Revert 118031 - Roll libjingle to r108:

* Add app/webrtc. * Add examples/peerconnection. * Bug fixes. Roll webrtc to 1311. BUG=110255 TEST=build and unit tests Review URL: http://codereview.chromium.org/9174012 TBR=ronghuawu@chromium.org Review URL: https://chromiumcodereview.appspot.com/9253007 git-svn-id: svn://svn.chromium.org/chrome/trunk/src@118058 0039d316-1c4b-4281-b951-d872f2087c98
rogerwang · Jan 18, 2012 · e08c430 · e08c430
1 parent 8720689
commit e08c430
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Showing 11 changed files with 1,650 additions and 35 deletions.
diff --git a/DEPS b/DEPS
@@ -30,7 +30,7 @@ vars = {
       "b41a2e565716b3ddc0212d185afa79881e971c58",
   "nacl_toolchain_revision": "7537",
 
-  "libjingle_revision": "108",
+  "libjingle_revision": "100",
   "libphonenumber_revision": "407",
   "libvpx_revision": "117975",
   "lss_revision": "9",
@@ -41,7 +41,7 @@ vars = {
   # the commit queue can handle CLs rolling Skia
   # and V8 without interference from each other.
   "v8_revision": "10408",
-  "webrtc_revision": "1311",
+  "webrtc_revision": "1264",
   "jsoncpp_revision": "248",
 }
 

diff --git a/content/renderer/media/media_stream_dependency_factory.cc b/content/renderer/media/media_stream_dependency_factory.cc
@@ -10,7 +10,7 @@
 #include "content/renderer/p2p/ipc_socket_factory.h"
 #include "content/renderer/p2p/port_allocator.h"
 #include "jingle/glue/thread_wrapper.h"
-#include "third_party/libjingle/source/talk/app/webrtcv1/peerconnection.h"
+#include "third_party/libjingle/source/talk/app/webrtc/peerconnection.h"
 #include "third_party/libjingle/source/talk/session/phone/dummydevicemanager.h"
 #include "third_party/libjingle/source/talk/session/phone/webrtcmediaengine.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/WebFrame.h"

diff --git a/content/renderer/media/media_stream_dependency_factory.h b/content/renderer/media/media_stream_dependency_factory.h
@@ -10,7 +10,7 @@
 #include "base/basictypes.h"
 #include "base/memory/scoped_ptr.h"
 #include "content/common/content_export.h"
-#include "third_party/libjingle/source/talk/app/webrtcv1/peerconnectionfactory.h"
+#include "third_party/libjingle/source/talk/app/webrtc/peerconnectionfactory.h"
 #include "webkit/glue/p2p_transport.h"
 
 namespace content {

diff --git a/content/renderer/media/mock_peer_connection_impl.h b/content/renderer/media/mock_peer_connection_impl.h
@@ -9,7 +9,7 @@
 
 #include "base/basictypes.h"
 #include "base/compiler_specific.h"
-#include "third_party/libjingle/source/talk/app/webrtcv1/peerconnection.h"
+#include "third_party/libjingle/source/talk/app/webrtc/peerconnection.h"
 
 namespace webrtc {
 

diff --git a/content/renderer/media/peer_connection_handler.h b/content/renderer/media/peer_connection_handler.h
@@ -13,7 +13,7 @@
 #include "base/memory/scoped_ptr.h"
 #include "base/message_loop_proxy.h"
 #include "content/common/content_export.h"
-#include "third_party/libjingle/source/talk/app/webrtcv1/peerconnection.h"
+#include "third_party/libjingle/source/talk/app/webrtc/peerconnection.h"
 #include "third_party/libjingle/source/talk/base/socketaddress.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebPeerConnectionHandler.h"
 

diff --git a/content/renderer/media/peer_connection_handler_unittest.cc b/content/renderer/media/peer_connection_handler_unittest.cc
@@ -15,7 +15,7 @@
 #include "content/renderer/media/rtc_video_decoder.h"
 #include "jingle/glue/thread_wrapper.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "third_party/libjingle/source/talk/app/webrtcv1/peerconnection.h"
+#include "third_party/libjingle/source/talk/app/webrtc/peerconnection.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebMediaStreamDescriptor.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebMediaStreamSource.h"
 

diff --git a/jingle/glue/thread_wrapper.cc b/jingle/glue/thread_wrapper.cc
@@ -1,4 +1,4 @@
-// Copyright (c) 2012 The Chromium Authors. All rights reserved.
+// Copyright (c) 2011 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.
 
@@ -49,7 +49,7 @@ JingleThreadWrapper::JingleThreadWrapper(MessageLoop* message_loop)
       pending_send_event_(true, false) {
   DCHECK_EQ(message_loop_, MessageLoop::current());
 
-  talk_base::ThreadManager::Instance()->SetCurrentThread(this);
+  talk_base::ThreadManager::SetCurrent(this);
   talk_base::MessageQueueManager::Instance()->Add(this);
   message_loop_->AddDestructionObserver(this);
 
@@ -63,7 +63,7 @@ void JingleThreadWrapper::WillDestroyCurrentMessageLoop() {
   DCHECK_EQ(talk_base::Thread::Current(), current());
   UnwrapCurrent();
   g_jingle_thread_wrapper.Get().Set(NULL);
-  talk_base::ThreadManager::Instance()->SetCurrentThread(NULL);
+  talk_base::ThreadManager::SetCurrent(NULL);
   talk_base::MessageQueueManager::Instance()->Remove(this);
   message_loop_->RemoveDestructionObserver(this);
   delete this;

diff --git a/third_party/libjingle/README.chromium b/third_party/libjingle/README.chromium
@@ -1,8 +1,8 @@
 Name: Libjingle library. Used for p2p voice and video communication.
 Short Name: libjingle
 URL: http://code.google.com/p/libjingle/
-Version: 0.6.10
-Revision: 108
+Version: 2.7.0
+Revision: 100
 License: BSD
 License File: source/COPYING
 Security Critical: yes

diff --git a/third_party/libjingle/libjingle.gyp b/third_party/libjingle/libjingle.gyp
@@ -282,10 +282,8 @@
         'source/talk/base/taskrunner.h',
         'source/talk/base/thread.cc',
         'source/talk/base/thread.h',
-        'source/talk/base/timeutils.cc',
-        'source/talk/base/timeutils.h',
-        'source/talk/base/timing.cc',
-        'source/talk/base/timing.h',
+        'source/talk/base/time.cc',
+        'source/talk/base/time.h',
         'source/talk/base/urlencode.cc',
         'source/talk/base/urlencode.h',
         'source/talk/base/worker.cc',
@@ -311,6 +309,8 @@
         'source/talk/xmpp/jid.h',
         'source/talk/xmpp/plainsaslhandler.h',
         'source/talk/xmpp/prexmppauth.h',
+        'source/talk/xmpp/ratelimitmanager.cc',
+        'source/talk/xmpp/ratelimitmanager.h',
         'source/talk/xmpp/saslcookiemechanism.h',
         'source/talk/xmpp/saslhandler.h',
         'source/talk/xmpp/saslmechanism.cc',
@@ -458,17 +458,17 @@
       'target_name': 'libjingle_peerconnection',
       'type': 'static_library',
       'sources': [
-        'source/talk/app/webrtcv1/peerconnection.h',
-        'source/talk/app/webrtcv1/peerconnectionfactory.cc',
-        'source/talk/app/webrtcv1/peerconnectionfactory.h',
-        'source/talk/app/webrtcv1/peerconnectionimpl.cc',
-        'source/talk/app/webrtcv1/peerconnectionimpl.h',
-        'source/talk/app/webrtcv1/peerconnectionproxy.cc',
-        'source/talk/app/webrtcv1/peerconnectionproxy.h',
-        'source/talk/app/webrtcv1/webrtcsession.cc',
-        'source/talk/app/webrtcv1/webrtcsession.h',
-        'source/talk/app/webrtcv1/webrtcjson.cc',
-        'source/talk/app/webrtcv1/webrtcjson.h',
+        'source/talk/app/webrtc/peerconnection.h',
+        'source/talk/app/webrtc/peerconnectionfactory.cc',
+        'source/talk/app/webrtc/peerconnectionfactory.h',
+        'source/talk/app/webrtc/peerconnectionimpl.cc',
+        'source/talk/app/webrtc/peerconnectionimpl.h',
+        'source/talk/app/webrtc/peerconnectionproxy.cc',
+        'source/talk/app/webrtc/peerconnectionproxy.h',
+        'source/talk/app/webrtc/webrtcsession.cc',
+        'source/talk/app/webrtc/webrtcsession.h',
+        'source/talk/app/webrtc/webrtcjson.cc',
+        'source/talk/app/webrtc/webrtcjson.h',
         'source/talk/session/phone/audiomonitor.cc',
         'source/talk/session/phone/audiomonitor.h',
         'source/talk/session/phone/call.cc',
@@ -512,7 +512,6 @@
         'source/talk/session/phone/srtpfilter.h',
         'source/talk/session/phone/ssrcmuxfilter.cc',
         'source/talk/session/phone/ssrcmuxfilter.h',
-        'source/talk/session/phone/streamparams.cc',
         'source/talk/session/phone/videocapturer.cc',
         'source/talk/session/phone/videocapturer.h',
         'source/talk/session/phone/videocommon.cc',
@@ -524,7 +523,9 @@
         'source/talk/session/phone/webrtcpassthroughrender.cc',
         'source/talk/session/phone/webrtcvideocapturer.cc',
         'source/talk/session/phone/webrtcvideocapturer.h',
-        'source/talk/session/phone/webrtcvideoengine.cc',
+        # TODO(ronghuawu): Fix this in libjingle and remove the override.
+        # See http://crbug.com/110255
+        'overrides/talk/session/phone/webrtcvideoengine.cc',
         'source/talk/session/phone/webrtcvideoengine.h',
         'source/talk/session/phone/webrtcvideoframe.cc',
         'source/talk/session/phone/webrtcvideoframe.h',

diff --git a/third_party/libjingle/overrides/talk/base/basictypes.h b/third_party/libjingle/overrides/talk/base/basictypes.h
@@ -1,4 +1,4 @@
-// Copyright (c) 2012 The Chromium Authors. All rights reserved.
+// Copyright (c) 2011 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.
 
@@ -60,9 +60,4 @@ const int kForever = -1;
   (reinterpret_cast<uint8*>(((reinterpret_cast<uintptr_t>(p) + \
   ((t)-1)) & ~((t)-1))))
 
-// LIBJINGLE_DEFINE_STATIC_LOCAL() is a libjingle's copy
-// of CR_DEFINE_STATIC_LOCAL().
-#define LIBJINGLE_DEFINE_STATIC_LOCAL(type, name, arguments) \
-  CR_DEFINE_STATIC_LOCAL(type, name, arguments)
-
 #endif // OVERRIDES_TALK_BASE_BASICTYPES_H__