Files
platform-external-webrtc/ortc/BUILD.gn
Ilya Nikolaevskiy 2ffe3e80db Reland Use runtime enabled features API to enable dual stream mode
This is an unchanged patch after dependency fixes in downstream projects are implemented.

Original patch was reviewed here:
https://webrtc-review.googlesource.com/c/src/+/39008

TBR=phoglund@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org,philipel@webrtc.org,lliuu@webrtc.org

Change-Id: I648bbf63d34282a48cabc854615005ec65b28cb3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8287
Reviewed-on: https://webrtc-review.googlesource.com/40420
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21674}
2018-01-18 12:22:49 +00:00

107 lines
3.2 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",
"../call:call_interfaces",
"../call:rtp_sender",
"../logging:rtc_event_log_api",
"../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",
"../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" ]
}
}
}