Add ScopedAllowIO in WebrtcTransport when shutting down WebRTC session.

remoting/protocol/webrtc_transport.cc

 // Copyright 2015 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "remoting/protocol/webrtc_transport.h"
 
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "base/base64.h"
 #include "base/callback_helpers.h"
 #include "base/command_line.h"
 #include "base/macros.h"
 #include "base/memory/ptr_util.h"
 #include "base/single_thread_task_runner.h"
 #include "base/strings/string_number_conversions.h"
 #include "base/strings/string_split.h"
 #include "base/strings/string_util.h"
 #include "base/task_runner_util.h"
+#include "base/threading/thread_restrictions.h"
 #include "base/threading/thread_task_runner_handle.h"
 #include "jingle/glue/thread_wrapper.h"
 #include "remoting/protocol/authenticator.h"
 #include "remoting/protocol/port_allocator_factory.h"
 #include "remoting/protocol/sdp_message.h"
 #include "remoting/protocol/stream_message_pipe_adapter.h"
 #include "remoting/protocol/transport_context.h"
 #include "remoting/protocol/webrtc_audio_module.h"
 #include "remoting/protocol/webrtc_dummy_video_encoder.h"
 #include "third_party/webrtc/api/test/fakeconstraints.h"
@@ skipping 156 matching lines @@
     rtc_config.bundle_policy =
         webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle;
     rtc_config.rtcp_mux_policy =
         webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire;
 
     rtc_config.media_config.video.periodic_alr_bandwidth_probing = true;
 
     peer_connection_ = peer_connection_factory_->CreatePeerConnection(
         rtc_config, &constraints, std::move(port_allocator), nullptr, this);
   }
-  virtual ~PeerConnectionWrapper() { peer_connection_->Close(); }
+  virtual ~PeerConnectionWrapper() {
+    // PeerConnection creates threads internally, which are stopped when the
+    // connection is closed. Thread.Stop() is a blocking operation.
+    // See crbug.com/660081.
+    base::ThreadRestrictions::ScopedAllowIO allow_io;
+    peer_connection_->Close();
+  }
 
   WebrtcAudioModule* audio_module() {
     return audio_module_.get();
   }
 
   webrtc::PeerConnectionInterface* peer_connection() {
     return peer_connection_.get();
   }
 
   webrtc::PeerConnectionFactoryInterface* peer_connection_factory() {
@@ skipping 487 matching lines @@
   // the stack and so it must be destroyed later.
   base::ThreadTaskRunnerHandle::Get()->DeleteSoon(
       FROM_HERE, peer_connection_wrapper_.release());
 
   if (error != OK)
     event_handler_->OnWebrtcTransportError(error);
 }
 
 }  // namespace protocol
 }  // namespace remoting