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