bb23c838f5
Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
675 lines
19 KiB
Plaintext
675 lines
19 KiB
Plaintext
# Copyright (c) 2016 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("//build/config/linux/pkg_config.gni")
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import("../webrtc.gni")
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group("media") {
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deps = []
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if (!build_with_mozilla) {
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deps += [
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":rtc_media",
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":rtc_media_base",
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]
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}
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}
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config("rtc_media_defines_config") {
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defines = [
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"HAVE_WEBRTC_VIDEO",
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"HAVE_WEBRTC_VOICE",
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]
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}
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config("rtc_media_warnings_config") {
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# GN orders flags on a target before flags from configs. The default config
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# adds these flags so to cancel them out they need to come from a config and
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# cannot be on the target directly.
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if (!is_win) {
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cflags = [ "-Wno-deprecated-declarations" ]
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}
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}
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rtc_source_set("rtc_h264_profile_id") {
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visibility = [ "*" ]
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sources = [
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"base/h264_profile_level_id.cc",
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"base/h264_profile_level_id.h",
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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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deps = [
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"..:webrtc_common",
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"../api:optional",
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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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}
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rtc_source_set("rtc_media_config") {
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visibility = [ "*" ]
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sources = [
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"base/mediaconfig.h",
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]
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}
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rtc_static_library("rtc_media_base") {
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visibility = [ "*" ]
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allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
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defines = []
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libs = []
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deps = [
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"../api:audio_options_api",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_task_queue",
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"../rtc_base:sanitizer",
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"../rtc_base:sequenced_task_checker",
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"../rtc_base:stringutils",
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]
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sources = [
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"base/adaptedvideotracksource.cc",
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"base/adaptedvideotracksource.h",
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"base/audiosource.h",
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"base/codec.cc",
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"base/codec.h",
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"base/cryptoparams.h",
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"base/device.h",
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"base/mediachannel.cc",
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"base/mediachannel.h",
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"base/mediaconstants.cc",
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"base/mediaconstants.h",
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"base/mediaengine.cc",
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"base/mediaengine.h",
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"base/rtpdataengine.cc",
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"base/rtpdataengine.h",
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"base/rtputils.cc",
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"base/rtputils.h",
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"base/streamparams.cc",
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"base/streamparams.h",
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"base/turnutils.cc",
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"base/turnutils.h",
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"base/videoadapter.cc",
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"base/videoadapter.h",
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"base/videobroadcaster.cc",
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"base/videobroadcaster.h",
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"base/videocapturer.cc",
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"base/videocapturer.h",
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"base/videocapturerfactory.h",
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"base/videocommon.cc",
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"base/videocommon.h",
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"base/videosourcebase.cc",
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"base/videosourcebase.h",
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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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deps += [
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":rtc_h264_profile_id",
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":rtc_media_config",
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"..:webrtc_common",
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"../api:libjingle_peerconnection_api",
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"../api:optional",
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"../api:video_frame_api",
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"../api:video_frame_api_i420",
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"../api/audio_codecs:audio_codecs_api",
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"../api/video_codecs:video_codecs_api",
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"../call:call_interfaces",
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"../call:video_stream_api",
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"../common_video",
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"../modules/audio_processing:audio_processing_statistics",
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"../rtc_base:rtc_base",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers:field_trial_api",
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]
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if (!build_with_mozilla) {
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deps += [ "../p2p" ]
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}
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if (is_nacl) {
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deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
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}
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}
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rtc_static_library("rtc_constants") {
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defines = []
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libs = []
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deps = []
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sources = [
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"engine/constants.cc",
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"engine/constants.h",
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]
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}
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rtc_static_library("rtc_internal_video_codecs") {
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visibility = [ "*" ]
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allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
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defines = []
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libs = []
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deps = [
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":rtc_h264_profile_id",
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"../modules/video_coding:video_codec_interface",
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]
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sources = [
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"engine/convert_legacy_video_factory.cc",
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"engine/convert_legacy_video_factory.h",
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"engine/internaldecoderfactory.cc",
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"engine/internaldecoderfactory.h",
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"engine/internalencoderfactory.cc",
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"engine/internalencoderfactory.h",
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"engine/multiplexcodecfactory.cc",
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"engine/multiplexcodecfactory.h",
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"engine/scopedvideodecoder.cc",
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"engine/scopedvideodecoder.h",
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"engine/scopedvideoencoder.cc",
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"engine/scopedvideoencoder.h",
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"engine/simulcast_encoder_adapter.cc",
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"engine/simulcast_encoder_adapter.h",
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"engine/vp8_encoder_simulcast_proxy.cc",
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"engine/vp8_encoder_simulcast_proxy.h",
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"engine/webrtcvideodecoderfactory.h",
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"engine/webrtcvideoencoderfactory.h",
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]
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configs += [ ":rtc_media_warnings_config" ]
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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_win) {
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cflags = [
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"/wd4245", # conversion from "int" to "size_t", signed/unsigned mismatch.
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"/wd4267", # conversion from "size_t" to "int", possible loss of data.
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"/wd4389", # signed/unsigned mismatch.
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]
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}
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include_dirs = []
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public_configs = []
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deps += [
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":rtc_constants",
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":rtc_media_base",
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":rtc_software_fallback_wrappers",
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"..:webrtc_common",
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"../api:video_frame_api_i420",
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"../api/video_codecs:video_codecs_api",
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"../call:call_interfaces",
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"../call:video_stream_api",
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"../modules/video_coding:webrtc_h264",
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"../modules/video_coding:webrtc_multiplex",
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"../modules/video_coding:webrtc_vp8",
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"../modules/video_coding:webrtc_vp8_helpers",
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"../modules/video_coding:webrtc_vp9",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:sequenced_task_checker",
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"../system_wrappers",
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"../system_wrappers:field_trial_api",
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"../system_wrappers:metrics_api",
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"//third_party/libyuv",
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]
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}
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rtc_static_library("rtc_software_fallback_wrappers") {
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sources = [
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"engine/videodecodersoftwarefallbackwrapper.cc",
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"engine/videodecodersoftwarefallbackwrapper.h",
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"engine/videoencodersoftwarefallbackwrapper.cc",
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"engine/videoencodersoftwarefallbackwrapper.h",
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]
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configs += [ ":rtc_media_warnings_config" ]
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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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deps = [
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":rtc_h264_profile_id",
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":rtc_media_base",
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"../api/video_codecs:video_codecs_api",
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"../modules/video_coding:video_codec_interface",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base/system:fallthrough",
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"../system_wrappers:field_trial_api",
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]
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}
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rtc_static_library("rtc_audio_video") {
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visibility = [ "*" ]
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allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
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defines = []
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libs = []
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deps = [
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"../modules/audio_processing/aec_dump:aec_dump",
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"../modules/video_coding:video_codec_interface",
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"../modules/video_coding:video_coding",
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"../rtc_base:audio_format_to_string",
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"../rtc_base:checks",
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]
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sources = [
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"engine/adm_helpers.cc",
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"engine/adm_helpers.h",
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"engine/apm_helpers.cc",
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"engine/apm_helpers.h",
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"engine/nullwebrtcvideoengine.h",
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"engine/payload_type_mapper.cc",
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"engine/payload_type_mapper.h",
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"engine/simulcast.cc",
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"engine/simulcast.h",
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"engine/webrtcmediaengine.cc",
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"engine/webrtcmediaengine.h",
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"engine/webrtcvideocapturer.cc",
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"engine/webrtcvideocapturer.h",
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"engine/webrtcvideocapturerfactory.cc",
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"engine/webrtcvideocapturerfactory.h",
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"engine/webrtcvideodecoderfactory.cc",
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"engine/webrtcvideodecoderfactory.h",
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"engine/webrtcvideoencoderfactory.cc",
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"engine/webrtcvideoencoderfactory.h",
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"engine/webrtcvideoengine.cc",
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"engine/webrtcvideoengine.h",
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"engine/webrtcvoiceengine.cc",
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"engine/webrtcvoiceengine.h",
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]
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configs += [ ":rtc_media_warnings_config" ]
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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_win) {
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cflags = [
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"/wd4245", # conversion from "int" to "size_t", signed/unsigned mismatch.
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"/wd4267", # conversion from "size_t" to "int", possible loss of data.
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"/wd4389", # signed/unsigned mismatch.
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]
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}
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if (rtc_enable_intelligibility_enhancer) {
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defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
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} else {
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defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
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}
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if (rtc_opus_support_120ms_ptime) {
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defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
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} else {
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defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
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}
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include_dirs = []
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public_configs = []
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if (build_with_chromium) {
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deps += [ "../modules/video_capture:video_capture" ]
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} else {
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public_configs += [ ":rtc_media_defines_config" ]
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deps += [ "../modules/video_capture:video_capture_internal_impl" ]
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}
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if (rtc_enable_protobuf) {
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deps += [ "../modules/audio_processing/aec_dump:aec_dump_impl" ]
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} else {
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deps += [ "../modules/audio_processing/aec_dump:null_aec_dump_factory" ]
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}
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if (rtc_use_builtin_sw_codecs) {
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deps += [ ":rtc_internal_video_codecs" ]
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}
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deps += [
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":rtc_constants",
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":rtc_media_base",
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":rtc_software_fallback_wrappers",
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"..:webrtc_common",
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"../api:call_api",
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"../api:libjingle_peerconnection_api",
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"../api:optional",
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"../api:transport_api",
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"../api:video_frame_api",
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"../api:video_frame_api_i420",
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"../api/audio_codecs:audio_codecs_api",
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"../api/video_codecs:video_codecs_api",
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"../call",
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"../call:call_interfaces",
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"../call:video_stream_api",
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"../common_video:common_video",
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"../modules/audio_device:audio_device",
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"../modules/audio_device:audio_device_impl",
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"../modules/audio_mixer:audio_mixer_impl",
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"../modules/audio_processing:audio_processing",
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"../modules/video_capture:video_capture_module",
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"../pc:rtc_pc_base",
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"../rtc_base:rtc_base",
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"../rtc_base:rtc_task_queue",
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"../rtc_base:stringutils",
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"../system_wrappers",
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"../system_wrappers:field_trial_api",
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"../system_wrappers:metrics_api",
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]
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}
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rtc_static_library("rtc_data") {
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defines = []
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deps = []
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if (rtc_enable_sctp) {
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sources = [
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"sctp/sctptransport.cc",
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"sctp/sctptransport.h",
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"sctp/sctptransportinternal.h",
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]
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} else {
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# libtool on mac does not like empty targets.
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sources = [
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"sctp/noop.cc",
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]
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}
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configs += [ ":rtc_media_warnings_config" ]
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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_win) {
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cflags = [
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"/wd4245", # conversion from "int" to "size_t", signed/unsigned mismatch.
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"/wd4267", # conversion from "size_t" to "int", possible loss of data.
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"/wd4389", # signed/unsigned mismatch.
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]
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}
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if (rtc_enable_sctp && rtc_build_usrsctp) {
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include_dirs = [
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# TODO(jiayl): move this into the public_configs of
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# //third_party/usrsctp/BUILD.gn.
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"//third_party/usrsctp/usrsctplib",
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]
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deps += [ "//third_party/usrsctp" ]
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}
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deps += [
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":rtc_media_base",
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"..:webrtc_common",
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"../api:call_api",
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"../api:transport_api",
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"../p2p:rtc_p2p",
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"../rtc_base:rtc_base",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers",
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]
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}
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rtc_source_set("rtc_media") {
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visibility = [ "*" ]
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allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
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deps = [
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":rtc_audio_video",
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":rtc_data",
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]
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}
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if (rtc_include_tests) {
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config("rtc_unittest_main_config") {
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# GN orders flags on a target before flags from configs. The default config
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# adds -Wall, and this flag have to be after -Wall -- so they need to
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# come from a config and can"t be on the target directly.
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if (is_clang && is_ios) {
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cflags = [ "-Wno-unused-variable" ]
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}
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}
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rtc_source_set("rtc_media_tests_utils") {
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testonly = true
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include_dirs = []
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deps = [
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":rtc_audio_video",
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"../api:libjingle_peerconnection_api",
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"../api:video_frame_api_i420",
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"../call:video_stream_api",
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"../common_video:common_video",
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"../modules/audio_coding:rent_a_codec",
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"../modules/audio_processing:audio_processing",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/video_coding:video_codec_interface",
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"../modules/video_coding:video_coding_utility",
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"../p2p:rtc_p2p",
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"../rtc_base:checks",
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"../rtc_base:rtc_task_queue",
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"../rtc_base:stringutils",
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]
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sources = [
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"base/fakeframesource.cc",
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"base/fakeframesource.h",
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"base/fakemediaengine.h",
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"base/fakenetworkinterface.h",
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"base/fakertp.cc",
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"base/fakertp.h",
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"base/fakevideocapturer.cc",
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"base/fakevideocapturer.h",
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"base/fakevideorenderer.h",
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"base/testutils.cc",
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"base/testutils.h",
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"engine/fakewebrtccall.cc",
|
|
"engine/fakewebrtccall.h",
|
|
"engine/fakewebrtcdeviceinfo.h",
|
|
"engine/fakewebrtcvcmfactory.h",
|
|
"engine/fakewebrtcvideocapturemodule.h",
|
|
"engine/fakewebrtcvideoengine.h",
|
|
]
|
|
|
|
configs += [ ":rtc_unittest_main_config" ]
|
|
|
|
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" ]
|
|
}
|
|
|
|
deps += [
|
|
":rtc_internal_video_codecs",
|
|
":rtc_media",
|
|
":rtc_media_base",
|
|
"..:webrtc_common",
|
|
"../api:call_api",
|
|
"../api:video_frame_api",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../call:call_interfaces",
|
|
"../call:mock_rtp_interfaces",
|
|
"../rtc_base:rtc_base",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_base_tests_utils",
|
|
"../rtc_base:rtc_task_queue_for_test",
|
|
"../test:test_support",
|
|
"//testing/gtest",
|
|
]
|
|
}
|
|
|
|
config("rtc_media_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.
|
|
# TODO(kjellander): Make the code compile without disabling these flags.
|
|
# See https://bugs.webrtc.org/3307.
|
|
if (is_clang && is_win) {
|
|
cflags = [
|
|
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=6266
|
|
# for -Wno-sign-compare
|
|
"-Wno-sign-compare",
|
|
]
|
|
}
|
|
if (!is_win) {
|
|
cflags = [ "-Wno-sign-compare" ]
|
|
}
|
|
}
|
|
|
|
rtc_media_unittests_resources = [
|
|
"../resources/media/captured-320x240-2s-48.frames",
|
|
"../resources/media/faces.1280x720_P420.yuv",
|
|
"../resources/media/faces_I420.jpg",
|
|
"../resources/media/faces_I422.jpg",
|
|
"../resources/media/faces_I444.jpg",
|
|
"../resources/media/faces_I411.jpg",
|
|
"../resources/media/faces_I400.jpg",
|
|
]
|
|
|
|
if (is_ios) {
|
|
bundle_data("rtc_media_unittests_bundle_data") {
|
|
testonly = true
|
|
sources = rtc_media_unittests_resources
|
|
outputs = [
|
|
"{{bundle_resources_dir}}/{{source_file_part}}",
|
|
]
|
|
}
|
|
}
|
|
|
|
rtc_test("rtc_media_unittests") {
|
|
testonly = true
|
|
|
|
defines = []
|
|
deps = [
|
|
":rtc_audio_video",
|
|
":rtc_constants",
|
|
":rtc_data",
|
|
"../api:video_frame_api_i420",
|
|
"../modules/audio_processing:mocks",
|
|
"../modules/video_coding:video_codec_interface",
|
|
"../pc:rtc_pc",
|
|
"../pc:rtc_pc_base",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:rtc_task_queue",
|
|
"../rtc_base:stringutils",
|
|
"../test:field_trial",
|
|
]
|
|
sources = [
|
|
"base/codec_unittest.cc",
|
|
"base/rtpdataengine_unittest.cc",
|
|
"base/rtputils_unittest.cc",
|
|
"base/streamparams_unittest.cc",
|
|
"base/turnutils_unittest.cc",
|
|
"base/videoadapter_unittest.cc",
|
|
"base/videobroadcaster_unittest.cc",
|
|
"base/videocapturer_unittest.cc",
|
|
"base/videocommon_unittest.cc",
|
|
"engine/apm_helpers_unittest.cc",
|
|
"engine/internaldecoderfactory_unittest.cc",
|
|
"engine/multiplexcodecfactory_unittest.cc",
|
|
"engine/nullwebrtcvideoengine_unittest.cc",
|
|
"engine/payload_type_mapper_unittest.cc",
|
|
"engine/simulcast_encoder_adapter_unittest.cc",
|
|
"engine/simulcast_unittest.cc",
|
|
"engine/videodecodersoftwarefallbackwrapper_unittest.cc",
|
|
"engine/videoencodersoftwarefallbackwrapper_unittest.cc",
|
|
"engine/vp8_encoder_simulcast_proxy_unittest.cc",
|
|
"engine/webrtcmediaengine_unittest.cc",
|
|
"engine/webrtcvideocapturer_unittest.cc",
|
|
"engine/webrtcvideoencoderfactory_unittest.cc",
|
|
"engine/webrtcvideoengine_unittest.cc",
|
|
]
|
|
|
|
# TODO(kthelgason): Reenable this test on iOS.
|
|
# See bugs.webrtc.org/5569
|
|
if (!is_ios) {
|
|
sources += [ "engine/webrtcvoiceengine_unittest.cc" ]
|
|
}
|
|
|
|
if (rtc_enable_sctp) {
|
|
sources += [ "sctp/sctptransport_unittest.cc" ]
|
|
}
|
|
|
|
configs += [ ":rtc_media_unittests_config" ]
|
|
|
|
if (rtc_use_h264) {
|
|
defines += [ "WEBRTC_USE_H264" ]
|
|
}
|
|
|
|
if (rtc_opus_support_120ms_ptime) {
|
|
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
|
|
} else {
|
|
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
|
|
}
|
|
|
|
if (is_win) {
|
|
cflags = [
|
|
"/wd4245", # conversion from int to size_t, signed/unsigned mismatch.
|
|
"/wd4389", # signed/unsigned mismatch.
|
|
]
|
|
}
|
|
|
|
if (!build_with_chromium && is_clang) {
|
|
suppressed_configs += [
|
|
"//build/config/clang:extra_warnings",
|
|
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
"//build/config/clang:find_bad_constructs",
|
|
]
|
|
}
|
|
|
|
data = rtc_media_unittests_resources
|
|
|
|
if (is_android) {
|
|
deps += [ "//testing/android/native_test:native_test_support" ]
|
|
shard_timeout = 900
|
|
}
|
|
|
|
if (is_ios) {
|
|
deps += [ ":rtc_media_unittests_bundle_data" ]
|
|
}
|
|
|
|
deps += [
|
|
":rtc_internal_video_codecs",
|
|
":rtc_media",
|
|
":rtc_media_base",
|
|
":rtc_media_tests_utils",
|
|
":rtc_software_fallback_wrappers",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:mock_video_codec_factory",
|
|
"../api:video_frame_api",
|
|
"../api/audio_codecs:builtin_audio_decoder_factory",
|
|
"../api/audio_codecs:builtin_audio_encoder_factory",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../audio",
|
|
"../call:call_interfaces",
|
|
"../common_video:common_video",
|
|
"../logging:rtc_event_log_api",
|
|
"../logging:rtc_event_log_impl_base",
|
|
"../modules/audio_device:mock_audio_device",
|
|
"../modules/audio_processing:audio_processing",
|
|
"../modules/video_coding:simulcast_test_utility",
|
|
"../modules/video_coding:video_coding_utility",
|
|
"../modules/video_coding:webrtc_vp8_helpers",
|
|
"../p2p:p2p_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",
|
|
"../test:audio_codec_mocks",
|
|
"../test:test_support",
|
|
"../test:video_test_common",
|
|
]
|
|
}
|
|
}
|