webrtc::RtpPacket name freed for better RtpPacket

webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index 1eecd0bca175bcd24a4283e3771c52348ff6c849..32ba26f54b88a5e9eaa6d2cf4d2c785aa857405e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -29,11 +29,6 @@
 namespace webrtc {
 enum { REDForFECHeaderLength = 1 };
 
-struct RtpPacket {
-  uint16_t rtpHeaderLength;
-  ForwardErrorCorrection::Packet* pkt;
-};
-
 RTPSenderVideo::RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender)
     : _rtpSender(*rtpSender),
       crit_(CriticalSectionWrapper::CreateCriticalSection()),