Wire up RTP keep-alive in ortc api.

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 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 "webrtc/test/call_test.h"
 
 #include <algorithm>
 
 #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/include/voe_base.h"
-
 namespace webrtc {
 namespace test {
 
 namespace {
 const int kVideoRotationRtpExtensionId = 4;
 }
 
 CallTest::CallTest()
     : clock_(Clock::GetRealTimeClock()),
       event_log_(RtcEventLog::CreateNull()),
+      sender_call_transport_controller_(nullptr),
       video_send_config_(nullptr),
       video_send_stream_(nullptr),
       audio_send_config_(nullptr),
       audio_send_stream_(nullptr),
       fake_encoder_(clock_),
       num_video_streams_(1),
       num_audio_streams_(0),
       num_flexfec_streams_(0),
       decoder_factory_(CreateBuiltinAudioDecoderFactory()),
       encoder_factory_(CreateBuiltinAudioEncoderFactory()),
@@ skipping 16 matching lines @@
     apm_send_ = AudioProcessing::Create();
     apm_recv_ = AudioProcessing::Create();
     CreateVoiceEngines();
     AudioState::Config audio_state_config;
     audio_state_config.voice_engine = voe_send_.voice_engine;
     audio_state_config.audio_mixer = AudioMixerImpl::Create();
     audio_state_config.audio_processing = apm_send_;
     send_config.audio_state = AudioState::Create(audio_state_config);
   }
   CreateSenderCall(send_config);
+  if (sender_call_transport_controller_ != nullptr) {
+    test->OnRtpTransportControllerSendCreated(
+        sender_call_transport_controller_);
+  }
   if (test->ShouldCreateReceivers()) {
     Call::Config recv_config(test->GetReceiverCallConfig());
     if (num_audio_streams_ > 0) {
       AudioState::Config audio_state_config;
       audio_state_config.voice_engine = voe_recv_.voice_engine;
       audio_state_config.audio_mixer = AudioMixerImpl::Create();
       audio_state_config.audio_processing = apm_recv_;
       recv_config.audio_state = AudioState::Create(audio_state_config);
     }
     CreateReceiverCall(recv_config);
@@ skipping 99 matching lines @@
     video_send_stream_->Stop();
 }
 
 void CallTest::CreateCalls(const Call::Config& sender_config,
                            const Call::Config& receiver_config) {
   CreateSenderCall(sender_config);
   CreateReceiverCall(receiver_config);
 }
 
 void CallTest::CreateSenderCall(const Call::Config& config) {
-  sender_call_.reset(Call::Create(config));
+  sender_call_transport_controller_ = new RtpTransportControllerSend(
+      Clock::GetRealTimeClock(), config.event_log);
+
+  sender_call_.reset(
+      Call::Create(config, std::unique_ptr<RtpTransportControllerSend>(
+                               sender_call_transport_controller_)));
 }
 
 void CallTest::CreateReceiverCall(const Call::Config& config) {
   receiver_call_.reset(Call::Create(config));
 }
 
 void CallTest::DestroyCalls() {
   sender_call_.reset();
   receiver_call_.reset();
 }
@@ skipping 299 matching lines @@
 }
 
 Call::Config BaseTest::GetSenderCallConfig() {
   return Call::Config(event_log_.get());
 }
 
 Call::Config BaseTest::GetReceiverCallConfig() {
   return Call::Config(event_log_.get());
 }
 
+void BaseTest::OnRtpTransportControllerSendCreated(
+    RtpTransportControllerSend* controller) {}
+
 void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
 }
 
 test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) {
   return new PacketTransport(sender_call, this, test::PacketTransport::kSender,
                              CallTest::payload_type_map_,
                              FakeNetworkPipe::Config());
 }
 
 test::PacketTransport* BaseTest::CreateReceiveTransport() {
@@ skipping 59 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc