Delete VieRemb class, move functionality to PacketRouter.

webrtc/modules/pacing/packet_router.cc

Index: webrtc/modules/pacing/packet_router.cc
diff --git a/webrtc/modules/pacing/packet_router.cc b/webrtc/modules/pacing/packet_router.cc
index ab86bfa4bd82551f8014cd5652540e743577ce8d..18ec23f67b93163ebaab2365b13511ba09d6d220 100644
--- a/webrtc/modules/pacing/packet_router.cc
+++ b/webrtc/modules/pacing/packet_router.cc
@@ -12,6 +12,7 @@
 
 #include "webrtc/base/atomicops.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
@@ -121,7 +122,50 @@ uint16_t PacketRouter::AllocateSequenceNumber() {
   return new_seq;
 }
 
-bool PacketRouter::SendFeedback(rtcp::TransportFeedback* packet) {
+void PacketRouter::OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
+                                           uint32_t bitrate) {
+  const int kRembSendIntervalMs = 200;
+
+  // % threshold for if we should send a new REMB asap.
+  const uint32_t kSendThresholdPercent = 97;
+
+  int64_t now = rtc::TimeMillis();
+  {
+    rtc::CritScope lock(&remb_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) {
+      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_ = now - kRembSendIntervalMs;
+      }
+    }
+    bitrate_ = bitrate;
+
+    if (now - last_remb_time_ < kRembSendIntervalMs) {
+      return;
+    }
+    // NOTE: Updated if we intend to send the data; we might not have
+    // a module to actually send it.
+    last_remb_time_ = now;
+    last_send_bitrate_ = bitrate;
+  }
+  {
+    rtc::CritScope lock(&modules_crit_);
+    // TODO(nisse): Check REMB status of the modules? Or add a loop,

[nisse-webrtc, 2017/04/03 14:45:02]

Can you have another look at this logic? Can we get away with using the first
module?

+    // similar to SendTransportFeedback below?
+    if (!rtp_send_modules_.empty())
+      rtp_send_modules_.front()->SetREMBData(bitrate, ssrcs);
+    else if (!rtp_receive_modules_.empty())
+      rtp_receive_modules_.front()->SetREMBData(bitrate, ssrcs);
+  }
+}
+
+bool PacketRouter::SendTransportFeedback(rtcp::TransportFeedback* packet) {
   RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
   rtc::CritScope cs(&modules_crit_);
   // Prefer send modules.