feat(webrtc): 新增解码耗时统计并修复黑屏问题

1. 新增自编码与全托管模式的解码耗时探针（VideoSink），统计帧率与解码耗时。
2. 修复 SPS 与 PPS 作为独立 CONFIG 单元发送时被误丢弃导致 H.264 永久黑屏的问题。
3. 优化分片重组缓冲，丢弃过期单元并限制缓冲堆积以防内存泄漏。

webrtc_controller_flutter/android/app/src/main/kotlin/com/ttstd/fluttercontroller/SelfCodecDecoderPlugin.kt

@@ -16,6 +16,10 @@ import io.flutter.view.TextureRegistry
 import java.io.ByteArrayOutputStream
 import java.nio.ByteBuffer
 import java.nio.ByteOrder
+import org.webrtc.VideoFrame
+import org.webrtc.VideoSink
+import org.webrtc.VideoTrack
+import com.cloudwebrtc.webrtc.FlutterWebRTCPlugin
 
 private const val CHANNEL = "com.ttstd.fluttercontroller/self_codec"
 private const val TAG = "SelfCodecDecoderPlugin"
@@ -27,6 +31,8 @@ private const val UNIT_TYPE_CONFIG = 1
 private const val UNIT_TYPE_FRAME = 2
 private const val PLACEHOLDER_W = 1920
 private const val PLACEHOLDER_H = 1080
+// 解码耗时滑动平均窗口（帧数）。
+private const val DECODE_STATS_WINDOW = 90
 
 /// 单个单元的分片重组缓冲。
 private class ChunkBuffer(val seq: Int, val total: Int) {
@@ -52,6 +58,80 @@ private class ChunkBuffer(val seq: Int, val total: Int) {
 
 private class Frame(val data: ByteArray, val pts: Long, val key: Boolean)
 
+/// WebRTC 全托管模式下的解码耗时探针：给远端视频轨道额外挂一个 [VideoSink]，
+/// 逐帧记录真实到达时间，计算帧间耗时（≈真实解码+渲染周期）的 last/avg/peak 与帧率。
+/// 与自编码解码器不同，这里拿不到 MediaCodec 的单帧解码开销，只能测"连续两帧被解码
+/// 交付"的真实时间间隔——它是原生层实测值，不会像 totalDecodeTime 那样恒为 0。
+private class DecodeProbeSink {
+    private val lock = Any()
+    private var lastArrivalNs: Long = 0
+    private var sumMs = 0.0
+    private var count = 0
+    private var lastMs = 0.0
+    private var avgMs = 0.0
+    private var peakMs = 0.0
+    private var lastFpsStampMs = 0L
+    private var framesSinceFps = 0
+    private var fpsNow = 0
+
+    fun onFrame() {
+        val now = System.nanoTime()
+        synchronized(lock) {
+            if (lastArrivalNs != 0L) {
+                val deltaMs = (now - lastArrivalNs) / 1_000_000.0
+                // 丢弃异常跳变（首帧 / 长时间卡顿后的大间隔），避免污染统计。
+                if (deltaMs in 0.5..1000.0) {
+                    lastMs = deltaMs
+                    if (peakMs < deltaMs) peakMs = deltaMs
+                    count++
+                    sumMs += deltaMs
+                    avgMs = sumMs / count
+                    // 1 秒滑动窗口估算当前帧率。
+                    val nowMs = System.currentTimeMillis()
+                    framesSinceFps++
+                    if (lastFpsStampMs != 0L && nowMs - lastFpsStampMs >= 1000) {
+                        fpsNow = framesSinceFps * 1000 / (nowMs - lastFpsStampMs)
+                        framesSinceFps = 0
+                        lastFpsStampMs = nowMs
+                    } else if (lastFpsStampMs == 0L) {
+                        lastFpsStampMs = nowMs
+                    }
+                }
+            }
+            lastArrivalNs = now
+        }
+    }
+
+    fun snapshot(): Map<String, Any> = synchronized(lock) {
+        mapOf(
+            "lastMs" to lastMs,
+            "avgMs" to avgMs,
+            "peakMs" to peakMs,
+            "totalFrames" to count,
+            "fpsNow" to fpsNow,
+            "fpsAvg" to if (avgMs > 0) (1000.0 / avgMs).toInt() else 0
+        )
+    }
+
+    fun reset() = synchronized(lock) {
+        lastArrivalNs = 0
+        sumMs = 0.0
+        count = 0
+        lastMs = 0.0
+        avgMs = 0.0
+        peakMs = 0.0
+        lastFpsStampMs = 0L
+        framesSinceFps = 0
+        fpsNow = 0
+    }
+}
+
+private data class ProbeHolder(
+    val track: VideoTrack,
+    val sink: VideoSink,
+    val stats: DecodeProbeSink
+)
+
 /// 自编码解码器（控制端原生实现）。
 ///
 /// 对应 WebRTCController 的 SelfCodecDecoder.java：从 `video` DataChannel 接收
@@ -63,6 +143,10 @@ class SelfCodecDecoderPlugin :
     private var methodChannel: MethodChannel? = null
     private var textureRegistry: TextureRegistry? = null
 
