Unify Transport and newapi::Transport interfaces.

webrtc/video/end_to_end_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 <algorithm>
@@ skipping 41 matching lines @@
 class EndToEndTest : public test::CallTest {
  public:
   EndToEndTest() {}
 
   virtual ~EndToEndTest() {
     EXPECT_EQ(nullptr, send_stream_);
     EXPECT_TRUE(receive_streams_.empty());
   }
 
  protected:
-  class UnusedTransport : public newapi::Transport {
+  class UnusedTransport : public Transport {
    private:
     bool SendRtp(const uint8_t* packet, size_t length) override {
       ADD_FAILURE() << "Unexpected RTP sent.";
       return false;
     }
 
     bool SendRtcp(const uint8_t* packet, size_t length) override {
       ADD_FAILURE() << "Unexpected RTCP sent.";
       return false;
     }
@@ skipping 3049 matching lines @@
   EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
 
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyFecConfig(default_receive_config.rtp.fec);
 }
 
 }  // namespace webrtc