| 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) {
|
|
|