| Index: webrtc/call/BUILD.gn
|
| diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
|
| index d86df1e379a9686c2b8243c84dc49f69cb4e6390..5a6257ec8b773f0543c4033c5214ccc6e427b1d6 100644
|
| --- a/webrtc/call/BUILD.gn
|
| +++ b/webrtc/call/BUILD.gn
|
| @@ -43,6 +43,7 @@ rtc_source_set("rtp_interfaces") {
|
| "rtp_transport_controller_send_interface.h",
|
| ]
|
| deps = [
|
| + "../api:array_view",
|
| "../rtc_base:rtc_base_approved",
|
| ]
|
| }
|
| @@ -64,6 +65,7 @@ rtc_source_set("rtp_receiver") {
|
| deps = [
|
| ":rtp_interfaces",
|
| "..:webrtc_common",
|
| + "../api:array_view",
|
| "../modules/rtp_rtcp",
|
| "../rtc_base:rtc_base_approved",
|
| ]
|
| @@ -167,6 +169,7 @@ 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",
|
|
|