Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors.

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 108 matching lines @@
 
   const Clock* const clock_;
 
   const int num_cpu_cores_;
   const rtc::scoped_ptr<ProcessThread> module_process_thread_;
   const rtc::scoped_ptr<CallStats> call_stats_;
   const rtc::scoped_ptr<BitrateAllocator> bitrate_allocator_;
   Call::Config config_;
   rtc::ThreadChecker configuration_thread_checker_;
 
- bool network_enabled_;
+  bool network_enabled_;
 
   rtc::scoped_ptr<RWLockWrapper> receive_crit_;
   // Audio and Video receive streams are owned by the client that creates them.
   std::map<uint32_t, AudioReceiveStream*> audio_receive_ssrcs_
       GUARDED_BY(receive_crit_);
   std::map<uint32_t, VideoReceiveStream*> video_receive_ssrcs_
       GUARDED_BY(receive_crit_);
   std::set<VideoReceiveStream*> video_receive_streams_
       GUARDED_BY(receive_crit_);
   std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
@@ skipping 595 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc