Move SrtpSession and tests to their own files.

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 32 matching lines @@
     "mediamonitor.cc",
     "mediamonitor.h",
     "mediasession.cc",
     "mediasession.h",
     "rtcpmuxfilter.cc",
     "rtcpmuxfilter.h",
     "rtptransport.cc",
     "rtptransport.h",
     "srtpfilter.cc",
     "srtpfilter.h",
+    "srtpsession.cc",
+    "srtpsession.h",
     "voicechannel.h",
   ]
 
   deps = [
     "..:webrtc_common",
     "../api:call_api",
     "../api:libjingle_peerconnection_api",
     "../api:ortc_api",
     "../base:rtc_base",
     "../base:rtc_task_queue",
@@ skipping 187 matching lines @@
 
     sources = [
       "bundlefilter_unittest.cc",
       "channel_unittest.cc",
       "channelmanager_unittest.cc",
       "currentspeakermonitor_unittest.cc",
       "mediasession_unittest.cc",
       "rtcpmuxfilter_unittest.cc",
       "rtptransport_unittest.cc",
       "srtpfilter_unittest.cc",
+      "srtpsession_unittest.cc",
+      "srtptestutil.h",
     ]
 
     include_dirs = [ "//third_party/libsrtp/srtp" ]
 
     configs += [ ":rtc_pc_unittests_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 177 matching lines @@
       "//testing/gmock",
     ]
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
 
       shard_timeout = 900
     }
   }
 }