Remove unused RTP-header parser.

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 426 matching lines @@
       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
     }
 
     void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out waiting for packets to be NACKed, retransmitted and "
              "rendered.";
     }
 
     rtc::CriticalSection crit_;
-    rtc::scoped_ptr<RtpHeaderParser> rtp_parser_;
     std::set<uint16_t> dropped_packets_;
     std::set<uint16_t> retransmitted_packets_;
     uint64_t sent_rtp_packets_;
     int packets_left_to_drop_;
     int nacks_left_ GUARDED_BY(&crit_);
   } test;
 
   RunBaseTest(&test, FakeNetworkPipe::Config());
 }
 
@@ skipping 2766 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