Fix flaky EndToEndTest.TransportSeqNumOnAudioAndVideo

webrtc/video/end_to_end_tests.cc

Index: webrtc/video/end_to_end_tests.cc
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index cb7edd422fa619bc9e36e1928950415b3b3208d2..3fb8f13d23e1500ca2e1992da184bcd0a9f6a2c2 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -4063,7 +4063,7 @@ TEST_P(EndToEndTest, TransportSeqNumOnAudioAndVideo) {
       if (header.ssrc == kAudioSendSsrc)
         audio_observed_ = true;
       if (audio_observed_ && video_observed_ &&
-          received_packet_ids_.size() == 50) {
+          received_packet_ids_.size() >= 50) {

[åsapersson, 2017/03/03 11:16:19]

maybe add a constant for 50

[danilchap, 2017/03/03 12:14:11]

Done.

         size_t packet_id_range =
             *received_packet_ids_.rbegin() - *received_packet_ids_.begin() + 1;
         EXPECT_EQ(received_packet_ids_.size(), packet_id_range);
@@ -4075,6 +4075,11 @@ TEST_P(EndToEndTest, TransportSeqNumOnAudioAndVideo) {
     void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out while waiting for audio and video "
                              "packets with transport sequence number.";
+      // Double check conditions for successfull test to produce better error
+      // message when this test fail.
+      EXPECT_TRUE(video_observed_);
+      EXPECT_TRUE(audio_observed_);
+      EXPECT_GE(received_packet_ids_.size(), 50u);
     }
 
    private: