| Index: content/content_renderer.gypi
|
| diff --git a/content/content_renderer.gypi b/content/content_renderer.gypi
|
| index 9b7a00f059ea502785e04e02a5b746c5691e1a4d..e37baef1e7d900a68c53588d2908b3153b5666b9 100644
|
| --- a/content/content_renderer.gypi
|
| +++ b/content/content_renderer.gypi
|
| @@ -625,6 +625,7 @@
|
| 'public/renderer/media_stream_renderer_factory.h',
|
| 'public/renderer/media_stream_video_sink.cc',
|
| 'public/renderer/media_stream_video_sink.h',
|
| + 'public/renderer/renderer_features.h',
|
| 'public/renderer/webrtc_log_message_delegate.h',
|
| ],
|
| # WebRTC-specific sources. Put WebRTC plugin-related stuff in the
|
| @@ -762,7 +763,6 @@
|
| 'renderer/p2p/socket_client_impl.h',
|
| 'renderer/p2p/socket_dispatcher.cc',
|
| 'renderer/p2p/socket_dispatcher.h',
|
| - 'renderer/renderer_features.h',
|
| ],
|
| # Stuff only used when both WebRTC and plugins are enabled.
|
| 'private_renderer_plugin_webrtc_sources': [
|
| @@ -837,6 +837,7 @@
|
| '../third_party/webrtc/modules/modules.gyp:audio_device',
|
| '../third_party/webrtc/modules/modules.gyp:audio_processing',
|
| '../third_party/webrtc/p2p/p2p.gyp:libstunprober',
|
| + '<(DEPTH)/content/public/renderer/media/webrtc/finch_h264_with_openh264_ffmpeg.gyp:finch_h264_with_openh264_ffmpeg',
|
| '<(DEPTH)/crypto/crypto.gyp:crypto',
|
| ],
|
| 'sources': [
|
|
|