| Index: webrtc/voice_engine/BUILD.gn
|
| diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
|
| index 8450d1dce3094efbad4d8e7c09fa6228d70e1420..dd5546be134681427eabe9f712e82e5d61b006a9 100644
|
| --- a/webrtc/voice_engine/BUILD.gn
|
| +++ b/webrtc/voice_engine/BUILD.gn
|
| @@ -92,8 +92,8 @@ rtc_static_library("voice_engine") {
|
| "..:webrtc_common",
|
| "../api:call_api",
|
| "../base:rtc_base_approved",
|
| - "../call:rtc_event_log",
|
| "../common_audio",
|
| + "../logging:rtc_event_log_api",
|
| "../modules/audio_conference_mixer",
|
| "../modules/audio_device",
|
| "../modules/audio_processing",
|
| @@ -244,7 +244,7 @@ if (rtc_include_tests) {
|
| ":voice_engine",
|
| "//testing/gtest",
|
| "//third_party/gflags",
|
| - "//webrtc/call:rtc_event_log",
|
| + "//webrtc/logging:rtc_event_log_api",
|
| "//webrtc/system_wrappers",
|
| "//webrtc/system_wrappers:system_wrappers_default",
|
| "//webrtc/test:test_support",
|
| @@ -271,7 +271,7 @@ if (rtc_include_tests) {
|
| "//testing/gmock",
|
| "//testing/gtest",
|
| "//third_party/gflags",
|
| - "//webrtc/call:rtc_event_log",
|
| + "//webrtc/logging:rtc_event_log_api",
|
| "//webrtc/modules/video_capture",
|
| "//webrtc/system_wrappers",
|
| "//webrtc/system_wrappers/:system_wrappers_default",
|
|
|