Wire up send-side bandwidth estimation.

webrtc/modules/rtp_rtcp/source/rtcp_receiver.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.
  */
 
@@ skipping 18 matching lines @@
 // The number of RTCP time intervals needed to trigger a timeout.
 const int kRrTimeoutIntervals = 3;
 
 RTCPReceiver::RTCPReceiver(
     int32_t id,
     Clock* clock,
     bool receiver_only,
     RtcpPacketTypeCounterObserver* packet_type_counter_observer,
     RtcpBandwidthObserver* rtcp_bandwidth_observer,
     RtcpIntraFrameObserver* rtcp_intra_frame_observer,
+    TransportFeedbackObserver* transport_feedback_observer,
     ModuleRtpRtcpImpl* owner)
     : TMMBRHelp(),
       _clock(clock),
       receiver_only_(receiver_only),
       _method(kRtcpOff),
       _lastReceived(0),
       _rtpRtcp(*owner),
       _criticalSectionFeedbacks(
           CriticalSectionWrapper::CreateCriticalSection()),
       _cbRtcpBandwidthObserver(rtcp_bandwidth_observer),
       _cbRtcpIntraFrameObserver(rtcp_intra_frame_observer),
+      _cbTransportFeedbackObserver(transport_feedback_observer),
       _criticalSectionRTCPReceiver(
           CriticalSectionWrapper::CreateCriticalSection()),
       main_ssrc_(0),
       _remoteSSRC(0),
       _remoteSenderInfo(),
       _lastReceivedSRNTPsecs(0),
       _lastReceivedSRNTPfrac(0),
       _lastReceivedXRNTPsecs(0),
       _lastReceivedXRNTPfrac(0),
       xr_rr_rtt_ms_(0),
@@ skipping 283 matching lines @@
             HandlePsfbApp(*rtcpParser, rtcpPacketInformation);
             break;
           case RTCPPacketTypes::kApp:
             // generic application messages
             HandleAPP(*rtcpParser, rtcpPacketInformation);
             break;
           case RTCPPacketTypes::kAppItem:
             // generic application messages
             HandleAPPItem(*rtcpParser, rtcpPacketInformation);
             break;
+          case RTCPPacketTypes::kTransportFeedback:
+            HandleTransportFeedback(rtcpParser, &rtcpPacketInformation);
+            break;
         default:
             rtcpParser->Iterate();
             break;
         }
         pktType = rtcpParser->PacketType();
     }
 
     if (packet_type_counter_observer_ != NULL) {
       packet_type_counter_observer_->RtcpPacketTypesCounterUpdated(
           main_ssrc_, packet_type_counter_);
@@ skipping 883 matching lines @@
 
 void RTCPReceiver::HandleAPPItem(RTCPUtility::RTCPParserV2& rtcpParser,
                                  RTCPPacketInformation& rtcpPacketInformation) {
   const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
 
   rtcpPacketInformation.AddApplicationData(rtcpPacket.APP.Data, rtcpPacket.APP.Size);
 
   rtcpParser.Iterate();
 }
 
+void RTCPReceiver::HandleTransportFeedback(
+    RTCPUtility::RTCPParserV2* rtcp_parser,
+    RTCPHelp::RTCPPacketInformation* rtcp_packet_information) {
+  rtcp::RtcpPacket* packet = rtcp_parser->ReleaseRtcpPacket();
+  DCHECK(packet != nullptr);
+  rtcp_packet_information->rtcpPacketTypeFlags |= kRtcpTransportFeedback;

[stefan-webrtc, 2015/09/17 10:33:19]

I find it a bit strange that we use this flag when we could check the pointer
below to see if it's null. What do you think? I guess how you are doing it is
how it's done elsewhere, right? Is it possible to clean up in a follow-up?

[sprang_webrtc, 2015/09/17 15:08:28]

Yeah. I'm just following suit. Definitely the whole rtcp receiver et al needs an
overhaul.

[stefan-webrtc, 2015/09/18 10:57:23]

I think we have to outline how this is going to look like in the end, otherwise
we are running risks of not ending up somewhere suboptimal.

[sprang_webrtc, 2015/09/21 10:44:10]

Why would we want to end up somewhere suboptimal?
This sounds like a large discussion, let's do it offline?

[stefan-webrtc, 2015/09/21 11:13:16]

I mostly meant that we probably should have an idea of how this is going to look
in the end. Not something we need for this CL though. :)

+  rtcp_packet_information->transport_feedback_.reset(
+      static_cast<rtcp::TransportFeedback*>(packet));
+
+  rtcp_parser->Iterate();
+}
 int32_t RTCPReceiver::UpdateTMMBR() {
   int32_t numBoundingSet = 0;
   uint32_t bitrate = 0;
   uint32_t accNumCandidates = 0;
 
   int32_t size = TMMBRReceived(0, 0, NULL);
   if (size > 0) {
     TMMBRSet* candidateSet = VerifyAndAllocateCandidateSet(size);
     // Get candidate set from receiver.
     accNumCandidates = TMMBRReceived(size, accNumCandidates, candidateSet);
@@ skipping 49 matching lines @@
     // Might trigger a OnReceivedBandwidthEstimateUpdate.
     UpdateTMMBR();
   }
   unsigned int local_ssrc;
   {
     // We don't want to hold this critsect when triggering the callbacks below.
     CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
     local_ssrc = main_ssrc_;
   }
   if (!receiver_only_ &&
-      rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSrReq) {
+      (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSrReq)) {
     _rtpRtcp.OnRequestSendReport();
   }
   if (!receiver_only_ &&
-      rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpNack) {
+      (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpNack)) {
     if (rtcpPacketInformation.nackSequenceNumbers.size() > 0) {
       LOG(LS_VERBOSE) << "Incoming NACK length: "
                    << rtcpPacketInformation.nackSequenceNumbers.size();
       _rtpRtcp.OnReceivedNACK(rtcpPacketInformation.nackSequenceNumbers);
     }
   }
   {
     // We need feedback that we have received a report block(s) so that we
     // can generate a new packet in a conference relay scenario, one received
     // report can generate several RTCP packets, based on number relayed/mixed
@@ skipping 30 matching lines @@
       }
       if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr) ||
           (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRr)) {
         int64_t now = _clock->TimeInMilliseconds();
         _cbRtcpBandwidthObserver->OnReceivedRtcpReceiverReport(
             rtcpPacketInformation.report_blocks,
             rtcpPacketInformation.rtt,
             now);
       }
     }
+    if (_cbTransportFeedbackObserver &&
+        (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpTransportFeedback)) {
+      uint32_t media_source_ssrc =
+          rtcpPacketInformation.transport_feedback_->GetMediaSourceSsrc();
+      if (media_source_ssrc == main_ssrc_ ||
+          registered_ssrcs_.find(media_source_ssrc) !=
+              registered_ssrcs_.end()) {
+        _cbTransportFeedbackObserver->OnTransportFeedback(
+            *rtcpPacketInformation.transport_feedback_.get());
+      }
+    }
   }
 
   if (!receiver_only_) {
     CriticalSectionScoped cs(_criticalSectionFeedbacks);
     if (stats_callback_) {
       for (ReportBlockList::const_iterator it =
           rtcpPacketInformation.report_blocks.begin();
           it != rtcpPacketInformation.report_blocks.end();
           ++it) {
         RtcpStatistics stats;
@@ skipping 56 matching lines @@
         return -1;
       }
       num += receiveInfo->TmmbrSet.lengthOfSet();
       receiveInfoIt++;
     }
   }
   return num;
 }
 
 }  // namespace webrtc