Remove remains of webrtc/base

webrtc/media/BUILD.gn

 # Copyright (c) 2016 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("//build/config/linux/pkg_config.gni")
 import("../webrtc.gni")
@@ skipping 27 matching lines @@
     "base/h264_profile_level_id.h",
   ]
 
   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" ]
   }
 
   deps = [
     "..:webrtc_common",
-    "../base:rtc_base",
-    "../base:rtc_base_approved",
+    "../rtc_base:rtc_base",
+    "../rtc_base:rtc_base_approved",
   ]
 }
 
 rtc_static_library("rtc_media_base") {
   # TODO(kjellander): Remove (bugs.webrtc.org/6828)
   # Enabling GN check triggers cyclic dependency error:
   #   //webrtc/media:rtc_media_base ->
   #   //webrtc/pc:rtc_pc_base ->
   #   //webrtc/media:rtc_data ->
   #   //webrtc/media:rtc_media_base
@@ skipping 48 matching lines @@
     ]
   } else {
     # Need to add a directory normally exported by libyuv.
     include_dirs += [ "$rtc_libyuv_dir/include" ]
   }
 
   deps += [
     ":rtc_h264_profile_id",
     "..:webrtc_common",
     "../api:libjingle_peerconnection_api",
-    "../base:rtc_base",
-    "../base:rtc_base_approved",
     "../p2p",
+    "../rtc_base:rtc_base",
+    "../rtc_base:rtc_base_approved",
   ]
 
   if (is_nacl) {
     deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
   }
 }
 
 rtc_static_library("rtc_audio_video") {
   defines = []
   libs = []
@@ skipping 89 matching lines @@
     "..:video_stream_api",
     "..:webrtc_common",
     "../api:call_api",
     "../api:libjingle_peerconnection_api",
     "../api:transport_api",
     "../api:video_frame_api",
     "../api/audio_codecs:audio_codecs_api",
     "../api/audio_codecs:builtin_audio_decoder_factory",
     "../api/audio_codecs:builtin_audio_encoder_factory",
     "../api/video_codecs:video_codecs_api",
-    "../base:rtc_base",
-    "../base:rtc_base_approved",
-    "../base:rtc_task_queue",
-    "../base:sequenced_task_checker",
     "../call",
     "../common_video:common_video",
     "../modules/audio_coding:rent_a_codec",
     "../modules/audio_device:audio_device",
     "../modules/audio_mixer:audio_mixer_impl",
     "../modules/audio_processing:audio_processing",
     "../modules/audio_processing/aec_dump",
     "../modules/video_capture:video_capture_module",
     "../modules/video_coding",
     "../modules/video_coding:webrtc_h264",
     "../modules/video_coding:webrtc_vp8",
     "../modules/video_coding:webrtc_vp9",
     "../p2p:rtc_p2p",
     "../pc:rtc_pc_base",
+    "../rtc_base:rtc_base",
+    "../rtc_base:rtc_base_approved",
+    "../rtc_base:rtc_task_queue",
+    "../rtc_base:sequenced_task_checker",
     "../system_wrappers",
     "../video",
     "../voice_engine",
   ]
 }
 
 rtc_static_library("rtc_data") {
   defines = []
   deps = []
 
@@ skipping 27 matching lines @@
       "//third_party/usrsctp/usrsctplib",
     ]
     deps += [ "//third_party/usrsctp" ]
   }
 
   deps += [
     ":rtc_media_base",
     "..:webrtc_common",
     "../api:call_api",
     "../api:transport_api",
-    "../base:rtc_base",
-    "../base:rtc_base_approved",
     "../p2p:rtc_p2p",
+    "../rtc_base:rtc_base",
+    "../rtc_base:rtc_base_approved",
     "../system_wrappers",
   ]
 }
 
 rtc_source_set("rtc_media") {
   public_deps = [
     ":rtc_audio_video",
     ":rtc_data",
   ]
 }
@@ skipping 53 matching lines @@
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
 
     deps += [
       ":rtc_media",
       ":rtc_media_base",
       "..:webrtc_common",
       "../api:call_api",
       "../api:video_frame_api",
       "../api/video_codecs:video_codecs_api",
-      "../base:rtc_base",
-      "../base:rtc_base_approved",
-      "../base:rtc_base_tests_utils",
       "../call:call_interfaces",
+      "../rtc_base:rtc_base",
+      "../rtc_base:rtc_base_approved",
+      "../rtc_base:rtc_base_tests_utils",
       "../test:test_support",
       "//testing/gtest",
     ]
     public_deps += [ "//testing/gmock" ]
   }
 
   config("rtc_media_unittests_config") {
     # GN orders flags on a target before flags from configs. The default config
     # adds -Wall, and this flag have to be after -Wall -- so they need to
     # come from a config and can"t be on the target directly.
@@ skipping 116 matching lines @@
 
     deps += [
       ":rtc_media",
       ":rtc_media_base",
       ":rtc_media_tests_utils",
       "../api:video_frame_api",
       "../api/audio_codecs:builtin_audio_decoder_factory",
       "../api/audio_codecs:builtin_audio_encoder_factory",
       "../api/video_codecs:video_codecs_api",
       "../audio",
-      "../base:rtc_base",
-      "../base:rtc_base_approved",
-      "../base:rtc_base_tests_main",
-      "../base:rtc_base_tests_utils",
       "../call:call_interfaces",
       "../common_video:common_video",
       "../logging:rtc_event_log_api",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_processing:audio_processing",
       "../modules/video_coding:simulcast_test_utility",
       "../modules/video_coding:video_coding_utility",
       "../modules/video_coding:webrtc_vp8",
       "../p2p:p2p_test_utils",
+      "../rtc_base:rtc_base",
+      "../rtc_base:rtc_base_approved",
+      "../rtc_base:rtc_base_tests_main",
+      "../rtc_base:rtc_base_tests_utils",
       "../system_wrappers:metrics_default",
       "../test:audio_codec_mocks",
       "../test:test_support",
       "../voice_engine:voice_engine",
     ]
   }
 }