docs(webrtc_controller_flutter): 更新项目文档以反映重构后的架构
AGENTS.md 与 README.md 同步更新：根据实际代码结构重写目录树、技术栈、架构分层及编码规范，移除旧版内联示例并补充新的开发约定与代码生成命令。
This commit is contained in:
@@ -0,0 +1,859 @@
|
||||
import 'package:flutter/foundation.dart';
|
||||
import 'package:flutter/services.dart';
|
||||
import 'package:flutter_webrtc/flutter_webrtc.dart';
|
||||
|
||||
import '../../../app/constants/app_constants.dart';
|
||||
import '../../../app/constants/ice_servers.dart';
|
||||
import '../../../core/proto/control_message.pb.dart';
|
||||
import '../../../core/utils/control_commands.dart';
|
||||
import '../domain/signal_message.dart';
|
||||
import 'self_codec_decoder.dart';
|
||||
import 'signaling_client.dart';
|
||||
|
||||
/// WebRTC 全托管模式下，由原生 VideoSink 探针逐帧实测得到的解码耗时统计。
|
||||
class _DecodeProbeStats {
|
||||
final double lastMs;
|
||||
final double avgMs;
|
||||
final double peakMs;
|
||||
final int totalFrames;
|
||||
final int fpsNow;
|
||||
final int fpsAvg;
|
||||
const _DecodeProbeStats({
|
||||
required this.lastMs,
|
||||
required this.avgMs,
|
||||
required this.peakMs,
|
||||
required this.totalFrames,
|
||||
required this.fpsNow,
|
||||
required this.fpsAvg,
|
||||
});
|
||||
}
|
||||
|
||||
/// 封装 WebRTC 连接逻辑（对应 Android 端 WebRtcClient）。
|
||||
///
|
||||
/// 职责：
|
||||
/// - 创建 PeerConnection（recvonly 视频 + 控制用 DataChannel）
|
||||
/// - 创建 Offer 并通过信令发送
|
||||
/// - 处理 Answer / ICE 候选
|
||||
/// - 通过 DataChannel 发送控制指令（触摸/滑动/按键）
|
||||
/// - 采集并解析连接统计信息
|
||||
///
|
||||
/// 本类同时兼容 Android 与 iOS（基于 flutter_webrtc）。
|
||||
class WebRtcController {
|
||||
final SignalingClient signaling;
|
||||
final String deviceId;
|
||||
final String targetDeviceId;
|
||||
final String? authType;
|
||||
final String? authValue;
|
||||
|
||||
RTCPeerConnection? _pc;
|
||||
RTCDataChannel? _dataChannel;
|
||||
RTCDataChannel? _videoDataChannel;
|
||||
SelfCodecDecoder? _selfCodecDecoder;
|
||||
int _currentStreamMode = streamModeWebRtc;
|
||||
bool _creatingDecoder = false;
|
||||
|
||||
/// 远端视频渲染器，由调用方持有并显示。
|
||||
final RTCVideoRenderer renderer = RTCVideoRenderer();
|
||||
|
||||
void Function()? onConnectionEstablished;
|
||||
void Function(String error)? onConnectionFailed;
|
||||
void Function()? onDisconnected;
|
||||
|
||||
/// ICE 连接断开（如网络中断、被控端退出），用于提示用户并返回连接设置。
|
||||
void Function(String message)? onIceDisconnected;
|
||||
|
||||
/// 远端视频流就绪（renderer 已绑定视频轨道）。
|
||||
void Function(RTCVideoRenderer renderer)? onRemoteStream;
|
||||
|
||||
/// 自编码解码纹理已就绪（textureId >= 0）。
|
||||
void Function(int)? onSelfCodecReady;
|
||||
|
||||
/// 自编码解码器已释放（连接断开时）。
|
||||
void Function()? onSelfCodecLost;
|
||||
|
||||
/// 被控端上报当前生效的串流模式。
|
||||
void Function(int mode)? onStreamModeReport;
|
||||
|
||||
/// 被控端上报分辨率（自编码解码器输出尺寸变化时也会触发）。
|
||||
void Function(int width, int height)? onResolutionReported;
|
||||
|
||||
/// 被控端上报当前采集帧率与支持的帧率档位（仅 REPORT_RESOLUTION 触发）。
|
||||
void Function(int width, int height, int fps, List<int> supportedFps)?
