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 1577316cd4b59d6fcd20697d9f3851294b7febe7..2b7d06fcfa988fdea9648938391d138cad8654de 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -37,8 +37,6 @@ 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",
@@ -50,6 +48,8 @@ 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",
   ]
@@ -77,14 +77,14 @@ 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",