From fef85da7712495c09a967e4181e8314075cd018c Mon Sep 17 00:00:00 2001
From: waken <33921191+mc373906408@users.noreply.github.com>
Date: Thu, 31 Aug 2023 10:53:46 +0800
Subject: [PATCH] =?UTF-8?q?=E4=BF=AE=E5=A4=8D=E8=B0=83=E7=94=A8close=5Fstr?=
 =?UTF-8?q?eam=E5=90=8C=E6=97=B6=E8=B0=83=E7=94=A8getMediaList=E5=BC=95?=
 =?UTF-8?q?=E5=8F=91=E7=9A=84=E5=B4=A9=E6=BA=83=20(#2800)?=
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复现条件：chrome打开webrtc
demo网页，网页会每秒调用getMediaList。添加一条rtsp拉流，然后再调用close_stream，会偶现进入RtspPlayer::getPacketLossRate函数，然后_rtcp_context为空拿不到指针
---
 src/Rtsp/RtspPlayer.cpp | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/src/Rtsp/RtspPlayer.cpp b/src/Rtsp/RtspPlayer.cpp
index 676d89b1..f6a13fda 100644
--- a/src/Rtsp/RtspPlayer.cpp
+++ b/src/Rtsp/RtspPlayer.cpp
@@ -544,6 +544,9 @@ float RtspPlayer::getPacketLossRate(TrackType type) const {
     size_t lost = 0, expected = 0;
     try {
         auto track_idx = getTrackIndexByTrackType(type);
+        if (_rtcp_context.empty()) {
+            return 0;
+        }
         auto ctx = _rtcp_context[track_idx];
         lost = ctx->getLost();
         expected = ctx->getExpectedPackets();