Move video send/receive stream headers to webrtc/call.

webrtc/call/BUILD.gn

Index: webrtc/call/BUILD.gn
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index afbc8228493ccac8da2c9ba6764c9c12760a8ef7..62bcff3385e9beca69b8306485e9958352b03ca8 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -22,7 +22,7 @@ rtc_source_set("call_interfaces") {
   ]
   deps = [
     ":rtp_interfaces",
-    "..:video_stream_api",
+    ":video_stream_api",
     "..:webrtc_common",
     "../api:audio_mixer_api",
     "../api:libjingle_peerconnection_api",
@@ -124,6 +124,21 @@ rtc_static_library("call") {
   ]
 }
 
+rtc_source_set("video_stream_api") {
+  sources = [
+    "video_receive_stream.cc",
+    "video_receive_stream.h",
+    "video_send_stream.cc",
+    "video_send_stream.h",
+  ]
+  deps = [
+    "../:webrtc_common",
+    "../api:transport_api",
+    "../common_video:common_video",
+    "../rtc_base:rtc_base_approved",
+  ]
+}
+
 if (rtc_include_tests) {
   rtc_source_set("call_tests") {
     testonly = true