From 7f1b0ae888bad8daf56a3ca965a1a3174ab2a3b2 Mon Sep 17 00:00:00 2001
From: "pbos@webrtc.org" <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Date: Wed, 5 Jun 2013 11:39:18 +0000
Subject: [PATCH] Fix init list for VideoSendStream::Config::Rtp.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1616004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4183 4adac7df-926f-26a2-2b94-8c16560cd09d
---
 webrtc/video_engine/new_include/video_send_stream.h | 7 +------
 1 file changed, 1 insertion(+), 6 deletions(-)

diff --git a/webrtc/video_engine/new_include/video_send_stream.h b/webrtc/video_engine/new_include/video_send_stream.h
index f957323b3a..75edad3045 100644
--- a/webrtc/video_engine/new_include/video_send_stream.h
+++ b/webrtc/video_engine/new_include/video_send_stream.h
@@ -86,12 +86,7 @@ class VideoSendStream {
     VideoCodec codec;
 
     struct Rtp {
-      Rtp()
-          : mode(kRtcpReducedSize),
-            max_packet_size(0),
-            nack(NULL),
-            fec(NULL),
-            rtx(NULL) {}
+      Rtp() : mode(kRtcpReducedSize), max_packet_size(0) {}
       RtcpMode mode;
 
       std::vector<uint32_t> ssrcs;