Index: webrtc/audio/audio_send_stream_unittest.cc |
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc |
index a01ef025d3be84a85f30aa5cec0d225b7608bc89..bffbad541dfdbf14573428243a53246ea87e8ec6 100644 |
--- a/webrtc/audio/audio_send_stream_unittest.cc |
+++ b/webrtc/audio/audio_send_stream_unittest.cc |
@@ -19,6 +19,7 @@ |
#include "webrtc/call/congestion_controller.h" |
#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h" |
#include "webrtc/modules/pacing/paced_sender.h" |
+#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h" |
#include "webrtc/test/mock_voe_channel_proxy.h" |
#include "webrtc/test/mock_voice_engine.h" |
#include "webrtc/video/call_stats.h" |
@@ -51,12 +52,15 @@ const uint32_t kTelephoneEventDuration = 6789; |
struct ConfigHelper { |
ConfigHelper() |
- : stream_config_(nullptr), |
- call_stats_(Clock::GetRealTimeClock()), |
+ : simulated_clock_(123456), |
+ stream_config_(nullptr), |
+ call_stats_(&simulated_clock_), |
process_thread_(ProcessThread::Create("AudioTestThread")), |
- congestion_controller_(process_thread_.get(), |
+ congestion_controller_(&simulated_clock_, |
+ process_thread_.get(), |
&call_stats_, |
- &bitrate_observer_) { |
+ &bitrate_observer_, |
+ &remote_bitrate_observer_) { |
using testing::Invoke; |
using testing::StrEq; |
@@ -153,12 +157,14 @@ struct ConfigHelper { |
} |
private: |
+ SimulatedClock simulated_clock_; |
testing::StrictMock<MockVoiceEngine> voice_engine_; |
rtc::scoped_refptr<AudioState> audio_state_; |
AudioSendStream::Config stream_config_; |
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr; |
CallStats call_stats_; |
testing::NiceMock<MockBitrateObserver> bitrate_observer_; |
+ testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_; |
rtc::scoped_ptr<ProcessThread> process_thread_; |
CongestionController congestion_controller_; |
}; |