Enable H.264 video WebRTC behind run-time flag and add WebRtcBrowserTest for H.264

content/renderer/media/webrtc/peer_connection_dependency_factory.cc

Index: content/renderer/media/webrtc/peer_connection_dependency_factory.cc
diff --git a/content/renderer/media/webrtc/peer_connection_dependency_factory.cc b/content/renderer/media/webrtc/peer_connection_dependency_factory.cc
index 966a295fdf8fdd3fa61f66ff58cc7a996eb21dc2..75b5cd2e054a95a539fee15a53d9f5e178bec0ab 100644
--- a/content/renderer/media/webrtc/peer_connection_dependency_factory.cc
+++ b/content/renderer/media/webrtc/peer_connection_dependency_factory.cc
@@ -19,8 +19,10 @@
 #include "base/synchronization/waitable_event.h"
 #include "build/build_config.h"
 #include "content/common/media/media_stream_messages.h"
+#include "content/public/common/common_features.h"
 #include "content/public/common/content_client.h"
 #include "content/public/common/content_switches.h"
+#include "content/public/common/feature_h264_with_openh264_ffmpeg.h"
 #include "content/public/common/renderer_preferences.h"
 #include "content/public/common/webrtc_ip_handling_policy.h"
 #include "content/public/renderer/content_renderer_client.h"
@@ -52,7 +54,6 @@
 #include "content/renderer/render_frame_impl.h"
 #include "content/renderer/render_thread_impl.h"
 #include "content/renderer/render_view_impl.h"
-#include "content/renderer/renderer_features.h"
 #include "crypto/openssl_util.h"
 #include "jingle/glue/thread_wrapper.h"
 #include "media/base/media_permission.h"
@@ -263,19 +264,17 @@ void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() {
   DVLOG(1) << "PeerConnectionDependencyFactory::CreatePeerConnectionFactory()";
 
 #if BUILDFLAG(RTC_USE_H264)
-  // TODO(hbos): This is temporary. Disable the runtime effects of building with
-  // |rtc_use_h264|. We are planning to default the |rtc_use_h264| flag to
-  // |proprietary_codecs| so that it will be used by Chromium trybots. This
-  // would also make it used by Chrome, but this feature is not ready to be
-  // launched yet. An upcoming CL will add browser tests for H264. That CL will
-  // remove this line. It should remain disabled until tested.
-  webrtc::DisableRtcUseH264();
-  // When building with |rtc_use_h264|, |H264DecoderImpl| may be used which
-  // depends on FFmpeg, therefore we need to initialize FFmpeg before going
-  // further.
-  // TODO(hbos): Temporarily commented out due to webrtc::DisableRtcUseH264(),
-  // no need to initialize FFmpeg when |H264DecoderImpl| is disabled.
-  // media::FFmpegGlue::InitializeFFmpeg();
+  // Building /w |rtc_use_h264|, is the corresponding run-time feature enabled?
+  if (base::FeatureList::IsEnabled(kWebRtcH264WithOpenH264FFmpeg)) {
+    LOG(WARNING) << "WebRTC H.264 with OpenH264/FFmpeg ENABLED.";

[brettw, 2016/02/22 22:50:21]

Are these log statements really necessary? It seems like there would be
reasonable ways to check if you want to know if this is enabled, and currently
this spams the warning log every time the browser starts (and also adds >100
bytes of code for each one to the shipping product.

[hbos_chromium, 2016/02/23 08:21:04]

Not every time the browser starts, just the first (or possibly every time?) you
create a new RTCPeerConnection in javascript in a tab. I think that is
reasonable for the time being. Though I can remove the logs if you want, I don't
have a strong opinion about it. (Currently not removed.)

[hbos_chromium, 2016/02/23 10:40:56]

Nevermind, you're right, not necessary. Removed.

+    // |H264DecoderImpl| may be used which depends on FFmpeg, therefore we need
+    // to initialize FFmpeg before going further.
+    media::FFmpegGlue::InitializeFFmpeg();
+  } else {
+    LOG(WARNING) << "WebRTC H.264 with OpenH264/FFmpeg DISABLED.";
+    // Feature is to be disabled, no need to make sure FFmpeg is initialized.
+    webrtc::DisableRtcUseH264();
+  }
 #endif
 
   base::MessageLoop::current()->AddDestructionObserver(this);