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

webrtc/voice_engine/voice_engine.gyp

 # Copyright (c) 2011 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.
 
 {
   'includes': [
@@ skipping 89 matching lines @@
       'type': 'static_library',
       'dependencies': [
         '<(webrtc_root)/base/base.gyp:rtc_base_approved',
         '<(webrtc_root)/common.gyp:webrtc_common',
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
       ],
       'sources': [
         'level_indicator.cc',
         'level_indicator.h',
       ]
-    }
+    },
+    {
+      'target_name': 'channel_transport',
+      'type': 'static_library',
+      'dependencies': [
+        '<(DEPTH)/testing/gtest.gyp:gtest',
+        '<(webrtc_root)/common.gyp:webrtc_common',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+      ],
+      '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_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_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_transport.h',
+        'test/channel_transport/udp_transport_impl.cc',
+        'test/channel_transport/udp_transport_impl.h',
+      ],
+      'msvs_disabled_warnings': [
+        4302,  # cast truncation
+      ],
+      'conditions': [
+        ['OS=="win" and clang==1', {
+          'msvs_settings': {
+            'VCCLCompilerTool': {
+              'AdditionalOptions': [
+                # Disable warnings failing when compiling with Clang on Windows.
+                # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+                '-Wno-parentheses-equality',
+                '-Wno-reorder',
+                '-Wno-tautological-constant-out-of-range-compare',
+                '-Wno-unused-private-field',
+              ],
+            },
+          },
+        }],
+      ],  # conditions.
+    },
+
   ],
   'conditions': [
     ['OS=="win"', {
       'defines': ['WEBRTC_DRIFT_COMPENSATION_SUPPORTED',],
     }],
     ['include_tests==1', {
       'targets': [
         {
           'target_name': 'voice_engine_unittests',
           'type': '<(gtest_target_type)',
           'dependencies': [
+            'channel_transport',
             'voice_engine',
             '<(DEPTH)/testing/gmock.gyp:gmock',
             '<(DEPTH)/testing/gtest.gyp:gtest',
             # The rest are to satisfy the unittests' include chain.
             # This would be unnecessary if we used qualified includes.
             '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
             '<(webrtc_root)/modules/modules.gyp:audio_device',
             '<(webrtc_root)/modules/modules.gyp:audio_processing',
             '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
             '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
             '<(webrtc_root)/modules/modules.gyp:media_file',
             '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
             '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
             '<(webrtc_root)/test/test.gyp:test_support_main',
           ],
           'sources': [
             'channel_unittest.cc',
             'network_predictor_unittest.cc',
+            'test/channel_transport/udp_transport_unittest.cc',
+            'test/channel_transport/udp_socket_manager_unittest.cc',
+            'test/channel_transport/udp_socket_wrapper_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',
           ],
           'conditions': [
             ['OS=="android"', {
               'dependencies': [
                 '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
               ],
             }],
           ],
         },
         {
           # command line test that should work on linux/mac/win
           'target_name': 'voe_cmd_test',
           'type': 'executable',
           'dependencies': [
+            'channel_transport',
             'voice_engine',
             '<(DEPTH)/testing/gtest.gyp:gtest',
             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
-            '<(webrtc_root)/test/test.gyp:channel_transport',
             '<(webrtc_root)/test/test.gyp:test_support',
             '<(webrtc_root)/webrtc.gyp:rtc_event_log',
           ],
           'sources': [
             'test/cmd_test/voe_cmd_test.cc',
           ],
         },
       ], # targets
       'conditions': [
         ['OS!="ios"', {
           'targets': [
             {
               'target_name': 'voe_auto_test',
               'type': 'executable',
               'dependencies': [
+                'channel_transport',
                 'voice_engine',
                 '<(DEPTH)/testing/gmock.gyp:gmock',
                 '<(DEPTH)/testing/gtest.gyp:gtest',
                 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
                 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
                 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
-                '<(webrtc_root)/test/test.gyp:channel_transport',
                 '<(webrtc_root)/test/test.gyp:test_common',
                 '<(webrtc_root)/test/test.gyp:test_support',
                 '<(webrtc_root)/webrtc.gyp:rtc_event_log',
                ],
               '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 116 matching lines @@
               'sources': [
                 'voe_auto_test.isolate',
               ],
             },
           ],
         }],
       ],  # conditions
     }], # include_tests==1
   ], # conditions
 }