362b9f238a
在 Android、Web 及 Flutter 控制端新增帧率选择 UI，被控端按屏幕刷新率生成支持的帧率档位并上报，允许在保持分辨率不变的情况下仅切换帧率。 另附带更新信号服务器的数据库配置。
489 lines
17 KiB
Swift
489 lines
17 KiB
Swift
import Foundation
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import Combine
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import UIKit
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import WebRTC
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/// 鉴权方式（与被控端约定一致）
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enum AuthType: String, CaseIterable, Identifiable {
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case none = "NONE"
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case code = "CODE"
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case password = "PASSWORD"
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var id: String { rawValue }
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var title: String {
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switch self {
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case .none: return "免密连接"
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case .code: return "动态验证码"
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case .password: return "固定密码"
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}
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}
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}
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/// 串流模式
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enum StreamMode: Int {
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case webrtc = 0
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case selfCodec = 1
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}
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/// 分辨率档位（与 Android 端一致，0 表示原始画质）
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struct ResolutionOption: Identifiable, Equatable {
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let id: Int
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let title: String
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let width: Int32
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let height: Int32
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static let all: [ResolutionOption] = [
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ResolutionOption(id: 0, title: "原始画质", width: 0, height: 0),
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ResolutionOption(id: 1, title: "1080P", width: 1920, height: 1080),
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ResolutionOption(id: 2, title: "720P", width: 1280, height: 720),
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ResolutionOption(id: 3, title: "480P", width: 854, height: 480)
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]
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}
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/// 主控端 ViewModel：串联信令、WebRTC、自编码解码与 UI 状态。
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/// 所有回调均已在主线程触发，@Published 属性直接驱动 SwiftUI。
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final class ControllerViewModel: NSObject, ObservableObject {
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// MARK: - UI 状态
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@Published var serverUrl: String = "wss://www.ttstd.com/signal"
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@Published var targetDeviceId: String = ""
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@Published var statusText: String = "未连接"
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/// 是否显示控制界面（对应 Android 的 setupPanel/controlPanel 切换）
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@Published var isControlling: Bool = false
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@Published var isConnecting: Bool = false
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@Published var statsText: String = ""
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@Published var streamMode: StreamMode = .webrtc
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@Published var selectedResolution: ResolutionOption = ResolutionOption.all[0]
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/// 帧率档位：默认常用档位，收到被控端上报的 supported_fps 后以上报列表为准
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@Published var fpsOptions: [Int] = [15, 24, 30, 60]
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/// 被控端当前采集帧率（0 表示尚未收到上报）
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@Published var currentFps: Int = 0
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/// 远端视频宽高比（宽/高），用于让触控层与画面精确对齐
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@Published var videoAspect: CGFloat = 9.0 / 16.0
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@Published var showAuthSheet: Bool = false
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@Published var alertMessage: String?
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let myDeviceId: String = DeviceUtils.deviceId()
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// MARK: - UIKit 渲染视图（由 ViewModel 持有，SwiftUI 通过 Representable 嵌入）
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let remoteVideoView = RTCMTLVideoView()
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let selfCodecView = SampleBufferDisplayView()
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// MARK: - 内部组件
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private var signalingClient: SignalingClient?
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private var webRTCClient: WebRTCClient?
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private let selfCodecDecoder = SelfCodecDecoder()
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private var statsTimer: Timer?
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private var connectionStartTime: Date?
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private var lastBytesReceived: Double = 0
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private var lastStatsTime: Date?
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private var pendingAuthType: AuthType = .none
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private var pendingAuthValue: String = ""
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/// 是否为本端主动切换串流模式（避免上报回环再次发送）
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private var suppressStreamModeSend = false
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/// 被控端最近上报的实际采集尺寸（仅切帧率时保持分辨率不变，0 表示原生）
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private var lastReportedWidth: Int32 = 0
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private var lastReportedHeight: Int32 = 0
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override init() {
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super.init()
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remoteVideoView.delegate = self
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selfCodecDecoder.displayLayer = selfCodecView.sampleBufferLayer
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selfCodecDecoder.onResolutionUpdate = { [weak self] w, h in
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guard let self, self.streamMode == .selfCodec, w > 0, h > 0 else { return }
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self.videoAspect = CGFloat(w) / CGFloat(h)
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}
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}
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// MARK: - 连接 / 断开
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/// 用户点击"连接"：弹出鉴权选择
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func requestConnect() {
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let target = targetDeviceId.trimmingCharacters(in: .whitespacesAndNewlines)
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guard !serverUrl.trimmingCharacters(in: .whitespaces).isEmpty else {
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alertMessage = "请输入信令服务器地址"
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return
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}
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guard !target.isEmpty else {
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alertMessage = "请输入目标设备 ID"
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return
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}
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showAuthSheet = true
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}
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/// 鉴权弹窗确认后开始连接
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func connect(authType: AuthType, authValue: String) {
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showAuthSheet = false
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pendingAuthType = authType
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pendingAuthValue = authValue
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isConnecting = true
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statusText = "正在连接信令服务器..."
