Update the remote bitrate estimator before passing the packet to the RTP module.

This solves the problem of reconstructed packets biasing the bandwidth estimate.

TEST=vie_auto_test --automated, trybots
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1594005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4171 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h

@@ -55,7 +55,7 @@ class RemoteBitrateEstimator : public CallStatsObserver, public Module {
   // the WebRtcRTPHeader to be initialized.
   virtual void IncomingPacket(int64_t arrival_time_ms,
                               int payload_size,
-                              const WebRtcRTPHeader& header) = 0;
+                              const RTPHeader& header) = 0;
 
   // Removes all data for |ssrc|.
   virtual void RemoveStream(unsigned int ssrc) = 0;

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_multi_stream.cc

@@ -44,7 +44,7 @@ class RemoteBitrateEstimatorMultiStream : public RemoteBitrateEstimator {
   // Note that |payload_size| is the packet size excluding headers.
   virtual void IncomingPacket(int64_t arrival_time_ms,
                               int payload_size,
-                              const WebRtcRTPHeader& header);
+                              const RTPHeader& header);
 
   // Triggers a new estimate calculation.
   // Implements the Module interface.
@@ -139,10 +139,10 @@ void RemoteBitrateEstimatorMultiStream::IncomingRtcp(unsigned int ssrc,
 void RemoteBitrateEstimatorMultiStream::IncomingPacket(
     int64_t arrival_time_ms,
     int payload_size,
-    const WebRtcRTPHeader& header) {
+    const RTPHeader& header) {
-  uint32_t ssrc = header.header.ssrc;
+  uint32_t ssrc = header.ssrc;
-  uint32_t rtp_timestamp = header.header.timestamp +
+  uint32_t rtp_timestamp = header.timestamp +
-      header.header.extension.transmissionTimeOffset;
+      header.extension.transmissionTimeOffset;
   CriticalSectionScoped cs(crit_sect_.get());
   incoming_bitrate_.Update(payload_size, arrival_time_ms);
   // Add this stream to the map of streams if it doesn't already exist.

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc

@@ -37,7 +37,7 @@ class RemoteBitrateEstimatorSingleStream : public RemoteBitrateEstimator {
   // packet size excluding headers.
   virtual void IncomingPacket(int64_t arrival_time_ms,
                               int payload_size,
-                              const WebRtcRTPHeader& header);
+                              const RTPHeader& header);
 
   // Triggers a new estimate calculation.
   // Implements the Module interface.
@@ -86,10 +86,10 @@ RemoteBitrateEstimatorSingleStream::RemoteBitrateEstimatorSingleStream(
 void RemoteBitrateEstimatorSingleStream::IncomingPacket(
     int64_t arrival_time_ms,
     int payload_size,
-    const WebRtcRTPHeader& header) {
+    const RTPHeader& header) {
-  uint32_t ssrc = header.header.ssrc;
+  uint32_t ssrc = header.ssrc;
-  uint32_t rtp_timestamp = header.header.timestamp +
+  uint32_t rtp_timestamp = header.timestamp +
-      header.header.extension.transmissionTimeOffset;
+      header.extension.transmissionTimeOffset;
   CriticalSectionScoped cs(crit_sect_.get());
   SsrcOveruseDetectorMap::iterator it = overuse_detectors_.find(ssrc);
   if (it == overuse_detectors_.end()) {

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc

@@ -223,11 +223,11 @@ void RemoteBitrateEstimatorTest::IncomingPacket(uint32_t ssrc,
                                                 int64_t arrival_time,
                                                 uint32_t rtp_timestamp,
                                                 uint32_t absolute_send_time) {
-  WebRtcRTPHeader header;
+  RTPHeader header;
   memset(&header, 0, sizeof(header));
-  header.header.ssrc = ssrc;
+  header.ssrc = ssrc;
-  header.header.timestamp = rtp_timestamp;
+  header.timestamp = rtp_timestamp;
-  header.header.extension.absoluteSendTime = absolute_send_time;
+  header.extension.absoluteSendTime = absolute_send_time;
   bitrate_estimator_->IncomingPacket(arrival_time, payload_size, header);
 }
 

webrtc/video_engine/vie_channel_group.cc

@@ -68,7 +68,7 @@ class WrappingBitrateEstimator : public RemoteBitrateEstimator {
 
   virtual void IncomingPacket(int64_t arrival_time_ms,
                               int payload_size,
-                              const WebRtcRTPHeader& header) {
+                              const RTPHeader& header) {
     CriticalSectionScoped cs(crit_sect_.get());
     rbe_->IncomingPacket(arrival_time_ms, payload_size, header);
   }

webrtc/video_engine/vie_receiver.cc

@@ -133,12 +133,6 @@ int32_t ViEReceiver::OnReceivedPayloadData(
   if (rtp_header == NULL) {
     return 0;
   }
-
-  // TODO(holmer): Make sure packets reconstructed using FEC are not passed to
-  // the bandwidth estimator.
-  const int packet_size = payload_size + rtp_header->header.paddingLength;
-  remote_bitrate_estimator_->IncomingPacket(TickTime::MillisecondTimestamp(),
-                                            packet_size, *rtp_header);
   if (vcm_->IncomingPacket(payload_data, payload_size, *rtp_header) != 0) {
     // Check this...
     return -1;
@@ -197,6 +191,9 @@ int ViEReceiver::InsertRTPPacket(const int8_t* rtp_packet,
                  "IncomingPacket invalid RTP header");
     return -1;
   }
+  const int payload_size = received_packet_length - header.headerLength;
+  remote_bitrate_estimator_->IncomingPacket(TickTime::MillisecondTimestamp(),
+                                            payload_size, header);
   assert(rtp_rtcp_);  // Should be set by owner at construction time.
   return rtp_rtcp_->IncomingRtpPacket(received_packet, received_packet_length,
                                       header);