Calculate the available bandwidth before sending a TMMBR

Also changed the way TMMBR was processed since it did not match the new bandwidth estimator.

Review URL: http://webrtc-codereview.appspot.com/270003
Patch from pwestin1 <pwestin@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@925 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -442,6 +442,10 @@ WebRtc_Word32 ModuleRtpRtcpImpl::Process()
     {
         WebRtc_UWord16 RTT = 0;
         _rtcpReceiver.RTT(_rtpReceiver.SSRC(), &RTT, NULL, NULL, NULL);
+        if (TMMBR())
+        {
+            _rtcpSender.CalculateNewTargetBitrate(RTT);
+        }
         _rtcpSender.SendRTCP(kRtcpReport, 0, 0, RTT);
     }
     if(_rtpSender.RTPKeepalive())
@@ -2489,7 +2493,7 @@ RateControlRegion ModuleRtpRtcpImpl::OnOverUseStateUpdate(
     bool firstOverUse = false;
     RateControlRegion region = _rtcpSender.UpdateOverUseState(rateControlInput,
                                                               firstOverUse);
-    if (firstOverUse && _rtcpSender.Status() == kRtcpNonCompound) {
+    if (firstOverUse) {
         // Send TMMBR immediately
         WebRtc_UWord16 RTT = 0;
         _rtcpReceiver.RTT(_rtpReceiver.SSRC(), &RTT, NULL, NULL, NULL);
@@ -2625,15 +2629,13 @@ void ModuleRtpRtcpImpl::OnReceivedReferencePictureSelectionIndication(
 
 void ModuleRtpRtcpImpl::OnReceivedBandwidthEstimateUpdate(
      const WebRtc_UWord16 bwEstimateKbit) {
+
     // We received a TMMBR
-    if (_defaultModule) {
-        CriticalSectionScoped lock(_criticalSectionModulePtrs);
-        if (_defaultModule) {
-            // if we use a default module pass this info to the default module
-            _defaultModule->OnReceivedBandwidthEstimateUpdate(bwEstimateKbit);
+    const bool defaultInstance(_childModules.empty() ? false : true);
+    if (defaultInstance) {
+        ProcessDefaultModuleBandwidth();
         return;
     }
-    }
     if (_audio) {
         _rtcpReceiver.UpdateBandwidthEstimate(bwEstimateKbit);
     } else {
@@ -2644,17 +2646,27 @@ void ModuleRtpRtcpImpl::OnReceivedBandwidthEstimateUpdate(
                                                          &newBitrate,
                                                          &fractionLost,
                                                          &roundTripTime) == 0) {
-            // video callback
+            if (!_defaultModule) {
+                // No default module check if we should trigger OnNetworkChanged
+                // via video callback
                 _rtpReceiver.UpdateBandwidthManagement(newBitrate,
                                                        fractionLost,
                                                        roundTripTime);
-            const bool defaultInstance = !_childModules.empty();
-            if ((newBitrate > 0) && !defaultInstance) {
+            }
+            if (newBitrate > 0) {
                 // update bitrate
                 _rtpSender.SetTargetSendBitrate(newBitrate);
             }
         }
     }
+    if (_defaultModule) {
+        CriticalSectionScoped lock(_criticalSectionModulePtrs);
+        if (_defaultModule) {
+            // if we use a default module pass this info to the default module
+            _defaultModule->OnReceivedBandwidthEstimateUpdate(bwEstimateKbit);
+            return;
+        }
+    }
 }
 
