b330688ef7
The previous effort of building WebRTC without SW codecs stopped when
libjingle_peerconnection was possible to build. In order to make the
group("default") target build, this basically updates a bunch of
tests to explicitly depend on the built-in software video codecs.
Bug: webrtc:7925
Change-Id: I2715414770c197fca01cb8dbde173a21f4434500
Reviewed-on: https://webrtc-review.googlesource.com/70503
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23216}
111 lines
3.4 KiB
Plaintext
111 lines
3.4 KiB
Plaintext
# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../webrtc.gni")
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if (is_android) {
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import("//build/config/android/config.gni")
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import("//build/config/android/rules.gni")
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}
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rtc_static_library("ortc") {
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defines = []
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sources = [
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"ortcfactory.cc",
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"ortcfactory.h",
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"ortcrtpreceiveradapter.cc",
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"ortcrtpreceiveradapter.h",
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"ortcrtpsenderadapter.cc",
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"ortcrtpsenderadapter.h",
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"rtpparametersconversion.cc",
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"rtpparametersconversion.h",
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"rtptransportadapter.cc",
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"rtptransportadapter.h",
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"rtptransportcontrolleradapter.cc",
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"rtptransportcontrolleradapter.h",
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]
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# TODO(deadbeef): Create a separate target for the common things ORTC and
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# PeerConnection code shares, so that ortc can depend on that instead of
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# libjingle_peerconnection.
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deps = [
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"../api:libjingle_peerconnection_api",
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"../api:optional",
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"../api:ortc_api",
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"../api/video_codecs:builtin_video_decoder_factory",
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"../api/video_codecs:builtin_video_encoder_factory",
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"../call:call_interfaces",
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"../call:rtp_sender",
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"../logging:rtc_event_log_api",
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"../logging:rtc_event_log_impl_base",
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"../media:rtc_audio_video",
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"../media:rtc_media",
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"../media:rtc_media_base",
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"../modules/audio_processing:audio_processing",
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"../p2p:rtc_p2p",
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"../pc:libjingle_peerconnection",
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"../pc:peerconnection",
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"../pc:rtc_pc",
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"../pc:rtc_pc_base",
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"../rtc_base:checks",
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"../rtc_base:rtc_base",
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"../rtc_base:rtc_base_approved",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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if (rtc_include_tests) {
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rtc_test("ortc_unittests") {
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testonly = true
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sources = [
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"ortcfactory_integrationtest.cc",
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"ortcfactory_unittest.cc",
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"ortcrtpreceiver_unittest.cc",
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"ortcrtpsender_unittest.cc",
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"rtpparametersconversion_unittest.cc",
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"rtptransport_unittest.cc",
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"rtptransportcontroller_unittest.cc",
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"srtptransport_unittest.cc",
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"testrtpparameters.cc",
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"testrtpparameters.h",
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]
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deps = [
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":ortc",
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"../api:libjingle_peerconnection_api",
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"../api:ortc_api",
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"../api/audio_codecs:builtin_audio_decoder_factory",
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"../api/audio_codecs:builtin_audio_encoder_factory",
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"../media:rtc_media_tests_utils",
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"../p2p:p2p_test_utils",
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"../p2p:rtc_p2p",
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"../pc:pc_test_utils",
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"../pc:peerconnection",
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"../rtc_base:rtc_base",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_base_tests_main",
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"../rtc_base:rtc_base_tests_utils",
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"../system_wrappers:metrics_default",
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"../system_wrappers:runtime_enabled_features_default",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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if (is_android) {
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deps += [ "//testing/android/native_test:native_test_support" ]
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}
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}
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}
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