| 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
|
|
|