Index: webrtc/pc/BUILD.gn |
diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn |
deleted file mode 100644 |
index 50bb26ad4641239e8244a364e0873dfb636fff91..0000000000000000000000000000000000000000 |
--- a/webrtc/pc/BUILD.gn |
+++ /dev/null |
@@ -1,70 +0,0 @@ |
-# 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" ] |
- } |
-} |