Index: webrtc/modules/rtp_rtcp/BUILD.gn |
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn |
index c1d69e07bb129c6f14973ecc49cc8420c98465e6..ec1766641ed9f94727870d720e378b27669b3f94 100644 |
--- a/webrtc/modules/rtp_rtcp/BUILD.gn |
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn |
@@ -167,6 +167,7 @@ rtc_static_library("rtp_rtcp") { |
deps = [ |
"../..:webrtc_common", |
+ "../../api:libjingle_peerconnection_api", |
"../../api:transport_api", |
"../../api/audio_codecs:audio_codecs_api", |
"../../base:gtest_prod", |
@@ -274,6 +275,7 @@ if (rtc_include_tests) { |
"source/rtp_packet_history_unittest.cc", |
"source/rtp_packet_unittest.cc", |
"source/rtp_payload_registry_unittest.cc", |
+ "source/rtp_receiver_unittest.cc", |
"source/rtp_rtcp_impl_unittest.cc", |
"source/rtp_sender_unittest.cc", |
"source/rtp_utility_unittest.cc", |