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let signaling = SignalingClient(
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serverUrl: serverUrl.trimmingCharacters(in: .whitespaces),
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deviceId: myDeviceId)
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signaling.delegate = self
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signalingClient = signaling
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signaling.connect()
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}
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func disconnect() {
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stopStatsTimer()
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selfCodecDecoder.setEnabled(false)
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webRTCClient?.close()
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webRTCClient = nil
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signalingClient?.disconnect()
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signalingClient = nil
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isConnecting = false
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isControlling = false
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statusText = "未连接"
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statsText = ""
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streamMode = .webrtc
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selectedResolution = ResolutionOption.all[0]
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fpsOptions = [15, 24, 30, 60]
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currentFps = 0
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lastReportedWidth = 0
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lastReportedHeight = 0
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videoAspect = 9.0 / 16.0
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}
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// MARK: - 远程输入
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func sendMotionEvent(action: Int32, x: Double, y: Double) {
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var msg = ControlMessage()
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msg.action = .motionEvent
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msg.motionAction = action
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msg.x = x
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msg.y = y
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webRTCClient?.sendControlCommand(msg)
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}
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func sendSwipe(x1: Double, y1: Double, x2: Double, y2: Double, duration: Int64) {
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var msg = ControlMessage()
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msg.action = .swipe
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msg.x1 = x1
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msg.y1 = y1
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msg.x2 = x2
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msg.y2 = y2
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msg.duration = duration
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webRTCClient?.sendControlCommand(msg)
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}
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/// 发送按键（Android KeyEvent keyCode），完整的按下+抬起
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func sendKey(_ keyCode: Int32) {
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var down = ControlMessage()
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down.action = .key
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down.keyCode = keyCode
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down.keyAction = 0
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webRTCClient?.sendControlCommand(down)
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var up = ControlMessage()
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up.action = .key
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up.keyCode = keyCode
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up.keyAction = 1
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webRTCClient?.sendControlCommand(up)
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}
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/// Android 按键常量
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enum AndroidKey {
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static let back: Int32 = 4
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static let home: Int32 = 3
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static let appSwitch: Int32 = 187
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}
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// MARK: - 分辨率 / 串流模式
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func selectResolution(_ option: ResolutionOption) {
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selectedResolution = option
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webRTCClient?.requestResolutionChange(width: option.width, height: option.height, fps: 0)
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}
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/// 仅切换帧率：分辨率沿用被控端最近上报的实际采集尺寸（0 表示原生）
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func selectFps(_ fps: Int) {
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guard fps > 0, fps != currentFps else { return }
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webRTCClient?.requestResolutionChange(width: lastReportedWidth,
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height: lastReportedHeight,
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fps: Int32(fps))
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}
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func toggleStreamMode(_ selfCodecOn: Bool) {
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let newMode: StreamMode = selfCodecOn ? .selfCodec : .webrtc
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guard newMode != streamMode else { return }
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streamMode = newMode
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applyStreamModeLocally(newMode)
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if !suppressStreamModeSend {
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webRTCClient?.sendStreamMode(Int32(newMode.rawValue))
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}
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}
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private func applyStreamModeLocally(_ mode: StreamMode) {
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selfCodecDecoder.setEnabled(mode == .selfCodec)
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}
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// MARK: - 私有
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fileprivate func startWebRTC() {
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guard let signaling = signalingClient else { return }
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statusText = "正在建立 WebRTC 连接..."
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let client = WebRTCClient(signaling: signaling, myDeviceId: myDeviceId)
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client.delegate = self
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client.selfCodecDecoder = selfCodecDecoder
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webRTCClient = client
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client.createOffer(
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targetDeviceId: targetDeviceId.trimmingCharacters(in: .whitespacesAndNewlines),
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renderer: remoteVideoView,
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authType: pendingAuthType.rawValue,
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authValue: pendingAuthType == .none ? nil : pendingAuthValue)
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}
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fileprivate func failConnection(_ reason: String) {
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alertMessage = reason
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disconnect()
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}
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// MARK: - 统计
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private func startStatsTimer() {
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connectionStartTime = Date()
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lastBytesReceived = 0
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lastStatsTime = nil
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statsTimer = Timer.scheduledTimer(withTimeInterval: 1.0, repeats: true) { [weak self] _ in
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self?.collectStats()
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}
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}
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private func stopStatsTimer() {
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statsTimer?.invalidate()
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statsTimer = nil
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}
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private func collectStats() {
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webRTCClient?.stats { [weak self] report in
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guard let self else { return }
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let text = self.buildStatsText(report)
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DispatchQueue.main.async {
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self.statsText = text
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}
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}
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}
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private func buildStatsText(_ report: RTCStatisticsReport) -> String {
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var width = 0, height = 0
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var fps: Double = 0
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var bytesReceived: Double = 0
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var framesDropped: Double = 0
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var codecId: String?
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var codecName = ""
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var decoderImpl = ""
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var rttMs: Double = -1
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var localCandidateId: String?
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var remoteCandidateId: String?
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for (_, stat) in report.statistics {
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switch stat.type {
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case "inbound-rtp":
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guard (stat.values["kind"] as? String) == "video" else { continue }
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width = (stat.values["frameWidth"] as? NSNumber)?.intValue ?? width
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height = (stat.values["frameHeight"] as? NSNumber)?.intValue ?? height
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fps = (stat.values["framesPerSecond"] as? NSNumber)?.doubleValue ?? fps
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bytesReceived = (stat.values["bytesReceived"] as? NSNumber)?.doubleValue ?? bytesReceived
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framesDropped = (stat.values["framesDropped"] as? NSNumber)?.doubleValue ?? framesDropped
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codecId = stat.values["codecId"] as? String
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decoderImpl = (stat.values["decoderImplementation"] as? String) ?? decoderImpl
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case "candidate-pair":
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let nominated = (stat.values["nominated"] as? NSNumber)?.boolValue ?? false
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let state = stat.values["state"] as? String
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if nominated && state == "succeeded" {
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rttMs = ((stat.values["currentRoundTripTime"] as? NSNumber)?.doubleValue ?? 0) * 1000
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localCandidateId = stat.values["localCandidateId"] as? String
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remoteCandidateId = stat.values["remoteCandidateId"] as? String
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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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if let codecId, let codecStat = report.statistics[codecId] {
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codecName = (codecStat.values["mimeType"] as? String)?
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.replacingOccurrences(of: "video/", with: "") ?? ""
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}
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// 码率（与 Android 一致：按字节增量/时间差计算）
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var bitrateText = "--"
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let now = Date()
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if let last = lastStatsTime, bytesReceived >= lastBytesReceived {
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let interval = now.timeIntervalSince(last)
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if interval > 0 {
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let bps = (bytesReceived - lastBytesReceived) * 8 / interval
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bitrateText = Self.formatBitrate(bps)
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}
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}
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lastBytesReceived = bytesReceived
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lastStatsTime = now
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// 连接类型
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var connType = "--"
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if let lid = localCandidateId, let rid = remoteCandidateId,
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let local = report.statistics[lid], let remote = report.statistics[rid] {
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let localType = (local.values["candidateType"] as? String) ?? ""
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let remoteType = (remote.values["candidateType"] as? String) ?? ""
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let proto = ((local.values["protocol"] as? String) ?? "").uppercased()
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if localType == "relay" || remoteType == "relay" {
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connType = "TURN 中继(\(proto))"
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} else if localType == "srflx" || remoteType == "srflx" {
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connType = "P2P 公网直连(\(proto))"
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} else if localType == "host" && remoteType == "host" {
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connType = "P2P 局域网直连(\(proto))"
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} else {
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connType = "\(localType)/\(remoteType)(\(proto))"
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}
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}
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var duration = ""
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if let start = connectionStartTime {
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let secs = Int(Date().timeIntervalSince(start))
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duration = String(format: "%02d:%02d", secs / 60, secs % 60)
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}
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let modeText = streamMode == .selfCodec ? "自编码" : "WebRTC"
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var lines: [String] = []
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lines.append("模式: \(modeText) 时长: \(duration)")
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if streamMode == .selfCodec {
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let w = selfCodecDecoder.videoWidth
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let h = selfCodecDecoder.videoHeight
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lines.append("分辨率: \(w > 0 ? "\(w)x\(h)" : "--") 解码: VideoToolbox(H264)")
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} else {
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lines.append("分辨率: \(width > 0 ? "\(width)x\(height)" : "--") 帧率: \(Int(fps))fps 丢帧: \(Int(framesDropped))")
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lines.append("编码: \(codecName.isEmpty ? "--" : codecName) 解码器: \(decoderImpl.isEmpty ? "--" : decoderImpl)")
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}
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lines.append("码率: \(bitrateText) 延迟: \(rttMs >= 0 ? "\(Int(rttMs))ms" : "--")")
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lines.append("链路: \(connType)")
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return lines.joined(separator: "\n")
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}
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private static func formatBitrate(_ bps: Double) -> String {
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if bps >= 1_000_000 {
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return String(format: "%.1f Mbps", bps / 1_000_000)
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}
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if bps >= 1_000 {
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return String(format: "%.0f Kbps", bps / 1_000)
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}
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return String(format: "%.0f bps", bps)
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}
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}
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// MARK: - SignalingClientDelegate
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extension ControllerViewModel: SignalingClientDelegate {
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func signalingDidConnect() {
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statusText = "信令已连接，正在发起会话..."
