From ef90c3227ebd4008bbcfabd17a9f422965f11a25 Mon Sep 17 00:00:00 2001
From: "phoglund@webrtc.org"
 <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Date: Mon, 26 Nov 2012 16:30:40 +0000
Subject: [PATCH] Will now correctly identify the first-ever received packet as
 the first packet in its frame.

We used to flag the _second_ packet in the first frame as the first. Subsequent frames worked as intended.

BUG=1103
TEST=vie_auto_test --automated, rtp_rtcp_unittests

Review URL: https://webrtc-codereview.appspot.com/964020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3164 4adac7df-926f-26a2-2b94-8c16560cd09d
---
 .../rtp_rtcp/source/mock/mock_rtp_receiver_video.h    |  2 ++
 webrtc/modules/rtp_rtcp/source/rtp_receiver.cc        |  9 +++++++--
 webrtc/modules/rtp_rtcp/source/rtp_receiver.h         |  2 ++
 webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc  | 11 +++++++----
 webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h   |  1 +
 5 files changed, 19 insertions(+), 6 deletions(-)

diff --git a/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h b/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h
index 37f08eba3f..3e3fce9f5b 100644
--- a/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h
+++ b/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h
@@ -40,6 +40,8 @@ class MockRTPReceiverVideo : public RTPReceiverVideo {
            const WebRtc_UWord32 rtpTimeStamp));
   MOCK_CONST_METHOD0(REDPayloadType,
       WebRtc_Word8());
+  MOCK_CONST_METHOD0(HaveNotReceivedPackets,
+        bool());
 };
 
 }  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
index b4a7e52891..33c66234a8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
@@ -157,6 +157,11 @@ RTPReceiver::SetPacketTimeout(const WebRtc_UWord32 timeoutMS)
     return 0;
 }
 
+bool RTPReceiver::HaveNotReceivedPackets() const
+{
+    return _lastReceiveTime == 0;
+}
+
 void RTPReceiver::PacketTimeout()
 {
     bool packetTimeOut = false;
@@ -168,7 +173,7 @@ void RTPReceiver::PacketTimeout()
             return;
         }
 
-        if(_lastReceiveTime == 0)
+        if (HaveNotReceivedPackets())
         {
             // not active
             return;
@@ -937,7 +942,7 @@ bool RTPReceiver::RetransmitOfOldPacket(
 
     // Min maxDelayMs is 1.
     if (maxDelayMs == 0) {
-      maxDelayMs = 1; 
+      maxDelayMs = 1;
     }
   } else {
     maxDelayMs = (minRTT / 3) + 1;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver.h
index 3893727eed..b07e8f4b6e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver.h
@@ -169,6 +169,8 @@ protected:
 
     virtual WebRtc_Word8 REDPayloadType() const;
 
+    bool HaveNotReceivedPackets() const;
+
 private:
     // Is RED configured with payload type payloadType
     bool REDPayloadType(const WebRtc_Word8 payloadType) const;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
index ad8c3377ce..ebaef9c940 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -321,14 +321,17 @@ WebRtc_Word32 RTPReceiverVideo::ReceiveGenericCodec(
     const WebRtc_UWord16 payloadDataLength) {
   rtpHeader->frameType = kVideoFrameKey;
 
-  if(((SequenceNumber() + 1) == rtpHeader->header.sequenceNumber) &&
-      (TimeStamp() != rtpHeader->header.timestamp)) {
+  bool isFirstPacketInFrame =
+      (SequenceNumber() + 1) == rtpHeader->header.sequenceNumber &&
+      TimeStamp() != rtpHeader->header.timestamp;
+
+  if (isFirstPacketInFrame || HaveNotReceivedPackets()) {
     rtpHeader->type.Video.isFirstPacket = true;
   }
   _criticalSectionReceiverVideo->Leave();
 
-  if(CallbackOfReceivedPayloadData(payloadData, payloadDataLength,
-                                   rtpHeader) != 0) {
+  if (CallbackOfReceivedPayloadData(payloadData, payloadDataLength,
+                                    rtpHeader) != 0) {
     return -1;
   }
   return 0;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
index ddfa69e75a..454ba810ac 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -71,6 +71,7 @@ class RTPReceiverVideo {
       const WebRtc_UWord32 rtpTimeStamp) const  = 0;
 
   virtual WebRtc_Word8 REDPayloadType() const = 0;
+  virtual bool HaveNotReceivedPackets() const = 0;
 
   WebRtc_Word32 SetCodecType(const RtpVideoCodecTypes videoType,
                              WebRtcRTPHeader* rtpHeader) const;