From ba853c9970588fc1451eaf4489c8309809b414df Mon Sep 17 00:00:00 2001
From: "mflodman@webrtc.org"
 <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Date: Fri, 10 Aug 2012 14:30:53 +0000
Subject: [PATCH] Fixes for RTP extension time offset.

Review URL: https://webrtc-codereview.appspot.com/720008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2592 4adac7df-926f-26a2-2b94-8c16560cd09d
---
 src/modules/rtp_rtcp/source/rtp_sender.cc  | 9 +++++----
 src/modules/rtp_rtcp/source/rtp_sender.h   | 2 +-
 src/modules/rtp_rtcp/source/rtp_utility.cc | 4 ++--
 3 files changed, 8 insertions(+), 7 deletions(-)

diff --git a/src/modules/rtp_rtcp/source/rtp_sender.cc b/src/modules/rtp_rtcp/source/rtp_sender.cc
index 5bfb76aa1..8652ff812 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender.cc
@@ -917,7 +917,7 @@ RTPSender::SendToNetwork(WebRtc_UWord8* buffer,
   if (capture_time_ms >= 0) {
     ModuleRTPUtility::RTPHeaderParser rtpParser(buffer, length);
     WebRtcRTPHeader rtp_header;
-    rtpParser.Parse(rtp_header);
+    rtpParser.Parse(rtp_header, &_rtpHeaderExtensionMap);
     int64_t time_now = _clock.GetTimeInMS();
     UpdateTransmissionTimeOffset(buffer, length, rtp_header,
                                  time_now - capture_time_ms);
@@ -1172,7 +1172,7 @@ void RTPSender::UpdateTransmissionTimeOffset(
     WebRtc_UWord8* rtp_packet,
     const WebRtc_UWord16 rtp_packet_length,
     const WebRtcRTPHeader& rtp_header,
-    const WebRtc_Word64 time_ms) const {
+    const WebRtc_Word64 time_diff_ms) const {
   CriticalSectionScoped cs(_sendCritsect);
 
   // Get length until start of transmission block.
@@ -1186,7 +1186,8 @@ void RTPSender::UpdateTransmissionTimeOffset(
   }
 
   int block_pos = 12 + rtp_header.header.numCSRCs + transmission_block_pos;
-  if ((rtp_packet_length < block_pos + 4)) {
+  if (rtp_packet_length < block_pos + 4 ||
+      rtp_header.header.headerLength < block_pos + 4) {
     WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
         "Failed to update transmission time offset, invalid length.");
     return;
@@ -1219,7 +1220,7 @@ void RTPSender::UpdateTransmissionTimeOffset(
 
   // Update transmission offset field.
   ModuleRTPUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1,
-                                          time_ms * 90);  // RTP timestamp
+                                          time_diff_ms * 90);  // RTP timestamp.
 }
 
 WebRtc_Word32
diff --git a/src/modules/rtp_rtcp/source/rtp_sender.h b/src/modules/rtp_rtcp/source/rtp_sender.h
index eadfca33b..4c843a710 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender.h
+++ b/src/modules/rtp_rtcp/source/rtp_sender.h
@@ -166,7 +166,7 @@ public:
     void UpdateTransmissionTimeOffset(WebRtc_UWord8* rtp_packet,
                                       const WebRtc_UWord16 rtp_packet_length,
                                       const WebRtcRTPHeader& rtp_header,
-                                      const WebRtc_Word64 time_ms) const;
+                                      const WebRtc_Word64 time_diff_ms) const;
 
     void SetTransmissionSmoothingStatus(const bool enable);
 
diff --git a/src/modules/rtp_rtcp/source/rtp_utility.cc b/src/modules/rtp_rtcp/source/rtp_utility.cc
index 5123278b3..67d53afd5 100644
--- a/src/modules/rtp_rtcp/source/rtp_utility.cc
+++ b/src/modules/rtp_rtcp/source/rtp_utility.cc
@@ -608,7 +608,7 @@ void RTPHeaderParser::ParseOneByteExtensionHeader(
     const WebRtc_UWord8* ptrRTPDataExtensionEnd,
     const WebRtc_UWord8* ptr) const {
   if (!ptrExtensionMap) {
-    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1, "No extension map.");
+    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "No extension map.");
     return;
   }
 
@@ -676,7 +676,7 @@ void RTPHeaderParser::ParseOneByteExtensionHeader(
         break;
       }
       default: {
-        WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+        WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
                      "Extension type not implemented.");
         return;
       }