Fix (1) EndToEndTest.InitialProbing and (2) EndToEndTest.TriggerMidCallProbing

webrtc/video/end_to_end_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>
@@ skipping 2288 matching lines @@
   Clock* const clock_;
   const int start_bitrate_bps_;
   int state_;
   Call* sender_call_;
 };
 
 TEST_F(EndToEndTest, MAYBE_InitialProbing) {
   class InitialProbingTest : public ProbingTest {
    public:
     explicit InitialProbingTest(bool* success)
-        : ProbingTest(300000), success_(success) {}
+        : ProbingTest(300000), success_(success) {
+      *success_ = false;

[eladalon, 2017/08/10 15:07:21]

This line was missing from this test, though not from the next one.

+    }
 
     void PerformTest() override {
       int64_t start_time_ms = clock_->TimeInMilliseconds();
       do {
         if (clock_->TimeInMilliseconds() - start_time_ms > kTimeoutMs)
           break;
 
         Call::Stats stats = sender_call_->GetStats();
         // Initial probing is done with a x3 and x6 multiplier of the start
         // bitrate, so a x4 multiplier is a high enough threshold.
         if (stats.send_bandwidth_bps > 4 * 300000) {
           *success_ = true;
           break;
         }
       } while (!observation_complete_.Wait(20));
     }
 
    private:
     const int kTimeoutMs = 1000;
     bool* const success_;
   };
 
-  bool success;
+  bool success = false;

[eladalon, 2017/08/10 15:07:21]

This is not strictly necessary, because it happens in the ctor, but it avoids
giving the reader a small heart-attack before he sees the ctor.

   const int kMaxAttempts = 3;
   for (int i = 0; i < kMaxAttempts; ++i) {
     InitialProbingTest test(&success);
     RunBaseTest(&test);
     if (success)
       return;
   }
-  RTC_DCHECK(success) << "Failed to perform mid initial probing ("
-                      << kMaxAttempts << " attempts).";
+  EXPECT_TRUE(success) << "Failed to perform mid initial probing ("
+                       << kMaxAttempts << " attempts).";
 }
 
 // Fails on Linux MSan: bugs.webrtc.org/7428
 #if defined(MEMORY_SANITIZER)
 TEST_F(EndToEndTest, DISABLED_TriggerMidCallProbing) {
 // Fails on iOS bots: bugs.webrtc.org/7851
 #elif defined(TARGET_IPHONE_SIMULATOR) && TARGET_IPHONE_SIMULATOR
 TEST_F(EndToEndTest, DISABLED_TriggerMidCallProbing) {
 #else
 TEST_F(EndToEndTest, TriggerMidCallProbing) {
@@ skipping 41 matching lines @@
             break;
         }
       } while (!observation_complete_.Wait(20));
     }
 
    private:
     const int kTimeoutMs = 5000;
     bool* const success_;
   };
 
-  bool success;
+  bool success = false;

[eladalon, 2017/08/10 15:07:21]

This is not strictly necessary, because it happens in the ctor, but it avoids
giving the reader a small heart-attack before he sees the ctor.

   const int kMaxAttempts = 3;
   for (int i = 0; i < kMaxAttempts; ++i) {
     TriggerMidCallProbingTest test(&success);
     RunBaseTest(&test);
     if (success)
       return;
   }
-  RTC_DCHECK(success) << "Failed to perform mid call probing (" << kMaxAttempts
-                      << " attempts).";
+  EXPECT_TRUE(success) << "Failed to perform mid call probing (" << kMaxAttempts
+                       << " attempts).";
 }
 
 TEST_F(EndToEndTest, VerifyNackStats) {
   static const int kPacketNumberToDrop = 200;
   class NackObserver : public test::EndToEndTest {
    public:
     NackObserver()
         : EndToEndTest(kLongTimeoutMs),
           sent_rtp_packets_(0),
           dropped_rtp_packet_(0),
@@ skipping 2334 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc