Build WebRTC with data channel only.

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) {
   import("//build/config/android/config.gni")
   import("//build/config/android/rules.gni")
 }
 
 group("pc") {
   public_deps = [
     ":rtc_pc",
   ]
 }
 
 config("rtc_pc_config") {
   defines = []
   if (rtc_enable_sctp) {
     defines += [ "HAVE_SCTP" ]
   }
 }
 
-rtc_static_library("rtc_pc") {
+rtc_static_library("rtc_pc_base") {
   defines = []
   sources = [
     "audiomonitor.cc",
     "audiomonitor.h",
     "bundlefilter.cc",
     "bundlefilter.h",
     "channel.cc",
     "channel.h",
     "channelmanager.cc",
     "channelmanager.h",
@@ skipping 13 matching lines @@
     "srtpfilter.h",
     "voicechannel.h",
   ]
 
   deps = [
     "..:webrtc_common",
     "../api:call_api",
     "../api:libjingle_peerconnection_api",
     "../api:ortc_api",
     "../base:rtc_base",
-    "../common_video:common_video",
-    "../media",
+    "../base:rtc_task_queue",
+    "../media:rtc_data",
+    "../media:rtc_media_base_data",
     "../p2p:rtc_p2p",
   ]
 
   if (rtc_build_libsrtp) {
     deps += [ "//third_party/libsrtp" ]
   }
 
   public_configs = [ ":rtc_pc_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" ]
   }
 }
 
+# TODO(zhihuang): Remove this once the downstream dependencies start using the
+# modular targets.
+rtc_source_set("rtc_pc") {
+  public_deps = [
+    ":rtc_pc_base",
+  ]
+
+  deps = [
+    "../media:rtc_audio_video",
+  ]
+}
+
 config("libjingle_peerconnection_warnings_config") {
   # GN orders flags on a target before flags from configs. The default config
   # adds these flags so to cancel them out they need to come from a config and
   # cannot be on the target directly.
   if (!is_win && !is_clang) {
     cflags = [ "-Wno-maybe-uninitialized" ]  # Only exists for GCC.
   }
 }
 
-rtc_static_library("libjingle_peerconnection") {
+rtc_static_library("peerconnection") {
   cflags = []
   sources = [
     "audiotrack.cc",
     "audiotrack.h",
     "datachannel.cc",
     "datachannel.h",
     "dtmfsender.cc",
     "dtmfsender.h",
     "iceserverparsing.cc",
     "iceserverparsing.h",
@@ skipping 40 matching lines @@
   ]
 
   configs += [ ":libjingle_peerconnection_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 = [
-    ":rtc_pc",
+    ":rtc_pc_base",
     "..:webrtc_common",
     "../api:call_api",
     "../api:rtc_stats_api",
-    "../api/audio_codecs:builtin_audio_decoder_factory",
-    "../api/audio_codecs:builtin_audio_encoder_factory",
     "../api/video_codecs:video_codecs_api",
     "../base:rtc_base",
     "../base:rtc_base_approved",
-    "../call",
+    "../call:call_interfaces",
     "../logging:rtc_event_log_api",
-    "../media",
-    "../modules/audio_device:audio_device",
+    "../media:rtc_data",
+    "../media:rtc_media_base_data",
     "../p2p:rtc_p2p",
     "../stats",
     "../system_wrappers:system_wrappers",
   ]
 
   public_deps = [
     "../api:libjingle_peerconnection_api",
   ]
+}
+
+# This target implements the CreatePeerConnectionFactory methods to build WebRTC
+# with full support(audio, video and datachannel). The applications would be
+# responsible to create their own implementation of CreatePeerConnectionFactory
+# methods with different dependencies based on their requirements.
+# The target "create_pc_factory_datachannelonly" is an example.
+rtc_static_library("create_pc_factory") {
+  sources = [
+    "createpeerconnectionfactory.cc",
+  ]
+
+  deps = [
+    "../api:audio_mixer_api",
+    "../api:libjingle_peerconnection_api",
+    "../api/audio_codecs:audio_codecs_api",
+    "../api/audio_codecs:builtin_audio_decoder_factory",
+    "../api/audio_codecs:builtin_audio_encoder_factory",
+    "../base:rtc_base",
+    "../base:rtc_base_approved",
+    "../call",
+    "../call:call_interfaces",
+    "../logging:rtc_event_log_api",
+    "../media:rtc_audio_video",
+    "../modules/audio_device:audio_device",
+  ]
+
+  configs += [ ":libjingle_peerconnection_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" ]
+  }
+}
+
+# This target implements the CreatePeerConnectionFactory methods to build WebRTC
+# with datachannel only support.
+rtc_static_library("create_pc_factory_datachannelonly") {
+  sources = [
+    "createpeerconnectionfactory_datachannelonly.cc",
+  ]
+
+  deps = [
+    "../api:audio_mixer_api",
+    "../api:libjingle_peerconnection_api",
+    "../base:rtc_base",
+    "../base:rtc_base_approved",
+    "../call:call_interfaces",
+    "../logging:rtc_event_log_api",
+  ]
+
+  configs += [ ":libjingle_peerconnection_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" ]
+  }
+}
+
+# TODO(zhihuang): Remove this once the downstream dependencies start using the
+# modular targets.
+rtc_source_set("libjingle_peerconnection") {
+  public_deps = [
+    ":create_pc_factory",
+    ":peerconnection",
+    "../api:libjingle_peerconnection_api",
+  ]
 
   if (rtc_use_quic) {
     sources += [
       "quicdatachannel.cc",
       "quicdatachannel.h",
       "quicdatatransport.cc",
       "quicdatatransport.h",
     ]
     deps += [ "//third_party/libquic" ]
     public_deps = [
@@ skipping 57 matching lines @@
     if (rtc_build_libsrtp) {
       deps += [ "//third_party/libsrtp" ]
     }
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
     }
   }
 
   rtc_source_set("pc_test_utils") {
+    # Cannot have GN check enabled because this target would also be used in the
+    # "peerconnection_datachannelonly_unittests" and we don't want to depend on
+    # the target "media:rtc_media_tests_utils" indrectly since it contains all
+    # the audio and video related classes.
+    # TODO(zhihuang): Enable the check once the "media:rtc_media_tests_utils" is
+    # broken down to modular sub-targets.
+    check_includes = false
     testonly = true
     sources = [
       "test/fakeaudiocapturemodule.cc",
       "test/fakeaudiocapturemodule.h",
       "test/fakedatachannelprovider.h",
       "test/fakeperiodicvideocapturer.h",
       "test/fakertccertificategenerator.h",
       "test/fakevideotrackrenderer.h",
       "test/fakevideotracksource.h",
       "test/mock_datachannel.h",
       "test/mock_peerconnection.h",
       "test/mock_webrtcsession.h",
       "test/mockpeerconnectionobservers.h",
       "test/peerconnectiontestwrapper.cc",
       "test/peerconnectiontestwrapper.h",
       "test/rtcstatsobtainer.h",
       "test/testsdpstrings.h",
     ]
 
     deps = [
-      ":libjingle_peerconnection",
+      ":peerconnection",
       "..:webrtc_common",
       "../api:libjingle_peerconnection_test_api",
       "../api:rtc_stats_api",
       "../base:rtc_base",
       "../base:rtc_base_approved",
       "../base:rtc_base_tests_utils",
-      "../media:rtc_media",
-      "../media:rtc_media_tests_utils",
       "../modules/audio_device:audio_device",
       "../p2p:p2p_test_utils",
       "../test:test_support",
       "//testing/gmock",
     ]
 
     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" ]
     }
@@ skipping 105 matching lines @@
       "../system_wrappers:metrics_default",
       "//testing/gmock",
     ]
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
 
       shard_timeout = 900
     }
   }
+
+  rtc_test("peerconnection_datachannelonly_unittests") {
+    check_includes = false  # TODO(zhihuang): Remove (bugs.webrtc.org/6828)
+    testonly = true
+    sources = [
+      "peerconnection_datachannelonly_unittest.cc",
+    ]
+
+    configs += [ ":peerconnection_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" ]
+    }
+
+    # TODO(jschuh): Bug 1348: fix this warning.
+    configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
+
+    if (is_win) {
+      cflags = [
+        "/wd4245",  # conversion from int to size_t, signed/unsigned mismatch.
+        "/wd4389",  # signed/unsigned mismatch.
+      ]
+    }
+
+    deps = []
+    if (is_android) {
+      sources += [
+        "test/androidtestinitializer.cc",
+        "test/androidtestinitializer.h",
+      ]
+      deps += [
+        "//testing/android/native_test:native_test_support",
+        "//webrtc/sdk/android:base_jni",
+        "//webrtc/sdk/android:libjingle_peerconnection_java",
+        "//webrtc/sdk/android:null_video_jni",
+      ]
+    }
+
+    deps += [
+      ":create_pc_factory_datachannelonly",
+      ":pc_test_utils",
+      ":peerconnection",
+      "..:webrtc_common",
+      "../api:fakemetricsobserver",
+      "../base:rtc_base_approved",
+      "../base:rtc_base_tests_main",
+      "../base:rtc_base_tests_utils",
+      "../modules/utility",
+      "../pc:rtc_pc_base",
+      "../system_wrappers:metrics_default",
+      "//testing/gmock",
+    ]
+
+    if (is_android) {
+      deps += [ "//testing/android/native_test:native_test_support" ]
+      shard_timeout = 900
+    }
+  }
 }