 // bw estimation
@@ -2708,39 +2720,7 @@ void ModuleRtpRtcpImpl::OnPacketLossStatisticsUpdate(
                                                roundTripTime);
     } else {
         if (!_simulcast) {
-            WebRtc_UWord32 minBitrateBps = 0xffffffff;
-            WebRtc_UWord32 maxBitrateBps = 0;
-            {
-                // get min and max for the sending channels
-                CriticalSectionScoped lock(_criticalSectionModulePtrs);
-
-                std::list<ModuleRtpRtcpImpl*>::iterator it =
-                    _childModules.begin();
-                while (it != _childModules.end()) {
-                    // Get child RTP sender and ask for bitrate estimate
-                    ModuleRtpRtcpImpl* childModule = *it;
-                    if (childModule->Sending()) {
-                        RTPSender& childRtpSender = (*it)->_rtpSender;
-                        const WebRtc_UWord32 childEstimateBps =
-                            1000 * childRtpSender.TargetSendBitrateKbit();
-                        if (childEstimateBps < minBitrateBps) {
-                            minBitrateBps = childEstimateBps;
-                        }
-                        if (childEstimateBps > maxBitrateBps) {
-                            maxBitrateBps = childEstimateBps;
-                        }
-                    }
-                    it++;
-                }
-            }  // end critsect
-            _bandwidthManagement.SetSendBitrate(minBitrateBps, 0, 0);
-            // Update default module bitrate. Don't care about min max.
-            // Check if we should trigger OnNetworkChanged via video callback
-            // TODO(pwestin) for this to work better fractionLost shoudl be the
-            // weighted average loss from all childs
-            _rtpReceiver.UpdateBandwidthManagement(minBitrateBps,
-                                                   fractionLost,
-                                                   roundTripTime);
+            ProcessDefaultModuleBandwidth();
         } else {
             // default and simulcast
             WebRtc_UWord32 newBitrate = 0;
@@ -2799,6 +2779,63 @@ void ModuleRtpRtcpImpl::OnPacketLossStatisticsUpdate(
     }
 }
 
+void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth() {
+    WebRtc_UWord32 minBitrateBps = 0xffffffff;
+    WebRtc_UWord32 maxBitrateBps = 0;
+    WebRtc_UWord32 count = 0;
+    WebRtc_UWord32 fractionLostAcc = 0;
+    WebRtc_UWord16 maxRoundTripTime = 0;
+    {
+        // get min and max for the sending channels
+        CriticalSectionScoped lock(_criticalSectionModulePtrs);
+
+        std::list<ModuleRtpRtcpImpl*>::iterator it =
+            _childModules.begin();
+        while (it != _childModules.end()) {
+            // Get child RTP sender and ask for bitrate estimate
+            ModuleRtpRtcpImpl* childModule = *it;
+            if (childModule->Sending()) {
+                count++;
+                RTPSender& childRtpSender = (*it)->_rtpSender;
+                const WebRtc_UWord32 childEstimateBps =
+                    1000 * childRtpSender.TargetSendBitrateKbit();
+                if (childEstimateBps < minBitrateBps) {
+                    minBitrateBps = childEstimateBps;
+                }
+                if (childEstimateBps > maxBitrateBps) {
+                    maxBitrateBps = childEstimateBps;
+                }
+                WebRtc_UWord16 RTT = 0;
+                WebRtc_UWord8 fractionLost = 0;
+                RTPReceiver& childRtpReceiver = (*it)->_rtpReceiver;
+                RTCPReceiver& childRtcpReceiver = (*it)->_rtcpReceiver;
+                childRtpReceiver.Statistics(&fractionLost,
+                                            NULL,
+                                            NULL,
+                                            NULL,
+                                            NULL,
+                                            false);
+                fractionLostAcc += fractionLost;
+                childRtcpReceiver.RTT(childRtpReceiver.SSRC(),
+                                      &RTT,
+                                      NULL,
+                                      NULL,
+                                      NULL);
+                maxRoundTripTime =
+                    (RTT > maxRoundTripTime) ? RTT : maxRoundTripTime;
+            }
+            it++;
+        }
+    }  // end critsect
+    _bandwidthManagement.SetSendBitrate(minBitrateBps, 0, 0);
+    // Update default module bitrate. Don't care about min max.
+    // Check if we should trigger OnNetworkChanged via video callback
+    WebRtc_UWord8 fractionLostAvg = WebRtc_UWord8(fractionLostAcc / count);
+    _rtpReceiver.UpdateBandwidthManagement(minBitrateBps,
+                                           fractionLostAvg ,
+                                           maxRoundTripTime);
+}
+
 void ModuleRtpRtcpImpl::OnRequestSendReport() {
     _rtcpSender.SendRTCP(kRtcpSr);
 }
@@ -2864,56 +2901,13 @@ WebRtc_Word32 ModuleRtpRtcpImpl::UpdateTMMBR() {
     WebRtc_Word32 newBitrates = 0;
     WebRtc_UWord32 minBitrateKbit = 0;
     WebRtc_UWord32 maxBitrateKbit = 0;
-
-    if (_defaultModule) {
-        CriticalSectionScoped lock(_criticalSectionModulePtrs);
-
-        // no callbacks allowed inside here
-        if (_defaultModule) {
-            // let the default module do the update
-            return _defaultModule->UpdateTMMBR();
-        }
-    }
-
     WebRtc_UWord32 accNumCandidates = 0;
 
