feat(decode): 用 WebRTC stats 替代原生探针计算解码耗时
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TongTongStudio
@@ -44,3 +44,10 @@ kotlin {
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flutter {
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source = "../.."
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}
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dependencies {
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// flutter_webrtc 依赖了 io.github.webrtc-sdk（org.webrtc.*），但它声明为
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// implementation，app 模块默认看不到 org.webrtc.VideoTrack/VideoSink 等类型。
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// 这里显式引入同版本，使解码耗时探针能直接引用 libwebrtc 的 VideoTrack/VideoSink。
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implementation("io.github.webrtc-sdk:android:144.7559.09")
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}
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@@ -16,7 +16,6 @@ import io.flutter.view.TextureRegistry
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import java.io.ByteArrayOutputStream
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import java.nio.ByteBuffer
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import java.nio.ByteOrder
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import org.webrtc.VideoFrame
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import org.webrtc.VideoSink
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import org.webrtc.VideoTrack
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import com.cloudwebrtc.webrtc.FlutterWebRTCPlugin
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@@ -90,7 +89,7 @@ private class DecodeProbeSink {
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val nowMs = System.currentTimeMillis()
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framesSinceFps++
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if (lastFpsStampMs != 0L && nowMs - lastFpsStampMs >= 1000) {
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fpsNow = framesSinceFps * 1000 / (nowMs - lastFpsStampMs)
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fpsNow = framesSinceFps * 1000 / (nowMs - lastFpsStampMs).toInt()
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framesSinceFps = 0
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lastFpsStampMs = nowMs
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} else if (lastFpsStampMs == 0L) {
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@@ -254,15 +253,22 @@ class SelfCodecDecoderPlugin :
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try {
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val plugin = FlutterWebRTCPlugin.sharedSingleton
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val track = plugin?.getRemoteTrack(trackId)
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if (track !is VideoTrack) {
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if (track == null) {
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result.error("PROBE", "remote video track not found: $trackId", null)
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return
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}
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// getRemoteTrack 返回 org.webrtc.MediaStreamTrack（超类），addSink 只在
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// 子类 org.webrtc.VideoTrack 上，用安全强转拿到 VideoTrack。
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val videoTrack = track as? VideoTrack
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if (videoTrack == null) {
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result.error("PROBE", "track is not a video track: $trackId", null)
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return
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}
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detachWebRtcDecodeProbeInternal()
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val stats = DecodeProbeSink()
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val sink = VideoSink { stats.onFrame() }
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track.addSink(sink)
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probeHolder = ProbeHolder(track, sink, stats)
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videoTrack.addSink(sink)
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probeHolder = ProbeHolder(videoTrack, sink, stats)
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result.success(null)
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} catch (e: Exception) {
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Log.e(TAG, "attachWebRtcDecodeProbe failed", e)
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@@ -235,6 +235,7 @@ class WebRtcController {
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MethodChannel('webrtc_decode_probe');
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Future<void> _attachWebRtcDecodeProbe(String trackId) async {
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debugPrint('[WebRtcController] Attaching decode probe for track: $trackId');
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try {
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await _decodeProbeChannel.invokeMethod<void>(
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'attachWebRtcDecodeProbe', {'trackId': trackId});
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@@ -471,6 +472,10 @@ class WebRtcController {
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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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/// 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。
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double _prevTotalDecodeTime = 0.0;
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int _prevFramesDecoded = 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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@@ -509,6 +514,9 @@ class WebRtcController {
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int dcMessagesSent = 0;
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int dcMessagesReceived = 0;
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double currentTotalDecodeTime = 0.0;
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int currentFramesDecoded = 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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@@ -518,6 +526,12 @@ class WebRtcController {
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final values = report.values;
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final type = report.type;
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if (type == 'inbound-rtp' && values['kind'] == 'video') {
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// 关键数据打印，用于排查 totalDecodeTime 是否存在于当前平台的统计中
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debugPrint('[WebRtcController] inbound-rtp: totalDecodeTime=${values['totalDecodeTime']}, '
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'googDecodeMs=${values['googDecodeMs']}, '
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'framesDecoded=${values['framesDecoded']}, '
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'fps=${values['framesPerSecond']}');
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width = values['frameWidth'] ?? '-';
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height = values['frameHeight'] ?? '-';
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fps = values['framesPerSecond'] ?? '-';
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@@ -535,6 +549,12 @@ class WebRtcController {
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final fd = values['framesDropped'];
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if (fd is num) framesDropped = fd;
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// 核心：解析解码统计（兼容标准字段与旧版 Google 私有字段）
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currentTotalDecodeTime = (values['totalDecodeTime'] as num?)?.toDouble() ??
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((values['googDecodeMs'] as num?)?.toDouble() ?? 0.0) / 1000.0;
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currentFramesDecoded = (values['framesDecoded'] as num?)?.toInt() ??
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(values['googFramesDecoded'] as num?)?.toInt() ?? 0;
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final codecId = values['codecId'];
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if (codecId != null) {
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final codecReport = reports.where((r) => r.id == codecId).firstOrNull;
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@@ -607,14 +627,14 @@ class WebRtcController {
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? '已连接'
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: (dcState == null ? '未连接' : dcState.name);
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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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// —— 分模式统计解码耗时 ——
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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 (_currentStreamMode == streamModeSelfCodec) {
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// 【自编码模式】：视频走 DataChannel 透传裸码流，WebRTC 的视频统计为空。
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// 使用原生解码器上报的真实分辨率、帧率与耗时。
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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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@@ -637,23 +657,43 @@ class WebRtcController {
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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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// 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。
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_prevTotalDecodeTime = currentTotalDecodeTime;
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_prevFramesDecoded = currentFramesDecoded;
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} else {
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// 【WebRTC 模式】：优先使用标准的 getStats() 增量计算耗时。
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double avg = 0;
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if (currentFramesDecoded > 0) {
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if (_prevFramesDecoded > 0 && currentFramesDecoded > _prevFramesDecoded) {
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final deltaMs = (currentTotalDecodeTime - _prevTotalDecodeTime) * 1000.0;
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final deltaFrames = currentFramesDecoded - _prevFramesDecoded;
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if (deltaFrames > 0) avg = deltaMs / deltaFrames;
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} else if (currentTotalDecodeTime > 0) {
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// 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
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avg = (currentTotalDecodeTime * 1000.0) / currentFramesDecoded;
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}
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}
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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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if (avg > 0) {
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nativeAvg = avg;
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decodeInfo = '解码耗时 (stats): ${avg.toStringAsFixed(1)} ms\n'
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'解码帧率: $fps fps 已解码: $currentFramesDecoded 帧';
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} else {
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// 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。
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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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'解码耗时 (probe): ${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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// WebRTC 模式下，自编码的计数重置。
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_prevDecodeCount = 0;
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_prevTotalDecodeTime = currentTotalDecodeTime;
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_prevFramesDecoded = currentFramesDecoded;
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}
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String decodeTimeText = '-';
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if (decodeInfo != null && nativeAvg > 0) {
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@@ -761,6 +801,8 @@ class WebRtcController {
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_prevBytesSent = 0;
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_prevStatsTimestamp = 0;
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_prevDecodeCount = 0;
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_prevTotalDecodeTime = 0.0;
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_prevFramesDecoded = 0;
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_webRtcDecodeProbe = null;
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_selfCodecWidth = 0;
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_selfCodecHeight = 0;
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