Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/

webrtc/voice_engine/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.
 
 import("../build/webrtc.gni")
 
@@ skipping 102 matching lines @@
   ]
 
   deps = [
     "..:webrtc_common",
     "../base:rtc_base_approved",
     "../common_audio",
   ]
 }
 
 if (rtc_include_tests) {
+  config("channel_transport_warnings_config") {
+    if (is_win) {
+      cflags = [ "/wd4302" ]  # cast truncation
+
+      if (is_clang) {
+        # GN orders flags on a target before flags from configs. The default
+        # config adds -Wall, and this flag have to be after -Wall -- so they need
+        # to come from a config and cannot be on the target directly.
+        cflags += [
+          "-Wno-parentheses-equality",
+          "-Wno-reorder",
+          "-Wno-tautological-constant-out-of-range-compare",
+
+          # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6268
+          # for -Wno-thread-safety-analysis
+          "-Wno-thread-safety-analysis",
+          "-Wno-unused-private-field",
+        ]
+      }
+    }
+  }
+
+  rtc_source_set("channel_transport") {
+    testonly = true
+    sources = [
+      "test/channel_transport/channel_transport.cc",
+      "test/channel_transport/channel_transport.h",
+      "test/channel_transport/traffic_control_win.cc",
+      "test/channel_transport/traffic_control_win.h",
+      "test/channel_transport/udp_socket2_manager_win.cc",
+      "test/channel_transport/udp_socket2_manager_win.h",
+      "test/channel_transport/udp_socket2_win.cc",
+      "test/channel_transport/udp_socket2_win.h",
+      "test/channel_transport/udp_socket_manager_posix.cc",
+      "test/channel_transport/udp_socket_manager_posix.h",
+      "test/channel_transport/udp_socket_manager_wrapper.cc",
+      "test/channel_transport/udp_socket_manager_wrapper.h",
+      "test/channel_transport/udp_socket_posix.cc",
+      "test/channel_transport/udp_socket_posix.h",
+      "test/channel_transport/udp_socket_wrapper.cc",
+      "test/channel_transport/udp_socket_wrapper.h",
+      "test/channel_transport/udp_transport.h",
+      "test/channel_transport/udp_transport_impl.cc",
+      "test/channel_transport/udp_transport_impl.h",
+    ]
+
+    configs += [ ":channel_transport_warnings_config" ]
+
+    if (is_clang && !is_nacl) {
+      # Suppress warnings from the Chromium Clang plugin.
+      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+    }
+
+    deps = [
+      "..:webrtc_common",
+      "../system_wrappers",
+      "//testing/gtest",
+    ]
+  }
+
   rtc_test("voice_engine_unittests") {
     deps = [
+      ":channel_transport",
       ":voice_engine",
       "//testing/gmock",
       "//testing/gtest",
       "//webrtc/common_audio",
       "//webrtc/modules/audio_coding",
       "//webrtc/modules/audio_conference_mixer",
       "//webrtc/modules/audio_device",
       "//webrtc/modules/audio_processing",
       "//webrtc/modules/media_file",
       "//webrtc/modules/rtp_rtcp",
       "//webrtc/modules/utility",
       "//webrtc/system_wrappers",
       "//webrtc/test:test_support_main",
     ]
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_native_code" ]
       shard_timeout = 900
     }
 
     sources = [
       "channel_unittest.cc",
       "network_predictor_unittest.cc",
+      "test/channel_transport/udp_socket_manager_unittest.cc",
+      "test/channel_transport/udp_socket_wrapper_unittest.cc",
+      "test/channel_transport/udp_transport_unittest.cc",
       "transmit_mixer_unittest.cc",
       "utility_unittest.cc",
       "voe_audio_processing_unittest.cc",
       "voe_base_unittest.cc",
       "voe_codec_unittest.cc",
       "voe_network_unittest.cc",
       "voice_engine_fixture.cc",
       "voice_engine_fixture.h",
     ]
 
@@ skipping 11 matching lines @@
       # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 
   if (!is_ios) {
     rtc_executable("voe_auto_test") {
       testonly = true
 
       deps = [
+        ":channel_transport",
         ":voice_engine",
         "//testing/gmock",
         "//testing/gtest",
         "//third_party/gflags",
         "//webrtc/:rtc_event_log",
         "//webrtc/modules/video_capture",
         "//webrtc/system_wrappers",
         "//webrtc/system_wrappers/:system_wrappers_default",
-        "//webrtc/test/:channel_transport",
         "//webrtc/test/:test_common",
         "//webrtc/test/:test_support",
       ]
 
       sources = [
         "test/auto_test/automated_mode.cc",
         "test/auto_test/extended/agc_config_test.cc",
         "test/auto_test/extended/ec_metrics_test.cc",
         "test/auto_test/fakes/conference_transport.cc",
         "test/auto_test/fakes/conference_transport.h",
@@ skipping 60 matching lines @@
       }
 
       if (is_clang) {
         # Suppress warnings from Chrome's Clang plugins.
         # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
         suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
       }
     }
   }
 }