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") {
-  defines = []
+rtc_source_set("rtc_pc_sources") {
   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",
     "rtptransport.h",
     "srtpfilter.cc",
     "srtpfilter.h",
     "voicechannel.h",
   ]
 
-  deps = [
-    "../api:call_api",
-    "../base:rtc_base",
-    "../media",
-  ]
-
   if (rtc_build_libsrtp) {
-    deps += [ "//third_party/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_sources",
+    "../api:call_api",
+    "../base:rtc_base",
+    "../media:rtc_media_datachannel_only",
+  ]
+
+  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") {
+  defines = [
+    "HAVE_WEBRTC_VOICE",
+    "HAVE_WEBRTC_VIDEO",
+  ]
+
+  sources = [
+    "channel.cc",
+    "channel.h",
+    "channelmanager.cc",
+    "channelmanager.h",
+    "mediasession.cc",
+    "mediasession.h",
+  ]
+
+  deps = [
+    ":rtc_pc_sources",
+    "../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).
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 }
 
 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_source_set("libjingle_peerconnection_sources") {
   cflags = []
   sources = [
     "audiotrack.cc",
     "audiotrack.h",
     "datachannel.cc",
     "datachannel.h",
     "dtmfsender.cc",
     "dtmfsender.h",
     "iceserverparsing.cc",
     "iceserverparsing.h",
     "jsepicecandidate.cc",
     "jsepsessiondescription.cc",
     "localaudiosource.cc",
     "localaudiosource.h",
     "mediastream.cc",
     "mediastream.h",
     "mediastreamobserver.cc",
     "mediastreamobserver.h",
     "mediastreamtrack.h",
-    "peerconnection.cc",
-    "peerconnection.h",
-    "peerconnectionfactory.cc",
-    "peerconnectionfactory.h",
     "remoteaudiosource.cc",
     "remoteaudiosource.h",
     "rtcstatscollector.cc",
     "rtcstatscollector.h",
     "rtpreceiver.cc",
     "rtpreceiver.h",
     "rtpsender.cc",
     "rtpsender.h",
     "sctputils.cc",
     "sctputils.h",
@@ skipping 17 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",
     "../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",
   ]
 
   if (rtc_use_quic) {
     sources += [
       "quicdatachannel.cc",
       "quicdatachannel.h",
       "quicdatatransport.cc",
       "quicdatatransport.h",
     ]
     deps += [ "//third_party/libquic" ]
     public_deps = [
       "//third_party/libquic",
     ]
   }
 }
 
+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_sources",
+  ]
+
+  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_sources",
+  ]
+
+  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
     # adds -Wall, and this flag have to be after -Wall -- so they need to
     # come from a config and can't be on the target directly.
     if (!is_win && !is_clang) {
       cflags = [ "-Wno-maybe-uninitialized" ]  # Only exists for GCC.
     }
   }
 
@@ skipping 35 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") {
+  rtc_source_set("pc_test_utils_sources") {
     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",
+    public_deps = [
       "../base:rtc_base_tests_utils",
       "//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" ]
     }
   }
 
+  rtc_source_set("pc_datachannel_only_test_utils") {
+    testonly = true
+    deps = [
+      ":libjingle_peerconnection_datachannel_only",
+    ]
+
+    public_deps = [
+      ":pc_test_utils_sources",
+    ]
+  }
+
+  rtc_source_set("pc_test_utils") {
+    testonly = true
+    deps = [
+      ":libjingle_peerconnection",
+    ]
+
+    public_deps = [
+      ":pc_test_utils_sources",
+    ]
+  }
+
   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
     # warnings is by having them in a separate config, loaded from the target.
     # TODO(kjellander): Make the code compile without disabling these flags.
     # See https://bugs.webrtc.org/3307.
     if (is_clang && is_win) {
       cflags = [
         # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6267
         # for -Wno-sign-compare
@@ skipping 93 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_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
+    }
+  }
 }