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.

[the sun, 2017/06/12 20:58:41]

I'd think create_pc_factory is example enough and the implementation(s) used in
the JNI layer are "application" specific.

[Zhi Huang, 2017/06/12 22:18:58]

Acknowledged.

+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") {

[the sun, 2017/06/12 20:58:41]

Is it possible to move these variants to the JNI code? IIUC we have "link time
polymorphism" again here. I understand we need it for the Java layer, to support
the different configs, but lets keep it there then.

[Zhi Huang, 2017/06/12 22:18:58]

So you are suggesting to compile this file in the jni target?  I think that
would work if the JNI owner is OK with it.

+  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") {

[the sun, 2017/06/12 20:58:41]

You shouldn't need a separate target for this - the test could be part of the
default pc test suite. If all link time polymorphism is gone, the compiler will
instead be used to verify how PCF is instantiated (and we could conceivably have
different configurations at once, within the same process).

[Zhi Huang, 2017/06/12 22:18:58]

Acknowledged.

+    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
+    }
+  }
 }