Index: webrtc/call/BUILD.gn |
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn |
index 263f16dd354da795e71f3a24bdd0eea4cd24864d..bc30871f43be0c20364b61655751920c90a1754f 100644 |
--- a/webrtc/call/BUILD.gn |
+++ b/webrtc/call/BUILD.gn |
@@ -45,6 +45,7 @@ rtc_source_set("rtp_interfaces") { |
"rtp_transport_controller_send_interface.h", |
] |
deps = [ |
+ "../api:array_view", |
"../rtc_base:rtc_base_approved", |
] |
} |
@@ -66,6 +67,7 @@ rtc_source_set("rtp_receiver") { |
deps = [ |
":rtp_interfaces", |
"..:webrtc_common", |
+ "../api:array_view", |
"../modules/rtp_rtcp", |
"../rtc_base:rtc_base_approved", |
] |
@@ -175,6 +177,7 @@ if (rtc_include_tests) { |
":rtp_receiver", |
":rtp_sender", |
"..:webrtc_common", |
+ "../api:array_view", |
"../api:mock_audio_mixer", |
"../logging:rtc_event_log_api", |
"../modules/audio_device:mock_audio_device", |