Create unit tests for RtpDemuxer

webrtc/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.
 
 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
 
@@ skipping 417 matching lines @@
         "{{bundle_resources_dir}}/{{source_file_part}}",
       ]
     }
   }
 
   rtc_test("video_engine_tests") {
     testonly = true
     deps = [
       "audio:audio_tests",
       "base:rtc_base_tests_utils",
+
+      # TODO(eladalon): call_tests aren't actually video-specific, so we
+      # should move them to a more appropriate test suite.
       "call:call_tests",
       "modules/video_capture",
       "test:test_common",
       "test:test_main",
       "test:video_test_common",
       "video:video_tests",
     ]
     data = video_engine_tests_resources
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
@@ skipping 78 matching lines @@
       ]
 
       deps = [
         "//base:base_java_test_support",
         "//webrtc/examples:AppRTCMobile_javalib",
         "//webrtc/sdk/android:libjingle_peerconnection_java",
       ]
     }
   }
 }