Changed FakeVoiceEngine into a MockVoiceEngine.

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 <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_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/fake_voice_engine.h"
+#include "webrtc/test/mock_voice_engine.h"
 #include "webrtc/test/frame_generator_capturer.h"
 
 namespace webrtc {
 namespace {
 // Note: If you consider to re-use this class, think twice and instead consider
 // writing tests that don't depend on the trace system.
 class TraceObserver {
  public:
   TraceObserver() {
     Trace::set_level_filter(kTraceTerseInfo);
 
     Trace::CreateTrace();
     Trace::SetTraceCallback(&callback_);
 
     // Call webrtc trace to initialize the tracer that would otherwise trigger a
     // data-race if left to be initialized by multiple threads (i.e. threads
     // spawned by test::DirectTransport members in BitrateEstimatorTest).
-    WEBRTC_TRACE(kTraceStateInfo,
-                 kTraceUtility,
-                 -1,
+    WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
                  "Instantiate without data races.");
   }
 
   ~TraceObserver() {
     Trace::SetTraceCallback(nullptr);
     Trace::ReturnTrace();
   }
 
   void PushExpectedLogLine(const std::string& expected_log_line) {
     callback_.PushExpectedLogLine(expected_log_line);
   }
 
-  EventTypeWrapper Wait() {
-    return callback_.Wait();
-  }
+  EventTypeWrapper Wait() { return callback_.Wait(); }
 
  private:
   class Callback : public TraceCallback {
    public:
     Callback() : done_(EventWrapper::Create()) {}
 
     void Print(TraceLevel level, const char* message, int length) override {
       rtc::CritScope lock(&crit_sect_);
       std::string msg(message);
       if (msg.find("BitrateEstimator") != std::string::npos) {
@@ skipping 37 matching lines @@
 };
 }  // namespace
 
 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 ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }
 
   virtual void SetUp() {
+    EXPECT_CALL(mock_voice_engine_, GetEventLog())
+        .WillRepeatedly(testing::Return(nullptr));
+
     Call::Config config;
-    config.voice_engine = &fake_voice_engine_;
+    config.voice_engine = &mock_voice_engine_;
     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());
 
     send_config_ = VideoSendStream::Config(send_transport_.get());
     send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]);
     // Encoders will be set separately per stream.
     send_config_.encoder_settings.encoder = nullptr;
     send_config_.encoder_settings.payload_name = "FAKE";
     send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
     encoder_config_.streams = test::CreateVideoStreams(1);
 
     receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
     // receive_config_.decoders will be set by every stream separately.
     receive_config_.rtp.remote_ssrc = send_config_.rtp.ssrcs[0];
     receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
     receive_config_.rtp.remb = true;
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
   }
 
   virtual void TearDown() {
     std::for_each(streams_.begin(), streams_.end(),
-        std::mem_fun(&Stream::StopSending));
+                  std::mem_fun(&Stream::StopSending));
 
     send_transport_->StopSending();
     receive_transport_->StopSending();
 
     while (!streams_.empty()) {
       delete streams_.back();
       streams_.pop_back();
     }
 
     receiver_call_.reset();
@@ skipping 12 matching lines @@
           video_receive_stream_(nullptr),
           frame_generator_capturer_(),
           fake_encoder_(Clock::GetRealTimeClock()),
           fake_decoder_() {
       test_->send_config_.rtp.ssrcs[0]++;
       test_->send_config_.encoder_settings.encoder = &fake_encoder_;
       send_stream_ = test_->sender_call_->CreateVideoSendStream(
           test_->send_config_, test_->encoder_config_);
       RTC_DCHECK_EQ(1u, test_->encoder_config_.streams.size());
       frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
-          send_stream_->Input(),
-          test_->encoder_config_.streams[0].width,
-          test_->encoder_config_.streams[0].height,
-          30,
+          send_stream_->Input(), test_->encoder_config_.streams[0].width,
+          test_->encoder_config_.streams[0].height, 30,
           Clock::GetRealTimeClock()));
       send_stream_->Start();
       frame_generator_capturer_->Start();
 
       if (receive_audio) {
         AudioReceiveStream::Config receive_config;
         receive_config.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0];
         // Bogus non-default id to prevent hitting a RTC_DCHECK when creating
         // the AudioReceiveStream. Every receive stream has to correspond to
         // an underlying channel id.
@@ skipping 51 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::FakeVoiceEngine fake_voice_engine_;
+  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