Fixing segfault caused by TurnServer.

webrtc/p2p/base/turnserver.cc

 /*
  *  Copyright 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 507 matching lines @@
   if (it != allocations_.end()) {
     it->second.release();
     allocations_.erase(it);
   }
 }
 
 void TurnServer::DestroyInternalSocket(rtc::AsyncPacketSocket* socket) {
   InternalSocketMap::iterator iter = server_sockets_.find(socket);
   if (iter != server_sockets_.end()) {
     rtc::AsyncPacketSocket* socket = iter->first;
+    server_sockets_.erase(iter);
     // We must destroy the socket async to avoid invalidating the sigslot
-    // callback list iterator inside a sigslot callback.
-    rtc::Thread::Current()->Dispose(socket);
-    server_sockets_.erase(iter);
+    // callback list iterator inside a sigslot callback. (In other words,
+    // deleting an object from within a callback from that object).
+    pending_delete_sockets_.push_back(
+        std::unique_ptr<rtc::AsyncPacketSocket>(socket));
+    invoker_.AsyncInvoke<void>(RTC_FROM_HERE, rtc::Thread::Current(),
+                               [this] { pending_delete_sockets_.clear(); });

[pthatcher1, 2016/08/23 00:36:55]

I think the style guide said not to store lambdas.

[Taylor Brandstetter, 2016/08/23 01:11:20]

The Chromium style guide says to never store *capturing* lambdas because
non-capturing lambdas used with base::Bind provide added protection:

invoker_.AsyncInvoke<void>(RTC_FROM_HERE, rtc::Thread::Current(),
  base::Bind([] (TurnServer* self) {
    self->pending_delete_sockets_.clear();
  }, this));

But we don't have Chromium's base::Bind. So what would you do here? Is a lambda
basically out of the question, and we have add a 1 line method instead?

[pthatcher1, 2016/08/23 01:15:32]

Ah, thanks for correcting my poor memory.  
That's what we've been doing elsewhere.  We could probably be less strict for
code that doesn't go in Chromium (like this one).  But it's probably worth
checking with the rest of the team via webrtc-eng@.

[Taylor Brandstetter, 2016/08/23 23:31:12]

I'll just bite the bullet, it's only a few extra lines of code.

   }
 }
 
 TurnServerConnection::TurnServerConnection(const rtc::SocketAddress& src,
                                            ProtocolType proto,
                                            rtc::AsyncPacketSocket* socket)
     : src_(src),
       dst_(socket->GetRemoteAddress()),
       proto_(proto),
       socket_(socket) {
@@ skipping 411 matching lines @@
                        MSG_ALLOCATION_TIMEOUT);
 }
 
 void TurnServerAllocation::Channel::OnMessage(rtc::Message* msg) {
   ASSERT(msg->message_id == MSG_ALLOCATION_TIMEOUT);
   SignalDestroyed(this);
   delete this;
 }
 
 }  // namespace cricket