Added various timestamps to FrameObject.

webrtc/modules/video_coding/packet_buffer.cc

 /*
  *  Copyright (c) 2016 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/modules/video_coding/packet_buffer.h"
 
 #include <algorithm>
 #include <limits>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/video_coding/frame_object.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 namespace video_coding {
 
-PacketBuffer::PacketBuffer(size_t start_buffer_size,
+PacketBuffer::PacketBuffer(Clock* clock,
+                           size_t start_buffer_size,
                            size_t max_buffer_size,
                            OnCompleteFrameCallback* frame_callback)
-    : size_(start_buffer_size),
+    : clock_(clock),
+      size_(start_buffer_size),
       max_size_(max_buffer_size),
       first_seq_num_(0),
       last_seq_num_(0),
       first_packet_received_(false),
       data_buffer_(start_buffer_size),
       sequence_buffer_(start_buffer_size),
       reference_finder_(frame_callback) {
   RTC_DCHECK_LE(start_buffer_size, max_buffer_size);
   // Buffer size must always be a power of 2.
   RTC_DCHECK((start_buffer_size & (start_buffer_size - 1)) == 0);
@@ skipping 119 matching lines @@
             max_nack_count, data_buffer_[start_index].timesNacked);
         sequence_buffer_[start_index].frame_created = true;
 
         if (sequence_buffer_[start_index].frame_begin)
           break;
 
         start_index = start_index > 0 ? start_index - 1 : size_ - 1;
         start_seq_num--;
       }
 
-      std::unique_ptr<RtpFrameObject> frame(new RtpFrameObject(
-          this, start_seq_num, seq_num, frame_size, max_nack_count));
+      std::unique_ptr<RtpFrameObject> frame(
+          new RtpFrameObject(this, start_seq_num, seq_num, frame_size,
+                             max_nack_count, clock_->TimeInMilliseconds()));
       reference_finder_.ManageFrame(std::move(frame));
     }
 
     index = (index + 1) % size_;
     ++seq_num;
   }
 }
 
 void PacketBuffer::ReturnFrame(RtpFrameObject* frame) {
   rtc::CritScope lock(&crit_);
@@ skipping 52 matching lines @@
 void PacketBuffer::Clear() {
   rtc::CritScope lock(&crit_);
   for (size_t i = 0; i < size_; ++i)
     sequence_buffer_[i].used = false;
 
   first_packet_received_ = false;
 }
 
 }  // namespace video_coding
 }  // namespace webrtc