Fixing a segfault that can occur when changing the track of an RtpSender.

webrtc/api/rtpsender.cc

Index: webrtc/api/rtpsender.cc
diff --git a/webrtc/api/rtpsender.cc b/webrtc/api/rtpsender.cc
index 58cb18c6cd2734e5bc37df470c5a3ba121abd401..360b6868fd377b99e1f64fdeb21bead5d7b490df 100644
--- a/webrtc/api/rtpsender.cc
+++ b/webrtc/api/rtpsender.cc
@@ -122,6 +122,9 @@ bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
 
   // Attach to new track.
   bool prev_can_send_track = can_send_track();
+  // Keep a reference to the old track to keep it alive until we call
+  // SetAudioSend.
+  rtc::scoped_refptr<AudioTrackInterface> old_track = track_;
   track_ = audio_track;
   if (track_) {
     cached_track_enabled_ = track_->enabled();
@@ -276,6 +279,9 @@ bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
 
   // Attach to new track.
   bool prev_can_send_track = can_send_track();
+  // Keep a reference to the old track to keep it alive until we call
+  // SetSource.
+  rtc::scoped_refptr<VideoTrackInterface> old_track = track_;
   track_ = video_track;
   if (track_) {
     cached_track_enabled_ = track_->enabled();