Move PacedSender ownership to RtpTransportControllerSend.

webrtc/call/call_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 <list>
 #include <map>
 #include <memory>
 #include <utility>
 
 #include "webrtc/api/test/mock_audio_mixer.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/call.h"
 #include "webrtc/call/fake_rtp_transport_controller_send.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_device/include/mock_audio_device.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/congestion_controller/include/mock/mock_send_side_congestion_controller.h"
+#include "webrtc/modules/pacing/mock/mock_paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_audio_decoder_factory.h"
 #include "webrtc/test/mock_transport.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace {
 
@@ skipping 283 matching lines @@
   for (auto s : streams) {
     call->DestroyFlexfecReceiveStream(s);
   }
 }
 
 namespace {
 struct CallBitrateHelper {
   CallBitrateHelper() : CallBitrateHelper(Call::Config::BitrateConfig()) {}
 
   explicit CallBitrateHelper(const Call::Config::BitrateConfig& bitrate_config)
-      : mock_cc_(Clock::GetRealTimeClock(), &event_log_, &packet_router_) {
+      : mock_cc_(Clock::GetRealTimeClock(), &event_log_, &pacer_) {
     Call::Config config(&event_log_);
     config.bitrate_config = bitrate_config;
     call_.reset(
         Call::Create(config, rtc::MakeUnique<FakeRtpTransportControllerSend>(
-                                 &packet_router_, &mock_cc_)));
+                                 &packet_router_, &pacer_, &mock_cc_)));
   }
 
   webrtc::Call* operator->() { return call_.get(); }
   testing::NiceMock<test::MockSendSideCongestionController>& mock_cc() {
     return mock_cc_;
   }
 
  private:
   webrtc::RtcEventLogNullImpl event_log_;
   PacketRouter packet_router_;
+  testing::NiceMock<MockPacedSender> pacer_;
   testing::NiceMock<test::MockSendSideCongestionController> mock_cc_;
   std::unique_ptr<Call> call_;
 };
 }  // namespace
 
 TEST(CallBitrateTest, SetBitrateConfigWithValidConfigCallsSetBweBitrates) {
   CallBitrateHelper call;
 
   Call::Config::BitrateConfig bitrate_config;
   bitrate_config.min_bitrate_bps = 1;
@@ skipping 354 matching lines @@
   mask.min_bitrate_bps = rtc::Optional<int>(2000);
   EXPECT_CALL(call.mock_cc(), SetBweBitrates(1000, -1, 1000));
   call->SetBitrateConfigMask(mask);
 
   // Set min to 3000; the clamped value stays the same so nothing happens.
   mask.min_bitrate_bps = rtc::Optional<int>(3000);
   call->SetBitrateConfigMask(mask);
 }
 
 }  // namespace webrtc