Remove VoEHardware interface usage.

webrtc/media/engine/webrtcvoiceengine_unittest.cc

 /*
  *  Copyright (c) 2008 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 46 matching lines @@
 const uint32_t kSsrcZ = 0x42;
 const uint32_t kSsrcW = 0x02;
 const uint32_t kSsrcs4[] = { 11, 200, 30, 44 };
 
 constexpr int kRtpHistoryMs = 5000;
 
 class FakeVoEWrapper : public cricket::VoEWrapper {
  public:
   explicit FakeVoEWrapper(cricket::FakeWebRtcVoiceEngine* engine)
       : cricket::VoEWrapper(engine,  // base
-                            engine,  // codec
-                            engine) {  // hw
+                            engine) {  // codec
   }
 };
 
 class MockTransmitMixer : public webrtc::voe::TransmitMixer {
  public:
   MockTransmitMixer() = default;
   virtual ~MockTransmitMixer() = default;
 
   MOCK_METHOD1(EnableStereoChannelSwapping, void(bool enable));
 };
+
+void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) {
+  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(
+      testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>(
+          webrtc::AudioDeviceModule::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(
+      testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>(
+          webrtc::AudioDeviceModule::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::_));
   EXPECT_CALL(apm, Initialize()).WillOnce(Return(0));
   StrictMock<MockTransmitMixer> transmit_mixer;
   EXPECT_CALL(transmit_mixer, EnableStereoChannelSwapping(false));
   cricket::FakeWebRtcVoiceEngine voe(&apm, &transmit_mixer);
   EXPECT_FALSE(voe.IsInited());
   {
     cricket::WebRtcVoiceEngine engine(
@@ skipping 16 matching lines @@
 class WebRtcVoiceEngineTestFake : public testing::Test {
  public:
   WebRtcVoiceEngineTestFake() : WebRtcVoiceEngineTestFake("") {}
 
   explicit WebRtcVoiceEngineTestFake(const char* field_trials)
       : apm_gc_(*apm_.gain_control()), apm_ec_(*apm_.echo_cancellation()),
         apm_ns_(*apm_.noise_suppression()), apm_vd_(*apm_.voice_detection()),
         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::_));
     EXPECT_CALL(apm_, Initialize()).WillOnce(Return(0));
     // Default Options.
     EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
     EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
     EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0));
     EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0));
     EXPECT_CALL(apm_vd_, Enable(true)).WillOnce(Return(0));
@@ skipping 3667 matching lines @@
   // Without this cast, the comparison turned unsigned and, thus, failed for -1.
   const int num_specs = static_cast<int>(specs.size());
   EXPECT_GE(find_codec({"cn", 8000, 1}), num_specs);
   EXPECT_GE(find_codec({"cn", 16000, 1}), num_specs);
   EXPECT_EQ(find_codec({"cn", 32000, 1}), -1);
   EXPECT_GE(find_codec({"telephone-event", 8000, 1}), num_specs);
   EXPECT_GE(find_codec({"telephone-event", 16000, 1}), num_specs);
   EXPECT_GE(find_codec({"telephone-event", 32000, 1}), num_specs);
   EXPECT_GE(find_codec({"telephone-event", 48000, 1}), num_specs);
 }