Split CongestionController into send- and receive-side classes.

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.
  */
 
@@ skipping 56 matching lines @@
  public:
   MOCK_CONST_METHOD0(AudioLevelFullRange, int16_t());
 };
 
 struct ConfigHelper {
   explicit ConfigHelper(bool audio_bwe_enabled)
       : simulated_clock_(123456),
         stream_config_(nullptr),
         congestion_controller_(&simulated_clock_,
                                &bitrate_observer_,
-                               nullptr,
                                &event_log_,
                                &packet_router_),
         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));
@@ skipping 31 matching lines @@
     // calls from the default ctor behavior.
     stream_config_.send_codec_spec.codec_inst = kIsacCodec;
     stream_config_.min_bitrate_bps = 10000;
     stream_config_.max_bitrate_bps = 65000;
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
   PacketRouter* packet_router() { return &packet_router_; }
-  CongestionController* congestion_controller() {
+  SendSideCongestionController* congestion_controller() {
     return &congestion_controller_;
   }
   BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
   rtc::TaskQueue* worker_queue() { return &worker_queue_; }
   RtcEventLog* event_log() { return &event_log_; }
   MockVoiceEngine* voice_engine() { return &voice_engine_; }
 
   void SetupDefaultChannelProxy(bool audio_bwe_enabled) {
     using testing::StrEq;
     channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
@@ skipping 102 matching lines @@
   SimulatedClock simulated_clock_;
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioSendStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
   testing::NiceMock<MockCongestionObserver> bitrate_observer_;
   MockAudioProcessing audio_processing_;
   MockTransmitMixer transmit_mixer_;
   AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
   PacketRouter packet_router_;
-  CongestionController congestion_controller_;
+  SendSideCongestionController congestion_controller_;
   MockRtcEventLog event_log_;
   MockRtcpRttStats rtcp_rtt_stats_;
   testing::NiceMock<MockLimitObserver> limit_observer_;
   BitrateAllocator bitrate_allocator_;
   // |worker_queue| is defined last to ensure all pending tasks are cancelled
   // and deleted before any other members.
   rtc::TaskQueue worker_queue_;
 };
 }  // namespace
 
@@ skipping 202 matching lines @@
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.packet_router(), helper.congestion_controller(),
       helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
   EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000));
   send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000);
 }
 
 }  // namespace test
 }  // namespace webrtc