Fix race in VideoSendStreamTest.RtcpSenderReportContainsMediaBytesSent.

webrtc/video/video_send_stream_tests.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <algorithm>  // max
@@ skipping 1504 matching lines @@
 
 TEST_F(VideoSendStreamTest, RtcpSenderReportContainsMediaBytesSent) {
   class RtcpSenderReportTest : public test::SendTest {
    public:
     RtcpSenderReportTest() : SendTest(kDefaultTimeoutMs),
                              rtp_packets_sent_(0),
                              media_bytes_sent_(0) {}
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
       ++rtp_packets_sent_;
       media_bytes_sent_ += length - header.headerLength - header.paddingLength;
       return SEND_PACKET;
     }
 
     Action OnSendRtcp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock(&crit_);
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
 
       RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
       while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
         if (packet_type == RTCPUtility::RTCPPacketTypes::kSr) {
           // 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();
       }
 
       return SEND_PACKET;
     }
 
     void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for RTCP sender report.";
     }
 
-    size_t rtp_packets_sent_;
-    size_t media_bytes_sent_;
+    rtc::CriticalSection crit_;
+    size_t rtp_packets_sent_ GUARDED_BY(&crit_);
+    size_t media_bytes_sent_ GUARDED_BY(&crit_);
   } test;
 
   RunBaseTest(&test, FakeNetworkPipe::Config());
 }
 
 TEST_F(VideoSendStreamTest, TranslatesTwoLayerScreencastToTargetBitrate) {
   static const int kScreencastTargetBitrateKbps = 200;
   class ScreencastTargetBitrateTest : public test::SendTest,
                                       public test::FakeEncoder {
    public:
@@ skipping 363 matching lines @@
         observation_complete_->Set();
       }
     }
 
   } test;
 
   RunBaseTest(&test, FakeNetworkPipe::Config());
 }
 
 }  // namespace webrtc