Nuke TickTime::UseFakeClock.

webrtc/video_engine/vie_remb.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_remb.h"
 
 #include <assert.h>
 
 #include <algorithm>
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
 const int kRembSendIntervalMs = 200;
 
 // % threshold for if we should send a new REMB asap.
 const unsigned int kSendThresholdPercent = 97;
 
-VieRemb::VieRemb()
-    : list_crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      last_remb_time_(TickTime::MillisecondTimestamp()),
+VieRemb::VieRemb(Clock* clock)
+    : clock_(clock),
+      list_crit_(CriticalSectionWrapper::CreateCriticalSection()),
+      last_remb_time_(clock_->TimeInMilliseconds()),
       last_send_bitrate_(0),
       bitrate_(0) {}
 
 VieRemb::~VieRemb() {}
 
 void VieRemb::AddReceiveChannel(RtpRtcp* rtp_rtcp) {
   assert(rtp_rtcp);
 
   CriticalSectionScoped cs(list_crit_.get());
   if (std::find(receive_modules_.begin(), receive_modules_.end(), rtp_rtcp) !=
@@ skipping 55 matching lines @@
                                       unsigned int bitrate) {
   list_crit_->Enter();
   // If we already have an estimate, check if the new total estimate is below
   // kSendThresholdPercent of the previous estimate.
   if (last_send_bitrate_ > 0) {
     unsigned int new_remb_bitrate = last_send_bitrate_ - bitrate_ + bitrate;
 
     if (new_remb_bitrate < kSendThresholdPercent * last_send_bitrate_ / 100) {
       // The new bitrate estimate is less than kSendThresholdPercent % of the
       // last report. Send a REMB asap.
-      last_remb_time_ = TickTime::MillisecondTimestamp() - kRembSendIntervalMs;
+      last_remb_time_ = clock_->TimeInMilliseconds() - kRembSendIntervalMs;
     }
   }
   bitrate_ = bitrate;
 
   // Calculate total receive bitrate estimate.
-  int64_t now = TickTime::MillisecondTimestamp();
+  int64_t now = clock_->TimeInMilliseconds();
 
   if (now - last_remb_time_ < kRembSendIntervalMs) {
     list_crit_->Leave();
     return;
   }
   last_remb_time_ = now;
 
   if (ssrcs.empty() || receive_modules_.empty()) {
     list_crit_->Leave();
     return;
   }
 
   // Send a REMB packet.
   RtpRtcp* sender = NULL;
   if (!rtcp_sender_.empty()) {
     sender = rtcp_sender_.front();
   } else {
     sender = receive_modules_.front();
   }
   last_send_bitrate_ = bitrate_;
 
   list_crit_->Leave();
 
   if (sender) {
     sender->SetREMBData(bitrate_, ssrcs);
   }
 }
 
 }  // namespace webrtc