Refactor webrtc/{api,audio} and modules/audio_coding for GN check

webrtc/audio/BUILD.gn

Index: webrtc/audio/BUILD.gn
diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
index bfc2372dd6414f3042efee728c6639eca3106df8..0669c603c024256d5e6fcec9dbc251fad0d50da2 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -30,11 +30,17 @@ rtc_static_library("audio") {
   deps = [
     "..:webrtc_common",
     "../api:audio_mixer_api",
+    "../api:call_api",
     "../base:rtc_base_approved",
+    "../base:rtc_task_queue",
     "../call:call_interfaces",
     "../common_audio",
     "../modules/audio_device",
     "../modules/audio_processing",
+    "../modules/congestion_controller:congestion_controller",
+    "../modules/pacing:pacing",
+    "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
+    "../modules/rtp_rtcp:rtp_rtcp",
     "../system_wrappers",
     "../voice_engine",
   ]
@@ -42,6 +48,11 @@ rtc_static_library("audio") {
 if (rtc_include_tests) {
   rtc_source_set("audio_tests") {
     testonly = true
+
+    # 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",
@@ -52,8 +63,11 @@ if (rtc_include_tests) {
       ":audio",
       "../api:mock_audio_mixer",
       "../base:rtc_base_approved",
+      "../base:rtc_task_queue",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer:audio_mixer_impl",
+      "../modules/congestion_controller:congestion_controller",
+      "../modules/pacing:pacing",
       "../test:test_common",
       "../test:test_support",
       "utility:audio_frame_operations",