mirror of
https://github.com/simplex-chat/simplex-chat.git
synced 2025-06-28 20:29:53 +00:00
54020250dc
* ios: native WebRTC * add video showing * make async function better working with main thread * wrapped code in main actor, just in case * small change * a little better * enable relay * removed unused code * allow switching calls * testing * enable encryption * testing more * another test * one more test * fix remote unencrypted video * deleted unused code related to PixelBuffer * added MediaEncryption playground * better playground * better playground * fixes * use new encryption api * media encryption works * small changes * added lib dependency * use commit reference for lib instead of version * video format, PIP size * remove sample.js --------- Co-authored-by: Evgeny Poberezkin <2769109+epoberezkin@users.noreply.github.com>
714 lines
30 KiB
Swift
714 lines
30 KiB
Swift
//
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// Created by Avently on 09.02.2023.
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// Copyright (c) 2023 SimpleX Chat. All rights reserved.
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//
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import WebRTC
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import LZString
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import SwiftUI
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import SimpleXChat
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final class WebRTCClient: NSObject, RTCVideoViewDelegate, RTCFrameEncryptorDelegate, RTCFrameDecryptorDelegate {
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private static let factory: RTCPeerConnectionFactory = {
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RTCInitializeSSL()
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let videoEncoderFactory = RTCDefaultVideoEncoderFactory()
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let videoDecoderFactory = RTCDefaultVideoDecoderFactory()
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return RTCPeerConnectionFactory(encoderFactory: videoEncoderFactory, decoderFactory: videoDecoderFactory)
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}()
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private static let ivTagBytes: Int = 28
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private static let enableEncryption: Bool = true
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struct Call {
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var connection: RTCPeerConnection
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var iceCandidates: [RTCIceCandidate]
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var localMedia: CallMediaType
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var localCamera: RTCVideoCapturer?
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var localVideoSource: RTCVideoSource?
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var localStream: RTCVideoTrack?
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var remoteStream: RTCVideoTrack?
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var device: AVCaptureDevice.Position = .front
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var aesKey: String?
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var frameEncryptor: RTCFrameEncryptor?
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var frameDecryptor: RTCFrameDecryptor?
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}
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private let rtcAudioSession = RTCAudioSession.sharedInstance()
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private let audioQueue = DispatchQueue(label: "audio")
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private var sendCallResponse: (WVAPIMessage) async -> Void
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private var activeCall: Binding<Call?>
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private var localRendererAspectRatio: Binding<CGFloat?>
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@available(*, unavailable)
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override init() {
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fatalError("Unimplemented")
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}
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required init(_ activeCall: Binding<Call?>, _ sendCallResponse: @escaping (WVAPIMessage) async -> Void, _ localRendererAspectRatio: Binding<CGFloat?>) {
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self.sendCallResponse = sendCallResponse
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self.activeCall = activeCall
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self.localRendererAspectRatio = localRendererAspectRatio
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super.init()
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}
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let defaultIceServers: [WebRTC.RTCIceServer] = [
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WebRTC.RTCIceServer(urlStrings: ["stun:stun.simplex.im:443"]),
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WebRTC.RTCIceServer(urlStrings: ["turn:turn.simplex.im:443?transport=udp"], username: "private", credential: "yleob6AVkiNI87hpR94Z"),
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WebRTC.RTCIceServer(urlStrings: ["turn:turn.simplex.im:443?transport=tcp"], username: "private", credential: "yleob6AVkiNI87hpR94Z"),
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]
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func initializeCall(_ iceServers: [WebRTC.RTCIceServer]?, _ remoteIceCandidates: [RTCIceCandidate], _ mediaType: CallMediaType, _ aesKey: String?, _ relay: Bool?) -> Call {
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let connection = createPeerConnection(iceServers ?? getWebRTCIceServers() ?? defaultIceServers, relay)
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connection.delegate = self
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createAudioSender(connection)
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var localStream: RTCVideoTrack? = nil
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var remoteStream: RTCVideoTrack? = nil
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var localCamera: RTCVideoCapturer? = nil
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var localVideoSource: RTCVideoSource? = nil
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if mediaType == .video {
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(localStream, remoteStream, localCamera, localVideoSource) = createVideoSender(connection)
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}
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var frameEncryptor: RTCFrameEncryptor? = nil
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var frameDecryptor: RTCFrameDecryptor? = nil
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if aesKey != nil {
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let encryptor = RTCFrameEncryptor.init(sizeChange: Int32(WebRTCClient.ivTagBytes))
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encryptor.delegate = self
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frameEncryptor = encryptor
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connection.senders.forEach { $0.setRtcFrameEncryptor(encryptor) }
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let decryptor = RTCFrameDecryptor.init(sizeChange: -Int32(WebRTCClient.ivTagBytes))
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decryptor.delegate = self
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frameDecryptor = decryptor
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// Has no video receiver in outgoing call if applied here, see [peerConnection(_ connection: RTCPeerConnection, didChange newState]
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// connection.receivers.forEach { $0.setRtcFrameDecryptor(decryptor) }
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}
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return Call(
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connection: connection,
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iceCandidates: remoteIceCandidates,
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localMedia: mediaType,
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localCamera: localCamera,
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localVideoSource: localVideoSource,
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localStream: localStream,
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remoteStream: remoteStream,
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aesKey: aesKey,
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frameEncryptor: frameEncryptor,
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frameDecryptor: frameDecryptor
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)
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}
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func createPeerConnection(_ iceServers: [WebRTC.RTCIceServer], _ relay: Bool?) -> RTCPeerConnection {
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let constraints = RTCMediaConstraints(mandatoryConstraints: nil,
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optionalConstraints: ["DtlsSrtpKeyAgreement": kRTCMediaConstraintsValueTrue])
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guard let connection = WebRTCClient.factory.peerConnection(
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with: getCallConfig(iceServers, relay),
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constraints: constraints, delegate: nil
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)
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else {
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fatalError("Unable to create RTCPeerConnection")
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}
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return connection
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}
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func getCallConfig(_ iceServers: [WebRTC.RTCIceServer], _ relay: Bool?) -> RTCConfiguration {
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let config = RTCConfiguration()
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config.iceServers = iceServers
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config.sdpSemantics = .unifiedPlan
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config.continualGatheringPolicy = .gatherContinually
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config.iceTransportPolicy = relay == true ? .relay : .all
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// Allows to wait 30 sec before `failing` connection if the answer from remote side is not received in time
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config.iceInactiveTimeout = 30_000
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return config
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}
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func addIceCandidates(_ connection: RTCPeerConnection, _ remoteIceCandidates: [RTCIceCandidate]) {
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remoteIceCandidates.forEach { candidate in
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connection.add(candidate.toWebRTCCandidate()) { error in
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if let error = error {
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logger.error("Adding candidate error \(error)")
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}
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}
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}
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}
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func sendCallCommand(command: WCallCommand) async {
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var resp: WCallResponse? = nil
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let pc = activeCall.wrappedValue?.connection
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switch command {
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case .capabilities:
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resp = .capabilities(capabilities: CallCapabilities(encryption: WebRTCClient.enableEncryption))
