Fix a problem in Thread::Send.

Previously if thread A->Send is called on thread B, B->ReceiveSends will be called, which enables an arbitrary thread to invoke calls on B while B is wait for A->Send to return. This caused mutliple problems like issue 3559, 3579.
The fix is to limit B->ReceiveSends to only process requests from A.
Also disallow the worker thread invoking other threads.

BUG=3559
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7290 4adac7df-926f-26a2-2b94-8c16560cd09d

4 files changed, 3 insertions, 11 deletions

diff --git a/app/webrtc/peerconnection_unittest.cc b/app/webrtc/peerconnection_unittest.cc
index 0d3e426..977fc11 100644
--- a/app/webrtc/peerconnection_unittest.cc
+++ b/app/webrtc/peerconnection_unittest.cc
@@ -481,9 +481,8 @@ class PeerConnectionTestClientBase
     if (!allocator_factory_) {
       return false;
     }
-    audio_thread_.Start();
     fake_audio_capture_module_ = FakeAudioCaptureModule::Create(
-        &audio_thread_);
+        rtc::Thread::Current());
 
     if (fake_audio_capture_module_ == NULL) {
       return false;
@@ -557,12 +556,6 @@ class PeerConnectionTestClientBase
   }
 
   std::string id_;
-  // Separate thread for executing |fake_audio_capture_module_| tasks. Audio
-  // processing must not be performed on the same thread as signaling due to
-  // signaling time constraints and relative complexity of the audio pipeline.
-  // This is consistent with the video pipeline that us a a separate thread for
-  // encoding and decoding.
-  rtc::Thread audio_thread_;
 
   rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
       allocator_factory_;
diff --git a/app/webrtc/peerconnectionfactory.cc b/app/webrtc/peerconnectionfactory.cc
index 5dccba8..862ceda 100644
--- a/app/webrtc/peerconnectionfactory.cc
+++ b/app/webrtc/peerconnectionfactory.cc
@@ -41,6 +41,7 @@
 #include "talk/media/webrtc/webrtcmediaengine.h"
 #include "talk/media/webrtc/webrtcvideodecoderfactory.h"
 #include "talk/media/webrtc/webrtcvideoencoderfactory.h"
+#include "webrtc/base/bind.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 
 using rtc::scoped_refptr;
diff --git a/app/webrtc/test/peerconnectiontestwrapper.cc b/app/webrtc/test/peerconnectiontestwrapper.cc
index 8a4f45c..24932b8 100644
--- a/app/webrtc/test/peerconnectiontestwrapper.cc
+++ b/app/webrtc/test/peerconnectiontestwrapper.cc
@@ -75,9 +75,8 @@ bool PeerConnectionTestWrapper::CreatePc(
     return false;
   }
 
-  audio_thread_.Start();
   fake_audio_capture_module_ = FakeAudioCaptureModule::Create(
-      &audio_thread_);
+      rtc::Thread::Current());
   if (fake_audio_capture_module_ == NULL) {
     return false;
   }
diff --git a/app/webrtc/test/peerconnectiontestwrapper.h b/app/webrtc/test/peerconnectiontestwrapper.h
index f3477ce..d4a0e4e 100644
--- a/app/webrtc/test/peerconnectiontestwrapper.h
+++ b/app/webrtc/test/peerconnectiontestwrapper.h
@@ -111,7 +111,6 @@ class PeerConnectionTestWrapper
       bool video, const webrtc::FakeConstraints& video_constraints);
 
   std::string name_;
-  rtc::Thread audio_thread_;
   rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
       allocator_factory_;
   rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;