Remove remains of webrtc/base

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 10 matching lines @@
     "syncable.h",
   ]
   deps = [
     ":rtp_interfaces",
     "..:video_stream_api",
     "..:webrtc_common",
     "../api:audio_mixer_api",
     "../api:libjingle_peerconnection_api",
     "../api:transport_api",
     "../api/audio_codecs:audio_codecs_api",
-    "../base:rtc_base",
-    "../base:rtc_base_approved",
+    "../rtc_base:rtc_base",
+    "../rtc_base:rtc_base_approved",
   ]
 }
 
 # TODO(nisse): These RTP targets should be moved elsewhere
 # when interfaces have stabilized.
 rtc_source_set("rtp_interfaces") {
   sources = [
     "rtcp_packet_sink_interface.h",
     "rtp_packet_sink_interface.h",
     "rtp_stream_receiver_controller_interface.h",
     "rtp_transport_controller_send_interface.h",
   ]
   deps = [
-    "../base:rtc_base_approved",
+    "../rtc_base:rtc_base_approved",
   ]
 }
 
 rtc_source_set("rtp_receiver") {
   sources = [
     "rsid_resolution_observer.h",
     "rtcp_demuxer.cc",
     "rtcp_demuxer.h",
     "rtp_demuxer.cc",
     "rtp_demuxer.h",
     "rtp_rtcp_demuxer_helper.cc",
     "rtp_rtcp_demuxer_helper.h",
     "rtp_stream_receiver_controller.cc",
     "rtp_stream_receiver_controller.h",
     "rtx_receive_stream.cc",
     "rtx_receive_stream.h",
   ]
   deps = [
     ":rtp_interfaces",
     "..:webrtc_common",
-    "../base:rtc_base_approved",
     "../modules/rtp_rtcp",
+    "../rtc_base:rtc_base_approved",
   ]
 }
 
 rtc_source_set("rtp_sender") {
   sources = [
     "rtp_transport_controller_send.cc",
     "rtp_transport_controller_send.h",
   ]
   deps = [
     ":rtp_interfaces",
-    "../base:rtc_base_approved",
     "../modules/congestion_controller",
+    "../rtc_base:rtc_base_approved",
   ]
 }
 
 rtc_static_library("call") {
   sources = [
     "bitrate_allocator.cc",
     "call.cc",
     "callfactory.cc",
     "callfactory.h",
     "flexfec_receive_stream_impl.cc",
@@ skipping 11 matching lines @@
   ]
 
   deps = [
     ":call_interfaces",
     ":rtp_interfaces",
     ":rtp_receiver",
     ":rtp_sender",
     "..:webrtc_common",
     "../api:transport_api",
     "../audio",
-    "../base:rtc_task_queue",
     "../logging:rtc_event_log_api",
     "../logging:rtc_event_log_impl",
     "../modules/bitrate_controller",
     "../modules/congestion_controller",
     "../modules/pacing",
     "../modules/rtp_rtcp",
     "../modules/utility",
+    "../rtc_base:rtc_task_queue",
     "../system_wrappers",
     "../video",
   ]
 }
 
 if (rtc_include_tests) {
   rtc_source_set("call_tests") {
     testonly = true
 
     # Skip restricting visibility on mobile platforms since the tests on those
@@ skipping 12 matching lines @@
       "rtp_rtcp_demuxer_helper_unittest.cc",
       "rtx_receive_stream_unittest.cc",
     ]
     deps = [
       ":call",
       ":rtp_interfaces",
       ":rtp_receiver",
       ":rtp_sender",
       "..:webrtc_common",
       "../api:mock_audio_mixer",
-      "../base:rtc_base_approved",
       "../logging:rtc_event_log_api",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer",
       "../modules/bitrate_controller",
       "../modules/congestion_controller:mock_congestion_controller",
       "../modules/pacing",
       "../modules/rtp_rtcp",
       "../modules/rtp_rtcp:mock_rtp_rtcp",
+      "../rtc_base:rtc_base_approved",
       "../system_wrappers",
       "../test:audio_codec_mocks",
       "../test:direct_transport",
       "../test:test_common",
       "../test:test_support",
       "../test:video_test_common",
       "//testing/gmock",
       "//testing/gtest",
     ]
     if (!build_with_chromium && is_clang) {
@@ skipping 13 matching lines @@
     }
     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",
+      "../rtc_base:rtc_base_approved",
       "../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",
     ]
     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" ]
     }
   }
 }