Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

1 files changed, 94 insertions, 107 deletions

diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc
index 685f3fd665..4b24bbd5ef 100644
--- a/webrtc/call/bitrate_estimator_tests.cc
+++ b/webrtc/call/bitrate_estimator_tests.cc
@@ -13,66 +13,54 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/audio_state.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
-#include "webrtc/test/fake_voice_engine.h"
+#include "webrtc/test/mock_voice_engine.h"
 #include "webrtc/test/frame_generator_capturer.h"
 
 namespace webrtc {
 namespace {
 // Note: If you consider to re-use this class, think twice and instead consider
-// writing tests that don't depend on the trace system.
-class TraceObserver {
+// writing tests that don't depend on the logging system.
+class LogObserver {
  public:
-  TraceObserver() {
-    Trace::set_level_filter(kTraceTerseInfo);
-
-    Trace::CreateTrace();
-    Trace::SetTraceCallback(&callback_);
-
-    // Call webrtc trace to initialize the tracer that would otherwise trigger a
-    // data-race if left to be initialized by multiple threads (i.e. threads
-    // spawned by test::DirectTransport members in BitrateEstimatorTest).
-    WEBRTC_TRACE(kTraceStateInfo,
-                 kTraceUtility,
-                 -1,
-                 "Instantiate without data races.");
-  }
+  LogObserver() { rtc::LogMessage::AddLogToStream(&callback_, rtc::LS_INFO); }
 
-  ~TraceObserver() {
-    Trace::SetTraceCallback(nullptr);
-    Trace::ReturnTrace();
-  }
+  ~LogObserver() { rtc::LogMessage::RemoveLogToStream(&callback_); }
 
   void PushExpectedLogLine(const std::string& expected_log_line) {
     callback_.PushExpectedLogLine(expected_log_line);
   }
 
-  EventTypeWrapper Wait() {
-    return callback_.Wait();
-  }
+  bool Wait() { return callback_.Wait(); }
 
  private:
-  class Callback : public TraceCallback {
+  class Callback : public rtc::LogSink {
    public:
-    Callback() : done_(EventWrapper::Create()) {}
+    Callback() : done_(false, false) {}
 
-    void Print(TraceLevel level, const char* message, int length) override {
+    void OnLogMessage(const std::string& message) override {
       rtc::CritScope lock(&crit_sect_);
-      std::string msg(message);
-      if (msg.find("BitrateEstimator") != std::string::npos) {
-        received_log_lines_.push_back(msg);
+      // Ignore log lines that are due to missing AST extensions, these are
+      // logged when we switch back from AST to TOF until the wrapping bitrate
+      // estimator gives up on using AST.
+      if (message.find("BitrateEstimator") != std::string::npos &&
+          message.find("packet is missing") == std::string::npos) {
+        received_log_lines_.push_back(message);
       }
+
       int num_popped = 0;
       while (!received_log_lines_.empty() && !expected_log_lines_.empty()) {
         std::string a = received_log_lines_.front();
@@ -80,19 +68,17 @@ class TraceObserver {
         received_log_lines_.pop_front();
         expected_log_lines_.pop_front();
         num_popped++;
-        EXPECT_TRUE(a.find(b) != std::string::npos);
+        EXPECT_TRUE(a.find(b) != std::string::npos) << a << " != " << b;
       }
       if (expected_log_lines_.size() <= 0) {
         if (num_popped > 0) {
-          done_->Set();
+          done_.Set();
         }
         return;
       }
     }
 
-    EventTypeWrapper Wait() {
-      return done_->Wait(test::CallTest::kDefaultTimeoutMs);
-    }
+    bool Wait() { return done_.Wait(test::CallTest::kDefaultTimeoutMs); }
 
     void PushExpectedLogLine(const std::string& expected_log_line) {
       rtc::CritScope lock(&crit_sect_);
@@ -104,7 +90,7 @@ class TraceObserver {
     rtc::CriticalSection crit_sect_;
     Strings received_log_lines_ GUARDED_BY(crit_sect_);
     Strings expected_log_lines_ GUARDED_BY(crit_sect_);
-    rtc::scoped_ptr<EventWrapper> done_;
+    rtc::Event done_;
   };
 
