Move some send stream configuration into webrtc::AudioSendStream.

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>
@@ skipping 100 matching lines @@
 static const int kTOFExtensionId = 4;
 static const int kASTExtensionId = 5;
 
 class BitrateEstimatorTest : public test::CallTest {
  public:
   BitrateEstimatorTest() : receive_config_(nullptr) {}
 
   virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }
 
   virtual void SetUp() {
-    EXPECT_CALL(mock_voice_engine_,
-        RegisterVoiceEngineObserver(testing::_)).WillOnce(testing::Return(0));
-    EXPECT_CALL(mock_voice_engine_,
-        DeRegisterVoiceEngineObserver()).WillOnce(testing::Return(0));
-    EXPECT_CALL(mock_voice_engine_, GetEventLog())
-        .WillRepeatedly(testing::Return(nullptr));
-
     AudioState::Config audio_state_config;
     audio_state_config.voice_engine = &mock_voice_engine_;
     Call::Config config;
     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()));
@@ skipping 120 matching lines @@
     BitrateEstimatorTest* test_;
     bool is_sending_receiving_;
     VideoSendStream* send_stream_;
     AudioReceiveStream* audio_receive_stream_;
     VideoReceiveStream* video_receive_stream_;
     rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
     test::FakeEncoder fake_encoder_;
     test::FakeDecoder fake_decoder_;
   };
 
-  test::MockVoiceEngine mock_voice_engine_;
+  testing::NiceMock<test::MockVoiceEngine> mock_voice_engine_;
   TraceObserver receiver_trace_;
   rtc::scoped_ptr<test::DirectTransport> send_transport_;
   rtc::scoped_ptr<test::DirectTransport> receive_transport_;
   rtc::scoped_ptr<Call> sender_call_;
   rtc::scoped_ptr<Call> receiver_call_;
   VideoReceiveStream::Config receive_config_;
   std::vector<Stream*> streams_;
 };
 
 static const char* kAbsSendTimeLog =
@@ skipping 81 matching lines @@
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
   receiver_trace_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
 }
 }  // namespace webrtc