| 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",
|
| ]
|
| }
|
| }
|
|
|