Index: webrtc/api/BUILD.gn |
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn |
index e87509bced713fdb371bff6921f8a3370651cf84..7c438086539105cfbbeeaac7cb91f91f084f5e54 100644 |
--- a/webrtc/api/BUILD.gn |
+++ b/webrtc/api/BUILD.gn |
@@ -28,7 +28,7 @@ rtc_source_set("call_api") { |
":audio_mixer_api", |
":transport_api", |
"..:webrtc_common", |
- "../base:rtc_base_approved", |
+ "../rtc_base:rtc_base_approved", |
"audio_codecs:audio_codecs_api", |
] |
} |
@@ -83,8 +83,8 @@ rtc_static_library("libjingle_peerconnection_api") { |
deps = [ |
":rtc_stats_api", |
"..:webrtc_common", |
- "../base:rtc_base", |
- "../base:rtc_base_approved", |
+ "../rtc_base:rtc_base", |
+ "../rtc_base:rtc_base_approved", |
"audio_codecs:audio_codecs_api", |
] |
@@ -143,7 +143,7 @@ rtc_source_set("rtc_stats_api") { |
] |
deps = [ |
- "../base:rtc_base_approved", |
+ "../rtc_base:rtc_base_approved", |
] |
} |
@@ -153,8 +153,8 @@ rtc_source_set("audio_mixer_api") { |
] |
deps = [ |
- "../base:rtc_base_approved", |
"../modules:module_api", |
+ "../rtc_base:rtc_base_approved", |
] |
} |
@@ -178,7 +178,7 @@ rtc_source_set("video_frame_api") { |
] |
deps = [ |
- "../base:rtc_base_approved", |
+ "../rtc_base:rtc_base_approved", |
"../system_wrappers", |
] |
@@ -206,7 +206,7 @@ rtc_source_set("libjingle_peerconnection_test_api") { |
] |
deps = [ |
- "../base:rtc_base_approved", |
+ "../rtc_base:rtc_base_approved", |
] |
} |
@@ -235,7 +235,7 @@ if (rtc_include_tests) { |
] |
deps = [ |
":libjingle_peerconnection_api", |
- "../base:rtc_base_approved", |
+ "../rtc_base:rtc_base_approved", |
] |
if (!build_with_chromium && is_clang) { |
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |