mirror of
https://github.com/klzgrad/naiveproxy.git
synced 2024-12-03 02:36:09 +03:00
111 lines
3.4 KiB
Plaintext
111 lines
3.4 KiB
Plaintext
|
# 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:libjingle_peerconnection_api",
|
||
|
"../api:optional",
|
||
|
"../api:ortc_api",
|
||
|
"../api/video_codecs:builtin_video_decoder_factory",
|
||
|
"../api/video_codecs:builtin_video_encoder_factory",
|
||
|
"../call:call_interfaces",
|
||
|
"../call:rtp_sender",
|
||
|
"../logging:rtc_event_log_api",
|
||
|
"../logging:rtc_event_log_impl_base",
|
||
|
"../media:rtc_audio_video",
|
||
|
"../media:rtc_media",
|
||
|
"../media:rtc_media_base",
|
||
|
"../modules/audio_processing:audio_processing",
|
||
|
"../p2p:rtc_p2p",
|
||
|
"../pc:libjingle_peerconnection",
|
||
|
"../pc:peerconnection",
|
||
|
"../pc:rtc_pc",
|
||
|
"../pc:rtc_pc_base",
|
||
|
"../rtc_base:checks",
|
||
|
"../rtc_base:rtc_base",
|
||
|
"../rtc_base:rtc_base_approved",
|
||
|
]
|
||
|
|
||
|
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",
|
||
|
"../api:libjingle_peerconnection_api",
|
||
|
"../api:ortc_api",
|
||
|
"../api/audio_codecs:builtin_audio_decoder_factory",
|
||
|
"../api/audio_codecs:builtin_audio_encoder_factory",
|
||
|
"../media:rtc_media_tests_utils",
|
||
|
"../p2p:p2p_test_utils",
|
||
|
"../p2p:rtc_p2p",
|
||
|
"../pc:pc_test_utils",
|
||
|
"../pc:peerconnection",
|
||
|
"../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",
|
||
|
"../system_wrappers:runtime_enabled_features_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" ]
|
||
|
}
|
||
|
}
|
||
|
}
|