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| 1 # Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | |
| 2 # | |
| 3 # Use of this source code is governed by a BSD-style license | |
| 4 # that can be found in the LICENSE file in the root of the source | |
| 5 # tree. An additional intellectual property rights grant can be found | |
| 6 # in the file PATENTS. All contributing project authors may | |
| 7 # be found in the AUTHORS file in the root of the source tree. | |
| 8 | |
| 9 import("../build/webrtc.gni") | |
| 10 | |
| 11 group("pc") { | |
| 12 deps = [ | |
| 13 ":rtc_pc", | |
| 14 ] | |
| 15 } | |
| 16 | |
| 17 config("rtc_pc_config") { | |
| 18 defines = [ | |
| 19 "SRTP_RELATIVE_PATH", | |
| 20 "HAVE_SCTP", | |
| 21 "HAVE_SRTP", | |
| 22 ] | |
| 23 } | |
| 24 | |
| 25 source_set("rtc_pc") { | |
| 26 defines = [] | |
| 27 sources = [ | |
| 28 "audiomonitor.cc", | |
| 29 "audiomonitor.h", | |
| 30 "bundlefilter.cc", | |
| 31 "bundlefilter.h", | |
| 32 "channel.cc", | |
| 33 "channel.h", | |
| 34 "channelmanager.cc", | |
| 35 "channelmanager.h", | |
| 36 "currentspeakermonitor.cc", | |
| 37 "currentspeakermonitor.h", | |
| 38 "mediamonitor.cc", | |
| 39 "mediamonitor.h", | |
| 40 "mediasession.cc", | |
| 41 "mediasession.h", | |
| 42 "mediasink.h", | |
| 43 "rtcpmuxfilter.cc", | |
| 44 "rtcpmuxfilter.h", | |
| 45 "srtpfilter.cc", | |
| 46 "srtpfilter.h", | |
| 47 "voicechannel.h", | |
| 48 ] | |
| 49 | |
| 50 deps = [ | |
| 51 "../base:rtc_base", | |
| 52 "../media", | |
| 53 ] | |
| 54 | |
| 55 if (rtc_build_libsrtp) { | |
| 56 deps += [ "//third_party/libsrtp" ] | |
| 57 } | |
| 58 | |
| 59 configs += [ "..:common_config" ] | |
| 60 public_configs = [ | |
| 61 "..:common_inherited_config", | |
| 62 ":rtc_pc_config", | |
| 63 ] | |
| 64 | |
| 65 if (is_clang) { | |
| 66 # Suppress warnings from Chrome's Clang plugins. | |
| 67 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. | |
| 68 configs -= [ "//build/config/clang:find_bad_constructs" ] | |
| 69 } | |
| 70 } | |
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Chromium Code Reviews
Issue 1975223002:
Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master