Move webrtc/audio_*.h to webrtc/api/call

webrtc/api/BUILD.gn

Index: webrtc/api/BUILD.gn
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index ecef934a3e610e6b3db3071ae165bdfc149b08d3..ba029078c56ae10275613250e0dbc296115e781c 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -14,6 +14,25 @@ group("api") {
   ]
 }
 
+source_set("audio_api") {
+  sources = [
+    "audio/audio_receive_stream.h",
+    "audio/audio_send_stream.h",
+    "audio/audio_send_stream.h",

[the sun, 2016/06/14 09:40:27]

!!

[kjellander_webrtc, 2016/06/14 20:17:46]

Whops, I only fixed api.gyp apparently.

+    "audio/audio_state.h",
+  ]
+
+  configs += [ "..:common_config" ]
+  public_configs = [ "..:common_inherited_config" ]
+
+  deps = [
+    # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
+    "..:webrtc_common",
+    "../base:rtc_base_approved",
+    "../modules/audio_coding:audio_encoder_interface",
+  ]
+}
+
 config("libjingle_peerconnection_warnings_config") {
   # GN orders flags on a target before flags from configs. The default config
   # adds these flags so to cancel them out they need to come from a config and
@@ -116,6 +135,7 @@ source_set("libjingle_peerconnection") {
   }
 
   deps = [
+    ":audio_api",
     "../call",
     "../media",
     "../pc",