| # 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") |
| import("//testing/test.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 (build_with_chromium) { |
| sources += [ |
| "externalhmac.cc", |
| "externalhmac.h", |
| ] |
| } |
| 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 the Chromium Clang plugin (bugs.webrtc.org/163). |
| configs -= [ "//build/config/clang:find_bad_constructs" ] |
| } |
| } |
| |
| if (rtc_include_tests) { |
| config("rtc_pc_unittests_config") { |
| # GN orders flags on a target before flags from configs. The default config |
| # adds -Wall, and this flag have to be after -Wall -- so they need to |
| # come from a config and can't be on the target directly. |
| if (!is_win && !is_clang) { |
| cflags = [ "-Wno-maybe-uninitialized" ] # Only exists for GCC. |
| } |
| } |
| |
| test("rtc_pc_unittests") { |
| testonly = true |
| |
| sources = [ |
| "bundlefilter_unittest.cc", |
| "channel_unittest.cc", |
| "channelmanager_unittest.cc", |
| "currentspeakermonitor_unittest.cc", |
| "mediasession_unittest.cc", |
| "rtcpmuxfilter_unittest.cc", |
| "srtpfilter_unittest.cc", |
| ] |
| |
| include_dirs = [ "//third_party/libsrtp/srtp" ] |
| |
| configs += [ |
| "..:common_config", |
| ":rtc_pc_unittests_config", |
| ] |
| public_configs = [ "..:common_inherited_config" ] |
| |
| if (is_clang) { |
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| configs -= [ "//build/config/clang:find_bad_constructs" ] |
| } |
| |
| if (is_win) { |
| libs = [ "strmiids.lib" ] |
| } |
| |
| deps = [ |
| ":rtc_pc", |
| "../api:libjingle_peerconnection", |
| "../base:rtc_base_tests_utils", |
| "../media:rtc_unittest_main", |
| "../system_wrappers:metrics_default", |
| ] |
| |
| if (rtc_build_libsrtp) { |
| deps += [ "//third_party/libsrtp" ] |
| } |
| |
| if (is_android) { |
| deps += [ "//testing/android/native_test:native_test_support" ] |
| } |
| } |
| } |