Chromium Code Reviews| Index: webrtc/call/bitrate_estimator_tests.cc |
| diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc |
| index 80949baba4d5933d672511385076e4c8b7ae5560..ff72c6688a61a20f38c31c15167281b2e95ba21c 100644 |
| --- a/webrtc/call/bitrate_estimator_tests.cc |
| +++ b/webrtc/call/bitrate_estimator_tests.cc |
| @@ -18,6 +18,7 @@ |
| #include "webrtc/base/logging.h" |
| #include "webrtc/base/thread_annotations.h" |
| #include "webrtc/call.h" |
| +#include "webrtc/modules/audio_device/include/mock_audio_device.h" |
| #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/include/trace.h" |
| #include "webrtc/test/call_test.h" |
| @@ -109,6 +110,8 @@ class BitrateEstimatorTest : public test::CallTest { |
| virtual void SetUp() { |
| AudioState::Config audio_state_config; |
| audio_state_config.voice_engine = &mock_voice_engine_; |
| + ON_CALL(mock_voice_engine_, audio_device_module()) |
| + .WillByDefault(Return(&mock_audio_device_)); |
|
stefan-webrtc
2016/11/03 14:33:53
Could you try to remove everything audio related h
aleloi
2016/11/08 11:46:41
I've done it in another CL; I'll rebase this one a
|
| Call::Config config(&event_log_); |
| config.audio_state = AudioState::Create(audio_state_config); |
| receiver_call_.reset(Call::Create(config)); |
| @@ -252,6 +255,7 @@ class BitrateEstimatorTest : public test::CallTest { |
| }; |
| testing::NiceMock<test::MockVoiceEngine> mock_voice_engine_; |
| + testing::NiceMock<webrtc::test::MockAudioDeviceModule> mock_audio_device_; |
| LogObserver receiver_log_; |
| std::unique_ptr<test::DirectTransport> send_transport_; |
| std::unique_ptr<test::DirectTransport> receive_transport_; |