276 lines
11 KiB
Swift
276 lines
11 KiB
Swift
//
|
|
// AudioWebRTCClient.swift
|
|
//
|
|
// Created by Tiago Daniel Jacobs on 20/04/22.
|
|
|
|
import Foundation
|
|
import WebRTC
|
|
import os
|
|
|
|
public protocol AudioWebRTCClientDelegate: AnyObject {
|
|
func webRTCClient(_ client: AudioWebRTCClient, didDiscoverLocalCandidate candidate: RTCIceCandidate)
|
|
func webRTCClient(_ client: AudioWebRTCClient, didChangeIceConnectionState state: RTCIceConnectionState)
|
|
func webRTCClient(_ client: AudioWebRTCClient, didChangeIceGatheringState state: RTCIceGatheringState)
|
|
func webRTCClient(_ client: AudioWebRTCClient, didChangeSignalingState state: RTCSignalingState)
|
|
}
|
|
|
|
open class AudioWebRTCClient: NSObject {
|
|
private var logger = os.Logger(subsystem: "BigBlueButtonMobileSDK", category: "AudioWebRTCClient")
|
|
private var iceGatheringComplete:Bool = false
|
|
|
|
// The `RTCPeerConnectionFactory` is in charge of creating new RTCPeerConnection instances.
|
|
// A new RTCPeerConnection should be created every new call, but the factory is shared.
|
|
private static let factory: RTCPeerConnectionFactory = {
|
|
RTCInitializeSSL()
|
|
let videoEncoderFactory = RTCDefaultVideoEncoderFactory()
|
|
let videoDecoderFactory = RTCDefaultVideoDecoderFactory()
|
|
videoEncoderFactory.preferredCodec = RTCVideoCodecInfo(name: kRTCVideoCodecVp8Name)
|
|
return RTCPeerConnectionFactory(encoderFactory: videoEncoderFactory, decoderFactory: videoDecoderFactory)
|
|
}()
|
|
|
|
public weak var delegate: AudioWebRTCClientDelegate?
|
|
private let peerConnection: RTCPeerConnection
|
|
private let rtcAudioSession = RTCAudioSession.sharedInstance()
|
|
private let audioQueue = DispatchQueue(label: "audio")
|
|
private let mediaConstrains = [kRTCMediaConstraintsOfferToReceiveAudio: kRTCMediaConstraintsValueTrue,
|
|
kRTCMediaConstraintsOfferToReceiveVideo: kRTCMediaConstraintsValueFalse]
|
|
private var videoSource: RTCVideoSource?
|
|
private var videoCapturer: RTCVideoCapturer?
|
|
private var localVideoTrack: RTCVideoTrack?
|
|
private var isRatioDefined:Bool=false
|
|
|
|
@available(*, unavailable)
|
|
override init() {
|
|
fatalError("WebRTCClient:init is unavailable")
|
|
}
|
|
|
|
public required init(iceServers: [String]) {
|
|
let config = RTCConfiguration()
|
|
config.iceServers = [RTCIceServer(urlStrings: iceServers)]
|
|
|
|
// Unified plan is more superior than planB
|
|
config.sdpSemantics = .unifiedPlan
|
|
|
|
// gatherContinually will let WebRTC to listen to any network changes and send any new candidates to the other client
|
|
// gatherOnce will get candidates only on beginning (this is how BBB expect to have it for now, so we use this one)
|
|
config.continualGatheringPolicy = .gatherOnce
|
|
|
|
|
|
// Define media constraints. DtlsSrtpKeyAgreement is required to be true to be able to connect with web browsers.
|
|
let constraints = RTCMediaConstraints(mandatoryConstraints: nil,
|
|
optionalConstraints: ["DtlsSrtpKeyAgreement":kRTCMediaConstraintsValueTrue])
|
|
|
|
guard let peerConnection = AudioWebRTCClient.factory.peerConnection(with: config, constraints: constraints, delegate: nil) else {
|
|
fatalError("Could not create new RTCPeerConnection")
|
|
}
|
|
|
|
self.peerConnection = peerConnection
|
|
|
|
super.init()
|
|
createMediaSenders()
|
|
configureAudioSession()
|
|
self.peerConnection.delegate = self
|
|
}
|
|
|
|
// MARK: Signaling
|
|
|
|
public func offer() async throws -> RTCSessionDescription {
|
|
let constrains = RTCMediaConstraints(mandatoryConstraints: self.mediaConstrains, optionalConstraints: nil)
|
|
let sdp = try await self.peerConnection.offer(for: constrains)
|
|
try await self.peerConnection.setLocalDescription(sdp)
|
|
return sdp
|
|
}
|
|
|
|
public func setRemoteSDP(remoteSDP: String) async throws {
|
|
let rtcSessionDescription = RTCSessionDescription(type: RTCSdpType.answer, sdp: remoteSDP)
|
|
try await self.peerConnection.setRemoteDescription(rtcSessionDescription)
|
|
}
|
|
|
|
// public func setRemoteCandidate(remoteIceCandidate: IceCandidate) async throws {
|
|
// let rtcRemoteCandidate = RTCIceCandidate(sdp: remoteIceCandidate.candidate, sdpMLineIndex: remoteIceCandidate.sdpMLineIndex, sdpMid: remoteIceCandidate.sdpMid)
|
|
// try await self.peerConnection.add(rtcRemoteCandidate)
|
|
// }
|
|
|
|
func set(remoteCandidate: RTCIceCandidate, completion: @escaping (Error?) -> ()) {
|
|
self.peerConnection.add(remoteCandidate, completionHandler: completion)
|
|
}
|
|
|
|
// MARK: Media
|
|
|
|
public func push(videoFrame: RTCVideoFrame) {
|
|
guard videoCapturer != nil, videoSource != nil else { return }
|
|
videoSource!.capturer(videoCapturer!, didCapture: videoFrame)
|
|
print("RTCVideoFrame pushed to server.")
|
|
}
|
|
|
|
private func configureAudioSession() {
|
|
self.rtcAudioSession.lockForConfiguration()
|
|
do {
|
|
try self.rtcAudioSession.setCategory(AVAudioSession.Category.playAndRecord.rawValue)
|
|
try self.rtcAudioSession.setMode(AVAudioSession.Mode.voiceChat.rawValue)
|
|
} catch let error {
|
|
debugPrint("Error changing AVAudioSession category: \(error)")
|
|
}
|
|
self.rtcAudioSession.unlockForConfiguration()
|
|
}
|
|
|
|
private func createMediaSenders() {
|
|
let streamId = "stream"
|
|
|
|
// Audio
|
|
let audioTrack = self.createAudioTrack()
|
|
self.peerConnection.add(audioTrack, streamIds: [streamId])
|
|
}
|
|
|
|
private func createAudioTrack() -> RTCAudioTrack {
|
|
let audioConstrains = RTCMediaConstraints(mandatoryConstraints: nil, optionalConstraints: nil)
|
|
let audioSource = AudioWebRTCClient.factory.audioSource(with: audioConstrains)
|
|
let audioTrack = AudioWebRTCClient.factory.audioTrack(with: audioSource, trackId: "audio0")
|
|
return audioTrack
|
|
}
|
|
}
|
|
|
|
// MARK: RTCPeerConnectionDelegate Methods
|
|
|
|
extension AudioWebRTCClient: RTCPeerConnectionDelegate {
|
|
|
|
public func peerConnection(_ peerConnection: RTCPeerConnection, didChange stateChanged: RTCSignalingState) {
|
|
self.logger.info("peerConnection new signaling state: \(stateChanged.rawValue)")
|
|
|
|
self.delegate?.webRTCClient(self, didChangeSignalingState: stateChanged)
|
|
}
|
|
|
|
public func peerConnection(_ peerConnection: RTCPeerConnection, didAdd stream: RTCMediaStream) {
|
|
self.logger.info("peerConnection did add stream \(stream.streamId)")
|
|
}
|
|
|
|
public func peerConnection(_ peerConnection: RTCPeerConnection, didRemove stream: RTCMediaStream) {
|
|
self.logger.info("peerConnection did remove stream \(stream.streamId)")
|
|
}
|
|
|
|
public func peerConnectionShouldNegotiate(_ peerConnection: RTCPeerConnection) {
|
|
self.logger.info("peerConnection should negotiate")
|
|
}
|
|
|
|
public func peerConnection(_ peerConnection: RTCPeerConnection, didChange newState: RTCIceConnectionState) {
|
|
self.logger.info("peerConnection new connection state: \(newState.rawValue)")
|
|
self.delegate?.webRTCClient(self, didChangeIceConnectionState: newState)
|
|
}
|
|
|
|
public func peerConnection(_ peerConnection: RTCPeerConnection, didChange newState: RTCIceGatheringState) {
|
|
self.logger.info("peerConnection new gathering state: \(newState.rawValue)")
|
|
self.delegate?.webRTCClient(self, didChangeIceGatheringState: newState)
|
|
|
|
if(newState == .complete) {
|
|
self.logger.info("peerConnection new gathering state is COMPLETE")
|
|
self.iceGatheringComplete = true
|
|
} else if(newState == .gathering) {
|
|
self.logger.info("peerConnection new gathering state is GATHERING")
|
|
} else if(newState == .new) {
|
|
self.logger.info("peerConnection new gathering state is NEW")
|
|
}
|
|
}
|
|
|
|
public func isIceGatheringComplete() -> Bool {
|
|
return iceGatheringComplete;
|
|
}
|
|
|
|
public func peerConnection(_ peerConnection: RTCPeerConnection, didGenerate candidate: RTCIceCandidate) {
|
|
self.logger.info("peerConnection discovered new candidate")
|
|
self.delegate?.webRTCClient(self, didDiscoverLocalCandidate: candidate)
|
|
}
|
|
|
|
public func peerConnection(_ peerConnection: RTCPeerConnection, didRemove candidates: [RTCIceCandidate]) {
|
|
self.logger.info("peerConnection did remove candidate(s)")
|
|
}
|
|
|
|
public func peerConnection(_ peerConnection: RTCPeerConnection, didOpen dataChannel: RTCDataChannel) {
|
|
self.logger.info("peerConnection did open data channel")
|
|
}
|
|
}
|
|
|
|
extension AudioWebRTCClient {
|
|
private func setTrackEnabled<T: RTCMediaStreamTrack>(_ type: T.Type, isEnabled: Bool) {
|
|
peerConnection.transceivers
|
|
.compactMap { return $0.sender.track as? T }
|
|
.forEach { $0.isEnabled = isEnabled }
|
|
}
|
|
}
|
|
|
|
// MARK: - Video control
|
|
|
|
extension AudioWebRTCClient {
|
|
func hideVideo() {
|
|
self.setVideoEnabled(false)
|
|
}
|
|
func showVideo() {
|
|
self.setVideoEnabled(true)
|
|
}
|
|
private func setVideoEnabled(_ isEnabled: Bool) {
|
|
setTrackEnabled(RTCVideoTrack.self, isEnabled: isEnabled)
|
|
}
|
|
}
|
|
|
|
// MARK:- Audio control
|
|
|
|
extension AudioWebRTCClient {
|
|
func muteAudio() {
|
|
self.setAudioEnabled(false)
|
|
}
|
|
|
|
func unmuteAudio() {
|
|
self.setAudioEnabled(true)
|
|
}
|
|
|
|
// Fallback to the default playing device: headphones/bluetooth/ear speaker
|
|
func speakerOff() {
|
|
self.audioQueue.async { [weak self] in
|
|
guard let self = self else {
|
|
return
|
|
}
|
|
|
|
self.rtcAudioSession.lockForConfiguration()
|
|
do {
|
|
try self.rtcAudioSession.setCategory(AVAudioSession.Category.playAndRecord.rawValue)
|
|
try self.rtcAudioSession.overrideOutputAudioPort(.none)
|
|
} catch let error {
|
|
debugPrint("Error setting AVAudioSession category: \(error)")
|
|
}
|
|
self.rtcAudioSession.unlockForConfiguration()
|
|
}
|
|
}
|
|
|
|
// Force speaker
|
|
func speakerOn() {
|
|
self.audioQueue.async { [weak self] in
|
|
guard let self = self else {
|
|
return
|
|
}
|
|
|
|
self.rtcAudioSession.lockForConfiguration()
|
|
do {
|
|
try self.rtcAudioSession.setCategory(AVAudioSession.Category.playAndRecord.rawValue)
|
|
try self.rtcAudioSession.overrideOutputAudioPort(.speaker)
|
|
try self.rtcAudioSession.setActive(true)
|
|
} catch let error {
|
|
debugPrint("Couldn't force audio to speaker: \(error)")
|
|
}
|
|
self.rtcAudioSession.unlockForConfiguration()
|
|
}
|
|
}
|
|
|
|
private func setAudioEnabled(_ isEnabled: Bool) {
|
|
setTrackEnabled(RTCAudioTrack.self, isEnabled: isEnabled)
|
|
}
|
|
}
|
|
|
|
extension AudioWebRTCClient: RTCDataChannelDelegate {
|
|
public func dataChannelDidChangeState(_ dataChannel: RTCDataChannel) {
|
|
debugPrint("dataChannel did change state: \(dataChannel.readyState)")
|
|
}
|
|
|
|
public func dataChannel(_ dataChannel: RTCDataChannel, didReceiveMessageWith buffer: RTCDataBuffer) {
|
|
debugPrint("dataChannel did receive message with buffer: \(buffer)")
|
|
}
|
|
}
|