Unify Transport and newapi::Transport interfaces.

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"
@@ skipping 72 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,
-                                newapi::Transport* send_transport) {
+                                Transport* send_transport) {
   assert(num_streams <= kNumSsrcs);
   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(
-    newapi::Transport* rtcp_send_transport) {
+    Transport* rtcp_send_transport) {
   assert(!send_config_.rtp.ssrcs.empty());
   assert(receive_configs_.empty());
   assert(allocated_decoders_.empty());
   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 =
@@ skipping 114 matching lines @@
                            const FakeNetworkPipe::Config& config)
     : BaseTest(timeout_ms, config) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc