Hook up WebRTC logging extension API to the underlying functionality.

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/strings/utf_string_conversions.h"
 #include "base/synchronization/waitable_event.h"
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream_source_extra_data.h"
 #include "content/renderer/media/peer_connection_identity_service.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/rtc_video_decoder_factory.h"
 #include "content/renderer/media/rtc_video_encoder_factory.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_initializer.h"

[no longer working on chromium, 2013/10/01 08:41:42]

remove if you don't need this

[Henrik Grunell, 2013/10/02 12:47:18]

Done.

 #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 "media/filters/gpu_video_accelerator_factories.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/WebKit/public/platform/WebMediaStream.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/WebKit/public/platform/WebURL.h"
 #include "third_party/WebKit/public/web/WebDocument.h"
 #include "third_party/WebKit/public/web/WebFrame.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
 
 #if defined(USE_OPENSSL)
 #include "third_party/libjingle/source/talk/base/ssladapter.h"
 #else
 #include "net/socket/nss_ssl_util.h"
 #endif
 
 #if defined(GOOGLE_TV)
 #include "content/renderer/media/rtc_video_decoder_factory_tv.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 enables default audio constraints on
 // mediastreams with audio.
 struct {
   const char* key;
   const char* value;
 } const kDefaultAudioConstraints[] = {
   { webrtc::MediaConstraintsInterface::kEchoCancellation,
     webrtc::MediaConstraintsInterface::kValueTrue },
 #if defined(OS_CHROMEOS) || defined(OS_MACOSX)
   // Enable the extended filter mode AEC on platforms with known echo issues.
@@ skipping 495 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 (optional_constraints.FindFirst(kWebRtcLoggingConstraint,
-                                     &constraint_value)) {
-    std::string url = web_frame->document().url().spec();
-    RenderThreadImpl::current()->GetIOMessageLoopProxy()->PostTask(
-        FROM_HERE, base::Bind(
-            &InitWebRtcLogging,
-            constraint_value,
-            url));
-  }
-
   scoped_refptr<P2PPortAllocatorFactory> pa_factory =
         new talk_base::RefCountedObject<P2PPortAllocatorFactory>(
             p2p_socket_dispatcher_.get(),
             network_manager_,
             socket_factory_.get(),
             web_frame);
 
   PeerConnectionIdentityService* identity_service =
       PeerConnectionIdentityService::Create(
           GURL(web_frame->document().url().spec()).GetOrigin());
@@ skipping 255 matching lines @@
   }
 
   // Add the capturer to the WebRtcAudioDeviceImpl if it is a new capturer.
   if (is_new_capturer)
     GetWebRtcAudioDevice()->AddAudioCapturer(capturer);
 
   return capturer;
 }
 
 }  // namespace content