From 9d4762e8b65b6694d06220c2a34b8b953c53c3c5 Mon Sep 17 00:00:00 2001
From: "stefan@webrtc.org"
 <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Date: Mon, 24 Mar 2014 17:13:00 +0000
Subject: [PATCH] Have changes to REMB trigger RTCP to be sent immediately.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5763 4adac7df-926f-26a2-2b94-8c16560cd09d
---
 webrtc/modules/rtp_rtcp/source/rtcp_sender.cc | 10 +++++++---
 webrtc/video_engine/vie_remb.cc               | 12 +++---------
 2 files changed, 10 insertions(+), 12 deletions(-)

diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index f3c33bf6d..6e6edf816 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -348,6 +348,9 @@ RTCPSender::SetREMBData(const uint32_t bitrate,
         _rembSSRC[i] = SSRC[i];
     }
     _sendREMB = true;
+    // Send a REMB immediately if we have a new REMB. The frequency of REMBs is
+    // throttled by the caller.
+    _nextTimeToSendRTCP = _clock->TimeInMilliseconds();
     return 0;
 }
 
@@ -483,14 +486,15 @@ RTCPSender::TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP) const
     For audio we use a fix 5 sec interval
 
     For video we use 1 sec interval fo a BW smaller than 360 kbit/s,
-        technicaly we break the max 5% RTCP BW for video below 10 kbit/s but that should be extreamly rare
+        technicaly we break the max 5% RTCP BW for video below 10 kbit/s but
+        that should be extremely rare
 
 
 From RFC 3550
 
     MAX RTCP BW is 5% if the session BW
         A send report is approximately 65 bytes inc CNAME
-        A report report is approximately 28 bytes
+        A receiver report is approximately 28 bytes
 
     The RECOMMENDED value for the reduced minimum in seconds is 360
       divided by the session bandwidth in kilobits/second.  This minimum
@@ -552,7 +556,7 @@ From RFC 3550
         now += RTCP_SEND_BEFORE_KEY_FRAME_MS;
     }
 
-    if(now > _nextTimeToSendRTCP)
+    if(now >= _nextTimeToSendRTCP)
     {
         return true;
 
diff --git a/webrtc/video_engine/vie_remb.cc b/webrtc/video_engine/vie_remb.cc
index 6b0f161d9..d318a4bb2 100644
--- a/webrtc/video_engine/vie_remb.cc
+++ b/webrtc/video_engine/vie_remb.cc
@@ -22,8 +22,7 @@
 
 namespace webrtc {
 
-const int kRembSendIntervallMs = 1000;
-const unsigned int kRembMinimumBitrateKbps = 50;
+const int kRembSendIntervalMs = 200;
 
 // % threshold for if we should send a new REMB asap.
 const unsigned int kSendThresholdPercent = 97;
@@ -117,7 +116,7 @@ void VieRemb::OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
     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() - kRembSendIntervallMs;
+      last_remb_time_ = TickTime::MillisecondTimestamp() - kRembSendIntervalMs;
     }
   }
   bitrate_ = bitrate;
@@ -125,7 +124,7 @@ void VieRemb::OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
   // Calculate total receive bitrate estimate.
   int64_t now = TickTime::MillisecondTimestamp();
 
-  if (now - last_remb_time_ < kRembSendIntervallMs) {
+  if (now - last_remb_time_ < kRembSendIntervalMs) {
     list_crit_->Leave();
     return;
   }
@@ -145,11 +144,6 @@ void VieRemb::OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
   }
   last_send_bitrate_ = bitrate_;
 
-  // Never send a REMB lower than last_send_bitrate_.
-  if (last_send_bitrate_ < kRembMinimumBitrateKbps) {
-    last_send_bitrate_ = kRembMinimumBitrateKbps;
-  }
-
   list_crit_->Leave();
 
   if (sender) {