Fix sequence-number replay race for padding.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 538 matching lines @@
   if (frame_count_observer_) {
     frame_count_observer_->FrameCountUpdated(frame_counts_, ssrc);
   }
 
   return ret_val;
 }
 
 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;
   }
 
   uint8_t buffer[IP_PACKET_SIZE];
   int bytes_left = static_cast<int>(bytes_to_send);
   while (bytes_left > 0) {
     size_t length = bytes_left;
     int64_t capture_time_ms;
     if (!packet_history_.GetBestFittingPacket(buffer, &length,
@@ skipping 42 matching lines @@
   for (; bytes > 0; bytes -= padding_bytes_in_packet) {
     if (bytes < padding_bytes_in_packet)
       bytes = padding_bytes_in_packet;
 
     uint32_t ssrc;
     uint16_t sequence_number;
     int payload_type;
     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_;
       }
       if (rtx_ == kRtxOff) {
         // Without RTX we can't send padding in the middle of frames.
         if (!last_packet_marker_bit_)
           return 0;
         ssrc = ssrc_;
         sequence_number = sequence_number_;
@@ skipping 373 matching lines @@
   uint8_t pt_fec;
   video_->GenericFECStatus(&fec_enabled, &pt_red, &pt_fec);
   return fec_enabled &&
       header.payloadType == pt_red &&
       buffer[header.headerLength] == pt_fec;
 }
 
 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);
   return bytes_sent;
 }
 
 // TODO(pwestin): send in the RtpHeaderParser to avoid parsing it again.
 int32_t RTPSender::SendToNetwork(uint8_t* buffer,
                                  size_t payload_length,
                                  size_t rtp_header_length,
@@ skipping 883 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc