Moved call.h and most of api/call/* into call/

webrtc/call/audio_state.h

Index: webrtc/call/audio_state.h
diff --git a/webrtc/api/call/audio_state.h b/webrtc/call/audio_state.h
similarity index 93%
rename from webrtc/api/call/audio_state.h
rename to webrtc/call/audio_state.h
index b8dca3fb4ec67f9144e69ccb8deb08ad7765b427..2c26a1749b5b62df61cef8aba5d606d23d1b92a9 100644
--- a/webrtc/api/call/audio_state.h
+++ b/webrtc/call/audio_state.h
@@ -7,8 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_API_CALL_AUDIO_STATE_H_
-#define WEBRTC_API_CALL_AUDIO_STATE_H_
+#ifndef WEBRTC_CALL_AUDIO_STATE_H_
+#define WEBRTC_CALL_AUDIO_STATE_H_
 
 #include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/base/refcount.h"
@@ -46,4 +46,4 @@ class AudioState : public rtc::RefCountInterface {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_API_CALL_AUDIO_STATE_H_
+#endif  // WEBRTC_CALL_AUDIO_STATE_H_