a76af0ca2e
Move base64.h to the proper location and put redirect header into the old place to be able to switch downstream users on new location. Bug: webrtc:8366 Change-Id: I5191fe631d32178d2efd1315ca9abd4250102291 Reviewed-on: https://webrtc-review.googlesource.com/88223 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24069}
563 lines
17 KiB
Plaintext
563 lines
17 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("../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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group("pc") {
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deps = [
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":rtc_pc",
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]
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}
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config("rtc_pc_config") {
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defines = []
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if (rtc_enable_sctp) {
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defines += [ "HAVE_SCTP" ]
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}
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}
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rtc_static_library("rtc_pc_base") {
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visibility = [ "*" ]
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defines = []
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sources = [
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"channel.cc",
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"channel.h",
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"channelmanager.cc",
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"channelmanager.h",
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"dtlssrtptransport.cc",
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"dtlssrtptransport.h",
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"externalhmac.cc",
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"externalhmac.h",
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"jseptransport.cc",
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"jseptransport.h",
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"jseptransportcontroller.cc",
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"jseptransportcontroller.h",
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"mediasession.cc",
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"mediasession.h",
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"rtcpmuxfilter.cc",
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"rtcpmuxfilter.h",
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"rtpmediautils.cc",
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"rtpmediautils.h",
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"rtptransport.cc",
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"rtptransport.h",
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"rtptransportinternal.h",
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"rtptransportinternaladapter.h",
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"sessiondescription.cc",
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"sessiondescription.h",
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"srtpfilter.cc",
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"srtpfilter.h",
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"srtpsession.cc",
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"srtpsession.h",
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"srtptransport.cc",
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"srtptransport.h",
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"transportstats.cc",
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"transportstats.h",
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]
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deps = [
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"..:webrtc_common",
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"../api:array_view",
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"../api:call_api",
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"../api:libjingle_peerconnection_api",
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"../api:ortc_api",
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"../api/video:video_frame",
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"../call:rtp_interfaces",
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"../call:rtp_receiver",
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"../common_video:common_video",
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"../logging:rtc_event_log_api",
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"../media:rtc_data",
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"../media:rtc_h264_profile_id",
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"../media:rtc_media_base",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../p2p:rtc_p2p",
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"../rtc_base:checks",
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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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"../rtc_base/third_party/base64",
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"../system_wrappers:metrics_api",
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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if (rtc_build_libsrtp) {
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deps += [ "//third_party/libsrtp" ]
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}
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public_configs = [ ":rtc_pc_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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}
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rtc_source_set("rtc_pc") {
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visibility = [ "*" ]
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allow_poison = [
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"audio_codecs", # TODO(bugs.webrtc.org/8396): Remove.
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"software_video_codecs", # TODO(bugs.webrtc.org/7925): Remove.
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]
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deps = [
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":rtc_pc_base",
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"../media:rtc_audio_video",
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]
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}
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rtc_static_library("peerconnection") {
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visibility = [ "*" ]
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cflags = []
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sources = [
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"audiotrack.cc",
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"audiotrack.h",
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"datachannel.cc",
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"datachannel.h",
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"dtmfsender.cc",
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"dtmfsender.h",
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"iceserverparsing.cc",
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"iceserverparsing.h",
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"jsepicecandidate.cc",
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"jsepsessiondescription.cc",
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"localaudiosource.cc",
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"localaudiosource.h",
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"mediastream.cc",
