Remove unused VoiceChannelTransport.

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("../webrtc.gni")
 
@@ skipping 161 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_static_library("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 (!build_with_chromium && is_clang) {
-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
-    }
-
-    deps = [
-      ":voice_engine",
-      "..:webrtc_common",
-      "../api:transport_api",
-      "../base:rtc_base_approved",
-      "../system_wrappers",
-      "../test:test_support",
-      "//testing/gtest",
-    ]
-  }
-
   rtc_test("voice_engine_unittests") {
     deps = [
-      ":channel_transport",
       ":file_player",
       ":voice_engine",
       "../base:rtc_base_approved",
       "../test:test_common",
       "//testing/gmock",
       "//testing/gtest",
       "//third_party/gflags",
       "//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/modules/video_capture:video_capture",
       "//webrtc/system_wrappers",
       "//webrtc/test:test_main",
       "//webrtc/test:video_test_common",
     ]
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_native_code" ]
       shard_timeout = 900
     }
 
     sources = [
       "channel_unittest.cc",
       "file_player_unittests.cc",
-      "test/channel_transport/udp_socket_manager_unittest.cc",
-      "test/channel_transport/udp_socket_wrapper_unittest.cc",
-      "test/channel_transport/udp_transport_unittest.cc",
       "transport_feedback_packet_loss_tracker_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 15 matching lines @@
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 
   if (!is_ios) {
     rtc_executable("voe_auto_test") {
       testonly = true
 
       deps = [
-        ":channel_transport",
         ":voice_engine",
         "..:webrtc_common",
         "../base:rtc_base_approved",
         "../modules/audio_device:audio_device",
         "../modules/audio_processing:audio_processing",
         "../modules/rtp_rtcp:rtp_rtcp",
         "//testing/gmock",
         "//testing/gtest",
         "//third_party/gflags",
         "//webrtc/logging:rtc_event_log_api",
@@ skipping 64 matching lines @@
         ]
       }
 
       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" ]
       }
     }
   }
 }