Use relative paths in GN files.

webrtc/call/BUILD.gn

 # Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 import("../webrtc.gni")
 
@@ skipping 112 matching lines @@
 }
 
 if (rtc_include_tests) {
   rtc_source_set("call_tests") {
     testonly = true
 
     # Skip restricting visibility on mobile platforms since the tests on those
     # 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:video_engine_tests" ]
+      visibility = [ "..:video_engine_tests" ]
     }
     sources = [
       "bitrate_allocator_unittest.cc",
       "bitrate_estimator_tests.cc",
       "call_unittest.cc",
       "flexfec_receive_stream_unittest.cc",
       "rtcp_demuxer_unittest.cc",
       "rtp_demuxer_unittest.cc",
       "rtp_rtcp_demuxer_helper_unittest.cc",
       "rtx_receive_stream_unittest.cc",
@@ skipping 29 matching lines @@
     }
   }
 
   rtc_source_set("call_perf_tests") {
     testonly = true
 
     # Skip restricting visibility on mobile platforms since the tests on those
     # 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:webrtc_perf_tests" ]
+      visibility = [ "..:webrtc_perf_tests" ]
     }
     sources = [
       "call_perf_tests.cc",
       "rampup_tests.cc",
       "rampup_tests.h",
     ]
     deps = [
       ":call_interfaces",
       "..:webrtc_common",
       "../api/audio_codecs:builtin_audio_encoder_factory",
       "../base:rtc_base_approved",
       "../logging:rtc_event_log_api",
       "../modules/audio_coding",
       "../modules/audio_mixer:audio_mixer_impl",
       "../modules/rtp_rtcp",
       "../system_wrappers",
       "../system_wrappers:metrics_default",
       "../test:direct_transport",
       "../test:fake_audio_device",
+      "../test:field_trial",
+      "../test:test_common",
       "../test:test_support",
       "../test:video_test_common",
       "../video",
       "../voice_engine",
       "//testing/gtest",
-      "//webrtc/test:field_trial",
-      "//webrtc/test:test_common",
     ]
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 }