Index: webrtc/call/bitrate_estimator_tests.cc |
diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc |
index 5b07c547449a109de1571c1aecb3d1fbf6bab9a5..d8c0d5e34d33b22e0cc0d1cb6b6805514f1f648a 100644 |
--- a/webrtc/call/bitrate_estimator_tests.cc |
+++ b/webrtc/call/bitrate_estimator_tests.cc |
@@ -15,12 +15,12 @@ |
#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" |
@@ -44,12 +44,12 @@ class LogObserver { |
callback_.PushExpectedLogLine(expected_log_line); |
} |
- EventTypeWrapper Wait() { return callback_.Wait(); } |
+ bool Wait() { return callback_.Wait(); } |
private: |
class Callback : public rtc::LogSink { |
public: |
- Callback() : done_(EventWrapper::Create()) {} |
+ Callback() : done_(false, false) {} |
void OnLogMessage(const std::string& message) override { |
rtc::CritScope lock(&crit_sect_); |
@@ -72,15 +72,13 @@ class LogObserver { |
} |
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_); |
@@ -92,7 +90,7 @@ class LogObserver { |
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_; |
@@ -271,7 +269,7 @@ TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) { |
receiver_log_.PushExpectedLogLine(kSingleStreamLog); |
receiver_log_.PushExpectedLogLine(kSingleStreamLog); |
streams_.push_back(new Stream(this, false)); |
- EXPECT_EQ(kEventSignaled, receiver_log_.Wait()); |
+ EXPECT_TRUE(receiver_log_.Wait()); |
} |
TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForAudio) { |
@@ -282,7 +280,7 @@ TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForAudio) { |
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE."); |
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog); |
streams_.push_back(new Stream(this, true)); |
- EXPECT_EQ(kEventSignaled, receiver_log_.Wait()); |
+ EXPECT_TRUE(receiver_log_.Wait()); |
} |
TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) { |
@@ -293,21 +291,21 @@ TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) { |
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE."); |
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog); |
streams_.push_back(new Stream(this, false)); |
- EXPECT_EQ(kEventSignaled, receiver_log_.Wait()); |
+ EXPECT_TRUE(receiver_log_.Wait()); |
} |
TEST_F(BitrateEstimatorTest, SwitchesToASTForAudio) { |
receiver_log_.PushExpectedLogLine(kSingleStreamLog); |
receiver_log_.PushExpectedLogLine(kSingleStreamLog); |
streams_.push_back(new Stream(this, true)); |
- EXPECT_EQ(kEventSignaled, receiver_log_.Wait()); |
+ EXPECT_TRUE(receiver_log_.Wait()); |
send_config_.rtp.extensions.push_back( |
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId)); |
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE."); |
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog); |
streams_.push_back(new Stream(this, true)); |
- EXPECT_EQ(kEventSignaled, receiver_log_.Wait()); |
+ EXPECT_TRUE(receiver_log_.Wait()); |
} |
TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) { |
@@ -316,14 +314,14 @@ TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) { |
receiver_log_.PushExpectedLogLine(kSingleStreamLog); |
receiver_log_.PushExpectedLogLine(kSingleStreamLog); |
streams_.push_back(new Stream(this, false)); |
- EXPECT_EQ(kEventSignaled, receiver_log_.Wait()); |
+ EXPECT_TRUE(receiver_log_.Wait()); |
send_config_.rtp.extensions[0] = |
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId); |
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE."); |
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog); |
streams_.push_back(new Stream(this, false)); |
- EXPECT_EQ(kEventSignaled, receiver_log_.Wait()); |
+ EXPECT_TRUE(receiver_log_.Wait()); |
} |
TEST_F(BitrateEstimatorTest, SwitchesToASTThenBackToTOFForVideo) { |
@@ -332,14 +330,14 @@ TEST_F(BitrateEstimatorTest, SwitchesToASTThenBackToTOFForVideo) { |
receiver_log_.PushExpectedLogLine(kSingleStreamLog); |
receiver_log_.PushExpectedLogLine(kSingleStreamLog); |
streams_.push_back(new Stream(this, false)); |
- EXPECT_EQ(kEventSignaled, receiver_log_.Wait()); |
+ EXPECT_TRUE(receiver_log_.Wait()); |
send_config_.rtp.extensions[0] = |
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId); |
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE."); |
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog); |
streams_.push_back(new Stream(this, false)); |
- EXPECT_EQ(kEventSignaled, receiver_log_.Wait()); |
+ EXPECT_TRUE(receiver_log_.Wait()); |
send_config_.rtp.extensions[0] = |
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId); |
@@ -349,6 +347,6 @@ TEST_F(BitrateEstimatorTest, SwitchesToASTThenBackToTOFForVideo) { |
streams_.push_back(new Stream(this, false)); |
streams_[0]->StopSending(); |
streams_[1]->StopSending(); |
- EXPECT_EQ(kEventSignaled, receiver_log_.Wait()); |
+ EXPECT_TRUE(receiver_log_.Wait()); |
} |
} // namespace webrtc |