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。
This commit is contained in:
TongTongStudio
@@ -1,4 +1,5 @@
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import 'package:flutter/foundation.dart';
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import 'package:flutter/services.dart';
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import 'package:flutter_webrtc/flutter_webrtc.dart';
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import 'self_codec_decoder.dart';
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@@ -9,6 +10,24 @@ import '../proto/control_message.pb.dart';
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import '../signaling/signaling_client.dart';
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import '../utils/control_commands.dart';
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/// 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。
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class _DecodeProbeStats {
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final double lastMs;
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final double avgMs;
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final double peakMs;
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final int totalFrames;
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final int fpsNow;
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final int fpsAvg;
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const _DecodeProbeStats({
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required this.lastMs,
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required this.avgMs,
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required this.peakMs,
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required this.totalFrames,
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required this.fpsNow,
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required this.fpsAvg,
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});
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}
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/// 封装 WebRTC 连接逻辑（对应 Android 端 WebRtcClient）。
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///
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/// 职责：
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@@ -204,6 +223,53 @@ class WebRtcController {
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}
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renderer.srcObject = stream;
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onRemoteStream?.call(renderer);
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// 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）。
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if (_selfCodecDecoder == null) {
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final trackId = event.track.id;
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if (trackId != null) _attachWebRtcDecodeProbe(trackId);
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}
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}
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/// 解码耗时探针通道（原生侧给远端视频轨道 attach 一个测量 VideoSink）。
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static const MethodChannel _decodeProbeChannel =
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MethodChannel('webrtc_decode_probe');
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Future<void> _attachWebRtcDecodeProbe(String trackId) async {
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try {
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await _decodeProbeChannel.invokeMethod<void>(
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'attachWebRtcDecodeProbe', {'trackId': trackId});
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} catch (e) {
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debugPrint('[decodeProbe] attach failed: $e');
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}
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}
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Future<void> _detachWebRtcDecodeProbe() async {
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try {
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await _decodeProbeChannel.invokeMethod<void>('detachWebRtcDecodeProbe');
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} catch (e) {
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debugPrint('[decodeProbe] detach failed: $e');
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}
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}
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Future<_DecodeProbeStats?> _getWebRtcDecodeStats() async {
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try {
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final res =
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await _decodeProbeChannel.invokeMethod<dynamic>('getWebRtcDecodeStats');
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if (res is Map) {
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final m = res.cast<String, dynamic>();
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return _DecodeProbeStats(
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lastMs: (m['lastMs'] as num? ?? 0).toDouble(),
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avgMs: (m['avgMs'] as num? ?? 0).toDouble(),
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peakMs: (m['peakMs'] as num? ?? 0).toDouble(),
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totalFrames: (m['totalFrames'] as num? ?? 0).toInt(),
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fpsNow: (m['fpsNow'] as num? ?? 0).toInt(),
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fpsAvg: (m['fpsAvg'] as num? ?? 0).toInt(),
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);
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}
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} catch (e) {
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debugPrint('[decodeProbe] getStats failed: $e');
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}
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return null;
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}
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void _onDataChannel(RTCDataChannel channel) {
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@@ -264,7 +330,11 @@ class WebRtcController {
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_creatingDecoder = true;
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try {
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final decoder = SelfCodecDecoder(
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onSize: (w, h) => onResolutionReported?.call(w, h),
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onSize: (w, h) {
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_selfCodecWidth = w;
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_selfCodecHeight = h;
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onResolutionReported?.call(w, h);
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},
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onError: (err) => debugPrint('自编码解码器: $err'),
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);
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final textureId = await decoder.create();
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@@ -294,6 +364,10 @@ class WebRtcController {
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onStreamModeReport?.call(msg.streamMode);
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break;
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case Action.REPORT_RESOLUTION:
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if (msg.width > 0 && msg.height > 0) {
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_selfCodecWidth = msg.width;
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_selfCodecHeight = msg.height;
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}
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onResolutionReported?.call(msg.width, msg.height);
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break;
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default:
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@@ -394,6 +468,16 @@ class WebRtcController {
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int _prevBytesSent = 0;
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int _prevStatsTimestamp = 0;
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/// 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。
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int _prevDecodeCount = 0;
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/// WebRTC 全托管模式下，由原生 VideoSink 探针实测的解码耗时统计缓存。
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_DecodeProbeStats? _webRtcDecodeProbe;
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/// 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）。
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int _selfCodecWidth = 0;
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int _selfCodecHeight = 0;
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/// 连接建立时的时间戳（毫秒），用于计算连接时长。
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int? _connectedAt;
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@@ -425,6 +509,11 @@ class WebRtcController {
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int dcMessagesSent = 0;
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int dcMessagesReceived = 0;
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// 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。
