| Index: content/content_renderer.gypi
|
| diff --git a/content/content_renderer.gypi b/content/content_renderer.gypi
|
| index b017e22ab5c253d837ae86e28e91928dd5b0518d..65473a334cdb94f6b02e84b0a129f2baca6ae6da 100644
|
| --- a/content/content_renderer.gypi
|
| +++ b/content/content_renderer.gypi
|
| @@ -773,7 +773,6 @@
|
| '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': [
|
| @@ -848,8 +847,19 @@
|
| '../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/content.gyp:feature_h264_with_openh264_ffmpeg',
|
| '<(DEPTH)/crypto/crypto.gyp:crypto',
|
| ],
|
| + 'includes': [
|
| + '../third_party/webrtc/build/common.gypi',
|
| + ],
|
| + 'conditions': [
|
| + ['rtc_use_h264==1', {
|
| + 'defines': [
|
| + 'BUILDFLAG_RTC_USE_H264',
|
| + ],
|
| + }],
|
| + ],
|
| 'sources': [
|
| '<@(public_renderer_webrtc_sources)',
|
| '<@(private_renderer_webrtc_sources)',
|
|
|