Index: webrtc/call/BUILD.gn |
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn |
index 8f0f38b9d56acd690877a4f397eaf242e0314b57..aa98053cb77344bb54cbe3c130c809c2f150161f 100644 |
--- a/webrtc/call/BUILD.gn |
+++ b/webrtc/call/BUILD.gn |
@@ -37,25 +37,16 @@ |
# when interfaces have stabilized. |
rtc_source_set("rtp_interfaces") { |
sources = [ |
- "rtcp_packet_sink_interface.h", |
"rtp_packet_sink_interface.h", |
"rtp_stream_receiver_controller_interface.h", |
"rtp_transport_controller_send_interface.h", |
] |
- deps = [ |
- "//webrtc/base:rtc_base_approved", |
- ] |
} |
rtc_source_set("rtp_receiver") { |
sources = [ |
- "rsid_resolution_observer.h", |
- "rtcp_demuxer.cc", |
- "rtcp_demuxer.h", |
"rtp_demuxer.cc", |
"rtp_demuxer.h", |
- "rtp_rtcp_demuxer_helper.cc", |
- "rtp_rtcp_demuxer_helper.h", |
"rtp_stream_receiver_controller.cc", |
"rtp_stream_receiver_controller.h", |
"rtx_receive_stream.cc", |
@@ -63,9 +54,8 @@ |
] |
deps = [ |
":rtp_interfaces", |
+ "../base:rtc_base_approved", |
"../modules/rtp_rtcp", |
- "//webrtc:webrtc_common", |
- "//webrtc/base:rtc_base_approved", |
] |
} |
@@ -137,9 +127,7 @@ |
"bitrate_estimator_tests.cc", |
"call_unittest.cc", |
"flexfec_receive_stream_unittest.cc", |
- "rtcp_demuxer_unittest.cc", |
"rtp_demuxer_unittest.cc", |
- "rtp_rtcp_demuxer_helper_unittest.cc", |
"rtx_receive_stream_unittest.cc", |
] |
deps = [ |
@@ -165,7 +153,6 @@ |
"../test:video_test_common", |
"//testing/gmock", |
"//testing/gtest", |
- "//webrtc:webrtc_common", |
] |
if (!build_with_chromium && is_clang) { |
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |