1 files changed, 15 insertions, 13 deletions
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index 7b36f7cced8..5d8ae166565 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -14,12 +14,13 @@
 #include <stdlib.h>
 #include <string.h>
 
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
@@ -30,11 +31,9 @@ struct RtpPacket {
   ForwardErrorCorrection::Packet* pkt;
 };
 
-RTPSenderVideo::RTPSenderVideo(const int32_t id,
-                               Clock* clock,
+RTPSenderVideo::RTPSenderVideo(Clock* clock,
                                RTPSenderInterface* rtpSender)
-    : _id(id),
-      _rtpSender(*rtpSender),
+    : _rtpSender(*rtpSender),
       _sendVideoCritsect(CriticalSectionWrapper::CreateCriticalSection()),
       _videoType(kRtpVideoGeneric),
       _videoCodecInformation(NULL),
@@ -42,7 +41,7 @@ RTPSenderVideo::RTPSenderVideo(const int32_t id,
       _retransmissionSettings(kRetransmitBaseLayer),
 
       // Generic FEC
-      _fec(id),
+      _fec(),
       _fecEnabled(false),
       _payloadTypeRED(-1),
       _payloadTypeFEC(-1),
@@ -253,8 +252,13 @@ RTPSenderVideo::FECPacketOverhead() const
 {
     if (_fecEnabled)
     {
-        return ForwardErrorCorrection::PacketOverhead() +
-            REDForFECHeaderLength;
+      // Overhead is FEC headers plus RED for FEC header plus anything in RTP
+      // header beyond the 12 bytes base header (CSRC list, extensions...)
+      // This reason for the header extensions to be included here is that
+      // from an FEC viewpoint, they are part of the payload to be protected.
+      // (The base RTP header is already protected by the FEC header.)
+      return ForwardErrorCorrection::PacketOverhead() + REDForFECHeaderLength +
+             (_rtpSender.RTPHeaderLength() - kRtpHeaderSize);
     }
     return 0;
 }
@@ -323,8 +327,6 @@ RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
     {
         return retVal;
     }
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, "%s(timestamp:%u)",
-                 __FUNCTION__, captureTimeStamp);
     return 0;
 }
 
@@ -470,9 +472,9 @@ RTPSenderVideo::SendVP8(const FrameType frameType,
                                   rtpHeaderLength, captureTimeStamp,
                                   capture_time_ms, storage, protect))
         {
-          WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
-                       "RTPSenderVideo::SendVP8 failed to send packet number"
-                       " %d", _rtpSender.SequenceNumber());
+          LOG(LS_WARNING)
+              << "RTPSenderVideo::SendVP8 failed to send packet number "
+              << _rtpSender.SequenceNumber();
         }
     }
     TRACE_EVENT_ASYNC_END1("webrtc", "Video", capture_time_ms,