Only one implementation of MockRtpPacketSink once

webrtc/call/BUILD.gn

 # Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 import("../webrtc.gni")
 
@@ skipping 15 matching lines @@
     "..:webrtc_common",
     "../api:audio_mixer_api",
     "../api:libjingle_peerconnection_api",
     "../api:transport_api",
     "../api/audio_codecs:audio_codecs_api",
     "../rtc_base:rtc_base",
     "../rtc_base:rtc_base_approved",
   ]
 }
 
+# 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",
     "rtp_packet_sink_interface.h",
     "rtp_stream_receiver_controller_interface.h",
     "rtp_transport_controller_send_interface.h",
   ]
   deps = [
     "../rtc_base:rtc_base_approved",
   ]
@@ skipping 89 matching lines @@
       "bitrate_estimator_tests.cc",
       "call_unittest.cc",
       "flexfec_receive_stream_unittest.cc",
       "rtcp_demuxer_unittest.cc",
       "rtp_demuxer_unittest.cc",
       "rtp_rtcp_demuxer_helper_unittest.cc",
       "rtx_receive_stream_unittest.cc",
     ]
     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",
       "..:webrtc_common",
       "../api:mock_audio_mixer",
       "../logging:rtc_event_log_api",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer",
       "../modules/bitrate_controller",
       "../modules/congestion_controller:mock_congestion_controller",
@@ skipping 50 matching lines @@
       "../video",
       "../voice_engine",
       "//testing/gtest",
     ]
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 }