libjingle roll 117:119

BUG=116125


Review URL: http://codereview.chromium.org/9567007

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@124466 0039d316-1c4b-4281-b951-d872f2087c98

DEPS

@@ -37,7 +37,7 @@ vars = {
   "nacl_toolchain_revision": "7799",
   "pnacl_toolchain_revision": "7799",
 
-  "libjingle_revision": "117",
+  "libjingle_revision": "119",
   "libphonenumber_revision": "425",
   "libvpx_revision": "123809",
   "lss_revision": "9",

content/renderer/media/media_stream_dependency_factory.cc

@@ -111,6 +111,13 @@ MediaStreamDependencyFactory::CreatePeerConnection(
   return pc_factory_->CreatePeerConnection(config, observer);
 }
 
+talk_base::scoped_refptr<webrtc::PeerConnectionInterface>
+MediaStreamDependencyFactory::CreateRoapPeerConnection(
+    const std::string& config,
+    webrtc::PeerConnectionObserver* observer) {
+  return pc_factory_->CreateRoapPeerConnection(config, observer);
+}
+
 talk_base::scoped_refptr<webrtc::LocalMediaStreamInterface>
 MediaStreamDependencyFactory::CreateLocalMediaStream(
     const std::string& label) {

content/renderer/media/media_stream_dependency_factory.h

@@ -56,6 +56,9 @@ class CONTENT_EXPORT MediaStreamDependencyFactory {
   virtual talk_base::scoped_refptr<webrtc::PeerConnectionInterface>
       CreatePeerConnection(const std::string& config,
                            webrtc::PeerConnectionObserver* observer);
+  virtual talk_base::scoped_refptr<webrtc::PeerConnectionInterface>
+      CreateRoapPeerConnection(const std::string& config,
+                               webrtc::PeerConnectionObserver* observer);
 
   // Asks the PeerConnection factory to create a Local MediaStream object.
   virtual talk_base::scoped_refptr<webrtc::LocalMediaStreamInterface>

content/renderer/media/mock_media_stream_dependency_factory.cc

@@ -5,7 +5,7 @@
 #include "base/logging.h"
 #include "content/renderer/media/mock_media_stream_dependency_factory.h"
 #include "content/renderer/media/mock_peer_connection_impl.h"
-#include "third_party/libjingle/source/talk/app/webrtc/mediastream.h"
+#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 #include "third_party/libjingle/source/talk/base/scoped_ref_ptr.h"
 
 namespace webrtc {
@@ -14,7 +14,7 @@ template <class TrackType>
 class MockMediaStreamTrackList
     : public MediaStreamTrackListInterface<TrackType> {
  public:
-  virtual size_t count() OVERRIDE {
+  virtual size_t count() const OVERRIDE {
     return tracks_.size();
   }
   virtual TrackType* at(size_t index) OVERRIDE {
@@ -56,7 +56,7 @@ class MockLocalMediaStream : public LocalMediaStreamInterface {
   virtual VideoTracks* video_tracks() OVERRIDE {
     return video_tracks_;
   }
-  virtual ReadyState ready_state() OVERRIDE {
+  virtual ReadyState ready_state() const OVERRIDE {
     NOTIMPLEMENTED();
     return kInitializing;
   }
@@ -157,6 +157,14 @@ MockMediaStreamDependencyFactory::CreatePeerConnection(
   return new talk_base::RefCountedObject<webrtc::MockPeerConnectionImpl>();
 }
 
+talk_base::scoped_refptr<webrtc::PeerConnectionInterface>
+MockMediaStreamDependencyFactory::CreateRoapPeerConnection(
+    const std::string& config,
+    webrtc::PeerConnectionObserver* observer) {
+  DCHECK(mock_pc_factory_created_);
+  return new talk_base::RefCountedObject<webrtc::MockPeerConnectionImpl>();
+}
+
 talk_base::scoped_refptr<webrtc::LocalMediaStreamInterface>
 MockMediaStreamDependencyFactory::CreateLocalMediaStream(
     const std::string& label) {

content/renderer/media/mock_media_stream_dependency_factory.h

@@ -63,6 +63,10 @@ class MockMediaStreamDependencyFactory : public MediaStreamDependencyFactory {
       CreatePeerConnection(
           const std::string& config,
           webrtc::PeerConnectionObserver* observer) OVERRIDE;
+  virtual talk_base::scoped_refptr<webrtc::PeerConnectionInterface>
+      CreateRoapPeerConnection(
+          const std::string& config,
+          webrtc::PeerConnectionObserver* observer) OVERRIDE;
   virtual talk_base::scoped_refptr<webrtc::LocalMediaStreamInterface>
       CreateLocalMediaStream(const std::string& label) OVERRIDE;
   virtual talk_base::scoped_refptr<webrtc::LocalVideoTrackInterface>

content/renderer/media/mock_peer_connection_impl.cc

@@ -91,6 +91,57 @@ MockPeerConnectionImpl::SdpState MockPeerConnectionImpl::sdp_state() {
   return kSdpNew;
 }
 
+void MockPeerConnectionImpl::StartIce(IceOptions options) {
+  NOTIMPLEMENTED();
+}
+
+webrtc::SessionDescriptionInterface* MockPeerConnectionImpl::CreateOffer(
+    const webrtc::MediaHints& hints) {
+  NOTIMPLEMENTED();
+  return NULL;
+}
+
+webrtc::SessionDescriptionInterface* MockPeerConnectionImpl::CreateAnswer(
+    const webrtc::MediaHints& hints,
+    const webrtc::SessionDescriptionInterface* offer) {
+  NOTIMPLEMENTED();
+  return NULL;
+}
+
+bool MockPeerConnectionImpl::SetLocalDescription(
+    Action action,
+    webrtc::SessionDescriptionInterface* desc) {
+  NOTIMPLEMENTED();
+  return false;
+}
+
+bool MockPeerConnectionImpl::SetRemoteDescription(
+    Action action,
+    webrtc::SessionDescriptionInterface* desc) {
+  NOTIMPLEMENTED();
+  return false;
+}
+
+bool MockPeerConnectionImpl::ProcessIceMessage(
+    const webrtc::IceCandidateInterface* ice_candidate) {
+  NOTIMPLEMENTED();
+  return false;
+}
+
+const webrtc::SessionDescriptionInterface*
+MockPeerConnectionImpl::local_description()
+    const {
+  NOTIMPLEMENTED();
+  return NULL;
+}
+
+const webrtc::SessionDescriptionInterface*
+MockPeerConnectionImpl::remote_description()
+    const {
+  NOTIMPLEMENTED();
+  return NULL;
+}
+
 void MockPeerConnectionImpl::AddRemoteStream(MediaStreamInterface* stream) {
   remote_streams_->AddStream(stream);
 }

content/renderer/media/mock_peer_connection_impl.h

@@ -32,6 +32,25 @@ class MockPeerConnectionImpl : public PeerConnectionInterface {
   virtual void Close() OVERRIDE;
   virtual ReadyState ready_state() OVERRIDE;
   virtual SdpState sdp_state() OVERRIDE;
+  virtual void StartIce(IceOptions options) OVERRIDE;
+
+  virtual webrtc::SessionDescriptionInterface* CreateOffer(
+      const webrtc::MediaHints& hints) OVERRIDE;
+  virtual webrtc::SessionDescriptionInterface* CreateAnswer(
+      const webrtc::MediaHints& hints,
+      const webrtc::SessionDescriptionInterface* offer) OVERRIDE;
+  virtual bool SetLocalDescription(
+      Action action,
+      webrtc::SessionDescriptionInterface* desc) OVERRIDE;
+  virtual bool SetRemoteDescription(
+      Action action,
+      webrtc::SessionDescriptionInterface* desc) OVERRIDE;
+  virtual bool ProcessIceMessage(
+      const webrtc::IceCandidateInterface* ice_candidate) OVERRIDE;
+  virtual const webrtc::SessionDescriptionInterface* local_description()
+      const OVERRIDE;
+  virtual const webrtc::SessionDescriptionInterface* remote_description()
+      const OVERRIDE;
 
   void AddRemoteStream(MediaStreamInterface* stream);
   void ClearStreamChangesCommitted() { stream_changes_committed_ = false; }

content/renderer/media/peer_connection_handler.cc

@@ -51,7 +51,7 @@ void PeerConnectionHandler::SetVideoRenderer(
 void PeerConnectionHandler::initialize(
     const WebKit::WebString& server_configuration,
     const WebKit::WebSecurityOrigin& security_origin) {
-  native_peer_connection_ = dependency_factory_->CreatePeerConnection(
+  native_peer_connection_ = dependency_factory_->CreateRoapPeerConnection(
       UTF16ToUTF8(server_configuration),
       this);
   CHECK(native_peer_connection_);
@@ -152,6 +152,17 @@ void PeerConnectionHandler::OnRemoveStream(
   }
 }
 
+void PeerConnectionHandler::OnIceCandidate(
+    const webrtc::IceCandidateInterface* candidate) {
+  // TODO(grunell): Implement.
+  NOTIMPLEMENTED();
+}
+
+void PeerConnectionHandler::OnIceComplete() {
+  // TODO(grunell): Implement.
+  NOTIMPLEMENTED();
+}
+
 void PeerConnectionHandler::AddStreams(
     const WebKit::WebVector<WebKit::WebMediaStreamDescriptor>& streams) {
   for (size_t i = 0; i < streams.size(); ++i) {

content/renderer/media/peer_connection_handler.h

@@ -64,6 +64,9 @@ class CONTENT_EXPORT PeerConnectionHandler
   virtual void OnStateChange(StateType state_changed) OVERRIDE;
   virtual void OnAddStream(webrtc::MediaStreamInterface* stream) OVERRIDE;
   virtual void OnRemoveStream(webrtc::MediaStreamInterface* stream) OVERRIDE;
+  virtual void OnIceCandidate(
+      const webrtc::IceCandidateInterface* candidate) OVERRIDE;
+  virtual void OnIceComplete() OVERRIDE;
 
  private:
   FRIEND_TEST_ALL_PREFIXES(PeerConnectionHandlerTest, Basic);

content/renderer/p2p/ipc_network_manager.cc

@@ -58,7 +58,10 @@ void IpcNetworkManager::OnNetworkListChanged(
             it->name, it->name, talk_base::IPAddress(address)));
   }
 
-  MergeNetworkList(networks);
+  bool changed = false;
+  MergeNetworkList(networks, &changed);
+  if (changed)
+    SignalNetworksChanged();
 }
 
 void IpcNetworkManager::SendNetworksChangedSignal() {

third_party/libjingle/README.chromium

 Name: Libjingle library. Used for p2p voice and video communication.
 Short Name: libjingle
 URL: http://code.google.com/p/libjingle/
-Version: 0.6.13
-Revision: 117
+Version: 0.6.14
+Revision: 119
 License: BSD
 License File: source/COPYING
 Security Critical: yes

third_party/libjingle/libjingle.gyp

@@ -486,13 +486,21 @@
       'target_name': 'libjingle_peerconnection',
       'type': 'static_library',
       'sources': [
-        'source/talk/app/webrtc/audiotrackimpl.cc',
-        'source/talk/app/webrtc/audiotrackimpl.h',
+        'source/talk/app/webrtc/audiotrack.cc',
+        'source/talk/app/webrtc/audiotrack.h',
+        'source/talk/app/webrtc/candidateobserver.h',
+        'source/talk/app/webrtc/jsep.h',
+        'source/talk/app/webrtc/jsepicecandidate.cc',
+        'source/talk/app/webrtc/jsepicecandidate.h',
+        'source/talk/app/webrtc/jsepsessiondescription.cc',
+        'source/talk/app/webrtc/jsepsessiondescription.h',
+        'source/talk/app/webrtc/jsepsignaling.cc',
+        'source/talk/app/webrtc/jsepsignaling.h',
+        'source/talk/app/webrtc/mediastream.cc',
         'source/talk/app/webrtc/mediastream.h',
         'source/talk/app/webrtc/mediastreamhandler.cc',
         'source/talk/app/webrtc/mediastreamhandler.h',
-        'source/talk/app/webrtc/mediastreamimpl.cc',
-        'source/talk/app/webrtc/mediastreamimpl.h',
+        'source/talk/app/webrtc/mediastreaminterface.h',
         'source/talk/app/webrtc/mediastreamprovider.h',
         'source/talk/app/webrtc/mediastreamproxy.cc',
         'source/talk/app/webrtc/mediastreamproxy.h',