Delete VieRemb class, move functionality to PacketRouter.

webrtc/modules/pacing/packet_router.cc

 /*
  *  Copyright (c) 2015 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/pacing/packet_router.h"
 
 #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"
 
 namespace webrtc {
 
 PacketRouter::PacketRouter() : transport_seq_(0) {
   pacer_thread_checker_.DetachFromThread();
 }
 
@@ skipping 89 matching lines @@
     // time the CAS operation was executed. Thus, if prev_seq is returned, the
     // operation was successful - otherwise we need to retry. Saving the
     // return value saves us a load on retry.
     prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq,
                                               new_seq);
   } while (prev_seq != desired_prev_seq);
 
   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.
   for (auto* rtp_module : rtp_send_modules_) {
     packet->SetSenderSsrc(rtp_module->SSRC());
     if (rtp_module->SendFeedbackPacket(*packet))
       return true;
   }
   for (auto* rtp_module : rtp_receive_modules_) {
     packet->SetSenderSsrc(rtp_module->SSRC());
     if (rtp_module->SendFeedbackPacket(*packet))
       return true;
   }
   return false;
 }
 
 }  // namespace webrtc