Index: webrtc/voice_engine/BUILD.gn |
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn |
index 78c92f6ec64f647bb6bc36faa7bc43d01f4e2673..e16b1762a78e3cd370030e0f086de2899e967eaf 100644 |
--- a/webrtc/voice_engine/BUILD.gn |
+++ b/webrtc/voice_engine/BUILD.gn |
@@ -37,10 +37,10 @@ rtc_static_library("file_player") { |
deps = [ |
":audio_coder", |
"..:webrtc_common", |
+ "../base:rtc_base_approved", |
"../common_audio", |
"../modules:module_api", |
"../modules/media_file", |
- "../rtc_base:rtc_base_approved", |
] |
if (!build_with_chromium && is_clang) { |
@@ -58,10 +58,10 @@ rtc_static_library("file_recorder") { |
":audio_coder", |
"..:webrtc_common", |
"../audio/utility:audio_frame_operations", |
+ "../base:rtc_base_approved", |
"../common_audio", |
"../modules:module_api", |
"../modules/media_file:media_file", |
- "../rtc_base:rtc_base_approved", |
"../system_wrappers", |
] |
@@ -143,6 +143,8 @@ rtc_static_library("voice_engine") { |
"../api/audio_codecs:builtin_audio_decoder_factory", |
"../api/audio_codecs:builtin_audio_encoder_factory", |
"../audio/utility:audio_frame_operations", |
+ "../base:rtc_base_approved", |
+ "../base:rtc_task_queue", |
"../call:rtp_interfaces", |
"../common_audio", |
"../logging:rtc_event_log_api", |
@@ -157,8 +159,6 @@ rtc_static_library("voice_engine") { |
"../modules/pacing", |
"../modules/rtp_rtcp", |
"../modules/utility", |
- "../rtc_base:rtc_base_approved", |
- "../rtc_base:rtc_task_queue", |
"../system_wrappers", |
] |
} |
@@ -171,9 +171,9 @@ rtc_static_library("audio_level") { |
deps = [ |
"..:webrtc_common", |
+ "../base:rtc_base_approved", |
"../common_audio", |
"../modules:module_api", |
- "../rtc_base:rtc_base_approved", |
] |
} |
@@ -182,9 +182,9 @@ if (rtc_include_tests) { |
deps = [ |
":file_player", |
":voice_engine", |
+ "../base:rtc_base_approved", |
+ "../base:rtc_base_tests_utils", |
"../modules:module_api", |
- "../rtc_base:rtc_base_approved", |
- "../rtc_base:rtc_base_tests_utils", |
"../test:test_common", |
"//testing/gmock", |
"//testing/gtest", |
@@ -247,11 +247,11 @@ if (rtc_include_tests) { |
deps = [ |
":voice_engine", |
"..:webrtc_common", |
+ "../base:rtc_base_approved", |
"../modules:module_api", |
"../modules/audio_device:audio_device", |
"../modules/audio_processing:audio_processing", |
"../modules/rtp_rtcp:rtp_rtcp", |
- "../rtc_base:rtc_base_approved", |
"//testing/gmock", |
"//testing/gtest", |
"//third_party/gflags", |