Build WebRTC with data channel only.

webrtc/pc/BUILD.gn

Index: webrtc/pc/BUILD.gn
diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn
index 483485012da481803fe641f220ef643832af61bb..96561e5ae0828189fb4c713bde65320207c87c1c 100644
--- a/webrtc/pc/BUILD.gn
+++ b/webrtc/pc/BUILD.gn
@@ -25,7 +25,7 @@ config("rtc_pc_config") {
   }
 }
 
-rtc_static_library("rtc_pc") {
+rtc_static_library("rtc_pc_base") {
   defines = []
   sources = [
     "audiomonitor.cc",
@@ -56,7 +56,8 @@ rtc_static_library("rtc_pc") {
   deps = [
     "../api:call_api",
     "../base:rtc_base",
-    "../media",
+    "../base:rtc_task_queue",
+    "../media:rtc_data",
   ]
 
   if (rtc_build_libsrtp) {
@@ -71,6 +72,18 @@ rtc_static_library("rtc_pc") {
   }
 }
 
+# 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
@@ -80,7 +93,93 @@ config("libjingle_peerconnection_warnings_config") {
   }
 }
 
-rtc_static_library("libjingle_peerconnection") {
+# This target contains the null implementation of the audio module and it is
+# used to build WebRTC without audio support.
+rtc_static_library("webrtc_null_audio") {
+  sources = [
+    "nullaudiofactory.cc",
+  ]
+
+  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 contains the real implementation of the audio module and it is
+# used to build WebRTC with audio support. It should never be used with
+# "webrtc_null_audio" at the same time and it should always be linked with the
+# "webrtc_media".
+rtc_static_library("webrtc_audio") {
+  sources = [
+    "audiofactory.cc",
+  ]
+
+  public_deps = [
+    "../media:rtc_audio_video",
+  ]
+
+  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 contains the null implementation of the audio/video related
+# objects and it is used to build WebRTC without audio and video support.
+rtc_source_set("webrtc_null_media") {
+  sources = [
+    "nullmediafactory.cc",
+  ]
+
+  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 contains the real implementation of the audio/video related
+# objects and it is used to build WebRTC with audio and video support.
+rtc_source_set("webrtc_media") {

[kjellander_webrtc, 2017/06/01 05:34:30]

Can we call this pc_media instead?

[Zhi Huang, 2017/06/02 05:16:43]

SGTM.

+  deps = [
+    "../call",
+    "../media:rtc_audio_video",
+  ]
+
+  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" ]
+  }
+}
+
+# The modular build targets can be used to build WebRTC with different
+# functionalities. The users can choose either the real implemenation
+# or the null implementation of the audio/video modules based on their
+# requirments.

[kjellander_webrtc, 2017/06/01 05:34:30]

requirements.

[Zhi Huang, 2017/06/02 05:16:43]

Done.

+#
+# For example, to build WebRTC with datachannel support only, we would need the
+# the peerconnection and the null implementation of the audio and video modules.
+#
+# rtc_source_set("webrtc_datachannel_only") {
+#    deps = [
+#      ":webrtc_null_audio",
+#      ":webrtc_null_media",
+#      ":webrtc_peerconnection",
+#    ]
+# }
+#
+# To build WebRTC with all the audio, video and datachannel support, we would
+# need the peerconnection and the real implementation of the audio and video
+# modules.
+#
+# rtc_source_set("webrtc_full") {
+#    deps = [
+#      ":webrtc_audio",
+#      ":webrtc_media",
+#      ":webrtc_peerconnection",
+#    ]
+# }
+rtc_static_library("webrtc_peerconnection") {
   cflags = []
   sources = [
     "audiotrack.cc",
@@ -141,18 +240,28 @@ rtc_static_library("libjingle_peerconnection") {
   }
 
   deps = [
-    ":rtc_pc",
+    ":rtc_pc_base",
     "../api:call_api",
     "../api:rtc_stats_api",
     "../api/video_codecs:video_codecs_api",
-    "../call",
-    "../media",
+    "../logging:rtc_event_log_api",
     "../stats",
   ]
 
   public_deps = [
     "../api:libjingle_peerconnection_api",
   ]
+}
+
+# TODO(zhihuang): Remove this once the downstream dependencies start using the
+# modular targets.
+rtc_source_set("libjingle_peerconnection") {
+  public_deps = [
+    ":webrtc_audio",
+    ":webrtc_media",
+    ":webrtc_peerconnection",
+    "../api:libjingle_peerconnection_api",
+  ]
 
   if (rtc_use_quic) {
     sources += [
@@ -244,7 +353,6 @@ if (rtc_include_tests) {
     ]
 
     deps = [
-      ":libjingle_peerconnection",
       "../base:rtc_base_tests_utils",
       "//testing/gmock",
     ]
@@ -368,4 +476,65 @@ if (rtc_include_tests) {
       shard_timeout = 900
     }
   }
+
+  rtc_test("peerconnection_datachannelonly_unittests") {
+    testonly = true
+    sources = [
+      "peerconnection_datachannelonly_unittest.cc",
+    ]
+
+    defines = [ "HAVE_SCTP" ]

[kjellander_webrtc, 2017/06/01 05:34:30]

You shouldn't need to define this since you depend on rtc_pc_base which has
rtc_pc_config in its public_configs.

[Zhi Huang, 2017/06/02 05:16:43]

Done.

+
+    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_audio_jni",
+        "//webrtc/sdk/android:null_video_jni",
+      ]
+    }
+
+    deps += [
+      ":pc_test_utils",
+      ":webrtc_null_audio",
+      ":webrtc_null_media",
+      ":webrtc_peerconnection",
+      "..:webrtc_common",
+      "../api:fakemetricsobserver",
+      "../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
+    }
+  }
 }