Implement WebRTC in Chrome for TV

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"
@@ skipping 21 matching lines @@
 #include "third_party/WebKit/public/platform/WebURL.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/WebDocument.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
 
+#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 disables all audio constraints.
 // Only used in combination with WebAudio sources.
 struct {
   const char* key;
@@ skipping 153 matching lines @@
   bool live_;
   typedef std::vector<scoped_refptr<webrtc::MediaSourceInterface> >
       ObservedSources;
   ObservedSources sources_;
 };
 
 MediaStreamDependencyFactory::MediaStreamDependencyFactory(
     VideoCaptureImplManager* vc_manager,
     P2PSocketDispatcher* p2p_socket_dispatcher)
     : network_manager_(NULL),
+#if defined(GOOGLE_TV)
+      decoder_factory_tv_(NULL),
+#endif
       vc_manager_(vc_manager),
       p2p_socket_dispatcher_(p2p_socket_dispatcher),
       signaling_thread_(NULL),
       worker_thread_(NULL),
       chrome_worker_thread_("Chrome_libJingle_WorkerThread") {
 }
 
 MediaStreamDependencyFactory::~MediaStreamDependencyFactory() {
   CleanupPeerConnectionFactory();
 }
@@ skipping 245 matching lines @@
   return type == WebKit::WebMediaStreamSource::TypeAudio ?
       native_stream->RemoveTrack(native_stream->FindAudioTrack(track_id)) :
       native_stream->RemoveTrack(native_stream->FindVideoTrack(track_id));
 }
 
 bool MediaStreamDependencyFactory::CreatePeerConnectionFactory() {
   DVLOG(1) << "MediaStreamDependencyFactory::CreatePeerConnectionFactory()";
   if (!pc_factory_.get()) {
     DCHECK(!audio_device_.get());
     audio_device_ = new WebRtcAudioDeviceImpl();
+
+    cricket::WebRtcVideoDecoderFactory* decoder_factory = NULL;
+#if defined(GOOGLE_TV)
+    // PeerConnectionFactory will hold the ownership of this
+    // VideoDecoderFactory.
+    decoder_factory = decoder_factory_tv_ = new RTCVideoDecoderFactoryTv;
+#endif
+
     scoped_refptr<webrtc::PeerConnectionFactoryInterface> factory(
         webrtc::CreatePeerConnectionFactory(worker_thread_,
                                             signaling_thread_,
                                             audio_device_.get(),
                                             NULL,
-                                            NULL));
+                                            decoder_factory));
     if (factory.get())
       pc_factory_ = factory;
     else
       audio_device_ = NULL;
   }
   return pc_factory_.get() != NULL;
 }
 
 bool MediaStreamDependencyFactory::PeerConnectionFactoryCreated() {
   return pc_factory_.get() != NULL;
@@ skipping 286 matching lines @@
       // 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.";
     }
   }
 }
 
 }  // namespace content