Clean up of CongestionController.

webrtc/video/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/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/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;
+const uint32_t kSendThresholdPercent = 97;
 
 VieRemb::VieRemb(Clock* clock)
     : clock_(clock),
       last_remb_time_(clock_->TimeInMilliseconds()),
       last_send_bitrate_(0),
       bitrate_(0) {}
 
 VieRemb::~VieRemb() {}
 
 void VieRemb::AddReceiveChannel(RtpRtcp* rtp_rtcp) {
@@ skipping 45 matching lines @@
       return;
     }
   }
 }
 
 bool VieRemb::InUse() const {
   rtc::CritScope lock(&list_crit_);
   return !receive_modules_.empty() || !rtcp_sender_.empty();
 }
 
-void VieRemb::OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
-                                      unsigned int bitrate) {
+void VieRemb::OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
+                                      uint32_t bitrate) {
   RtpRtcp* sender = NULL;
   {
     rtc::CritScope lock(&list_crit_);
     // 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;
+      uint32_t 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_ = clock_->TimeInMilliseconds() - kRembSendIntervalMs;
       }
     }
     bitrate_ = bitrate;
 
     // Calculate total receive bitrate estimate.
@@ skipping 16 matching lines @@
     }
     last_send_bitrate_ = bitrate_;
   }
 
   if (sender) {
     sender->SetREMBData(bitrate_, ssrcs);
   }
 }
 
 }  // namespace webrtc