Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.

webrtc/BUILD.gn

Index: webrtc/BUILD.gn
diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index 6d84c504a38e3158e35e731d9d584e37e0afa340..1517d85c6b492654f633bf1715b6c2d09aa1e5a4 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -167,10 +167,12 @@ config("common_config") {
 
 source_set("webrtc") {
   sources = [
+    "audio_send_stream.h",
+    "audio_state.h",
     "call.h",
-    "config.h",
-    "frame_callback.h",
-    "transport.h",
+    "video_decoder.h",
+    "video_encoder.h",
+    "video_frame.h",
   ]
 
   defines = []
@@ -228,12 +230,20 @@ if (!build_with_chromium) {
 
 source_set("webrtc_common") {
   sources = [
+    "audio_receive_stream.h",
+    "audio_sink.h",
     "common_types.cc",
     "common_types.h",
     "config.cc",
     "config.h",
     "engine_configurations.h",
+    "frame_callback.h",
+    "stream.h",
+    "transport.h",
     "typedefs.h",
+    "video_receive_stream.h",
+    "video_renderer.h",
+    "video_send_stream.h",
   ]
 
   configs += [ ":common_config" ]