From 82ea9a86c21846211550d80561919de4249db40e Mon Sep 17 00:00:00 2001
From: Gagan Suie <gagan.suie@codecrow.io>
Date: Thu, 23 Mar 2023 20:59:18 -0500
Subject: [PATCH] Fix: updated codecov file and added streaming utils

---
 .github/codecov.yml                           |   2 +-
 src/lib/WHEPClient.ts                         |  79 +++++++++++
 src/lib/WHIPClient.ts                         | 134 ++++++++++++++++++
 src/lib/negotiateConnectionWithClientOffer.ts |  84 +++++++++++
 4 files changed, 298 insertions(+), 1 deletion(-)
 create mode 100644 src/lib/WHEPClient.ts
 create mode 100644 src/lib/WHIPClient.ts
 create mode 100644 src/lib/negotiateConnectionWithClientOffer.ts

diff --git a/.github/codecov.yml b/.github/codecov.yml
index 223e1875..dbf7cfc0 100644
--- a/.github/codecov.yml
+++ b/.github/codecov.yml
@@ -1,2 +1,2 @@
 fixes:
-  - "::/mage-website"
+  - "/home/runner/work/mage-website/::"
diff --git a/src/lib/WHEPClient.ts b/src/lib/WHEPClient.ts
new file mode 100644
index 00000000..722e124d
--- /dev/null
+++ b/src/lib/WHEPClient.ts
@@ -0,0 +1,79 @@
+import negotiateConnectionWithClientOffer from '$lib/negotiateConnectionWithClientOffer'
+
+/**
+ * Example implementation of a client that uses WHEP to playback video over WebRTC
+ *
+ * https://www.ietf.org/id/draft-murillo-whep-00.html
+ */
+export default class WHEPClient {
+	private peerConnection: RTCPeerConnection
+	private stream: MediaStream
+
+	constructor(private endpoint: string, private videoElement: HTMLVideoElement) {
+		this.stream = new MediaStream()
+
+		/**
+		 * Create a new WebRTC connection, using public STUN servers with ICE,
+		 * allowing the client to disover its own IP address.
+		 * https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Protocols#ice
+		 */
+		this.peerConnection = new RTCPeerConnection({
+			iceServers: [
+				{
+					urls: 'stun:stun.cloudflare.com:3478'
+				}
+			],
+			bundlePolicy: 'max-bundle'
+		})
+
+		/** https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addTransceiver */
+		this.peerConnection.addTransceiver('video', {
+			direction: 'recvonly'
+		})
+		this.peerConnection.addTransceiver('audio', {
+			direction: 'recvonly'
+		})
+
+		/**
+		 * When new tracks are received in the connection, store local references,
+		 * so that they can be added to a MediaStream, and to the <video> element.
+		 *
+		 * https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/track_event
+		 */
+		this.peerConnection.ontrack = (event) => {
+			const track = event.track
+			const currentTracks = this.stream.getTracks()
+			const streamAlreadyHasVideoTrack = currentTracks.some((track) => track.kind === 'video')
+			const streamAlreadyHasAudioTrack = currentTracks.some((track) => track.kind === 'audio')
+			switch (track.kind) {
+				case 'video':
+					if (streamAlreadyHasVideoTrack) {
+						break
+					}
+					this.stream.addTrack(track)
+					break
+				case 'audio':
+					if (streamAlreadyHasAudioTrack) {
+						break
+					}
+					this.stream.addTrack(track)
+					break
+				default:
+					console.log('got unknown track ' + track)
+			}
+		}
+
+		this.peerConnection.addEventListener('connectionstatechange', (ev) => {
+			if (this.peerConnection.connectionState !== 'connected') {
+				return
+			}
+			if (!this.videoElement.srcObject) {
+				this.videoElement.srcObject = this.stream
+			}
+		})
+
+		this.peerConnection.addEventListener('negotiationneeded', (ev) => {
+			negotiateConnectionWithClientOffer(this.peerConnection, this.endpoint)
+		})
+	}
+}
diff --git a/src/lib/WHIPClient.ts b/src/lib/WHIPClient.ts
new file mode 100644
index 00000000..ed357bb0
--- /dev/null
+++ b/src/lib/WHIPClient.ts
@@ -0,0 +1,134 @@
+import negotiateConnectionWithClientOffer from '$lib/negotiateConnectionWithClientOffer'
+
+/**
+ * Example implementation of a client that uses WHIP to broadcast video over WebRTC
+ *
+ * https://www.ietf.org/archive/id/draft-ietf-wish-whip-01.html
+ */
+export default class WHIPClient {
+	private peerConnection: RTCPeerConnection
+	private localStream?: MediaStream
+
+	constructor(private endpoint: string, private videoElement: HTMLVideoElement, trackType: string) {
+		/**
+		 * Create a new WebRTC connection, using public STUN servers with ICE,
+		 * allowing the client to disover its own IP address.
+		 * https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Protocols#ice
+		 */
+		this.peerConnection = new RTCPeerConnection({
+			iceServers: [
+				{
+					urls: 'stun:stun.cloudflare.com:3478'
+				}
+			],
+			bundlePolicy: 'max-bundle'
+		})
+
+		/**
+		 * Listen for negotiationneeded events, and use WHIP as the signaling protocol to establish a connection
+		 *
+		 * https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/negotiationneeded_event
+		 * https://www.ietf.org/archive/id/draft-ietf-wish-whip-01.html
+		 */
+		this.peerConnection.addEventListener('negotiationneeded', async (ev) => {
+			console.log('Connection negotiation starting')
+			await negotiateConnectionWithClientOffer(this.peerConnection, this.endpoint)
+			console.log('Connection negotiation ended')
+		})
+
+		/**
+		 * While the connection is being initialized,
+		 * connect the video stream to the provided <video> element.
+		 */
+		this.accessLocalMediaSources(trackType)
+			.then((stream) => {
+				this.localStream = stream
+				videoElement.srcObject = stream as MediaProvider
+			})
+			.catch(console.error)
+	}
+
+	/**
+	 * Ask for camera and microphone permissions and
+	 * add video and audio tracks to the peerConnection.
+	 *
+	 * https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
+	 */
+	private async accessLocalMediaSources(trackType: string) {
+		if (trackType === 'screen') {
+			return navigator.mediaDevices.getDisplayMedia({ video: true, audio: true }).then((stream) => {
+				stream.getTracks().forEach((track) => {
+					const transceiver = this.peerConnection.addTransceiver(track, {
+						/** WHIP is only for sending streaming media */
+						direction: 'sendonly'
+					})
+					if (track.kind == 'video' && transceiver.sender.track) {
+						transceiver.sender.track.applyConstraints({
+							width: 1280,
+							height: 720
+						})
+					}
+				})
+				return stream
+			})
+		} else if (trackType === 'webcam') {
+			return navigator.mediaDevices.getUserMedia({ video: true, audio: true }).then((stream) => {
+				stream.getTracks().forEach((track) => {
+					const transceiver = this.peerConnection.addTransceiver(track, {
+						/** WHIP is only for sending streaming media */
+						direction: 'sendonly'
+					})
+					if (track.kind == 'video' && transceiver.sender.track) {
+						transceiver.sender.track.applyConstraints({
+							width: 1280,
+							height: 720
+						})
+					}
+				})
+				return stream
+			})
+		} else if (trackType === 'audio') {
+			return navigator.mediaDevices
+				.getUserMedia({
+					video: false,
+					audio: {
+						echoCancellation: true,
+						noiseSuppression: true,
+						deviceId: 'default'
+					}
+				})
+				.then((stream) => {
+					stream.getTracks().forEach((track) => {
+						const transceiver = this.peerConnection.addTransceiver(track, {
+							/** WHIP is only for sending streaming media */
+							direction: 'sendonly'
+						})
+						if (track.kind == 'video' && transceiver.sender.track) {
+							transceiver.sender.track.applyConstraints({
+								width: 1280,
+								height: 720
+							})
+						}
+					})
+					return stream
+				})
+		}
+	}
+
+	/**
+	 * Terminate the streaming session
+	 * 1. Notify the WHIP server by sending a DELETE request
+	 * 2. Close the WebRTC connection
+	 * 3. Stop using the local camera and microphone
+	 *
+	 * Note that once you call this method, this instance of this WHIPClient cannot be reused.
+	 */
+	public async disconnectStream() {
+		const response = await fetch(this.endpoint, {
+			method: 'DELETE',
+			mode: 'cors'
+		})
+		this.peerConnection.close()
+		this.localStream?.getTracks().forEach((track) => track.stop())
+	}
+}
diff --git a/src/lib/negotiateConnectionWithClientOffer.ts b/src/lib/negotiateConnectionWithClientOffer.ts
new file mode 100644
index 00000000..91f0cfaf
--- /dev/null
+++ b/src/lib/negotiateConnectionWithClientOffer.ts
@@ -0,0 +1,84 @@
+/**
+ * Performs the actual SDP exchange.
+ *
+ * 1. Constructs the client's SDP offer
+ * 2. Sends the SDP offer to the server,
+ * 3. Awaits the server's offer.
+ *
+ * SDP describes what kind of media we can send and how the server and client communicate.
+ *
+ * https://developer.mozilla.org/en-US/docs/Glossary/SDP
+ * https://www.ietf.org/archive/id/draft-ietf-wish-whip-01.html#name-protocol-operation
+ */
+export default async function negotiateConnectionWithClientOffer(
+	peerConnection: RTCPeerConnection,
+	endpoint: string
+) {
+	/** https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/createOffer */
+	const offer = await peerConnection.createOffer()
+	/** https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/setLocalDescription */
+	await peerConnection.setLocalDescription(offer)
+
+	/** Wait for ICE gathering to complete */
+	const ofr = await waitToCompleteICEGathering(peerConnection)
+	if (!ofr) {
+		throw Error('failed to gather ICE candidates for offer')
+	}
+
+	/**
+	 * As long as the connection is open, attempt to...
+	 */
+	while (peerConnection.connectionState !== 'closed') {
+		/**
+		 * This response contains the server's SDP offer.
+		 * This specifies how the client should communicate,
+		 * and what kind of media client and server have negotiated to exchange.
+		 */
+		const response = await postSDPOffer(endpoint, ofr.sdp)
+		if (response.status === 201) {
+			const answerSDP = await response.text()
+			await peerConnection.setRemoteDescription(
+				new RTCSessionDescription({ type: 'answer', sdp: answerSDP })
+			)
+			return response.headers.get('Location')
+		} else if (response.status === 405) {
+			console.log('Remember to update the URL passed into the WHIP or WHEP client')
+		} else {
+			const errorMessage = await response.text()
+			console.error(errorMessage)
+		}
+
+		/** Limit reconnection attempts to at-most once every 5 seconds */
+		await new Promise((r) => setTimeout(r, 5000))
+	}
+}
+
+async function postSDPOffer(endpoint: string, data: string) {
+	return await fetch(endpoint, {
+		method: 'POST',
+		mode: 'cors',
+		headers: {
+			'content-type': 'application/sdp'
+		},
+		body: data
+	})
+}
+
+/**
+ * Receives an RTCPeerConnection and waits until
+ * the connection is initialized or a timeout passes.
+ *
+ * https://www.ietf.org/archive/id/draft-ietf-wish-whip-01.html#section-4.1
+ * https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/iceGatheringState
+ * https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/icegatheringstatechange_event
+ */
+async function waitToCompleteICEGathering(peerConnection: RTCPeerConnection) {
+	return new Promise<RTCSessionDescription | null>((resolve) => {
+		/** Wait at most 1 second for ICE gathering. */
+		setTimeout(function () {
+			resolve(peerConnection.localDescription)
+		}, 1000)
+		peerConnection.onicegatheringstatechange = (ev) =>
+			peerConnection.iceGatheringState === 'complete' && resolve(peerConnection.localDescription)
+	})
+}