Added an empty AudioTransportProxy to AudioState.

webrtc/audio/audio_send_stream_unittest.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
 #include <string>
 #include <vector>
 
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "webrtc/modules/audio_device/include/mock_audio_device.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace webrtc {
 namespace test {
@@ skipping 33 matching lines @@
       : simulated_clock_(123456),
         stream_config_(nullptr),
         congestion_controller_(&simulated_clock_,
                                &bitrate_observer_,
                                &remote_bitrate_observer_,
                                &event_log_),
         bitrate_allocator_(&limit_observer_),
         worker_queue_("ConfigHelper_worker_queue") {
     using testing::Invoke;
 
-    EXPECT_CALL(voice_engine_,
-        RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
-    EXPECT_CALL(voice_engine_,
-        DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
+    EXPECT_CALL(voice_engine_, RegisterVoiceEngineObserver(_))
+        .WillOnce(Return(0));
+    EXPECT_CALL(voice_engine_, DeRegisterVoiceEngineObserver())
+        .WillOnce(Return(0));
+    EXPECT_CALL(voice_engine_, audio_transport());
+    EXPECT_CALL(voice_engine_, audio_processing());
+    EXPECT_CALL(voice_engine_, audio_device_module());
+    ON_CALL(voice_engine_, audio_device_module())
+        .WillByDefault(Return(&mock_audio_device_));
     AudioState::Config config;
     config.voice_engine = &voice_engine_;
     audio_state_ = AudioState::Create(config);
 
     SetupDefaultChannelProxy();
 
     EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
         .WillOnce(Invoke([this](int channel_id) {
           return channel_proxy_;
         }));
@@ skipping 102 matching lines @@
                         SetArgReferee<1>(kEchoReturnLossEnhancement),
                         Return(0)));
     EXPECT_CALL(voice_engine_, GetEcDelayMetrics(_, _, _))
         .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoDelayMedian),
                         SetArgReferee<1>(kEchoDelayStdDev), Return(0)));
   }
 
  private:
   SimulatedClock simulated_clock_;
   testing::StrictMock<MockVoiceEngine> voice_engine_;
+  testing::NiceMock<webrtc::test::MockAudioDeviceModule> mock_audio_device_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioSendStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
   testing::NiceMock<MockCongestionObserver> bitrate_observer_;
   testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
   CongestionController congestion_controller_;
   MockRtcEventLog event_log_;
   testing::NiceMock<MockLimitObserver> limit_observer_;
   BitrateAllocator bitrate_allocator_;
   // |worker_queue| is defined last to ensure all pending tasks are cancelled
@@ skipping 168 matching lines @@
   EXPECT_CALL(*helper.voice_engine(), SetVADStatus(kChannelId, true, _, _))
       .WillOnce(Return(0));
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
       helper.event_log());
 }
 
 }  // namespace test
 }  // namespace webrtc