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final now = DateTime.now().millisecondsSinceEpoch;
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final double dtSec =
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_prevStatsTimestamp != 0 ? (now - _prevStatsTimestamp) / 1000.0 : 0;
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for (final report in reports) {
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final values = report.values;
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final type = report.type;
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@@ -482,17 +571,13 @@ class WebRtcController {
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}
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// 计算每秒网速（与上次采样差值）。
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final now = DateTime.now().millisecondsSinceEpoch;
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String downSpeed = '-';
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String upSpeed = '-';
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if (_prevStatsTimestamp != 0) {
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final dt = (now - _prevStatsTimestamp) / 1000.0;
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if (dt > 0) {
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final downBps = (bytesReceived - _prevBytesReceived) / dt;
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final upBps = (bytesSent - _prevBytesSent) / dt;
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downSpeed = _formatSpeed(downBps);
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upSpeed = _formatSpeed(upBps);
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}
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if (dtSec > 0) {
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final downBps = (bytesReceived - _prevBytesReceived) / dtSec;
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final upBps = (bytesSent - _prevBytesSent) / dtSec;
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downSpeed = _formatSpeed(downBps);
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upSpeed = _formatSpeed(upBps);
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}
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_prevBytesReceived = bytesReceived;
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_prevBytesSent = bytesSent;
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@@ -522,11 +607,66 @@ class WebRtcController {
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? '已连接'
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: (dcState == null ? '未连接' : dcState.name);
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return '分辨率: ${width}x$height 帧率: $fps 延迟: $delay ms\n'
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// —— 自编码（自建 MediaCodec 硬解）补充：真实分辨率 + 解码耗时 ——
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// 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/
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// 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
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// 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，
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// 确保「分辨率」与「解码耗时」这些信息显示在一起。
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String? decodeInfo;
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// 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。
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double nativeAvg = 0;
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if (_selfCodecDecoder != null) {
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final ds = await _selfCodecDecoder!.getStats();
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if (ds != null) {
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final last = (ds['lastMs'] as num?)?.toDouble() ?? 0;
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final avg = (ds['avgMs'] as num?)?.toDouble() ?? 0;
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final peak = (ds['maxMs'] as num?)?.toDouble() ?? 0;
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final cnt = (ds['count'] as num?)?.toInt() ?? 0;
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nativeAvg = avg;
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if (_selfCodecWidth > 0 && _selfCodecHeight > 0) {
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width = _selfCodecWidth;
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height = _selfCodecHeight;
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}
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double decodeFps = 0;
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if (dtSec > 0 && cnt > 0) {
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decodeFps = (cnt - _prevDecodeCount) / dtSec;
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}
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_prevDecodeCount = cnt;
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if (decodeFps > 0) fps = decodeFps.round();
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decodeInfo = '解码耗时: ${last.toStringAsFixed(1)} ms 平均: ${avg.toStringAsFixed(1)} ms 峰值: ${peak.toStringAsFixed(1)} ms\n'
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'解码帧率: ${decodeFps.toStringAsFixed(0)} fps 已解码: $cnt 帧';
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}
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}
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// 无活动解码器时清零计数，避免重连后帧率计算异常。
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if (_selfCodecDecoder == null) _prevDecodeCount = 0;
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// WebRTC 全托管模式：flutter_webrtc 1.5.2 不报 totalDecodeTime / framesDecoded，
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// 改为由原生 VideoSink 探针逐帧实测真实解码耗时（last/avg/peak）与帧率。
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// 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
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// 耗时（≈ 真实解码+渲染周期），是原生层实测值，不会恒为 0。
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if (_selfCodecDecoder == null) {
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_webRtcDecodeProbe = await _getWebRtcDecodeStats();
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final p = _webRtcDecodeProbe;
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if (p != null && p.totalFrames > 0) {
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nativeAvg = p.avgMs;
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decodeInfo =
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'解码耗时: ${p.lastMs.toStringAsFixed(1)} ms 平均: ${p.avgMs.toStringAsFixed(1)} ms 峰值: ${p.peakMs.toStringAsFixed(1)} ms\n'
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'解码帧率: ${p.fpsNow} fps (平均 ${p.fpsAvg} fps) 已解码: ${p.totalFrames} 帧';
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}
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}
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String decodeTimeText = '-';
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if (decodeInfo != null && nativeAvg > 0) {
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decodeTimeText = '${nativeAvg.toStringAsFixed(1)} ms';
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}
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final base = '分辨率: ${width}x$height 帧率: $fps 延迟: $delay ms 解码耗时: $decodeTimeText\n'
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'解码格式: $codec 抖动: $jitter ms 丢包: $lossRate%\n'
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'丢帧: $framesDropped ↓下载: $downSpeed ↑上传: $upSpeed\n'
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'连接类型: $connType ($protocol) 时长: $duration\n'
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'控制通道: $dcStatus 收发: $dcMessagesSent/$dcMessagesReceived';
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return decodeInfo != null ? '$base\n$decodeInfo' : base;
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}
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/// 依据选定候选对的本地/远端候选类型，返回 P2P / TURN 中继描述。
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@@ -591,6 +731,9 @@ class WebRtcController {
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try {
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await _dataChannel?.close();
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} catch (_) {}
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try {
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await _detachWebRtcDecodeProbe();
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} catch (_) {}
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try {
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await _pc?.close();
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} catch (_) {}
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@@ -617,6 +760,10 @@ class WebRtcController {
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_prevBytesReceived = 0;
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_prevBytesSent = 0;
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_prevStatsTimestamp = 0;
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_prevDecodeCount = 0;
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_webRtcDecodeProbe = null;
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_selfCodecWidth = 0;
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_selfCodecHeight = 0;
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_connectedAt = null;
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}
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}
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