Added an empty AudioTransportProxy to AudioState.

webrtc/call/bitrate_estimator_tests.cc

 /*
  *  Copyright (c) 2013 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 <functional>
 #include <list>
 #include <memory>
 #include <string>
 
 #include "webrtc/api/call/audio_state.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #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"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voice_engine.h"
@@ skipping 71 matching lines @@
 class BitrateEstimatorTest : public test::CallTest {
  public:
   BitrateEstimatorTest() : mock_voice_engine_(decoder_factory_),
                            receive_config_(nullptr) {}
 
   virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }
 
   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 here instead? I don't think it
uses it any longer.

[aleloi, 2016/11/08 11:46:41]

I've done it in another CL; I'll rebase this one as soon as
https://codereview.webrtc.org/2479383003/has landed.

     Call::Config config(&event_log_);
     config.audio_state = AudioState::Create(audio_state_config);
     receiver_call_.reset(Call::Create(config));
     sender_call_.reset(Call::Create(config));
 
     send_transport_.reset(new test::DirectTransport(sender_call_.get()));
     send_transport_->SetReceiver(receiver_call_->Receiver());
     receive_transport_.reset(new test::DirectTransport(receiver_call_.get()));
     receive_transport_->SetReceiver(sender_call_->Receiver());
 
@@ skipping 123 matching lines @@
     bool is_sending_receiving_;
     VideoSendStream* send_stream_;
     AudioReceiveStream* audio_receive_stream_;
     VideoReceiveStream* video_receive_stream_;
     std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
     test::FakeEncoder fake_encoder_;
     test::FakeDecoder fake_decoder_;
   };
 
   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_;
   std::unique_ptr<Call> sender_call_;
   std::unique_ptr<Call> receiver_call_;
   VideoReceiveStream::Config receive_config_;
   std::vector<Stream*> streams_;
 };
 
 static const char* kAbsSendTimeLog =
@@ skipping 58 matching lines @@
       RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc