Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/

webrtc/call/call.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.
  */
 
@@ skipping 190 matching lines @@
     ReceiveRtpConfig(const std::vector<RtpExtension>& extensions,
                      bool transport_cc)
         : extensions(extensions), transport_cc(transport_cc) {}
 
     // Registered RTP header extensions for each stream. Note that RTP header
     // extensions are negotiated per track ("m= line") in the SDP, but we have
     // no notion of tracks at the Call level. We therefore store the RTP header
     // extensions per SSRC instead, which leads to some storage overhead.
     RtpHeaderExtensionMap extensions;
     // Set if the RTCP feedback message needed for send side BWE was negotiated.
-    bool transport_cc;
+    bool transport_cc = false;
   };
   std::map<uint32_t, ReceiveRtpConfig> receive_rtp_config_
       GUARDED_BY(receive_crit_);
 
   std::unique_ptr<RWLockWrapper> send_crit_;
   // Audio and Video send streams are owned by the client that creates them.
   std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_ GUARDED_BY(send_crit_);
   std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_);
   std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_);
 
@@ skipping 1035 matching lines @@
   if (media_type != MediaType::AUDIO ||
       (transport_cc && header.extension.hasTransportSequenceNumber)) {
     congestion_controller_->OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc