Remove remains of webrtc/base

webrtc/audio/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")
 if (is_android) {
@@ skipping 19 matching lines @@
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 
   deps = [
     "..:webrtc_common",
     "../api:audio_mixer_api",
     "../api:call_api",
     "../api/audio_codecs:audio_codecs_api",
     "../api/audio_codecs:builtin_audio_encoder_factory",
-    "../base:rtc_base_approved",
-    "../base:rtc_task_queue",
     "../call:call_interfaces",
     "../call:rtp_interfaces",
     "../common_audio",
     "../modules/audio_coding:cng",
     "../modules/audio_device",
     "../modules/audio_processing",
     "../modules/bitrate_controller:bitrate_controller",
     "../modules/congestion_controller:congestion_controller",
     "../modules/pacing:pacing",
     "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
     "../modules/rtp_rtcp:rtp_rtcp",
+    "../rtc_base:rtc_base_approved",
+    "../rtc_base:rtc_task_queue",
     "../system_wrappers",
     "../voice_engine",
   ]
 }
 if (rtc_include_tests) {
   rtc_source_set("audio_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" ]
     }
 
     # TODO(kjellander): Remove (bugs.webrtc.org/6828)
     # This needs remote_bitrate_estimator to be moved to webrtc/api first.
     check_includes = false
 
     sources = [
       "audio_receive_stream_unittest.cc",
       "audio_send_stream_unittest.cc",
       "audio_state_unittest.cc",
     ]
     deps = [
       ":audio",
       "../api:mock_audio_mixer",
-      "../base:rtc_base_approved",
-      "../base:rtc_task_queue",
       "../call:rtp_receiver",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer:audio_mixer_impl",
       "../modules/congestion_controller:congestion_controller",
       "../modules/congestion_controller:mock_congestion_controller",
       "../modules/pacing:pacing",
+      "../rtc_base:rtc_base_approved",
+      "../rtc_base:rtc_task_queue",
       "../test:test_common",
       "../test:test_support",
       "utility:utility_tests",
       "//testing/gmock",
       "//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" ]
@@ skipping 26 matching lines @@
         "//resources/voice_engine/audio_tiny48.wav",
       ]
 
       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" ]
       }
     }
   }
 }