Index: webrtc/call/call_unittest.cc |
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc |
index 25fb11989583cfcf93aafd53689c7618d83ce31b..b26024d91e5cb86582a7d5f5ea95049b86644827 100644 |
--- a/webrtc/call/call_unittest.cc |
+++ b/webrtc/call/call_unittest.cc |
@@ -12,22 +12,31 @@ |
#include "testing/gtest/include/gtest/gtest.h" |
+#include "webrtc/audio_state.h" |
#include "webrtc/call.h" |
#include "webrtc/test/mock_voice_engine.h" |
namespace { |
struct CallHelper { |
- CallHelper() : voice_engine_(new webrtc::test::MockVoiceEngine()) { |
+ CallHelper() { |
+ EXPECT_CALL(voice_engine_, |
+ RegisterVoiceEngineObserver(testing::_)).WillOnce(testing::Return(0)); |
+ EXPECT_CALL(voice_engine_, |
+ DeRegisterVoiceEngineObserver()).WillOnce(testing::Return(0)); |
+ EXPECT_CALL(voice_engine_, |
+ GetEventLog()).WillOnce(testing::Return(nullptr)); |
+ webrtc::AudioState::Config audio_state_config; |
+ audio_state_config.voice_engine = &voice_engine_; |
webrtc::Call::Config config; |
- config.voice_engine = voice_engine_.get(); |
+ config.audio_state = webrtc::AudioState::Create(audio_state_config); |
call_.reset(webrtc::Call::Create(config)); |
} |
webrtc::Call* operator->() { return call_.get(); } |
private: |
- rtc::scoped_ptr<webrtc::test::MockVoiceEngine> voice_engine_; |
+ webrtc::test::MockVoiceEngine voice_engine_; |
rtc::scoped_ptr<webrtc::Call> call_; |
}; |
} // namespace |