Index: webrtc/modules/audio_device/BUILD.gn |
diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn |
index c56516519245a3a90c6dab71674ccc04702e2c17..2c262ba9629f4b34cc292a76bc674158544e113a 100644 |
--- a/webrtc/modules/audio_device/BUILD.gn |
+++ b/webrtc/modules/audio_device/BUILD.gn |
@@ -273,7 +273,7 @@ if (rtc_include_tests) { |
# gets additional generated targets which would require many lines here to |
# cover (which would be confusing to read and hard to maintain). |
if (!is_android && !is_ios) { |
- visibility = [ "//webrtc/modules:modules_unittests" ] |
+ visibility = [ "..:modules_unittests" ] |
} |
sources = [ |
"fine_audio_buffer_unittest.cc", |
@@ -304,7 +304,7 @@ if (rtc_include_tests) { |
] |
deps += [ |
"../../../base", |
- "//webrtc/sdk/android:libjingle_peerconnection_java", |
+ "../../sdk/android:libjingle_peerconnection_java", |
] |
} |
if (is_ios && !use_ios_simulator) { |
@@ -355,7 +355,7 @@ if (!build_with_chromium && is_android) { |
"android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java", |
] |
deps = [ |
- "//webrtc/rtc_base:base_java", |
+ "../../rtc_base:base_java", |
] |
} |
} |