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
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 @@
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 @@
}
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 @@
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 @@
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 @@
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() {
