| Index: webrtc/BUILD.gn | 
| diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn | 
| index 01fa042af36da554506a0b82da03fd219b4575dd..1b209dda6aa67052f53efd79c6ac8415189e0d92 100644 | 
| --- a/webrtc/BUILD.gn | 
| +++ b/webrtc/BUILD.gn | 
| @@ -173,9 +173,14 @@ config("common_config") { | 
|  | 
| source_set("webrtc") { | 
| sources = [ | 
| +    "audio_receive_stream.h", | 
| +    "audio_send_stream.h", | 
| +    "audio_state.h", | 
| "call.h", | 
| -    "config.h", | 
| +    "common.h", | 
| "transport.h", | 
| +    "video_receive_stream.h", | 
| +    "video_send_stream.h", | 
| ] | 
|  | 
| defines = [] | 
| @@ -236,6 +241,7 @@ source_set("webrtc_common") { | 
| "config.h", | 
| "engine_configurations.h", | 
| "typedefs.h", | 
| +    "video_frame.h", | 
| ] | 
|  | 
| configs += [ ":common_config" ] | 
| @@ -271,6 +277,7 @@ source_set("rtc_event_log") { | 
|  | 
| deps = [ | 
| ":webrtc_common", | 
| +    "base:rtc_base_approved", | 
| ] | 
|  | 
| if (rtc_enable_protobuf) { | 
| @@ -297,6 +304,7 @@ if (rtc_enable_protobuf) { | 
| deps = [ | 
| ":rtc_event_log_proto", | 
| ":webrtc_common", | 
| +      "base:rtc_base_approved", | 
| ] | 
|  | 
| if (is_clang && !is_nacl) { | 
|  |