blob: 485771c9a412c7df473d01a244450b5a0ab4cdec [file] [log] [blame]
# Copyright (c) 2017 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("../webrtc.gni")
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
rtc_static_library("ortc") {
defines = []
sources = [
"ortcfactory.cc",
"ortcfactory.h",
"ortcrtpreceiveradapter.cc",
"ortcrtpreceiveradapter.h",
"ortcrtpsenderadapter.cc",
"ortcrtpsenderadapter.h",
"rtpparametersconversion.cc",
"rtpparametersconversion.h",
"rtptransportadapter.cc",
"rtptransportadapter.h",
"rtptransportcontrolleradapter.cc",
"rtptransportcontrolleradapter.h",
]
# TODO(deadbeef): Create a separate target for the common things ORTC and
# PeerConnection code shares, so that ortc can depend on that instead of
# libjingle_peerconnection.
deps = [
"../api:optional",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../call:call_interfaces",
"../call:rtp_sender",
"../logging:rtc_event_log_api",
"../media:rtc_media",
"../media:rtc_media_base",
"../modules/audio_processing:audio_processing",
"../p2p:rtc_p2p",
"../pc:libjingle_peerconnection",
"../pc:rtc_pc",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
]
public_deps = [
"../api:ortc_api",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
if (rtc_include_tests) {
rtc_test("ortc_unittests") {
testonly = true
sources = [
"ortcfactory_integrationtest.cc",
"ortcfactory_unittest.cc",
"ortcrtpreceiver_unittest.cc",
"ortcrtpsender_unittest.cc",
"rtpparametersconversion_unittest.cc",
"rtptransport_unittest.cc",
"rtptransportcontroller_unittest.cc",
"srtptransport_unittest.cc",
"testrtpparameters.cc",
"testrtpparameters.h",
]
deps = [
":ortc",
"../media:rtc_media_tests_utils",
"../p2p:p2p_test_utils",
"../p2p:rtc_p2p",
"../pc:pc_test_utils",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_main",
"../rtc_base:rtc_base_tests_utils",
"../system_wrappers:metrics_default",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
if (is_android) {
deps += [ "//testing/android/native_test:native_test_support" ]
}
}
}