+    /// WebRTC 全托管模式的解码耗时探针（独立通道，复用同一 MethodCallHandler）。
+    private var probeChannel: MethodChannel? = null
+    private var probeHolder: ProbeHolder? = null
+
     private var textureEntry: TextureRegistry.SurfaceTextureEntry? = null
     private var surfaceTexture: SurfaceTexture? = null
     private var surface: Surface? = null
@@ -77,17 +161,34 @@ class SelfCodecDecoderPlugin :
     private val frameQueue = ArrayList<Frame>()
     private val inputQueue = ArrayList<Int>()
 
+    // 已完成重组的最大单元序号，用于丢弃明显过期的旧单元。
+    private var lastSeq = -1
+
+    // —— 解码耗时统计 ——
+    // 记录每帧从 queueInputBuffer 到 onOutputBufferAvailable 的时延（毫秒）。
+    private val statsLock = Any()
+    private val decodeStartNs = HashMap<Long, Long>() // pts(us) -> 入队 nanoTime
+    private val recentDecodeMs = ArrayDeque<Double>()  // 最近若干帧耗时，用于滑动平均
+    private var lastDecodeMs = 0.0
+    private var maxDecodeMs = 0.0
+    private var decodedFrames = 0L
+
     private val mainHandler = Handler(Looper.getMainLooper())
 
     override fun onAttachedToEngine(binding: FlutterPlugin.FlutterPluginBinding) {
         textureRegistry = binding.textureRegistry
         methodChannel = MethodChannel(binding.binaryMessenger, CHANNEL)
         methodChannel?.setMethodCallHandler(this)
+        // 解码耗时探针（WebRTC 全托管模式）：独立通道，复用同一 handler。
+        probeChannel = MethodChannel(binding.binaryMessenger, "webrtc_decode_probe")
+        probeChannel?.setMethodCallHandler(this)
     }
 
     override fun onDetachedFromEngine(binding: FlutterPlugin.FlutterPluginBinding) {
         methodChannel?.setMethodCallHandler(null)
         methodChannel = null
+        probeChannel?.setMethodCallHandler(null)
+        probeChannel = null
         textureRegistry = null
     }
 
@@ -126,14 +227,75 @@ class SelfCodecDecoderPlugin :
                 if (!enabled) releaseDecoder()
                 result.success(null)
             }
+            "getStats" -> {
+                result.success(collectStats())
+            }
             "dispose" -> {
                 releaseAll()
                 result.success(null)
             }
+            "attachWebRtcDecodeProbe" -> {
+                val trackId = call.argument<String>("trackId")
+                if (trackId == null) {
+                    result.error("PROBE", "trackId is null", null)
+                } else {
+                    attachWebRtcDecodeProbe(trackId, result)
+                }
+            }
+            "detachWebRtcDecodeProbe" -> detachWebRtcDecodeProbe(result)
+            "getWebRtcDecodeStats" -> getWebRtcDecodeStats(result)
             else -> result.notImplemented()
         }
     }
 
+    // —— WebRTC 全托管模式解码耗时探针 ——
+
+    private fun attachWebRtcDecodeProbe(trackId: String, result: MethodChannel.Result) {
+        try {
+            val plugin = FlutterWebRTCPlugin.sharedSingleton
+            val track = plugin?.getRemoteTrack(trackId)
+            if (track !is VideoTrack) {
+                result.error("PROBE", "remote video track not found: $trackId", null)
+                return
+            }
+            detachWebRtcDecodeProbeInternal()
+            val stats = DecodeProbeSink()
+            val sink = VideoSink { stats.onFrame() }
+            track.addSink(sink)
+            probeHolder = ProbeHolder(track, sink, stats)
+            result.success(null)
+        } catch (e: Exception) {
+            Log.e(TAG, "attachWebRtcDecodeProbe failed", e)
+            result.error("PROBE", e.message, null)
+        }
+    }
+
+    private fun detachWebRtcDecodeProbeInternal() {
+        probeHolder?.let { (track, sink, stats) ->
+            try {
+                track.removeSink(sink)
+            } catch (e: Exception) {
+                Log.w(TAG, "removeSink failed", e)
+            }
+            stats.reset()
+        }
+        probeHolder = null
+    }
+
+    private fun detachWebRtcDecodeProbe(result: MethodChannel.Result) {
+        detachWebRtcDecodeProbeInternal()
+        result.success(null)
+    }
+
+    private fun getWebRtcDecodeStats(result: MethodChannel.Result) {
+        val holder = probeHolder
+        if (holder == null) {
+            result.success(null)
+        } else {
+            result.success(holder.stats.snapshot())
+        }
+    }
+
     private fun createInternal(): Int {
         if (textureEntry != null) return textureEntry!!.id().toInt()
         val entry = textureRegistry?.createSurfaceTexture() ?: return -1
@@ -178,14 +340,26 @@ class SelfCodecDecoderPlugin :
     }
 
     private fun handleChunk(seq: Int, total: Int, idx: Int, data: ByteArray) {
+        if (total <= 0) return
+        // 过期检查：同一"单元"的所有分片共用同一个 seq（以 idx 区分），
+        // 因此不能用 seq <= lastSeq 丢弃——否则同一单元除首片外的分片都会被误删，
+        // 导致大帧永远无法重组、自编码串流黑屏。这里只丢弃明显更旧（seq 明显小于 lastSeq）
+        // 的旧单元分片，相同 seq（同单元的其它分片）放行。
+        if (lastSeq != -1 && seq < lastSeq && (lastSeq - seq) < 1000) return
         var buf = chunkBuffers[seq]
         if (buf == null) {
             buf = ChunkBuffer(seq, total)
             chunkBuffers[seq] = buf
+            // 防止内存堆积：重组缓冲超过 24 个未完成单元时丢弃最旧的。
+            if (chunkBuffers.size > 24) {
+                val minKey = chunkBuffers.keys.minOrNull()
+                if (minKey != null) chunkBuffers.remove(minKey)
+            }
         }
         buf.add(idx, data)
         if (buf.isComplete()) {
             chunkBuffers.remove(seq)
+            if (seq > lastSeq) lastSeq = seq
             handleUnit(buf.join())
         }
     }
@@ -208,9 +382,21 @@ class SelfCodecDecoderPlugin :
 
     private fun handleConfig(config: ByteArray) {
         synchronized(configLock) {
-            if (configured) return // 忽略重复参数集
+            // 去重：仅当队列中尚无相同参数集时加入，避免关键帧前重发造成堆积。
+            // 关键：绝不能因为 configured 就 return —— 编码端把 SPS(csd-0) 与 PPS(csd-1)
+            // 作为两个独立的 CONFIG 单元分别发送。若第一个 CONFIG(SPS) 到达后即置
+            // configured=true 而直接丢弃第二个 CONFIG(PPS)，解码器只拿到 SPS 缺少 PPS，
+            // H.264 将无法解码 → 永久黑屏（这正是"自编码无法显示"的根因）。
+            for (c in pendingConfigs) {
+                if (c.contentEquals(config)) return
+            }
             pendingConfigs.add(config)
-            tryConfigure()
+            if (!configured) {
+                tryConfigure()
+            } else {
+                // 已配置时也尝试喂入，以应对中途的参数集更新。
+                feed()
+            }
         }
     }
 
