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>
 
 #include "base/command_line.h"
 #include "base/location.h"
 #include "base/logging.h"
 #include "base/macros.h"
 #include "base/metrics/field_trial.h"
 #include "base/strings/string_util.h"
 #include "base/strings/utf_string_conversions.h"
 #include "base/synchronization/waitable_event.h"
 #include "build/build_config.h"
 #include "content/common/media/media_stream_messages.h"
 #include "content/public/common/content_client.h"
 #include "content/public/common/content_switches.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"
+#include "content/public/renderer/media/webrtc/finch_h264_with_openh264_ffmpeg.h"
+#include "content/public/renderer/renderer_features.h"
 #include "content/renderer/media/media_stream.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 #include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/media_stream_video_source.h"
 #include "content/renderer/media/media_stream_video_track.h"
 #include "content/renderer/media/peer_connection_identity_store.h"
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "content/renderer/media/rtc_peer_connection_handler.h"
 #include "content/renderer/media/rtc_video_decoder_factory.h"
 #include "content/renderer/media/rtc_video_encoder_factory.h"
 #include "content/renderer/media/webaudio_capturer_source.h"
 #include "content/renderer/media/webrtc/media_stream_remote_audio_track.h"
 #include "content/renderer/media/webrtc/stun_field_trial.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.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.h"
 #include "content/renderer/media/webrtc_uma_histograms.h"
 #include "content/renderer/p2p/empty_network_manager.h"
 #include "content/renderer/p2p/filtering_network_manager.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_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"
 #include "media/filters/ffmpeg_glue.h"
 #include "media/renderers/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"
@@ skipping 190 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();
+  // Building with |rtc_use_h264| - is the corresponding Finch feature enabled?
+  if (base::FeatureList::IsEnabled(kWebRtcH264WithOpenH264FFmpeg)) {
+    LOG(WARNING) << "WebRTC H.264 with OpenH264/FFmpeg ENABLED.";

[jochen (gone - plz use gerrit), 2016/02/17 10:55:41]

please use DVLOG()

+    // |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);
   // 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