| 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",
|
|
|