Fixing flaky test: WebRtcSessionTest.TestPacketOptionsAndOnPacketSent

The test sent a media packet, then verified it was sent by checking the
"last packet sent"'s ID. But the last packet sent may have been
a STUN packet that came *after* the media packet.

BUG=webrtc:5978

Review-Url: https://codereview.webrtc.org/2071573002
Cr-Commit-Position: refs/heads/master@{#13156}

tools/valgrind-webrtc/gtest_exclude/peerconnection_unittests.gtest-drmemory_win32.txt

@@ -6,6 +6,3 @@ PeerConnectionEndToEndTest.*
 PeerConnectionInterfaceTest.*
 # Issue 3453
 WebRtcSessionTest.TestReceiveSdesOfferCreateSdesAnswer
-# Flakily fails or crashes on Dr Memory Light.
-# Issue 5978
-WebRtcSessionTest.TestPacketOptionsAndOnPacketSent

webrtc/api/webrtcsession_unittest.cc

@@ -1326,7 +1326,8 @@ class WebRtcSessionTest
         ->SendRtp(test_packet, sizeof(test_packet), options);
 
     const int kPacketTimeout = 2000;
-    EXPECT_EQ_WAIT(fake_call_.last_sent_packet().packet_id, 10, kPacketTimeout);
+    EXPECT_EQ_WAIT(10, fake_call_.last_sent_nonnegative_packet_id(),
+                   kPacketTimeout);
     EXPECT_GT(fake_call_.last_sent_packet().send_time_ms, -1);
   }
 

webrtc/media/engine/fakewebrtccall.cc

@@ -465,5 +465,9 @@ void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
+  if (sent_packet.packet_id >= 0) {
+    last_sent_nonnegative_packet_id_ = sent_packet.packet_id;
+  }
 }
+
 }  // namespace cricket

webrtc/media/engine/fakewebrtccall.h

@@ -177,6 +177,13 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
   const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc);
 
   rtc::SentPacket last_sent_packet() const { return last_sent_packet_; }
+
+  // This is useful if we care about the last media packet (with id populated)
+  // but not the last ICE packet (with -1 ID).
+  int last_sent_nonnegative_packet_id() const {
+    return last_sent_nonnegative_packet_id_;
+  }
+
   webrtc::NetworkState GetNetworkState(webrtc::MediaType media) const;
   int GetNumCreatedSendStreams() const;
   int GetNumCreatedReceiveStreams() const;
@@ -222,6 +229,7 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
   webrtc::NetworkState audio_network_state_;
   webrtc::NetworkState video_network_state_;
   rtc::SentPacket last_sent_packet_;
+  int last_sent_nonnegative_packet_id_ = -1;
   webrtc::Call::Stats stats_;
   std::vector<FakeVideoSendStream*> video_send_streams_;
   std::vector<FakeAudioSendStream*> audio_send_streams_;