PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added.

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) {
@@ skipping 66 matching lines @@
     "peerconnection.cc",
     "peerconnection.h",
     "peerconnectionfactory.cc",
     "peerconnectionfactory.h",
     "peerconnectionfactoryproxy.h",
     "peerconnectioninterface.h",
     "peerconnectionproxy.h",
     "proxy.h",
     "remoteaudiosource.cc",
     "remoteaudiosource.h",
-    "rtcstats.h",
-    "rtcstats_objects.h",
-    "rtcstatsreport.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",
@@ skipping 21 matching lines @@
   if (is_clang) {
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 
   deps = [
     ":call_api",
     "../call",
     "../media",
     "../pc",
+    "../stats",
   ]
 
   if (rtc_use_quic) {
     sources += [
       "quicdatachannel.cc",
       "quicdatachannel.h",
       "quicdatatransport.cc",
       "quicdatatransport.h",
     ]
     deps += [ "//third_party/libquic" ]
@@ skipping 150 matching lines @@
       "android/java/src/org/webrtc/VideoSource.java",
       "android/java/src/org/webrtc/VideoTrack.java",
     ]
 
     deps = [
       "//webrtc/base:base_java",
     ]
   }
 }
 
+# GYP version: webrtc/api/api.gyp:rtc_stats_api
+rtc_source_set("rtc_stats_api") {
+  cflags = []
+  sources = [
+    "stats/rtcstats.h",
+    "stats/rtcstats_objects.h",
+    "stats/rtcstatsreport.h",
+  ]
+
+  if (is_clang) {
+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]

[kjellander_webrtc, 2016/09/14 13:55:27]

For new code, I won't tolerate violating these excellent warnings. Are they hard
to fix?

[hbos, 2016/09/14 18:48:03]

Removed. It's not necessary in the header target, but it is necessary in the
rtc_stats target (there since before this CL). Now that rtc_stats does not
include libjingle_peerconnection stuff this can be addressed. I should do that
in a follow-up CL though because it requires changing a macro used (not simple
enough to do in this CL).

[kjellander_webrtc, 2016/09/15 10:16:37]

Acknowledged.

+  }
+
+  deps = [
+    "../base:rtc_base_approved",
+  ]
+}
+
 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
@@ skipping 18 matching lines @@
       "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",
+      "rtcstatscollector_unittest.cc",

[kjellander_webrtc, 2016/09/14 13:55:27]

nit: sort alphabetically.

[hbos, 2016/09/14 18:48:03]

It is sorted.

[kjellander_webrtc, 2016/09/15 10:16:37]

Sorry, I misread rtc/rtp :P

       "rtpsenderreceiver_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",
@@ skipping 97 matching lines @@
 
       shared_libraries = [ ":libjingle_peerconnection_so" ]
     }
 
     android_resources("libjingle_peerconnection_android_unittest_resources") {
       resource_dirs = [ "androidtests/res" ]
       custom_package = "org.webrtc"
     }
   }
 }