Video/Screenshare loopback tool.

webrtc/video/BUILD.gn

 # Copyright (c) 2014 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 161 matching lines @@
       "//testing/gmock",
       "//testing/gtest",
       "//third_party/gflags",
     ]
     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" ]
     }
   }
 
+  rtc_executable("vs_loopback") {
+    testonly = true
+    sources = [
+      "vs_loopback.cc",
+    ]
+    deps = [
+      ":video_quality_test",
+      "../rtc_base:rtc_base_approved",
+      "../system_wrappers:metrics_default",
+      "../test:field_trial",
+      "../test:run_test",
+      "../test:test_common",
+      "../test:test_renderer",
+      "../test:test_support",
+      "//testing/gmock",
+      "//testing/gtest",
+      "//third_party/gflags",
+    ]
+    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" ]
+    }
+  }
+
   rtc_executable("screenshare_loopback") {
     testonly = true
     sources = [
       "screenshare_loopback.cc",
     ]
 
     deps = [
       ":video_quality_test",
       "../rtc_base:rtc_base_approved",
       "../system_wrappers:metrics_default",
@@ skipping 113 matching lines @@
     ]
     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" ]
     }
     if (rtc_use_h264) {
       defines += [ "WEBRTC_USE_H264" ]
     }
   }
 }