Add send transports to individual webrtc::Call streams.

webrtc/video/bitrate_estimator_tests.cc

 /*
  *  Copyright (c) 2013 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 <functional>
@@ skipping 103 matching lines @@
 static const int kASTExtensionId = 5;
 
 class BitrateEstimatorTest : public test::CallTest {
  public:
   BitrateEstimatorTest()
       : receiver_trace_(),
         send_transport_(),
         receive_transport_(),
         sender_call_(),
         receiver_call_(),
-        receive_config_(),
+        receive_config_(nullptr),
         streams_() {
   }
 
   virtual ~BitrateEstimatorTest() {
     EXPECT_TRUE(streams_.empty());
   }
 
   virtual void SetUp() {
-    Call::Config receiver_call_config(&receive_transport_);
-    receiver_call_.reset(Call::Create(receiver_call_config));
-
-    Call::Config sender_call_config(&send_transport_);
-    sender_call_.reset(Call::Create(sender_call_config));
+    receiver_call_.reset(Call::Create(Call::Config()));
+    sender_call_.reset(Call::Create(Call::Config()));
 
     send_transport_.SetReceiver(receiver_call_->Receiver());
     receive_transport_.SetReceiver(sender_call_->Receiver());
 
-    send_config_ = VideoSendStream::Config();
+    send_config_ = VideoSendStream::Config(&send_transport_);
     send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]);
     // Encoders will be set separately per stream.
     send_config_.encoder_settings.encoder = nullptr;
     send_config_.encoder_settings.payload_name = "FAKE";
     send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
     encoder_config_.streams = test::CreateVideoStreams(1);
 
-    receive_config_ = VideoReceiveStream::Config();
+    receive_config_ = VideoReceiveStream::Config(&receive_transport_);
     // receive_config_.decoders will be set by every stream separately.
     receive_config_.rtp.remote_ssrc = send_config_.rtp.ssrcs[0];
     receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
     receive_config_.rtp.remb = true;
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
   }
 
@@ skipping 201 matching lines @@
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
   receiver_trace_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
 }
 }  // namespace webrtc