| 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).
|
|
|