commit 1457b4737a82cc3624daf3a997f6257d13a3f974
author asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon May 26 13:06:04 2014 +0000
committer asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon May 26 13:06:04 2014 +0000
tree c1470ce01d0f72f5bb0085d07e113bffc8c4828b
parent 727ff698298677ea221a176698cb47e8648da621

First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class.
Receive stats is reset if the payload type changes. Update stats after a possible reset.

R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6247 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/video/call_tests.cc

diff --git a/webrtc/video/call_tests.cc b/webrtc/video/call_tests.cc
index b305246..7b78af3 100644
--- a/webrtc/video/call_tests.cc
+++ b/webrtc/video/call_tests.cc
@@ -1557,4 +1557,52 @@
   TestXrReceiverReferenceTimeReport(false);
 }
 
+TEST_F(CallTest, TestReceivedRtpPacketStats) {
+  static const size_t kNumRtpPacketsToSend = 5;
+  class ReceivedRtpStatsObserver : public test::RtpRtcpObserver {
+   public:
+    ReceivedRtpStatsObserver()
+        : test::RtpRtcpObserver(kDefaultTimeoutMs),
+          receive_stream_(NULL),
+          sent_rtp_(0) {}
+
+    void SetReceiveStream(VideoReceiveStream* stream) {
+      receive_stream_ = stream;
+    }
+
+   private:
+    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+      if (sent_rtp_ >= kNumRtpPacketsToSend) {
+        VideoReceiveStream::Stats stats = receive_stream_->GetStats();
+        if (kNumRtpPacketsToSend == stats.rtp_stats.packets) {
+          observation_complete_->Set();
+        }
+        return DROP_PACKET;
+      }
+      ++sent_rtp_;
+      return SEND_PACKET;
+    }
+
+    VideoReceiveStream* receive_stream_;
+    uint32_t sent_rtp_;
+  } observer;
+
+  CreateCalls(Call::Config(observer.SendTransport()),
+              Call::Config(observer.ReceiveTransport()));
+  observer.SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver());
+
+  CreateTestConfigs();
+  CreateStreams();
+  observer.SetReceiveStream(receive_stream_);
+  CreateFrameGenerator();
+  StartSending();
+
+  EXPECT_EQ(kEventSignaled, observer.Wait())
+      << "Timed out while verifying number of received RTP packets.";
+
+  StopSending();
+  observer.StopSending();
+  DestroyStreams();
+}
+
 }  // namespace webrtc

webrtc/video_engine/vie_receiver.cc

diff --git a/webrtc/video_engine/vie_receiver.cc b/webrtc/video_engine/vie_receiver.cc
index c8d7508..c588d78 100644
--- a/webrtc/video_engine/vie_receiver.cc
+++ b/webrtc/video_engine/vie_receiver.cc
@@ -238,12 +238,16 @@
   header.payload_type_frequency = kVideoPayloadTypeFrequency;
 
   bool in_order = IsPacketInOrder(header);
-  rtp_receive_statistics_->IncomingPacket(
-      header, rtp_packet_length, IsPacketRetransmitted(header, in_order));
   rtp_payload_registry_->SetIncomingPayloadType(header);
-  return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order)
+  int ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order)
       ? 0
       : -1;
+  // Update receive statistics after ReceivePacket.
+  // Receive statistics will be reset if the payload type changes (make sure
+  // that the first packet is included in the stats).
+  rtp_receive_statistics_->IncomingPacket(
+      header, rtp_packet_length, IsPacketRetransmitted(header, in_order));
+  return ret;
 }
 
 bool ViEReceiver::ReceivePacket(const uint8_t* packet,