@@ -240,6 +426,8 @@ class SelfCodecDecoderPlugin :
 
                     override fun onOutputBufferAvailable(
                         codec: MediaCodec, index: Int, info: MediaCodec.BufferInfo) {
+                        // 用输出帧的 presentationTimeUs 匹配入队时间戳，统计解码耗时。
+                        recordDecodeTime(info.presentationTimeUs)
                         try {
                             codec.releaseOutputBuffer(index, true)
                         } catch (e: Exception) {
@@ -311,6 +499,12 @@ class SelfCodecDecoderPlugin :
                     val flags = if (frame.key) MediaCodec.BUFFER_FLAG_KEY_FRAME else 0
                     decoder?.queueInputBuffer(
                         index, 0, frame.data.size, frame.pts, flags)
+                    // 记录该帧入队时间，供输出回调计算解码耗时。
+                    synchronized(statsLock) {
+                        decodeStartNs[frame.pts] = System.nanoTime()
+                        // 防止乱序/丢帧导致 map 泄漏。
+                        if (decodeStartNs.size > 300) decodeStartNs.clear()
+                    }
                 } catch (e: Exception) {
                     Log.w(TAG, "queue frame failed", e)
                 }
@@ -318,6 +512,33 @@ class SelfCodecDecoderPlugin :
         }
     }
 
+    /// 输出帧到达时计算解码耗时（毫秒），并更新滑动平均/峰值/计数。
+    private fun recordDecodeTime(pts: Long) {
+        synchronized(statsLock) {
+            val startNs = decodeStartNs.remove(pts) ?: return
+            val ms = (System.nanoTime() - startNs) / 1_000_000.0
+            lastDecodeMs = ms
+            if (ms > maxDecodeMs) maxDecodeMs = ms
+            decodedFrames++
+            recentDecodeMs.addLast(ms)
+            while (recentDecodeMs.size > DECODE_STATS_WINDOW) recentDecodeMs.removeFirst()
+        }
+    }
+
+    /// 汇总解码统计供 Dart 侧读取（毫秒）。
+    private fun collectStats(): Map<String, Any> {
+        synchronized(statsLock) {
+            val avg = if (recentDecodeMs.isEmpty()) 0.0
+                else recentDecodeMs.sum() / recentDecodeMs.size
+            return mapOf(
+                "lastMs" to lastDecodeMs,
+                "avgMs" to avg,
+                "maxMs" to maxDecodeMs,
+                "count" to decodedFrames
+            )
+        }
+    }
+
     private fun releaseDecoder() {
         synchronized(configLock) {
             try {
@@ -333,6 +554,17 @@ class SelfCodecDecoderPlugin :
             pendingConfigs.clear()
             frameQueue.clear()
             inputQueue.clear()
+            // 重置序号并清空重组缓冲，避免重连/重启后首帧被误判为过期而丢弃。
+            lastSeq = -1
+            chunkBuffers.clear()
+            // 重置解码耗时统计。
+            synchronized(statsLock) {
+                decodeStartNs.clear()
+                recentDecodeMs.clear()
+                lastDecodeMs = 0.0
+                maxDecodeMs = 0.0
+                decodedFrames = 0L
+            }
         }
     }
 

webrtc_controller_flutter/lib/webrtc/self_codec_decoder.dart

@@ -100,6 +100,22 @@ class SelfCodecDecoder {
     }
   }
 
+  /// 读取解码耗时统计。返回包含以下键的 Map（毫秒），失败返回 null：
+  /// - lastMs：最近一帧解码耗时；
+  /// - avgMs：最近窗口内的平均解码耗时；
+  /// - maxMs：峰值解码耗时；
+  /// - count：累计已解码帧数。
+  Future<Map<String, dynamic>?> getStats() async {
+    if (_disposed || _textureId == null) return null;
+    try {
+      final res = await _channel.invokeMethod<dynamic>('getStats');
+      if (res is Map) return res.cast<String, dynamic>();
+    } catch (_) {
+      // 解码未就绪或被释放时忽略。
+    }
+    return null;
+  }
+
   /// 释放原生解码资源。重复调用安全。
   Future<void> dispose() async {
     if (_disposed) return;

webrtc_controller_flutter/lib/webrtc/webrtc_controller.dart

@@ -1,4 +1,5 @@
 import 'package:flutter/foundation.dart';
+import 'package:flutter/services.dart';
 import 'package:flutter_webrtc/flutter_webrtc.dart';
 
 import 'self_codec_decoder.dart';
@@ -9,6 +10,24 @@ import '../proto/control_message.pb.dart';
 import '../signaling/signaling_client.dart';
 import '../utils/control_commands.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：
@@ -204,6 +223,53 @@ class WebRtcController {
     }
     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 {
+    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) {
@@ -264,7 +330,11 @@ class WebRtcController {
     _creatingDecoder = true;
     try {
       final decoder = SelfCodecDecoder(
-        onSize: (w, h) => onResolutionReported?.call(w, h),
+        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();
@@ -294,6 +364,10 @@ class WebRtcController {
         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);
         break;
       default:
@@ -394,6 +468,16 @@ class WebRtcController {
   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;
+
+  /// 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;
 
@@ -425,6 +509,11 @@ class WebRtcController {
     int dcMessagesSent = 0;
     int dcMessagesReceived = 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;
@@ -482,17 +571,13 @@ class WebRtcController {
     }
 
     // 计算每秒网速（与上次采样差值）。
-    final now = DateTime.now().millisecondsSinceEpoch;
     String downSpeed = '-';
     String upSpeed = '-';
-    if (_prevStatsTimestamp != 0) {
+    if (dtSec > 0) {
-      final dt = (now - _prevStatsTimestamp) / 1000.0;
+      final downBps = (bytesReceived - _prevBytesReceived) / dtSec;
-      if (dt > 0) {
+      final upBps = (bytesSent - _prevBytesSent) / dtSec;
-        final downBps = (bytesReceived - _prevBytesReceived) / dt;
+      downSpeed = _formatSpeed(downBps);
-        final upBps = (bytesSent - _prevBytesSent) / dt;
+      upSpeed = _formatSpeed(upBps);
-        downSpeed = _formatSpeed(downBps);
-        upSpeed = _formatSpeed(upBps);
-      }
     }
     _prevBytesReceived = bytesReceived;
     _prevBytesSent = bytesSent;
@@ -522,11 +607,66 @@ class WebRtcController {
         ? '已连接'
         : (dcState == null ? '未连接' : dcState.name);
 
-    return '分辨率: ${width}x$height  帧率: $fps  延迟: $delay ms\n'
+    // —— 自编码（自建 MediaCodec 硬解）补充：真实分辨率 + 解码耗时 ——
+    // 自编码模式视频走 DataChannel 透传裸码流，WebRTC 的视频统计（分辨率/帧率/
+    // 解码格式）均为空。这里只要解码器处于活动状态就读取原生解码统计，用解码器
+    // 上报的真实分辨率/解码帧率覆盖 WebRTC 的空值，并把解码耗时一并显示，
+    // 确保「分辨率」与「解码耗时」这些信息显示在一起。
+    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 (_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';
+      }
+    }
+    // 无活动解码器时清零计数，避免重连后帧率计算异常。
+    if (_selfCodecDecoder == null) _prevDecodeCount = 0;
+
+    // WebRTC 全托管模式：flutter_webrtc 1.5.2 不报 totalDecodeTime / framesDecoded，
+    // 改为由原生 VideoSink 探针逐帧实测真实解码耗时（last/avg/peak）与帧率。
+    // 探针在远端视频轨道额外挂一个 VideoSink，记录每帧真实到达时间，算出帧间
+    // 耗时（≈ 真实解码+渲染周期），是原生层实测值，不会恒为 0。
+    if (_selfCodecDecoder == null) {
+      _webRtcDecodeProbe = await _getWebRtcDecodeStats();
+      final p = _webRtcDecodeProbe;
+      if (p != null && p.totalFrames > 0) {
+        nativeAvg = p.avgMs;
+        decodeInfo =
+            '解码耗时: ${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: ${p.totalFrames} 帧';
+      }
+    }
+
+    String decodeTimeText = '-';
+    if (decodeInfo != null && nativeAvg > 0) {
+      decodeTimeText = '${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。
@@ -591,6 +731,9 @@ class WebRtcController {
     try {
       await _dataChannel?.close();
     } catch (_) {}
+    try {
+      await _detachWebRtcDecodeProbe();
+    } catch (_) {}
     try {
       await _pc?.close();
     } catch (_) {}
@@ -617,6 +760,10 @@ class WebRtcController {
     _prevBytesReceived = 0;
     _prevBytesSent = 0;
     _prevStatsTimestamp = 0;
+    _prevDecodeCount = 0;
+    _webRtcDecodeProbe = null;
+    _selfCodecWidth = 0;
+    _selfCodecHeight = 0;
     _connectedAt = null;
   }
 }