|
||||
onFpsReport;
|
||||
|
||||
/// 当前平台（Windows / Linux / Web / iOS 等）不支持原生硬解时回调。
|
||||
void Function()? onSelfCodecNotSupported;
|
||||
|
||||
static const String kVideoChannelLabel = 'video_channel';
|
||||
static const int streamModeWebRtc = 0;
|
||||
static const int streamModeSelfCodec = 1;
|
||||
|
||||
/// 由服务端下发的 TURN 短期凭证覆盖默认 ICE 配置（为空则用 kIceServers）。
|
||||
static List<Map<String, dynamic>>? iceServersOverride;
|
||||
|
||||
WebRtcController({
|
||||
required this.signaling,
|
||||
required this.deviceId,
|
||||
required this.targetDeviceId,
|
||||
this.authType,
|
||||
this.authValue,
|
||||
});
|
||||
|
||||
/// 初始化渲染器、PeerConnection，并创建 Offer。
|
||||
Future<void> initialize() async {
|
||||
await renderer.initialize();
|
||||
|
||||
final iceServers = iceServersOverride ?? kIceServers;
|
||||
final configuration = {
|
||||
'iceServers': iceServers,
|
||||
'sdpSemantics': 'unified-plan',
|
||||
'iceCandidatePoolSize': 10,
|
||||
// 增强复杂网络下的稳定性，参考 Android 端配置。
|
||||
'continualGatheringPolicy': 'gatherContinually',
|
||||
'iceTransportsType': 'all',
|
||||
'tcpCandidatePolicy': 'enabled',
|
||||
};
|
||||
|
||||
final iceUrls = iceServers
|
||||
.map((e) => '${e['username'] != null ? '(${e['username']})' : ''}${e['urls']}')
|
||||
.join(', ');
|
||||
debugPrint('[WebRtcController] signaling server: ${signaling.serverUrl} | '
|
||||
'deviceId=$deviceId target=$targetDeviceId');
|
||||
debugPrint('[WebRtcController] ICE servers: $iceUrls');
|
||||
|
||||
_pc = await createPeerConnection(configuration);
|
||||
|
||||
_pc!.onIceCandidate = _onIceCandidate;
|
||||
_pc!.onIceConnectionState = _onIceConnectionState;
|
||||
_pc!.onIceGatheringState = (state) {
|
||||
debugPrint('[WebRtcController] ICE gathering state: $state');
|
||||
};
|
||||
_pc!.onTrack = _onTrack;
|
||||
_pc!.onDataChannel = _onDataChannel;
|
||||
|
||||
// 仅接收远端视频（recvonly）。
|
||||
await _pc!.addTransceiver(
|
||||
kind: RTCRtpMediaType.RTCRtpMediaTypeVideo,
|
||||
init: RTCRtpTransceiverInit(direction: TransceiverDirection.RecvOnly),
|
||||
);
|
||||
|
||||
// 创建控制用 DataChannel。
|
||||
// 使用非可靠、无序模式以降低延迟，解决"不跟手"问题。
|
||||
final dcInit = RTCDataChannelInit()
|
||||
..ordered = false
|
||||
..maxRetransmits = 0;
|
||||
_dataChannel = await _pc!.createDataChannel(kDataChannelLabel, dcInit);
|
||||
_setupDataChannel(_dataChannel!);
|
||||
|
||||
// 创建自编码视频透传通道（无序、不可靠，类似 RTP，最低延迟）。
|
||||
// 被控端仅在收到 SET_STREAM_MODE=自编码 后才经此通道推送 H.264 裸流。
|
||||
final videoInit = RTCDataChannelInit()
|
||||
..ordered = false
|
||||
..maxRetransmits = 0;
|
||||
_videoDataChannel = await _pc!.createDataChannel(
|
||||
kVideoChannelLabel,
|
||||
videoInit,
|
||||
);
|
||||
_setupVideoDataChannel(_videoDataChannel!);
|
||||
|
||||
// 创建并发送 Offer。
|
||||
final constraints = {
|
||||
'mandatory': {
|
||||
'OfferToReceiveVideo': true,
|
||||
'OfferToReceiveAudio': false,
|
||||
},
|
||||
'optional': [],
|
||||
};
|
||||
final offer = await _pc!.createOffer(constraints);
|
||||
// 与 WebRTCControlled 的 optimizeSdp 保持一致：Offer 的 m=video 行优先 VP8/VP9，
|
||||
// H264 兜底，确保各控制端（桌面/浏览器/Chromium/Linux）都能稳定解码，
|
||||
// 即编码端（被控端）与解码端（控制端）协商出一致的、可解码的视频格式。
|
||||
final reorderedSdp = _preferVideoCodecs(offer.sdp!);
|
||||
final offerWithPref = RTCSessionDescription(reorderedSdp, offer.type);
|
||||
await _pc!.setLocalDescription(offerWithPref);
|
||||
_sendOffer(reorderedSdp, authType: authType, authValue: authValue);
|
||||
}
|
||||
|
||||
/// 重排 Offer SDP 中 `m=video` 行的视频编解码器顺序：VP8/VP9 优先，H264 兜底。
|
||||
///
|
||||
/// 与 Android 被控端 [WebRtcClient.optimizeSdp] 的编解码优先级保持一致，
|
||||
/// 保证协商出的编码格式控制端一定可解码（编码/解码一致性）。
|
||||
String _preferVideoCodecs(String sdp) {
|
||||
final lines = sdp.split('\r\n');
|
||||
final vp8 = <String>[];
|
||||
final vp9 = <String>[];
|
||||
final h264 = <String>[];
|
||||
for (final line in lines) {
|
||||
final t = line.trim();
|
||||
if (t.startsWith('a=rtpmap:')) {
|
||||
final payload = t.split(':')[1].split(' ')[0];
