From 79b63206b99912d9a5f97a35b546409886a8fed2 Mon Sep 17 00:00:00 2001
From: "stefan@webrtc.org"
 <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Date: Wed, 4 Dec 2013 13:34:28 +0000
Subject: [PATCH] Fixes a crash in fullstack tests introduced with r5209.

TBR=mflodman@webrtc.org
BUG=1812

Review URL: https://webrtc-codereview.appspot.com/4689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5211 4adac7df-926f-26a2-2b94-8c16560cd09d
---
 webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc       | 6 ++----
 .../modules/rtp_rtcp/source/rtp_packet_history_unittest.cc | 5 ++---
 webrtc/modules/rtp_rtcp/source/rtp_sender.cc               | 7 ++++++-
 3 files changed, 10 insertions(+), 8 deletions(-)

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
index fc288b7411..14ca821f7e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -254,12 +254,10 @@ bool RTPPacketHistory::GetPacketAndSetSendTime(uint16_t sequence_number,
       ((now - stored_send_times_.at(index)) < min_elapsed_time_ms)) {
     WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, 
         "Skip getting packet %u, packet recently resent.", sequence_number);
-    *packet_length = 0;
-    return true;
+    return false;
   }
 
-  if (length == 0 ||
-      (retransmit && stored_types_.at(index) == kDontRetransmit)) {
+  if (retransmit && stored_types_.at(index) == kDontRetransmit) {
     // No bytes copied since this packet shouldn't be retransmitted or is
     // of zero size.
     return false;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
index d0bbf92220..1682b7c338 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
@@ -249,8 +249,7 @@ TEST_F(RtpPacketHistoryTest, MinResendTime) {
 
   // Time has not elapsed. Packet should be found, but no bytes copied.
   len = kMaxPacketLength;
-  EXPECT_TRUE(hist_->GetPacketAndSetSendTime(kSeqNum, 101, false, packet_, &len,
-                                             &time));
-  EXPECT_EQ(0, len);
+  EXPECT_FALSE(hist_->GetPacketAndSetSendTime(kSeqNum, 101, false, packet_,
+                                              &len, &time));
 }
 }  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 60be774742..0f08573efd 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -560,7 +560,12 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) {
 
   ModuleRTPUtility::RTPHeaderParser rtp_parser(data_buffer, length);
   RTPHeader header;
-  rtp_parser.Parse(header);
+  if (!rtp_parser.Parse(header)) {
+    assert(false);
+    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
+                 "Failed to parse RTP header of packet to be retransmitted.");
+    return -1;
+  }
   TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPSender::ReSendPacket",
                        "timestamp", header.timestamp,
                        "seqnum", header.sequenceNumber);