tongtongstudio
6eb2c7321a
- 添加API客户端、加密存储和provision/token激活逻辑 - WebSocket改用Bearer令牌认证，移除REGISTER请求 - 设备ID改为服务端下发，支持令牌刷新和强制下线处理 - 新增deviceSecret加密存储和accessToken自动刷新 - 更新设备ID获取方式为出厂SN，添加安全存储依赖
347 lines
13 KiB
Swift
347 lines
13 KiB
Swift
import Foundation
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import WebRTC
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protocol WebRTCClientDelegate: AnyObject {
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/// ICE 连接建立（可以开始远程控制）
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func webRTCClientDidConnect()
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/// 连接断开（远端断开或网络故障）
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func webRTCClientDidDisconnect()
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/// 连接失败
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func webRTCClient(didFail error: String)
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/// 被控端上报当前串流模式（0=WebRTC 1=自编码）
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func webRTCClient(didReportStreamMode mode: Int)
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/// 被控端上报当前采集分辨率
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func webRTCClient(didReportResolution width: Int, height: Int)
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/// 被控端上报当前采集帧率与支持的帧率档位（按屏幕刷新率筛选，升序）
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func webRTCClient(didReportFps fps: Int, supportedFps: [Int])
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}
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/// WebRTC 客户端：负责 PeerConnection 的创建、Offer/Answer 协商、
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/// ICE 候选交换、控制/视频 DataChannel 管理与统计信息获取。
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/// 与 Android 端 WebRtcClient 行为保持一致。
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final class WebRTCClient: NSObject {
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static let streamModeWebRTC: Int32 = 0
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static let streamModeSelfCodec: Int32 = 1
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private static let controlChannelLabel = "control_channel"
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private static let videoChannelLabel = "video_channel"
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private static let factory: RTCPeerConnectionFactory = {
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RTCInitializeSSL()
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return RTCPeerConnectionFactory(
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encoderFactory: RTCDefaultVideoEncoderFactory(),
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decoderFactory: RTCDefaultVideoDecoderFactory())
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}()
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weak var delegate: WebRTCClientDelegate?
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/// 自编码模式的 H.264 裸流解码器（video_channel 二进制数据直接转发给它）
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var selfCodecDecoder: SelfCodecDecoder?
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private let signaling: SignalingClient
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private let myDeviceId: String
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private var targetDeviceId: String = ""
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private var peerConnection: RTCPeerConnection?
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private var controlChannel: RTCDataChannel?
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private var videoChannel: RTCDataChannel?
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private weak var remoteRenderer: RTCVideoRenderer?
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private var remoteVideoTrack: RTCVideoTrack?
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/// 额外的渲染器（用于把远端视频帧 fan-out 给录制器，WebRTC 模式）
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weak var extraRenderer: RTCVideoRenderer?
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private var connected = false
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init(signaling: SignalingClient, myDeviceId: String) {
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self.signaling = signaling
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self.myDeviceId = myDeviceId
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super.init()
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}
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/// 由服务端下发的 TURN 短期凭证覆盖默认 ICE 配置（为空时用内置 STUN/TURN）。
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static var iceServersOverride: [[String: Any]]?
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var isDataChannelOpen: Bool {
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controlChannel?.readyState == .open
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}
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// MARK: - 连接流程
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/// 创建 PeerConnection 与 DataChannel，并向目标设备发送 Offer（携带鉴权信息）。
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func createOffer(targetDeviceId: String,
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renderer: RTCVideoRenderer,
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authType: String?,
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authValue: String?) {
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self.targetDeviceId = targetDeviceId
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self.remoteRenderer = renderer
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let config = RTCConfiguration()
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if let override = WebRTCClient.iceServersOverride {
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// 服务端下发的 TURN 短期凭证（RFC 7635）。
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config.iceServers = override.compactMap { entry in
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guard let urls = entry["urls"] as? String else { return nil }
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return RTCIceServer(urlStrings: [urls],
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username: entry["username"] as? String,
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credential: entry["credential"] as? String)
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}
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} else {
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config.iceServers = [
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RTCIceServer(urlStrings: ["stun:stun.l.google.com:19302"]),
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RTCIceServer(urlStrings: ["stun:www.ttstd.com:3478"]),
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RTCIceServer(urlStrings: ["turn:www.ttstd.com:3478"],
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username: "ttstd",
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credential: "ttstd123")
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]
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}
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config.sdpSemantics = .unifiedPlan
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config.continualGatheringPolicy = .gatherContinually
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config.iceCandidatePoolSize = 10
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config.iceTransportPolicy = .all
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let pcConstraints = RTCMediaConstraints(mandatoryConstraints: nil, optionalConstraints: nil)
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guard let pc = Self.factory.peerConnection(with: config, constraints: pcConstraints, delegate: self) else {
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notifyFail("PeerConnection 创建失败")
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return
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}
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peerConnection = pc
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// 只接收视频，不发送本地媒体
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let transceiverInit = RTCRtpTransceiverInit()
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transceiverInit.direction = .recvOnly
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pc.addTransceiver(of: .video, init: transceiverInit)
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// 控制通道：可靠、有序
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let controlConfig = RTCDataChannelConfiguration()
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controlConfig.isOrdered = true
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controlChannel = pc.dataChannel(forLabel: Self.controlChannelLabel, configuration: controlConfig)
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controlChannel?.delegate = self
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// 自编码视频通道：不可靠、低延迟（丢帧优先）
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let videoConfig = RTCDataChannelConfiguration()
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videoConfig.isOrdered = false
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videoConfig.maxRetransmits = 0
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videoChannel = pc.dataChannel(forLabel: Self.videoChannelLabel, configuration: videoConfig)
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videoChannel?.delegate = self
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let offerConstraints = RTCMediaConstraints(
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mandatoryConstraints: [
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"OfferToReceiveVideo": "true",
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"OfferToReceiveAudio": "false"
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],
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optionalConstraints: nil)
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pc.offer(for: offerConstraints) { [weak self] sdp, error in
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guard let self else { return }
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if let error {
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self.notifyFail("创建 Offer 失败: \(error.localizedDescription)")
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return
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}
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guard let sdp else {
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self.notifyFail("创建 Offer 失败: SDP 为空")
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return
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}
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pc.setLocalDescription(sdp) { [weak self] error in
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guard let self else { return }
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if let error {
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self.notifyFail("设置本地 SDP 失败: \(error.localizedDescription)")
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return
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}
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self.sendOffer(sdp: sdp.sdp, authType: authType, authValue: authValue)
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}
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}
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}
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private func sendOffer(sdp: String, authType: String?, authValue: String?) {
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var msg = SignalMessage()
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msg.type = "OFFER"
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msg.fromDeviceId = myDeviceId
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msg.toDeviceId = targetDeviceId
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msg.payload = SignalMessage.encodePayload(["sdp": sdp])
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msg.authType = authType
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msg.authValue = authValue
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signaling.send(msg)
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}
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/// 处理被控端返回的 Answer
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func handleAnswer(sdp: String) {
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let desc = RTCSessionDescription(type: .answer, sdp: sdp)
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peerConnection?.setRemoteDescription(desc) { [weak self] error in
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if let error {
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self?.notifyFail("设置远端 SDP 失败: \(error.localizedDescription)")
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}
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}
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}
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/// 添加远端 ICE 候选
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func addIceCandidate(sdpMid: String, sdpMLineIndex: Int32, candidate: String) {
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let ice = RTCIceCandidate(sdp: candidate, sdpMLineIndex: sdpMLineIndex, sdpMid: sdpMid)
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peerConnection?.add(ice) { error in
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if let error {
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NSLog("[WebRTC] addIceCandidate error: \(error.localizedDescription)")
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}
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}
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}
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// MARK: - 控制指令
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func sendControlCommand(_ message: ControlMessage) {
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guard let channel = controlChannel, channel.readyState == .open else { return }
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let buffer = RTCDataBuffer(data: message.serializedData(), isBinary: true)
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channel.sendData(buffer)
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}
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/// 请求被控端切换采集分辨率（width=0 表示恢复原生分辨率）
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func requestResolutionChange(width: Int32, height: Int32, fps: Int32) {
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var msg = ControlMessage()
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msg.action = .setResolution
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msg.width = width
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msg.height = height
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msg.fps = fps
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sendControlCommand(msg)
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}
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/// 请求被控端切换串流模式（0=WebRTC 1=自编码）
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func sendStreamMode(_ mode: Int32) {
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var msg = ControlMessage()
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msg.action = .setStreamMode
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msg.streamMode = mode
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sendControlCommand(msg)
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}
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// MARK: - 统计
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func stats(_ completion: @escaping (RTCStatisticsReport) -> Void) {
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peerConnection?.statistics(completionHandler: completion)
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}
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// MARK: - 关闭
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func close() {
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connected = false
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controlChannel?.close()
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videoChannel?.close()
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controlChannel = nil
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videoChannel = nil
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if let track = remoteVideoTrack, let renderer = remoteRenderer {
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track.remove(renderer)
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}
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if let track = remoteVideoTrack, let extra = extraRenderer {
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track.remove(extra)
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}
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remoteVideoTrack = nil
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peerConnection?.close()
