Moving WebRTC logging related files from content to chrome.

content/renderer/media/media_stream_dependency_factory.cc

 // Copyright (c) 2012 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.
 
 #include "content/renderer/media/media_stream_dependency_factory.h"
 
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/synchronization/waitable_event.h"
 #include "base/utf_string_conversions.h"
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream_source_extra_data.h"
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "content/renderer/media/rtc_peer_connection_handler.h"
 #include "content/renderer/media/rtc_video_capturer.h"
 #include "content/renderer/media/video_capture_impl_manager.h"
 #include "content/renderer/media/webaudio_capturer_source.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
-#include "content/renderer/media/webrtc_logging_handler_impl.h"
-#include "content/renderer/media/webrtc_logging_message_filter.h"
 #include "content/renderer/media/webrtc_uma_histograms.h"
 #include "content/renderer/p2p/ipc_network_manager.h"
 #include "content/renderer/p2p/ipc_socket_factory.h"
 #include "content/renderer/p2p/port_allocator.h"
 #include "content/renderer/render_thread_impl.h"
 #include "jingle/glue/thread_wrapper.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaConstraints.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStream.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamSource.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamTrack.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/WebFrame.h"
 
 #if defined(USE_OPENSSL)
 #include "third_party/libjingle/source/talk/base/ssladapter.h"
 #else
 #include "net/socket/nss_ssl_util.h"
 #endif
 
 namespace content {
 
-// The constraint key for the PeerConnection constructor for enabling diagnostic
-// WebRTC logging. It's a Google specific key, hence the "goog" prefix.
-const char kWebRtcLoggingConstraint[] = "googLog";
-
 // Constant constraint keys which disables all audio constraints.
 // Only used in combination with WebAudio sources.
 struct {
   const char* key;
   const char* value;
 } const kWebAudioConstraints[] = {
     {webrtc::MediaConstraintsInterface::kEchoCancellation,
      webrtc::MediaConstraintsInterface::kValueFalse},
     {webrtc::MediaConstraintsInterface::kAutoGainControl,
      webrtc::MediaConstraintsInterface::kValueFalse},
@@ skipping 151 matching lines @@
 };
 
 MediaStreamDependencyFactory::MediaStreamDependencyFactory(
     VideoCaptureImplManager* vc_manager,
     P2PSocketDispatcher* p2p_socket_dispatcher)
     : network_manager_(NULL),
       vc_manager_(vc_manager),
       p2p_socket_dispatcher_(p2p_socket_dispatcher),
       signaling_thread_(NULL),
       worker_thread_(NULL),
-      chrome_worker_thread_("Chrome_libJingle_WorkerThread"),
-      webrtc_log_open_(false) {
+      chrome_worker_thread_("Chrome_libJingle_WorkerThread") {
 }
 
 MediaStreamDependencyFactory::~MediaStreamDependencyFactory() {
   CleanupPeerConnectionFactory();
 }
 
 WebKit::WebRTCPeerConnectionHandler*
 MediaStreamDependencyFactory::CreateRTCPeerConnectionHandler(
     WebKit::WebRTCPeerConnectionHandlerClient* client) {
   // Save histogram data so we can see how much PeerConnetion is used.
@@ skipping 264 matching lines @@
 
 scoped_refptr<webrtc::PeerConnectionInterface>
 MediaStreamDependencyFactory::CreatePeerConnection(
     const webrtc::PeerConnectionInterface::IceServers& ice_servers,
     const webrtc::MediaConstraintsInterface* constraints,
     WebKit::WebFrame* web_frame,
     webrtc::PeerConnectionObserver* observer) {
   CHECK(web_frame);
   CHECK(observer);
 
-  webrtc::MediaConstraintsInterface::Constraints optional_constraints =
-      constraints->GetOptional();
-  std::string constraint_value;
-  if (!webrtc_log_open_ &&
-      optional_constraints.FindFirst(kWebRtcLoggingConstraint,
-                                     &constraint_value)) {
-    webrtc_log_open_ = true;
-
-    RenderThreadImpl::current()->GetIOMessageLoopProxy()->PostTask(
-        FROM_HERE, base::Bind(
-            &MediaStreamDependencyFactory::CreateWebRtcLoggingHandler,
-            base::Unretained(this),
-            RenderThreadImpl::current()->webrtc_logging_message_filter(),
-            constraint_value));
-  }
-
   scoped_refptr<P2PPortAllocatorFactory> pa_factory =
         new talk_base::RefCountedObject<P2PPortAllocatorFactory>(
             p2p_socket_dispatcher_.get(),
             network_manager_,
             socket_factory_.get(),
             web_frame);
   return pc_factory_->CreatePeerConnection(
       ice_servers, constraints, pa_factory, observer).get();
 }
 
@@ skipping 250 matching lines @@
       // Stopping the thread will wait until all tasks have been
       // processed before returning. We wait for the above task to finish before
       // letting the the function continue to avoid any potential race issues.
       chrome_worker_thread_.Stop();
     } else {
       NOTREACHED() << "Worker thread not running.";
     }
   }
 }
 
-void MediaStreamDependencyFactory::CreateWebRtcLoggingHandler(
-    WebRtcLoggingMessageFilter* filter,
-    const std::string& app_session_id) {
-  WebRtcLoggingHandlerImpl* handler =
-      new WebRtcLoggingHandlerImpl(filter->io_message_loop());
-  filter->InitLogging(handler, app_session_id);
-}
-
 }  // namespace content