Remove remains of webrtc/base

webrtc/audio/BUILD.gn

Index: webrtc/audio/BUILD.gn
diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
index 495875de0364e20f23bfa12487b357b8f6958913..a75d8ef4dcc95542ec2f38411b5a090d20b5124f 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -39,8 +39,6 @@ rtc_static_library("audio") {
     "../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",
@@ -52,6 +50,8 @@ rtc_static_library("audio") {
     "../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",
   ]
@@ -80,14 +80,14 @@ if (rtc_include_tests) {
     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",
@@ -151,8 +151,8 @@ if (rtc_include_tests) {
       "test/audio_bwe_integration_test.h",
     ]
     deps = [
-      "../base:rtc_base_approved",
       "../common_audio",
+      "../rtc_base:rtc_base_approved",
       "../system_wrappers",
       "../test:fake_audio_device",
       "../test:field_trial",