| Index: webrtc/call/BUILD.gn
|
| diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
|
| index 2d1853063acfb3ef7f6f80ea0df17d6b0302a4ee..6a50ea09a2a0043a12e66acb1d551fb2830c4b3a 100644
|
| --- a/webrtc/call/BUILD.gn
|
| +++ b/webrtc/call/BUILD.gn
|
| @@ -28,8 +28,8 @@ rtc_source_set("call_interfaces") {
|
| "../api:libjingle_peerconnection_api",
|
| "../api:transport_api",
|
| "../api/audio_codecs:audio_codecs_api",
|
| - "../rtc_base:rtc_base",
|
| - "../rtc_base:rtc_base_approved",
|
| + "../base:rtc_base",
|
| + "../base:rtc_base_approved",
|
| ]
|
| }
|
|
|
| @@ -43,7 +43,7 @@ rtc_source_set("rtp_interfaces") {
|
| "rtp_transport_controller_send_interface.h",
|
| ]
|
| deps = [
|
| - "../rtc_base:rtc_base_approved",
|
| + "../base:rtc_base_approved",
|
| ]
|
| }
|
|
|
| @@ -64,8 +64,8 @@ rtc_source_set("rtp_receiver") {
|
| deps = [
|
| ":rtp_interfaces",
|
| "..:webrtc_common",
|
| + "../base:rtc_base_approved",
|
| "../modules/rtp_rtcp",
|
| - "../rtc_base:rtc_base_approved",
|
| ]
|
| }
|
|
|
| @@ -76,8 +76,8 @@ rtc_source_set("rtp_sender") {
|
| ]
|
| deps = [
|
| ":rtp_interfaces",
|
| + "../base:rtc_base_approved",
|
| "../modules/congestion_controller",
|
| - "../rtc_base:rtc_base_approved",
|
| ]
|
| }
|
|
|
| @@ -109,6 +109,7 @@ rtc_static_library("call") {
|
| "..:webrtc_common",
|
| "../api:transport_api",
|
| "../audio",
|
| + "../base:rtc_task_queue",
|
| "../logging:rtc_event_log_api",
|
| "../logging:rtc_event_log_impl",
|
| "../modules/bitrate_controller",
|
| @@ -116,7 +117,6 @@ rtc_static_library("call") {
|
| "../modules/pacing",
|
| "../modules/rtp_rtcp",
|
| "../modules/utility",
|
| - "../rtc_base:rtc_task_queue",
|
| "../system_wrappers",
|
| "../video",
|
| ]
|
| @@ -149,6 +149,7 @@ if (rtc_include_tests) {
|
| ":rtp_sender",
|
| "..:webrtc_common",
|
| "../api:mock_audio_mixer",
|
| + "../base:rtc_base_approved",
|
| "../logging:rtc_event_log_api",
|
| "../modules/audio_device:mock_audio_device",
|
| "../modules/audio_mixer",
|
| @@ -157,7 +158,6 @@ if (rtc_include_tests) {
|
| "../modules/pacing",
|
| "../modules/rtp_rtcp",
|
| "../modules/rtp_rtcp:mock_rtp_rtcp",
|
| - "../rtc_base:rtc_base_approved",
|
| "../system_wrappers",
|
| "../test:audio_codec_mocks",
|
| "../test:direct_transport",
|
| @@ -191,11 +191,11 @@ if (rtc_include_tests) {
|
| ":call_interfaces",
|
| "..:webrtc_common",
|
| "../api/audio_codecs:builtin_audio_encoder_factory",
|
| + "../base:rtc_base_approved",
|
| "../logging:rtc_event_log_api",
|
| "../modules/audio_coding",
|
| "../modules/audio_mixer:audio_mixer_impl",
|
| "../modules/rtp_rtcp",
|
| - "../rtc_base:rtc_base_approved",
|
| "../system_wrappers",
|
| "../system_wrappers:metrics_default",
|
| "../test:direct_transport",
|
|
|