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case let .start(media: media, aesKey, iceServers, relay):
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logger.debug("starting incoming call - create webrtc session")
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if activeCall.wrappedValue != nil { endCall() }
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let encryption = WebRTCClient.enableEncryption
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let call = initializeCall(iceServers?.toWebRTCIceServers(), [], media, encryption ? aesKey : nil, relay)
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activeCall.wrappedValue = call
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call.connection.offer { answer in
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Task {
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let gotCandidates = await self.waitWithTimeout(10_000, stepMs: 1000, until: { self.activeCall.wrappedValue?.iceCandidates.count ?? 0 > 0 })
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if gotCandidates {
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await self.sendCallResponse(.init(
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corrId: nil,
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resp: .offer(
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offer: compressToBase64(input: encodeJSON(CustomRTCSessionDescription(type: answer.type.toSdpType(), sdp: answer.sdp))),
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iceCandidates: compressToBase64(input: encodeJSON(self.activeCall.wrappedValue?.iceCandidates ?? [])),
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capabilities: CallCapabilities(encryption: encryption)
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),
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command: command)
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)
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} else {
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self.endCall()
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}
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}
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}
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case let .offer(offer, iceCandidates, media, aesKey, iceServers, relay):
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if activeCall.wrappedValue != nil {
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resp = .error(message: "accept: call already started")
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} else if !WebRTCClient.enableEncryption && aesKey != nil {
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resp = .error(message: "accept: encryption is not supported")
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} else if let offer: CustomRTCSessionDescription = decodeJSON(decompressFromBase64(input: offer)),
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let remoteIceCandidates: [RTCIceCandidate] = decodeJSON(decompressFromBase64(input: iceCandidates)) {
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let call = initializeCall(iceServers?.toWebRTCIceServers(), remoteIceCandidates, media, WebRTCClient.enableEncryption ? aesKey : nil, relay)
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activeCall.wrappedValue = call
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let pc = call.connection
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if let type = offer.type, let sdp = offer.sdp {
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if (try? await pc.setRemoteDescription(RTCSessionDescription(type: type.toWebRTCSdpType(), sdp: sdp))) != nil {
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pc.answer { answer in
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self.addIceCandidates(pc, remoteIceCandidates)
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// Task {
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// try? await Task.sleep(nanoseconds: 32_000 * 1000000)
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Task {
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await self.sendCallResponse(.init(
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corrId: nil,
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resp: .answer(
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answer: compressToBase64(input: encodeJSON(CustomRTCSessionDescription(type: answer.type.toSdpType(), sdp: answer.sdp))),
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iceCandidates: compressToBase64(input: encodeJSON(call.iceCandidates))
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),
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command: command)
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)
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}
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// }
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}
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} else {
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resp = .error(message: "accept: remote description is not set")
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}
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}
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}
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case let .answer(answer, iceCandidates):
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if pc == nil {
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resp = .error(message: "answer: call not started")
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} else if pc?.localDescription == nil {
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resp = .error(message: "answer: local description is not set")
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} else if pc?.remoteDescription != nil {
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resp = .error(message: "answer: remote description already set")
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} else if let answer: CustomRTCSessionDescription = decodeJSON(decompressFromBase64(input: answer)),
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let remoteIceCandidates: [RTCIceCandidate] = decodeJSON(decompressFromBase64(input: iceCandidates)),
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let type = answer.type, let sdp = answer.sdp,
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let pc = pc {
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if (try? await pc.setRemoteDescription(RTCSessionDescription(type: type.toWebRTCSdpType(), sdp: sdp))) != nil {
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addIceCandidates(pc, remoteIceCandidates)
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resp = .ok
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} else {
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resp = .error(message: "answer: remote description is not set")
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}
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}
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case let .ice(iceCandidates):
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if let pc = pc,
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let remoteIceCandidates: [RTCIceCandidate] = decodeJSON(decompressFromBase64(input: iceCandidates)) {
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addIceCandidates(pc, remoteIceCandidates)
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resp = .ok
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} else {
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resp = .error(message: "ice: call not started")
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}
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case let .media(media, enable):
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if activeCall.wrappedValue == nil {
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resp = .error(message: "media: call not started")
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} else if activeCall.wrappedValue?.localMedia == .audio && media == .video {
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resp = .error(message: "media: no video")
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} else {
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enableMedia(media, enable)
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resp = .ok
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}
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case .end:
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await sendCallResponse(.init(corrId: nil, resp: .ok, command: command))
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endCall()
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}
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if let resp = resp {
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await sendCallResponse(.init(corrId: nil, resp: resp, command: command))
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}
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}
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func enableMedia(_ media: CallMediaType, _ enable: Bool) {
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media == .video ? setVideoEnabled(enable) : setAudioEnabled(enable)
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}
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func addLocalRenderer(_ activeCall: Call, _ renderer: RTCEAGLVideoView) {
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activeCall.localStream?.add(renderer)
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// To get width and height of a frame, see videoView(videoView:, didChangeVideoSize)
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renderer.delegate = self
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}
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func videoView(_ videoView: RTCVideoRenderer, didChangeVideoSize size: CGSize) {
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guard size.height > 0 else { return }
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localRendererAspectRatio.wrappedValue = size.width / size.height
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}
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func frameDecryptor(_ decryptor: RTCFrameDecryptor, mediaType: RTCRtpMediaType, withFrame encrypted: Data) -> Data? {
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guard encrypted.count > 0 else { return nil }
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if var key: [CChar] = activeCall.wrappedValue?.aesKey?.cString(using: .utf8),
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let pointer: UnsafeMutableRawPointer = malloc(encrypted.count) {
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memcpy(pointer, (encrypted as NSData).bytes, encrypted.count)
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let isKeyFrame = encrypted[0] & 1 == 0
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let clearTextBytesSize = mediaType.rawValue == 0 ? 1 : isKeyFrame ? 10 : 3
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logCrypto("decrypt", chat_decrypt_media(&key, pointer.advanced(by: clearTextBytesSize), Int32(encrypted.count - clearTextBytesSize)))
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return Data(bytes: pointer, count: encrypted.count - WebRTCClient.ivTagBytes)
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} else {
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return nil
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}
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}
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func frameEncryptor(_ encryptor: RTCFrameEncryptor, mediaType: RTCRtpMediaType, withFrame unencrypted: Data) -> Data? {
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guard unencrypted.count > 0 else { return nil }
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if var key: [CChar] = activeCall.wrappedValue?.aesKey?.cString(using: .utf8),
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let pointer: UnsafeMutableRawPointer = malloc(unencrypted.count + WebRTCClient.ivTagBytes) {
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memcpy(pointer, (unencrypted as NSData).bytes, unencrypted.count)
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let isKeyFrame = unencrypted[0] & 1 == 0
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let clearTextBytesSize = mediaType.rawValue == 0 ? 1 : isKeyFrame ? 10 : 3
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logCrypto("encrypt", chat_encrypt_media(&key, pointer.advanced(by: clearTextBytesSize), Int32(unencrypted.count + WebRTCClient.ivTagBytes - clearTextBytesSize)))
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return Data(bytes: pointer, count: unencrypted.count + WebRTCClient.ivTagBytes)
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} else {
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return nil
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}
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}
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private func logCrypto(_ op: String, _ r: UnsafeMutablePointer<CChar>?) {
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if let r = r {
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let err = fromCString(r)
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if err != "" {
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logger.error("\(op) error: \(err)")
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// } else {
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// logger.debug("\(op) ok")
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}
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}
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}
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func addRemoteRenderer(_ activeCall: Call, _ renderer: RTCVideoRenderer) {
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activeCall.remoteStream?.add(renderer)
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}
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func startCaptureLocalVideo(_ activeCall: Call) {
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guard
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let capturer = activeCall.localCamera as? RTCCameraVideoCapturer,
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let camera = (RTCCameraVideoCapturer.captureDevices().first { $0.position == activeCall.device })
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else {
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logger.error("Unable to find a camera")
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return
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}
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let supported = RTCCameraVideoCapturer.supportedFormats(for: camera)
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let height: (AVCaptureDevice.Format) -> Int32 = { (format: AVCaptureDevice.Format) in CMVideoFormatDescriptionGetDimensions(format.formatDescription).height }
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let format = supported.first(where: { height($0) == 1280 })
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?? supported.first(where: { height($0) >= 480 && height($0) < 1280 })
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?? supported.first(where: { height($0) > 1280 })
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guard
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let format = format,
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let fps = format.videoSupportedFrameRateRanges.max(by: { $0.maxFrameRate < $1.maxFrameRate })
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else {
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logger.error("Unable to find any format for camera or to choose FPS")
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return
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}
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logger.debug("Format for camera is \(format.description)")
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capturer.stopCapture()
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capturer.startCapture(with: camera,
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format: format,
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fps: Int(min(24, fps.maxFrameRate)))
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}
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private func createAudioSender(_ connection: RTCPeerConnection) {
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let streamId = "stream"
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let audioTrack = createAudioTrack()
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connection.add(audioTrack, streamIds: [streamId])
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}
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private func createVideoSender(_ connection: RTCPeerConnection) -> (RTCVideoTrack?, RTCVideoTrack?, RTCVideoCapturer?, RTCVideoSource?) {
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let streamId = "stream"
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let (localVideoTrack, localCamera, localVideoSource) = createVideoTrack()
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connection.add(localVideoTrack, streamIds: [streamId])
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return (localVideoTrack, connection.transceivers.first { $0.mediaType == .video }?.receiver.track as? RTCVideoTrack, localCamera, localVideoSource)
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}
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private func createAudioTrack() -> RTCAudioTrack {
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let audioConstrains = RTCMediaConstraints(mandatoryConstraints: nil, optionalConstraints: nil)
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let audioSource = WebRTCClient.factory.audioSource(with: audioConstrains)
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let audioTrack = WebRTCClient.factory.audioTrack(with: audioSource, trackId: "audio0")
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return audioTrack
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}
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private func createVideoTrack() -> (RTCVideoTrack, RTCVideoCapturer, RTCVideoSource) {
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let localVideoSource = WebRTCClient.factory.videoSource()
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#if targetEnvironment(simulator)
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let localCamera = RTCFileVideoCapturer(delegate: localVideoSource)
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#else
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let localCamera = RTCCameraVideoCapturer(delegate: localVideoSource)
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#endif
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let localVideoTrack = WebRTCClient.factory.videoTrack(with: localVideoSource, trackId: "video0")
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return (localVideoTrack, localCamera, localVideoSource)
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}
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func endCall() {
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guard let call = activeCall.wrappedValue else { return }
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activeCall.wrappedValue = nil
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call.connection.close()
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call.connection.delegate = nil
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call.frameEncryptor?.delegate = nil
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call.frameDecryptor?.delegate = nil
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audioSessionToDefaults()
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}
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func waitWithTimeout(_ timeoutMs: UInt64, stepMs: UInt64, until success: () -> Bool) async -> Bool {
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let startedAt = DispatchTime.now()
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while !success() && startedAt.uptimeNanoseconds + timeoutMs * 1000000 > DispatchTime.now().uptimeNanoseconds {
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guard let _ = try? await Task.sleep(nanoseconds: stepMs * 1000000) else { break }
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}
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return success()
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}
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}
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extension WebRTC.RTCPeerConnection {
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func mediaConstraints() -> RTCMediaConstraints {
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RTCMediaConstraints(
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mandatoryConstraints: [kRTCMediaConstraintsOfferToReceiveAudio: kRTCMediaConstraintsValueTrue,
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kRTCMediaConstraintsOfferToReceiveVideo: kRTCMediaConstraintsValueTrue],
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optionalConstraints: nil)
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}
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func offer(_ completion: @escaping (_ sdp: RTCSessionDescription) -> Void) {
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offer(for: mediaConstraints()) { (sdp, error) in
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guard let sdp = sdp else {
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return
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}
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self.setLocalDescription(sdp, completionHandler: { (error) in
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completion(sdp)
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})
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}
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}
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func answer(_ completion: @escaping (_ sdp: RTCSessionDescription) -> Void) {
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answer(for: mediaConstraints()) { (sdp, error) in
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guard let sdp = sdp else {
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return
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}
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self.setLocalDescription(sdp, completionHandler: { (error) in
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completion(sdp)
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})
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}
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}
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}
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extension WebRTCClient: RTCPeerConnectionDelegate {
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func peerConnection(_ connection: RTCPeerConnection, didChange stateChanged: RTCSignalingState) {
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logger.debug("Connection new signaling state: \(stateChanged.rawValue)")
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}
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func peerConnection(_ connection: RTCPeerConnection, didAdd stream: RTCMediaStream) {
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logger.debug("Connection did add stream")
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}
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func peerConnection(_ connection: RTCPeerConnection, didRemove stream: RTCMediaStream) {
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logger.debug("Connection did remove stream")
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}
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func peerConnectionShouldNegotiate(_ connection: RTCPeerConnection) {
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logger.debug("Connection should negotiate")
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}
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func peerConnection(_ connection: RTCPeerConnection, didChange newState: RTCIceConnectionState) {
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debugPrint("Connection new connection state: \(newState.toString() ?? "" + newState.rawValue.description) \(connection.receivers)")
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guard let call = activeCall.wrappedValue,
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let connectionStateString = newState.toString(),
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let iceConnectionStateString = connection.iceConnectionState.toString(),
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let iceGatheringStateString = connection.iceGatheringState.toString(),
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let signalingStateString = connection.signalingState.toString()
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else {
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return
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}
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Task {
|
|
await sendCallResponse(.init(
|
|
corrId: nil,
|
|
resp: .connection(state: ConnectionState(
|
|
connectionState: connectionStateString,
|
|
iceConnectionState: iceConnectionStateString,
|
|
iceGatheringState: iceGatheringStateString,
|
|
signalingState: signalingStateString)
|
|
),
|
|
command: nil)
|
|
)
|
|
|
|
switch newState {
|
|
case .checking:
|
|
if let frameDecryptor = activeCall.wrappedValue?.frameDecryptor {
|
|
connection.receivers.forEach { $0.setRtcFrameDecryptor(frameDecryptor) }
|
|
}
|
|
let enableSpeaker: Bool
|
|
switch call.localMedia {
|
|
case .video: enableSpeaker = true
|
|
default: enableSpeaker = false
|
|
}
|
|
setSpeakerEnabledAndConfigureSession(enableSpeaker)
|
|
case .disconnected, .failed: endCall()
|
|
default: do {}
|
|
}
|
|
}
|
|
}
|
|
|
|
func peerConnection(_ connection: RTCPeerConnection, didChange newState: RTCIceGatheringState) {
|
|
logger.debug("connection new gathering state: \(newState.toString() ?? "" + newState.rawValue.description)")
|
|
}
|
|
|
|
func peerConnection(_ connection: RTCPeerConnection, didGenerate candidate: WebRTC.RTCIceCandidate) {
|
|
// logger.debug("Connection generated candidate \(candidate.debugDescription)")
|
|
activeCall.wrappedValue?.iceCandidates.append(candidate.toCandidate(nil, nil, nil))
|
|
}
|
|
|
|
func peerConnection(_ connection: RTCPeerConnection, didRemove candidates: [WebRTC.RTCIceCandidate]) {
|
|
logger.debug("Connection did remove candidates")
|
|
}
|
|
|
|
func peerConnection(_ connection: RTCPeerConnection, didOpen dataChannel: RTCDataChannel) {}
|
|
|
|
func peerConnection(_ connection: RTCPeerConnection,
|
|
didChangeLocalCandidate local: WebRTC.RTCIceCandidate,
|
|
remoteCandidate remote: WebRTC.RTCIceCandidate,
|
|
lastReceivedMs lastDataReceivedMs: Int32,
|
|
changeReason reason: String) {
|
|
// logger.debug("Connection changed candidate \(reason) \(remote.debugDescription) \(remote.description)")
|
|
sendConnectedEvent(connection, local: local, remote: remote)
|
|
}
|
|
|
|
func sendConnectedEvent(_ connection: WebRTC.RTCPeerConnection, local: WebRTC.RTCIceCandidate, remote: WebRTC.RTCIceCandidate) {
|
|
connection.statistics { (stats: RTCStatisticsReport) in
|
|
stats.statistics.values.forEach { stat in
|
|
// logger.debug("Stat \(stat.debugDescription)")
|
|
if stat.type == "candidate-pair", stat.values["state"] as? String == "succeeded",
|
|
let localId = stat.values["localCandidateId"] as? String,
|
|
let remoteId = stat.values["remoteCandidateId"] as? String,
|
|
let localStats = stats.statistics[localId],
|
|
let remoteStats = stats.statistics[remoteId],
|
|
local.sdp.contains("\((localStats.values["ip"] as? String ?? "--")) \((localStats.values["port"] as? String ?? "--"))") &&
|
|
remote.sdp.contains("\((remoteStats.values["ip"] as? String ?? "--")) \((remoteStats.values["port"] as? String ?? "--"))")
|
|
{
|
|
Task {
|
|
await self.sendCallResponse(.init(
|
|
corrId: nil,
|
|
resp: .connected(connectionInfo: ConnectionInfo(
|
|
localCandidate: local.toCandidate(
|
|
RTCIceCandidateType.init(rawValue: localStats.values["candidateType"] as! String),
|
|
localStats.values["protocol"] as? String,
|
|
localStats.values["relayProtocol"] as? String
|
|
),
|
|
remoteCandidate: remote.toCandidate(
|
|
RTCIceCandidateType.init(rawValue: remoteStats.values["candidateType"] as! String),
|
|
remoteStats.values["protocol"] as? String,
|
|
remoteStats.values["relayProtocol"] as? String
|
|
))),
|
|
command: nil)
|
|
)
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
extension WebRTCClient {
|
|
func setAudioEnabled(_ enabled: Bool) {
|
|
setTrackEnabled(RTCAudioTrack.self, enabled)
|
|
}
|
|
|
|
func setSpeakerEnabledAndConfigureSession( _ enabled: Bool) {
|
|
audioQueue.async { [weak self] in
|
|
guard let self = self else { return }
|
|
self.rtcAudioSession.lockForConfiguration()
|
|
defer {
|
|
self.rtcAudioSession.unlockForConfiguration()
|
|
}
|
|
do {
|
|
try self.rtcAudioSession.setCategory(AVAudioSession.Category.playAndRecord.rawValue)
|
|
try self.rtcAudioSession.setMode(AVAudioSession.Mode.voiceChat.rawValue)
|
|
try self.rtcAudioSession.overrideOutputAudioPort(enabled ? .speaker : .none)
|
|
try self.rtcAudioSession.setActive(true)
|
|
} catch let error {
|
|
logger.debug("Error configuring AVAudioSession: \(error)")
|
|
}
|
|
}
|
|
}
|
|
|
|
func audioSessionToDefaults() {
|
|
audioQueue.async { [weak self] in
|
|
guard let self = self else { return }
|
|
self.rtcAudioSession.lockForConfiguration()
|
|
defer {
|
|
self.rtcAudioSession.unlockForConfiguration()
|
|
}
|
|
do {
|
|
try self.rtcAudioSession.setCategory(AVAudioSession.Category.ambient.rawValue)
|
|
try self.rtcAudioSession.setMode(AVAudioSession.Mode.default.rawValue)
|
|
try self.rtcAudioSession.overrideOutputAudioPort(.none)
|
|
try self.rtcAudioSession.setActive(false)
|
|
} catch let error {
|
|
logger.debug("Error configuring AVAudioSession: \(error)")
|
|
}
|
|
}
|
|
}
|
|
|
|
func setVideoEnabled(_ enabled: Bool) {
|
|
setTrackEnabled(RTCVideoTrack.self, enabled)
|
|
}
|
|
|
|
func flipCamera() {
|
|
switch activeCall.wrappedValue?.device {
|
|
case .front: activeCall.wrappedValue?.device = .back
|
|
case .back: activeCall.wrappedValue?.device = .front
|
|
default: ()
|
|
}
|
|
if let call = activeCall.wrappedValue {
|
|
startCaptureLocalVideo(call)
|
|
}
|
|
}
|
|
|
|
private func setTrackEnabled<T: RTCMediaStreamTrack>(_ type: T.Type, _ enabled: Bool) {
|
|
activeCall.wrappedValue?.connection.transceivers
|
|
.compactMap { $0.sender.track as? T }
|
|
.forEach { $0.isEnabled = enabled }
|
|
}
|
|
}
|
|
|
|
struct CustomRTCSessionDescription: Codable {
|
|
public var type: RTCSdpType?
|
|
public var sdp: String?
|
|
}
|
|
|
|
enum RTCSdpType: String, Codable {
|
|
case answer
|
|
case offer
|
|
case pranswer
|
|
case rollback
|
|
}
|
|
|
|
extension RTCIceCandidate {
|
|
func toWebRTCCandidate() -> WebRTC.RTCIceCandidate {
|
|
WebRTC.RTCIceCandidate(
|
|
sdp: candidate,
|
|
sdpMLineIndex: Int32(sdpMLineIndex ?? 0),
|
|
sdpMid: sdpMid
|
|
)
|
|
}
|
|
}
|
|
|
|
extension WebRTC.RTCIceCandidate {
|
|
func toCandidate(_ candidateType: RTCIceCandidateType?, _ protocol: String?, _ relayProtocol: String?) -> RTCIceCandidate {
|
|
RTCIceCandidate(
|
|
candidateType: candidateType,
|
|
protocol: `protocol`,
|
|
relayProtocol: relayProtocol,
|
|
sdpMid: sdpMid,
|
|
sdpMLineIndex: Int(sdpMLineIndex),
|
|
candidate: sdp
|
|
)
|
|
}
|
|
}
|
|
|
|
|
|
extension [RTCIceServer] {
|
|
func toWebRTCIceServers() -> [WebRTC.RTCIceServer] {
|
|
self.map {
|
|
WebRTC.RTCIceServer(
|
|
urlStrings: $0.urls,
|
|
username: $0.username,
|
|
credential: $0.credential
|
|
) }
|
|
}
|
|
}
|
|
|
|
extension RTCSdpType {
|
|
func toWebRTCSdpType() -> WebRTC.RTCSdpType {
|
|
switch self {
|
|
case .answer: return WebRTC.RTCSdpType.answer
|
|
case .offer: return WebRTC.RTCSdpType.offer
|
|
case .pranswer: return WebRTC.RTCSdpType.prAnswer
|
|
case .rollback: return WebRTC.RTCSdpType.rollback
|
|
}
|
|
}
|
|
}
|
|
|
|
extension WebRTC.RTCSdpType {
|
|
func toSdpType() -> RTCSdpType {
|
|
switch self {
|
|
case .answer: return RTCSdpType.answer
|
|
case .offer: return RTCSdpType.offer
|
|
case .prAnswer: return RTCSdpType.pranswer
|
|
case .rollback: return RTCSdpType.rollback
|
|
default: return RTCSdpType.answer // should never be here
|
|
}
|
|
}
|
|
}
|
|
|
|
extension RTCPeerConnectionState {
|
|
func toString() -> String? {
|
|
switch self {
|
|
case .new: return "new"
|
|
case .connecting: return "connecting"
|
|
case .connected: return "connected"
|
|
case .failed: return"failed"
|
|
case .disconnected: return "disconnected"
|
|
case .closed: return "closed"
|
|
default: return nil // unknown
|
|
}
|
|
}
|
|
}
|
|
|
|
extension RTCIceConnectionState {
|
|
func toString() -> String? {
|
|
switch self {
|
|
case .new: return "new"
|
|
case .checking: return "checking"
|
|
case .connected: return "connected"
|
|
case .completed: return "completed"
|
|
case .failed: return "failed"
|
|
case .disconnected: return "disconnected"
|
|
case .closed: return "closed"
|
|
default: return nil // unknown or unused on the other side
|
|
}
|
|
}
|
|
}
|
|
|
|
extension RTCIceGatheringState {
|
|
func toString() -> String? {
|
|
switch self {
|
|
case .new: return "new"
|
|
case .gathering: return "gathering"
|
|
case .complete: return "complete"
|
|
default: return nil // unknown
|
|
}
|
|
}
|
|
}
|
|
|
|
extension RTCSignalingState {
|
|
func toString() -> String? {
|
|
switch self {
|
|
case .stable: return "stable"
|
|
case .haveLocalOffer: return "have-local-offer"
|
|
case .haveLocalPrAnswer: return "have-local-pranswer"
|
|
case .haveRemoteOffer: return "have-remote-offer"
|
|
case .haveRemotePrAnswer: return "have-remote-pranswer"
|
|
case .closed: return "closed"
|
|
default: return nil // unknown
|
|
}
|
|
}
|
|
}
|