| Index: webrtc/modules/rtp_rtcp/BUILD.gn
|
| diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
|
| index 8d17d1a799db5fa1f02f409dd20696697546746c..2095aadb284f4abeab59d39010a9490551761a96 100644
|
| --- a/webrtc/modules/rtp_rtcp/BUILD.gn
|
| +++ b/webrtc/modules/rtp_rtcp/BUILD.gn
|
| @@ -169,6 +169,7 @@ rtc_static_library("rtp_rtcp") {
|
| "../..:webrtc_common",
|
| "../../api:array_view",
|
| "../../api:libjingle_peerconnection_api",
|
| + "../../api:optional",
|
| "../../api:transport_api",
|
| "../../api/audio_codecs:audio_codecs_api",
|
| "../../common_video",
|
| @@ -222,6 +223,7 @@ rtc_source_set("mock_rtp_rtcp") {
|
| deps = [
|
| ":rtp_rtcp",
|
| "..:module_api",
|
| + "../../api:optional",
|
| "../../rtc_base:rtc_base_approved",
|
| "../../test:test_support",
|
| ]
|
|
|