Create //webrtc/api:libjingle_peerconnection_api + refactorings.

webrtc/api/BUILD.gn

 # Copyright (c) 2015 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("../build/webrtc.gni")
 if (is_android) {
   import("//build/config/android/config.gni")
   import("//build/config/android/rules.gni")
 }
 
 group("api") {
   public_deps = [
-    ":libjingle_peerconnection",
+    ":libjingle_peerconnection_api",
   ]
 }
 
 rtc_source_set("call_api") {
   sources = [
     "call/audio_sink.h",
     "call/flexfec_receive_stream.h",
   ]
 
   deps = [
     # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
     ":audio_mixer_api",
     ":transport_api",
     "..:webrtc_common",
     "../base:rtc_base_approved",
     "../modules/audio_coding:audio_decoder_factory_interface",
     "../modules/audio_coding:audio_encoder_interface",
   ]
 }
 
-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("libjingle_peerconnection_api") {
   check_includes = false  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
   cflags = []
   sources = [
-    "audiotrack.cc",
-    "audiotrack.h",
-    "datachannel.cc",
     "datachannel.h",
     "datachannelinterface.h",
-    "dtmfsender.cc",
-    "dtmfsender.h",
     "dtmfsenderinterface.h",
+
+    # TODO(ossu): Put fakemetricobserver in a separate target for api mock/fake
+    # classes.
+    "fakemetricsobserver.cc",
+    "fakemetricsobserver.h",
     "jsep.h",
-    "jsepicecandidate.cc",
-    "jsepicecandidate.h",
-    "jsepsessiondescription.cc",
     "jsepsessiondescription.h",
-    "localaudiosource.cc",
-    "localaudiosource.h",
     "mediaconstraintsinterface.cc",
     "mediaconstraintsinterface.h",
-    "mediacontroller.cc",
     "mediacontroller.h",
-    "mediastream.cc",
     "mediastream.h",
     "mediastreaminterface.h",
-    "mediastreamobserver.cc",
-    "mediastreamobserver.h",
     "mediastreamproxy.h",
     "mediastreamtrack.h",
     "mediastreamtrackproxy.h",
+    "mediatypes.cc",
+    "mediatypes.h",
     "notifier.h",
-    "peerconnection.cc",
-    "peerconnection.h",
-    "peerconnectionfactory.cc",
-    "peerconnectionfactory.h",
     "peerconnectionfactoryproxy.h",
     "peerconnectioninterface.h",
     "peerconnectionproxy.h",
     "proxy.h",
-    "remoteaudiosource.cc",
-    "remoteaudiosource.h",
-    "rtcstatscollector.cc",
-    "rtcstatscollector.h",
     "rtpparameters.h",
-    "rtpreceiver.cc",
-    "rtpreceiver.h",
     "rtpreceiverinterface.h",
-    "rtpsender.cc",
     "rtpsender.h",
     "rtpsenderinterface.h",
-    "sctputils.cc",
-    "sctputils.h",
-    "statscollector.cc",
-    "statscollector.h",
     "statstypes.cc",
     "statstypes.h",
     "streamcollection.h",
-    "videocapturertracksource.cc",
-    "videocapturertracksource.h",
+    "test/fakeconstraints.h",

[kjellander_webrtc, 2016/12/20 07:17:06]

This wasn't here before and seems to be only included by tests +
webrtc/api/test/peerconnectiontestwrapper.h.

It used to be in the peerconnection_unittests target so why add in the api
target? I understand it makes sense keeping it in api as it's a fake, but I'm
wondering if we should have *_test_api targets for those or something similar. 
mock_audio_mixer is such an example:
https://cs.chromium.org/chromium/src/third_party/webrtc/api/BUILD.gn?rcl=0&l=325

[ossu, 2017/01/12 14:03:06]

Yeah, putting testing-specific things into a separate target (i.e. fakes and
mocks) makes sense. I've created a libjingle_peerconnection_test_api target and
put this fake in there for now; or do you prefer separate target for each fake
or mock, such as the mock_audio_mixer target? Regardless of target, no
downstream project will properly depend on it to get at fakeconstraints.h but
AFAICT, they didn't before either, so we should be fine to move forward.

