Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(2101)

Unified Diff: webrtc/media/engine/webrtcvoiceengine_unittest.cc

Issue 2737633002: Remove VoEHardware interface usage. (Closed)
Patch Set: test Created 3 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/media/engine/webrtcvoiceengine_unittest.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 4c8599008ab49dbef6ca365ad461b0759357ce90..f7cfa7f3f075bfb30561065cba058e072994bf50 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -62,8 +62,7 @@ class FakeVoEWrapper : public cricket::VoEWrapper {
public:
explicit FakeVoEWrapper(cricket::FakeWebRtcVoiceEngine* engine)
: cricket::VoEWrapper(engine, // base
- engine, // codec
- engine) { // hw
+ engine) { // codec
}
};
@@ -74,17 +73,50 @@ class MockTransmitMixer : public webrtc::voe::TransmitMixer {
MOCK_METHOD1(EnableStereoChannelSwapping, void(bool enable));
};
+
+void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) {
henrika_webrtc 2017/03/13 11:15:55 These changes does not correlate 100% with the rem
the sun 2017/03/15 12:47:57 Certainly! The previous calls to SetRecordingDevi
henrika_webrtc 2017/03/15 12:50:19 Acknowledged.
+ RTC_DCHECK(adm);
+ EXPECT_CALL(*adm, AddRef()).WillOnce(Return(0));
+ EXPECT_CALL(*adm, Release()).WillOnce(Return(0));
+#if !defined(WEBRTC_IOS)
+ EXPECT_CALL(*adm, Recording()).WillOnce(Return(false));
+ EXPECT_CALL(*adm, SetRecordingChannel(webrtc::AudioDeviceModule::
+ ChannelType::kChannelBoth)).WillOnce(Return(0));
+#if defined(WEBRTC_WIN)
+ EXPECT_CALL(*adm, SetRecordingDevice(
+ static_cast<webrtc::AudioDeviceModule::WindowsDeviceType>(
+ webrtc::AudioDeviceModule::WindowsDeviceType::kDefaultCommunicationDevice)))
+ .WillOnce(Return(0));
+#else
+ EXPECT_CALL(*adm, SetRecordingDevice(0)).WillOnce(Return(0));
+#endif // #if defined(WEBRTC_WIN)
+ EXPECT_CALL(*adm, InitMicrophone()).WillOnce(Return(0));
+ EXPECT_CALL(*adm, StereoRecordingIsAvailable(testing::_)).WillOnce(Return(0));
+ EXPECT_CALL(*adm, SetStereoRecording(false)).WillOnce(Return(0));
+ EXPECT_CALL(*adm, Playing()).WillOnce(Return(false));
+#if defined(WEBRTC_WIN)
+ EXPECT_CALL(*adm, SetPlayoutDevice(
+ static_cast<webrtc::AudioDeviceModule::WindowsDeviceType>(
+ webrtc::AudioDeviceModule::WindowsDeviceType::kDefaultCommunicationDevice)))
+ .WillOnce(Return(0));
+#else
+ EXPECT_CALL(*adm, SetPlayoutDevice(0)).WillOnce(Return(0));
+#endif // #if defined(WEBRTC_WIN)
+ EXPECT_CALL(*adm, InitSpeaker()).WillOnce(Return(0));
+ EXPECT_CALL(*adm, StereoPlayoutIsAvailable(testing::_)).WillOnce(Return(0));
+ EXPECT_CALL(*adm, SetStereoPlayout(false)).WillOnce(Return(0));
+#endif // #if !defined(WEBRTC_IOS)
+ EXPECT_CALL(*adm, BuiltInAECIsAvailable()).WillOnce(Return(false));
+ EXPECT_CALL(*adm, BuiltInAGCIsAvailable()).WillOnce(Return(false));
+ EXPECT_CALL(*adm, BuiltInNSIsAvailable()).WillOnce(Return(false));
+ EXPECT_CALL(*adm, SetAGC(true)).WillOnce(Return(0));
+}
} // namespace
// Tests that our stub library "works".
TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {
StrictMock<webrtc::test::MockAudioDeviceModule> adm;
- EXPECT_CALL(adm, AddRef()).WillOnce(Return(0));
- EXPECT_CALL(adm, Release()).WillOnce(Return(0));
- EXPECT_CALL(adm, BuiltInAECIsAvailable()).WillOnce(Return(false));
- EXPECT_CALL(adm, BuiltInAGCIsAvailable()).WillOnce(Return(false));
- EXPECT_CALL(adm, BuiltInNSIsAvailable()).WillOnce(Return(false));
- EXPECT_CALL(adm, SetAGC(true)).WillOnce(Return(0));
+ AdmSetupExpectations(&adm);
StrictMock<webrtc::test::MockAudioProcessing> apm;
EXPECT_CALL(apm, ApplyConfig(testing::_));
EXPECT_CALL(apm, SetExtraOptions(testing::_));
@@ -121,12 +153,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
call_(webrtc::Call::Config(&event_log_)), voe_(&apm_, &transmit_mixer_),
override_field_trials_(field_trials) {
// AudioDeviceModule.
- EXPECT_CALL(adm_, AddRef()).WillOnce(Return(0));
- EXPECT_CALL(adm_, Release()).WillOnce(Return(0));
- EXPECT_CALL(adm_, BuiltInAECIsAvailable()).WillOnce(Return(false));
- EXPECT_CALL(adm_, BuiltInAGCIsAvailable()).WillOnce(Return(false));
- EXPECT_CALL(adm_, BuiltInNSIsAvailable()).WillOnce(Return(false));
- EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0));
+ AdmSetupExpectations(&adm_);
// AudioProcessing.
EXPECT_CALL(apm_, ApplyConfig(testing::_));
EXPECT_CALL(apm_, SetExtraOptions(testing::_));

Powered by Google App Engine
This is Rietveld 408576698