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startWebRTC()
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}
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func signalingDidDisconnect() {
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guard isConnecting || isControlling else { return }
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failConnection("信令服务器连接断开")
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}
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func signaling(didFail error: String) {
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guard isConnecting || isControlling else { return }
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failConnection("信令连接失败: \(error)")
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}
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func signaling(didReceive message: SignalMessage) {
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switch (message.type ?? "").uppercased() {
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case "ANSWER":
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if let sdp = message.payloadJSON()?["sdp"] as? String {
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statusText = "收到应答，正在协商..."
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webRTCClient?.handleAnswer(sdp: sdp)
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}
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case "ICE_CANDIDATE":
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if let json = message.payloadJSON(),
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let candidate = json["candidate"] as? String {
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let sdpMid = json["sdpMid"] as? String ?? ""
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let index = Int32((json["sdpMLineIndex"] as? NSNumber)?.intValue ?? 0)
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webRTCClient?.addIceCandidate(sdpMid: sdpMid, sdpMLineIndex: index, candidate: candidate)
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}
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case "TARGET_OFFLINE":
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failConnection("目标设备不在线")
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case "CONNECTION_REJECTED":
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let reason = message.payloadJSON()?["reason"] as? String
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failConnection(reason ?? "对方拒绝了连接请求")
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case "REQUEST_ERROR":
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let reason = message.payloadJSON()?["reason"] as? String
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failConnection(reason ?? "请求错误")
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case "REQUEST_TIMEOUT":
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failConnection("连接请求超时，对方未响应")
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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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// MARK: - WebRTCClientDelegate
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extension ControllerViewModel: WebRTCClientDelegate {
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func webRTCClientDidConnect() {
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isConnecting = false
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isControlling = true
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statusText = "已连接: \(targetDeviceId)"
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applyStreamModeLocally(streamMode)
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startStatsTimer()
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}
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func webRTCClientDidDisconnect() {
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guard isControlling else { return }
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alertMessage = "连接已断开"
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disconnect()
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}
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func webRTCClient(didFail error: String) {
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failConnection(error)
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}
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func webRTCClient(didReportStreamMode mode: Int) {
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let reported = StreamMode(rawValue: mode) ?? .webrtc
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guard reported != streamMode else { return }
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// 与被控端实际模式同步（不再回发 SET_STREAM_MODE）
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suppressStreamModeSend = true
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streamMode = reported
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applyStreamModeLocally(reported)
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suppressStreamModeSend = false
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}
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func webRTCClient(didReportResolution width: Int, height: Int) {
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guard width > 0, height > 0 else { return }
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lastReportedWidth = Int32(width)
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lastReportedHeight = Int32(height)
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if streamMode == .webrtc {
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videoAspect = CGFloat(width) / CGFloat(height)
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}
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}
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func webRTCClient(didReportFps fps: Int, supportedFps: [Int]) {
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if fps > 0 { currentFps = fps }
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if !supportedFps.isEmpty { fpsOptions = supportedFps }
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}
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}
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|
||
// MARK: - RTCVideoViewDelegate（WebRTC 模式下画面尺寸变化）
|
||
|
||
extension ControllerViewModel: RTCVideoViewDelegate {
|
||
|
||
func videoView(_ videoView: RTCVideoRenderer, didChangeVideoSize size: CGSize) {
|
||
DispatchQueue.main.async {
|
||
guard size.width > 0, size.height > 0 else { return }
|
||
if self.streamMode == .webrtc {
|
||
self.videoAspect = size.width / size.height
|
||
}
|
||
}
|
||
}
|
||
}
|