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 10 matching lines @@
     "rtptransport.cc",
     "rtptransport.h",
     "srtpfilter.cc",
     "srtpfilter.h",
     "voicechannel.h",
   ]
 
   deps = [
     "../api:call_api",
     "../base:rtc_base",
-    "../media",
+    "../media:rtc_data",
   ]
 
   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_static_library("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("webrtc_null_audio") {

[kjellander_webrtc, 2017/05/29 20:59:15]

How about using rtc instead of webrtc in all these target names? It's what we
use in many places.

[Taylor Brandstetter, 2017/05/30 17:30:52]

It would be useful to have a comment explaining how the "modular build targets"
are expected to be used. 

[Zhi Huang, 2017/05/31 00:03:29]

Done.

[Zhi Huang, 2017/05/31 00:03:29]

Sounds good to me.

[Zhi Huang, 2017/05/31 03:55:37]

I just realized that if we follow the "rtc_..." pattern, then we'll end up
having pc:rtc_null_media and pc:rtc_media and there is also a target named
media:rtc_media. Is it OK?

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

AFAIK that is OK but you won't be able to build rtc_media from command line.
That's the only drawback as far as I understand it. It's still a confusing fact
so I'd prefer to rename pc:rtc_media to something like pc:pc_media instead.

+  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" ]
+  }
+}
+
+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" ]
+  }
+}
+
+rtc_static_library("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" ]
+  }
+}
+
+rtc_static_library("webrtc_media") {
+  sources = [
+    "mediafactory.cc",
+  ]
+
+  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" ]
+  }
+}
+
+rtc_static_library("webrtc_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",
     "../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_static_library("libjingle_peerconnection") {
+  public_deps = [
+    ":webrtc_audio",
+    ":webrtc_media",
+    ":webrtc_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 71 matching lines @@
       "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",
       "../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" ]
     }
   }
 
@@ skipping 103 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(kjellander): Remove (bugs.webrtc.org/6828)

[kjellander_webrtc, 2017/05/29 20:59:15]

Is it possible to avoid this?

[Zhi Huang, 2017/05/31 00:03:29]

Yep. :)

+    testonly = true
+    sources = [
+      "peerconnection_datachannelonly_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:base_jni",
+        "//webrtc/sdk/android:libjingle_peerconnection_java",
+        "//webrtc/sdk/android:webrtc_null_audio_jni",
+        "//webrtc/sdk/android:webrtc_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",
+      "../media:rtc_media_tests_utils",
+      "../pc:rtc_pc_base",
+      "../system_wrappers:metrics_default",
+      "//testing/gmock",
+    ]
+
+    if (is_android) {
+      deps += [ "//testing/android/native_test:native_test_support" ]
+

[kjellander_webrtc, 2017/05/29 20:59:15]

Remove blank line.

[Zhi Huang, 2017/05/31 00:03:29]

Done.

+      shard_timeout = 900
+    }
+  }
 }