   Callback callback_;
@@ -118,13 +104,13 @@ class BitrateEstimatorTest : public test::CallTest {
  public:
   BitrateEstimatorTest() : receive_config_(nullptr) {}
 
-  virtual ~BitrateEstimatorTest() {
-    EXPECT_TRUE(streams_.empty());
-  }
+  virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }
 
   virtual void SetUp() {
+    AudioState::Config audio_state_config;
+    audio_state_config.voice_engine = &mock_voice_engine_;
     Call::Config config;
-    config.voice_engine = &fake_voice_engine_;
+    config.audio_state = AudioState::Create(audio_state_config);
     receiver_call_.reset(Call::Create(config));
     sender_call_.reset(Call::Create(config));
 
@@ -133,18 +119,19 @@ class BitrateEstimatorTest : public test::CallTest {
     receive_transport_.reset(new test::DirectTransport(receiver_call_.get()));
     receive_transport_->SetReceiver(sender_call_->Receiver());
 
-    send_config_ = VideoSendStream::Config(send_transport_.get());
-    send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]);
+    video_send_config_ = VideoSendStream::Config(send_transport_.get());
+    video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
     // Encoders will be set separately per stream.
-    send_config_.encoder_settings.encoder = nullptr;
-    send_config_.encoder_settings.payload_name = "FAKE";
-    send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
-    encoder_config_.streams = test::CreateVideoStreams(1);
+    video_send_config_.encoder_settings.encoder = nullptr;
+    video_send_config_.encoder_settings.payload_name = "FAKE";
+    video_send_config_.encoder_settings.payload_type =
+        kFakeVideoSendPayloadType;
+    video_encoder_config_.streams = test::CreateVideoStreams(1);
 
     receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
     // receive_config_.decoders will be set by every stream separately.
-    receive_config_.rtp.remote_ssrc = send_config_.rtp.ssrcs[0];
-    receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
+    receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
+    receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
     receive_config_.rtp.remb = true;
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
@@ -154,7 +141,7 @@ class BitrateEstimatorTest : public test::CallTest {
 
   virtual void TearDown() {
     std::for_each(streams_.begin(), streams_.end(),
-        std::mem_fun(&Stream::StopSending));
+                  std::mem_fun(&Stream::StopSending));
 
     send_transport_->StopSending();
     receive_transport_->StopSending();
@@ -165,6 +152,7 @@ class BitrateEstimatorTest : public test::CallTest {
     }
 
     receiver_call_.reset();
+    sender_call_.reset();
   }
 
  protected:
@@ -181,23 +169,21 @@ class BitrateEstimatorTest : public test::CallTest {
           frame_generator_capturer_(),
           fake_encoder_(Clock::GetRealTimeClock()),
           fake_decoder_() {
-      test_->send_config_.rtp.ssrcs[0]++;
-      test_->send_config_.encoder_settings.encoder = &fake_encoder_;
+      test_->video_send_config_.rtp.ssrcs[0]++;
+      test_->video_send_config_.encoder_settings.encoder = &fake_encoder_;
       send_stream_ = test_->sender_call_->CreateVideoSendStream(
-          test_->send_config_, test_->encoder_config_);
-      RTC_DCHECK_EQ(1u, test_->encoder_config_.streams.size());
+          test_->video_send_config_, test_->video_encoder_config_);
+      RTC_DCHECK_EQ(1u, test_->video_encoder_config_.streams.size());
       frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
-          send_stream_->Input(),
-          test_->encoder_config_.streams[0].width,
-          test_->encoder_config_.streams[0].height,
-          30,
+          send_stream_->Input(), test_->video_encoder_config_.streams[0].width,
+          test_->video_encoder_config_.streams[0].height, 30,
           Clock::GetRealTimeClock()));
       send_stream_->Start();
       frame_generator_capturer_->Start();
 
