Only one implementation of MockRtpPacketSink once

webrtc/call/BUILD.gn

Index: webrtc/call/BUILD.gn
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index c40b557d4cda89b7ff9a53e4fc727d32962002b9..2f0483092a076f34f0395033d2a8db861b5f9308 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -33,8 +33,15 @@ rtc_source_set("call_interfaces") {
   ]
 }
 
+# TODO(eladalon): This should be moved, as with the TODO for |rtp_interfaces|.
+rtc_source_set("mock_rtp_interfaces") {

[danilchap, 2017/07/26 10:28:59]

probably better to move inside if (rtc_include_tests) section below

[eladalon, 2017/07/26 11:24:27]

Done.

+  sources = [

[danilchap, 2017/07/26 10:28:59]

add
testonly = true

[eladalon, 2017/07/26 11:24:27]

Done.

+    "test/mock_rtp_packet_sink_interface.h",
+  ]

[danilchap, 2017/07/26 10:28:59]

add deps on rtp_interfaces and //testing/gmock
It is header-only dependencies, but still nicer to have direct dependencies
explicit.

+}
+
 # TODO(nisse): These RTP targets should be moved elsewhere
-# when interfaces have stabilized.
+# when interfaces have stabilized. See also TODO for |mock_rtp_interfaces|.
 rtc_source_set("rtp_interfaces") {
   sources = [
     "rtcp_packet_sink_interface.h",
@@ -144,6 +151,7 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":call",
+      ":mock_rtp_interfaces",

[eladalon, 2017/07/26 09:01:05]

I find it odd that this didn't actually appear to be necessary.
1. Any idea why?
2. Shall we keep it, then?
3. rtp_video_stream_receiver_unittest.cc does not seem to depend on
mock_rtp_interfaces, which I also find odd.

[danilchap, 2017/07/26 10:28:59]

Because we currently do not check that only headers from the dependent targets
are included and there is no code to link, so no linker error either.

It would be nice to eventually turn that feature on for stricter dependency
graph,
so yes, better keep it.

and better make other unittests that depend on the mock to explicitly depend on
this mock target too.

[eladalon, 2017/07/26 11:24:27]

Done.

       ":rtp_interfaces",
       ":rtp_receiver",
       ":rtp_sender",