-    // Find candidate set
     if (!_childModules.empty()) {
-        CriticalSectionScoped lock(_criticalSectionModulePtrsFeedback);
-
-        // this module is the default module
-        // loop over all modules using the default codec
-        WebRtc_UWord32 size = 0;
-        std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
-        while (it != _childModules.end()) {
-            ModuleRtpRtcpImpl* module = *it;
-            WebRtc_Word32 tmpSize = module->TMMBRReceived(0, 0, NULL);
-            if (tmpSize > 0) {
-                size += tmpSize;
-            }
-            it++;
-        }
-        TMMBRSet* candidateSet = VerifyAndAllocateCandidateSet(size);
-        if (candidateSet == NULL) {
+        // Default module should not handle this
         return -1;
     }
 
-        it = _childModules.begin();
-        while (it != _childModules.end()) {
-            ModuleRtpRtcpImpl* module = *it;
-            if (size > accNumCandidates && module) {
-                WebRtc_Word32 accSize = module->TMMBRReceived(size,
-                                                              accNumCandidates,
-                                                              candidateSet);
-                if (accSize > 0) {
-                    accNumCandidates = accSize;
-                }
-            }
-            it++;
-        }
-    } else {
-        // this module don't use the default module and is not the
-        // default module
     WebRtc_Word32 size = _rtcpReceiver.TMMBRReceived(0, 0, NULL);
     if (size > 0) {
         TMMBRSet* candidateSet = VerifyAndAllocateCandidateSet(size);
@@ -2925,7 +2919,6 @@ WebRtc_Word32 ModuleRtpRtcpImpl::UpdateTMMBR() {
         // candidate set empty
         VerifyAndAllocateCandidateSet(0);  // resets candidate set
     }
-    }
     // Find bounding set
     TMMBRSet* boundingSet = NULL;
     numBoundingSet = FindTMMBRBoundingSet(boundingSet);
@@ -2936,33 +2929,17 @@ WebRtc_Word32 ModuleRtpRtcpImpl::UpdateTMMBR() {
                      "Failed to find TMMBR bounding set.");
         return -1;
     }
-
     // Set bounding set
     // Inform remote clients about the new bandwidth
-    if (_childModules.empty()) {
     // inform the remote client
     _rtcpSender.SetTMMBN(boundingSet,
                         _rtpSender.MaxConfiguredBitrateVideo()/1000);
     // might trigger a TMMBN
-    } else {
-        // inform child modules using the default codec
-        CriticalSectionScoped lock(_criticalSectionModulePtrsFeedback);
-        std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
-        while (it != _childModules.end()) {
-            ModuleRtpRtcpImpl* module = *it;
-            if (module) {
-                module->SetTMMBN(boundingSet,
-                                 _rtpSender.MaxConfiguredBitrateVideo()/1000);
-            }
-            it++;
-        }
-    }
     if (numBoundingSet == 0) {
         // owner of max bitrate request has timed out
         // empty bounding set has been sent
         return 0;
     }
-
     // Get net bitrate from bounding set depending on sent packet rate
     newBitrates = CalcMinMaxBitRate(_rtpSender.PacketRate(),
                                     (WebRtc_UWord32)numBoundingSet,

src/modules/rtp_rtcp/source/rtp_rtcp_impl.h

@@ -532,6 +532,7 @@ protected:
     RTCPReceiver              _rtcpReceiver;
 private:
     void SendKeyFrame();
+    void ProcessDefaultModuleBandwidth();
 
     WebRtc_Word32             _id;
     const bool                _audio;