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", |