       if (receive_audio) {
         AudioReceiveStream::Config receive_config;
-        receive_config.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0];
+        receive_config.rtp.remote_ssrc = test_->video_send_config_.rtp.ssrcs[0];
         // Bogus non-default id to prevent hitting a RTC_DCHECK when creating
         // the AudioReceiveStream. Every receive stream has to correspond to
         // an underlying channel id.
@@ -211,12 +197,13 @@ class BitrateEstimatorTest : public test::CallTest {
         VideoReceiveStream::Decoder decoder;
         decoder.decoder = &fake_decoder_;
         decoder.payload_type =
-            test_->send_config_.encoder_settings.payload_type;
+            test_->video_send_config_.encoder_settings.payload_type;
         decoder.payload_name =
-            test_->send_config_.encoder_settings.payload_name;
+            test_->video_send_config_.encoder_settings.payload_name;
+        test_->receive_config_.decoders.clear();
         test_->receive_config_.decoders.push_back(decoder);
         test_->receive_config_.rtp.remote_ssrc =
-            test_->send_config_.rtp.ssrcs[0];
+            test_->video_send_config_.rtp.ssrcs[0];
         test_->receive_config_.rtp.local_ssrc++;
         video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
             test_->receive_config_);
@@ -262,8 +249,8 @@ class BitrateEstimatorTest : public test::CallTest {
     test::FakeDecoder fake_decoder_;
   };
 
-  test::FakeVoiceEngine fake_voice_engine_;
-  TraceObserver receiver_trace_;
+  testing::NiceMock<test::MockVoiceEngine> mock_voice_engine_;
+  LogObserver receiver_log_;
   rtc::scoped_ptr<test::DirectTransport> send_transport_;
   rtc::scoped_ptr<test::DirectTransport> receive_transport_;
   rtc::scoped_ptr<Call> sender_call_;
@@ -278,89 +265,89 @@ static const char* kSingleStreamLog =
     "RemoteBitrateEstimatorSingleStream: Instantiating.";
 
 TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
-  send_config_.rtp.extensions.push_back(
+  video_send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
-  receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+  EXPECT_TRUE(receiver_log_.Wait());
 }
 
 TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForAudio) {
-  send_config_.rtp.extensions.push_back(
+  video_send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
-  receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
-  receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   streams_.push_back(new Stream(this, true));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+  EXPECT_TRUE(receiver_log_.Wait());
 }
 
 TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
-  send_config_.rtp.extensions.push_back(
+  video_send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
-  receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
-  receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   streams_.push_back(new Stream(this, false));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+  EXPECT_TRUE(receiver_log_.Wait());
 }
 
 TEST_F(BitrateEstimatorTest, SwitchesToASTForAudio) {
-  receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, true));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+  EXPECT_TRUE(receiver_log_.Wait());
 
-  send_config_.rtp.extensions.push_back(
+  video_send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
-  receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
-  receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   streams_.push_back(new Stream(this, true));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+  EXPECT_TRUE(receiver_log_.Wait());
 }
 
 TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
-  send_config_.rtp.extensions.push_back(
+  video_send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
-  receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+  EXPECT_TRUE(receiver_log_.Wait());
 
-  send_config_.rtp.extensions[0] =
+  video_send_config_.rtp.extensions[0] =
       RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
-  receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
-  receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   streams_.push_back(new Stream(this, false));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+  EXPECT_TRUE(receiver_log_.Wait());
 }
 
 TEST_F(BitrateEstimatorTest, SwitchesToASTThenBackToTOFForVideo) {
-  send_config_.rtp.extensions.push_back(
+  video_send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
-  receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
-  receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+  EXPECT_TRUE(receiver_log_.Wait());
 
-  send_config_.rtp.extensions[0] =
+  video_send_config_.rtp.extensions[0] =
       RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
-  receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
-  receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   streams_.push_back(new Stream(this, false));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+  EXPECT_TRUE(receiver_log_.Wait());
 
-  send_config_.rtp.extensions[0] =
+  video_send_config_.rtp.extensions[0] =
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
-  receiver_trace_.PushExpectedLogLine(
+  receiver_log_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
-  receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+  EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc