Add send transports to individual webrtc::Call streams.

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 "webrtc/test/encoder_settings.h"
 
 namespace webrtc {
 namespace test {
 
 namespace {
 const int kVideoRotationRtpExtensionId = 4;
 }
 
 CallTest::CallTest()
     : clock_(Clock::GetRealTimeClock()),
+      send_config_(nullptr),
       send_stream_(NULL),
       fake_encoder_(clock_) {
 }
 
 CallTest::~CallTest() {
 }
 
 void CallTest::RunBaseTest(BaseTest* test) {
   CreateSenderCall(test->GetSenderCallConfig());
   if (test->ShouldCreateReceivers())
     CreateReceiverCall(test->GetReceiverCallConfig());
   test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
 
   if (test->ShouldCreateReceivers()) {
     test->SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver());
   } else {
     // Sender-only call delivers to itself.
     test->SetReceivers(sender_call_->Receiver(), NULL);
   }
 
-  CreateSendConfig(test->GetNumStreams());
+  CreateSendConfig(test->GetNumStreams(), test->SendTransport());
   if (test->ShouldCreateReceivers()) {
-    CreateMatchingReceiveConfigs();
+    CreateMatchingReceiveConfigs(test->ReceiveTransport());
   }
   test->ModifyConfigs(&send_config_, &receive_configs_, &encoder_config_);
   CreateStreams();
   test->OnStreamsCreated(send_stream_, receive_streams_);
 
   CreateFrameGeneratorCapturer();
   test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
 
   Start();
   test->PerformTest();
@@ skipping 26 matching lines @@
 }
 
 void CallTest::CreateSenderCall(const Call::Config& config) {
   sender_call_.reset(Call::Create(config));
 }
 
 void CallTest::CreateReceiverCall(const Call::Config& config) {
   receiver_call_.reset(Call::Create(config));
 }
 
-void CallTest::CreateSendConfig(size_t num_streams) {
+void CallTest::CreateSendConfig(size_t num_streams,
+                                newapi::Transport* send_transport) {
   assert(num_streams <= kNumSsrcs);
-  send_config_ = VideoSendStream::Config();
+  send_config_ = VideoSendStream::Config(send_transport);
   send_config_.encoder_settings.encoder = &fake_encoder_;
   send_config_.encoder_settings.payload_name = "FAKE";
   send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
   send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
   encoder_config_.streams = test::CreateVideoStreams(num_streams);
   for (size_t i = 0; i < num_streams; ++i)
     send_config_.rtp.ssrcs.push_back(kSendSsrcs[i]);
   send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
 }
 
-void CallTest::CreateMatchingReceiveConfigs() {
+void CallTest::CreateMatchingReceiveConfigs(
+    newapi::Transport* rtcp_send_transport) {
   assert(!send_config_.rtp.ssrcs.empty());
   assert(receive_configs_.empty());
   assert(allocated_decoders_.empty());
-  VideoReceiveStream::Config config;
+  VideoReceiveStream::Config config(rtcp_send_transport);
   config.rtp.remb = true;
   config.rtp.local_ssrc = kReceiverLocalSsrc;
   for (const RtpExtension& extension : send_config_.rtp.extensions)
     config.rtp.extensions.push_back(extension);
   for (size_t i = 0; i < send_config_.rtp.ssrcs.size(); ++i) {
     VideoReceiveStream::Decoder decoder =
         test::CreateMatchingDecoder(send_config_.encoder_settings);
     allocated_decoders_.push_back(decoder.decoder);
     config.decoders.clear();
     config.decoders.push_back(decoder);
@@ skipping 55 matching lines @@
 
 BaseTest::BaseTest(unsigned int timeout_ms,
                    const FakeNetworkPipe::Config& config)
     : RtpRtcpObserver(timeout_ms, config) {
 }
 
 BaseTest::~BaseTest() {
 }
 
 Call::Config BaseTest::GetSenderCallConfig() {
-  return Call::Config(SendTransport());
+  return Call::Config();
 }
 
 Call::Config BaseTest::GetReceiverCallConfig() {
-  return Call::Config(ReceiveTransport());
+  return Call::Config();
 }
 
 void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
 }
 
 size_t BaseTest::GetNumStreams() const {
   return 1;
 }
 
 void BaseTest::ModifyConfigs(
@@ skipping 30 matching lines @@
                            const FakeNetworkPipe::Config& config)
     : BaseTest(timeout_ms, config) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc