Simplify CongestionController.

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 "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/call/congestion_controller.h"
 #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_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/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
-#include "webrtc/video/call_stats.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
 using testing::_;
 using testing::Return;
 
 const int kChannelId = 1;
 const uint32_t kSsrc = 1234;
@@ skipping 11 matching lines @@
 const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, 0, -671};
 const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
 const int kTelephoneEventPayloadType = 123;
 const uint8_t kTelephoneEventCode = 45;
 const uint32_t kTelephoneEventDuration = 6789;
 
 struct ConfigHelper {
   ConfigHelper()
       : simulated_clock_(123456),
         stream_config_(nullptr),
-        call_stats_(&simulated_clock_),
-        process_thread_(ProcessThread::Create("AudioTestThread")),
         congestion_controller_(&simulated_clock_,
-                               process_thread_.get(),
-                               &call_stats_,
                                &bitrate_observer_,
                                &remote_bitrate_observer_) {
     using testing::Invoke;
     using testing::StrEq;
 
     EXPECT_CALL(voice_engine_,
         RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
     AudioState::Config config;
@@ skipping 83 matching lines @@
         .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoDelayMedian),
                         SetArgReferee<1>(kEchoDelayStdDev), Return(0)));
   }
 
  private:
   SimulatedClock simulated_clock_;
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioSendStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
-  CallStats call_stats_;
   testing::NiceMock<MockBitrateObserver> bitrate_observer_;
   testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
-  rtc::scoped_ptr<ProcessThread> process_thread_;
   CongestionController congestion_controller_;
 };
 }  // namespace
 
 TEST(AudioSendStreamTest, ConfigToString) {
   AudioSendStream::Config config(nullptr);
   config.rtp.ssrc = kSsrc;
   config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
   config.rtp.c_name = kCName;
@@ skipping 62 matching lines @@
       static_cast<internal::AudioState*>(helper.audio_state().get());
   VoiceEngineObserver* voe_observer =
       static_cast<VoiceEngineObserver*>(internal_audio_state);
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
   EXPECT_TRUE(send_stream.GetStats().typing_noise_detected);
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
   EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
 }
 }  // namespace test
 }  // namespace webrtc