Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

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/base/ptr_util.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/call.h"
 #include "webrtc/call/fake_rtp_transport_controller_send.h"
+#include "webrtc/call/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/rtp_rtcp/include/rtp_rtcp.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 {
 
 struct CallHelper {
   explicit CallHelper(
       rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr)
       : voice_engine_(decoder_factory) {
     webrtc::AudioState::Config audio_state_config;
     audio_state_config.voice_engine = &voice_engine_;
     audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create();
     EXPECT_CALL(voice_engine_, audio_device_module());
     EXPECT_CALL(voice_engine_, audio_processing());
     EXPECT_CALL(voice_engine_, audio_transport());
+
+    rtp_transport_send_ = rtc::MakeUnique<webrtc::RtpTransportControllerSend>(
+        webrtc::Clock::GetRealTimeClock(), &event_log_);
+
     webrtc::Call::Config config(&event_log_);
     config.audio_state = webrtc::AudioState::Create(audio_state_config);
-    call_.reset(webrtc::Call::Create(config));
+    config.audio_rtp_transport_send = rtp_transport_send_.get();
+    config.video_rtp_transport_send = rtp_transport_send_.get();
+    config.send_side_cc = rtp_transport_send_->send_side_cc();
+    call_ = rtc::WrapUnique(webrtc::Call::Create(config));
   }
 
   webrtc::Call* operator->() { return call_.get(); }
   webrtc::test::MockVoiceEngine* voice_engine() { return &voice_engine_; }
 
  private:
   testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_;
   webrtc::RtcEventLogNullImpl event_log_;
+  std::unique_ptr<webrtc::RtpTransportControllerSend> rtp_transport_send_;
   std::unique_ptr<webrtc::Call> call_;
 };
 }  // namespace
 
 namespace webrtc {
 
 TEST(CallTest, ConstructDestruct) {
   CallHelper call;
 }
 
@@ skipping 249 matching lines @@
   EXPECT_NE(stream, nullptr);
   streams.push_back(stream);
 
   for (auto s : streams) {
     call->DestroyFlexfecReceiveStream(s);
   }
 }
 
 namespace {
 struct CallBitrateHelper {
-  CallBitrateHelper() : CallBitrateHelper(Call::Config(&event_log_)) {}
+  CallBitrateHelper()
+      : mock_cc_(Clock::GetRealTimeClock(), &event_log_, &packet_router_),
+        transport_send(&packet_router_, &mock_cc_) {
+    Call::Config config(&event_log_);
+    config.audio_rtp_transport_send = &transport_send;
+    config.video_rtp_transport_send = &transport_send;
 
-  explicit CallBitrateHelper(const Call::Config& config)
-      : mock_cc_(Clock::GetRealTimeClock(), &event_log_, &packet_router_),
-        call_(Call::Create(
-            config,
-            rtc::MakeUnique<FakeRtpTransportControllerSend>(&packet_router_,
-                                                            &mock_cc_))) {}
-
+    call_ = rtc::WrapUnique(Call::Create(config));
+  }
   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<test::MockSendSideCongestionController> mock_cc_;
+  FakeRtpTransportControllerSend transport_send;
   std::unique_ptr<Call> call_;
 };
 }  // namespace
 
 TEST(CallBitrateTest, SetBitrateConfigWithValidConfigCallsSetBweBitrates) {
   CallBitrateHelper call;
 
   Call::Config::BitrateConfig bitrate_config;
   bitrate_config.min_bitrate_bps = 1;
   bitrate_config.start_bitrate_bps = 2;
@@ skipping 100 matching lines @@
     VoEBase* base;
   };
   ScopedVoiceEngine voice_engine;
 
   voice_engine.base->Init(&mock_adm);
   AudioState::Config audio_state_config;
   audio_state_config.voice_engine = voice_engine.voe;
   audio_state_config.audio_mixer = mock_mixer;
   auto audio_state = AudioState::Create(audio_state_config);
   RtcEventLogNullImpl event_log;
+  RtpTransportControllerSend rtp_transport_send(
+      webrtc::Clock::GetRealTimeClock(), &event_log);
   Call::Config call_config(&event_log);
   call_config.audio_state = audio_state;
+  call_config.audio_rtp_transport_send = &rtp_transport_send;
+  call_config.video_rtp_transport_send = &rtp_transport_send;
+  call_config.send_side_cc = rtp_transport_send.send_side_cc();
   std::unique_ptr<Call> call(Call::Create(call_config));
 
   auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) {
     AudioSendStream::Config config(nullptr);
     config.rtp.ssrc = ssrc;
     config.voe_channel_id = voice_engine.base->CreateChannel();
     AudioSendStream* stream = call->CreateAudioSendStream(config);
     VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine.voe);
     auto channel_proxy = voe_impl->GetChannelProxy(config.voe_channel_id);
     RtpRtcp* rtp_rtcp = nullptr;
@@ skipping 11 matching lines @@
   EXPECT_EQ(rtp_state1.sequence_number, rtp_state2.sequence_number);
   EXPECT_EQ(rtp_state1.start_timestamp, rtp_state2.start_timestamp);
   EXPECT_EQ(rtp_state1.timestamp, rtp_state2.timestamp);
   EXPECT_EQ(rtp_state1.capture_time_ms, rtp_state2.capture_time_ms);
   EXPECT_EQ(rtp_state1.last_timestamp_time_ms,
             rtp_state2.last_timestamp_time_ms);
   EXPECT_EQ(rtp_state1.media_has_been_sent, rtp_state2.media_has_been_sent);
 }
 
 }  // namespace webrtc