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

content/renderer/media/webrtc/peer_connection_dependency_factory.cc

 // Copyright 2014 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/webrtc/peer_connection_dependency_factory.h"
 
 #include <stddef.h>
 
 #include <utility>
 #include <vector>
@@ skipping 245 matching lines @@
   DCHECK(!signaling_thread_);
   DCHECK(!worker_thread_);
   DCHECK(!network_manager_);
   DCHECK(!socket_factory_);
   DCHECK(!chrome_signaling_thread_.IsRunning());
   DCHECK(!chrome_worker_thread_.IsRunning());
 
   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();
+  media::FFmpegGlue::InitializeFFmpeg();
 #endif
 
   base::MessageLoop::current()->AddDestructionObserver(this);
   // To allow sending to the signaling/worker threads.
   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 
   CHECK(chrome_signaling_thread_.Start());
   CHECK(chrome_worker_thread_.Start());
 
@@ skipping 511 matching lines @@
 }
 
 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
   if (audio_device_.get())
     return;
 
   audio_device_ = new WebRtcAudioDeviceImpl();
 }
 
 }  // namespace content