('./package.json').dependencies['@antmedia/web_player'].replace(/\\^/, '')") echo "OLD_EMBEDDED_PLAYER_VERSION=$OLD_EMBEDDED_PLAYER_VERSION" >> $GITHUB_ENV echo "NEW_EMBEDDED_PLAYER_VERSION=$NEW_EMBEDDED_PLAYER_VERSION" >> $GITHUB_ENV From f01ca450d5377547f5a0599f8ce05c5788befd53 Mon Sep 17 00:00:00 2001 From: mekya Date: Sun, 13 Oct 2024 21:55:40 +0300 Subject: [PATCH 20/34] Support role in conference sample --- src/main/webapp/conference.html | 47 +++++++++++++++++++++++---------- 1 file changed, 33 insertions(+), 14 deletions(-) diff --git a/src/main/webapp/conference.html b/src/main/webapp/conference.html index f6ba55f5..27dd9122 100644 --- a/src/main/webapp/conference.html +++ b/src/main/webapp/conference.html @@ -244,15 +244,18 @@

WebRTC Multitrack Conference

var token = getUrlParameter("token"); var publishStreamId = getUrlParameter("streamId"); var playOnly = getUrlParameter("playOnly"); - var dcOnly = getUrlParameter("dcOnly"); - if (playOnly == null) { - playOnly = false; + playOnly = false; } - + + var dcOnly = getUrlParameter("dcOnly"); + if (dcOnly == null) { dcOnly = false; } + + var role = getUrlParameter("role"); + var roomId = getUrlParameter("roomId"); var streamName = getUrlParameter("streamName"); @@ -405,21 +408,32 @@

WebRTC Multitrack Conference

function sendData() { try { var iceState = webRTCAdaptor.iceConnectionState(publishStreamId); + var msg = $("#dataTextbox").val(); + var notEvent = { streamId: publishStreamId, eventType: "CHAT_MESSAGE", message: msg }; + if (iceState != null && iceState != "failed" && iceState != "disconnected") { - var msg = $("#dataTextbox").val(); - var notEvent = { streamId: publishStreamId, eventType: "CHAT_MESSAGE", message: msg }; + webRTCAdaptor.sendData(publishStreamId, JSON.stringify(notEvent)); $("#all-messages").append("Sent: " + msg + "
"); $("#dataTextbox").val(""); } else { - $.notify("WebRTC playing is not active. Please click Start Playing first", { - autoHideDelay: 5000, - className: 'error', - position: 'top center' - }); + iceState = webRTCAdaptor.iceConnectionState(roomNameBox.value); + if (iceState != null && iceState != "failed" && iceState != "disconnected") + { + webRTCAdaptor.sendData(roomNameBox.value, JSON.stringify(notEvent)); + $("#all-messages").append("Sent: " + msg + "
"); + $("#dataTextbox").val(""); + } + else { + $.notify("WebRTC playing is not active. Please click Start Playing first", { + autoHideDelay: 5000, + className: 'error', + position: 'top center' + }); + } } } catch (exception) { @@ -504,8 +518,10 @@

WebRTC Multitrack Conference

publishStreamId = generateRandomString(12); } - publish(publishStreamId, token); - webRTCAdaptor.play(roomNameBox.value, token, roomNameBox.value, [], subscriberId, subscriberCode); + if (!playOnly) { + publish(publishStreamId, token); + } + webRTCAdaptor.play(roomNameBox.value, token, roomNameBox.value, [], subscriberId, subscriberCode, null, role); } function leaveRoom() { @@ -526,7 +542,7 @@

WebRTC Multitrack Conference

if (streamName == null || streamName == 'undefined' || streamName == "") { streamName = streamId; } - webRTCAdaptor.publish(publishStreamId, token, subscriberId, subscriberCode, streamName, roomNameBox.value, JSON.stringify(metadata)); + webRTCAdaptor.publish(publishStreamId, token, subscriberId, subscriberCode, streamName, roomNameBox.value, JSON.stringify(metadata), role); } @@ -765,6 +781,9 @@

WebRTC Multitrack Conference

else if (info == "play_started") { isPlaying = true; isNoSreamExist = false; + join_publish_button.disabled = true; + stop_publish_button.disabled = false; + webRTCAdaptor.getBroadcastObject(roomNameBox.value); if (reconnecting) { playReconnected = true; From d33fe79742d253201c161771a22d165b02eb4694 Mon Sep 17 00:00:00 2001 From: mekya Date: Fri, 18 Oct 2024 20:17:56 +0300 Subject: [PATCH 21/34] Add playback stats for WebRTC and add test codes --- src/main/js/peer_stats.js | 37 +++++- src/main/js/webrtc_adaptor.js | 28 +++- src/main/js/websocket_adaptor.js | 7 +- src/test/js/webrtc_adaptor.test.js | 198 ++++++++++++++++++++++++++++- 4 files changed, 260 insertions(+), 10 deletions(-) diff --git a/src/main/js/peer_stats.js b/src/main/js/peer_stats.js index 22df29f5..72ae43df 100644 --- a/src/main/js/peer_stats.js +++ b/src/main/js/peer_stats.js @@ -66,18 +66,44 @@ export class PeerStats { this.lastBytesSent = 0; /** + * @deprecated use videoPacketsSent * The total number of video packets sent. * @type {number} */ this.totalVideoPacketsSent = 0; + /** + * The total number of video packets sent. + * @type {number} + */ + this.videoPacketsSent = 0; + /** + * The total number of video packets received. + * @type {number} + */ + this.videoPacketsReceived = 0; + /** - * The total number of audio packets sent. + * @deprecated use audioPacketsSent + * The total number of audio packets sent. * @type {number} */ this.totalAudioPacketsSent = 0; - - /** + + /** + * + * The total number of audio packets sent. + * @type {number} + */ + this.audioPacketsSent = 0; + /* + * The total number of audio packets received. + * @type {number} + * + * */ + this.audioPacketsReceived = 0; + + /** * The current timestamp. * @type {number} */ @@ -216,6 +242,11 @@ export class PeerStats { * @type {*[]} */ this.inboundRtpList = []; + + /** + * The current round trip time for the candidate pair + */ + this.currentRoundTripTime = 0; } //kbits/sec get averageOutgoingBitrate() { diff --git a/src/main/js/webrtc_adaptor.js b/src/main/js/webrtc_adaptor.js index 26953be0..7e9ad425 100644 --- a/src/main/js/webrtc_adaptor.js +++ b/src/main/js/webrtc_adaptor.js @@ -1499,12 +1499,18 @@ export class WebRTCAdaptor { var audioJitterAverageDelay = -1; var videoJitterAverageDelay = -1; var availableOutgoingBitrate = Infinity; + var currentRoundTripTime = -1; + + var audioPacketsReceived = -1; + var videoPacketsReceived = -1; var inboundRtp = []; stats.forEach(value => { //Logger.debug(value); if (value.type == "inbound-rtp" && typeof value.kind != "undefined") { + //this is coming when webrtc playing + let inboundRtpObj = {}; inboundRtpObj.trackIdentifier = value.trackIdentifier; @@ -1512,6 +1518,9 @@ export class WebRTCAdaptor { bytesReceived += value.bytesReceived; if (value.kind == "audio") { audioPacketsLost = value.packetsLost; + audioJitter = value.jitter; + audioPacketsReceived = value.packetsReceived; + inboundRtpObj.audioPacketsLost = value.packetsLost; } else if (value.kind == "video") { videoPacketsLost = value.packetsLost; @@ -1519,6 +1528,8 @@ export class WebRTCAdaptor { inboundRtpObj.framesDropped = value.framesDropped; inboundRtpObj.framesDecoded = value.framesDecoded; inboundRtpObj.framesPerSecond = value.framesPerSecond; + videoJitter = value.jitter; + videoPacketsReceived = value.packetsReceived; } inboundRtpObj.bytesReceived = value.bytesReceived; @@ -1556,7 +1567,11 @@ export class WebRTCAdaptor { inboundRtp.push(inboundRtpObj); - } else if (value.type == "outbound-rtp") {//TODO: SPLIT AUDIO AND VIDEO BITRATES + } + else if (value.type == "outbound-rtp") + { + //TODO: SPLIT AUDIO AND VIDEO BITRATES + //it is for the publishing if (value.kind == "audio") { audioPacketsSent = value.packetsSent; } else if (value.kind == "video") { @@ -1640,6 +1655,8 @@ export class WebRTCAdaptor { } else if(value.type == "candidate-pair" && value.state == "succeeded" && value.availableOutgoingBitrate !=undefined){ availableOutgoingBitrate = value.availableOutgoingBitrate/1000 + //currentRoundTripTime + currentRoundTripTime = value.currentRoundTripTime; } }); @@ -1655,6 +1672,9 @@ export class WebRTCAdaptor { this.remotePeerConnectionStats[streamId].totalBytesSent = bytesSent; this.remotePeerConnectionStats[streamId].totalVideoPacketsSent = videoPacketsSent; this.remotePeerConnectionStats[streamId].totalAudioPacketsSent = audioPacketsSent; + this.remotePeerConnectionStats[streamId].videoPacketsSent = videoPacketsSent; + this.remotePeerConnectionStats[streamId].audioPacketsSent = audioPacketsSent; + this.remotePeerConnectionStats[streamId].audioLevel = audioLevel; this.remotePeerConnectionStats[streamId].qualityLimitationReason = qlr; this.remotePeerConnectionStats[streamId].totalFramesEncoded = framesEncoded; @@ -1676,9 +1696,13 @@ export class WebRTCAdaptor { this.remotePeerConnectionStats[streamId].availableOutgoingBitrate = availableOutgoingBitrate; this.remotePeerConnectionStats[streamId].inboundRtpList = inboundRtp; + + this.remotePeerConnectionStats[streamId].currentRoundTripTime = currentRoundTripTime; + this.remotePeerConnectionStats[streamId].audioPacketsReceived = audioPacketsReceived; + this.remotePeerConnectionStats[streamId].videoPacketsReceived = videoPacketsReceived; this.notifyEventListeners("updated_stats", this.remotePeerConnectionStats[streamId]); - resolve(true); + resolve(this.remotePeerConnectionStats[streamId]); }).catch(err=>{ resolve(false); }); diff --git a/src/main/js/websocket_adaptor.js b/src/main/js/websocket_adaptor.js index a34ecb60..3ad46873 100644 --- a/src/main/js/websocket_adaptor.js +++ b/src/main/js/websocket_adaptor.js @@ -50,9 +50,8 @@ export class WebSocketAdaptor { this.wsConn = new WebSocket(this.websocket_url); this.wsConn.onopen = () => { - if (this.debug) { - Logger.debug("websocket connected"); - } + Logger.debug("websocket connected"); + this.pingTimerId = setInterval(() => { this.sendPing(); @@ -182,7 +181,7 @@ export class WebSocketAdaptor { } } catch (error) { - Logger.warn("Cannot send message:" + text); + Logger.warn("Make sure you call methods after you receive initialized callback. Cannot send message:" + text + " Error is " + error); } } diff --git a/src/test/js/webrtc_adaptor.test.js b/src/test/js/webrtc_adaptor.test.js index 1d19038e..cd5927be 100644 --- a/src/test/js/webrtc_adaptor.test.js +++ b/src/test/js/webrtc_adaptor.test.js @@ -12,6 +12,8 @@ describe("WebRTCAdaptor", function() { var initialized = false; var currentTest; + + var processStarted = false; beforeEach(function() { clock = sinon.useFakeTimers(); @@ -197,7 +199,7 @@ describe("WebRTCAdaptor", function() { }); - it.only("should set connected and connecting to false and log the correct message", function() { + it("should set connected and connecting to false and log the correct message", function() { var adaptor = new WebRTCAdaptor({ websocketURL: "ws://example.com", @@ -1711,6 +1713,200 @@ describe("WebRTCAdaptor", function() { }); + + it("WebRTCGetStats", async function() + { + + clock.restore(); + + this.timeout(15000); + + var websocketURL = "wss://test.antmedia.io/live/websocket"; + processStarted = false; + initialized = false; + var adaptor = new WebRTCAdaptor({ + websocketURL: websocketURL, + callback: (info, obj) => { + console.log("callback info: " + info); + if (info == "initialized") { + initialized = true; + } + else if (info == "publish_started") { + console.log("publish started"); + processStarted = true; + } + else if (info == "publish_finished") { + console.log("publish finished") + } + }, + }); + + await new Promise((resolve, reject)=>{ + setTimeout(()=> { + resolve(); + }, 3000); + }); + + expect(initialized).to.be.true; + + var streamId = "stream1desadafg23424"; + + adaptor.publish(streamId); + + await new Promise((resolve, reject)=>{ + + setTimeout(()=> { + expect(processStarted).to.be.true; + resolve(); + }, 3000); + }); + + //getStats + var peerStats = await adaptor.getStats(streamId); + + console.log("publish peerStats: " + JSON.stringify(peerStats)); + expect(peerStats.streamId).to.be.equal(streamId); + expect(peerStats.audioPacketsSent).to.be.above(0); + expect(peerStats.videoPacketsSent).to.be.above(0); + expect(peerStats.frameWidth).to.be.above(0); + expect(peerStats.frameHeight).to.be.above(0); + expect(peerStats.currentRoundTripTime).to.be.above(0); + expect(peerStats.currentRoundTripTime).to.be.most(1); + + expect(peerStats.videoPacketsLost).to.be.least(0); + expect(peerStats.audioPacketsLost).to.be.least(0); + expect(peerStats.videoJitter).to.be.least(0); + expect(peerStats.audioJitter).to.be.least(0); + expect(peerStats.totalBytesSentCount).to.be.above(0); + expect(peerStats.lastFramesEncoded).to.be.above(0); + expect(peerStats.totalFramesEncodedCount).to.be.above(0); + expect(peerStats.frameWidth).to.be.equal(640); + expect(peerStats.frameHeight).to.be.equal(480); + expect(peerStats.qualityLimitationReason).to.be.equal("none"); + expect(peerStats.firstByteSentCount).to.be.not.equal(0); + expect(peerStats.srcFps).to.be.above(0); + expect(peerStats.videoRoundTripTime).to.be.above(0); + //expect(peerStats.audioRoundTripTime).to.be.above(0); + expect(peerStats.availableOutgoingBitrate).to.be.above(0); + + + + + expect(peerStats.totalBytesReceivedCount).to.be.equal(-1); + expect(peerStats.lastBytesSent).to.be.equal(0); + expect(peerStats.videoPacketsLost).to.be.equal(0); + expect(peerStats.fractionLost).to.be.equal(-1); + expect(peerStats.startTime).to.be.not.equal(0); + expect(peerStats.lastBytesReceived).to.be.equal(0); + expect(peerStats.currentTimestamp).to.be.not.equal(0); + expect(peerStats.lastTime).to.be.equal(0); + expect(peerStats.timerId).to.be.equal(0); + expect(peerStats.firstBytesReceivedCount).to.be.equal(-1); + expect(peerStats.audioLevel).to.be.equal(-1); + expect(peerStats.resWidth).to.be.equal(640); + expect(peerStats.resHeight).to.be.equal(480); + expect(peerStats.framesReceived).to.be.equal(-1); + expect(peerStats.framesDropped).to.be.equal(-1); + expect(peerStats.framesDecoded).to.be.equal(-1); + expect(peerStats.audioJitterAverageDelay).to.be.equal(-1); + expect(peerStats.videoJitterAverageDelay).to.be.equal(-1); + expect(peerStats.inboundRtpList).to.be.empty; + expect(peerStats.audioPacketsReceived).to.be.equal(-1); + expect(peerStats.videoPacketsReceived).to.be.equal(-1); + + //getStats + processStarted = false; + initialized = false; + var playAdaptor = new WebRTCAdaptor({ + websocketURL: websocketURL, + callback: (info, obj) => { + console.log("callback info: " + info); + if (info == "initialized") { + initialized = true; + } + else if (info == "play_started") { + console.log("play started"); + processStarted = true; + } + else if (info == "play_finished") { + console.log("play finished") + } + }, + }); + await new Promise((resolve, reject)=>{ + setTimeout(()=> { + resolve(); + }, 3000); + }); + + expect(initialized).to.be.true; + + playAdaptor.play(streamId); + + await new Promise((resolve, reject)=>{ + + setTimeout(()=> { + expect(processStarted).to.be.true; + resolve(); + }, 3000); + }); + + peerStats = await playAdaptor.getStats(streamId); + + console.log("play peerStats: " + JSON.stringify(peerStats)); + expect(peerStats.streamId).to.be.equal(streamId); + expect(peerStats.frameWidth).to.be.equal(640); + expect(peerStats.frameHeight).to.be.equal(480); + expect(peerStats.currentRoundTripTime).to.be.above(0); + expect(peerStats.currentRoundTripTime).to.be.most(1); + + expect(peerStats.videoPacketsLost).to.be.least(0); + expect(peerStats.audioPacketsLost).to.be.least(0); + expect(peerStats.videoJitter).to.be.least(0); + expect(peerStats.audioJitter).to.be.least(0); + expect(peerStats.lastFramesEncoded).to.be.equal(-1); + expect(peerStats.totalFramesEncodedCount).to.be.equal(-1); + expect(peerStats.frameWidth).to.be.equal(640); + expect(peerStats.frameHeight).to.be.equal(480); + expect(peerStats.qualityLimitationReason).to.be.equal(""); + expect(peerStats.firstByteSentCount).to.be.not.equal(0); + expect(peerStats.srcFps).to.be.equal(-1); + expect(peerStats.videoRoundTripTime).to.be.equal(-1); + expect(peerStats.audioRoundTripTime).to.be.equal(-1); + expect(peerStats.availableOutgoingBitrate).to.be.above(-1); + + + + + expect(peerStats.totalBytesReceivedCount).to.be.above(0); + expect(peerStats.lastBytesSent).to.be.equal(0); + expect(peerStats.videoPacketsLost).to.be.equal(0); + //expect(peerStats.fractionLost).to.be.equal(-1); + expect(peerStats.startTime).to.be.not.equal(0); + expect(peerStats.lastBytesReceived).to.be.equal(0); + expect(peerStats.currentTimestamp).to.be.not.equal(0); + expect(peerStats.lastTime).to.be.equal(0); + expect(peerStats.timerId).to.be.equal(0); + expect(peerStats.firstBytesReceivedCount).to.be.above(0); + expect(peerStats.audioLevel).to.be.equal(-1); + expect(peerStats.resWidth).to.be.equal(-1); + expect(peerStats.resHeight).to.be.equal(-1); + expect(peerStats.framesReceived).to.be.above(0); + expect(peerStats.framesDropped).to.be.least(0); + expect(peerStats.framesDecoded).to.be.above(0); + expect(peerStats.audioJitterAverageDelay).to.be.equal(-1); + expect(peerStats.videoJitterAverageDelay).to.be.equal(-1); + expect(peerStats.audioPacketsReceived).to.be.above(0); + expect(peerStats.videoPacketsReceived).to.be.above(0); + + + expect(peerStats.totalBytesSentCount).to.be.equal(-1); + expect(peerStats.totalAudioPacketsSent).to.be.equal(-1); + expect(peerStats.totalVideoPacketsSent).to.be.equal(-1); + + + + }); From 3bcafe7e0f9a2900dfd79f670872ae32d96c60ae Mon Sep 17 00:00:00 2001 From: mekya Date: Fri, 18 Oct 2024 21:38:37 +0300 Subject: [PATCH 22/34] Fix test case --- src/test/js/stream.merger.test.js | 4 ++ src/test/js/webrtc_adaptor.test.js | 68 ++++++++++++++++++++++++------ 2 files changed, 58 insertions(+), 14 deletions(-) diff --git a/src/test/js/stream.merger.test.js b/src/test/js/stream.merger.test.js index 697cf46e..cfd0c017 100644 --- a/src/test/js/stream.merger.test.js +++ b/src/test/js/stream.merger.test.js @@ -340,6 +340,7 @@ describe("StreamMerger", function () { }); it("should update the layout based on the received message", function () { + this.timeout(10000); const layoutData = { layoutOptions: { canvas: { @@ -524,6 +525,7 @@ it('should mute and unmute the stream correctly', function() { }); it("should remove the specified stream from the streams array", function () { + this.timeout(10000); const mediaStream1 = createMockMediaStream(); const options1 = { streamId: "stream1", width: 320, height: 240, Xindex: 0, Yindex: 0, mute: false }; @@ -547,6 +549,8 @@ it("should remove the specified stream from the streams array", function () { it("should remove all streams from the streams array", function () { + this.timeout(10000); + const mediaStream1 = createMockMediaStream(); const options1 = { streamId: "stream1", width: 320, height: 240, Xindex: 0, Yindex: 0, mute: false }; diff --git a/src/test/js/webrtc_adaptor.test.js b/src/test/js/webrtc_adaptor.test.js index cd5927be..6d3e2946 100644 --- a/src/test/js/webrtc_adaptor.test.js +++ b/src/test/js/webrtc_adaptor.test.js @@ -130,7 +130,7 @@ describe("WebRTCAdaptor", function() { adaptor.enableStats(streamId); expect(adaptor.remotePeerConnectionStats[streamId]).to.not.be.undefined - expect(await adaptor.getStats(streamId)).to.be.true; + expect(await adaptor.getStats(streamId)).to.be.not.null; console.log(adaptor.remotePeerConnectionStats[streamId]) @@ -586,6 +586,7 @@ describe("WebRTCAdaptor", function() { adaptor.enableAudioLevelForLocalStream((level) => { console.log("sound level -> " + level); if (level > 0) { + adaptor.disableAudioLevelForLocalStream(); done(); } }); @@ -1201,7 +1202,7 @@ describe("WebRTCAdaptor", function() { it("should resolve with true when getStats is successful", async function() { mockPeerConnection.getStats.resolves(mockStats); const result = await adaptor.getStats("stream1"); - expect(result).to.be.true; + expect(result).to.be.not.null; }); it("should correctly process inbound RTP with audio kind", async function() { @@ -1261,9 +1262,17 @@ describe("WebRTCAdaptor", function() { ] }; - assert(consoleSpy.calledWith(JSON.stringify(localMockStatsProcessed)), 'console.log was not called with the expected arguments'); - expect(result).to.be.true; + expect(result).to.be.not.null; + expect(result.inboundRtpList[0].trackIdentifier).to.equal("audioTrack1"); + expect(result.inboundRtpList[0].audioPacketsLost).to.equal(10); + expect(result.inboundRtpList[0].bytesReceived).to.equal(1000); + expect(result.inboundRtpList[0].jitterBufferDelay).to.equal(5); + expect(result.inboundRtpList[0].lastPacketReceivedTimestamp).to.equal(160000); + expect(result.inboundRtpList[0].fractionLost).to.equal(0.1); + expect(result.inboundRtpList[0].currentTime).to.equal(0); + + consoleSpy.restore(); }); @@ -1334,9 +1343,23 @@ describe("WebRTCAdaptor", function() { ] }; - assert(consoleSpy.calledWith(JSON.stringify(localMockStatsProcessed)), 'console.log was not called with the expected arguments'); + //assert(consoleSpy.calledWith(JSON.stringify(localMockStatsProcessed)), 'console.log was not called with the expected arguments'); + + expect(result).to.be.not.null; + expect(result.inboundRtpList[0].trackIdentifier).to.equal("videoTrack2"); + expect(result.inboundRtpList[0].videoPacketsLost).to.equal(5); + expect(result.inboundRtpList[0].framesDropped).to.equal(2); + expect(result.inboundRtpList[0].framesDecoded).to.equal(50); + expect(result.inboundRtpList[0].framesPerSecond).to.equal(25); + expect(result.inboundRtpList[0].bytesReceived).to.equal(2000); + expect(result.inboundRtpList[0].jitterBufferDelay).to.equal(10); + expect(result.inboundRtpList[0].lastPacketReceivedTimestamp).to.equal(160000); + expect(result.inboundRtpList[0].fractionLost).to.equal(0.05); + expect(result.inboundRtpList[0].currentTime).to.equal(0); + expect(result.inboundRtpList[0].frameWidth).to.equal(1920); + expect(result.inboundRtpList[0].frameHeight).to.equal(1080); + - expect(result).to.be.true; consoleSpy.restore(); }); @@ -1717,9 +1740,18 @@ describe("WebRTCAdaptor", function() { it("WebRTCGetStats", async function() { + const randomAlphaNumeric = length => { + let s = ''; + Array.from({ length }).some(() => { + s += Math.random().toString(36).slice(2); + return s.length >= length; + }); + return s.slice(0, length); + }; + clock.restore(); - this.timeout(15000); + this.timeout(25000); var websocketURL = "wss://test.antmedia.io/live/websocket"; processStarted = false; @@ -1744,12 +1776,12 @@ describe("WebRTCAdaptor", function() { await new Promise((resolve, reject)=>{ setTimeout(()=> { resolve(); - }, 3000); + }, 5000); }); expect(initialized).to.be.true; - var streamId = "stream1desadafg23424"; + var streamId = "stream1desadafg23424" + randomAlphaNumeric(24); adaptor.publish(streamId); @@ -1758,7 +1790,7 @@ describe("WebRTCAdaptor", function() { setTimeout(()=> { expect(processStarted).to.be.true; resolve(); - }, 3000); + }, 5000); }); //getStats @@ -1819,6 +1851,7 @@ describe("WebRTCAdaptor", function() { initialized = false; var playAdaptor = new WebRTCAdaptor({ websocketURL: websocketURL, + isPlayMode: true, callback: (info, obj) => { console.log("callback info: " + info); if (info == "initialized") { @@ -1836,7 +1869,7 @@ describe("WebRTCAdaptor", function() { await new Promise((resolve, reject)=>{ setTimeout(()=> { resolve(); - }, 3000); + }, 5000); }); expect(initialized).to.be.true; @@ -1846,10 +1879,12 @@ describe("WebRTCAdaptor", function() { await new Promise((resolve, reject)=>{ setTimeout(()=> { - expect(processStarted).to.be.true; + resolve(); - }, 3000); + }, 5000); }); + + expect(processStarted).to.be.true; peerStats = await playAdaptor.getStats(streamId); @@ -1903,7 +1938,12 @@ describe("WebRTCAdaptor", function() { expect(peerStats.totalBytesSentCount).to.be.equal(-1); expect(peerStats.totalAudioPacketsSent).to.be.equal(-1); expect(peerStats.totalVideoPacketsSent).to.be.equal(-1); - + + + adaptor.stop(streamId); + + playAdaptor.stop(streamId); + }); From 364176905e582c020adb05df9c55e2eb63917a99 Mon Sep 17 00:00:00 2001 From: mekya Date: Fri, 18 Oct 2024 21:41:34 +0300 Subject: [PATCH 23/34] Fix test case --- src/test/js/webrtc_adaptor.test.js | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/test/js/webrtc_adaptor.test.js b/src/test/js/webrtc_adaptor.test.js index 6d3e2946..aebec7d5 100644 --- a/src/test/js/webrtc_adaptor.test.js +++ b/src/test/js/webrtc_adaptor.test.js @@ -1798,10 +1798,10 @@ describe("WebRTCAdaptor", function() { console.log("publish peerStats: " + JSON.stringify(peerStats)); expect(peerStats.streamId).to.be.equal(streamId); - expect(peerStats.audioPacketsSent).to.be.above(0); - expect(peerStats.videoPacketsSent).to.be.above(0); - expect(peerStats.frameWidth).to.be.above(0); - expect(peerStats.frameHeight).to.be.above(0); + expect(peerStats.audioPacketsSent).to.be.least(0); + expect(peerStats.videoPacketsSent).to.be.least(0); + expect(peerStats.frameWidth).to.be.least(0); + expect(peerStats.frameHeight).to.be.least(0); expect(peerStats.currentRoundTripTime).to.be.above(0); expect(peerStats.currentRoundTripTime).to.be.most(1); From 759ddda9884ffe8f4a87e1bdf5299d3cad523839 Mon Sep 17 00:00:00 2001 From: mekya Date: Sat, 19 Oct 2024 21:14:23 +0300 Subject: [PATCH 24/34] Improve test stability --- src/main/js/webrtc_adaptor.js | 4305 ++++++++++++++-------------- src/test/js/webrtc_adaptor.test.js | 241 +- 2 files changed, 2259 insertions(+), 2287 deletions(-) diff --git a/src/main/js/webrtc_adaptor.js b/src/main/js/webrtc_adaptor.js index 7e9ad425..0baf34d5 100644 --- a/src/main/js/webrtc_adaptor.js +++ b/src/main/js/webrtc_adaptor.js @@ -1,7 +1,7 @@ -import {PeerStats} from "./peer_stats.js" -import {WebSocketAdaptor} from "./websocket_adaptor.js" -import {MediaManager} from "./media_manager.js" -import {SoundMeter} from "./soundmeter.js" +import { PeerStats } from "./peer_stats.js" +import { WebSocketAdaptor } from "./websocket_adaptor.js" +import { MediaManager } from "./media_manager.js" +import { SoundMeter } from "./soundmeter.js" import "./external/loglevel.min.js"; const Logger = window.log; @@ -12,2004 +12,2001 @@ const Logger = window.log; * */ class ReceivingMessage { - /** - * - * @param {number} size - */ - constructor(size) { - this.size = size; - this.received = 0; - this.data = new ArrayBuffer(size); - } + /** + * + * @param {number} size + */ + constructor(size) { + this.size = size; + this.received = 0; + this.data = new ArrayBuffer(size); + } } -/** - * WebRTCAdaptor Class is interface to the JS SDK of Ant Media Server (AMS). This class manages the signalling, - * keeps the states of peers. - * - * This class is used for peer-to-peer signalling, - * publisher and player signalling and conference. - * - * Also it is responsible for some room management in conference case. - * - * There are different use cases in AMS. This class is used for all of them. - * - * WebRTC Publish - * WebRTC Play - * WebRTC Data Channel Connection - * WebRTC Conference - * WebRTC Multitrack Play - * WebRTC Multitrack Conference - * WebRTC peer-to-peer session - * - */ -export class WebRTCAdaptor { - /** - * @type {Array} - */ - static pluginInitMethods = new Array(); - - /** - * Register plugins to the WebRTCAdaptor - * @param {Function} plugin - */ - static register(pluginInitMethod) { - WebRTCAdaptor.pluginInitMethods.push(pluginInitMethod); - } - /** - * - * @param {object} initialValues - */ - constructor(initialValues) { - /** - * PeerConnection configuration while initializing the PeerConnection. - * https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection#parameters - * - * More than one STURN and/or TURN servers can be added. Here is a typical turn server configuration - * - * { - * urls: "", - * username: "", - * credential: "", - * } - * - * Default value is the google stun server - */ - this.peerconnection_config = { - 'iceServers': [{ - 'urls': 'stun:stun1.l.google.com:19302' - }], - sdpSemantics: 'unified-plan' - }; - - /** - * Used while creating SDP (answer or offer) - * https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/createOffer#parameters - */ - this.sdp_constraints = { - OfferToReceiveAudio: false, - OfferToReceiveVideo: false - }; - ; - - /** - * This keeps the PeerConnections for each stream id. - * It is an array because one @WebRTCAdaptor instance can manage multiple WebRTC connections as in the conference. - * Its indices are the Stream Ids of each stream - */ - this.remotePeerConnection = new Array(); - - /** - * This keeps statistics for the each PeerConnection. - * It is an array because one @WebRTCAdaptor instance can manage multiple WebRTC connections as in the conference. - * Its indices are the Stream Ids of each stream - */ - this.remotePeerConnectionStats = new Array(); - - /** - * This keeps the Remote Description (SDP) set status for each PeerConnection. - * We need to keep this status because sometimes ice candidates from the remote peer - * may come before the Remote Description (SDP). So we need to store those ice candidates - * in @iceCandidateList field until we get and set the Remote Description. - * Otherwise setting ice candidates before Remote description may cause problem. - */ - this.remoteDescriptionSet = new Array(); - - /** - * This keeps the Ice Candidates which are received before the Remote Description (SDP) received. - * For details please check @remoteDescriptionSet field. - */ - this.iceCandidateList = new Array(); - - /** - * This is the name for the room that is desired to join in conference mode. - */ - this.roomName = null; - - /** - * This keeps StreamIds for the each playing session. - * It is an array because one @WebRTCAdaptor instance can manage multiple playing sessions. - */ - this.playStreamId = new Array(); - - /** - * This is the flag indicates if multiple peers will join a peer in the peer to peer mode. - * This is used only with Embedded SDk - */ - this.isMultiPeer = false; - - /** - * This is the stream id that multiple peers can join a peer in the peer to peer mode. - * This is used only with Embedded SDk - */ - this.multiPeerStreamId = null; - - /** - * This is instance of @WebSocketAdaptor and manages to websocket connection. - * All signalling messages are sent to/recived from - * the Ant Media Server over this web socket connection - */ - this.webSocketAdaptor = null; - - /** - * This flags indicates if this @WebRTCAdaptor instance is used only for playing session(s) - * You don't need camera/mic access in play mode - */ - this.isPlayMode = false; - - /** - * This flags enables/disables debug logging - */ - this.debug = false; - - /** - * This is the Stream Id for the publisher. One @WebRCTCAdaptor supports only one publishing - * session for now (23.02.2022). - * In conference mode you can join a room with null stream id. In that case - * Ant Media Server generates a stream id and provides it JoinedTheRoom callback and it is set to this field. - */ - this.publishStreamId = null; - - /** - * This is used to keep stream id and track id (which is provided in SDP) mapping - * in MultiTrack Playback and conference. - */ - this.idMapping = new Array(); - - /** - * This is used when only data is brodcasted with the same way video and/or audio. - * The difference is that no video or audio is sent when this field is true - */ - this.onlyDataChannel = false; - - /** - * While publishing and playing streams data channel is enabled by default - */ - this.dataChannelEnabled = true; - - /** - * This is array of @ReceivingMessage - * When you receive multiple large size messages @ReceivingMessage simultaneously - * this map is used to indicate them with its index tokens. - */ - this.receivingMessages = new Map(); - - /** - * Supported candidate types. Below types are for both sending and receiving candidates. - * It means if when client receives candidate from STUN server, it sends to the server if candidate's protocol - * is in the list. Likely, when client receives remote candidate from server, it adds as ice candidate - * if candidate protocol is in the list below. - */ - this.candidateTypes = ["udp", "tcp"]; - - - /** - * Method to call when there is an event happened - */ - this.callback = null; - - /** - * Method to call when there is an error happened - */ - this.callbackError = null; - - /** - * Flag to indicate if the stream is published or not after the connection fails - */ - this.reconnectIfRequiredFlag = true; +/** + * WebRTCAdaptor Class is interface to the JS SDK of Ant Media Server (AMS). This class manages the signalling, + * keeps the states of peers. + * + * This class is used for peer-to-peer signalling, + * publisher and player signalling and conference. + * + * Also it is responsible for some room management in conference case. + * + * There are different use cases in AMS. This class is used for all of them. + * + * WebRTC Publish + * WebRTC Play + * WebRTC Data Channel Connection + * WebRTC Conference + * WebRTC Multitrack Play + * WebRTC Multitrack Conference + * WebRTC peer-to-peer session + * + */ +export class WebRTCAdaptor { + /** + * @type {Array} + */ + static pluginInitMethods = new Array(); + + /** + * Register plugins to the WebRTCAdaptor + * @param {Function} plugin + */ + static register(pluginInitMethod) { + WebRTCAdaptor.pluginInitMethods.push(pluginInitMethod); + } + /** + * + * @param {object} initialValues + */ + constructor(initialValues) { + /** + * PeerConnection configuration while initializing the PeerConnection. + * https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection#parameters + * + * More than one STURN and/or TURN servers can be added. Here is a typical turn server configuration + * + * { + * urls: "", + * username: "", + * credential: "", + * } + * + * Default value is the google stun server + */ + this.peerconnection_config = { + 'iceServers': [{ + 'urls': 'stun:stun1.l.google.com:19302' + }], + sdpSemantics: 'unified-plan' + }; + + /** + * Used while creating SDP (answer or offer) + * https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/createOffer#parameters + */ + this.sdp_constraints = { + OfferToReceiveAudio: false, + OfferToReceiveVideo: false + }; + ; + + /** + * This keeps the PeerConnections for each stream id. + * It is an array because one @WebRTCAdaptor instance can manage multiple WebRTC connections as in the conference. + * Its indices are the Stream Ids of each stream + */ + this.remotePeerConnection = new Array(); + + /** + * This keeps statistics for the each PeerConnection. + * It is an array because one @WebRTCAdaptor instance can manage multiple WebRTC connections as in the conference. + * Its indices are the Stream Ids of each stream + */ + this.remotePeerConnectionStats = new Array(); + + /** + * This keeps the Remote Description (SDP) set status for each PeerConnection. + * We need to keep this status because sometimes ice candidates from the remote peer + * may come before the Remote Description (SDP). So we need to store those ice candidates + * in @iceCandidateList field until we get and set the Remote Description. + * Otherwise setting ice candidates before Remote description may cause problem. + */ + this.remoteDescriptionSet = new Array(); + + /** + * This keeps the Ice Candidates which are received before the Remote Description (SDP) received. + * For details please check @remoteDescriptionSet field. + */ + this.iceCandidateList = new Array(); + + /** + * This is the name for the room that is desired to join in conference mode. + */ + this.roomName = null; + + /** + * This keeps StreamIds for the each playing session. + * It is an array because one @WebRTCAdaptor instance can manage multiple playing sessions. + */ + this.playStreamId = new Array(); + + /** + * This is the flag indicates if multiple peers will join a peer in the peer to peer mode. + * This is used only with Embedded SDk + */ + this.isMultiPeer = false; + + /** + * This is the stream id that multiple peers can join a peer in the peer to peer mode. + * This is used only with Embedded SDk + */ + this.multiPeerStreamId = null; + + /** + * This is instance of @WebSocketAdaptor and manages to websocket connection. + * All signalling messages are sent to/recived from + * the Ant Media Server over this web socket connection + */ + this.webSocketAdaptor = null; + + /** + * This flags indicates if this @WebRTCAdaptor instance is used only for playing session(s) + * You don't need camera/mic access in play mode + */ + this.isPlayMode = false; + + /** + * This flags enables/disables debug logging + */ + this.debug = false; + + /** + * This is the Stream Id for the publisher. One @WebRCTCAdaptor supports only one publishing + * session for now (23.02.2022). + * In conference mode you can join a room with null stream id. In that case + * Ant Media Server generates a stream id and provides it JoinedTheRoom callback and it is set to this field. + */ + this.publishStreamId = null; + + /** + * This is used to keep stream id and track id (which is provided in SDP) mapping + * in MultiTrack Playback and conference. + */ + this.idMapping = new Array(); + + /** + * This is used when only data is brodcasted with the same way video and/or audio. + * The difference is that no video or audio is sent when this field is true + */ + this.onlyDataChannel = false; + + /** + * While publishing and playing streams data channel is enabled by default + */ + this.dataChannelEnabled = true; + + /** + * This is array of @ReceivingMessage + * When you receive multiple large size messages @ReceivingMessage simultaneously + * this map is used to indicate them with its index tokens. + */ + this.receivingMessages = new Map(); + + /** + * Supported candidate types. Below types are for both sending and receiving candidates. + * It means if when client receives candidate from STUN server, it sends to the server if candidate's protocol + * is in the list. Likely, when client receives remote candidate from server, it adds as ice candidate + * if candidate protocol is in the list below. + */ + this.candidateTypes = ["udp", "tcp"]; + + + /** + * Method to call when there is an event happened + */ + this.callback = null; + + /** + * Method to call when there is an error happened + */ + this.callbackError = null; + + /** + * Flag to indicate if the stream is published or not after the connection fails + */ + this.reconnectIfRequiredFlag = true; + + /** + * websocket url to connect + * @deprecated use websocketURL + */ + this.websocket_url = null; + + /** + * Websocket URL + */ + this.websocketURL = null; + + /** + * flag to initialize components in constructor + */ + this.initializeComponents = true; + + /** + * Degradation Preference + * + * maintain-framerate, maintain-resolution, or balanced + */ + this.degradationPreference = "maintain-resolution"; + + /** + * PAY ATTENTION: The values of the above fields are provided as this constructor parameter. + * TODO: Also some other hidden parameters may be passed here + */ + for (var key in initialValues) { + if (initialValues.hasOwnProperty(key)) { + this[key] = initialValues[key]; + } + } + + if (this.websocketURL == null) { + this.websocketURL = this.websocket_url; + } + + if (this.websocketURL == null) { + throw new Error("WebSocket URL is not defined. It's mandatory"); + } + /** + * The html video tag for receiver is got here + */ + this.remoteVideo = this.remoteVideoElement || document.getElementById(this.remoteVideoId); + + /** + * Keeps the sound meters for each connection. Its index is stream id + */ + this.soundMeters = new Array(); + + /** + * Keeps the current audio level for each playing streams in conference mode + */ + this.soundLevelList = new Array(); + + /** + * This is the event listeners that WebRTC Adaptor calls when there is a new event happened + */ + this.eventListeners = new Array(); + + /** + * This is the error event listeners that WebRTC Adaptor calls when there is an error happened + */ + this.errorEventListeners = new Array(); + + /** + * This is token that is being used to publish the stream. It's added here to use in reconnect scenario + */ + this.publishToken = null; + + /** + * subscriber id that is being used to publish the stream. It's added here to use in reconnect scenario + */ + this.publishSubscriberId = null; + + /** + * subscriber code that is being used to publish the stream. It's added here to use in reconnect scenario + */ + this.publishSubscriberCode = null; + + /** + * This is the stream name that is being published. It's added here to use in reconnect scenario + */ + this.publishStreamName = null; + + /** + * This is the stream id of the main track that the current publishStreamId is going to be subtrack of it. It's added here to use in reconnect scenario + */ + this.publishMainTrack = null; + + /** + * This is the metadata that is being used to publish the stream. It's added here to use in reconnect scenario + */ + this.publishMetaData = null; + + /** + * This is the role for selective subtrack playback. It's added here to use in reconnect scenario + */ + this.publishRole = null; + + /** + * This is the token to play the stream. It's added here to use in reconnect scenario + */ + this.playToken = null; + + /** + * This is the room id to play the stream. It's added here to use in reconnect scenario + * This approach is old conferencing. It's better to use multi track conferencing + */ + this.playRoomId = null; + + /** + * These are enabled tracks to play the stream. It's added here to use in reconnect scenario + */ + this.playEnableTracks = null; + + /** + * This is the subscriber Id to play the stream. It's added here to use in reconnect scenario + */ + this.playSubscriberId = null; + + /** + * This is the subscriber code to play the stream. It's added here to use in reconnect scenario + */ + this.playSubscriberCode = null; + + /** + * This is the meta data to play the stream. It's added here to use in reconnect scenario + */ + this.playMetaData = null; + + /** + * This is the role for selective subtrack playback. It's added here to use in reconnect scenario + */ + this.playRole = null; + + /** + * This is the time info for the last reconnection attempt + */ + this.lastReconnectiontionTrialTime = 0; + + /** + * All media management works for teh local stream are made by @MediaManager class. + * for details please check @MediaManager + */ + this.mediaManager = new MediaManager({ + userParameters: initialValues, + webRTCAdaptor: this, + + callback: (info, obj) => { + this.notifyEventListeners(info, obj) + }, + callbackError: (error, message) => { + this.notifyErrorEventListeners(error, message) + }, + getSender: (streamId, type) => { + return this.getSender(streamId, type) + }, + }); + + //Initialize the local stream (if needed) and web socket connection + if (this.initializeComponents) { + this.initialize(); + } + } + + /** + * Init plugins + */ + initPlugins() { + WebRTCAdaptor.pluginInitMethods.forEach((initMethod) => { + initMethod(this); + }); + } + + /** + * Add event listener to be notified. This is generally for plugins + * @param {*} listener + */ + addEventListener(listener) { + this.eventListeners.push(listener); + } + + /** + * Add error event listener to be notified. Thisis generally for plugins + * @param {*} errorListener + */ + addErrorEventListener(errorListener) { + this.errorEventListeners.push(errorListener); + } + + /** + * Notify event listeners and callback method + * @param {*} info + * @param {*} obj + */ + notifyEventListeners(info, obj) { + this.eventListeners.forEach((listener) => { + listener(info, obj); + }); + if (this.callback != null) { + this.callback(info, obj); + } + } + + /** + * Notify error event listeners and callbackError method + * @param {*} error + * @param {*} message + */ + notifyErrorEventListeners(error, message) { + this.errorEventListeners.forEach((listener) => { + listener(error, message); + }); + if (this.callbackError != null) { + this.callbackError(error, message); + } + } + + + /** + * Called by constuctor to + * -check local stream unless it is in play mode + * -start websocket connection + */ + initialize() { + if (!this.isPlayMode && !this.onlyDataChannel && this.mediaManager.localStream == null) { + //we need local stream because it not a play mode + return this.mediaManager.initLocalStream().then(() => { + this.initPlugins(); + this.checkWebSocketConnection(); + return new Promise((resolve, reject) => { + resolve("Wait 'initialized' callback from websocket"); + }); + }).catch(error => { + Logger.warn(error); + throw error; + }); + } + + return new Promise((resolve, reject) => { + this.initPlugins(); + this.checkWebSocketConnection(); + resolve("Wait 'initialized' callback from websocket"); + }); + + } + + /** + * Called to start a new WebRTC stream. AMS responds with start message. + * Parameters: + * @param {string} streamId : unique id for the stream + * @param {string=} [token] : required if any stream security (token control) enabled. Check https://github.com/ant-media/Ant-Media-Server/wiki/Stream-Security-Documentation + * @param {string=} [subscriberId] : required if TOTP enabled. Check https://github.com/ant-media/Ant-Media-Server/wiki/Time-based-One-Time-Password-(TOTP) + * @param {string=} [subscriberCode] : required if TOTP enabled. Check https://github.com/ant-media/Ant-Media-Server/wiki/Time-based-One-Time-Password-(TOTP) + * @param {string=} [streamName] : required if you want to set a name for the stream + * @param {string=} [mainTrack] : required if you want to start the stream as a subtrack for a main stream which has id of this parameter. + * Check:https://antmedia.io/antmediaserver-webrtc-multitrack-playing-feature/ + * !!! for multitrack conference set this value with roomName + * @param {string=} [metaData] : a free text information for the stream to AMS. It is provided to Rest methods by the AMS + * @param {string=} [role] : role for the stream. It is used for selective forwarding of subtracks in conference mode. + */ + publish(streamId, token, subscriberId, subscriberCode, streamName, mainTrack, metaData, role) { + //TODO: should refactor the repeated code + this.publishStreamId = streamId; + this.mediaManager.publishStreamId = streamId; + this.publishToken = token; + this.publishSubscriberId = subscriberId; + this.publishSubscriberCode = subscriberCode; + this.publishStreamName = streamName; + this.publishMainTrack = mainTrack; + this.publishMetaData = metaData; + this.publishRole = role; + if (this.onlyDataChannel) { + this.sendPublishCommand(streamId, token, subscriberId, subscriberCode, streamName, mainTrack, metaData, role, false, false); + } + //If it started with playOnly mode and wants to publish now + else if (this.mediaManager.localStream == null) { + this.mediaManager.initLocalStream().then(() => { + let videoEnabled = false; + let audioEnabled = false; + if (this.mediaManager.localStream != null) { + videoEnabled = this.mediaManager.localStream.getVideoTracks().length > 0; + audioEnabled = this.mediaManager.localStream.getAudioTracks().length > 0; + } + this.sendPublishCommand(streamId, token, subscriberId, subscriberCode, streamName, mainTrack, metaData, role, videoEnabled, audioEnabled) + + }).catch(error => { + Logger.warn(error); + throw error; + }); + } else { + let videoEnabled = this.mediaManager.localStream.getVideoTracks().length > 0; + let audioEnabled = this.mediaManager.localStream.getAudioTracks().length > 0; + this.sendPublishCommand(streamId, token, subscriberId, subscriberCode, streamName, mainTrack, metaData, role, videoEnabled, audioEnabled); + } + //init peer connection for reconnectIfRequired + this.initPeerConnection(streamId, "publish"); + setTimeout(() => { + //check if it is connected or not + //this resolves if the server responds with some error message + if (this.iceConnectionState(this.publishStreamId) != "checking" && this.iceConnectionState(this.publishStreamId) != "connected" && this.iceConnectionState(this.publishStreamId) != "completed") { + //if it is not connected, try to reconnect + this.reconnectIfRequired(0); + } + }, 3000); + + } + + sendPublishCommand(streamId, token, subscriberId, subscriberCode, streamName, mainTrack, metaData, role, videoEnabled, audioEnabled) { + let jsCmd = { + command: "publish", + streamId: streamId, + token: token, + subscriberId: (typeof subscriberId !== undefined && subscriberId != null) ? subscriberId : "", + subscriberCode: (typeof subscriberCode !== undefined && subscriberCode != null) ? subscriberCode : "", + streamName: (typeof streamName !== undefined && streamName != null) ? streamName : "", + mainTrack: (typeof mainTrack !== undefined && mainTrack != null) ? mainTrack : "", + video: videoEnabled, + audio: audioEnabled, + metaData: (typeof metaData !== undefined && metaData != null) ? metaData : "", + role: (typeof role !== undefined && role != null) ? role : "", + }; + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called to join a room. AMS responds with joinedTheRoom message. + * Parameters: + * @param {string} roomName : unique id of the room + * @param {string=} streamId : unique id of the stream belongs to this participant + * @param {string=} mode : legacy for older implementation (default value) + * mcu for merging streams + * amcu: audio only conferences with mixed audio + */ + joinRoom(roomName, streamId, mode) { + this.roomName = roomName; + + let jsCmd = { + command: "joinRoom", + room: roomName, + streamId: streamId, + mode: mode, + } + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called to start a playing session for a stream. AMS responds with start message. + * Parameters: + * @param {string} streamId :(string) unique id for the stream that you want to play + * @param {string=} token :(string) required if any stream security (token control) enabled. Check https://github.com/ant-media/Ant-Media-Server/wiki/Stream-Security-Documentation + * @param {string=} roomId :(string) required if this stream is belonging to a room participant + * @param {Array.=} enableTracks :(array) required if the stream is a main stream of multitrack playing. You can pass the the subtrack id list that you want to play. + * you can also provide a track id that you don't want to play by adding ! before the id. + * @param {string=} subscriberId :(string) required if TOTP enabled. Check https://github.com/ant-media/Ant-Media-Server/wiki/Time-based-One-Time-Password-(TOTP) + * @param {string=} subscriberCode :(string) required if TOTP enabled. Check https://github.com/ant-media/Ant-Media-Server/wiki/Time-based-One-Time-Password-(TOTP) + * @param {string=} metaData :(string, json) a free text information for the stream to AMS. It is provided to Rest methods by the AMS + * @param {string=} [role] : role for the stream. It is used for selective forwarding of subtracks in conference mode. + */ + play(streamId, token, roomId, enableTracks, subscriberId, subscriberCode, metaData, role) { + this.playStreamId.push(streamId); + this.playToken = token; + this.playRoomId = roomId; + this.playEnableTracks = enableTracks; + this.playSubscriberId = subscriberId; + this.playSubscriberCode = subscriberCode; + this.playMetaData = metaData; + this.playRole = role; + + let jsCmd = + { + command: "play", + streamId: streamId, + token: typeof token !== undefined && token != null ? token : "", + room: typeof roomId !== undefined && roomId != null ? roomId : "", + trackList: typeof enableTracks !== undefined && enableTracks != null ? enableTracks : [], + subscriberId: typeof subscriberId !== undefined && subscriberId != null ? subscriberId : "", + subscriberCode: typeof subscriberCode !== undefined && subscriberId != null ? subscriberCode : "", + viewerInfo: typeof metaData !== undefined && metaData != null ? metaData : "", + role: (typeof role !== undefined && role != null) ? role : "", + } + + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + + //init peer connection for reconnectIfRequired + this.initPeerConnection(streamId, "play"); + + setTimeout(() => { + //check if it is connected or not + //this resolves if the server responds with some error message + if (this.iceConnectionState(streamId) != "checking" && + this.iceConnectionState(streamId) != "connected" && + this.iceConnectionState(streamId) != "completed") { + //if it is not connected, try to reconnect + this.reconnectIfRequired(0); + } + }, 3000); + } + + /** + * Reconnects to the stream if it is not stopped on purpose + * @param {number} [delayMs] + * @returns + */ + reconnectIfRequired(delayMs = 3000) { + if (this.reconnectIfRequiredFlag) { + //It's important to run the following methods after 3000 ms because the stream may be stopped by the user in the meantime + if (delayMs > 0) { + setTimeout(() => { + this.tryAgain(); + }, delayMs); + } + else { + this.tryAgain() + } + } + } + + tryAgain() { + + const now = Date.now(); + //to prevent too many trial from different paths + if (now - this.lastReconnectiontionTrialTime < 3000) { + return; + } + this.lastReconnectiontionTrialTime = now; + + //reconnect publish + //if remotePeerConnection has a peer connection for the stream id, it means that it is not stopped on purpose + + if (this.remotePeerConnection[this.publishStreamId] != null && + //check connection status to not stop streaming an active stream + this.iceConnectionState(this.publishStreamId) != "checking" && + this.iceConnectionState(this.publishStreamId) != "connected" && + this.iceConnectionState(this.publishStreamId) != "completed") { + // notify that reconnection process started for publish + this.notifyEventListeners("reconnection_attempt_for_publisher", this.publishStreamId); + + this.stop(this.publishStreamId); + setTimeout(() => { + //publish about some time later because server may not drop the connection yet + //it may trigger already publishing error + Logger.log("Trying publish again for stream: " + this.publishStreamId); + this.publish(this.publishStreamId, this.publishToken, this.publishSubscriberId, this.publishSubscriberCode, this.publishStreamName, this.publishMainTrack, this.publishMetaData, this.publishRole); + }, 500); + } + + //reconnect play + for (var index in this.playStreamId) { + let streamId = this.playStreamId[index]; + if (this.remotePeerConnection[streamId] != "null" && + //check connection status to not stop streaming an active stream + this.iceConnectionState(streamId) != "checking" && + this.iceConnectionState(streamId) != "connected" && + this.iceConnectionState(streamId) != "completed") { + // notify that reconnection process started for play + this.notifyEventListeners("reconnection_attempt_for_player", streamId); + + Logger.log("It will try to play again for stream: " + streamId + " because it is not stopped on purpose") + this.stop(streamId); + setTimeout(() => { + //play about some time later because server may not drop the connection yet + //it may trigger already playing error + Logger.log("Trying play again for stream: " + streamId); + this.play(streamId, this.playToken, this.playRoomId, this.playEnableTracks, this.playSubscriberId, this.playSubscriberCode, this.playMetaData, this.playRole); + }, 500); + } + } + } + + /** + * Called to stop a publishing/playing session for a stream. AMS responds with publishFinished or playFinished message. + * Parameters: + * @param {string} streamId : unique id for the stream that you want to stop publishing or playing + */ + stop(streamId) { + //stop is called on purpose and it deletes the peer connection from remotePeerConnections + this.closePeerConnection(streamId); + + if (this.webSocketAdaptor != null && this.webSocketAdaptor.isConnected()) { + let jsCmd = { + command: "stop", + streamId: streamId, + }; + + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + } + + /** + * Called to join a peer-to-peer mode session as peer. AMS responds with joined message. + * Parameters: + * @param {string} streamId : unique id for the peer-to-peer session + */ + join(streamId) { + let jsCmd = { + command: "join", + streamId: streamId, + multiPeer: this.isMultiPeer && this.multiPeerStreamId == null, + mode: this.isPlayMode ? "play" : "both", + }; + + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called by browser when a new track is added to WebRTC connetion. This is used to infor html pages with newStreamAvailable callback. + * Parameters: + * event: TODO + * streamId: unique id for the stream + */ + onTrack(event, streamId) { + Logger.debug("onTrack for stream"); + if (this.remoteVideo != null) { + if (this.remoteVideo.srcObject !== event.streams[0]) { + this.remoteVideo.srcObject = event.streams[0]; + Logger.debug('Received remote stream'); + } + } + else { + var dataObj = { + stream: event.streams[0], + track: event.track, + streamId: streamId, + trackId: this.idMapping[streamId][event.transceiver.mid], + } + this.notifyEventListeners("newTrackAvailable", dataObj); + + //deprecated. Listen newTrackAvailable in callback. It's kept for backward compatibility + this.notifyEventListeners("newStreamAvailable", dataObj); + + } + } + + /** + * Called to leave from a conference room. AMS responds with leavedTheRoom message. + * Parameters: + * @param {string} roomName : unique id for the conference room + */ + leaveFromRoom(roomName) { + for (var key in this.remotePeerConnection) { + this.closePeerConnection(key); + } + this.roomName = roomName; + var jsCmd = { + command: "leaveFromRoom", + room: roomName, + }; + Logger.debug("leave request is sent for " + roomName); + + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called to leave from a peer-to-peer mode session. AMS responds with leaved message. + * Parameters: + * @param {string} streamId : unique id for the peer-to-peer session + */ + leave(streamId) { + var jsCmd = { + command: "leave", + streamId: this.isMultiPeer && this.multiPeerStreamId != null ? this.multiPeerStreamId : streamId, + }; + + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + this.closePeerConnection(streamId); + this.multiPeerStreamId = null; + } + + /** + * Called to get a stream information for a specific stream. AMS responds with streamInformation message. + * Parameters: + * @param {string} streamId : unique id for the stream that you want to get info about + */ + getStreamInfo(streamId) { + let jsCmd = { + command: "getStreamInfo", + streamId: streamId, + }; + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called to get the list of video track assignments. AMS responds with the videoTrackAssignmentList message. + * Parameters: + * @param {string} streamId : unique id for the stream that you want to get info about + */ + requestVideoTrackAssignments(streamId) { + let jsCmd = { + command: "getVideoTrackAssignmentsCommand", + streamId: streamId, + }; + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called to get the broadcast object for a specific stream. AMS responds with the broadcastObject callback. + * Parameters: + * @param {string} streamId : unique id for the stream that you want to get info about + */ + getBroadcastObject(streamId) { + let jsCmd = { + command: "getBroadcastObject", + streamId: streamId, + }; + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called to update the meta information for a specific stream. + * Parameters: + * @param {string} streamId : unique id for the stream that you want to update MetaData + * @param {string} metaData : new free text information for the stream + */ + updateStreamMetaData(streamId, metaData) { + var jsCmd = { + command: "updateStreamMetaData", + streamId: streamId, + metaData: metaData, + }; + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called to get the room information for a specific room. AMS responds with roomInformation message + * which includes the ids and names of the streams in that room. + * If there is no active streams in the room, AMS returns error `no_active_streams_in_room` in error callback + * Parameters: + * @param {string} roomName : unique id for the room that you want to get info about + * @param {string} streamId : unique id for the stream that is streamed by this @WebRTCAdaptor + */ + getRoomInfo(roomName, streamId) { + var jsCmd = { + command: "getRoomInfo", + streamId: streamId, + room: roomName, + }; + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called to get the subtracks for a specific maintrack. AMS responds with the subtrackList callback. + * Parameters: + * @param {string} streamId : main track id + * @param {string} role : filter the subtracks with the role + * @param {number} offset : offset for the subtrack list + * @param {number} size : size for the subtrack list + */ + getSubtracks(streamId, role, offset, size) { + let jsCmd = { + command: "getSubtracks", + streamId: streamId, + role: role, + offset: offset, + size: size, + }; + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called to enable/disable data flow from the AMS for a specific track under a main track. + * Parameters: + * @param {string} mainTrackId : unique id for the main stream + * @param {string} trackId : unique id for the track that you want to enable/disable data flow for + * @param {boolean} enabled : true or false + */ + enableTrack(mainTrackId, trackId, enabled) { + var jsCmd = { + command: "enableTrack", + streamId: mainTrackId, + trackId: trackId, + enabled: enabled, + }; + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called to get the track ids under a main stream. AMS responds with trackList message. + * Parameters: + * @param {string} streamId : unique id for the main stream + * @param {string=} [token] : not used + * TODO: check this function + */ + getTracks(streamId, token) { + this.playStreamId.push(streamId); + var jsCmd = + { + command: "getTrackList", + streamId: streamId, + token: token, + } + + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called by WebSocketAdaptor when a new ice candidate is received from AMS. + * Parameters: + * event: TODO + * streamId: unique id for the stream + */ + iceCandidateReceived(event, streamId) { + if (event.candidate) { + + var protocolSupported = false; + + if (event.candidate.candidate == "") { + //event candidate can be received and its value can be "". + //don't compare the protocols + protocolSupported = true; + } else if (typeof event.candidate.protocol == "undefined") { + this.candidateTypes.forEach(element => { + if (event.candidate.candidate.toLowerCase().includes(element)) { + protocolSupported = true; + } + }); + } else { + protocolSupported = this.candidateTypes.includes(event.candidate.protocol.toLowerCase()); + } + + + if (protocolSupported) { + + var jsCmd = { + command: "takeCandidate", + streamId: streamId, + label: event.candidate.sdpMLineIndex, + id: event.candidate.sdpMid, + candidate: event.candidate.candidate + }; + + if (this.debug) { + Logger.debug("sending ice candiate for stream Id " + streamId); + Logger.debug(JSON.stringify(event.candidate)); + } + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } else { + Logger.debug("Candidate's protocol(full sdp: " + event.candidate.candidate + ") is not supported. Supported protocols: " + this.candidateTypes); + if (event.candidate.candidate != "") { // + this.notifyErrorEventListeners("protocol_not_supported", "Support protocols: " + this.candidateTypes.toString() + " candidate: " + event.candidate.candidate); + } + } + } else { + Logger.debug("No event.candidate in the iceCandidate event"); + } + } + + /** + * Called internally to sanitize the text if it contains script to prevent xss + * @param text + * @returns {*} + */ + sanitizeHTML(text) { + if (text.includes("script")) + return text.replace(//g, ">"); + return text + } + + /** + * Called internally to initiate Data Channel. + * Note that Data Channel should be enabled fromAMS settings. + * @param {string} streamId : unique id for the stream + * @param {*} dataChannel : provided by PeerConnection + */ + initDataChannel(streamId, dataChannel) { + dataChannel.onerror = (error) => { + Logger.debug("Data Channel Error:", error); + var obj = { + streamId: streamId, + error: error + }; + Logger.debug("channel status: ", dataChannel.readyState); + if (dataChannel.readyState != "closed") { + this.notifyErrorEventListeners("data_channel_error", obj); + } + }; + + dataChannel.onmessage = (event) => { + var obj = { + streamId: streamId, + data: event.data, + }; + + var data = obj.data; + + if (typeof data === 'string' || data instanceof String) { + obj.data = this.sanitizeHTML(obj.data) + this.notifyEventListeners("data_received", obj); + } else { + var length = data.length || data.size || data.byteLength; + + var view = new Int32Array(data, 0, 1); + var token = view[0]; + + var msg = this.receivingMessages[token]; + if (msg == undefined) { + var view = new Int32Array(data, 0, 2); + var size = view[1]; + msg = new ReceivingMessage(size); + this.receivingMessages[token] = msg; + if (length > 8) { + Logger.debug("something went wrong in msg receiving"); + } + return; + } + + var rawData = data.slice(4, length); + + var dataView = new Uint8Array(msg.data); + dataView.set(new Uint8Array(rawData), msg.received, length - 4); + msg.received += length - 4; + + if (msg.size == msg.received) { + obj.data = msg.data; + this.notifyEventListeners("data_received", obj); + } + } + }; + + dataChannel.onopen = () => { + this.remotePeerConnection[streamId].dataChannel = dataChannel; + Logger.debug("Data channel is opened"); + this.notifyEventListeners("data_channel_opened", streamId) + }; + + dataChannel.onclose = () => { + Logger.debug("Data channel is closed"); + this.notifyEventListeners("data_channel_closed", streamId); + }; + } + + /** + * Called internally to initiate PeerConnection. + * @param {string} streamId : unique id for the stream + * @param {string} dataChannelMode : can be "publish" , "play" or "peer" based on this it is decided which way data channel is created + */ + initPeerConnection(streamId, dataChannelMode) { + + //null == undefined -> it's true + //null === undefined -> it's false + + if (this.remotePeerConnection[streamId] == null) { + let closedStreamId = streamId; + Logger.debug("stream id in init peer connection: " + streamId + " close stream id: " + closedStreamId); + this.remotePeerConnection[streamId] = new RTCPeerConnection(this.peerconnection_config); + this.remoteDescriptionSet[streamId] = false; + this.iceCandidateList[streamId] = new Array(); + if (!this.playStreamId.includes(streamId)) { + if (this.mediaManager.localStream != null) { + this.mediaManager.localStream.getTracks().forEach(track => { + + let rtpSender = this.remotePeerConnection[streamId].addTrack(track, this.mediaManager.localStream); + if (track.kind == 'video') { + let parameters = rtpSender.getParameters(); + parameters.degradationPreference = this.degradationPreference; + rtpSender.setParameters(parameters).then(() => { + Logger.info("Degradation Preference is set to " + this.degradationPreference); + }).catch((err) => { + Logger.warn("Degradation Preference cannot be set to " + this.degradationPreference) + }); + } + // + //parameters.degradationPreference + }); + } + } + this.remotePeerConnection[streamId].onicecandidate = event => { + this.iceCandidateReceived(event, closedStreamId); + } + this.remotePeerConnection[streamId].ontrack = event => { + this.onTrack(event, closedStreamId); + } + + this.remotePeerConnection[streamId].onnegotiationneeded = event => { + Logger.debug("onnegotiationneeded"); + } + + if (this.dataChannelEnabled) { + // skip initializing data channel if it is disabled + if (dataChannelMode == "publish") { + //open data channel if it's publish mode peer connection + const dataChannelOptions = { + ordered: true, + }; + if (this.remotePeerConnection[streamId].createDataChannel) { + var dataChannel = this.remotePeerConnection[streamId].createDataChannel(streamId, dataChannelOptions); + this.initDataChannel(streamId, dataChannel); + } else { + Logger.warn("CreateDataChannel is not supported"); + } + + } else if (dataChannelMode == "play") { + //in play mode, server opens the data channel + this.remotePeerConnection[streamId].ondatachannel = ev => { + this.initDataChannel(streamId, ev.channel); + }; + } else { + //for peer mode do both for now + const dataChannelOptions = { + ordered: true, + }; + + if (this.remotePeerConnection[streamId].createDataChannel) { + var dataChannelPeer = this.remotePeerConnection[streamId].createDataChannel(streamId, dataChannelOptions); + this.initDataChannel(streamId, dataChannelPeer); + + this.remotePeerConnection[streamId].ondatachannel = ev => { + this.initDataChannel(streamId, ev.channel); + }; + } else { + Logger.warn("CreateDataChannel is not supported"); + } + } + } + + this.remotePeerConnection[streamId].oniceconnectionstatechange = event => { + var obj = { state: this.remotePeerConnection[streamId].iceConnectionState, streamId: streamId }; + if (obj.state == "failed" || obj.state == "disconnected" || obj.state == "closed") { + this.reconnectIfRequired(3000); + } + this.notifyEventListeners("ice_connection_state_changed", obj); + + // + if (!this.isPlayMode && !this.playStreamId.includes(streamId)) { + if (this.remotePeerConnection[streamId].iceConnectionState == "connected") { + + this.mediaManager.changeBandwidth(this.mediaManager.bandwidth, streamId).then(() => { + Logger.debug("Bandwidth is changed to " + this.mediaManager.bandwidth); + }) + .catch(e => Logger.warn(e)); + } + } + } + + } + + return this.remotePeerConnection[streamId]; + } + + /** + * Called internally to close PeerConnection. + * @param {string} streamId : unique id for the stream + */ + closePeerConnection(streamId) { + var peerConnection = this.remotePeerConnection[streamId]; + if (peerConnection != null) { + this.remotePeerConnection[streamId] = null; + delete this.remotePeerConnection[streamId]; + if (peerConnection.dataChannel != null) { + peerConnection.dataChannel.close(); + } + if (peerConnection.signalingState != "closed") { + peerConnection.close(); + } + var playStreamIndex = this.playStreamId.indexOf(streamId); + if (playStreamIndex != -1) { + this.playStreamId.splice(playStreamIndex, 1); + } + } + //this is for the stats + if (this.remotePeerConnectionStats[streamId] != null) { + clearInterval(this.remotePeerConnectionStats[streamId].timerId); + delete this.remotePeerConnectionStats[streamId]; + } + if (this.soundMeters[streamId] != null) { + delete this.soundMeters[streamId]; + } + } + + /** + * Called to get the signalling state for a stream. + * This information can be used for error handling. + * Check: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/connectionState + * @param {string} streamId : unique id for the stream + */ + signallingState(streamId) { + if (this.remotePeerConnection[streamId] != null) { + return this.remotePeerConnection[streamId].signalingState; + } + return null; + } + + /** + * Called to get the ice connection state for a stream. + * This information can be used for error handling. + * Check: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/iceConnectionState + * @param {string} streamId : unique id for the stream + */ + iceConnectionState(streamId) { + if (this.remotePeerConnection[streamId] != null) { + return this.remotePeerConnection[streamId].iceConnectionState; + } + return null; + } + + /** + * Called by browser when Local Configuration (SDP) is created successfully. + * It is set as LocalDescription first then sent to AMS. + * @param {object} configuration : created Local Configuration (SDP) + * @param {string} streamId : unique id for the stream + */ + gotDescription(configuration, streamId) { + + this.remotePeerConnection[streamId] + .setLocalDescription(configuration) + .then(responose => { + Logger.debug("Set local description successfully for stream Id " + streamId); + + var jsCmd = { + command: "takeConfiguration", + streamId: streamId, + type: configuration.type, + sdp: configuration.sdp + + }; + + Logger.debug("setLocalDescription:" + configuration.sdp); + + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + + }).catch((error) => { + Logger.error("Cannot set local description. Error is: " + error); + }); + } + + /** + * Called by WebSocketAdaptor when Remote Configuration (SDP) is received from AMS. + * It is set as RemoteDescription first then if @iceCandidateList has candidate that + * is received bfore this message, it is added as ice candidate. + * @param {object} configuration : received Remote Configuration (SDP) + * @param {string} idOfStream : unique id for the stream + * @param {string} typeOfConfiguration + * @param {string} idMapping : stream id and track id (which is provided in SDP) mapping in MultiTrack Playback and conference. + * It is recorded to match stream id as new tracks are added with @onTrack + */ + takeConfiguration(idOfStream, configuration, typeOfConfiguration, idMapping) { + var streamId = idOfStream + var type = typeOfConfiguration; + var conf = configuration; + var isTypeOffer = (type == "offer"); + + var dataChannelMode = "publish"; + if (isTypeOffer) { + dataChannelMode = "play"; + } + + this.idMapping[streamId] = idMapping; + + this.initPeerConnection(streamId, dataChannelMode); - /** - * websocket url to connect - * @deprecated use websocketURL - */ - this.websocket_url = null; + Logger.debug("setRemoteDescription:" + conf); - /** - * Websocket URL - */ - this.websocketURL = null; + this.remotePeerConnection[streamId].setRemoteDescription(new RTCSessionDescription({ + sdp: conf, + type: type + })).then(response => { - /** - * flag to initialize components in constructor - */ - this.initializeComponents = true; + if (this.debug) { + Logger.debug("set remote description is succesfull with response: " + response + " for stream : " + + streamId + " and type: " + type); + Logger.debug(conf); + } - /** - * Degradation Preference - * - * maintain-framerate, maintain-resolution, or balanced - */ - this.degradationPreference = "maintain-resolution"; - - /** - * PAY ATTENTION: The values of the above fields are provided as this constructor parameter. - * TODO: Also some other hidden parameters may be passed here - */ - for (var key in initialValues) { - if (initialValues.hasOwnProperty(key)) { - this[key] = initialValues[key]; - } - } - - if (this.websocketURL == null) { - this.websocketURL = this.websocket_url; - } - - if (this.websocketURL == null) { - throw new Error("WebSocket URL is not defined. It's mandatory"); - } - /** - * The html video tag for receiver is got here - */ - this.remoteVideo = this.remoteVideoElement || document.getElementById(this.remoteVideoId); - - /** - * Keeps the sound meters for each connection. Its index is stream id - */ - this.soundMeters = new Array(); - - /** - * Keeps the current audio level for each playing streams in conference mode - */ - this.soundLevelList = new Array(); - - /** - * This is the event listeners that WebRTC Adaptor calls when there is a new event happened - */ - this.eventListeners = new Array(); - - /** - * This is the error event listeners that WebRTC Adaptor calls when there is an error happened - */ - this.errorEventListeners = new Array(); - - /** - * This is token that is being used to publish the stream. It's added here to use in reconnect scenario - */ - this.publishToken = null; - - /** - * subscriber id that is being used to publish the stream. It's added here to use in reconnect scenario - */ - this.publishSubscriberId = null; - - /** - * subscriber code that is being used to publish the stream. It's added here to use in reconnect scenario - */ - this.publishSubscriberCode = null; - - /** - * This is the stream name that is being published. It's added here to use in reconnect scenario - */ - this.publishStreamName = null; - - /** - * This is the stream id of the main track that the current publishStreamId is going to be subtrack of it. It's added here to use in reconnect scenario - */ - this.publishMainTrack = null; - - /** - * This is the metadata that is being used to publish the stream. It's added here to use in reconnect scenario - */ - this.publishMetaData = null; - - /** - * This is the role for selective subtrack playback. It's added here to use in reconnect scenario - */ - this.publishRole = null; - - /** - * This is the token to play the stream. It's added here to use in reconnect scenario - */ - this.playToken = null; - - /** - * This is the room id to play the stream. It's added here to use in reconnect scenario - * This approach is old conferencing. It's better to use multi track conferencing - */ - this.playRoomId = null; - - /** - * These are enabled tracks to play the stream. It's added here to use in reconnect scenario - */ - this.playEnableTracks = null; - - /** - * This is the subscriber Id to play the stream. It's added here to use in reconnect scenario - */ - this.playSubscriberId = null; - - /** - * This is the subscriber code to play the stream. It's added here to use in reconnect scenario - */ - this.playSubscriberCode = null; - - /** - * This is the meta data to play the stream. It's added here to use in reconnect scenario - */ - this.playMetaData = null; - - /** - * This is the role for selective subtrack playback. It's added here to use in reconnect scenario - */ - this.playRole = null; - - /** - * This is the time info for the last reconnection attempt - */ - this.lastReconnectiontionTrialTime = 0; - - /** - * All media management works for teh local stream are made by @MediaManager class. - * for details please check @MediaManager - */ - this.mediaManager = new MediaManager({ - userParameters: initialValues, - webRTCAdaptor: this, - - callback: (info, obj) => { - this.notifyEventListeners(info, obj) - }, - callbackError: (error, message) => { - this.notifyErrorEventListeners(error, message) - }, - getSender: (streamId, type) => { - return this.getSender(streamId, type) - }, - }); - - //Initialize the local stream (if needed) and web socket connection - if (this.initializeComponents) { - this.initialize(); - } - } - - /** - * Init plugins - */ - initPlugins() { - WebRTCAdaptor.pluginInitMethods.forEach((initMethod) => { - initMethod(this); - }); - } - - /** - * Add event listener to be notified. This is generally for plugins - * @param {*} listener - */ - addEventListener(listener) { - this.eventListeners.push(listener); - } - - /** - * Add error event listener to be notified. Thisis generally for plugins - * @param {*} errorListener - */ - addErrorEventListener(errorListener) { - this.errorEventListeners.push(errorListener); - } - - /** - * Notify event listeners and callback method - * @param {*} info - * @param {*} obj - */ - notifyEventListeners(info, obj) { - this.eventListeners.forEach((listener) => { - listener(info, obj); - }); - if (this.callback != null) { - this.callback(info, obj); - } - } - - /** - * Notify error event listeners and callbackError method - * @param {*} error - * @param {*} message - */ - notifyErrorEventListeners(error, message) { - this.errorEventListeners.forEach((listener) => { - listener(error, message); - }); - if (this.callbackError != null) { - this.callbackError(error, message); - } - } - - - /** - * Called by constuctor to - * -check local stream unless it is in play mode - * -start websocket connection - */ - initialize() { - if (!this.isPlayMode && !this.onlyDataChannel && this.mediaManager.localStream == null) - { - //we need local stream because it not a play mode - return this.mediaManager.initLocalStream().then(() => { - this.initPlugins(); - this.checkWebSocketConnection(); - return new Promise((resolve, reject) => { - resolve("Wait 'initialized' callback from websocket"); - }); - }).catch(error => { - Logger.warn(error); - throw error; - }); - } - - return new Promise((resolve, reject) => { - this.initPlugins(); - this.checkWebSocketConnection(); - resolve("Wait 'initialized' callback from websocket"); - }); - - } - - /** - * Called to start a new WebRTC stream. AMS responds with start message. - * Parameters: - * @param {string} streamId : unique id for the stream - * @param {string=} [token] : required if any stream security (token control) enabled. Check https://github.com/ant-media/Ant-Media-Server/wiki/Stream-Security-Documentation - * @param {string=} [subscriberId] : required if TOTP enabled. Check https://github.com/ant-media/Ant-Media-Server/wiki/Time-based-One-Time-Password-(TOTP) - * @param {string=} [subscriberCode] : required if TOTP enabled. Check https://github.com/ant-media/Ant-Media-Server/wiki/Time-based-One-Time-Password-(TOTP) - * @param {string=} [streamName] : required if you want to set a name for the stream - * @param {string=} [mainTrack] : required if you want to start the stream as a subtrack for a main stream which has id of this parameter. - * Check:https://antmedia.io/antmediaserver-webrtc-multitrack-playing-feature/ - * !!! for multitrack conference set this value with roomName - * @param {string=} [metaData] : a free text information for the stream to AMS. It is provided to Rest methods by the AMS - * @param {string=} [role] : role for the stream. It is used for selective forwarding of subtracks in conference mode. - */ - publish(streamId, token, subscriberId, subscriberCode, streamName, mainTrack, metaData, role) { - //TODO: should refactor the repeated code - this.publishStreamId = streamId; - this.mediaManager.publishStreamId = streamId; - this.publishToken = token; - this.publishSubscriberId = subscriberId; - this.publishSubscriberCode = subscriberCode; - this.publishStreamName = streamName; - this.publishMainTrack = mainTrack; - this.publishMetaData = metaData; - this.publishRole = role; - if (this.onlyDataChannel) { - this.sendPublishCommand(streamId, token, subscriberId, subscriberCode, streamName, mainTrack, metaData, role, false, false); - } - //If it started with playOnly mode and wants to publish now - else if (this.mediaManager.localStream == null) { - this.mediaManager.initLocalStream().then(() => { - let videoEnabled = false; - let audioEnabled = false; - if (this.mediaManager.localStream != null) { - videoEnabled = this.mediaManager.localStream.getVideoTracks().length > 0; - audioEnabled = this.mediaManager.localStream.getAudioTracks().length > 0; - } - this.sendPublishCommand(streamId, token, subscriberId, subscriberCode, streamName, mainTrack, metaData, role, videoEnabled, audioEnabled) - - }).catch(error => { - Logger.warn(error); - throw error; - }); - } else { - let videoEnabled = this.mediaManager.localStream.getVideoTracks().length > 0; - let audioEnabled = this.mediaManager.localStream.getAudioTracks().length > 0; - this.sendPublishCommand(streamId, token, subscriberId, subscriberCode, streamName, mainTrack, metaData, role, videoEnabled, audioEnabled); - } - //init peer connection for reconnectIfRequired - this.initPeerConnection(streamId, "publish"); - setTimeout(() => { - //check if it is connected or not - //this resolves if the server responds with some error message - if (this.iceConnectionState(this.publishStreamId) != "checking" && this.iceConnectionState(this.publishStreamId) != "connected" && this.iceConnectionState(this.publishStreamId) != "completed") { - //if it is not connected, try to reconnect - this.reconnectIfRequired(0); - } - }, 3000); - - } - - sendPublishCommand(streamId, token, subscriberId, subscriberCode, streamName, mainTrack, metaData, role, videoEnabled, audioEnabled) { - let jsCmd = { - command: "publish", - streamId: streamId, - token: token, - subscriberId: (typeof subscriberId !== undefined && subscriberId != null) ? subscriberId : "", - subscriberCode: (typeof subscriberCode !== undefined && subscriberCode != null) ? subscriberCode : "", - streamName: (typeof streamName !== undefined && streamName != null) ? streamName : "", - mainTrack: (typeof mainTrack !== undefined && mainTrack != null) ? mainTrack : "", - video: videoEnabled, - audio: audioEnabled, - metaData: (typeof metaData !== undefined && metaData != null) ? metaData : "", - role: (typeof role !== undefined && role != null) ? role : "", - }; - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called to join a room. AMS responds with joinedTheRoom message. - * Parameters: - * @param {string} roomName : unique id of the room - * @param {string=} streamId : unique id of the stream belongs to this participant - * @param {string=} mode : legacy for older implementation (default value) - * mcu for merging streams - * amcu: audio only conferences with mixed audio - */ - joinRoom(roomName, streamId, mode) { - this.roomName = roomName; - - let jsCmd = { - command: "joinRoom", - room: roomName, - streamId: streamId, - mode: mode, - } - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called to start a playing session for a stream. AMS responds with start message. - * Parameters: - * @param {string} streamId :(string) unique id for the stream that you want to play - * @param {string=} token :(string) required if any stream security (token control) enabled. Check https://github.com/ant-media/Ant-Media-Server/wiki/Stream-Security-Documentation - * @param {string=} roomId :(string) required if this stream is belonging to a room participant - * @param {Array.=} enableTracks :(array) required if the stream is a main stream of multitrack playing. You can pass the the subtrack id list that you want to play. - * you can also provide a track id that you don't want to play by adding ! before the id. - * @param {string=} subscriberId :(string) required if TOTP enabled. Check https://github.com/ant-media/Ant-Media-Server/wiki/Time-based-One-Time-Password-(TOTP) - * @param {string=} subscriberCode :(string) required if TOTP enabled. Check https://github.com/ant-media/Ant-Media-Server/wiki/Time-based-One-Time-Password-(TOTP) - * @param {string=} metaData :(string, json) a free text information for the stream to AMS. It is provided to Rest methods by the AMS - * @param {string=} [role] : role for the stream. It is used for selective forwarding of subtracks in conference mode. - */ - play(streamId, token, roomId, enableTracks, subscriberId, subscriberCode, metaData, role) { - this.playStreamId.push(streamId); - this.playToken = token; - this.playRoomId = roomId; - this.playEnableTracks = enableTracks; - this.playSubscriberId = subscriberId; - this.playSubscriberCode = subscriberCode; - this.playMetaData = metaData; - this.playRole = role; - - let jsCmd = - { - command: "play", - streamId: streamId, - token: typeof token !== undefined && token != null ? token : "", - room: typeof roomId !== undefined && roomId != null ? roomId : "", - trackList: typeof enableTracks !== undefined && enableTracks != null ? enableTracks : [], - subscriberId: typeof subscriberId !== undefined && subscriberId != null ? subscriberId : "", - subscriberCode: typeof subscriberCode !== undefined && subscriberId != null ? subscriberCode : "", - viewerInfo: typeof metaData !== undefined && metaData != null ? metaData : "", - role: (typeof role !== undefined && role != null) ? role : "", - } - - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - - //init peer connection for reconnectIfRequired - this.initPeerConnection(streamId, "play"); - - setTimeout(() => { - //check if it is connected or not - //this resolves if the server responds with some error message - if (this.iceConnectionState(streamId) != "checking" && - this.iceConnectionState(streamId) != "connected" && - this.iceConnectionState(streamId) != "completed") - { - //if it is not connected, try to reconnect - this.reconnectIfRequired(0); - } - }, 3000); - } - - /** - * Reconnects to the stream if it is not stopped on purpose - * @param {number} [delayMs] - * @returns - */ - reconnectIfRequired(delayMs=3000) - { - if (this.reconnectIfRequiredFlag) - { - //It's important to run the following methods after 3000 ms because the stream may be stopped by the user in the meantime - if (delayMs > 0) - { - setTimeout(() => { - this.tryAgain(); - }, delayMs); + this.remoteDescriptionSet[streamId] = true; + var length = this.iceCandidateList[streamId].length; + Logger.debug("Ice candidate list size to be added: " + length); + for (var i = 0; i < length; i++) { + this.addIceCandidate(streamId, this.iceCandidateList[streamId][i]); } - else { - this.tryAgain() + this.iceCandidateList[streamId] = []; + + if (isTypeOffer) { + //SDP constraints may be different in play mode + Logger.debug("try to create answer for stream id: " + streamId); + + this.remotePeerConnection[streamId].createAnswer(this.sdp_constraints) + .then(configuration => { + Logger.debug("created answer for stream id: " + streamId); + //support for stereo + configuration.sdp = configuration.sdp.replace("useinbandfec=1", "useinbandfec=1; stereo=1"); + this.gotDescription(configuration, streamId); + }) + .catch((error) => { + Logger.error("create answer error :" + error); + }); } - } - } - tryAgain() { + }).catch((error) => { + if (this.debug) { + Logger.error("set remote description is failed with error: " + error); + } + if (error.toString().indexOf("InvalidAccessError") > -1 || error.toString().indexOf("setRemoteDescription") > -1) { + /** + * This error generally occurs in codec incompatibility. + * AMS for a now supports H.264 codec. This error happens when some browsers try to open it from VP8. + */ + this.notifyErrorEventListeners("notSetRemoteDescription"); + } + }); - const now = Date.now(); - //to prevent too many trial from different paths - if(now - this.lastReconnectiontionTrialTime < 3000) { - return; - } - this.lastReconnectiontionTrialTime = now; + } - //reconnect publish - //if remotePeerConnection has a peer connection for the stream id, it means that it is not stopped on purpose + /** + * Called by WebSocketAdaptor when new ice candidate is received from AMS. + * If Remote Description (SDP) is already set, the candidate is added immediately, + * otherwise stored in @iceCandidateList to add after Remote Description (SDP) set. + * @param {string} idOfTheStream : unique id for the stream + * @param {number|null} tmpLabel : sdpMLineIndex + * @param {string} tmpCandidate : ice candidate + */ + takeCandidate(idOfTheStream, tmpLabel, tmpCandidate) { + var streamId = idOfTheStream; + var label = tmpLabel; + var candidateSdp = tmpCandidate; + + var candidate = new RTCIceCandidate({ + sdpMLineIndex: label, + candidate: candidateSdp + }); + + var dataChannelMode = "peer"; + this.initPeerConnection(streamId, dataChannelMode); + + Logger.debug("takeCandidate:" + candidateSdp) + + if (this.remoteDescriptionSet[streamId] == true) { + this.addIceCandidate(streamId, candidate); + } else { + Logger.debug("Ice candidate is added to list because remote description is not set yet"); + this.iceCandidateList[streamId].push(candidate); + } + }; - if (this.remotePeerConnection[this.publishStreamId] != null && - //check connection status to not stop streaming an active stream - this.iceConnectionState(this.publishStreamId) != "checking" && - this.iceConnectionState(this.publishStreamId) != "connected" && - this.iceConnectionState(this.publishStreamId) != "completed") - { - // notify that reconnection process started for publish - this.notifyEventListeners("reconnection_attempt_for_publisher", this.publishStreamId); - - this.stop(this.publishStreamId); - setTimeout(() => { - //publish about some time later because server may not drop the connection yet - //it may trigger already publishing error - Logger.log("Trying publish again for stream: " + this.publishStreamId); - this.publish(this.publishStreamId, this.publishToken, this.publishSubscriberId, this.publishSubscriberCode, this.publishStreamName, this.publishMainTrack, this.publishMetaData, this.publishRole); - }, 500); - } - - //reconnect play - for (var index in this.playStreamId) - { - let streamId = this.playStreamId[index]; - if (this.remotePeerConnection[streamId] != "null" && - //check connection status to not stop streaming an active stream - this.iceConnectionState(streamId) != "checking" && - this.iceConnectionState(streamId) != "connected" && - this.iceConnectionState(streamId) != "completed") - { - // notify that reconnection process started for play - this.notifyEventListeners("reconnection_attempt_for_player", streamId); - - Logger.log("It will try to play again for stream: " + streamId + " because it is not stopped on purpose") - this.stop(streamId); - setTimeout(() => { - //play about some time later because server may not drop the connection yet - //it may trigger already playing error - Logger.log("Trying play again for stream: " + streamId); - this.play(streamId, this.playToken, this.playRoomId, this.playEnableTracks, this.playSubscriberId, this.playSubscriberCode, this.playMetaData, this.playRole); - }, 500); - } - } - } - - /** - * Called to stop a publishing/playing session for a stream. AMS responds with publishFinished or playFinished message. - * Parameters: - * @param {string} streamId : unique id for the stream that you want to stop publishing or playing - */ - stop(streamId) { - //stop is called on purpose and it deletes the peer connection from remotePeerConnections - this.closePeerConnection(streamId); - - if (this.webSocketAdaptor != null && this.webSocketAdaptor.isConnected()) { - let jsCmd = { - command: "stop", - streamId: streamId, - }; - - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - } - - /** - * Called to join a peer-to-peer mode session as peer. AMS responds with joined message. - * Parameters: - * @param {string} streamId : unique id for the peer-to-peer session - */ - join(streamId) { - let jsCmd = { - command: "join", - streamId: streamId, - multiPeer: this.isMultiPeer && this.multiPeerStreamId == null, - mode: this.isPlayMode ? "play" : "both", - }; - - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called by browser when a new track is added to WebRTC connetion. This is used to infor html pages with newStreamAvailable callback. - * Parameters: - * event: TODO - * streamId: unique id for the stream - */ - onTrack(event, streamId) - { - Logger.debug("onTrack for stream"); - if (this.remoteVideo != null) { - if (this.remoteVideo.srcObject !== event.streams[0]) { - this.remoteVideo.srcObject = event.streams[0]; - Logger.debug('Received remote stream'); - } - } - else { - var dataObj = { - stream: event.streams[0], - track: event.track, - streamId: streamId, - trackId: this.idMapping[streamId][event.transceiver.mid], - } - this.notifyEventListeners("newTrackAvailable", dataObj); - - //deprecated. Listen newTrackAvailable in callback. It's kept for backward compatibility - this.notifyEventListeners("newStreamAvailable", dataObj); - - } - } - - /** - * Called to leave from a conference room. AMS responds with leavedTheRoom message. - * Parameters: - * @param {string} roomName : unique id for the conference room - */ - leaveFromRoom(roomName) { - for (var key in this.remotePeerConnection) { - this.closePeerConnection(key); - } - this.roomName = roomName; - var jsCmd = { - command: "leaveFromRoom", - room: roomName, - }; - Logger.debug("leave request is sent for " + roomName); - - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called to leave from a peer-to-peer mode session. AMS responds with leaved message. - * Parameters: - * @param {string} streamId : unique id for the peer-to-peer session - */ - leave(streamId) { - var jsCmd = { - command: "leave", - streamId: this.isMultiPeer && this.multiPeerStreamId != null ? this.multiPeerStreamId : streamId, - }; - - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - this.closePeerConnection(streamId); - this.multiPeerStreamId = null; - } - - /** - * Called to get a stream information for a specific stream. AMS responds with streamInformation message. - * Parameters: - * @param {string} streamId : unique id for the stream that you want to get info about - */ - getStreamInfo(streamId) { - let jsCmd = { - command: "getStreamInfo", - streamId: streamId, - }; - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called to get the list of video track assignments. AMS responds with the videoTrackAssignmentList message. - * Parameters: - * @param {string} streamId : unique id for the stream that you want to get info about - */ - requestVideoTrackAssignments(streamId) { - let jsCmd = { - command: "getVideoTrackAssignmentsCommand", - streamId: streamId, - }; - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called to get the broadcast object for a specific stream. AMS responds with the broadcastObject callback. - * Parameters: - * @param {string} streamId : unique id for the stream that you want to get info about - */ - getBroadcastObject(streamId) { - let jsCmd = { - command: "getBroadcastObject", - streamId: streamId, - }; - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called to update the meta information for a specific stream. - * Parameters: - * @param {string} streamId : unique id for the stream that you want to update MetaData - * @param {string} metaData : new free text information for the stream - */ - updateStreamMetaData(streamId, metaData) { - var jsCmd = { - command: "updateStreamMetaData", - streamId: streamId, - metaData: metaData, - }; - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called to get the room information for a specific room. AMS responds with roomInformation message - * which includes the ids and names of the streams in that room. - * If there is no active streams in the room, AMS returns error `no_active_streams_in_room` in error callback - * Parameters: - * @param {string} roomName : unique id for the room that you want to get info about - * @param {string} streamId : unique id for the stream that is streamed by this @WebRTCAdaptor - */ - getRoomInfo(roomName, streamId) { - var jsCmd = { - command: "getRoomInfo", - streamId: streamId, - room: roomName, - }; - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called to get the subtracks for a specific maintrack. AMS responds with the subtrackList callback. - * Parameters: - * @param {string} streamId : main track id - * @param {string} role : filter the subtracks with the role - * @param {number} offset : offset for the subtrack list - * @param {number} size : size for the subtrack list - */ - getSubtracks(streamId, role, offset, size) { - let jsCmd = { - command: "getSubtracks", - streamId: streamId, - role: role, - offset: offset, - size: size, - }; - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called to enable/disable data flow from the AMS for a specific track under a main track. - * Parameters: - * @param {string} mainTrackId : unique id for the main stream - * @param {string} trackId : unique id for the track that you want to enable/disable data flow for - * @param {boolean} enabled : true or false - */ - enableTrack(mainTrackId, trackId, enabled) { - var jsCmd = { - command: "enableTrack", - streamId: mainTrackId, - trackId: trackId, - enabled: enabled, - }; - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called to get the track ids under a main stream. AMS responds with trackList message. - * Parameters: - * @param {string} streamId : unique id for the main stream - * @param {string=} [token] : not used - * TODO: check this function - */ - getTracks(streamId, token) { - this.playStreamId.push(streamId); - var jsCmd = - { - command: "getTrackList", - streamId: streamId, - token: token, - } - - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called by WebSocketAdaptor when a new ice candidate is received from AMS. - * Parameters: - * event: TODO - * streamId: unique id for the stream - */ - iceCandidateReceived(event, streamId) { - if (event.candidate) { - - var protocolSupported = false; - - if (event.candidate.candidate == "") { - //event candidate can be received and its value can be "". - //don't compare the protocols - protocolSupported = true; - } else if (typeof event.candidate.protocol == "undefined") { - this.candidateTypes.forEach(element => { - if (event.candidate.candidate.toLowerCase().includes(element)) { - protocolSupported = true; - } - }); - } else { - protocolSupported = this.candidateTypes.includes(event.candidate.protocol.toLowerCase()); - } - - - if (protocolSupported) { - - var jsCmd = { - command: "takeCandidate", - streamId: streamId, - label: event.candidate.sdpMLineIndex, - id: event.candidate.sdpMid, - candidate: event.candidate.candidate - }; - - if (this.debug) { - Logger.debug("sending ice candiate for stream Id " + streamId); - Logger.debug(JSON.stringify(event.candidate)); - } - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } else { - Logger.debug("Candidate's protocol(full sdp: " + event.candidate.candidate + ") is not supported. Supported protocols: " + this.candidateTypes); - if (event.candidate.candidate != "") { // - this.notifyErrorEventListeners("protocol_not_supported", "Support protocols: " + this.candidateTypes.toString() + " candidate: " + event.candidate.candidate); - } - } - } else { - Logger.debug("No event.candidate in the iceCandidate event"); - } - } - - /** - * Called internally to sanitize the text if it contains script to prevent xss - * @param text - * @returns {*} - */ - sanitizeHTML(text) { - if(text.includes("script")) - return text.replace(//g, ">"); - return text - } - - /** - * Called internally to initiate Data Channel. - * Note that Data Channel should be enabled fromAMS settings. - * @param {string} streamId : unique id for the stream - * @param {*} dataChannel : provided by PeerConnection - */ - initDataChannel(streamId, dataChannel) { - dataChannel.onerror = (error) => { - Logger.debug("Data Channel Error:", error); - var obj = { - streamId: streamId, - error: error - }; - Logger.debug("channel status: ", dataChannel.readyState); - if (dataChannel.readyState != "closed") { - this.notifyErrorEventListeners("data_channel_error", obj); - } - }; - - dataChannel.onmessage = (event) => { - var obj = { - streamId: streamId, - data: event.data, - }; - - var data = obj.data; - - if (typeof data === 'string' || data instanceof String) { - obj.data = this.sanitizeHTML(obj.data) - this.notifyEventListeners("data_received", obj); - } else { - var length = data.length || data.size || data.byteLength; - - var view = new Int32Array(data, 0, 1); - var token = view[0]; - - var msg = this.receivingMessages[token]; - if (msg == undefined) { - var view = new Int32Array(data, 0, 2); - var size = view[1]; - msg = new ReceivingMessage(size); - this.receivingMessages[token] = msg; - if (length > 8) { - Logger.debug("something went wrong in msg receiving"); - } - return; - } - - var rawData = data.slice(4, length); - - var dataView = new Uint8Array(msg.data); - dataView.set(new Uint8Array(rawData), msg.received, length - 4); - msg.received += length - 4; - - if (msg.size == msg.received) { - obj.data = msg.data; - this.notifyEventListeners("data_received", obj); - } - } - }; - - dataChannel.onopen = () => { - this.remotePeerConnection[streamId].dataChannel = dataChannel; - Logger.debug("Data channel is opened"); - this.notifyEventListeners("data_channel_opened", streamId) - }; - - dataChannel.onclose = () => { - Logger.debug("Data channel is closed"); - this.notifyEventListeners("data_channel_closed", streamId); - }; - } - - /** - * Called internally to initiate PeerConnection. - * @param {string} streamId : unique id for the stream - * @param {string} dataChannelMode : can be "publish" , "play" or "peer" based on this it is decided which way data channel is created - */ - initPeerConnection(streamId, dataChannelMode) { + /** + * Called internally to add the Ice Candidate to PeerConnection + * @param {string} streamId : unique id for the stream + * @param {object} candidate : ice candidate + */ + addIceCandidate(streamId, candidate) { + var protocolSupported = false; + if (candidate.candidate == "") { + //candidate can be received and its value can be "". + //don't compare the protocols + protocolSupported = true; + } else if (typeof candidate.protocol == "undefined") { + this.candidateTypes.forEach(element => { + if (candidate.candidate.toLowerCase().includes(element)) { + protocolSupported = true; + } + }); + } else { + protocolSupported = this.candidateTypes.includes(candidate.protocol.toLowerCase()); + } - //null == undefined -> it's true - //null === undefined -> it's false + if (protocolSupported) { + + this.remotePeerConnection[streamId].addIceCandidate(candidate) + .then(response => { + if (this.debug) { + Logger.debug("Candidate is added for stream " + streamId); + } + }) + .catch((error) => { + Logger.error("ice candiate cannot be added for stream id: " + streamId + " error is: " + error); + Logger.error(candidate); + }); + } else { + if (this.debug) { + Logger.debug("Candidate's protocol(" + candidate.protocol + ") is not supported." + + "Candidate: " + candidate.candidate + " Supported protocols:" + this.candidateTypes); + } + } + }; - if (this.remotePeerConnection[streamId] == null) { - let closedStreamId = streamId; - Logger.debug("stream id in init peer connection: " + streamId + " close stream id: " + closedStreamId); - this.remotePeerConnection[streamId] = new RTCPeerConnection(this.peerconnection_config); - this.remoteDescriptionSet[streamId] = false; - this.iceCandidateList[streamId] = new Array(); - if (!this.playStreamId.includes(streamId)) { - if (this.mediaManager.localStream != null) { - this.mediaManager.localStream.getTracks().forEach(track => { - - let rtpSender = this.remotePeerConnection[streamId].addTrack(track, this.mediaManager.localStream); - if (track.kind == 'video') - { - let parameters = rtpSender.getParameters(); - parameters.degradationPreference = this.degradationPreference; - rtpSender.setParameters(parameters).then(() => { - Logger.info("Degradation Preference is set to " + this.degradationPreference); - }).catch((err) => { - Logger.warn("Degradation Preference cannot be set to " + this.degradationPreference) - }); - } - // - //parameters.degradationPreference - }); - } - } - this.remotePeerConnection[streamId].onicecandidate = event => { - this.iceCandidateReceived(event, closedStreamId); - } - this.remotePeerConnection[streamId].ontrack = event => { - this.onTrack(event, closedStreamId); - } - - this.remotePeerConnection[streamId].onnegotiationneeded = event => { - Logger.debug("onnegotiationneeded"); - } - - if (this.dataChannelEnabled) { - // skip initializing data channel if it is disabled - if (dataChannelMode == "publish") { - //open data channel if it's publish mode peer connection - const dataChannelOptions = { - ordered: true, - }; - if (this.remotePeerConnection[streamId].createDataChannel) { - var dataChannel = this.remotePeerConnection[streamId].createDataChannel(streamId, dataChannelOptions); - this.initDataChannel(streamId, dataChannel); - } else { - Logger.warn("CreateDataChannel is not supported"); - } - - } else if (dataChannelMode == "play") { - //in play mode, server opens the data channel - this.remotePeerConnection[streamId].ondatachannel = ev => { - this.initDataChannel(streamId, ev.channel); - }; - } else { - //for peer mode do both for now - const dataChannelOptions = { - ordered: true, - }; - - if (this.remotePeerConnection[streamId].createDataChannel) { - var dataChannelPeer = this.remotePeerConnection[streamId].createDataChannel(streamId, dataChannelOptions); - this.initDataChannel(streamId, dataChannelPeer); - - this.remotePeerConnection[streamId].ondatachannel = ev => { - this.initDataChannel(streamId, ev.channel); - }; - } else { - Logger.warn("CreateDataChannel is not supported"); - } - } - } - - this.remotePeerConnection[streamId].oniceconnectionstatechange = event => { - var obj = {state: this.remotePeerConnection[streamId].iceConnectionState, streamId: streamId}; - if (obj.state == "failed" || obj.state == "disconnected" || obj.state == "closed") { - this.reconnectIfRequired(3000); - } - this.notifyEventListeners("ice_connection_state_changed", obj); - - // - if (!this.isPlayMode && !this.playStreamId.includes(streamId)) { - if (this.remotePeerConnection[streamId].iceConnectionState == "connected") { - - this.mediaManager.changeBandwidth(this.mediaManager.bandwidth, streamId).then(() => { - Logger.debug("Bandwidth is changed to " + this.mediaManager.bandwidth); - }) - .catch(e => Logger.warn(e)); - } - } - } - - } - - return this.remotePeerConnection[streamId]; - } - - /** - * Called internally to close PeerConnection. - * @param {string} streamId : unique id for the stream - */ - closePeerConnection(streamId) { - var peerConnection = this.remotePeerConnection[streamId]; - if (peerConnection != null) { - this.remotePeerConnection[streamId] = null; - delete this.remotePeerConnection[streamId]; - if (peerConnection.dataChannel != null) { - peerConnection.dataChannel.close(); - } - if (peerConnection.signalingState != "closed") { - peerConnection.close(); - } - var playStreamIndex = this.playStreamId.indexOf(streamId); - if (playStreamIndex != -1) { - this.playStreamId.splice(playStreamIndex, 1); - } - } - //this is for the stats - if (this.remotePeerConnectionStats[streamId] != null) { - clearInterval(this.remotePeerConnectionStats[streamId].timerId); - delete this.remotePeerConnectionStats[streamId]; - } - if (this.soundMeters[streamId] != null) { - delete this.soundMeters[streamId]; - } - } - - /** - * Called to get the signalling state for a stream. - * This information can be used for error handling. - * Check: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/connectionState - * @param {string} streamId : unique id for the stream - */ - signallingState(streamId) { - if (this.remotePeerConnection[streamId] != null) { - return this.remotePeerConnection[streamId].signalingState; - } - return null; - } - - /** - * Called to get the ice connection state for a stream. - * This information can be used for error handling. - * Check: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/iceConnectionState - * @param {string} streamId : unique id for the stream - */ - iceConnectionState(streamId) { - if (this.remotePeerConnection[streamId] != null) { - return this.remotePeerConnection[streamId].iceConnectionState; - } - return null; - } - - /** - * Called by browser when Local Configuration (SDP) is created successfully. - * It is set as LocalDescription first then sent to AMS. - * @param {object} configuration : created Local Configuration (SDP) - * @param {string} streamId : unique id for the stream - */ - gotDescription(configuration, streamId) { - - this.remotePeerConnection[streamId] - .setLocalDescription(configuration) - .then(responose => { - Logger.debug("Set local description successfully for stream Id " + streamId); - - var jsCmd = { - command: "takeConfiguration", - streamId: streamId, - type: configuration.type, - sdp: configuration.sdp - - }; - - Logger.debug("setLocalDescription:"+configuration.sdp); - - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - - }).catch((error) => { - Logger.error("Cannot set local description. Error is: " + error); - }); - } - - /** - * Called by WebSocketAdaptor when Remote Configuration (SDP) is received from AMS. - * It is set as RemoteDescription first then if @iceCandidateList has candidate that - * is received bfore this message, it is added as ice candidate. - * @param {object} configuration : received Remote Configuration (SDP) - * @param {string} idOfStream : unique id for the stream - * @param {string} typeOfConfiguration - * @param {string} idMapping : stream id and track id (which is provided in SDP) mapping in MultiTrack Playback and conference. - * It is recorded to match stream id as new tracks are added with @onTrack - */ - takeConfiguration(idOfStream, configuration, typeOfConfiguration, idMapping) { - var streamId = idOfStream - var type = typeOfConfiguration; - var conf = configuration; - var isTypeOffer = (type == "offer"); - - var dataChannelMode = "publish"; - if (isTypeOffer) { - dataChannelMode = "play"; - } - - this.idMapping[streamId] = idMapping; - - this.initPeerConnection(streamId, dataChannelMode); + /** + * Called by WebSocketAdaptor when start message is received //TODO: may be changed. this logic shouldn't be in WebSocketAdaptor + * @param {string} idOfStream : unique id for the stream + */ + startPublishing(idOfStream) { + let streamId = idOfStream; + + let peerConnection = this.initPeerConnection(streamId, "publish"); + + //this.remotePeerConnection[streamId] + peerConnection.createOffer(this.sdp_constraints) + .then(configuration => { + this.gotDescription(configuration, streamId); + }) + .catch((error) => { + Logger.error("create offer error for stream id: " + streamId + " error: " + error); + }); + } - Logger.debug("setRemoteDescription:" + conf); - this.remotePeerConnection[streamId].setRemoteDescription(new RTCSessionDescription({ - sdp: conf, - type: type - })).then(response => { - - if (this.debug) { - Logger.debug("set remote description is succesfull with response: " + response + " for stream : " - + streamId + " and type: " + type); - Logger.debug(conf); - } - - this.remoteDescriptionSet[streamId] = true; - var length = this.iceCandidateList[streamId].length; - Logger.debug("Ice candidate list size to be added: " + length); - for (var i = 0; i < length; i++) { - this.addIceCandidate(streamId, this.iceCandidateList[streamId][i]); - } - this.iceCandidateList[streamId] = []; - - if (isTypeOffer) { - //SDP constraints may be different in play mode - Logger.debug("try to create answer for stream id: " + streamId); - - this.remotePeerConnection[streamId].createAnswer(this.sdp_constraints) - .then(configuration => { - Logger.debug("created answer for stream id: " + streamId); - //support for stereo - configuration.sdp = configuration.sdp.replace("useinbandfec=1", "useinbandfec=1; stereo=1"); - this.gotDescription(configuration, streamId); - }) - .catch((error) => { - Logger.error("create answer error :" + error); - }); - } - - }).catch((error) => { - if (this.debug) { - Logger.error("set remote description is failed with error: " + error); - } - if (error.toString().indexOf("InvalidAccessError") > -1 || error.toString().indexOf("setRemoteDescription") > -1) { - /** - * This error generally occurs in codec incompatibility. - * AMS for a now supports H.264 codec. This error happens when some browsers try to open it from VP8. - */ - this.notifyErrorEventListeners("notSetRemoteDescription"); - } - }); - - } - - /** - * Called by WebSocketAdaptor when new ice candidate is received from AMS. - * If Remote Description (SDP) is already set, the candidate is added immediately, - * otherwise stored in @iceCandidateList to add after Remote Description (SDP) set. - * @param {string} idOfTheStream : unique id for the stream - * @param {number|null} tmpLabel : sdpMLineIndex - * @param {string} tmpCandidate : ice candidate - */ - takeCandidate(idOfTheStream, tmpLabel, tmpCandidate) { - var streamId = idOfTheStream; - var label = tmpLabel; - var candidateSdp = tmpCandidate; - - var candidate = new RTCIceCandidate({ - sdpMLineIndex: label, - candidate: candidateSdp - }); - - var dataChannelMode = "peer"; - this.initPeerConnection(streamId, dataChannelMode); - - Logger.debug("takeCandidate:" + candidateSdp) - - if (this.remoteDescriptionSet[streamId] == true) { - this.addIceCandidate(streamId, candidate); - } else { - Logger.debug("Ice candidate is added to list because remote description is not set yet"); - this.iceCandidateList[streamId].push(candidate); - } - }; - - /** - * Called internally to add the Ice Candidate to PeerConnection - * @param {string} streamId : unique id for the stream - * @param {object} candidate : ice candidate - */ - addIceCandidate(streamId, candidate) { - var protocolSupported = false; - if (candidate.candidate == "") { - //candidate can be received and its value can be "". - //don't compare the protocols - protocolSupported = true; - } else if (typeof candidate.protocol == "undefined") { - this.candidateTypes.forEach(element => { - if (candidate.candidate.toLowerCase().includes(element)) { - protocolSupported = true; - } - }); - } else { - protocolSupported = this.candidateTypes.includes(candidate.protocol.toLowerCase()); - } - - if (protocolSupported) { - - this.remotePeerConnection[streamId].addIceCandidate(candidate) - .then(response => { - if (this.debug) { - Logger.debug("Candidate is added for stream " + streamId); - } - }) - .catch((error) => { - Logger.error("ice candiate cannot be added for stream id: " + streamId + " error is: " + error); - Logger.error(candidate); - }); - } else { - if (this.debug) { - Logger.debug("Candidate's protocol(" + candidate.protocol + ") is not supported." + - "Candidate: " + candidate.candidate + " Supported protocols:" + this.candidateTypes); - } - } - }; - - /** - * Called by WebSocketAdaptor when start message is received //TODO: may be changed. this logic shouldn't be in WebSocketAdaptor - * @param {string} idOfStream : unique id for the stream - */ - startPublishing(idOfStream) { - let streamId = idOfStream; - - let peerConnection = this.initPeerConnection(streamId, "publish"); - - //this.remotePeerConnection[streamId] - peerConnection.createOffer(this.sdp_constraints) - .then(configuration => { - this.gotDescription(configuration, streamId); - }) - .catch((error) => { - Logger.error("create offer error for stream id: " + streamId + " error: " + error); - }); - } - - - /** - * Toggle video track on the server side. - * - * @param {string} streamId : is the id of the stream - * @param {string} trackId : is the id of the track. streamId is also one of the trackId of the stream. If you are having just a single track on your - * stream, you need to give streamId as trackId parameter as well. - * @param {boolean} enabled : is the enable/disable video track. If it's true, server sends video track. If it's false, server does not send video - */ - toggleVideo(streamId, trackId, enabled) { - let jsCmd = { - command: "toggleVideo", - streamId: streamId, - trackId: trackId, - enabled: enabled, - }; - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Toggle audio track on the server side. - * - * @param {string} streamId : is the id of the stream - * @param {string} trackId : is the id of the track. streamId is also one of the trackId of the stream. If you are having just a single track on your - * stream, you need to give streamId as trackId parameter as well. - * @param {boolean} enabled : is the enable/disable video track. If it's true, server sends audio track. If it's false, server does not send audio - * - */ - toggleAudio(streamId, trackId, enabled) { - var jsCmd = { - command: "toggleAudio", - streamId: streamId, - trackId: trackId, - enabled: enabled, - }; - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called to get statistics for a PeerConnection. It can be publisher or player. - * - * @param {string} streamId : unique id for the stream - */ - getStats(streamId) { - Logger.debug("peerstatsgetstats = " + this.remotePeerConnectionStats[streamId]); - return new Promise((resolve, reject) => { - - this.remotePeerConnection[streamId].getStats(null).then(stats => { - var bytesReceived = -1; - var videoPacketsLost = -1; - var audioPacketsLost = -1; - var fractionLost = -1; - var currentTime = -1; - var bytesSent = -1; - var videoPacketsSent = -1; - var audioPacketsSent = -1; - var audioLevel = -1; - var qlr = ""; - var framesEncoded = -1; - var width = -1; - var height = -1; - var fps = -1; - var frameWidth = -1; - var frameHeight = -1; - var videoRoundTripTime = -1; - var videoJitter = -1; - - var audioRoundTripTime = -1; - var audioJitter = -1; - - var framesDecoded = -1; - var framesDropped = -1; - var framesReceived = -1; - - var audioJitterAverageDelay = -1; - var videoJitterAverageDelay = -1; - var availableOutgoingBitrate = Infinity; - var currentRoundTripTime = -1; - - var audioPacketsReceived = -1; - var videoPacketsReceived = -1; - - var inboundRtp = []; - - stats.forEach(value => { - //Logger.debug(value); - if (value.type == "inbound-rtp" && typeof value.kind != "undefined") { - //this is coming when webrtc playing - - let inboundRtpObj = {}; - - inboundRtpObj.trackIdentifier = value.trackIdentifier; - - bytesReceived += value.bytesReceived; - if (value.kind == "audio") { - audioPacketsLost = value.packetsLost; - audioJitter = value.jitter; - audioPacketsReceived = value.packetsReceived; - - inboundRtpObj.audioPacketsLost = value.packetsLost; - } else if (value.kind == "video") { - videoPacketsLost = value.packetsLost; - inboundRtpObj.videoPacketsLost = value.packetsLost; - inboundRtpObj.framesDropped = value.framesDropped; - inboundRtpObj.framesDecoded = value.framesDecoded; - inboundRtpObj.framesPerSecond = value.framesPerSecond; + /** + * Toggle video track on the server side. + * + * @param {string} streamId : is the id of the stream + * @param {string} trackId : is the id of the track. streamId is also one of the trackId of the stream. If you are having just a single track on your + * stream, you need to give streamId as trackId parameter as well. + * @param {boolean} enabled : is the enable/disable video track. If it's true, server sends video track. If it's false, server does not send video + */ + toggleVideo(streamId, trackId, enabled) { + let jsCmd = { + command: "toggleVideo", + streamId: streamId, + trackId: trackId, + enabled: enabled, + }; + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Toggle audio track on the server side. + * + * @param {string} streamId : is the id of the stream + * @param {string} trackId : is the id of the track. streamId is also one of the trackId of the stream. If you are having just a single track on your + * stream, you need to give streamId as trackId parameter as well. + * @param {boolean} enabled : is the enable/disable video track. If it's true, server sends audio track. If it's false, server does not send audio + * + */ + toggleAudio(streamId, trackId, enabled) { + var jsCmd = { + command: "toggleAudio", + streamId: streamId, + trackId: trackId, + enabled: enabled, + }; + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called to get statistics for a PeerConnection. It can be publisher or player. + * + * @param {string} streamId : unique id for the stream + */ + getStats(streamId) { + Logger.debug("peerstatsgetstats = " + this.remotePeerConnectionStats[streamId]); + return new Promise((resolve, reject) => { + + this.remotePeerConnection[streamId].getStats(null).then(stats => { + + var peerStats = this.parseStats(stats, streamId); + this.notifyEventListeners("updated_stats", peerStats); + resolve(peerStats); + }).catch(err => { + resolve(false); + }); + }); + } + + + parseStats(stats, streamId) { + var bytesReceived = -1; + var videoPacketsLost = -1; + var audioPacketsLost = -1; + var fractionLost = -1; + var currentTime = -1; + var bytesSent = -1; + var videoPacketsSent = -1; + var audioPacketsSent = -1; + var audioLevel = -1; + var qlr = ""; + var framesEncoded = -1; + var width = -1; + var height = -1; + var fps = -1; + var frameWidth = -1; + var frameHeight = -1; + var videoRoundTripTime = -1; + var videoJitter = -1; + + var audioRoundTripTime = -1; + var audioJitter = -1; + + var framesDecoded = -1; + var framesDropped = -1; + var framesReceived = -1; + + var audioJitterAverageDelay = -1; + var videoJitterAverageDelay = -1; + var availableOutgoingBitrate = Infinity; + var currentRoundTripTime = -1; + + var audioPacketsReceived = -1; + var videoPacketsReceived = -1; + + var inboundRtp = []; + + stats.forEach(value => { + //Logger.debug(value); + if (value.type == "inbound-rtp" && typeof value.kind != "undefined") { + //this is coming when webrtc playing + + let inboundRtpObj = {}; + + inboundRtpObj.trackIdentifier = value.trackIdentifier; + + bytesReceived += value.bytesReceived; + if (value.kind == "audio") { + audioPacketsLost = value.packetsLost; + audioJitter = value.jitter; + audioPacketsReceived = value.packetsReceived; + + inboundRtpObj.audioPacketsLost = value.packetsLost; + } else if (value.kind == "video") { + videoPacketsLost = value.packetsLost; + inboundRtpObj.videoPacketsLost = value.packetsLost; + inboundRtpObj.framesDropped = value.framesDropped; + inboundRtpObj.framesDecoded = value.framesDecoded; + inboundRtpObj.framesPerSecond = value.framesPerSecond; + videoJitter = value.jitter; + videoPacketsReceived = value.packetsReceived; + } + + inboundRtpObj.bytesReceived = value.bytesReceived; + inboundRtpObj.jitterBufferDelay = value.jitterBufferDelay; + inboundRtpObj.lastPacketReceivedTimestamp = value.lastPacketReceivedTimestamp; + + fractionLost += value.fractionLost; + inboundRtpObj.fractionLost = value.fractionLost; + currentTime = value.timestamp; + inboundRtpObj.currentTime = value.timestamp; + + if (typeof value.frameWidth != "undefined") { + frameWidth = value.frameWidth; + inboundRtpObj.frameWidth = value.frameWidth; + } + if (typeof value.frameHeight != "undefined") { + frameHeight = value.frameHeight; + inboundRtpObj.frameHeight = value.frameHeight; + } + + if (typeof value.framesDecoded != "undefined") { + framesDecoded = value.framesDecoded; + inboundRtpObj.framesDecoded = value.framesDecoded; + } + + if (typeof value.framesDropped != "undefined") { + framesDropped = value.framesDropped; + inboundRtpObj.framesDropped = value.framesDropped; + } + + if (typeof value.framesReceived != "undefined") { + framesReceived = value.framesReceived; + inboundRtpObj.framesReceived = value.framesReceived; + } + + inboundRtp.push(inboundRtpObj); + + } + else if (value.type == "outbound-rtp") { + //TODO: SPLIT AUDIO AND VIDEO BITRATES + //it is for the publishing + if (value.kind == "audio") { + audioPacketsSent = value.packetsSent; + } else if (value.kind == "video") { + videoPacketsSent = value.packetsSent; + frameWidth = value.frameWidth; + frameHeight = value.frameHeight; + } + bytesSent += value.bytesSent + currentTime = value.timestamp + qlr = value.qualityLimitationReason; + if (value.framesEncoded != null) { //audio tracks are undefined here + framesEncoded += value.framesEncoded; + } + } else if (value.type == "track" && typeof value.kind != "undefined" && value.kind == "audio") { + if (typeof value.audioLevel != "undefined") { + audioLevel = value.audioLevel; + } + + if (typeof value.jitterBufferDelay != "undefined" && typeof value.jitterBufferEmittedCount != "undefined") { + audioJitterAverageDelay = value.jitterBufferDelay / value.jitterBufferEmittedCount; + } + } else if (value.type == "track" && typeof value.kind != "undefined" && value.kind == "video") { + if (typeof value.frameWidth != "undefined") { + frameWidth = value.frameWidth; + } + if (typeof value.frameHeight != "undefined") { + frameHeight = value.frameHeight; + } + + if (typeof value.framesDecoded != "undefined") { + framesDecoded = value.framesDecoded; + } + + if (typeof value.framesDropped != "undefined") { + framesDropped = value.framesDropped; + } + + if (typeof value.framesReceived != "undefined") { + framesReceived = value.framesReceived; + } + + if (typeof value.jitterBufferDelay != "undefined" && typeof value.jitterBufferEmittedCount != "undefined") { + videoJitterAverageDelay = value.jitterBufferDelay / value.jitterBufferEmittedCount; + } + } + + else if (value.type == "remote-inbound-rtp" && typeof value.kind != "undefined") { + //this is coming when webrtc publishing + + if (typeof value.packetsLost != "undefined") { + if (value.kind == "video") { + //this is the packetsLost for publishing + videoPacketsLost = value.packetsLost; + } else if (value.kind == "audio") { + //this is the packetsLost for publishing + audioPacketsLost = value.packetsLost; + } + } + + if (typeof value.roundTripTime != "undefined") { + if (value.kind == "video") { + videoRoundTripTime = value.roundTripTime; + } else if (value.kind == "audio") { + audioRoundTripTime = value.roundTripTime; + } + } + + if (typeof value.jitter != "undefined") { + if (value.kind == "video") { videoJitter = value.jitter; - videoPacketsReceived = value.packetsReceived; - } - - inboundRtpObj.bytesReceived = value.bytesReceived; - inboundRtpObj.jitterBufferDelay = value.jitterBufferDelay; - inboundRtpObj.lastPacketReceivedTimestamp = value.lastPacketReceivedTimestamp; - - fractionLost += value.fractionLost; - inboundRtpObj.fractionLost = value.fractionLost; - currentTime = value.timestamp; - inboundRtpObj.currentTime = value.timestamp; - - if (typeof value.frameWidth != "undefined") { - frameWidth = value.frameWidth; - inboundRtpObj.frameWidth = value.frameWidth; - } - if (typeof value.frameHeight != "undefined") { - frameHeight = value.frameHeight; - inboundRtpObj.frameHeight = value.frameHeight; - } - - if (typeof value.framesDecoded != "undefined") { - framesDecoded = value.framesDecoded; - inboundRtpObj.framesDecoded = value.framesDecoded; - } - - if (typeof value.framesDropped != "undefined") { - framesDropped = value.framesDropped; - inboundRtpObj.framesDropped = value.framesDropped; - } - - if (typeof value.framesReceived != "undefined") { - framesReceived = value.framesReceived; - inboundRtpObj.framesReceived = value.framesReceived; - } - - inboundRtp.push(inboundRtpObj); - - } - else if (value.type == "outbound-rtp") - { - //TODO: SPLIT AUDIO AND VIDEO BITRATES - //it is for the publishing - if (value.kind == "audio") { - audioPacketsSent = value.packetsSent; - } else if (value.kind == "video") { - videoPacketsSent = value.packetsSent; - frameWidth = value.frameWidth; - frameHeight = value.frameHeight; - } - bytesSent += value.bytesSent - currentTime = value.timestamp - qlr = value.qualityLimitationReason; - if (value.framesEncoded != null) { //audio tracks are undefined here - framesEncoded += value.framesEncoded; - } - } else if (value.type == "track" && typeof value.kind != "undefined" && value.kind == "audio") { - if (typeof value.audioLevel != "undefined") { - audioLevel = value.audioLevel; - } - - if (typeof value.jitterBufferDelay != "undefined" && typeof value.jitterBufferEmittedCount != "undefined") { - audioJitterAverageDelay = value.jitterBufferDelay / value.jitterBufferEmittedCount; - } - } else if (value.type == "track" && typeof value.kind != "undefined" && value.kind == "video") { - if (typeof value.frameWidth != "undefined") { - frameWidth = value.frameWidth; - } - if (typeof value.frameHeight != "undefined") { - frameHeight = value.frameHeight; - } - - if (typeof value.framesDecoded != "undefined") { - framesDecoded = value.framesDecoded; - } - - if (typeof value.framesDropped != "undefined") { - framesDropped = value.framesDropped; - } - - if (typeof value.framesReceived != "undefined") { - framesReceived = value.framesReceived; - } - - if (typeof value.jitterBufferDelay != "undefined" && typeof value.jitterBufferEmittedCount != "undefined") { - videoJitterAverageDelay = value.jitterBufferDelay / value.jitterBufferEmittedCount; - } - } - - else if (value.type == "remote-inbound-rtp" && typeof value.kind != "undefined") { - //this is coming when webrtc publishing - - if (typeof value.packetsLost != "undefined") { - if (value.kind == "video") { - //this is the packetsLost for publishing - videoPacketsLost = value.packetsLost; - } else if (value.kind == "audio") { - //this is the packetsLost for publishing - audioPacketsLost = value.packetsLost; - } - } - - if (typeof value.roundTripTime != "undefined") { - if (value.kind == "video") { - videoRoundTripTime = value.roundTripTime; - } else if (value.kind == "audio") { - audioRoundTripTime = value.roundTripTime; - } - } - - if (typeof value.jitter != "undefined") { - if (value.kind == "video") { - videoJitter = value.jitter; - } else if (value.kind == "audio") { - audioJitter = value.jitter; - } - } - } else if (value.type == "media-source") { - if (value.kind == "video") { //returns video source dimensions, not necessarily dimensions being encoded by browser - width = value.width; - height = value.height; - fps = value.framesPerSecond; - } - } - else if(value.type == "candidate-pair" && value.state == "succeeded" && value.availableOutgoingBitrate !=undefined){ - availableOutgoingBitrate = value.availableOutgoingBitrate/1000 - //currentRoundTripTime - currentRoundTripTime = value.currentRoundTripTime; - } - }); - - if (typeof this.remotePeerConnectionStats[streamId] == 'undefined' || this.remotePeerConnectionStats[streamId] == null) { - this.remotePeerConnectionStats[streamId] = new PeerStats(streamId); - } - - this.remotePeerConnectionStats[streamId].totalBytesReceived = bytesReceived; - this.remotePeerConnectionStats[streamId].videoPacketsLost = videoPacketsLost; - this.remotePeerConnectionStats[streamId].audioPacketsLost = audioPacketsLost; - this.remotePeerConnectionStats[streamId].fractionLost = fractionLost; - this.remotePeerConnectionStats[streamId].currentTime = currentTime; - this.remotePeerConnectionStats[streamId].totalBytesSent = bytesSent; - this.remotePeerConnectionStats[streamId].totalVideoPacketsSent = videoPacketsSent; - this.remotePeerConnectionStats[streamId].totalAudioPacketsSent = audioPacketsSent; - this.remotePeerConnectionStats[streamId].videoPacketsSent = videoPacketsSent; - this.remotePeerConnectionStats[streamId].audioPacketsSent = audioPacketsSent; - - this.remotePeerConnectionStats[streamId].audioLevel = audioLevel; - this.remotePeerConnectionStats[streamId].qualityLimitationReason = qlr; - this.remotePeerConnectionStats[streamId].totalFramesEncoded = framesEncoded; - this.remotePeerConnectionStats[streamId].resWidth = width; - this.remotePeerConnectionStats[streamId].resHeight = height; - this.remotePeerConnectionStats[streamId].srcFps = fps; - this.remotePeerConnectionStats[streamId].frameWidth = frameWidth; - this.remotePeerConnectionStats[streamId].frameHeight = frameHeight; - this.remotePeerConnectionStats[streamId].videoRoundTripTime = videoRoundTripTime; - this.remotePeerConnectionStats[streamId].videoJitter = videoJitter; - this.remotePeerConnectionStats[streamId].audioRoundTripTime = audioRoundTripTime; - this.remotePeerConnectionStats[streamId].audioJitter = audioJitter; - this.remotePeerConnectionStats[streamId].framesDecoded = framesDecoded; - this.remotePeerConnectionStats[streamId].framesDropped = framesDropped; - this.remotePeerConnectionStats[streamId].framesReceived = framesReceived; - - this.remotePeerConnectionStats[streamId].videoJitterAverageDelay = videoJitterAverageDelay; - this.remotePeerConnectionStats[streamId].audioJitterAverageDelay = audioJitterAverageDelay; - this.remotePeerConnectionStats[streamId].availableOutgoingBitrate = availableOutgoingBitrate; - - this.remotePeerConnectionStats[streamId].inboundRtpList = inboundRtp; - - this.remotePeerConnectionStats[streamId].currentRoundTripTime = currentRoundTripTime; - this.remotePeerConnectionStats[streamId].audioPacketsReceived = audioPacketsReceived; - this.remotePeerConnectionStats[streamId].videoPacketsReceived = videoPacketsReceived; - - this.notifyEventListeners("updated_stats", this.remotePeerConnectionStats[streamId]); - resolve(this.remotePeerConnectionStats[streamId]); - }).catch(err=>{ - resolve(false); - }); - }); - } - - /** - * Called to start a periodic timer to get statistics periodically (5 seconds) for a specific stream. - * - * @param {string} streamId : unique id for the stream - * @param {number} periodMs : period in milliseconds. Default value is 5000 ms. - */ - enableStats(streamId, periodMs = 5000) { - if (this.remotePeerConnectionStats[streamId] == null) { - this.remotePeerConnectionStats[streamId] = new PeerStats(streamId); - this.remotePeerConnectionStats[streamId].timerId = setInterval(() => { - this.getStats(streamId); - - }, periodMs); - } - } - - /** - * Called to stop the periodic timer which is set by @enableStats - * - * @param {string} streamId : unique id for the stream - */ - disableStats(streamId) { - if (this.remotePeerConnectionStats[streamId] != null || typeof this.remotePeerConnectionStats[streamId] != 'undefined') { - clearInterval(this.remotePeerConnectionStats[streamId].timerId); - delete this.remotePeerConnectionStats[streamId]; - } - } - - /** - * Called to check and start Web Socket connection if it is not started - */ - checkWebSocketConnection() { - if (this.webSocketAdaptor == null || (this.webSocketAdaptor.isConnected() == false && this.webSocketAdaptor.isConnecting() == false)) - { - Logger.debug("websocket url : " + this.websocketURL); - this.webSocketAdaptor = new WebSocketAdaptor({ - websocket_url: this.websocketURL, - webrtcadaptor: this, - callback: (info, obj) => { - if (info == "closed") { - this.reconnectIfRequired(); - } - this.notifyEventListeners(info, obj); - }, - callbackError: (error, message) => { - this.notifyErrorEventListeners(error, message) - }, - debug: this.debug - }); - } - } - - /** - * Called to stop Web Socket connection - * After calling this function, create new WebRTCAdaptor instance, don't use the the same object - * Because all streams are closed on server side as well when websocket connection is closed. - */ - closeWebSocket() { - for (var key in this.remotePeerConnection) { - this.closePeerConnection(key); - } - //free the remote peer connection by initializing again - this.remotePeerConnection = new Array(); - this.webSocketAdaptor.close(); - } - - /** - * @param {string} streamId Called to send a text message to other peer in the peer-to-peer sessionnnection is closed. - * @param {*} definition - * @param {*} data - */ - peerMessage(streamId, definition, data) { - var jsCmd = { - command: "peerMessageCommand", - streamId: streamId, - definition: definition, - data: data, - }; - - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called to force AMS to send the video with the specified resolution in case of Adaptive Streaming (ABR) enabled. - * Normally the resolution is automatically determined by AMS according to the network condition. - * @param {string} streamId : unique id for the stream - * @param {*} resolution : default is auto. You can specify any height value from the ABR list. - */ - forceStreamQuality(streamId, resolution) { - var jsCmd = { - command: "forceStreamQuality", - streamId: streamId, - streamHeight: resolution - }; - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called to send data via DataChannel. DataChannel should be enabled on AMS settings. - * @param {string} streamId : unique id for the stream - * @param {*} data : data that you want to send. It may be a text (may in Json format or not) or binary - */ - sendData(streamId, data) { - var CHUNK_SIZE = 16000; - if (this.remotePeerConnection[streamId] !== undefined) { - var dataChannel = this.remotePeerConnection[streamId].dataChannel; - if (dataChannel === undefined || dataChannel === null || typeof dataChannel === 'undefined') { - Logger.warn('dataChannel is null or undefined'); - return; - } else if (dataChannel.readyState !== 'open') { - Logger.warn('dataChannel.readyState is not open: ' + dataChannel.readyState); - return; - } - var length = data.length || data.size || data.byteLength; - var sent = 0; - - if (typeof data === 'string' || data instanceof String) { - dataChannel.send(data); - } else { - var token = Math.floor(Math.random() * 999999); - let header = new Int32Array(2); - header[0] = token; - header[1] = length; - - dataChannel.send(header); - - var sent = 0; - while (sent < length) { - var size = Math.min(length - sent, CHUNK_SIZE); - var buffer = new Uint8Array(size + 4); - var tokenArray = new Int32Array(1); - tokenArray[0] = token; - buffer.set(new Uint8Array(tokenArray.buffer, 0, 4), 0); - - var chunk = data.slice(sent, sent + size); - buffer.set(new Uint8Array(chunk), 4); - sent += size; - - dataChannel.send(buffer); - } - } - } else { - Logger.warn("Send data is called for undefined peer connection with stream id: " + streamId); - } - } - - /** - * Called by user - * to add SoundMeter to a stream (remote stream) - * to measure audio level. This sound Meters are added to a map with the key of StreamId. - * When user called @getSoundLevelList, the instant levels are provided. - * - * This list can be used to add a sign to talking participant - * in conference room. And also to determine the dominant audio to focus that player. - * @param {MediaStream} stream - * @param {string} streamId - */ - enableAudioLevel(stream, streamId) { - const soundMeter = new SoundMeter(this.mediaManager.audioContext); - - // Put variables in global scope to make them available to the - // browser console. - // this function fetches getSoundLevelList and this list get instant levels from soundmeter directly - // so we don't need to fill inside of levelCallback here, just pass an empty function - soundMeter.connectToSource(stream, () => {}, function (e) { - if (e) { - alert(e); - return; - } - Logger.debug("Added sound meter for stream: " + streamId + " = " + soundMeter.instant.toFixed(2)); - }); - - this.soundMeters[streamId] = soundMeter; - } - - /** - * Called by the user - * to get the audio levels for the streams for the provided StreamIds - * - * @param {*} streamsList - */ - getSoundLevelList(streamsList) { - for (let i = 0; i < streamsList.length; i++) { - this.soundLevelList[streamsList[i]] = this.soundMeters[streamsList[i]].instant.toFixed(2); - } - this.notifyEventListeners("gotSoundList", this.soundLevelList); - } - - /** - * Called media manaher to get video/audio sender for the local peer connection - * - * @param {string} streamId : - * @param {string} type : "video" or "audio" - * @returns - */ - getSender(streamId, type) { - var sender = null; - if (this.remotePeerConnection[streamId] != null) { - sender = this.remotePeerConnection[streamId].getSenders().find(function (s) { - return s.track.kind == type; - }); - } - return sender; - } - - /** - * Called by user - * - * @param {string} videoTrackId : track id associated with pinned video - * @param {string} streamId : streamId of the pinned video - * @param {boolean} enabled : true | false - * @returns - */ - assignVideoTrack(videoTrackId, streamId, enabled) { - var jsCmd = { - command: "assignVideoTrackCommand", - streamId: streamId, - videoTrackId: videoTrackId, - enabled: enabled, - }; - - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called by user - * video tracks may be less than the participants count - * so these parameters are used for assigning video tracks to participants. - * This message is used to make pagination in conference. - * @param {string} streamId - * @param {number} offset : start index for participant list to play - * @param {number} size : number of the participants to play - * @returns - */ - updateVideoTrackAssignments(streamId, offset, size) { - var jsCmd = { - streamId: streamId, - command: "updateVideoTrackAssignmentsCommand", - offset: offset, - size: size, - }; - - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called by user - * This message is used to set max video track count in a conference. - * @param {string} streamId - * @param {number} maxTrackCount : maximum video track count - * @returns - */ - setMaxVideoTrackCount(streamId, maxTrackCount) { - var jsCmd = { - streamId: streamId, - command: "setMaxVideoTrackCountCommand", - maxTrackCount: maxTrackCount, - }; - - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** - * Called by user - * This message is used to send audio level in a conference. - * + } else if (value.kind == "audio") { + audioJitter = value.jitter; + } + } + } else if (value.type == "media-source") { + if (value.kind == "video") { //returns video source dimensions, not necessarily dimensions being encoded by browser + width = value.width; + height = value.height; + fps = value.framesPerSecond; + } + } + else if (value.type == "candidate-pair" && value.state == "succeeded" && value.availableOutgoingBitrate != undefined) { + availableOutgoingBitrate = value.availableOutgoingBitrate / 1000 + //currentRoundTripTime + currentRoundTripTime = value.currentRoundTripTime; + } + }); + + if (typeof this.remotePeerConnectionStats[streamId] == 'undefined' || this.remotePeerConnectionStats[streamId] == null) { + this.remotePeerConnectionStats[streamId] = new PeerStats(streamId); + } + + this.remotePeerConnectionStats[streamId].totalBytesReceived = bytesReceived; + this.remotePeerConnectionStats[streamId].videoPacketsLost = videoPacketsLost; + this.remotePeerConnectionStats[streamId].audioPacketsLost = audioPacketsLost; + this.remotePeerConnectionStats[streamId].fractionLost = fractionLost; + this.remotePeerConnectionStats[streamId].currentTime = currentTime; + this.remotePeerConnectionStats[streamId].totalBytesSent = bytesSent; + this.remotePeerConnectionStats[streamId].totalVideoPacketsSent = videoPacketsSent; + this.remotePeerConnectionStats[streamId].totalAudioPacketsSent = audioPacketsSent; + this.remotePeerConnectionStats[streamId].videoPacketsSent = videoPacketsSent; + this.remotePeerConnectionStats[streamId].audioPacketsSent = audioPacketsSent; + + this.remotePeerConnectionStats[streamId].audioLevel = audioLevel; + this.remotePeerConnectionStats[streamId].qualityLimitationReason = qlr; + this.remotePeerConnectionStats[streamId].totalFramesEncoded = framesEncoded; + this.remotePeerConnectionStats[streamId].resWidth = width; + this.remotePeerConnectionStats[streamId].resHeight = height; + this.remotePeerConnectionStats[streamId].srcFps = fps; + this.remotePeerConnectionStats[streamId].frameWidth = frameWidth; + this.remotePeerConnectionStats[streamId].frameHeight = frameHeight; + this.remotePeerConnectionStats[streamId].videoRoundTripTime = videoRoundTripTime; + this.remotePeerConnectionStats[streamId].videoJitter = videoJitter; + this.remotePeerConnectionStats[streamId].audioRoundTripTime = audioRoundTripTime; + this.remotePeerConnectionStats[streamId].audioJitter = audioJitter; + this.remotePeerConnectionStats[streamId].framesDecoded = framesDecoded; + this.remotePeerConnectionStats[streamId].framesDropped = framesDropped; + this.remotePeerConnectionStats[streamId].framesReceived = framesReceived; + + this.remotePeerConnectionStats[streamId].videoJitterAverageDelay = videoJitterAverageDelay; + this.remotePeerConnectionStats[streamId].audioJitterAverageDelay = audioJitterAverageDelay; + this.remotePeerConnectionStats[streamId].availableOutgoingBitrate = availableOutgoingBitrate; + + this.remotePeerConnectionStats[streamId].inboundRtpList = inboundRtp; + + this.remotePeerConnectionStats[streamId].currentRoundTripTime = currentRoundTripTime; + this.remotePeerConnectionStats[streamId].audioPacketsReceived = audioPacketsReceived; + this.remotePeerConnectionStats[streamId].videoPacketsReceived = videoPacketsReceived; + + return this.remotePeerConnectionStats[streamId]; + } + + + + /** + * Called to start a periodic timer to get statistics periodically (5 seconds) for a specific stream. + * + * @param {string} streamId : unique id for the stream + * @param {number} periodMs : period in milliseconds. Default value is 5000 ms. + */ + enableStats(streamId, periodMs = 5000) { + if (this.remotePeerConnectionStats[streamId] == null) { + this.remotePeerConnectionStats[streamId] = new PeerStats(streamId); + this.remotePeerConnectionStats[streamId].timerId = setInterval(() => { + this.getStats(streamId); + + }, periodMs); + } + } + + /** + * Called to stop the periodic timer which is set by @enableStats + * + * @param {string} streamId : unique id for the stream + */ + disableStats(streamId) { + if (this.remotePeerConnectionStats[streamId] != null || typeof this.remotePeerConnectionStats[streamId] != 'undefined') { + clearInterval(this.remotePeerConnectionStats[streamId].timerId); + delete this.remotePeerConnectionStats[streamId]; + } + } + + /** + * Called to check and start Web Socket connection if it is not started + */ + checkWebSocketConnection() { + if (this.webSocketAdaptor == null || (this.webSocketAdaptor.isConnected() == false && this.webSocketAdaptor.isConnecting() == false)) { + Logger.debug("websocket url : " + this.websocketURL); + this.webSocketAdaptor = new WebSocketAdaptor({ + websocket_url: this.websocketURL, + webrtcadaptor: this, + callback: (info, obj) => { + if (info == "closed") { + this.reconnectIfRequired(); + } + this.notifyEventListeners(info, obj); + }, + callbackError: (error, message) => { + this.notifyErrorEventListeners(error, message) + }, + debug: this.debug + }); + } + } + + /** + * Called to stop Web Socket connection + * After calling this function, create new WebRTCAdaptor instance, don't use the the same object + * Because all streams are closed on server side as well when websocket connection is closed. + */ + closeWebSocket() { + for (var key in this.remotePeerConnection) { + this.closePeerConnection(key); + } + //free the remote peer connection by initializing again + this.remotePeerConnection = new Array(); + this.webSocketAdaptor.close(); + } + + /** + * @param {string} streamId Called to send a text message to other peer in the peer-to-peer sessionnnection is closed. + * @param {*} definition + * @param {*} data + */ + peerMessage(streamId, definition, data) { + var jsCmd = { + command: "peerMessageCommand", + streamId: streamId, + definition: definition, + data: data, + }; + + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called to force AMS to send the video with the specified resolution in case of Adaptive Streaming (ABR) enabled. + * Normally the resolution is automatically determined by AMS according to the network condition. + * @param {string} streamId : unique id for the stream + * @param {*} resolution : default is auto. You can specify any height value from the ABR list. + */ + forceStreamQuality(streamId, resolution) { + var jsCmd = { + command: "forceStreamQuality", + streamId: streamId, + streamHeight: resolution + }; + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called to send data via DataChannel. DataChannel should be enabled on AMS settings. + * @param {string} streamId : unique id for the stream + * @param {*} data : data that you want to send. It may be a text (may in Json format or not) or binary + */ + sendData(streamId, data) { + var CHUNK_SIZE = 16000; + if (this.remotePeerConnection[streamId] !== undefined) { + var dataChannel = this.remotePeerConnection[streamId].dataChannel; + if (dataChannel === undefined || dataChannel === null || typeof dataChannel === 'undefined') { + Logger.warn('dataChannel is null or undefined'); + return; + } else if (dataChannel.readyState !== 'open') { + Logger.warn('dataChannel.readyState is not open: ' + dataChannel.readyState); + return; + } + var length = data.length || data.size || data.byteLength; + var sent = 0; + + if (typeof data === 'string' || data instanceof String) { + dataChannel.send(data); + } else { + var token = Math.floor(Math.random() * 999999); + let header = new Int32Array(2); + header[0] = token; + header[1] = length; + + dataChannel.send(header); + + var sent = 0; + while (sent < length) { + var size = Math.min(length - sent, CHUNK_SIZE); + var buffer = new Uint8Array(size + 4); + var tokenArray = new Int32Array(1); + tokenArray[0] = token; + buffer.set(new Uint8Array(tokenArray.buffer, 0, 4), 0); + + var chunk = data.slice(sent, sent + size); + buffer.set(new Uint8Array(chunk), 4); + sent += size; + + dataChannel.send(buffer); + } + } + } else { + Logger.warn("Send data is called for undefined peer connection with stream id: " + streamId); + } + } + + /** + * Called by user + * to add SoundMeter to a stream (remote stream) + * to measure audio level. This sound Meters are added to a map with the key of StreamId. + * When user called @getSoundLevelList, the instant levels are provided. + * + * This list can be used to add a sign to talking participant + * in conference room. And also to determine the dominant audio to focus that player. + * @param {MediaStream} stream + * @param {string} streamId + */ + enableAudioLevel(stream, streamId) { + const soundMeter = new SoundMeter(this.mediaManager.audioContext); + + // Put variables in global scope to make them available to the + // browser console. + // this function fetches getSoundLevelList and this list get instant levels from soundmeter directly + // so we don't need to fill inside of levelCallback here, just pass an empty function + soundMeter.connectToSource(stream, () => { }, function(e) { + if (e) { + alert(e); + return; + } + Logger.debug("Added sound meter for stream: " + streamId + " = " + soundMeter.instant.toFixed(2)); + }); + + this.soundMeters[streamId] = soundMeter; + } + + /** + * Called by the user + * to get the audio levels for the streams for the provided StreamIds + * + * @param {*} streamsList + */ + getSoundLevelList(streamsList) { + for (let i = 0; i < streamsList.length; i++) { + this.soundLevelList[streamsList[i]] = this.soundMeters[streamsList[i]].instant.toFixed(2); + } + this.notifyEventListeners("gotSoundList", this.soundLevelList); + } + + /** + * Called media manaher to get video/audio sender for the local peer connection + * + * @param {string} streamId : + * @param {string} type : "video" or "audio" + * @returns + */ + getSender(streamId, type) { + var sender = null; + if (this.remotePeerConnection[streamId] != null) { + sender = this.remotePeerConnection[streamId].getSenders().find(function(s) { + return s.track.kind == type; + }); + } + return sender; + } + + /** + * Called by user + * + * @param {string} videoTrackId : track id associated with pinned video + * @param {string} streamId : streamId of the pinned video + * @param {boolean} enabled : true | false + * @returns + */ + assignVideoTrack(videoTrackId, streamId, enabled) { + var jsCmd = { + command: "assignVideoTrackCommand", + streamId: streamId, + videoTrackId: videoTrackId, + enabled: enabled, + }; + + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called by user + * video tracks may be less than the participants count + * so these parameters are used for assigning video tracks to participants. + * This message is used to make pagination in conference. + * @param {string} streamId + * @param {number} offset : start index for participant list to play + * @param {number} size : number of the participants to play + * @returns + */ + updateVideoTrackAssignments(streamId, offset, size) { + var jsCmd = { + streamId: streamId, + command: "updateVideoTrackAssignmentsCommand", + offset: offset, + size: size, + }; + + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called by user + * This message is used to set max video track count in a conference. + * @param {string} streamId + * @param {number} maxTrackCount : maximum video track count + * @returns + */ + setMaxVideoTrackCount(streamId, maxTrackCount) { + var jsCmd = { + streamId: streamId, + command: "setMaxVideoTrackCountCommand", + maxTrackCount: maxTrackCount, + }; + + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** + * Called by user + * This message is used to send audio level in a conference. + * * IMPORTANT: AMS v2.7+ can get the audio level from the RTP header and sends audio level to the viewers the same way here. - * Just one difference, AMS sends the audio level in the range of 0 and 127. 0 is max, 127 is ms - - * It means that likely you don't need to send UPDATE_AUDIO_LEVEL anymore - * - * @param {string} streamId - * @param {*} value : audio level - * @returns - */ - updateAudioLevel(streamId, value) { - var jsCmd = { - streamId: streamId, - eventType: "UPDATE_AUDIO_LEVEL", - audioLevel: value, - }; - - this.sendData(streamId, JSON.stringify(jsCmd)); - } - - /** - * Called by user - * This message is used to get debug data from server for debugging purposes in conference. - * @param {string} streamId - * @returns - */ - getDebugInfo(streamId) { - var jsCmd = { - streamId: streamId, - command: "getDebugInfo", - }; - - this.webSocketAdaptor.send(JSON.stringify(jsCmd)); - } - - /** + * Just one difference, AMS sends the audio level in the range of 0 and 127. 0 is max, 127 is ms + + * It means that likely you don't need to send UPDATE_AUDIO_LEVEL anymore + * + * @param {string} streamId + * @param {*} value : audio level + * @returns + */ + updateAudioLevel(streamId, value) { + var jsCmd = { + streamId: streamId, + eventType: "UPDATE_AUDIO_LEVEL", + audioLevel: value, + }; + + this.sendData(streamId, JSON.stringify(jsCmd)); + } + + /** + * Called by user + * This message is used to get debug data from server for debugging purposes in conference. + * @param {string} streamId + * @returns + */ + getDebugInfo(streamId) { + var jsCmd = { + streamId: streamId, + command: "getDebugInfo", + }; + + this.webSocketAdaptor.send(JSON.stringify(jsCmd)); + } + + /** * Register user push notification token to Ant Media Server according to subscriberId and authToken * @param {string} subscriberId: subscriber id it can be anything that defines the user * @param {string} authToken: JWT token with the issuer field is the subscriberId and secret is the application's subscriberAuthenticationKey, @@ -2040,8 +2037,8 @@ export class WebRTCAdaptor { }; this.webSocketAdaptor.send(JSON.stringify(jsCmd)); } - - + + /** * Send push notification to subscribers * @param {string} subscriberId: subscriber id it can be anything(email, username, id) that defines the user in your applicaiton @@ -2062,19 +2059,19 @@ export class WebRTCAdaptor { * } */ sendPushNotification(subscriberId, authToken, pushNotificationContent, subscriberIdsToNotify) { - + //type check for pushNotificationContent if json if (typeof pushNotificationContent !== "object") { Logger.error("Push Notification Content is not JSON format"); throw new Error("Push Notification Content is not JSON format"); } - + //type check if subscriberIdsToNotify is array if (!Array.isArray(subscriberIdsToNotify)) { Logger.error("subscriberIdsToNotify is not an array. Please put the subscriber ids to notify in an array such as [user1], [user1, user2]"); throw new Error("subscriberIdsToNotify is not an array. Please put the subscriber ids to notify in an array such as [user1], [user1, user2]"); } - + let jsCmd = { command: "sendPushNotification", subscriberId: subscriberId, @@ -2084,7 +2081,7 @@ export class WebRTCAdaptor { }; this.webSocketAdaptor.send(JSON.stringify(jsCmd)); } - + /** * Send push notification to topic * @param {string} subscriberId: subscriber id it can be anything(email, username, id) that defines the user in your applicaiton @@ -2110,7 +2107,7 @@ export class WebRTCAdaptor { Logger.error("Push Notification Content is not JSON format"); throw new Error("Push Notification Content is not JSON format"); } - + let jsCmd = { command: "sendPushNotification", subscriberId: subscriberId, @@ -2121,169 +2118,169 @@ export class WebRTCAdaptor { this.webSocketAdaptor.send(JSON.stringify(jsCmd)); } - /** - * The following messages are forwarded to MediaManager. They are also kept here because of backward compatibility. - * You can find the details about them in media_manager.js - * @param {string} streamId - * @returns - */ - turnOffLocalCamera(streamId) { - return this.mediaManager.turnOffLocalCamera(streamId); - } - /** - * - * @param {string} streamId - * @returns - */ - turnOnLocalCamera(streamId) { - return this.mediaManager.turnOnLocalCamera(streamId); - } - - muteLocalMic() { - this.mediaManager.muteLocalMic(); - } - - unmuteLocalMic() { - this.mediaManager.unmuteLocalMic(); - } - /** - * - * @param {string} streamId - * @returns - */ - switchDesktopCapture(streamId) { - return this.mediaManager.switchDesktopCapture(streamId); - } - - /** - * Switch to Video camera capture again. Updates the video track on the fly as well. - * @param {string} streamId - * @param {string} deviceId - * @returns {Promise} - */ - switchVideoCameraCapture(streamId, deviceId,onEndedCallback) { - return this.mediaManager.switchVideoCameraCapture(streamId,deviceId,onEndedCallback); - } - - /** - * Update video track of the stream. Updates the video track on the fly as well. - * @param {string} stream - * @param {string} streamId - * @param {function} onEndedCallback - * @param {boolean} stopDesktop - * @returns {Promise} - */ - updateVideoTrack(stream, streamId, onEndedCallback, stopDesktop) { - return this.mediaManager.updateVideoTrack(stream, streamId, onEndedCallback, stopDesktop); - } - - /** - * Update audio track of the stream. Updates the audio track on the fly as well. It just replaces the audio track with the first one in the stream - * @param {*} stream - * @param {*} streamId - * @param {*} onEndedCallback - * @returns - */ - updateAudioTrack(stream, streamId, onEndedCallback) { - return this.mediaManager.updateAudioTrack(stream, streamId, onEndedCallback); - } - - /** - * Called by User - * to switch between front and back camera on mobile devices - * - * @param {string} streamId Id of the stream to be changed. - * @param {string} facingMode it can be ""user" or "environment" - * - * This method is used to switch front and back camera. - */ - switchVideoCameraFacingMode(streamId, facingMode) { - return this.mediaManager.switchVideoCameraFacingMode(streamId, facingMode); - } - /** - * - * @param {string} streamId - * @returns - */ - switchDesktopCaptureWithCamera(streamId) { - return this.mediaManager.switchDesktopCaptureWithCamera(streamId); - } - /** - * - * @param {string} streamId - * @param {string} deviceId - * @returns - */ - switchAudioInputSource(streamId, deviceId) { - return this.mediaManager.switchAudioInputSource(streamId, deviceId); - } - /** - * - * @param {number} volumeLevel - */ - setVolumeLevel(volumeLevel) { - this.mediaManager.setVolumeLevel(volumeLevel); - } - /** - * - * Using sound meter in order to get audio level may cause audio distortion in Windows browsers - * @param {Function} levelCallback - * @param {number} period - * @returns - */ - enableAudioLevelForLocalStream(levelCallback, period) { - return this.mediaManager.enableAudioLevelForLocalStream(levelCallback); - } - - disableAudioLevelForLocalStream() { - this.mediaManager.disableAudioLevelForLocalStream(); - } - /** - * - * @param {object} constraints - * @returns - */ - applyConstraints(constraints) { - return this.mediaManager.applyConstraints(constraints) - }; - - /** - * - * @param {number} bandwidth - * @param {string} streamId - */ - changeBandwidth(bandwidth, streamId) { - this.mediaManager.changeBandwidth(bandwidth, streamId); - } - - enableAudioLevelWhenMuted() { - return this.mediaManager.enableAudioLevelWhenMuted(); - } - - disableAudioLevelWhenMuted() { - this.mediaManager.disableAudioLevelWhenMuted(); - } - /** - * - * @param {string} streamId - * @returns - */ - getVideoSender(streamId) { - return this.mediaManager.getVideoSender(streamId); - } - /** - * - * @param {MediaStreamConstraints} mediaConstraints : media constraints to be used for opening the stream - * @param {string} streamId : id of the stream to replace tracks with - * @returns - */ - openStream(mediaConstraints, streamId) { - return this.mediaManager.openStream(mediaConstraints, streamId); - } - - closeStream() { - return this.mediaManager.closeStream(); - }; + /** + * The following messages are forwarded to MediaManager. They are also kept here because of backward compatibility. + * You can find the details about them in media_manager.js + * @param {string} streamId + * @returns + */ + turnOffLocalCamera(streamId) { + return this.mediaManager.turnOffLocalCamera(streamId); + } + /** + * + * @param {string} streamId + * @returns + */ + turnOnLocalCamera(streamId) { + return this.mediaManager.turnOnLocalCamera(streamId); + } + + muteLocalMic() { + this.mediaManager.muteLocalMic(); + } + + unmuteLocalMic() { + this.mediaManager.unmuteLocalMic(); + } + /** + * + * @param {string} streamId + * @returns + */ + switchDesktopCapture(streamId) { + return this.mediaManager.switchDesktopCapture(streamId); + } + + /** + * Switch to Video camera capture again. Updates the video track on the fly as well. + * @param {string} streamId + * @param {string} deviceId + * @returns {Promise} + */ + switchVideoCameraCapture(streamId, deviceId, onEndedCallback) { + return this.mediaManager.switchVideoCameraCapture(streamId, deviceId, onEndedCallback); + } + + /** + * Update video track of the stream. Updates the video track on the fly as well. + * @param {string} stream + * @param {string} streamId + * @param {function} onEndedCallback + * @param {boolean} stopDesktop + * @returns {Promise} + */ + updateVideoTrack(stream, streamId, onEndedCallback, stopDesktop) { + return this.mediaManager.updateVideoTrack(stream, streamId, onEndedCallback, stopDesktop); + } + + /** + * Update audio track of the stream. Updates the audio track on the fly as well. It just replaces the audio track with the first one in the stream + * @param {*} stream + * @param {*} streamId + * @param {*} onEndedCallback + * @returns + */ + updateAudioTrack(stream, streamId, onEndedCallback) { + return this.mediaManager.updateAudioTrack(stream, streamId, onEndedCallback); + } + + /** + * Called by User + * to switch between front and back camera on mobile devices + * + * @param {string} streamId Id of the stream to be changed. + * @param {string} facingMode it can be ""user" or "environment" + * + * This method is used to switch front and back camera. + */ + switchVideoCameraFacingMode(streamId, facingMode) { + return this.mediaManager.switchVideoCameraFacingMode(streamId, facingMode); + } + /** + * + * @param {string} streamId + * @returns + */ + switchDesktopCaptureWithCamera(streamId) { + return this.mediaManager.switchDesktopCaptureWithCamera(streamId); + } + /** + * + * @param {string} streamId + * @param {string} deviceId + * @returns + */ + switchAudioInputSource(streamId, deviceId) { + return this.mediaManager.switchAudioInputSource(streamId, deviceId); + } + /** + * + * @param {number} volumeLevel + */ + setVolumeLevel(volumeLevel) { + this.mediaManager.setVolumeLevel(volumeLevel); + } + /** + * + * Using sound meter in order to get audio level may cause audio distortion in Windows browsers + * @param {Function} levelCallback + * @param {number} period + * @returns + */ + enableAudioLevelForLocalStream(levelCallback, period) { + return this.mediaManager.enableAudioLevelForLocalStream(levelCallback); + } + + disableAudioLevelForLocalStream() { + this.mediaManager.disableAudioLevelForLocalStream(); + } + /** + * + * @param {object} constraints + * @returns + */ + applyConstraints(constraints) { + return this.mediaManager.applyConstraints(constraints) + }; + + /** + * + * @param {number} bandwidth + * @param {string} streamId + */ + changeBandwidth(bandwidth, streamId) { + this.mediaManager.changeBandwidth(bandwidth, streamId); + } + + enableAudioLevelWhenMuted() { + return this.mediaManager.enableAudioLevelWhenMuted(); + } + + disableAudioLevelWhenMuted() { + this.mediaManager.disableAudioLevelWhenMuted(); + } + /** + * + * @param {string} streamId + * @returns + */ + getVideoSender(streamId) { + return this.mediaManager.getVideoSender(streamId); + } + /** + * + * @param {MediaStreamConstraints} mediaConstraints : media constraints to be used for opening the stream + * @param {string} streamId : id of the stream to replace tracks with + * @returns + */ + openStream(mediaConstraints, streamId) { + return this.mediaManager.openStream(mediaConstraints, streamId); + } + + closeStream() { + return this.mediaManager.closeStream(); + }; } diff --git a/src/test/js/webrtc_adaptor.test.js b/src/test/js/webrtc_adaptor.test.js index aebec7d5..94a1b018 100644 --- a/src/test/js/webrtc_adaptor.test.js +++ b/src/test/js/webrtc_adaptor.test.js @@ -12,7 +12,7 @@ describe("WebRTCAdaptor", function() { var initialized = false; var currentTest; - + var processStarted = false; beforeEach(function() { @@ -207,10 +207,10 @@ describe("WebRTCAdaptor", function() { }); let webSocketAdaptor = adaptor.webSocketAdaptor; - + webSocketAdaptor.connected = true; webSocketAdaptor.connecting = true; - + expect(webSocketAdaptor.connected).to.be.true; expect(webSocketAdaptor.connecting).to.be.true; // Simulate offline event @@ -1736,94 +1736,77 @@ describe("WebRTCAdaptor", function() { }); - - it("WebRTCGetStats", async function() - { - - const randomAlphaNumeric = length => { - let s = ''; - Array.from({ length }).some(() => { - s += Math.random().toString(36).slice(2); - return s.length >= length; - }); - return s.slice(0, length); - }; - - clock.restore(); - this.timeout(25000); + it("parseStats-publish", async function() { + + //sample publish statistics + var stats = [{ "id": "AP", "timestamp": 1729330520732.352, "type": "media-playout", "kind": "audio", "synthesizedSamplesDuration": 0, "synthesizedSamplesEvents": 0, "totalPlayoutDelay": 0, "totalSamplesCount": 0, "totalSamplesDuration": 0 }, + { "id": "CF4D:0B:4F:4D:8D:38:1C:AC:2A:F3:9C:17:29:92:EE:18:DF:B0:21:35:08:99:93:6D:12:F7:7D:A3:8E:72:8E:B8", "timestamp": 1729330520732.352, "type": "certificate", "base64Certificate": "MIIBFTCBvKADAgECAgh9MrqnGcYy7zAKBggqhkjOPQQDAjARMQ8wDQYDVQQDDAZXZWJSVEMwHhcNMjQxMDE4MDkzMzMxWhcNMjQxMTE4MDkzMzMxWjARMQ8wDQYDVQQDDAZXZWJSVEMwWTATBgcqhkjOPQIBBggqhkjOPQMBBwNCAAR6BNLECVPeybudk4KkGtMX1S3CkYJ+TCOgwtxVBKtGdPL8d4xUhbYC403jHZqrMwtbkY+IH1+uBbfuvz66FDGiMAoGCCqGSM49BAMCA0gAMEUCIQCQ9sr21P+NWMqg9QZthhOvTyhtkGnvDrww7I+ZqBamhgIgfqMp2YhIZC2QTONtIZnSs27vvEPWjomCowBzygeusTA=", "fingerprint": "4D:0B:4F:4D:8D:38:1C:AC:2A:F3:9C:17:29:92:EE:18:DF:B0:21:35:08:99:93:6D:12:F7:7D:A3:8E:72:8E:B8", "fingerprintAlgorithm": "sha-256" }, + { "id": "CFF9:6F:09:D5:F9:01:7F:A3:4F:00:0D:AE:A7:7E:6F:A9:54:C7:67:4C:9E:F7:02:0E:CF:84:44:11:59:71:A7:BE", "timestamp": 1729330520732.352, "type": "certificate", "base64Certificate": "MIIBFzCBvaADAgECAgkAglBApme+IRgwCgYIKoZIzj0EAwIwETEPMA0GA1UEAwwGV2ViUlRDMB4XDTI0MTAxODA5MzMzMFoXDTI0MTExODA5MzMzMFowETEPMA0GA1UEAwwGV2ViUlRDMFkwEwYHKoZIzj0CAQYIKoZIzj0DAQcDQgAEseI9xyxWMKs3R2LGUPjzyYUrm8q2Ev+YhAIftwIYQ0JrE4piYUV+5j8DXJe1ldC5Yd+mlFL1PNi7qRYP9j4cwDAKBggqhkjOPQQDAgNJADBGAiEAotZ2yCw7yVXjqzgmzjm6Rh5zfSQNVqs/rA3385tuaD8CIQDfMC+JI7joQw04rFL0M0ik1YbvrKrNokqKu0N3rPMYVQ==", "fingerprint": "F9:6F:09:D5:F9:01:7F:A3:4F:00:0D:AE:A7:7E:6F:A9:54:C7:67:4C:9E:F7:02:0E:CF:84:44:11:59:71:A7:BE", "fingerprintAlgorithm": "sha-256" }, + { "id": "COT01_111_minptime=10;stereo=1;useinbandfec=1", "timestamp": 1729330520732.352, "type": "codec", "channels": 2, "clockRate": 48000, "mimeType": "audio/opus", "payloadType": 111, "sdpFmtpLine": "minptime=10;stereo=1;useinbandfec=1", "transportId": "T01" }, + { "id": "COT01_96", "timestamp": 1729330520732.352, "type": "codec", "clockRate": 90000, "mimeType": "video/VP8", "payloadType": 96, "transportId": "T01" }, + { "id": "CPXuLks4I7_lSNilYVM", "timestamp": 1729330520732.352, "type": "candidate-pair", "availableOutgoingBitrate": 2563076, "bytesDiscardedOnSend": 0, "bytesReceived": 106219, "bytesSent": 14450360, "consentRequestsSent": 44, "currentRoundTripTime": 0, "lastPacketReceivedTimestamp": 1729330520728, "lastPacketSentTimestamp": 1729330520726, "localCandidateId": "IXuLks4I7", "nominated": true, "packetsDiscardedOnSend": 0, "packetsReceived": 2301, "packetsSent": 18196, "priority": 9115038255631187000, "remoteCandidateId": "IlSNilYVM", "requestsReceived": 47, "requestsSent": 45, "responsesReceived": 45, "responsesSent": 47, "state": "succeeded", "totalRoundTripTime": 0.023, "transportId": "T01", "writable": true }, + { "id": "D1", "timestamp": 1729330520732.352, "type": "data-channel", "bytesReceived": 0, "bytesSent": 0, "dataChannelIdentifier": 1, "label": "streamId_dyvi2Oovo", "messagesReceived": 0, "messagesSent": 0, "protocol": "", "state": "open" }, + { "id": "I9AXE+Zl0", "timestamp": 1729330520732.352, "type": "local-candidate", "address": "31.142.67.147", "candidateType": "srflx", "foundation": "518501176", "ip": "31.142.67.147", "isRemote": false, "networkType": "wifi", "port": 54238, "priority": 1686052607, "protocol": "udp", "relatedAddress": "172.20.10.2", "relatedPort": 53816, "transportId": "T01", "url": "stun:stun1.l.google.com:19302", "usernameFragment": "5Ukp" }, + { "id": "IXuLks4I7", "timestamp": 1729330520732.352, "type": "local-candidate", "address": "172.20.10.2", "candidateType": "host", "foundation": "1579008401", "ip": "172.20.10.2", "isRemote": false, "networkType": "wifi", "port": 53816, "priority": 2122260223, "protocol": "udp", "transportId": "T01", "usernameFragment": "5Ukp" }, + { "id": "IesfU0nCG", "timestamp": 1729330520732.352, "type": "local-candidate", "address": "172.20.10.2", "candidateType": "host", "foundation": "2696353029", "ip": "172.20.10.2", "isRemote": false, "networkType": "wifi", "port": 9, "priority": 1518280447, "protocol": "tcp", "tcpType": "active", "transportId": "T01", "usernameFragment": "5Ukp" }, + { "id": "IlSNilYVM", "timestamp": 1729330520732.352, "type": "remote-candidate", "address": "172.20.10.2", "candidateType": "host", "foundation": "1478312482", "ip": "172.20.10.2", "isRemote": true, "port": 50000, "priority": 2122260223, "protocol": "udp", "transportId": "T01", "usernameFragment": "wtUu" }, + { "id": "OT01A2527777913", "timestamp": 1729330520732.352, "type": "outbound-rtp", "codecId": "COT01_111_minptime=10;stereo=1;useinbandfec=1", "kind": "audio", "mediaType": "audio", "ssrc": 2527777913, "transportId": "T01", "bytesSent": 881500, "packetsSent": 5475, "active": true, "headerBytesSent": 153300, "mediaSourceId": "SA1", "mid": "0", "nackCount": 0, "remoteId": "RIA2527777913", "retransmittedBytesSent": 0, "retransmittedPacketsSent": 0, "targetBitrate": 64000, "totalPacketSendDelay": 0.000051 }, + { "id": "OT01V1354817864", "timestamp": 1729330520732.352, "type": "outbound-rtp", "codecId": "COT01_96", "kind": "video", "mediaType": "video", "ssrc": 1354817864, "transportId": "T01", "bytesSent": 12902907, "packetsSent": 12386, "active": true, "encoderImplementation": "libvpx", "firCount": 0, "frameHeight": 1080, "frameWidth": 1920, "framesEncoded": 4089, "framesPerSecond": 37, "framesSent": 4089, "headerBytesSent": 309481, "hugeFramesSent": 2, "keyFramesEncoded": 2, "mediaSourceId": "SV2", "mid": "1", "nackCount": 0, "pliCount": 0, "powerEfficientEncoder": false, "qpSum": 38909, "qualityLimitationDurations": { "bandwidth": 0, "cpu": 0, "none": 109.53, "other": 0 }, "qualityLimitationReason": "none", "qualityLimitationResolutionChanges": 0, "remoteId": "RIV1354817864", "retransmittedBytesSent": 0, "retransmittedPacketsSent": 0, "rtxSsrc": 2624029362, "scalabilityMode": "L1T1", "targetBitrate": 1200000, "totalEncodeTime": 22.745, "totalEncodedBytesTarget": 0, "totalPacketSendDelay": 1.3370279999999999 }, + { "id": "P", "timestamp": 1729330520732.352, "type": "peer-connection", "dataChannelsClosed": 0, "dataChannelsOpened": 1 }, + { "id": "RIA2527777913", "timestamp": 1729330516942, "type": "remote-inbound-rtp", "codecId": "COT01_111_minptime=10;stereo=1;useinbandfec=1", "kind": "audio", "mediaType": "audio", "ssrc": 2527777913, "transportId": "T01", "jitter": 0, "packetsLost": 0, "fractionLost": 0, "localId": "OT01A2527777913", "roundTripTime": 0.001, "roundTripTimeMeasurements": 22, "totalRoundTripTime": 0.022 }, + { "id": "RIV1354817864", "timestamp": 1729330520576, "type": "remote-inbound-rtp", "codecId": "COT01_96", "kind": "video", "mediaType": "video", "ssrc": 1354817864, "transportId": "T01", "jitter": 0.0006659999999999999, "packetsLost": 0, "fractionLost": 0, "localId": "OT01V1354817864", "roundTripTime": 0.001, "roundTripTimeMeasurements": 106, "totalRoundTripTime": 0.10915 }, + { "id": "SA1", "timestamp": 1729330520732.352, "type": "media-source", "kind": "audio", "trackIdentifier": "c67ebf63-f9dc-43e7-95fd-87ac5ccf6734", "audioLevel": 0.02252265999328593, "totalAudioEnergy": 0.6837220094184183, "totalSamplesDuration": 110.10000000001942 }, + { "id": "SV2", "timestamp": 1729330520732.352, "type": "media-source", "kind": "video", "trackIdentifier": "dc8e387c-57bf-40a3-8b67-8f62216e7c2e", "frames": 4104, "framesPerSecond": 39, "height": 1080, "width": 1920 }, + { "id": "T01", "timestamp": 1729330520732.352, "type": "transport", "bytesReceived": 106219, "bytesSent": 14450360, "dtlsCipher": "TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256", "dtlsRole": "server", "dtlsState": "connected", "iceLocalUsernameFragment": "5Ukp", "iceRole": "controlling", "iceState": "connected", "localCertificateId": "CFF9:6F:09:D5:F9:01:7F:A3:4F:00:0D:AE:A7:7E:6F:A9:54:C7:67:4C:9E:F7:02:0E:CF:84:44:11:59:71:A7:BE", "packetsReceived": 2301, "packetsSent": 18196, "remoteCertificateId": "CF4D:0B:4F:4D:8D:38:1C:AC:2A:F3:9C:17:29:92:EE:18:DF:B0:21:35:08:99:93:6D:12:F7:7D:A3:8E:72:8E:B8", "selectedCandidatePairChanges": 1, "selectedCandidatePairId": "CPXuLks4I7_lSNilYVM", "srtpCipher": "AES_CM_128_HMAC_SHA1_80", "tlsVersion": "FEFD" }]; + var websocketURL = "wss://test.antmedia.io/live/websocket"; processStarted = false; initialized = false; var adaptor = new WebRTCAdaptor({ - websocketURL: websocketURL, - callback: (info, obj) => { - console.log("callback info: " + info); - if (info == "initialized") { - initialized = true; - } - else if (info == "publish_started") { - console.log("publish started"); - processStarted = true; - } - else if (info == "publish_finished") { - console.log("publish finished") - } - }, - }); - - await new Promise((resolve, reject)=>{ - setTimeout(()=> { - resolve(); - }, 5000); - }); - - expect(initialized).to.be.true; - - var streamId = "stream1desadafg23424" + randomAlphaNumeric(24); + websocketURL: websocketURL, + callback: (info, obj) => { + console.log("callback info: " + info); + if (info == "initialized") { + initialized = true; + } + }, + }); - adaptor.publish(streamId); - await new Promise((resolve, reject)=>{ + var streamId = "stream1"; - setTimeout(()=> { - expect(processStarted).to.be.true; - resolve(); - }, 5000); - }); - //getStats - var peerStats = await adaptor.getStats(streamId); - + var peerStats = adaptor.parseStats(stats, streamId); + console.log("publish peerStats: " + JSON.stringify(peerStats)); expect(peerStats.streamId).to.be.equal(streamId); - expect(peerStats.audioPacketsSent).to.be.least(0); - expect(peerStats.videoPacketsSent).to.be.least(0); - expect(peerStats.frameWidth).to.be.least(0); - expect(peerStats.frameHeight).to.be.least(0); - expect(peerStats.currentRoundTripTime).to.be.above(0); - expect(peerStats.currentRoundTripTime).to.be.most(1); + expect(peerStats.audioPacketsSent).to.be.equal(5475); + expect(peerStats.videoPacketsSent).to.be.equal(12386); + expect(peerStats.frameWidth).to.be.equal(1920); + expect(peerStats.frameHeight).to.be.equal(1080); + expect(peerStats.currentRoundTripTime).to.be.least(0); expect(peerStats.videoPacketsLost).to.be.least(0); expect(peerStats.audioPacketsLost).to.be.least(0); - expect(peerStats.videoJitter).to.be.least(0); - expect(peerStats.audioJitter).to.be.least(0); - expect(peerStats.totalBytesSentCount).to.be.above(0); - expect(peerStats.lastFramesEncoded).to.be.above(0); - expect(peerStats.totalFramesEncodedCount).to.be.above(0); - expect(peerStats.frameWidth).to.be.equal(640); - expect(peerStats.frameHeight).to.be.equal(480); + expect(peerStats.videoJitter).to.be.least(0.0006); + expect(peerStats.audioJitter).to.be.equal(0); + expect(peerStats.totalBytesSentCount).to.be.equal(13784406); + expect(peerStats.lastFramesEncoded).to.be.equal(4088); + expect(peerStats.totalFramesEncodedCount).to.be.equal(4088); + expect(peerStats.frameWidth).to.be.equal(1920); + expect(peerStats.frameHeight).to.be.equal(1080); expect(peerStats.qualityLimitationReason).to.be.equal("none"); expect(peerStats.firstByteSentCount).to.be.not.equal(0); - expect(peerStats.srcFps).to.be.above(0); - expect(peerStats.videoRoundTripTime).to.be.above(0); + expect(peerStats.srcFps).to.be.equal(39); + expect(peerStats.videoRoundTripTime).to.be.equal(0.001); //expect(peerStats.audioRoundTripTime).to.be.above(0); - expect(peerStats.availableOutgoingBitrate).to.be.above(0); + expect(peerStats.availableOutgoingBitrate).to.be.equal(2563.076); + + - - expect(peerStats.totalBytesReceivedCount).to.be.equal(-1); expect(peerStats.lastBytesSent).to.be.equal(0); expect(peerStats.videoPacketsLost).to.be.equal(0); @@ -1835,8 +1818,8 @@ describe("WebRTCAdaptor", function() { expect(peerStats.timerId).to.be.equal(0); expect(peerStats.firstBytesReceivedCount).to.be.equal(-1); expect(peerStats.audioLevel).to.be.equal(-1); - expect(peerStats.resWidth).to.be.equal(640); - expect(peerStats.resHeight).to.be.equal(480); + expect(peerStats.resWidth).to.be.equal(1920); + expect(peerStats.resHeight).to.be.equal(1080); expect(peerStats.framesReceived).to.be.equal(-1); expect(peerStats.framesDropped).to.be.equal(-1); expect(peerStats.framesDecoded).to.be.equal(-1); @@ -1846,54 +1829,50 @@ describe("WebRTCAdaptor", function() { expect(peerStats.audioPacketsReceived).to.be.equal(-1); expect(peerStats.videoPacketsReceived).to.be.equal(-1); - //getStats - processStarted = false; - initialized = false; + }); + + it("parseStats-play", async function() { + + var stats = [{ "id": "AP", "timestamp": 1729360465698.781, "type": "media-playout", "kind": "audio", "synthesizedSamplesDuration": 0, "synthesizedSamplesEvents": 0, "totalPlayoutDelay": 174058.96224, "totalSamplesCount": 3614400, "totalSamplesDuration": 75.3 }, + { "id": "CF48:5F:93:78:DB:03:CB:A8:ED:6E:0C:52:34:00:55:80:50:0B:7B:73:3C:AB:F1:C9:15:63:59:E2:8E:7B:09:DD", "timestamp": 1729360465698.781, "type": "certificate", "base64Certificate": "MIIBFzCBvaADAgECAgkApIGSBIC4JckwCgYIKoZIzj0EAwIwETEPMA0GA1UEAwwGV2ViUlRDMB4XDTI0MTAxODE3NTMxMFoXDTI0MTExODE3NTMxMFowETEPMA0GA1UEAwwGV2ViUlRDMFkwEwYHKoZIzj0CAQYIKoZIzj0DAQcDQgAEuAAt92Y89uymKvP0E2sc8vA8IAD2YrPMol4uO8VSsFZIevCMoBGcSkgFbDyqsV4zzEmNA9uyp2Qr3njOI/2lIDAKBggqhkjOPQQDAgNJADBGAiEAj0S++4Go0R6pbocel9F3AevVIRcBFERHQ/JbsDrRCEACIQCfHKdmFN6dl+7vBW1VXl1qqD9dhSRtl2sFo1knPYd6tA==", "fingerprint": "48:5F:93:78:DB:03:CB:A8:ED:6E:0C:52:34:00:55:80:50:0B:7B:73:3C:AB:F1:C9:15:63:59:E2:8E:7B:09:DD", "fingerprintAlgorithm": "sha-256" }, + { "id": "CFDB:33:70:CB:A5:84:C4:9C:65:2E:7C:D9:61:87:5D:09:BF:A4:C2:04:CB:AB:CC:C6:AA:D9:57:D8:C5:1E:D8:E6", "timestamp": 1729360465698.781, "type": "certificate", "base64Certificate": "MIIBFjCBvaADAgECAgkAyueWBvMtqW4wCgYIKoZIzj0EAwIwETEPMA0GA1UEAwwGV2ViUlRDMB4XDTI0MTAxODE3NTMxMFoXDTI0MTExODE3NTMxMFowETEPMA0GA1UEAwwGV2ViUlRDMFkwEwYHKoZIzj0CAQYIKoZIzj0DAQcDQgAEb3mgpzNXygFciDx+zpI3FhEMWEVExL6L03ZF6q9dwzEErPo7cmileEz/h+57fWKHuImu4dw+oyHKRU30PHP0oTAKBggqhkjOPQQDAgNIADBFAiB+umpKbmN4F0iLvMQX9wXrxqwdOorxC/ADn6dWh2q/dQIhAJE4axj9umROWU9phNlMcU2AkxPSDqjVM/hvaV84YSzA", "fingerprint": "DB:33:70:CB:A5:84:C4:9C:65:2E:7C:D9:61:87:5D:09:BF:A4:C2:04:CB:AB:CC:C6:AA:D9:57:D8:C5:1E:D8:E6", "fingerprintAlgorithm": "sha-256" }, + { "id": "CIT01_111_minptime=10;stereo=1;useinbandfec=1", "timestamp": 1729360465698.781, "type": "codec", "channels": 2, "clockRate": 48000, "mimeType": "audio/opus", "payloadType": 111, "sdpFmtpLine": "minptime=10;stereo=1;useinbandfec=1", "transportId": "T01" }, + { "id": "CIT01_127_level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f", "timestamp": 1729360465698.781, "type": "codec", "clockRate": 90000, "mimeType": "video/H264", "payloadType": 127, "sdpFmtpLine": "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f", "transportId": "T01" }, + { "id": "CPdtvDwfjB_kGw81ZH2", "timestamp": 1729360465698.781, "type": "candidate-pair", "availableOutgoingBitrate": 300000, "bytesDiscardedOnSend": 0, "bytesReceived": 1475790, "bytesSent": 61385, "consentRequestsSent": 31, "currentRoundTripTime": 0, "lastPacketReceivedTimestamp": 1729360465684, "lastPacketSentTimestamp": 1729360465697, "localCandidateId": "IdtvDwfjB", "nominated": true, "packetsDiscardedOnSend": 0, "packetsReceived": 7261, "packetsSent": 1340, "priority": 9115038255631187000, "remoteCandidateId": "IkGw81ZH2", "requestsReceived": 34, "requestsSent": 32, "responsesReceived": 32, "responsesSent": 34, "state": "succeeded", "totalRoundTripTime": 0.011, "transportId": "T01", "writable": true }, + { "id": "D3", "timestamp": 1729360465698.781, "type": "data-channel", "bytesReceived": 5355, "bytesSent": 0, "dataChannelIdentifier": 1, "label": "stream1", "messagesReceived": 40, "messagesSent": 0, "protocol": "", "state": "open" }, + { "id": "IT01A3202491249", "timestamp": 1729360465698.781, "type": "inbound-rtp", "codecId": "CIT01_111_minptime=10;stereo=1;useinbandfec=1", "kind": "audio", "mediaType": "audio", "ssrc": 3202491249, "transportId": "T01", "jitter": 0, "packetsLost": 0, "packetsReceived": 3764, "audioLevel": 0.0076906643879512925, "bytesReceived": 482689, "concealedSamples": 1774, "concealmentEvents": 2, "estimatedPlayoutTimestamp": 3938327816566, "fecPacketsDiscarded": 0, "fecPacketsReceived": 0, "headerBytesReceived": 105392, "insertedSamplesForDeceleration": 2862, "jitterBufferDelay": 113116.8, "jitterBufferEmittedCount": 3612480, "jitterBufferMinimumDelay": 76147.2, "jitterBufferTargetDelay": 76300.8, "lastPacketReceivedTimestamp": 1729339016620.57, "mid": "1", "packetsDiscarded": 0, "playoutId": "AP", "remoteId": "ROA3202491249", "removedSamplesForAcceleration": 2670, "silentConcealedSamples": 0, "totalAudioEnergy": 0.9479140477134788, "totalProcessingDelay": 108273.23712, "totalSamplesDuration": 75.29000000000161, "totalSamplesReceived": 3613920, "trackIdentifier": "ARDAMSaaudioTrack0" }, + { "id": "IT01V3262821271", "timestamp": 1729360465698.781, "type": "inbound-rtp", "codecId": "CIT01_127_level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f", "kind": "video", "mediaType": "video", "ssrc": 3262821271, "transportId": "T01", "jitter": 0.01, "packetsLost": 0, "packetsReceived": 3346, "bytesReceived": 709312, "estimatedPlayoutTimestamp": 3938338821605, "firCount": 0, "frameHeight": 360, "frameWidth": 640, "framesAssembledFromMultiplePackets": 75, "framesDecoded": 2226, "framesDropped": 0, "framesPerSecond": 29, "framesReceived": 2226, "freezeCount": 0, "headerBytesReceived": 90444, "jitterBufferDelay": 31.145653, "jitterBufferEmittedCount": 2227, "jitterBufferMinimumDelay": 44.558583999999996, "jitterBufferTargetDelay": 44.558583999999996, "keyFramesDecoded": 75, "lastPacketReceivedTimestamp": 1729339016617.0981, "mid": "0", "nackCount": 0, "pauseCount": 0, "pliCount": 0, "remoteId": "ROV3262821271", "retransmittedBytesReceived": 112295, "retransmittedPacketsReceived": 519, "rtxSsrc": 3654956870, "totalAssemblyTime": 0.6750689999999999, "totalDecodeTime": 1.81971, "totalFreezesDuration": 0, "totalInterFrameDelay": 74.305, "totalPausesDuration": 0, "totalProcessingDelay": 33.002987, "totalSquaredInterFrameDelay": 2.597927000000023, "trackIdentifier": "ARDAMSvvideoTrack0" }, + { "id": "IdtvDwfjB", "timestamp": 1729360465698.781, "type": "local-candidate", "address": "192.168.1.31", "candidateType": "host", "foundation": "2770254034", "ip": "192.168.1.31", "isRemote": false, "networkType": "wifi", "port": 54322, "priority": 2122260223, "protocol": "udp", "transportId": "T01", "usernameFragment": "KlG/" }, + { "id": "IkGw81ZH2", "timestamp": 1729360465698.781, "type": "remote-candidate", "address": "192.168.1.31", "candidateType": "host", "foundation": "3335006257", "ip": "192.168.1.31", "isRemote": true, "port": 50001, "priority": 2122260223, "protocol": "udp", "transportId": "T01", "usernameFragment": "+T7W" }, + { "id": "P", "timestamp": 1729360465698.781, "type": "peer-connection", "dataChannelsClosed": 0, "dataChannelsOpened": 1 }, + { "id": "ROA3202491249", "timestamp": 1729339015650, "type": "remote-outbound-rtp", "codecId": "CIT01_111_minptime=10;stereo=1;useinbandfec=1", "kind": "audio", "mediaType": "audio", "ssrc": 3202491249, "transportId": "T01", "bytesSent": 476489, "packetsSent": 3715, "localId": "IT01A3202491249", "remoteTimestamp": 1729339015650, "reportsSent": 17, "roundTripTimeMeasurements": 0, "totalRoundTripTime": 0 }, + { "id": "ROV3262821271", "timestamp": 1729339016116, "type": "remote-outbound-rtp", "codecId": "CIT01_127_level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f", "kind": "video", "mediaType": "video", "ssrc": 3262821271, "transportId": "T01", "bytesSent": 702462, "packetsSent": 3323, "localId": "IT01V3262821271", "remoteTimestamp": 1729339016116, "reportsSent": 84, "roundTripTimeMeasurements": 0, "totalRoundTripTime": 0 }, + { "id": "T01", "timestamp": 1729360465698.781, "type": "transport", "bytesReceived": 1475790, "bytesSent": 61385, "dtlsCipher": "TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256", "dtlsRole": "client", "dtlsState": "connected", "iceLocalUsernameFragment": "KlG/", "iceRole": "controlled", "iceState": "connected", "localCertificateId": "CF48:5F:93:78:DB:03:CB:A8:ED:6E:0C:52:34:00:55:80:50:0B:7B:73:3C:AB:F1:C9:15:63:59:E2:8E:7B:09:DD", "packetsReceived": 7261, "packetsSent": 1340, "remoteCertificateId": "CFDB:33:70:CB:A5:84:C4:9C:65:2E:7C:D9:61:87:5D:09:BF:A4:C2:04:CB:AB:CC:C6:AA:D9:57:D8:C5:1E:D8:E6", "selectedCandidatePairChanges": 1, "selectedCandidatePairId": "CPdtvDwfjB_kGw81ZH2", "srtpCipher": "AES_CM_128_HMAC_SHA1_80", "tlsVersion": "FEFD" }] + + var websocketURL = "wss://test.antmedia.io/live/websocket"; + var playAdaptor = new WebRTCAdaptor({ - websocketURL: websocketURL, - isPlayMode: true, - callback: (info, obj) => { - console.log("callback info: " + info); - if (info == "initialized") { - initialized = true; - } - else if (info == "play_started") { - console.log("play started"); - processStarted = true; - } - else if (info == "play_finished") { - console.log("play finished") - } - }, - }); - await new Promise((resolve, reject)=>{ - setTimeout(()=> { - resolve(); - }, 5000); - }); + websocketURL: websocketURL, + isPlayMode: true, + callback: (info, obj) => { + console.log("callback info: " + info); + if (info == "initialized") { + initialized = true; + } + + }, + }); + + + var streamId = "stream1"; + + var peerStats = playAdaptor.parseStats(stats, streamId); - expect(initialized).to.be.true; - - playAdaptor.play(streamId); - - await new Promise((resolve, reject)=>{ - - setTimeout(()=> { - - resolve(); - }, 5000); - }); - - expect(processStarted).to.be.true; - - peerStats = await playAdaptor.getStats(streamId); - console.log("play peerStats: " + JSON.stringify(peerStats)); expect(peerStats.streamId).to.be.equal(streamId); expect(peerStats.frameWidth).to.be.equal(640); - expect(peerStats.frameHeight).to.be.equal(480); - expect(peerStats.currentRoundTripTime).to.be.above(0); - expect(peerStats.currentRoundTripTime).to.be.most(1); + expect(peerStats.frameHeight).to.be.equal(360); + expect(peerStats.currentRoundTripTime).to.be.equal(0); expect(peerStats.videoPacketsLost).to.be.least(0); expect(peerStats.audioPacketsLost).to.be.least(0); @@ -1902,7 +1881,7 @@ describe("WebRTCAdaptor", function() { expect(peerStats.lastFramesEncoded).to.be.equal(-1); expect(peerStats.totalFramesEncodedCount).to.be.equal(-1); expect(peerStats.frameWidth).to.be.equal(640); - expect(peerStats.frameHeight).to.be.equal(480); + expect(peerStats.frameHeight).to.be.equal(360); expect(peerStats.qualityLimitationReason).to.be.equal(""); expect(peerStats.firstByteSentCount).to.be.not.equal(0); expect(peerStats.srcFps).to.be.equal(-1); @@ -1911,9 +1890,9 @@ describe("WebRTCAdaptor", function() { expect(peerStats.availableOutgoingBitrate).to.be.above(-1); - - - expect(peerStats.totalBytesReceivedCount).to.be.above(0); + + + expect(peerStats.totalBytesReceivedCount).to.be.equal(1192000); expect(peerStats.lastBytesSent).to.be.equal(0); expect(peerStats.videoPacketsLost).to.be.equal(0); //expect(peerStats.fractionLost).to.be.equal(-1); @@ -1926,26 +1905,22 @@ describe("WebRTCAdaptor", function() { expect(peerStats.audioLevel).to.be.equal(-1); expect(peerStats.resWidth).to.be.equal(-1); expect(peerStats.resHeight).to.be.equal(-1); - expect(peerStats.framesReceived).to.be.above(0); - expect(peerStats.framesDropped).to.be.least(0); - expect(peerStats.framesDecoded).to.be.above(0); + expect(peerStats.framesReceived).to.be.equal(2226); + expect(peerStats.framesDropped).to.be.equal(0); + expect(peerStats.framesDecoded).to.be.equal(2226); expect(peerStats.audioJitterAverageDelay).to.be.equal(-1); expect(peerStats.videoJitterAverageDelay).to.be.equal(-1); - expect(peerStats.audioPacketsReceived).to.be.above(0); - expect(peerStats.videoPacketsReceived).to.be.above(0); - - + expect(peerStats.audioPacketsReceived).to.be.equal(3764); + expect(peerStats.videoPacketsReceived).to.be.equal(3346); + + expect(peerStats.totalBytesSentCount).to.be.equal(-1); expect(peerStats.totalAudioPacketsSent).to.be.equal(-1); expect(peerStats.totalVideoPacketsSent).to.be.equal(-1); - - - adaptor.stop(streamId); - - playAdaptor.stop(streamId); - + + }); From 60c08092c8d95c2dd01b8e4d96cbfd7ca7fa6086 Mon Sep 17 00:00:00 2001 From: mekya Date: Sat, 19 Oct 2024 21:21:40 +0300 Subject: [PATCH 25/34] Increase timeout value for test stability --- src/test/js/webrtc_adaptor.test.js | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/test/js/webrtc_adaptor.test.js b/src/test/js/webrtc_adaptor.test.js index 94a1b018..c90bec64 100644 --- a/src/test/js/webrtc_adaptor.test.js +++ b/src/test/js/webrtc_adaptor.test.js @@ -556,7 +556,7 @@ describe("WebRTCAdaptor", function() { }); it("testSoundMeter", function(done) { - this.timeout(5000); + this.timeout(15000); console.log("Starting testSoundMeter"); var adaptor = new WebRTCAdaptor({ From 4a659a544b09b8ec9fa44a085affea4c976d7b7d Mon Sep 17 00:00:00 2001 From: mekya Date: Sat, 19 Oct 2024 21:26:18 +0300 Subject: [PATCH 26/34] Remove disabling soundMeter --- src/test/js/webrtc_adaptor.test.js | 1 - 1 file changed, 1 deletion(-) diff --git a/src/test/js/webrtc_adaptor.test.js b/src/test/js/webrtc_adaptor.test.js index c90bec64..3dd975a0 100644 --- a/src/test/js/webrtc_adaptor.test.js +++ b/src/test/js/webrtc_adaptor.test.js @@ -586,7 +586,6 @@ describe("WebRTCAdaptor", function() { adaptor.enableAudioLevelForLocalStream((level) => { console.log("sound level -> " + level); if (level > 0) { - adaptor.disableAudioLevelForLocalStream(); done(); } }); From b60ccad365cd0878a3562c74b6bf614e77cbb4fe Mon Sep 17 00:00:00 2001 From: mekya Date: Sat, 19 Oct 2024 21:57:14 +0300 Subject: [PATCH 27/34] Make the files served:true for karma --- karma.conf.cjs | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/karma.conf.cjs b/karma.conf.cjs index 0798de18..25b7535b 100644 --- a/karma.conf.cjs +++ b/karma.conf.cjs @@ -3,7 +3,7 @@ module.exports = function(config) { frameworks: ['mocha', 'chai', 'sinon'], files: [ { pattern: "src/test/js/**/*.js", type: "module" }, - { pattern: "src/main/js/**/*.js", included: false } + { pattern: "src/main/js/**/*.js", included: false, served:true } ], From d1025d9dfc702bb9bacc9052c60f412b3f12aadd Mon Sep 17 00:00:00 2001 From: mekya Date: Sat, 19 Oct 2024 22:17:15 +0300 Subject: [PATCH 28/34] Set proxy for volume-meter-processor.js --- karma.conf.cjs | 4 ++++ src/test/js/webrtc_adaptor.test.js | 2 +- 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/karma.conf.cjs b/karma.conf.cjs index 25b7535b..83471d4b 100644 --- a/karma.conf.cjs +++ b/karma.conf.cjs @@ -6,6 +6,10 @@ module.exports = function(config) { { pattern: "src/main/js/**/*.js", included: false, served:true } ], + + proxies: { + '/volume-meter-processor.js': '/base/src/main/js/volume-meter-processor.js' + }, reporters: ['progress', 'coverage'], diff --git a/src/test/js/webrtc_adaptor.test.js b/src/test/js/webrtc_adaptor.test.js index 3dd975a0..a0e4f0a6 100644 --- a/src/test/js/webrtc_adaptor.test.js +++ b/src/test/js/webrtc_adaptor.test.js @@ -566,7 +566,7 @@ describe("WebRTCAdaptor", function() { audio: true }, initializeComponents: false, - volumeMeterUrl: 'base/src/main/js/volume-meter-processor.js', + volumeMeterUrl: '/volume-meter-processor.js', }); //fake stream in te browser is a period audio and silence, so getting sound level more than 0 requires From bac9f8bb637c7d36de60ca297c259cb3ad67b103 Mon Sep 17 00:00:00 2001 From: mekya Date: Sat, 19 Oct 2024 22:21:35 +0300 Subject: [PATCH 29/34] Only run testSoundMeter to debug --- src/test/js/webrtc_adaptor.test.js | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/test/js/webrtc_adaptor.test.js b/src/test/js/webrtc_adaptor.test.js index a0e4f0a6..ae0172f9 100644 --- a/src/test/js/webrtc_adaptor.test.js +++ b/src/test/js/webrtc_adaptor.test.js @@ -555,7 +555,7 @@ describe("WebRTCAdaptor", function() { await adaptor.updateAudioTrack(stream, null, null); }); - it("testSoundMeter", function(done) { + it.only("testSoundMeter", function(done) { this.timeout(15000); console.log("Starting testSoundMeter"); @@ -566,7 +566,7 @@ describe("WebRTCAdaptor", function() { audio: true }, initializeComponents: false, - volumeMeterUrl: '/volume-meter-processor.js', + volumeMeterUrl: 'volume-meter-processor.js', }); //fake stream in te browser is a period audio and silence, so getting sound level more than 0 requires From 405ba257314be610c638fed4ddad92b04c382f42 Mon Sep 17 00:00:00 2001 From: mekya Date: Sat, 19 Oct 2024 22:23:07 +0300 Subject: [PATCH 30/34] Run all tests again --- src/test/js/webrtc_adaptor.test.js | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/test/js/webrtc_adaptor.test.js b/src/test/js/webrtc_adaptor.test.js index ae0172f9..4603d54f 100644 --- a/src/test/js/webrtc_adaptor.test.js +++ b/src/test/js/webrtc_adaptor.test.js @@ -555,7 +555,7 @@ describe("WebRTCAdaptor", function() { await adaptor.updateAudioTrack(stream, null, null); }); - it.only("testSoundMeter", function(done) { + it("testSoundMeter", function(done) { this.timeout(15000); console.log("Starting testSoundMeter"); From 2ea22d1ac063b71a220378633ac0d95bb3f8068a Mon Sep 17 00:00:00 2001 From: mekya Date: Sat, 19 Oct 2024 22:29:40 +0300 Subject: [PATCH 31/34] Add slash to access volume-meter-processor --- src/test/js/webrtc_adaptor.test.js | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/test/js/webrtc_adaptor.test.js b/src/test/js/webrtc_adaptor.test.js index 4603d54f..a0e4f0a6 100644 --- a/src/test/js/webrtc_adaptor.test.js +++ b/src/test/js/webrtc_adaptor.test.js @@ -566,7 +566,7 @@ describe("WebRTCAdaptor", function() { audio: true }, initializeComponents: false, - volumeMeterUrl: 'volume-meter-processor.js', + volumeMeterUrl: '/volume-meter-processor.js', }); //fake stream in te browser is a period audio and silence, so getting sound level more than 0 requires From 9c00e560594b3b0880de9eeb4e5be639879655d4 Mon Sep 17 00:00:00 2001 From: mekya Date: Sat, 19 Oct 2024 22:31:25 +0300 Subject: [PATCH 32/34] Change the location of testSoundMeter --- src/test/js/webrtc_adaptor.test.js | 78 +++++++++++++++--------------- 1 file changed, 39 insertions(+), 39 deletions(-) diff --git a/src/test/js/webrtc_adaptor.test.js b/src/test/js/webrtc_adaptor.test.js index a0e4f0a6..113995a9 100644 --- a/src/test/js/webrtc_adaptor.test.js +++ b/src/test/js/webrtc_adaptor.test.js @@ -32,6 +32,45 @@ describe("WebRTCAdaptor", function() { sandbox.restore(); }); + + it("testSoundMeter", function(done) { + this.timeout(15000); + console.log("Starting testSoundMeter"); + + var adaptor = new WebRTCAdaptor({ + websocketURL: "ws://localhost", + mediaConstraints: { + video: true, + audio: true + }, + initializeComponents: false, + volumeMeterUrl: '/volume-meter-processor.js', + }); + + //fake stream in te browser is a period audio and silence, so getting sound level more than 0 requires + + adaptor.initialize().then(() => { + var audioContext = new (window.AudioContext || window.webkitAudioContext)(); + var oscillator = audioContext.createOscillator(); + oscillator.type = "sine"; + oscillator.frequency.value = 800; + var mediaStreamSource = audioContext.createMediaStreamDestination(); + oscillator.connect(mediaStreamSource); + var mediaStreamTrack = mediaStreamSource.stream.getAudioTracks()[0]; + oscillator.start(); + + adaptor.mediaManager.localStream = new MediaStream([mediaStreamTrack]) + adaptor.mediaManager.audioContext = audioContext; + adaptor.enableAudioLevelForLocalStream((level) => { + console.log("sound level -> " + level); + if (level > 0) { + done(); + } + }); + + expect(adaptor.mediaManager.localStreamSoundMeter).to.not.be.null; + }) + }) it("Initialize", async function() { @@ -555,45 +594,6 @@ describe("WebRTCAdaptor", function() { await adaptor.updateAudioTrack(stream, null, null); }); - it("testSoundMeter", function(done) { - this.timeout(15000); - console.log("Starting testSoundMeter"); - - var adaptor = new WebRTCAdaptor({ - websocketURL: "ws://localhost", - mediaConstraints: { - video: true, - audio: true - }, - initializeComponents: false, - volumeMeterUrl: '/volume-meter-processor.js', - }); - - //fake stream in te browser is a period audio and silence, so getting sound level more than 0 requires - - adaptor.initialize().then(() => { - var audioContext = new (window.AudioContext || window.webkitAudioContext)(); - var oscillator = audioContext.createOscillator(); - oscillator.type = "sine"; - oscillator.frequency.value = 800; - var mediaStreamSource = audioContext.createMediaStreamDestination(); - oscillator.connect(mediaStreamSource); - var mediaStreamTrack = mediaStreamSource.stream.getAudioTracks()[0]; - oscillator.start(); - - adaptor.mediaManager.localStream = new MediaStream([mediaStreamTrack]) - adaptor.mediaManager.audioContext = audioContext; - adaptor.enableAudioLevelForLocalStream((level) => { - console.log("sound level -> " + level); - if (level > 0) { - done(); - } - }); - - expect(adaptor.mediaManager.localStreamSoundMeter).to.not.be.null; - }) - }) - it("takeConfiguration", async function() { var adaptor = new WebRTCAdaptor({ From 6b4bb5ec808aa9622796011ae844611510f7c0a3 Mon Sep 17 00:00:00 2001 From: mekya Date: Sat, 19 Oct 2024 22:34:47 +0300 Subject: [PATCH 33/34] Add comment about testSoundMeter --- src/test/js/webrtc_adaptor.test.js | 1 + 1 file changed, 1 insertion(+) diff --git a/src/test/js/webrtc_adaptor.test.js b/src/test/js/webrtc_adaptor.test.js index 113995a9..be34a32d 100644 --- a/src/test/js/webrtc_adaptor.test.js +++ b/src/test/js/webrtc_adaptor.test.js @@ -33,6 +33,7 @@ describe("WebRTCAdaptor", function() { }); + //I've put this test first because it fails when run in the middle, ı thnk one test breaks it. Let's revisit this case in another time - mekya it("testSoundMeter", function(done) { this.timeout(15000); console.log("Starting testSoundMeter"); From bc248f16b5b4bcbc2f7ee164b11e9f737a12e29f Mon Sep 17 00:00:00 2001 From: mekya Date: Wed, 30 Oct 2024 19:49:40 +0300 Subject: [PATCH 34/34] Add option to debug signaling endpoint --- src/main/webapp/player.html | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/src/main/webapp/player.html b/src/main/webapp/player.html index edbfad4e..bcb5e45f 100644 --- a/src/main/webapp/player.html +++ b/src/main/webapp/player.html @@ -181,6 +181,13 @@

} }); + var signaling = getUrlParameter("signaling"); + if (signaling == "true") { + signaling = true; + } + else { + signaling = false; + } var subscriberId = getUrlParameter("subscriberId"); var subscriberCode = getUrlParameter("subscriberCode"); @@ -327,6 +334,11 @@

if (location.protocol.startsWith("https")) { websocketURL = "wss://" + path; } + + if (signaling) { + //add signaling suffix to the websocket URL + websocketURL += "/signaling"; + } const ctx = document.getElementById('fpsChart').getContext('2d'); const fpsChart = new Chart(ctx, {