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"mediastream.h",
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"mediastreamobserver.cc",
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"mediastreamobserver.h",
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"mediastreamtrack.h",
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"peerconnection.cc",
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"peerconnection.h",
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"peerconnectionfactory.cc",
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"peerconnectionfactory.h",
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"peerconnectioninternal.h",
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"remoteaudiosource.cc",
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"remoteaudiosource.h",
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"rtcstatscollector.cc",
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"rtcstatscollector.h",
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"rtcstatstraversal.cc",
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"rtcstatstraversal.h",
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"rtpparametersconversion.cc",
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"rtpparametersconversion.h",
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"rtpreceiver.cc",
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"rtpreceiver.h",
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"rtpsender.cc",
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"rtpsender.h",
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"rtptransceiver.cc",
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"rtptransceiver.h",
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"sctputils.cc",
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"sctputils.h",
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"sdputils.cc",
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"sdputils.h",
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"statscollector.cc",
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"statscollector.h",
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"streamcollection.h",
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"trackmediainfomap.cc",
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"trackmediainfomap.h",
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"videocapturertracksource.cc",
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"videocapturertracksource.h",
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"videotrack.cc",
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"videotrack.h",
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"videotracksource.cc",
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"videotracksource.h",
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"webrtcsdp.cc",
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"webrtcsdp.h",
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"webrtcsessiondescriptionfactory.cc",
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"webrtcsessiondescriptionfactory.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_pc_base",
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"..:webrtc_common",
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"../api:call_api",
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"../api:fec_controller_api",
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"../api:libjingle_peerconnection_api",
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"../api:rtc_stats_api",
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"../api/video:video_frame",
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"../api/video_codecs:video_codecs_api",
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"../call:call_interfaces",
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"../common_video:common_video",
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"../logging:ice_log",
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"../logging:rtc_event_log_api",
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"../logging:rtc_event_log_impl_output",
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"../media:rtc_data",
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"../media:rtc_media_base",
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"../modules/congestion_controller/bbr",
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"../p2p:rtc_p2p",
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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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"../rtc_base:stringutils",
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"../rtc_base/experiments:congestion_controller_experiment",
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"../rtc_base/third_party/base64",
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"../stats",
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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/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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# This target implements CreatePeerConnectionFactory methods that will create a
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# PeerConnection will full functionality (audio, video and data). Applications
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# that wish to reduce their binary size by ommitting functionality they don't
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# need should use CreateModularCreatePeerConnectionFactory instead, using the
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# "peerconnection" build target and other targets specific to their
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# requrements. See comment in peerconnectionfactoryinterface.h.
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rtc_static_library("create_pc_factory") {
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sources = [
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"createpeerconnectionfactory.cc",
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]
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deps = [
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"../api:callfactory_api",
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"../api:libjingle_peerconnection_api",
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"../api/audio:audio_mixer_api",
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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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"../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_base",
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"../modules/audio_device:audio_device",
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"../modules/audio_processing:audio_processing",
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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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rtc_source_set("libjingle_peerconnection") {
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visibility = [ "*" ]
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allow_poison = [
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"audio_codecs", # TODO(bugs.webrtc.org/8396): Remove.
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"software_video_codecs", # TODO(bugs.webrtc.org/7925): Remove.
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]
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deps = [
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":create_pc_factory",
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":peerconnection",
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"../api:libjingle_peerconnection_api",
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]
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}
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if (rtc_include_tests) {
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rtc_test("rtc_pc_unittests") {
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testonly = true
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sources = [
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"channel_unittest.cc",
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"channelmanager_unittest.cc",
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"dtlssrtptransport_unittest.cc",
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"jseptransport_unittest.cc",
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"jseptransportcontroller_unittest.cc",
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"mediasession_unittest.cc",
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"rtcpmuxfilter_unittest.cc",
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"rtptransport_unittest.cc",
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"rtptransporttestutil.h",
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"srtpfilter_unittest.cc",
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"srtpsession_unittest.cc",
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"srtptestutil.h",
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"srtptransport_unittest.cc",
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]
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include_dirs = [ "//third_party/libsrtp/srtp" ]
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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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libs = [ "strmiids.lib" ]
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}
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deps = [
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":libjingle_peerconnection",
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":pc_test_utils",
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":rtc_pc",
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":rtc_pc_base",
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"../api:array_view",
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"../api:libjingle_peerconnection_api",
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"../call:rtp_interfaces",
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"../logging:rtc_event_log_api",
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"../media:rtc_media_base",
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"../media:rtc_media_tests_utils",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../p2p:p2p_test_utils",
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"../p2p:rtc_p2p",
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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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"../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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"../test:test_support",
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"//third_party/abseil-cpp/absl/memory",
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]
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if (rtc_build_libsrtp) {
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deps += [ "//third_party/libsrtp" ]
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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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rtc_source_set("peerconnection_perf_tests") {
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testonly = true
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sources = [
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"peerconnection_rampup_tests.cc",
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"peerconnectionwrapper.cc",
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"peerconnectionwrapper.h",
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]
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deps = [
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":pc_test_utils",
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"../api:libjingle_peerconnection_api",
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"../api:libjingle_peerconnection_test_api",
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"../api:rtc_stats_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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"../api/video_codecs:builtin_video_decoder_factory",
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"../api/video_codecs:builtin_video_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: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_utils",
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"../test:perf_test",
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"../test:test_support",
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"//third_party/abseil-cpp/absl/memory",
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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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rtc_source_set("pc_test_utils") {
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testonly = true
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sources = [
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"test/fakeaudiocapturemodule.cc",
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"test/fakeaudiocapturemodule.h",
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"test/fakedatachannelprovider.h",
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"test/fakepeerconnectionbase.h",
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"test/fakepeerconnectionforstats.h",
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"test/fakeperiodicvideosource.h",
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"test/fakeperiodicvideotracksource.h",
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"test/fakertccertificategenerator.h",
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"test/fakesctptransport.h",
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"test/fakevideotrackrenderer.h",
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"test/fakevideotracksource.h",
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"test/framegeneratorcapturervideotracksource.h",
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"test/mock_datachannel.h",
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"test/mock_peerconnection.h",
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"test/mock_rtpreceiverinternal.h",
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"test/mock_rtpsenderinternal.h",
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"test/mockpeerconnectionobservers.h",
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"test/peerconnectiontestwrapper.cc",
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"test/peerconnectiontestwrapper.h",
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"test/rtcstatsobtainer.h",
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"test/testsdpstrings.h",
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]
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deps = [
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":libjingle_peerconnection",
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":peerconnection",
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":rtc_pc_base",
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"..:webrtc_common",
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"../api:libjingle_peerconnection_api",
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"../api:libjingle_peerconnection_test_api",
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"../api:rtc_stats_api",
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"../api/video:video_frame",
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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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"../logging:rtc_event_log_api",
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"../media:rtc_data",
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"../media:rtc_media",
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"../media:rtc_media_base",
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"../media:rtc_media_tests_utils",
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"../modules/audio_device:audio_device",
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"../modules/audio_processing:audio_processing",
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"../p2p:p2p_test_utils",
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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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"../rtc_base:rtc_base_tests_utils",
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"../rtc_base:rtc_task_queue",
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"../test:test_support",
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"../test:video_test_common",
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"//third_party/abseil-cpp/absl/memory",
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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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rtc_test("peerconnection_unittests") {
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testonly = true
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sources = [
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"datachannel_unittest.cc",
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"dtmfsender_unittest.cc",
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"iceserverparsing_unittest.cc",
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"jsepsessiondescription_unittest.cc",
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"localaudiosource_unittest.cc",
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"mediaconstraintsinterface_unittest.cc",
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"mediastream_unittest.cc",
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"peerconnection_bundle_unittest.cc",
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"peerconnection_crypto_unittest.cc",
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"peerconnection_datachannel_unittest.cc",
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"peerconnection_histogram_unittest.cc",
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"peerconnection_ice_unittest.cc",
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"peerconnection_integrationtest.cc",
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"peerconnection_jsep_unittest.cc",
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"peerconnection_media_unittest.cc",
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"peerconnection_rtp_unittest.cc",
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"peerconnection_signaling_unittest.cc",
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"peerconnectionendtoend_unittest.cc",
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"peerconnectionfactory_unittest.cc",
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"peerconnectioninterface_unittest.cc",
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"peerconnectionwrapper.cc",
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"peerconnectionwrapper.h",
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"proxy_unittest.cc",
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"rtcstats_integrationtest.cc",
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"rtcstatscollector_unittest.cc",
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"rtcstatstraversal_unittest.cc",
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"rtpmediautils_unittest.cc",
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"rtpparametersconversion_unittest.cc",
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"rtpsenderreceiver_unittest.cc",
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"sctputils_unittest.cc",
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"statscollector_unittest.cc",
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"test/fakeaudiocapturemodule_unittest.cc",
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"test/testsdpstrings.h",
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"trackmediainfomap_unittest.cc",
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"videocapturertracksource_unittest.cc",
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"videotrack_unittest.cc",
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"webrtcsdp_unittest.cc",
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]
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if (rtc_enable_sctp) {
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defines = [ "HAVE_SCTP" ]
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}
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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 = [
|
|
":peerconnection",
|
|
":rtc_pc_base",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:mock_rtp",
|
|
"../api/units:time_delta",
|
|
"../logging:fake_rtc_event_log",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:stringutils",
|
|
"../rtc_base/third_party/base64",
|
|
"../test:fileutils",
|
|
"//third_party/abseil-cpp/absl/memory",
|
|
]
|
|
if (is_android) {
|
|
deps += [ ":android_black_magic" ]
|
|
}
|
|
|
|
deps += [
|
|
":libjingle_peerconnection",
|
|
":pc_test_utils",
|
|
"..:webrtc_common",
|
|
"../api:callfactory_api",
|
|
"../api:libjingle_peerconnection_test_api",
|
|
"../api:rtc_stats_api",
|
|
"../api/audio_codecs:audio_codecs_api",
|
|
"../api/audio_codecs:builtin_audio_decoder_factory",
|
|
"../api/audio_codecs:builtin_audio_encoder_factory",
|
|
"../api/audio_codecs/L16:audio_decoder_L16",
|
|
"../api/audio_codecs/L16:audio_encoder_L16",
|
|
"../api/video_codecs:builtin_video_decoder_factory",
|
|
"../api/video_codecs:builtin_video_encoder_factory",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../call:call_interfaces",
|
|
"../logging:rtc_event_log_api",
|
|
"../logging:rtc_event_log_impl_base",
|
|
"../logging:rtc_event_log_impl_output",
|
|
"../media:rtc_audio_video",
|
|
"../media:rtc_data", # TODO(phoglund): AFAIK only used for one sctp constant.
|
|
"../media:rtc_media_base",
|
|
"../media:rtc_media_tests_utils",
|
|
"../modules/audio_processing:audio_processing",
|
|
"../modules/utility:utility",
|
|
"../p2p:p2p_test_utils",
|
|
"../p2p:rtc_p2p",
|
|
"../pc:rtc_pc",
|
|
"../rtc_base:rtc_base",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_base_tests_main",
|
|
"../rtc_base:rtc_base_tests_utils",
|
|
"../rtc_base:rtc_task_queue",
|
|
"../rtc_base:safe_conversions",
|
|
"../system_wrappers:metrics_default",
|
|
"../system_wrappers:runtime_enabled_features_default",
|
|
"../test:audio_codec_mocks",
|
|
"../test:test_support",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
|
|
if (is_android) {
|
|
deps += [
|
|
"//testing/android/native_test:native_test_support",
|
|
|
|
# We need to depend on this one directly, or classloads will fail for
|
|
# the voice engine BuildInfo, for instance.
|
|
"../sdk/android:libjingle_peerconnection_java",
|
|
]
|
|
|
|
shard_timeout = 900
|
|
}
|
|
}
|
|
|
|
if (is_android) {
|
|
rtc_source_set("android_black_magic") {
|
|
# The android code uses hacky includes to chromium-base and the ssl code;
|
|
# having this in a separate target enables us to keep the peerconnection
|
|
# unit tests clean.
|
|
check_includes = false
|
|
testonly = true
|
|
sources = [
|
|
"test/androidtestinitializer.cc",
|
|
"test/androidtestinitializer.h",
|
|
]
|
|
deps = [
|
|
"../sdk/android:libjingle_peerconnection_jni",
|
|
"//testing/android/native_test:native_test_support",
|
|
]
|
|
}
|
|
}
|
|
}
|