Introduce RtpTransportInternal and SrtpTransport.

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 30 matching lines @@
     "externalhmac.cc",
     "externalhmac.h",
     "mediamonitor.cc",
     "mediamonitor.h",
     "mediasession.cc",
     "mediasession.h",
     "rtcpmuxfilter.cc",
     "rtcpmuxfilter.h",
     "rtptransport.cc",
     "rtptransport.h",
+    "rtptransportinternal.h",
     "srtpfilter.cc",
     "srtpfilter.h",
     "srtpsession.cc",
     "srtpsession.h",
+    "srtptransport.cc",
+    "srtptransport.h",
     "voicechannel.h",
   ]
 
   deps = [
     "..:webrtc_common",
     "../api:call_api",
     "../api:libjingle_peerconnection_api",
     "../api:ortc_api",
     "../media:rtc_data",
     "../media:rtc_h264_profile_id",
@@ skipping 186 matching lines @@
     testonly = true
 
     sources = [
       "bundlefilter_unittest.cc",
       "channel_unittest.cc",
       "channelmanager_unittest.cc",
       "currentspeakermonitor_unittest.cc",
       "mediasession_unittest.cc",
       "rtcpmuxfilter_unittest.cc",
       "rtptransport_unittest.cc",
+      "rtptransporttestutil.h",
       "srtpfilter_unittest.cc",
       "srtpsession_unittest.cc",
       "srtptestutil.h",
+      "srtptransport_unittest.cc",
     ]
 
     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" ]
     }
 
     if (is_win) {
       libs = [ "strmiids.lib" ]
     }
 
     deps = [
       ":libjingle_peerconnection",
       ":rtc_pc",
       "../logging:rtc_event_log_api",
       "../media:rtc_media_base",
       "../media:rtc_media_tests_utils",
       "../p2p:p2p_test_utils",
       "../p2p:rtc_p2p",
       "../rtc_base:rtc_base",
       "../rtc_base:rtc_base_approved",
       "../rtc_base:rtc_base_tests_main",
       "../rtc_base:rtc_base_tests_utils",
       "../system_wrappers:metrics_default",
+      "../test:test_support",
     ]
 
     if (rtc_build_libsrtp) {
       deps += [ "//third_party/libsrtp" ]
     }
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
     }
   }
@@ skipping 149 matching lines @@
       "//testing/gmock",
     ]
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
 
       shard_timeout = 900
     }
   }
 }