Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback

webrtc/video/BUILD.gn

Index: webrtc/video/BUILD.gn
diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn
index e6f63f25eb7174213a8c1ca04d297613dfe5e3d7..430b00de6a05d93a125bccad63414944aab33ee2 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -112,6 +112,9 @@ if (rtc_include_tests) {
       "../test:video_test_common",
       "../voice_engine",
       "//testing/gtest",
+      "//webrtc/test:rtp_test_utils",
+      "//webrtc/test:test_renderer",
+      "//webrtc/test:video_test_common",
     ]
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).