Add a Preference to allow WebRTC only bind to "any address" (all 0s)

content/renderer/media/webrtc/peer_connection_dependency_factory.cc

 // Copyright 2014 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 "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
 
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/strings/utf_string_conversions.h"
 #include "base/synchronization/waitable_event.h"
 #include "content/common/media/media_stream_messages.h"
 #include "content/public/common/content_switches.h"
+#include "content/public/common/renderer_preferences.h"
 #include "content/renderer/media/media_stream.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 #include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/media_stream_video_source.h"
 #include "content/renderer/media/media_stream_video_track.h"
 #include "content/renderer/media/peer_connection_identity_service.h"
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "content/renderer/media/rtc_peer_connection_handler.h"
 #include "content/renderer/media/rtc_video_decoder_factory.h"
 #include "content/renderer/media/rtc_video_encoder_factory.h"
 #include "content/renderer/media/webaudio_capturer_source.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "content/renderer/media/webrtc_logging.h"
 #include "content/renderer/media/webrtc_uma_histograms.h"
 #include "content/renderer/p2p/ipc_network_manager.h"
 #include "content/renderer/p2p/ipc_socket_factory.h"
 #include "content/renderer/p2p/port_allocator.h"
 #include "content/renderer/render_thread_impl.h"
+#include "content/renderer/render_view_impl.h"
 #include "jingle/glue/thread_wrapper.h"
 #include "media/filters/gpu_video_accelerator_factories.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/WebKit/public/platform/WebMediaStream.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/WebKit/public/platform/WebURL.h"
 #include "third_party/WebKit/public/web/WebDocument.h"
 #include "third_party/WebKit/public/web/WebFrame.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
@@ skipping 61 matching lines @@
             webrtc::MediaConstraintsInterface::kValueFalse, true);
         // No need to modify |effects| since the ducking flag is already off.
         DCHECK((*effects & media::AudioParameters::DUCKING) == 0);
       }
     }
   }
 }
 
 class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface {
  public:
-  P2PPortAllocatorFactory(
-      P2PSocketDispatcher* socket_dispatcher,
-      rtc::NetworkManager* network_manager,
-      rtc::PacketSocketFactory* socket_factory)
+  P2PPortAllocatorFactory(P2PSocketDispatcher* socket_dispatcher,
+                          rtc::NetworkManager* network_manager,
+                          rtc::PacketSocketFactory* socket_factory,
+                          bool enable_multiple_routes)
       : socket_dispatcher_(socket_dispatcher),
         network_manager_(network_manager),
-        socket_factory_(socket_factory) {
-  }
+        socket_factory_(socket_factory),
+        enable_multiple_routes_(enable_multiple_routes) {}
 
   cricket::PortAllocator* CreatePortAllocator(
       const std::vector<StunConfiguration>& stun_servers,
       const std::vector<TurnConfiguration>& turn_configurations) override {
     P2PPortAllocator::Config config;
     for (size_t i = 0; i < stun_servers.size(); ++i) {
       config.stun_servers.insert(rtc::SocketAddress(
           stun_servers[i].server.hostname(),
           stun_servers[i].server.port()));
     }
     for (size_t i = 0; i < turn_configurations.size(); ++i) {
       P2PPortAllocator::Config::RelayServerConfig relay_config;
       relay_config.server_address = turn_configurations[i].server.hostname();
       relay_config.port = turn_configurations[i].server.port();
       relay_config.username = turn_configurations[i].username;
       relay_config.password = turn_configurations[i].password;
       relay_config.transport_type = turn_configurations[i].transport_type;
       relay_config.secure = turn_configurations[i].secure;
       config.relays.push_back(relay_config);
 
       // Use turn servers as stun servers.
       config.stun_servers.insert(rtc::SocketAddress(
           turn_configurations[i].server.hostname(),
           turn_configurations[i].server.port()));
     }
+    config.enable_multiple_routes = enable_multiple_routes_;
 
     return new P2PPortAllocator(
         socket_dispatcher_.get(), network_manager_, socket_factory_, config);
   }
 
  protected:
   ~P2PPortAllocatorFactory() override {}
 
  private:
   scoped_refptr<P2PSocketDispatcher> socket_dispatcher_;
   // |network_manager_| and |socket_factory_| are a weak references, owned by
   // PeerConnectionDependencyFactory.
   rtc::NetworkManager* network_manager_;
   rtc::PacketSocketFactory* socket_factory_;
+
+  // When false, only 'any' address (all 0s) will be bound for address
+  // discovery.
+  bool enable_multiple_routes_;
 };
 
 PeerConnectionDependencyFactory::PeerConnectionDependencyFactory(
     P2PSocketDispatcher* p2p_socket_dispatcher)
     : network_manager_(NULL),
       p2p_socket_dispatcher_(p2p_socket_dispatcher),
       signaling_thread_(NULL),
       worker_thread_(NULL),
       chrome_signaling_thread_("Chrome_libJingle_Signaling"),
       chrome_worker_thread_("Chrome_libJingle_WorkerThread") {
@@ skipping 213 matching lines @@
 PeerConnectionDependencyFactory::CreatePeerConnection(
     const webrtc::PeerConnectionInterface::RTCConfiguration& config,
     const webrtc::MediaConstraintsInterface* constraints,
     blink::WebFrame* web_frame,
     webrtc::PeerConnectionObserver* observer) {
   CHECK(web_frame);
   CHECK(observer);
   if (!GetPcFactory().get())
     return NULL;
 
+  // Copy the flag from Preference associated with this WebFrame.
+  bool enable_multiple_routes = true;
+  if (web_frame && web_frame->view()) {
+    RenderViewImpl* renderer_view_impl =
+        RenderViewImpl::FromWebView(web_frame->view());
+    if (renderer_view_impl) {
+      enable_multiple_routes = renderer_view_impl->renderer_preferences()
+                                    .enable_webrtc_multiple_routes;
+    }
+  }
+
   scoped_refptr<P2PPortAllocatorFactory> pa_factory =
-        new rtc::RefCountedObject<P2PPortAllocatorFactory>(
-            p2p_socket_dispatcher_.get(),
-            network_manager_,
-            socket_factory_.get());
+      new rtc::RefCountedObject<P2PPortAllocatorFactory>(
+          p2p_socket_dispatcher_.get(), network_manager_, socket_factory_.get(),
+          enable_multiple_routes);
 
   PeerConnectionIdentityService* identity_service =
       new PeerConnectionIdentityService(
           GURL(web_frame->document().url().spec()).GetOrigin());
 
   return GetPcFactory()->CreatePeerConnection(config,
                                               constraints,
                                               pa_factory.get(),
                                               identity_service,
                                               observer).get();
@@ skipping 210 matching lines @@
 }
 
 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
   if (audio_device_.get())
     return;
 
   audio_device_ = new WebRtcAudioDeviceImpl();
 }
 
 }  // namespace content