|
||||
if (t.contains('VP8/90000')) {
|
||||
vp8.add(payload);
|
||||
} else if (t.contains('VP9/90000')) {
|
||||
vp9.add(payload);
|
||||
} else if (t.contains('H264/90000')) {
|
||||
h264.add(payload);
|
||||
}
|
||||
}
|
||||
}
|
||||
final ordered = <String>[...vp8, ...vp9, ...h264];
|
||||
if (ordered.isEmpty) return sdp;
|
||||
|
||||
final result = <String>[];
|
||||
for (final line in lines) {
|
||||
final t = line.trim();
|
||||
if (t.startsWith('m=video')) {
|
||||
final parts = t.split(' ');
|
||||
if (parts.length > 3) {
|
||||
final head = '${parts[0]} ${parts[1]} ${parts[2]}';
|
||||
final rest = parts
|
||||
.skip(3)
|
||||
.where((p) => !ordered.contains(p))
|
||||
.toList();
|
||||
result.add([head, ...ordered, ...rest].join(' '));
|
||||
continue;
|
||||
}
|
||||
}
|
||||
result.add(line);
|
||||
}
|
||||
return result.join('\r\n');
|
||||
}
|
||||
|
||||
void _onTrack(RTCTrackEvent event) {
|
||||
if (event.track.kind != 'video') return;
|
||||
_bindRemoteVideo(event);
|
||||
}
|
||||
|
||||
/// 绑定远端视频轨道到渲染器。
|
||||
///
|
||||
/// 与 Web 端 WebRtcController.ontrack 的兜底逻辑保持一致：
|
||||
/// 某些平台/协商场景下（Unified Plan + recvonly）`event.streams` 可能为空，
|
||||
/// 此时必须用 `event.track` 自行构造 MediaStream，否则 renderer.srcObject
|
||||
/// 为空 -> 控制端拿不到画面（黑屏/一直显示"等待画面"），但控制通道不受影响。
|
||||
Future<void> _bindRemoteVideo(RTCTrackEvent event) async {
|
||||
MediaStream stream;
|
||||
if (event.streams.isNotEmpty) {
|
||||
stream = event.streams[0];
|
||||
} else {
|
||||
// event.streams 为空：用 track 自行构造 MediaStream。
|
||||
stream = await createLocalMediaStream('remoteVideo');
|
||||
await stream.addTrack(event.track);
|
||||
}
|
||||
renderer.srcObject = stream;
|
||||
onRemoteStream?.call(renderer);
|
||||
// 仅在 WebRTC 全托管模式 attach 解码耗时探针（自编码模式用自己的解码器统计）。
|
||||
if (_selfCodecDecoder == null) {
|
||||
final trackId = event.track.id;
|
||||
if (trackId != null) _attachWebRtcDecodeProbe(trackId);
|
||||
}
|
||||
}
|
||||
|
||||
/// 解码耗时探针通道（原生侧给远端视频轨道 attach 一个测量 VideoSink）。
|
||||
static const MethodChannel _decodeProbeChannel =
|
||||
MethodChannel('webrtc_decode_probe');
|
||||
|
||||
Future<void> _attachWebRtcDecodeProbe(String trackId) async {
|
||||
debugPrint('[WebRtcController] Attaching decode probe for track: $trackId');
|
||||
try {
|
||||
await _decodeProbeChannel.invokeMethod<void>(
|
||||
'attachWebRtcDecodeProbe', {'trackId': trackId});
|
||||
} catch (e) {
|
||||
debugPrint('[decodeProbe] attach failed: $e');
|
||||
}
|
||||
}
|
||||
|
||||
Future<void> _detachWebRtcDecodeProbe() async {
|
||||
try {
|
||||
await _decodeProbeChannel.invokeMethod<void>('detachWebRtcDecodeProbe');
|
||||
} catch (e) {
|
||||
debugPrint('[decodeProbe] detach failed: $e');
|
||||
}
|
||||
}
|
||||
|
||||
Future<_DecodeProbeStats?> _getWebRtcDecodeStats() async {
|
||||
try {
|
||||
final res =
|
||||
await _decodeProbeChannel.invokeMethod<dynamic>('getWebRtcDecodeStats');
|
||||
if (res is Map) {
|
||||
final m = res.cast<String, dynamic>();
|
||||
return _DecodeProbeStats(
|
||||
lastMs: (m['lastMs'] as num? ?? 0).toDouble(),
|
||||
avgMs: (m['avgMs'] as num? ?? 0).toDouble(),
|
||||
peakMs: (m['peakMs'] as num? ?? 0).toDouble(),
|
||||
totalFrames: (m['totalFrames'] as num? ?? 0).toInt(),
|
||||
fpsNow: (m['fpsNow'] as num? ?? 0).toInt(),
|
||||
fpsAvg: (m['fpsAvg'] as num? ?? 0).toInt(),
|
||||
);
|
||||
}
|
||||
} catch (e) {
|
||||
debugPrint('[decodeProbe] getStats failed: $e');
|
||||
}
|
||||
return null;
|
||||
}
|
||||
|
||||
void _onDataChannel(RTCDataChannel channel) {
|
||||
// 被控端也可能主动创建 DataChannel，按标签统一处理。
|
||||
if (channel.label == kDataChannelLabel) {
|
||||
_dataChannel ??= channel;
|
||||
_setupDataChannel(channel);
|
||||
} else if (channel.label == kVideoChannelLabel) {
|
||||
_videoDataChannel ??= channel;
|
||||
_setupVideoDataChannel(channel);
|
||||
}
|
||||
}
|
||||
|
||||
void _setupDataChannel(RTCDataChannel channel) {
|
||||
channel.onMessage = (RTCDataChannelMessage message) {
|
||||
if (message.isBinary) {
|
||||
try {
|
||||
final command = ControlMessage.fromBuffer(message.binary);
|
||||
_handleControlMessage(command);
|
||||
} catch (e) {
|
||||
// ignore: avoid_print
|
||||
print('DataChannel message decode failed: $e');
|
||||
}
|
||||
} else {
|
||||
// ignore: avoid_print
|
||||
print('DataChannel text message: ${message.text}');
|
||||
}
|
||||
};
|
||||
channel.onDataChannelState = (RTCDataChannelState state) {
|
||||
// ignore: avoid_print
|
||||
print('DataChannel state: $state');
|
||||
};
|
||||
}
|
||||
|
||||
void _setupVideoDataChannel(RTCDataChannel channel) {
|
||||
channel.onMessage = (RTCDataChannelMessage message) {
|
||||
if (message.isBinary) {
|
||||
_selfCodecDecoder?.feed(message.binary);
|
||||
}
|
||||
};
|
||||
channel.onDataChannelState = (RTCDataChannelState state) {
|
||||
if (state == RTCDataChannelState.RTCDataChannelOpen) {
|
||||
_onVideoChannelOpen();
|
||||
} else if (state == RTCDataChannelState.RTCDataChannelClosed) {
|
||||
_selfCodecDecoder?.setEnabled(false);
|
||||
}
|
||||
};
|
||||
}
|
||||
|
||||
void _onVideoChannelOpen() {
|
||||
_ensureSelfCodecDecoder().then((_) {
|
||||
_selfCodecDecoder?.setEnabled(_currentStreamMode == streamModeSelfCodec);
|
||||
});
|
||||
}
|
||||
|
||||
Future<void> _ensureSelfCodecDecoder() async {
|
||||
if (_selfCodecDecoder != null || _creatingDecoder) return;
|
||||
_creatingDecoder = true;
|
||||
try {
|
||||
final decoder = SelfCodecDecoder(
|
||||
onSize: (w, h) {
|
||||
_selfCodecWidth = w;
|
||||
_selfCodecHeight = h;
|
||||
onResolutionReported?.call(w, h);
|
||||
},
|
||||
onError: (err) => debugPrint('自编码解码器: $err'),
|
||||
);
|
||||
final textureId = await decoder.create();
|
||||
if (textureId != null) {
|
||||
_selfCodecDecoder = decoder;
|
||||
onSelfCodecReady?.call(textureId);
|
||||
} else {
|
||||
// 当前平台（Windows / Linux / Web / iOS 等）无原生硬解能力，回退 WebRTC 媒体流。
|
||||
onSelfCodecNotSupported?.call();
|
||||
}
|
||||
} finally {
|
||||
_creatingDecoder = false;
|
||||
}
|
||||
}
|
||||
|
||||
void _handleControlMessage(ControlMessage msg) {
|
||||
switch (msg.action) {
|
||||
case Action.REPORT_STREAM_MODE:
|
||||
_currentStreamMode = msg.streamMode;
|
||||
if (msg.streamMode == streamModeSelfCodec) {
|
||||
_ensureSelfCodecDecoder().then((_) {
|
||||
_selfCodecDecoder?.setEnabled(true);
|
||||
});
|
||||
} else {
|
||||
_selfCodecDecoder?.setEnabled(false);
|
||||
}
|
||||
onStreamModeReport?.call(msg.streamMode);
|
||||
break;
|
||||
case Action.REPORT_RESOLUTION:
|
||||
if (msg.width > 0 && msg.height > 0) {
|
||||
_selfCodecWidth = msg.width;
|
||||
_selfCodecHeight = msg.height;
|
||||
}
|
||||
onResolutionReported?.call(msg.width, msg.height);
|
||||
// 同步帧率信息（当前帧率 + 被控端按刷新率筛选的帧率档位）
|
||||
onFpsReport?.call(
|
||||
msg.width, msg.height, msg.fps, msg.supportedFps.toList());
|
||||
break;
|
||||
default:
|
||||
// ignore: avoid_print
|
||||
print('DataChannel message: ${msg.action}');
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
/// 请求被控端切换屏幕串流模式（SET_STREAM_MODE 指令）。
|
||||
/// [mode] 为 0=WebRTC 全托管 / 1=自编码。
|
||||
Future<void> sendStreamMode(int mode) async {
|
||||
_currentStreamMode = mode;
|
||||
if (mode == streamModeSelfCodec) {
|
||||
await _ensureSelfCodecDecoder();
|
||||
await _selfCodecDecoder?.setEnabled(true);
|
||||
} else {
|
||||
await _selfCodecDecoder?.setEnabled(false);
|
||||
}
|
||||
sendControlCommand(ControlCommands.streamMode(mode));
|
||||
}
|
||||
|
||||
void _onIceCandidate(RTCIceCandidate candidate) {
|
||||
debugPrint('[WebRtcController] local ICE candidate: ${candidate.candidate}');
|
||||
final payload = {
|
||||
'sdpMid': candidate.sdpMid,
|
||||
'sdpMLineIndex': candidate.sdpMLineIndex,
|
||||
'candidate': candidate.candidate,
|
||||
};
|
||||
final msg = SignalMessage.withPayload(
|
||||
type: 'ICE_CANDIDATE',
|
||||
fromDeviceId: deviceId,
|
||||
toDeviceId: targetDeviceId,
|
||||
deviceType: 'CONTROLLER',
|
||||
payload: payload,
|
||||
);
|
||||
signaling.send(msg);
|
||||
}
|
||||
|
||||
/// 打印 ICE candidate-pair 统计，用于诊断连接失败原因。
|
||||
Future<void> _dumpIceStats() async {
|
||||
try {
|
||||
final stats = await _pc?.getStats();
|
||||
final pairs = <String>[];
|
||||
for (final report in stats ?? []) {
|
||||
if (report.type == 'candidate-pair') {
|
||||
final state = report.values['state'] ?? report.values['nominated'];
|
||||
pairs.add('${report.values['localCandidateId']}->'
|
||||
'${report.values['remoteCandidateId']} state=$state');
|
||||
}
|
||||
}
|
||||
debugPrint('[WebRtcController] ICE Failed. candidate-pair 数量=${pairs.length}');
|
||||
for (final p in pairs) {
|
||||
debugPrint('[WebRtcController] pair: $p');
|
||||
}
|
||||
} catch (e) {
|
||||
debugPrint('[WebRtcController] getStats 失败: $e');
|
||||
}
|
||||
}
|
||||
|
||||
void _onIceConnectionState(RTCIceConnectionState state) {
|
||||
// ignore: avoid_print
|
||||
print('ICE connection state: $state');
|
||||
if (state == RTCIceConnectionState.RTCIceConnectionStateChecking) {
|
||||
debugPrint('[WebRtcController] ICE Checking: 正在收集/交换候选，等待连通...');
|
||||
} else if (state == RTCIceConnectionState.RTCIceConnectionStateConnected) {
|
||||
_connectedAt = DateTime.now().millisecondsSinceEpoch;
|
||||
onConnectionEstablished?.call();
|
||||
} else if (state == RTCIceConnectionState.RTCIceConnectionStateDisconnected) {
|
||||
onIceDisconnected?.call('ICE 连接已断开');
|
||||
} else if (state == RTCIceConnectionState.RTCIceConnectionStateFailed) {
|
||||
_dumpIceStats();
|
||||
onConnectionFailed?.call('ICE 连接失败');
|
||||
onDisconnected?.call();
|
||||
}
|
||||
}
|
||||
|
||||
void _sendOffer(String sdp, {String? authType, String? authValue}) {
|
||||
final payload = {'sdp': sdp};
|
||||
final msg = SignalMessage.withPayload(
|
||||
type: 'OFFER',
|
||||
fromDeviceId: deviceId,
|
||||
toDeviceId: targetDeviceId,
|
||||
deviceType: 'CONTROLLER',
|
||||
payload: payload,
|
||||
authType: authType,
|
||||
authValue: authValue,
|
||||
);
|
||||
signaling.send(msg);
|
||||
}
|
||||
|
||||
/// 处理来自信令服务器的 Answer。
|
||||
Future<void> handleAnswer(String sdp) async {
|
||||
final answer = RTCSessionDescription(sdp, 'answer');
|
||||
await _pc?.setRemoteDescription(answer);
|
||||
}
|
||||
|
||||
/// 处理来自信令服务器的 ICE 候选。
|
||||
Future<void> handleIceCandidate(Map<String, dynamic> payload) async {
|
||||
final candidate = RTCIceCandidate(
|
||||
payload['candidate'] as String,
|
||||
payload['sdpMid'] as String?,
|
||||
payload['sdpMLineIndex'] as int?,
|
||||
);
|
||||
debugPrint('[WebRtcController] remote ICE candidate: ${candidate.candidate}');
|
||||
await _pc?.addCandidate(candidate);
|
||||
}
|
||||
|
||||
/// 通过 DataChannel 发送控制指令（protobuf 二进制）。
|
||||
void sendControlCommand(ControlMessage command) {
|
||||
if (_dataChannel?.state == RTCDataChannelState.RTCDataChannelOpen) {
|
||||
debugPrint('WebRtcController: Sending command -> ${command.action}');
|
||||
_dataChannel!.send(RTCDataChannelMessage.fromBinary(command.writeToBuffer()));
|
||||
}
|
||||
}
|
||||
|
||||
/// 请求被控端切换屏幕采集分辨率（SET_RESOLUTION 指令）。
|
||||
void sendResolutionChange(int width, int height, int fps) {
|
||||
sendControlCommand(ControlCommands.setResolution(width, height, fps));
|
||||
}
|
||||
|
||||
/// 上一次统计采样的字节数与时间戳，用于计算每秒网速。
|
||||
int _prevBytesReceived = 0;
|
||||
int _prevBytesSent = 0;
|
||||
int _prevStatsTimestamp = 0;
|
||||
|
||||
/// 上一次采样时的累计解码帧数，用于计算自编码模式下的解码帧率。
|
||||
int _prevDecodeCount = 0;
|
||||
|
||||
/// 上一次采样时的累计解码时间（秒）与解码帧数，用于计算 WebRTC 模式下的解码耗时。
|
||||
double _prevTotalDecodeTime = 0.0;
|
||||
int _prevFramesDecoded = 0;
|
||||
|
||||
/// WebRTC 全托管模式下，由原生 VideoSink 探针实测的解码耗时统计缓存。
|
||||
_DecodeProbeStats? _webRtcDecodeProbe;
|
||||
|
||||
/// 解码器上报的真实分辨率（自编码模式下 WebRTC 视频统计为空，需用此覆盖）。
|
||||
int _selfCodecWidth = 0;
|
||||
int _selfCodecHeight = 0;
|
||||
|
||||
/// 连接建立时的时间戳（毫秒），用于计算连接时长。
