e830e683c4
The TransportController will be replaced by the JsepTransportController and JsepTransport will be replace be JsepTransport2. The JsepTransportController will take the entire SessionDescription and handle the RtcpMux, Sdes and BUNDLE internally. The ownership model is also changed. The P2P layer transports are not ref-counted and will be owned by the JsepTransport2. In ORTC aspect, RtpTransportAdapter is now a wrapper over RtpTransport or SrtpTransport and it implements the public and internal interface by calling the transport underneath. Bug: webrtc:8587 Change-Id: Ia7fa61288a566f211f8560072ea0eecaf19e48df Reviewed-on: https://webrtc-review.googlesource.com/59586 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22693}
552 lines
16 KiB
Plaintext
552 lines
16 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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"audiomonitor.h",
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"bundlefilter.cc",
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"bundlefilter.h",
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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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"currentspeakermonitor.cc",
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"currentspeakermonitor.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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"jseptransport2.cc",
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"jseptransport2.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:optional",
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"../api:ortc_api",
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"../api:video_frame_api",
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"../common_video:common_video",
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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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"../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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]
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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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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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config("libjingle_peerconnection_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 && !is_clang) {
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cflags = [ "-Wno-maybe-uninitialized" ] # Only exists for GCC.
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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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"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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configs += [ ":libjingle_peerconnection_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_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:optional",
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"../api:rtc_stats_api",
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"../api:video_frame_api",
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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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"../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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"../stats",
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"../system_wrappers",
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"../system_wrappers:field_trial_api",
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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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configs += [ ":libjingle_peerconnection_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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}
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rtc_source_set("libjingle_peerconnection") {
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visibility = [ "*" ]
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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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config("rtc_pc_unittests_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_win && !is_clang) {
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cflags = [ "-Wno-maybe-uninitialized" ] # Only exists for GCC.
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}
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}
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rtc_test("rtc_pc_unittests") {
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testonly = true
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sources = [
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"bundlefilter_unittest.cc",
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"channel_unittest.cc",
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"channelmanager_unittest.cc",
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"currentspeakermonitor_unittest.cc",
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"dtlssrtptransport_unittest.cc",
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"jseptransport2_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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configs += [ ":rtc_pc_unittests_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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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:fakemetricsobserver",
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"../api:libjingle_peerconnection_api",
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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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"../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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]
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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("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/fakeperiodicvideocapturer.h",
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"test/fakeperiodicvideosource.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/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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"../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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"../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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]
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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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config("peerconnection_unittests_config") {
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# The warnings below are enabled by default. Since GN orders compiler flags
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# for a target before flags from configs, the only way to disable such
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# warnings is by having them in a separate config, loaded from the target.
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# TODO(kjellander): Make the code compile without disabling these flags.
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# See https://bugs.webrtc.org/3307.
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if (is_clang && is_win) {
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cflags = [
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# See https://bugs.chromium.org/p/webrtc/issues/detail?id=6267
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# for -Wno-sign-compare
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"-Wno-sign-compare",
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]
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}
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if (!is_win) {
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cflags = [ "-Wno-sign-compare" ]
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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_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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"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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configs += [ ":peerconnection_unittests_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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# TODO(jschuh): Bug 1348: fix this warning.
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configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
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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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"/wd4389", # signed/unsigned mismatch.
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]
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}
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deps = [
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":peerconnection",
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":rtc_pc_base",
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"../api:libjingle_peerconnection_api",
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"../api:mock_rtp",
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"../rtc_base:checks",
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"../rtc_base:stringutils",
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"../test:fileutils",
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]
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if (is_android) {
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deps += [ ":android_black_magic" ]
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|
}
|
|
|
|
deps += [
|
|
":libjingle_peerconnection",
|
|
":pc_test_utils",
|
|
"..:webrtc_common",
|
|
"../api:callfactory_api",
|
|
"../api:fakemetricsobserver",
|
|
"../api:libjingle_peerconnection_test_api",
|
|
"../api:optional",
|
|
"../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",
|
|
"../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",
|
|
"../system_wrappers:metrics_default",
|
|
"../system_wrappers:runtime_enabled_features_default",
|
|
"../test:audio_codec_mocks",
|
|
"../test:test_support",
|
|
]
|
|
|
|
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",
|
|
]
|
|
}
|
|
}
|
|
}
|