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

webrtc/call/BUILD.gn

Index: webrtc/call/BUILD.gn
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 64829e9b9d43e8ff07c023523b6ca65bf3449210..5572941f30efeec1f7d8497b333ef4d24ec7f3de 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -8,6 +8,16 @@
 
 import("../build/webrtc.gni")
 
+rtc_source_set("call_interfaces") {
+  sources = [
+    "audio_receive_stream.h",
+    "audio_send_stream.cc",
+    "audio_send_stream.h",
+    "audio_state.h",
+    "call.h",
+  ]
+}
+
 rtc_static_library("call") {
   sources = [
     "bitrate_allocator.cc",
@@ -22,10 +32,12 @@ rtc_static_library("call") {
   }
 
   public_deps = [
+    ":call_interfaces",
     "../api:call_api",
   ]
 
   deps = [
+    ":call_interfaces",
     "..:webrtc_common",
     "../api:transport_api",
     "../audio",