Index: webrtc/audio/BUILD.gn |
diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn |
index 5296f1fa827f07963d513cdc56efd42d291fadb6..e94e354919258a723a9b7678a37fc4c17dca8ccd 100644 |
--- a/webrtc/audio/BUILD.gn |
+++ b/webrtc/audio/BUILD.gn |
@@ -24,6 +24,8 @@ rtc_static_library("audio") { |
"audio_transport_proxy.h", |
"conversion.h", |
"scoped_voe_interface.h", |
+ "time_interval.cc", |
+ "time_interval.h", |
] |
if (!build_with_chromium && is_clang) { |
@@ -73,6 +75,7 @@ if (rtc_include_tests) { |
"audio_receive_stream_unittest.cc", |
"audio_send_stream_unittest.cc", |
"audio_state_unittest.cc", |
+ "time_interval_unittest.cc", |
] |
deps = [ |
":audio", |