Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 40539b82d5a2c9bcf23d078e997ce0368160f5a3

This commit was generated by merge_from_chromium.py.

Change-Id: I01a4e13d3b66df627fcc0992da660f11cfe85646

5 files changed, 9 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_;
diff --git a/session/media/channelmanager.cc b/session/media/channelmanager.cc
index 45e7e47..199bc86 100644
--- a/session/media/channelmanager.cc
+++ b/session/media/channelmanager.cc
@@ -137,6 +137,12 @@ void ChannelManager::Construct(MediaEngineInterface* me,
       this, &ChannelManager::OnVideoCaptureStateChange);
   capture_manager_->SignalCapturerStateChange.connect(
       this, &ChannelManager::OnVideoCaptureStateChange);
+
+  if (worker_thread_ != rtc::Thread::Current()) {
+    // Do not allow invoking calls to other threads on the worker thread.
+    worker_thread_->Invoke<bool>(
+        rtc::Bind(&rtc::Thread::SetAllowBlockingCalls, worker_thread_, false));
+  }
 }
 
 ChannelManager::~ChannelManager() {