GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}

webrtc/pc/BUILD.gn

Index: webrtc/pc/BUILD.gn
diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn
new file mode 100644
index 0000000000000000000000000000000000000000..50bb26ad4641239e8244a364e0873dfb636fff91
--- /dev/null
+++ b/webrtc/pc/BUILD.gn
@@ -0,0 +1,70 @@
+# 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("../build/webrtc.gni")
+
+group("pc") {
+  deps = [
+    ":rtc_pc",
+  ]
+}
+
+config("rtc_pc_config") {
+  defines = [
+    "SRTP_RELATIVE_PATH",
+    "HAVE_SCTP",
+    "HAVE_SRTP",
+  ]
+}
+
+source_set("rtc_pc") {
+  defines = []
+  sources = [
+    "audiomonitor.cc",
+    "audiomonitor.h",
+    "bundlefilter.cc",
+    "bundlefilter.h",
+    "channel.cc",
+    "channel.h",
+    "channelmanager.cc",
+    "channelmanager.h",
+    "currentspeakermonitor.cc",
+    "currentspeakermonitor.h",
+    "mediamonitor.cc",
+    "mediamonitor.h",
+    "mediasession.cc",
+    "mediasession.h",
+    "mediasink.h",
+    "rtcpmuxfilter.cc",
+    "rtcpmuxfilter.h",
+    "srtpfilter.cc",
+    "srtpfilter.h",
+    "voicechannel.h",
+  ]
+
+  deps = [
+    "../base:rtc_base",
+    "../media",
+  ]
+
+  if (rtc_build_libsrtp) {
+    deps += [ "//third_party/libsrtp" ]
+  }
+
+  configs += [ "..:common_config" ]
+  public_configs = [
+    "..:common_inherited_config",
+    ":rtc_pc_config",
+  ]
+
+  if (is_clang) {
+    # Suppress warnings from Chrome's Clang plugins.
+    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+    configs -= [ "//build/config/clang:find_bad_constructs" ]
+  }
+}