Improve testing of SRTP external auth code paths.

webrtc/pc/BUILD.gn

 # Copyright (c) 2016 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")
 if (is_android) {
@@ skipping 20 matching lines @@
     "audiomonitor.cc",
     "audiomonitor.h",
     "bundlefilter.cc",
     "bundlefilter.h",
     "channel.cc",
     "channel.h",
     "channelmanager.cc",
     "channelmanager.h",
     "currentspeakermonitor.cc",
     "currentspeakermonitor.h",
+    "externalhmac.cc",
+    "externalhmac.h",
     "mediamonitor.cc",
     "mediamonitor.h",
     "mediasession.cc",
     "mediasession.h",
     "rtcpmuxfilter.cc",
     "rtcpmuxfilter.h",
     "srtpfilter.cc",
     "srtpfilter.h",
     "voicechannel.h",
   ]
 
   deps = [
     "../api:call_api",
     "../base:rtc_base",
     "../media",
   ]
 
-  if (rtc_enable_external_auth) {
-    sources += [
-      "externalhmac.cc",
-      "externalhmac.h",
-    ]
-  }
   if (rtc_build_libsrtp) {
     deps += [ "//third_party/libsrtp" ]
   }
 
   public_configs = [ ":rtc_pc_config" ]
 
   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" ]
   }
@@ skipping 287 matching lines @@
       "//testing/gmock",
     ]
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
 
       shard_timeout = 900
     }
   }
 }