modules: more interface -> include renames

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.
  */
 
 #include <string.h>
 
 #include <map>
 #include <vector>
 
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call.h"
 #include "webrtc/call/congestion_controller.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/video/video_receive_stream.h"
 #include "webrtc/video/video_send_stream.h"
 #include "webrtc/video_engine/call_stats.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 
@@ skipping 550 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