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peerConnection = nil
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}
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private func notifyFail(_ text: String) {
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DispatchQueue.main.async {
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self.delegate?.webRTCClient(didFail: text)
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}
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}
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}
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// MARK: - RTCPeerConnectionDelegate
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extension WebRTCClient: RTCPeerConnectionDelegate {
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func peerConnection(_ peerConnection: RTCPeerConnection, didChange stateChanged: RTCSignalingState) {}
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func peerConnection(_ peerConnection: RTCPeerConnection, didAdd stream: RTCMediaStream) {}
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func peerConnection(_ peerConnection: RTCPeerConnection, didRemove stream: RTCMediaStream) {}
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func peerConnectionShouldNegotiate(_ peerConnection: RTCPeerConnection) {}
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func peerConnection(_ peerConnection: RTCPeerConnection, didChange newState: RTCIceConnectionState) {
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DispatchQueue.main.async {
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switch newState {
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case .connected, .completed:
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if !self.connected {
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self.connected = true
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self.delegate?.webRTCClientDidConnect()
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}
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case .disconnected, .failed, .closed:
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if self.connected {
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self.connected = false
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self.delegate?.webRTCClientDidDisconnect()
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} else if newState == .failed {
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self.delegate?.webRTCClient(didFail: "ICE 连接失败")
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}
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default:
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break
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}
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}
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}
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func peerConnection(_ peerConnection: RTCPeerConnection, didChange newState: RTCIceGatheringState) {}
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func peerConnection(_ peerConnection: RTCPeerConnection, didGenerate candidate: RTCIceCandidate) {
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var msg = SignalMessage()
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msg.type = "ICE_CANDIDATE"
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msg.fromDeviceId = myDeviceId
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msg.toDeviceId = targetDeviceId
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msg.payload = SignalMessage.encodePayload([
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"sdpMid": candidate.sdpMid ?? "",
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"sdpMLineIndex": Int(candidate.sdpMLineIndex),
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"candidate": candidate.sdp
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])
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signaling.send(msg)
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}
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func peerConnection(_ peerConnection: RTCPeerConnection, didRemove candidates: [RTCIceCandidate]) {}
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func peerConnection(_ peerConnection: RTCPeerConnection, didOpen dataChannel: RTCDataChannel) {
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// 作为 Offer 方，通道由本端创建；此回调用于兜底远端创建的通道
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dataChannel.delegate = self
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if dataChannel.label == Self.videoChannelLabel {
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videoChannel = dataChannel
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} else if dataChannel.label == Self.controlChannelLabel {
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controlChannel = dataChannel
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}
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}
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func peerConnection(_ peerConnection: RTCPeerConnection,
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didAdd rtpReceiver: RTCRtpReceiver,
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streams mediaStreams: [RTCMediaStream]) {
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guard let track = rtpReceiver.track as? RTCVideoTrack else { return }
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DispatchQueue.main.async {
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self.remoteVideoTrack = track
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if let renderer = self.remoteRenderer {
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track.add(renderer)
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}
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if let extra = self.extraRenderer {
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track.add(extra)
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}
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}
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}
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}
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// MARK: - RTCDataChannelDelegate
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extension WebRTCClient: RTCDataChannelDelegate {
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func dataChannelDidChangeState(_ dataChannel: RTCDataChannel) {
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NSLog("[WebRTC] DataChannel \(dataChannel.label) state: \(dataChannel.readyState.rawValue)")
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}
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func dataChannel(_ dataChannel: RTCDataChannel, didReceiveMessageWith buffer: RTCDataBuffer) {
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if dataChannel.label == Self.videoChannelLabel {
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// 自编码 H.264 裸流分片
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selfCodecDecoder?.onBinaryMessage(buffer.data)
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return
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}
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// 控制通道：被控端上报（串流模式 / 分辨率）
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guard buffer.isBinary, let msg = ControlMessage.parse(from: buffer.data) else { return }
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DispatchQueue.main.async {
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switch msg.action {
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case .reportStreamMode:
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self.delegate?.webRTCClient(didReportStreamMode: Int(msg.streamMode))
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case .reportResolution:
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self.delegate?.webRTCClient(didReportResolution: Int(msg.width), height: Int(msg.height))
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self.delegate?.webRTCClient(didReportFps: Int(msg.fps),
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supportedFps: msg.supportedFps.map(Int.init))
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default:
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break
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}
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}
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}
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}
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