|
||||
int? _connectedAt;
|
||||
|
||||
/// 采集连接统计信息，回调格式化后的文本（对应 Android 端 updateStats）。
|
||||
///
|
||||
/// 包含：分辨率/帧率/延迟/解码格式、抖动、丢包率、丢帧、
|
||||
/// 每秒上下行网速、连接类型(P2P/TURN)+候选协议、连接时长、
|
||||
/// 控制 DataChannel 状态与收发消息数。
|
||||
Future<String> getStatsText() async {
|
||||
if (_pc == null) return '';
|
||||
final reports = await _pc!.getStats();
|
||||
|
||||
dynamic width = '-';
|
||||
dynamic height = '-';
|
||||
dynamic fps = '-';
|
||||
dynamic delay = '-';
|
||||
dynamic codec = '-';
|
||||
dynamic jitter = '-';
|
||||
dynamic lossRate = '-';
|
||||
dynamic framesDropped = '-';
|
||||
|
||||
int bytesReceived = 0;
|
||||
int bytesSent = 0;
|
||||
String? localCandidateId;
|
||||
String? remoteCandidateId;
|
||||
bool hasNominatedPair = false;
|
||||
|
||||
// 控制 DataChannel 统计（仅统计与对端的控制通道）。
|
||||
int dcMessagesSent = 0;
|
||||
int dcMessagesReceived = 0;
|
||||
|
||||
double currentTotalDecodeTime = 0.0;
|
||||
int currentFramesDecoded = 0;
|
||||
|
||||
// 与上次采样的时间差（秒），供循环内估算解码帧率，也供下方计算每秒网速。
|
||||
final now = DateTime.now().millisecondsSinceEpoch;
|
||||
final double dtSec =
|
||||
_prevStatsTimestamp != 0 ? (now - _prevStatsTimestamp) / 1000.0 : 0;
|
||||
|
||||
for (final report in reports) {
|
||||
final values = report.values;
|
||||
final type = report.type;
|
||||
if (type == 'inbound-rtp' && values['kind'] == 'video') {
|
||||
// 关键数据打印，用于排查 totalDecodeTime 是否存在于当前平台的统计中
|
||||
debugPrint('[WebRtcController] inbound-rtp: totalDecodeTime=${values['totalDecodeTime']}, '
|
||||
'googDecodeMs=${values['googDecodeMs']}, '
|
||||
'framesDecoded=${values['framesDecoded']}, '
|
||||
'fps=${values['framesPerSecond']}');
|
||||
|
||||
width = values['frameWidth'] ?? '-';
|
||||
height = values['frameHeight'] ?? '-';
|
||||
fps = values['framesPerSecond'] ?? '-';
|
||||
|
||||
final j = values['jitter'];
|
||||
if (j is num) jitter = (j * 1000).toStringAsFixed(0);
|
||||
|
||||
final pr = values['packetsReceived'];
|
||||
final pl = values['packetsLost'];
|
||||
if (pr is num && pl is num) {
|
||||
final total = pr + pl;
|
||||
lossRate = total > 0 ? ((pl / total) * 100).toStringAsFixed(1) : '0.0';
|
||||
}
|
||||
|
||||
final fd = values['framesDropped'];
|
||||
if (fd is num) framesDropped = fd;
|
||||
|
||||
// 核心：解析解码统计（兼容标准字段与旧版 Google 私有字段）
|
||||
currentTotalDecodeTime = (values['totalDecodeTime'] as num?)?.toDouble() ??
|
||||
((values['googDecodeMs'] as num?)?.toDouble() ?? 0.0) / 1000.0;
|
||||
currentFramesDecoded = (values['framesDecoded'] as num?)?.toInt() ??
|
||||
(values['googFramesDecoded'] as num?)?.toInt() ?? 0;
|
||||
|
||||
final codecId = values['codecId'];
|
||||
if (codecId != null) {
|
||||
final codecReport = reports.where((r) => r.id == codecId).firstOrNull;
|
||||
final mime = codecReport?.values['mimeType'];
|
||||
if (mime is String && mime.startsWith('video/')) {
|
||||
codec = mime.substring(6);
|
||||
}
|
||||
}
|
||||
} else if (type == 'candidate-pair') {
|
||||
if (values['nominated'] == true) {
|
||||
hasNominatedPair = true;
|
||||
final rtt = values['currentRoundTripTime'];
|
||||
if (rtt is num) {
|
||||
delay = (rtt * 1000).toStringAsFixed(0);
|
||||
}
|
||||
// candidate-pair 的 bytesReceived/bytesSent 为整条连接（视频+DataChannel+ICE）的累计值。
|
||||
final br = values['bytesReceived'];
|
||||
final bs = values['bytesSent'];
|
||||
if (br is num) bytesReceived = br.toInt();
|
||||
if (bs is num) bytesSent = bs.toInt();
|
||||
final lc = values['localCandidateId'];
|
||||
final rc = values['remoteCandidateId'];
|
||||
if (lc is String) localCandidateId = lc;
|
||||
if (rc is String) remoteCandidateId = rc;
|
||||
}
|
||||
} else if (type == 'data-channel') {
|
||||
if (values['label'] == kDataChannelLabel) {
|
||||
final ms = values['messagesSent'];
|
||||
final mr = values['messagesReceived'];
|
||||
if (ms is num) dcMessagesSent = ms.toInt();
|
||||
if (mr is num) dcMessagesReceived = mr.toInt();
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// 计算每秒网速（与上次采样差值）。
|
||||
String downSpeed = '-';
|
||||
String upSpeed = '-';
|
||||
if (dtSec > 0) {
|
||||
final downBps = (bytesReceived - _prevBytesReceived) / dtSec;
|
||||
final upBps = (bytesSent - _prevBytesSent) / dtSec;
|
||||
downSpeed = _formatSpeed(downBps);
|
||||
upSpeed = _formatSpeed(upBps);
|
||||
}
|
||||
_prevBytesReceived = bytesReceived;
|
||||
_prevBytesSent = bytesSent;
|
||||
_prevStatsTimestamp = now;
|
||||
|
||||
// 连接类型（P2P/TURN）与候选协议（udp/tcp）。
|
||||
String connType = '-';
|
||||
String protocol = '-';
|
||||
if (hasNominatedPair) {
|
||||
connType =
|
||||
_resolveConnectionType(reports, localCandidateId, remoteCandidateId);
|
||||
protocol = _resolveProtocol(reports, localCandidateId, remoteCandidateId);
|
||||
}
|
||||
|
||||
// 连接时长。
|
||||
String duration = '-';
|
||||
if (_connectedAt != null) {
|
||||
final secs = ((now - _connectedAt!) / 1000).floor();
|
||||
final m = (secs ~/ 60).toString().padLeft(2, '0');
|
||||
final s = (secs % 60).toString().padLeft(2, '0');
|
||||
duration = '$m:$s';
|
||||
}
|
||||
|
||||
// 控制 DataChannel 状态。
|
||||
final dcState = _dataChannel?.state;
|
||||
final dcStatus = dcState == RTCDataChannelState.RTCDataChannelOpen
|
||||
? '已连接'
|
||||
: (dcState == null ? '未连接' : dcState.name);
|
||||
|
||||
// —— 分模式统计解码耗时 ——
|
||||
String? decodeInfo;
|
||||
// 基础行要展示的解码耗时：WebRTC 模式用 inbound-rtp 估算，自编码模式用原生平均。
|
||||
double nativeAvg = 0;
|
||||
|
||||
if (_currentStreamMode == streamModeSelfCodec) {
|
||||
// 【自编码模式】：视频走 DataChannel 透传裸码流，WebRTC 的视频统计为空。
|
||||
// 使用原生解码器上报的真实分辨率、帧率与耗时。
|
||||
if (_selfCodecDecoder != null) {
|
||||
final ds = await _selfCodecDecoder!.getStats();
|
||||
if (ds != null) {
|
||||
final last = (ds['lastMs'] as num?)?.toDouble() ?? 0;
|
||||
final avg = (ds['avgMs'] as num?)?.toDouble() ?? 0;
|
||||
final peak = (ds['maxMs'] as num?)?.toDouble() ?? 0;
|
||||
final cnt = (ds['count'] as num?)?.toInt() ?? 0;
|
||||
nativeAvg = avg;
|
||||
if (_selfCodecWidth > 0 && _selfCodecHeight > 0) {
|
||||
width = _selfCodecWidth;
|
||||
height = _selfCodecHeight;
|
||||
}
|
||||
double decodeFps = 0;
|
||||
if (dtSec > 0 && cnt > 0) {
|
||||
decodeFps = (cnt - _prevDecodeCount) / dtSec;
|
||||
}
|
||||
_prevDecodeCount = cnt;
|
||||
if (decodeFps > 0) fps = decodeFps.round();
|
||||
decodeInfo = '解码耗时: ${last.toStringAsFixed(1)} ms 平均: ${avg.toStringAsFixed(1)} ms 峰值: ${peak.toStringAsFixed(1)} ms\n'
|
||||
'解码帧率: ${decodeFps.toStringAsFixed(0)} fps 已解码: $cnt 帧';
|
||||
}
|
||||
}
|
||||
// 自编码模式下，WebRTC 的相关计数建议重置，避免切回时出现跳变。
|
||||
_prevTotalDecodeTime = currentTotalDecodeTime;
|
||||
_prevFramesDecoded = currentFramesDecoded;
|
||||
} else {
|
||||
// 【WebRTC 模式】：优先使用标准的 getStats() 增量计算耗时。
|
||||
double avg = 0;
|
||||
if (currentFramesDecoded > 0) {
|
||||
if (_prevFramesDecoded > 0 && currentFramesDecoded > _prevFramesDecoded) {
|
||||
final deltaMs = (currentTotalDecodeTime - _prevTotalDecodeTime) * 1000.0;
|
||||
final deltaFrames = currentFramesDecoded - _prevFramesDecoded;
|
||||
if (deltaFrames > 0) avg = deltaMs / deltaFrames;
|
||||
} else if (currentTotalDecodeTime > 0) {
|
||||
// 兜底：第一次采样或帧数无增量时，用全量平均作为初始参考
|
||||
avg = (currentTotalDecodeTime * 1000.0) / currentFramesDecoded;
|
||||
}
|
||||
}
|
||||
|
||||
if (avg > 0) {
|
||||
nativeAvg = avg;
|
||||
decodeInfo = '解码耗时 (stats): ${avg.toStringAsFixed(1)} ms\n'
|
||||
'解码帧率: $fps fps 已解码: $currentFramesDecoded 帧';
|
||||
} else {
|
||||
// 备选方案：如果 getStats 没报数据（部分平台/环境限制），则回退到原生 VideoSink 探针实测模式。
|
||||
_webRtcDecodeProbe = await _getWebRtcDecodeStats();
|
||||
final p = _webRtcDecodeProbe;
|
||||
if (p != null && p.totalFrames > 0) {
|
||||
nativeAvg = p.avgMs;
|
||||
decodeInfo =
|
||||
'解码耗时 (probe): ${p.lastMs.toStringAsFixed(1)} ms 平均: ${p.avgMs.toStringAsFixed(1)} ms 峰值: ${p.peakMs.toStringAsFixed(1)} ms\n'
|
||||
'解码帧率: ${p.fpsNow} fps (平均 ${p.fpsAvg} fps) 已解码: ${p.totalFrames} 帧';
|
||||
}
|
||||
}
|
||||
// WebRTC 模式下，自编码的计数重置。
|
||||
_prevDecodeCount = 0;
|
||||
_prevTotalDecodeTime = currentTotalDecodeTime;
|
||||
_prevFramesDecoded = currentFramesDecoded;
|
||||
}
|
||||
|
||||
String decodeTimeText = '-';
|
||||
if (decodeInfo != null && nativeAvg > 0) {
|
||||
decodeTimeText = '${nativeAvg.toStringAsFixed(1)} ms';
|
||||
}
|
||||
|
||||
final base = '分辨率: ${width}x$height 帧率: $fps 延迟: $delay ms 解码耗时: $decodeTimeText\n'
|
||||
'解码格式: $codec 抖动: $jitter ms 丢包: $lossRate%\n'
|
||||
'丢帧: $framesDropped ↓下载: $downSpeed ↑上传: $upSpeed\n'
|
||||
'连接类型: $connType ($protocol) 时长: $duration\n'
|
||||
'控制通道: $dcStatus 收发: $dcMessagesSent/$dcMessagesReceived';
|
||||
|
||||
return decodeInfo != null ? '$base\n$decodeInfo' : base;
|
||||
}
|
||||
|
||||
/// 依据选定候选对的本地/远端候选类型，返回 P2P / TURN 中继描述。
|
||||
String _resolveConnectionType(
|
||||
List<StatsReport> reports,
|
||||
String? localCandidateId,
|
||||
String? remoteCandidateId,
|
||||
) {
|
||||
String? typeOf(String? id) {
|
||||
if (id == null) return null;
|
||||
final report = reports.where((r) => r.id == id).firstOrNull;
|
||||
final t = report?.values['candidateType'];
|
||||
return t is String ? t : null;
|
||||
}
|
||||
|
||||
final local = typeOf(localCandidateId);
|
||||
final remote = typeOf(remoteCandidateId);
|
||||
final isRelay = local == 'relay' || remote == 'relay';
|
||||
|
||||
if (isRelay) {
|
||||
// 进一步用本地候选具体类型细化（优先说明本端是否经 TURN 中继）。
|
||||
if (local == 'relay') return 'TURN 中继 (本端经服务器)';
|
||||
return 'TURN 中继';
|
||||
}
|
||||
// host: 同一局域网直连; srflx/prflx: 经 NAT 打洞的 P2P。
|
||||
return 'P2P 直连';
|
||||
}
|
||||
|
||||
/// 返回选定候选对的传输协议（udp/tcp）。
|
||||
String _resolveProtocol(
|
||||
List<StatsReport> reports,
|
||||
String? localCandidateId,
|
||||
String? remoteCandidateId,
|
||||
) {
|
||||
String protoOf(String? id) {
|
||||
if (id == null) return '-';
|
||||
final report = reports.where((r) => r.id == id).firstOrNull;
|
||||
final p = report?.values['protocol'];
|
||||
return p is String ? p : '-';
|
||||
}
|
||||
|
||||
final local = protoOf(localCandidateId);
|
||||
final remote = protoOf(remoteCandidateId);
|
||||
if (local != '-' && remote != '-' && local != remote) {
|
||||
return '$local/$remote';
|
||||
}
|
||||
return local != '-' ? local : remote;
|
||||
}
|
||||
|
||||
/// 将字节/秒格式化为人类可读字符串（B/s、KB/s、MB/s）。
|
||||
String _formatSpeed(double bytesPerSecond) {
|
||||
if (bytesPerSecond >= 1024 * 1024) {
|
||||
return '${(bytesPerSecond / (1024 * 1024)).toStringAsFixed(1)} MB/s';
|
||||
} else if (bytesPerSecond >= 1024) {
|
||||
return '${(bytesPerSecond / 1024).toStringAsFixed(1)} KB/s';
|
||||
}
|
||||
return '${bytesPerSecond.toStringAsFixed(0)} B/s';
|
||||
}
|
||||
|
||||
/// 释放所有资源。
|
||||
Future<void> close() async {
|
||||
try {
|
||||
await _dataChannel?.close();
|
||||
} catch (_) {}
|
||||
try {
|
||||
await _detachWebRtcDecodeProbe();
|
||||
} catch (_) {}
|
||||
try {
|
||||
await _pc?.close();
|
||||
} catch (_) {}
|
||||
try {
|
||||
await renderer.dispose();
|
||||
} catch (_) {}
|
||||
_dataChannel = null;
|
||||
try {
|
||||
await _videoDataChannel?.close();
|
||||
} catch (_) {}
|
||||
_videoDataChannel = null;
|
||||
try {
|
||||
await _selfCodecDecoder?.dispose();
|
||||
} catch (_) {}
|
||||
_selfCodecDecoder = null;
|
||||
_currentStreamMode = streamModeWebRtc;
|
||||
onSelfCodecLost?.call();
|
||||
onSelfCodecReady = null;
|
||||
onStreamModeReport = null;
|
||||
onResolutionReported = null;
|
||||
onFpsReport = null;
|
||||
onSelfCodecNotSupported = null;
|
||||
_pc = null;
|
||||
// 重置网速统计基线，避免下次连接首帧显示错误的瞬时速率。
|
||||
_prevBytesReceived = 0;
|
||||
_prevBytesSent = 0;
|
||||
_prevStatsTimestamp = 0;
|
||||
_prevDecodeCount = 0;
|
||||
_prevTotalDecodeTime = 0.0;
|
||||
_prevFramesDecoded = 0;
|
||||
_webRtcDecodeProbe = null;
|
||||
_selfCodecWidth = 0;
|
||||
_selfCodecHeight = 0;
|
||||
_connectedAt = null;
|
||||
}
|
||||
}
|
||||
Reference in New Issue
Block a user