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..088477bcc908cafa32c9e6d919e4631b88ae8bc8 100644
--- a/webrtc/pc/BUILD.gn
+++ b/webrtc/pc/BUILD.gn
@@ -25,25 +25,18 @@ config("rtc_pc_config") {
   }
 }
 
-rtc_static_library("rtc_pc") {
-  defines = []
+rtc_source_set("rtc_pc_common") {
   sources = [
     "audiomonitor.cc",
     "audiomonitor.h",
     "bundlefilter.cc",
     "bundlefilter.h",
-    "channel.cc",
-    "channel.h",
-    "channelmanager.cc",
-    "channelmanager.h",
     "currentspeakermonitor.cc",
     "currentspeakermonitor.h",
     "externalhmac.cc",
     "externalhmac.h",
     "mediamonitor.cc",
     "mediamonitor.h",
-    "mediasession.cc",
-    "mediasession.h",
     "rtcpmuxfilter.cc",
     "rtcpmuxfilter.h",
     "rtptransport.cc",
@@ -53,17 +46,64 @@ rtc_static_library("rtc_pc") {
     "voicechannel.h",
   ]
 
+  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" ]
+  }
+}
+
+rtc_static_library("rtc_pc_datachannel_only") {
+  sources = [
+    "channel.cc",
+    "channel.h",
+    "channelmanager.cc",
+    "channelmanager.h",
+    "mediasession.cc",
+    "mediasession.h",
+  ]
+
   deps = [
+    ":rtc_pc_common",
     "../api:call_api",
     "../base:rtc_base",
-    "../media",
+    "../media:rtc_media_datachannel_only",
   ]
 
-  if (rtc_build_libsrtp) {
-    deps += [ "//third_party/libsrtp" ]
+  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" ]
   }
+}
 
-  public_configs = [ ":rtc_pc_config" ]
+rtc_static_library("rtc_pc") {
+  defines = [
+    "HAVE_WEBRTC_VOICE",
+    "HAVE_WEBRTC_VIDEO",
+  ]
+
+  sources = [
+    "channel.cc",
+    "channel.h",
+    "channelmanager.cc",
+    "channelmanager.h",
+    "mediasession.cc",
+    "mediasession.h",
+  ]
+
+  deps = [
+    ":rtc_pc_common",
+    "../api:call_api",
+    "../base:rtc_base",
+    "../media:rtc_media",
+  ]
 
   if (!build_with_chromium && is_clang) {
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
@@ -80,7 +120,7 @@ config("libjingle_peerconnection_warnings_config") {
   }
 }
 
-rtc_static_library("libjingle_peerconnection") {
+rtc_source_set("libjingle_peerconnection_common") {
   cflags = []
   sources = [
     "audiotrack.cc",
@@ -100,10 +140,6 @@ rtc_static_library("libjingle_peerconnection") {
     "mediastreamobserver.cc",
     "mediastreamobserver.h",
     "mediastreamtrack.h",
-    "peerconnection.cc",
-    "peerconnection.h",
-    "peerconnectionfactory.cc",
-    "peerconnectionfactory.h",
     "remoteaudiosource.cc",
     "remoteaudiosource.h",
     "rtcstatscollector.cc",
@@ -141,12 +177,10 @@ rtc_static_library("libjingle_peerconnection") {
   }
 
   deps = [
-    ":rtc_pc",
     "../api:call_api",
     "../api:rtc_stats_api",
     "../api/video_codecs:video_codecs_api",
-    "../call",
-    "../media",
+    "../logging:rtc_event_log_api",
     "../stats",
   ]
 
@@ -168,6 +202,56 @@ rtc_static_library("libjingle_peerconnection") {
   }
 }
 
+rtc_static_library("libjingle_peerconnection_datachannel_only") {
+  sources = [
+    "peerconnection.cc",
+    "peerconnection.h",
+    "peerconnectionfactory.cc",
+    "peerconnectionfactory.h",
+  ]
+
+  deps = [
+    ":rtc_pc_datachannel_only",
+  ]
+
+  public_deps = [
+    ":libjingle_peerconnection_common",
+  ]
+
+  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" ]
+  }
+}
+
+rtc_static_library("libjingle_peerconnection") {
+  defines = [
+    "HAVE_WEBRTC_VOICE",
+    "HAVE_WEBRTC_VIDEO",
+  ]
+
+  sources = [
+    "peerconnection.cc",
+    "peerconnection.h",
+    "peerconnectionfactory.cc",
+    "peerconnectionfactory.h",
+  ]
+
+  deps = [
+    ":rtc_pc",
+    "../call",
+  ]
+
+  public_deps = [
+    ":libjingle_peerconnection_common",
+  ]
+
+  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" ]
+  }
+}
+
 if (rtc_include_tests) {
   config("rtc_pc_unittests_config") {
     # GN orders flags on a target before flags from configs. The default config
@@ -223,7 +307,7 @@ if (rtc_include_tests) {
     }
   }
 
-  rtc_source_set("pc_test_utils") {
+  rtc_source_set("pc_test_utils_common") {
     testonly = true
     sources = [
       "test/fakeaudiocapturemodule.cc",
@@ -243,8 +327,7 @@ if (rtc_include_tests) {
       "test/testsdpstrings.h",
     ]
 
-    deps = [
-      ":libjingle_peerconnection",
+    public_deps = [
       "../base:rtc_base_tests_utils",
       "//testing/gmock",
     ]
@@ -255,6 +338,28 @@ if (rtc_include_tests) {
     }
   }
 
+  rtc_source_set("pc_datachannel_only_test_utils") {
+    testonly = true
+    deps = [
+      ":libjingle_peerconnection_datachannel_only",
+    ]
+
+    public_deps = [
+      ":pc_test_utils_common",
+    ]
+  }
+
+  rtc_source_set("pc_test_utils") {
+    testonly = true
+    deps = [
+      ":libjingle_peerconnection",
+    ]
+
+    public_deps = [
+      ":pc_test_utils_common",
+    ]
+  }
+
   config("peerconnection_unittests_config") {
     # The warnings below are enabled by default. Since GN orders compiler flags
     # for a target before flags from configs, the only way to disable such
@@ -368,4 +473,64 @@ if (rtc_include_tests) {
       shard_timeout = 900
     }
   }
+
+  rtc_test("peerconnection_datachannel_only_unittests") {
+    check_includes = false  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+    testonly = true
+    sources = [
+      "peerconnectionendtoend_datachannel_only_unittest.cc",
+    ]
+
+    if (rtc_enable_sctp) {
+      defines = [ "HAVE_SCTP" ]
+    }
+
+    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:libjingle_peerconnection_datachannel_only_jni",
+        "//webrtc/sdk/android:libjingle_peerconnection_java",
+      ]
+    }
+
+    deps += [
+      ":libjingle_peerconnection_datachannel_only",
+      ":pc_datachannel_only_test_utils",
+      "..:webrtc_common",
+      "../api:fakemetricsobserver",
+      "../base:rtc_base_tests_main",
+      "../base:rtc_base_tests_utils",
+      "../pc:rtc_pc_datachannel_only",
+      "../system_wrappers:metrics_default",
+      "//testing/gmock",
+    ]
+
+    if (is_android) {
+      deps += [ "//testing/android/native_test:native_test_support" ]
+
+      shard_timeout = 900
+    }
+  }
 }