Move PacedSender ownership to RtpTransportControllerSend.

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 <utility>
 #include <vector>
 
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/call/fake_rtp_transport_controller_send.h"
 #include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_observer.h"
 #include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
-#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/pacing/mock/mock_paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/rtc_base/task_queue.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_audio_encoder.h"
 #include "webrtc/test/mock_audio_encoder_factory.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
 #include "webrtc/voice_engine/transmit_mixer.h"
@@ skipping 96 matching lines @@
 
 struct ConfigHelper {
   ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call)
       : stream_config_(nullptr),
         audio_processing_(new rtc::RefCountedObject<MockAudioProcessing>()),
         simulated_clock_(123456),
         send_side_cc_(rtc::MakeUnique<SendSideCongestionController>(
             &simulated_clock_,
             nullptr /* observer */,
             &event_log_,
-            &packet_router_)),
-        fake_transport_(&packet_router_, send_side_cc_.get()),
+            &pacer_)),
+        fake_transport_(&packet_router_, &pacer_, send_side_cc_.get()),
         bitrate_allocator_(&limit_observer_),
         worker_queue_("ConfigHelper_worker_queue"),
         audio_encoder_(nullptr) {
     using testing::Invoke;
 
     EXPECT_CALL(voice_engine_,
         RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_, audio_device_module());
@@ skipping 174 matching lines @@
  private:
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioSendStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
   rtc::scoped_refptr<MockAudioProcessing> audio_processing_;
   MockTransmitMixer transmit_mixer_;
   AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
   SimulatedClock simulated_clock_;
   PacketRouter packet_router_;
+  testing::NiceMock<MockPacedSender> pacer_;
   std::unique_ptr<SendSideCongestionController> send_side_cc_;
   FakeRtpTransportControllerSend fake_transport_;
   MockRtcEventLog event_log_;
   MockRtpRtcp rtp_rtcp_;
   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_;
@@ skipping 256 matching lines @@
     EXPECT_CALL(*helper.channel_proxy(), RegisterSenderCongestionControlObjects(
         helper.transport(), Ne(nullptr)))
         .Times(1);
   }
   send_stream.Reconfigure(new_config);
 }
 
 
 }  // namespace test
 }  // namespace webrtc