+    "umametrics.h",
     "videosourceproxy.h",
-    "videotrack.cc",
-    "videotrack.h",
-    "videotracksource.cc",
     "videotracksource.h",
-    "webrtcsdp.cc",
-    "webrtcsdp.h",
-    "webrtcsession.cc",
-    "webrtcsession.h",
-    "webrtcsessiondescriptionfactory.cc",
-    "webrtcsessiondescriptionfactory.h",
   ]
 
-  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 = [
-    ":call_api",
     ":rtc_stats_api",
-    "../call",
-    "../media",
-    "../pc",
-    "../stats",
   ]
+}
 
-  if (rtc_use_quic) {
-    sources += [
-      "quicdatachannel.cc",
-      "quicdatachannel.h",
-      "quicdatatransport.cc",
-      "quicdatatransport.h",
-    ]
-    deps += [ "//third_party/libquic" ]
-    public_deps = [
-      "//third_party/libquic",
-    ]
-  }
+# TODO(ossu): Remove once downstream projects have updated.
+rtc_source_set("libjingle_peerconnection") {
+  deps = [
+    "../pc:libjingle_peerconnection",
+  ]
 }
 
 rtc_source_set("rtc_stats_api") {
   cflags = []
   sources = [
     "stats/rtcstats.h",
     "stats/rtcstats_objects.h",
+    "stats/rtcstatscollectorcallback.h",
     "stats/rtcstatsreport.h",
   ]
 
   deps = [
     "../base:rtc_base_approved",
   ]
 }
 
 rtc_source_set("audio_mixer_api") {
   sources = [
@@ skipping 27 matching lines @@
     public_deps = [
       "$rtc_libyuv_dir",
     ]
   } else {
     # Need to add a directory normally exported by libyuv.
     include_dirs = [ "$rtc_libyuv_dir/include" ]
   }
 }
 
 if (rtc_include_tests) {
-  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
-        "-Wno-sign-compare",
-        "-Wno-unused-function",
-      ]
-    }
-
-    if (!is_win) {
-      cflags = [ "-Wno-sign-compare" ]
-    }
-  }
-
-  rtc_test("peerconnection_unittests") {
-    check_includes = false  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
-    testonly = true
-    sources = [
-      "datachannel_unittest.cc",
-      "dtmfsender_unittest.cc",
-      "fakemetricsobserver.cc",
-      "fakemetricsobserver.h",
-      "jsepsessiondescription_unittest.cc",
-      "localaudiosource_unittest.cc",
-      "mediaconstraintsinterface_unittest.cc",
-      "mediastream_unittest.cc",
-      "peerconnection_unittest.cc",
-      "peerconnectionendtoend_unittest.cc",
-      "peerconnectionfactory_unittest.cc",
-      "peerconnectioninterface_unittest.cc",
-      "proxy_unittest.cc",
-      "rtcstats_integrationtest.cc",
-      "rtcstatscollector_unittest.cc",
-      "rtpsenderreceiver_unittest.cc",
-      "sctputils_unittest.cc",
-      "statscollector_unittest.cc",
-      "test/fakeaudiocapturemodule.cc",
-      "test/fakeaudiocapturemodule.h",
-      "test/fakeaudiocapturemodule_unittest.cc",
-      "test/fakeconstraints.h",
-      "test/fakedatachannelprovider.h",
-      "test/fakeperiodicvideocapturer.h",
-      "test/fakertccertificategenerator.h",
-      "test/fakevideotrackrenderer.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",
-      "videocapturertracksource_unittest.cc",
-      "videotrack_unittest.cc",
-      "webrtcsdp_unittest.cc",
-      "webrtcsession_unittest.cc",
-    ]
-
-    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.
-      ]
-    }
-
-    if (rtc_use_quic) {
-      public_deps = [
-        "//third_party/libquic",
-      ]
-      sources += [
-        "quicdatachannel_unittest.cc",
-        "quicdatatransport_unittest.cc",
-      ]
-    }
-
-    deps = []
-    if (is_android) {
-      sources += [
-        "test/androidtestinitializer.cc",
-        "test/androidtestinitializer.h",
-      ]
-      deps += [
-        "//testing/android/native_test:native_test_support",
-        "//webrtc/sdk/android:libjingle_peerconnection_java",
-        "//webrtc/sdk/android:libjingle_peerconnection_jni",
-      ]
-    }
-
-    deps += [
-      ":libjingle_peerconnection",
-      "..:webrtc_common",
-      "../base:rtc_base_tests_utils",
-      "../media:rtc_unittest_main",
-      "../pc:rtc_pc",
-      "../system_wrappers:metrics_default",
-      "//testing/gmock",
-    ]
-
-    if (is_android) {
-      deps += [ "//testing/android/native_test:native_test_support" ]
-
-      shard_timeout = 900
-    }
-  }
-
   rtc_source_set("mock_audio_mixer") {
     testonly = true
     sources = [
       "test/mock_audio_mixer.h",
     ]
 
     public_deps = [
       ":audio_mixer_api",
     ]
 
     deps = [
       "//testing/gmock",
       "//webrtc/test:test_support",
     ]
   }
 }