Create //webrtc/api:libjingle_peerconnection_api + refactorings.

webrtc/api/BUILD.gn

Index: webrtc/api/BUILD.gn
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index fa3c62952464d4cd815b29a7bcdd1d85ac1bf410..9d893e0c01fdce5765d1289702fc2cc4b1d8de8f 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -14,7 +14,7 @@ if (is_android) {
 
 group("api") {
   public_deps = [
-    ":libjingle_peerconnection",
+    ":libjingle_peerconnection_api",
   ]
 }
 
@@ -34,122 +34,63 @@ rtc_source_set("call_api") {
   ]
 }
 
-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",
     "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.cc",
     "mediastreaminterface.h",
-    "mediastreamobserver.cc",
-    "mediastreamobserver.h",
     "mediastreamproxy.h",
     "mediastreamtrack.h",
     "mediastreamtrackproxy.h",
+    "mediatypes.cc",
+    "mediatypes.h",
     "notifier.h",
-    "ortcfactory.cc",
-    "ortcfactory.h",
     "ortcfactoryinterface.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",
     "trackmediainfomap.cc",
     "trackmediainfomap.h",
     "udptransportinterface.h",
-    "videocapturertracksource.cc",
-    "videocapturertracksource.h",
+    "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") {
@@ -157,6 +98,7 @@ rtc_source_set("rtc_stats_api") {
   sources = [
     "stats/rtcstats.h",
     "stats/rtcstats_objects.h",
+    "stats/rtcstatscollectorcallback.h",
     "stats/rtcstatsreport.h",
   ]
 
@@ -210,146 +152,34 @@ rtc_source_set("video_frame_api") {
 }
 
 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)
+  rtc_source_set("mock_audio_mixer") {
     testonly = true
     sources = [
-      "datachannel_unittest.cc",
-      "dtmfsender_unittest.cc",
-      "jsepsessiondescription_unittest.cc",
-      "localaudiosource_unittest.cc",
-      "mediaconstraintsinterface_unittest.cc",
-      "mediastream_unittest.cc",
-      "ortcfactory_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_rtpreceiver.h",
-      "test/mock_rtpsender.h",
-      "test/mock_webrtcsession.h",
-      "test/mockpeerconnectionobservers.h",
-      "test/peerconnectiontestwrapper.cc",
-      "test/peerconnectiontestwrapper.h",
-      "test/rtcstatsobtainer.h",
-      "test/testsdpstrings.h",
-      "trackmediainfomap_unittest.cc",
-      "videocapturertracksource_unittest.cc",
-      "videotrack_unittest.cc",
-      "webrtcsdp_unittest.cc",
-      "webrtcsession_unittest.cc",
+      "test/mock_audio_mixer.h",
     ]
 
-    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.
-      ]
-    }
-
-    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",
-      ]
-    }
+    public_deps = [
+      ":audio_mixer_api",
+    ]
 
-    deps += [
-      ":fakemetricsobserver",
-      ":libjingle_peerconnection",
-      "..:webrtc_common",
-      "../base:rtc_base_tests_utils",
-      "../media:rtc_unittest_main",
-      "../pc:rtc_pc",
-      "../system_wrappers:metrics_default",
+    deps = [
       "//testing/gmock",
+      "//webrtc/test:test_support",
     ]
-
-    if (is_android) {
-      deps += [ "//testing/android/native_test:native_test_support" ]
-
-      shard_timeout = 900
-    }
   }
 
-  rtc_source_set("mock_audio_mixer") {
+  rtc_source_set("libjingle_peerconnection_test_api") {
     testonly = true
     sources = [
-      "test/mock_audio_mixer.h",
+      "test/fakeconstraints.h",
     ]
 
     public_deps = [
-      ":audio_mixer_api",
+      ":libjingle_peerconnection_api",
     ]
 
     deps = [
-      "//testing/gmock",
+      "../base:rtc_base_approved",
       "//webrtc/test:test_support",
     ]
   }
@@ -361,7 +191,7 @@ if (rtc_include_tests) {
       "fakemetricsobserver.h",
     ]
     deps = [
-      ":libjingle_peerconnection",
+      ":libjingle_peerconnection_api",
       "../base:rtc_base_approved",
     ]
     if (!build_with_chromium && is_clang) {