commit 146742164664f9e35ed57575854e14f8011f02a8
author asapersson@webrtc.org <asapersson@webrtc.org> Mon Feb 23 08:14:07 2015 +0000
committer asapersson@webrtc.org <asapersson@webrtc.org> Mon Feb 23 08:14:46 2015 +0000
tree 02b7e793e92a3ee653a3e15ce9960dd37a26a812
parent 50e28166afcdf4b2fcc6e331b70e77a284c3a560

Fix for flaky test: VideoSendStreamTest.RtcpSenderReportContainsMediaBytesSent.

Only compare media bytes sent if number of sent packets in rtcp packet are equal to sent rtp packets.

BUG=4327
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34299004

Cr-Commit-Position: refs/heads/master@{#8454}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8454 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/video/video_send_stream_tests.cc

diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 8330b9d..f5ec59b 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -1423,14 +1423,17 @@
 }
 
 TEST_F(VideoSendStreamTest, RtcpSenderReportContainsMediaBytesSent) {
-  class RtcpByeTest : public test::SendTest {
+  class RtcpSenderReportTest : public test::SendTest {
    public:
-    RtcpByeTest() : SendTest(kDefaultTimeoutMs), media_bytes_sent_(0) {}
+    RtcpSenderReportTest() : SendTest(kDefaultTimeoutMs),
+                             rtp_packets_sent_(0),
+                             media_bytes_sent_(0) {}
 
    private:
     virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
+      ++rtp_packets_sent_;
       media_bytes_sent_ += length - header.headerLength - header.paddingLength;
       return SEND_PACKET;
     }
@@ -1440,15 +1443,17 @@
       EXPECT_TRUE(parser.IsValid());
 
       RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
-      uint32_t sender_octet_count = 0;
       while (packet_type != RTCPUtility::kRtcpNotValidCode) {
         if (packet_type == RTCPUtility::kRtcpSrCode) {
-          sender_octet_count = parser.Packet().SR.SenderOctetCount;
-          EXPECT_EQ(sender_octet_count, media_bytes_sent_);
-          if (sender_octet_count > 0)
+          // Only compare sent media bytes if SenderPacketCount matches the
+          // number of sent rtp packets (a new rtp packet could be sent before
+          // the rtcp packet).
+          if (parser.Packet().SR.SenderOctetCount > 0 &&
+              parser.Packet().SR.SenderPacketCount == rtp_packets_sent_) {
+            EXPECT_EQ(media_bytes_sent_, parser.Packet().SR.SenderOctetCount);
             observation_complete_->Set();
+          }
         }
-
         packet_type = parser.Iterate();
       }
 
@@ -1460,6 +1465,7 @@
           << "Timed out while waiting for RTCP sender report.";
     }
 
+    size_t rtp_packets_sent_;
     size_t media_bytes_sent_;
   } test;