Add stats for frequency offset when converting RTP timestamp to NTP time.

webrtc/video/stream_synchronization.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_
 #define WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_
 
 #include <list>
 
 #include "webrtc/system_wrappers/include/rtp_to_ntp.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class StreamSynchronization {
  public:
   struct Measurements {
     Measurements() : rtcp(), latest_receive_time_ms(0), latest_timestamp(0) {}
-    RtcpList rtcp;
+    RtcpMeasurements rtcp;
     int64_t latest_receive_time_ms;
     uint32_t latest_timestamp;
   };
 
   StreamSynchronization(uint32_t video_primary_ssrc, int audio_channel_id);
 
   bool ComputeDelays(int relative_delay_ms,
                      int current_audio_delay_ms,
                      int* extra_audio_delay_ms,
                      int* total_video_delay_target_ms);
@@ skipping 18 matching lines @@
 
   SynchronizationDelays channel_delay_;
   const uint32_t video_primary_ssrc_;
   const int audio_channel_id_;
   int base_target_delay_ms_;
   int avg_diff_ms_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_