Fix sequence-number replay race for padding.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index c831b5132a7c9befdd4187a2e008b3b225ef99e9..97469ca35ecf27defdfc971376e4d6ff6399b23d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -556,6 +556,8 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
 size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send) {
   {
     rtc::CritScope lock(&send_critsect_);
+    if (!sending_media_)
+      return 0;
     if ((rtx_ & kRtxRedundantPayloads) == 0)
       return 0;
   }
@@ -618,6 +620,8 @@ size_t RTPSender::SendPadData(size_t bytes,
     bool over_rtx;
     {
       rtc::CritScope lock(&send_critsect_);
+      if (!sending_media_)
+        return bytes_sent;
       if (!timestamp_provided) {
         timestamp = timestamp_;
         capture_time_ms = capture_time_ms_;
@@ -1011,11 +1015,6 @@ bool RTPSender::IsFecPacket(const uint8_t* buffer,
 size_t RTPSender::TimeToSendPadding(size_t bytes) {
   if (audio_configured_ || bytes == 0)
     return 0;
-  {

[pbos-webrtc, 2016/02/19 15:11:29]

This check is now done in TrySendRedundantPayloads as well as SendPadData, so no
need to acquire the lock once more, and we have to check inside both those
functions as well.

-    rtc::CritScope lock(&send_critsect_);
-    if (!sending_media_)
-      return 0;
-  }
   size_t bytes_sent = TrySendRedundantPayloads(bytes);
   if (bytes_sent < bytes)
     bytes_sent += SendPadData(bytes - bytes_sent, false, 0, 0);