modules/video_coding refactorings

webrtc/video_engine/vie_sync_module.cc

 /*
  *  Copyright (c) 2012 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 "webrtc/video_engine/vie_sync_module.h"
 
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/video_engine/stream_synchronization.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {
 
 int UpdateMeasurements(StreamSynchronization::Measurements* stream,
                        const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) {
   if (!receiver.Timestamp(&stream->latest_timestamp))
     return -1;
@@ skipping 137 matching lines @@
 
   if (voe_sync_interface_->SetMinimumPlayoutDelay(
       voe_channel_id_, target_audio_delay_ms) == -1) {
     LOG(LS_ERROR) << "Error setting voice delay.";
   }
   vcm_->SetMinimumPlayoutDelay(target_video_delay_ms);
   return 0;
 }
 
 }  // namespace webrtc