Remove remains of webrtc/base

webrtc/BUILD.gn

 # Copyright (c) 2014 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.
 
 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
 
@@ skipping 214 matching lines @@
 }
 
 rtc_source_set("video_stream_api") {
   sources = [
     "video_receive_stream.h",
     "video_send_stream.h",
   ]
   deps = [
     ":webrtc_common",
     "api:transport_api",
-    "base:rtc_base_approved",
     "common_video:common_video",
+    "rtc_base:rtc_base_approved",
   ]
 }
 
 if (!build_with_chromium) {
   # Target to build all the WebRTC production code.
   rtc_static_library("webrtc") {
     # Only the root target should depend on this.
     visibility = [ "//:default" ]
 
     sources = []
     complete_static_lib = true
     defines = []
 
     deps = [
       ":webrtc_common",
       "api",
       "api:transport_api",
       "audio",
-      "base",
       "call",
       "common_audio",
       "common_video",
       "logging",
       "media",
       "modules",
       "modules/video_capture:video_capture_internal_impl",
       "ortc",
       "p2p",
       "pc",
@@ skipping 18 matching lines @@
       testonly = true
 
       # Only the root target should depend on this.
       visibility = [ "//:default" ]
 
       deps = [
         ":rtc_unittests",
         ":video_engine_tests",
         ":webrtc_nonparallel_tests",
         ":webrtc_perf_tests",
-        "base:rtc_base_tests_utils",
         "common_audio:common_audio_unittests",
         "common_video:common_video_unittests",
         "media:rtc_media_unittests",
         "modules:modules_tests",
         "modules:modules_unittests",
         "modules/audio_coding:audio_coding_tests",
         "modules/audio_processing:audio_processing_tests",
         "modules/remote_bitrate_estimator:bwe_simulations_tests",
         "modules/rtp_rtcp:test_packet_masks_metrics",
         "modules/video_capture:video_capture_internal_impl",
         "modules/video_coding:plot_videoprocessor_integrationtest",
         "ortc:ortc_unittests",
         "pc:peerconnection_unittests",
         "pc:rtc_pc_unittests",
+        "rtc_base:rtc_base_tests_utils",
         "stats:rtc_stats_unittests",
         "system_wrappers:system_wrappers_unittests",
         "test",
         "video:screenshare_loopback",
         "video:video_loopback",
         "voice_engine:voice_engine_unittests",
       ]
       if (is_android) {
         deps += [
           ":android_junit_tests",
@@ skipping 67 matching lines @@
   rtc_test("rtc_unittests") {
     testonly = true
     sources = [
       "config_unittest.cc",
     ]
 
     deps = [
       ":webrtc_common",
       "api:rtc_api_unittests",
       "api/audio_codecs/test:audio_codecs_api_unittests",
-      "base:rtc_base_approved_unittests",
-      "base:rtc_base_tests_main",
-      "base:rtc_base_tests_utils",
-      "base:rtc_base_unittests",
-      "base:rtc_numerics_unittests",
-      "base:rtc_task_queue_unittests",
-      "base:sequenced_task_checker_unittests",
-      "base:weak_ptr_unittests",
       "p2p:libstunprober_unittests",
       "p2p:rtc_p2p_unittests",
+      "rtc_base:rtc_base_approved_unittests",
+      "rtc_base:rtc_base_tests_main",
+      "rtc_base:rtc_base_tests_utils",
+      "rtc_base:rtc_base_unittests",
+      "rtc_base:rtc_numerics_unittests",
+      "rtc_base:rtc_task_queue_unittests",
+      "rtc_base:sequenced_task_checker_unittests",
+      "rtc_base:weak_ptr_unittests",
       "system_wrappers:metrics_default",
     ]
 
     if (rtc_enable_protobuf) {
       deps += [ "logging:rtc_event_log_tests" ]
     }
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
       shard_timeout = 900
@@ skipping 17 matching lines @@
       outputs = [
         "{{bundle_resources_dir}}/{{source_file_part}}",
       ]
     }
   }
 
   rtc_test("video_engine_tests") {
     testonly = true
     deps = [
       "audio:audio_tests",
-      "base:rtc_base_tests_utils",
 
       # TODO(eladalon): call_tests aren't actually video-specific, so we
       # should move them to a more appropriate test suite.
       "call:call_tests",
       "modules/video_capture",
+      "rtc_base:rtc_base_tests_utils",
       "test:test_common",
       "test:test_main",
       "test:video_test_common",
       "video:video_tests",
     ]
     data = video_engine_tests_resources
     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" ]
     }
@@ skipping 51 matching lines @@
       shard_timeout = 2700
     }
     if (is_ios) {
       deps += [ ":webrtc_perf_tests_bundle_data" ]
     }
   }
 
   rtc_test("webrtc_nonparallel_tests") {
     testonly = true
     deps = [
-      "base:rtc_base_nonparallel_tests",
+      "rtc_base:rtc_base_nonparallel_tests",
     ]
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
       shard_timeout = 900
     }
   }
 
   if (is_android) {
     junit_binary("android_junit_tests") {
       java_files = [
         "examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java",
         "examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java",
         "examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java",
         "sdk/android/tests/src/org/webrtc/CameraEnumerationTest.java",
       ]
 
       deps = [
         "//base:base_java_test_support",
         "//webrtc/examples:AppRTCMobile_javalib",
         "//webrtc/sdk/android:libjingle_peerconnection_java",
       ]
     }
   }
 }