GN: New conventions, default target and refactorings

webrtc/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.
 
 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
 
@@ skipping 225 matching lines @@
     defines += [ "WEBRTC_UNSAFE_FUZZER_MODE" ]
   }
 }
 
 config("common_objc") {
   libs = [ "Foundation.framework" ]
   precompiled_header = "sdk/objc/WebRTC-Prefix.pch"
   precompiled_source = "sdk/objc/WebRTC-Prefix.pch"
 }
 
-if (!is_ios || !build_with_chromium) {
+if (!build_with_chromium) {
+  # Target to build all the WebRTC production code.
   rtc_static_library("webrtc") {
+    # Only the root target should depend on this.
+    visibility = [ "//:default" ]
+
     sources = [
       # TODO(kjellander): Remove this whenever possible. GN's static_library
       # target type requires at least one object to avoid errors linking.
       "build/no_op_function.cc",
       "call.h",
       "config.h",
       "transport.h",
     ]
 
     defines = []
 
     deps = [
       ":webrtc_common",
+      "api",
       "audio",
-      "base:rtc_base",
+      "base",
       "call",
       "common_audio",
       "common_video",
+      "libjingle/xmllite",
+      "libjingle/xmpp",
+      "logging",
+      "media",
       "modules",
+      "modules/video_capture:video_capture_internal_impl",
+      "p2p",
+      "pc",
+      "sdk",
       "stats",
       "system_wrappers",
-      "tools",
       "video",
       "voice_engine",
     ]
 
-    if (build_with_chromium) {
-      deps += [ "modules/video_capture" ]
-    } else {
-      # TODO(kjellander): Enable for Chromium as well when bugs.webrtc.org/4256
-      # is fixed. Right now it's not possible due to circular dependencies.
-      deps += [
-        "api",
-        "media",
-        "p2p",
-        "pc",
-      ]
-    }
-
     if (rtc_enable_protobuf) {
       defines += [ "ENABLE_RTC_EVENT_LOG" ]
       deps += [ "logging:rtc_event_log_proto" ]
     }
   }
+
+  if (rtc_include_tests) {
+    # Target to build all the WebRTC tests (but not examples or tools).
+    # Executable in order to get a target that links all WebRTC code.
+    rtc_executable("webrtc_tests") {
+      testonly = true
+
+      # Only the root target should depend on this.
+      visibility = [ "//:default" ]
+
+      deps = [
+        ":rtc_unittests",
+        ":video_engine_tests",
+        ":webrtc_nonparallel_tests",
+        ":webrtc_perf_tests",
+        ":xmllite_xmpp_unittests",
+        "api:peerconnection_unittests",
+        "common_audio:common_audio_unittests",
+        "common_video:common_video_unittests",
+        "media:rtc_media_unittests",
+        "modules:modules_tests",
+        "modules:modules_unittests",
+        "modules/audio_coding:audio_coding_tests",
+        "modules/audio_processing:audio_processing_tests",
+        "modules/rtp_rtcp:test_packet_masks_metrics",
+        "modules/video_capture:video_capture_internal_impl",
+        "pc:rtc_pc_unittests",
+        "stats:rtc_stats_unittests",
+        "system_wrappers:system_wrappers_unittests",
+        "test",
+        "video:screenshare_loopback",
+        "video:video_loopback",
+        "video:video_tests",
+        "voice_engine:voe_cmd_test",
+        "voice_engine:voice_engine_unittests",
+      ]
+      if (is_android) {
+        deps += [
+          ":android_junit_tests",
+          "api:libjingle_peerconnection_android_unittest",
+        ]
+      } else {
+        deps += [ "modules/video_capture:video_capture_tests" ]
+      }
+      if (!is_ios) {
+        deps += [
+          "modules/audio_device:audio_device_tests",
+          "voice_engine:voe_auto_test",
+        ]
+      }
+    }
+  }
 }
 
 rtc_static_library("webrtc_common") {
   sources = [
     "common_types.cc",
     "common_types.h",
     "config.cc",
     "config.h",
     "typedefs.h",
     "voice_engine_configurations.h",
@@ skipping 328 matching lines @@
   if (is_ios) {
     bundle_data("webrtc_perf_tests_bundle_data") {
       testonly = true
       sources = webrtc_perf_tests_resources
       outputs = [
         "{{bundle_resources_dir}}/{{source_file_part}}",
       ]
     }
   }
 
-  rtc_executable("webrtc_tests") {
-    testonly = true
-    deps = [
-      ":webrtc",
-      "modules/video_capture:video_capture_internal_impl",
-      "test",
-    ]
-  }
-
   rtc_test("webrtc_perf_tests") {
     testonly = true
     configs += [ ":rtc_unittests_config" ]
 
     sources = [
       "call/call_perf_tests.cc",
       "call/rampup_tests.cc",
       "call/rampup_tests.h",
       "modules/audio_coding/neteq/test/neteq_performance_unittest.cc",
       "modules/audio_processing/audio_processing_performance_unittest.cc",
@@ skipping 76 matching lines @@
 
       deps = [
         "//base:base_java_test_support",
         "//webrtc/api:libjingle_peerconnection_java",
         "//webrtc/api:libjingle_peerconnection_jni",
         "//webrtc/examples:AppRTCMobile_javalib",
       ]
     }
   }
 }