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..471ee07b6a66a4f9a996bdc991ccd23572cebd0a 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -167,10 +167,10 @@ config("common_config") {
 
 source_set("webrtc") {
   sources = [
+    "audio_send_stream.h",

[kjellander_webrtc, 2016/02/25 11:12:06]

These were missing from any .gyp* file.

+    "audio_state.h",
     "call.h",
-    "config.h",

[kjellander_webrtc, 2016/02/25 11:12:05]

moved to common.gyp:webrtc_common since they only depend on other headers in
there (of each other).

-    "frame_callback.h",
-    "transport.h",
+    "video_frame.h",
   ]
 
   defines = []
@@ -228,12 +228,19 @@ if (!build_with_chromium) {
 
 source_set("webrtc_common") {
   sources = [
+    "audio_receive_stream.h",

[kjellander_webrtc, 2016/02/25 11:12:06]

Maybe it's confusing to have audio_recieve_stream.h here and audio_send_stream.h
in the 'webrtc' target? 
I based it only on what they include. This one can be here since it only
includes headers in this target. The other one can't since it includes
"webrtc/base/scoped_ptr.h" and 
"webrtc/modules/audio_coding/codecs/audio_encoder.h"

+    "audio_sink.h",
     "common_types.cc",
     "common_types.h",
     "config.cc",
     "config.h",
     "engine_configurations.h",
+    "frame_callback.h",
+    "transport.h",
     "typedefs.h",
+    "video_receive_stream.h",
+    "video_renderer.h",
+    "video_send_stream.h",
   ]
 
   configs += [ ":common_config" ]