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 16 matching lines @@
     "../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(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",
       ":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 49 matching lines @@
       "../test:video_test_common",
       "../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" ]
     }
   }
+
+  # TODO(eladalon): This should be moved, as with the TODO for |rtp_interfaces|.
+  rtc_source_set("mock_rtp_interfaces") {
+    testonly = true
+
+    sources = [
+      "test/mock_rtp_packet_sink_interface.h",
+    ]
+    deps = [
+      ":rtp_interfaces",
+      "//testing/gmock",
+    ]
+  }
 }