| Index: webrtc/call/call_perf_tests.cc
|
| diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
|
| index 44ecae325ef884d1ca65b94a319c6a8ab2df8aab..5141ed285af0a06334469bb76ab72cf59523ff56 100644
|
| --- a/webrtc/call/call_perf_tests.cc
|
| +++ b/webrtc/call/call_perf_tests.cc
|
| @@ -172,7 +172,7 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
|
| false);
|
| }
|
| if (time_since_creation > kMinRunTimeMs)
|
| - observation_complete_->Set();
|
| + observation_complete_.Set();
|
| }
|
| }
|
|
|
| @@ -311,7 +311,7 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
|
| EXPECT_EQ(0, voe_base->StartReceive(channel));
|
| EXPECT_EQ(0, voe_base->StartSend(channel));
|
|
|
| - EXPECT_EQ(kEventSignaled, observer.Wait())
|
| + EXPECT_TRUE(observer.Wait())
|
| << "Timed out while waiting for audio and video to be synchronized.";
|
|
|
| EXPECT_EQ(0, voe_base->StopSend(channel));
|
| @@ -388,7 +388,7 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
|
| }
|
|
|
| if (time_since_creation > run_time_ms_) {
|
| - observation_complete_->Set();
|
| + observation_complete_.Set();
|
| }
|
|
|
| FrameCaptureTimeList::iterator iter =
|
| @@ -448,9 +448,9 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
|
| }
|
|
|
| void PerformTest() override {
|
| - EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for "
|
| - "estimated capture NTP time to be "
|
| - "within bounds.";
|
| + EXPECT_TRUE(Wait()) << "Timed out while waiting for "
|
| + "estimated capture NTP time to be "
|
| + "within bounds.";
|
| }
|
|
|
| rtc::CriticalSection crit_;
|
| @@ -503,7 +503,7 @@ void CallPerfTest::TestCpuOveruse(LoadObserver::Load tested_load,
|
|
|
| void OnLoadUpdate(Load load) override {
|
| if (load == tested_load_)
|
| - observation_complete_->Set();
|
| + observation_complete_.Set();
|
| }
|
|
|
| void ModifyConfigs(VideoSendStream::Config* send_config,
|
| @@ -514,8 +514,7 @@ void CallPerfTest::TestCpuOveruse(LoadObserver::Load tested_load,
|
| }
|
|
|
| void PerformTest() override {
|
| - EXPECT_EQ(kEventSignaled, Wait())
|
| - << "Timed out before receiving an overuse callback.";
|
| + EXPECT_TRUE(Wait()) << "Timed out before receiving an overuse callback.";
|
| }
|
|
|
| LoadObserver::Load tested_load_;
|
| @@ -583,7 +582,7 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
|
| }
|
| if (num_bitrate_observations_in_range_ ==
|
| kNumBitrateObservationsInRange)
|
| - observation_complete_->Set();
|
| + observation_complete_.Set();
|
| }
|
| }
|
| return SEND_PACKET;
|
| @@ -606,8 +605,7 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
|
| }
|
|
|
| void PerformTest() override {
|
| - EXPECT_EQ(kEventSignaled, Wait())
|
| - << "Timeout while waiting for send-bitrate stats.";
|
| + EXPECT_TRUE(Wait()) << "Timeout while waiting for send-bitrate stats.";
|
| }
|
|
|
| VideoSendStream* send_stream_;
|
| @@ -635,7 +633,7 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
|
| BitrateObserver()
|
| : EndToEndTest(kDefaultTimeoutMs),
|
| FakeEncoder(Clock::GetRealTimeClock()),
|
| - time_to_reconfigure_(webrtc::EventWrapper::Create()),
|
| + time_to_reconfigure_(false, false),
|
| encoder_inits_(0),
|
| last_set_bitrate_(0),
|
| send_stream_(nullptr) {}
|
| @@ -654,7 +652,7 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
|
| last_set_bitrate_,
|
| kPermittedReconfiguredBitrateDiffKbps)
|
| << "Encoder reconfigured with bitrate too far away from last set.";
|
| - observation_complete_->Set();
|
| + observation_complete_.Set();
|
| }
|
| return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
|
| }
|
| @@ -664,7 +662,7 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
|
| last_set_bitrate_ = new_target_bitrate_kbps;
|
| if (encoder_inits_ == 1 &&
|
| new_target_bitrate_kbps > kReconfigureThresholdKbps) {
|
| - time_to_reconfigure_->Set();
|
| + time_to_reconfigure_.Set();
|
| }
|
| return FakeEncoder::SetRates(new_target_bitrate_kbps, framerate);
|
| }
|
| @@ -693,18 +691,18 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
|
| }
|
|
|
| void PerformTest() override {
|
| - ASSERT_EQ(kEventSignaled, time_to_reconfigure_->Wait(kDefaultTimeoutMs))
|
| + ASSERT_TRUE(time_to_reconfigure_.Wait(kDefaultTimeoutMs))
|
| << "Timed out before receiving an initial high bitrate.";
|
| encoder_config_.streams[0].width *= 2;
|
| encoder_config_.streams[0].height *= 2;
|
| EXPECT_TRUE(send_stream_->ReconfigureVideoEncoder(encoder_config_));
|
| - EXPECT_EQ(kEventSignaled, Wait())
|
| + EXPECT_TRUE(Wait())
|
| << "Timed out while waiting for a couple of high bitrate estimates "
|
| "after reconfiguring the send stream.";
|
| }
|
|
|
| private:
|
| - rtc::scoped_ptr<webrtc::EventWrapper> time_to_reconfigure_;
|
| + rtc::Event time_to_reconfigure_;
|
| int encoder_inits_;
|
| uint32_t last_set_bitrate_;
|
| VideoSendStream* send_stream_;
|
|
|