mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
316386747a
Description for changes from the original CL:
Calling legacy SRD, implemented using
SetRemoteDescriptionObserverAdapter wrapping the old observer, was
meant to have the exact same behavior as the legacy SRD implementation
which invokes the callbacks in a Post.
However, in the CL that landed and got reverted (PS1), the Adapter had
its own message handler, and callbacks would be invoked even if the PC
was destroyed.
In PS2 I've changed the Adapter to use the PeerConnection's message
handler. If the PC is destroyed, the callback will not be invoked.
This gives identical behavior to before this CL, and the legacy
behavior is covered by a new unittest.
I changed the adapter to be an implementation detail of
peerconnection.cc, therefor some stuff was moved, and the only tests
covering this is now in peerconnection_rtp_unittest.cc.
This is a reland of 6c7ec32bd6
Original change's description:
> SetRemoteDescriptionObserverInterface added.
>
> The new observer replaced SetSessionDescriptionObserver for
> SetRemoteDescription. Unlike SetSessionDescriptionObserver,
> SetRemoteDescriptionObserverInterface is invoked synchronously so
> that the you can rely on the state of the PeerConnection to represent
> the result of the SetRemoteDescription call in the callback.
>
> The new observer succeeds or fails with an RTCError.
>
> This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack
> and SetSessionDescriptionObserver, with the benefit that all media
> object changes can be processed in a single callback by the application
> in a synchronous callback. This will help Chromium keep objects in-sync
> across layers and threads in a non-racy and straight-forward way, see
> design doc (Proposal 2):
> https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing
>
> An adapter for SetSessionDescriptionObserver is added to allow calling
> the old SetRemoteDescription signature and get the old behavior
> (OnSuccess/OnFailure callback in a Post) until third parties switch.
>
> Bug: webrtc:8473
> Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99
> Reviewed-on: https://webrtc-review.googlesource.com/17523
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20841}
TBR=pthatcher@webrtc.org
Bug: webrtc:8473
Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5
Reviewed-on: https://webrtc-review.googlesource.com/25640
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20854}
504 lines
14 KiB
Text
504 lines
14 KiB
Text
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
|
|
#
|
|
# Use of this source code is governed by a BSD-style license
|
|
# that can be found in the LICENSE file in the root of the source
|
|
# tree. An additional intellectual property rights grant can be found
|
|
# in the file PATENTS. All contributing project authors may
|
|
# be found in the AUTHORS file in the root of the source tree.
|
|
|
|
import("../webrtc.gni")
|
|
if (is_android) {
|
|
import("//build/config/android/config.gni")
|
|
import("//build/config/android/rules.gni")
|
|
}
|
|
|
|
group("pc") {
|
|
public_deps = [
|
|
":rtc_pc",
|
|
]
|
|
}
|
|
|
|
config("rtc_pc_config") {
|
|
defines = []
|
|
if (rtc_enable_sctp) {
|
|
defines += [ "HAVE_SCTP" ]
|
|
}
|
|
}
|
|
|
|
rtc_static_library("rtc_pc_base") {
|
|
defines = []
|
|
sources = [
|
|
"audiomonitor.cc",
|
|
"audiomonitor.h",
|
|
"bundlefilter.cc",
|
|
"bundlefilter.h",
|
|
"channel.cc",
|
|
"channel.h",
|
|
"channelmanager.cc",
|
|
"channelmanager.h",
|
|
"currentspeakermonitor.cc",
|
|
"currentspeakermonitor.h",
|
|
"dtlssrtptransport.cc",
|
|
"dtlssrtptransport.h",
|
|
"externalhmac.cc",
|
|
"externalhmac.h",
|
|
"mediamonitor.cc",
|
|
"mediamonitor.h",
|
|
"mediasession.cc",
|
|
"mediasession.h",
|
|
"rtcpmuxfilter.cc",
|
|
"rtcpmuxfilter.h",
|
|
"rtptransport.cc",
|
|
"rtptransport.h",
|
|
"rtptransportinternal.h",
|
|
"rtptransportinternaladapter.h",
|
|
"srtpfilter.cc",
|
|
"srtpfilter.h",
|
|
"srtpsession.cc",
|
|
"srtpsession.h",
|
|
"srtptransport.cc",
|
|
"srtptransport.h",
|
|
"transportcontroller.cc",
|
|
"transportcontroller.h",
|
|
"voicechannel.h",
|
|
]
|
|
|
|
deps = [
|
|
"..:webrtc_common",
|
|
"../api:call_api",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:optional",
|
|
"../api:ortc_api",
|
|
"../media:rtc_data",
|
|
"../media:rtc_h264_profile_id",
|
|
"../media:rtc_media_base",
|
|
"../p2p:rtc_p2p",
|
|
"../rtc_base:rtc_base",
|
|
"../rtc_base:rtc_task_queue",
|
|
]
|
|
|
|
if (rtc_build_libsrtp) {
|
|
deps += [ "//third_party/libsrtp" ]
|
|
}
|
|
|
|
public_configs = [ ":rtc_pc_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" ]
|
|
}
|
|
}
|
|
|
|
rtc_source_set("rtc_pc") {
|
|
public_deps = [
|
|
":rtc_pc_base",
|
|
]
|
|
|
|
deps = [
|
|
"../media:rtc_audio_video",
|
|
]
|
|
}
|
|
|
|
config("libjingle_peerconnection_warnings_config") {
|
|
# GN orders flags on a target before flags from configs. The default config
|
|
# adds these flags so to cancel them out they need to come from a config and
|
|
# cannot be on the target directly.
|
|
if (!is_win && !is_clang) {
|
|
cflags = [ "-Wno-maybe-uninitialized" ] # Only exists for GCC.
|
|
}
|
|
}
|
|
|
|
rtc_static_library("peerconnection") {
|
|
cflags = []
|
|
sources = [
|
|
"audiotrack.cc",
|
|
"audiotrack.h",
|
|
"datachannel.cc",
|
|
"datachannel.h",
|
|
"dtmfsender.cc",
|
|
"dtmfsender.h",
|
|
"iceserverparsing.cc",
|
|
"iceserverparsing.h",
|
|
"jsepicecandidate.cc",
|
|
"jsepsessiondescription.cc",
|
|
"localaudiosource.cc",
|
|
"localaudiosource.h",
|
|
"mediastream.cc",
|
|
"mediastream.h",
|
|
"mediastreamobserver.cc",
|
|
"mediastreamobserver.h",
|
|
"mediastreamtrack.h",
|
|
"peerconnection.cc",
|
|
"peerconnection.h",
|
|
"peerconnectionfactory.cc",
|
|
"peerconnectionfactory.h",
|
|
"remoteaudiosource.cc",
|
|
"remoteaudiosource.h",
|
|
"rtcstatscollector.cc",
|
|
"rtcstatscollector.h",
|
|
"rtpreceiver.cc",
|
|
"rtpreceiver.h",
|
|
"rtpsender.cc",
|
|
"rtpsender.h",
|
|
"rtptransceiver.cc",
|
|
"rtptransceiver.h",
|
|
"sctputils.cc",
|
|
"sctputils.h",
|
|
"sdputils.cc",
|
|
"sdputils.h",
|
|
"statscollector.cc",
|
|
"statscollector.h",
|
|
"streamcollection.h",
|
|
"trackmediainfomap.cc",
|
|
"trackmediainfomap.h",
|
|
"videocapturertracksource.cc",
|
|
"videocapturertracksource.h",
|
|
"videotrack.cc",
|
|
"videotrack.h",
|
|
"videotracksource.cc",
|
|
"videotracksource.h",
|
|
"webrtcsdp.cc",
|
|
"webrtcsdp.h",
|
|
"webrtcsessiondescriptionfactory.cc",
|
|
"webrtcsessiondescriptionfactory.h",
|
|
]
|
|
|
|
configs += [ ":libjingle_peerconnection_warnings_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_pc_base",
|
|
"..:webrtc_common",
|
|
"../api:call_api",
|
|
"../api:optional",
|
|
"../api:rtc_stats_api",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../call:call_interfaces",
|
|
"../logging:rtc_event_log_api",
|
|
"../media:rtc_data",
|
|
"../media:rtc_media_base",
|
|
"../p2p:rtc_p2p",
|
|
"../rtc_base:rtc_base",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../stats",
|
|
"../system_wrappers:system_wrappers",
|
|
]
|
|
|
|
public_deps = [
|
|
"../api:libjingle_peerconnection_api",
|
|
]
|
|
}
|
|
|
|
# This target implements CreatePeerConnectionFactory methods that will create a
|
|
# PeerConnection will full functionality (audio, video and data). Applications
|
|
# that wish to reduce their binary size by ommitting functionality they don't
|
|
# need should use CreateModularCreatePeerConnectionFactory instead, using the
|
|
# "peerconnection" build target and other targets specific to their
|
|
# requrements. See comment in peerconnectionfactoryinterface.h.
|
|
rtc_static_library("create_pc_factory") {
|
|
sources = [
|
|
"createpeerconnectionfactory.cc",
|
|
]
|
|
|
|
deps = [
|
|
"../api:audio_mixer_api",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:peerconnection_and_implicit_call_api",
|
|
"../api/audio_codecs:audio_codecs_api",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../call",
|
|
"../call:call_interfaces",
|
|
"../logging:rtc_event_log_api",
|
|
"../media:rtc_audio_video",
|
|
"../media:rtc_media_base",
|
|
"../modules/audio_device:audio_device",
|
|
"../modules/audio_processing:audio_processing",
|
|
"../rtc_base:rtc_base",
|
|
"../rtc_base:rtc_base_approved",
|
|
]
|
|
|
|
configs += [ ":libjingle_peerconnection_warnings_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" ]
|
|
}
|
|
}
|
|
|
|
rtc_source_set("libjingle_peerconnection") {
|
|
public_deps = [
|
|
":create_pc_factory",
|
|
":peerconnection",
|
|
"../api:libjingle_peerconnection_api",
|
|
]
|
|
}
|
|
|
|
if (rtc_include_tests) {
|
|
config("rtc_pc_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.
|
|
if (!is_win && !is_clang) {
|
|
cflags = [ "-Wno-maybe-uninitialized" ] # Only exists for GCC.
|
|
}
|
|
}
|
|
|
|
rtc_test("rtc_pc_unittests") {
|
|
testonly = true
|
|
|
|
sources = [
|
|
"bundlefilter_unittest.cc",
|
|
"channel_unittest.cc",
|
|
"channelmanager_unittest.cc",
|
|
"currentspeakermonitor_unittest.cc",
|
|
"dtlssrtptransport_unittest.cc",
|
|
"mediasession_unittest.cc",
|
|
"rtcpmuxfilter_unittest.cc",
|
|
"rtptransport_unittest.cc",
|
|
"rtptransporttestutil.h",
|
|
"srtpfilter_unittest.cc",
|
|
"srtpsession_unittest.cc",
|
|
"srtptestutil.h",
|
|
"srtptransport_unittest.cc",
|
|
"transportcontroller_unittest.cc",
|
|
]
|
|
|
|
include_dirs = [ "//third_party/libsrtp/srtp" ]
|
|
|
|
configs += [ ":rtc_pc_unittests_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" ]
|
|
}
|
|
|
|
if (is_win) {
|
|
libs = [ "strmiids.lib" ]
|
|
}
|
|
|
|
deps = [
|
|
":libjingle_peerconnection",
|
|
":pc_test_utils",
|
|
":rtc_pc",
|
|
"../api:array_view",
|
|
"../logging:rtc_event_log_api",
|
|
"../media:rtc_media_base",
|
|
"../media:rtc_media_tests_utils",
|
|
"../p2p:p2p_test_utils",
|
|
"../p2p:rtc_p2p",
|
|
"../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",
|
|
"../test:test_support",
|
|
]
|
|
|
|
if (rtc_build_libsrtp) {
|
|
deps += [ "//third_party/libsrtp" ]
|
|
}
|
|
|
|
if (is_android) {
|
|
deps += [ "//testing/android/native_test:native_test_support" ]
|
|
}
|
|
}
|
|
|
|
rtc_source_set("pc_test_utils") {
|
|
testonly = true
|
|
sources = [
|
|
"test/fakeaudiocapturemodule.cc",
|
|
"test/fakeaudiocapturemodule.h",
|
|
"test/fakedatachannelprovider.h",
|
|
"test/fakeperiodicvideocapturer.h",
|
|
"test/fakertccertificategenerator.h",
|
|
"test/fakesctptransport.h",
|
|
"test/faketransportcontroller.h",
|
|
"test/fakevideotrackrenderer.h",
|
|
"test/fakevideotracksource.h",
|
|
"test/mock_datachannel.h",
|
|
"test/mock_peerconnection.h",
|
|
"test/mockpeerconnectionobservers.h",
|
|
"test/peerconnectiontestwrapper.cc",
|
|
"test/peerconnectiontestwrapper.h",
|
|
"test/rtcstatsobtainer.h",
|
|
"test/testsdpstrings.h",
|
|
]
|
|
|
|
deps = [
|
|
":libjingle_peerconnection",
|
|
":rtc_pc_base",
|
|
"..:webrtc_common",
|
|
"../api:libjingle_peerconnection_test_api",
|
|
"../api:peerconnection_and_implicit_call_api",
|
|
"../api:rtc_stats_api",
|
|
"../call:call_interfaces",
|
|
"../logging:rtc_event_log_api",
|
|
"../media:rtc_media",
|
|
"../media:rtc_media_base",
|
|
"../media:rtc_media_tests_utils",
|
|
"../modules/audio_device:audio_device",
|
|
"../p2p:p2p_test_utils",
|
|
"../rtc_base:rtc_base",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_base_tests_utils",
|
|
"../test:test_support",
|
|
"//testing/gmock",
|
|
]
|
|
|
|
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" ]
|
|
}
|
|
}
|
|
|
|
config("peerconnection_unittests_config") {
|
|
# The warnings below are enabled by default. Since GN orders compiler flags
|
|
# for a target before flags from configs, the only way to disable such
|
|
# warnings is by having them in a separate config, loaded from the target.
|
|
# 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=6267
|
|
# for -Wno-sign-compare
|
|
"-Wno-sign-compare",
|
|
]
|
|
}
|
|
|
|
if (!is_win) {
|
|
cflags = [ "-Wno-sign-compare" ]
|
|
}
|
|
}
|
|
|
|
rtc_test("peerconnection_unittests") {
|
|
testonly = true
|
|
sources = [
|
|
"datachannel_unittest.cc",
|
|
"dtmfsender_unittest.cc",
|
|
"iceserverparsing_unittest.cc",
|
|
"jsepsessiondescription_unittest.cc",
|
|
"localaudiosource_unittest.cc",
|
|
"mediaconstraintsinterface_unittest.cc",
|
|
"mediastream_unittest.cc",
|
|
"peerconnection_bundle_unittest.cc",
|
|
"peerconnection_crypto_unittest.cc",
|
|
"peerconnection_datachannel_unittest.cc",
|
|
"peerconnection_ice_unittest.cc",
|
|
"peerconnection_integrationtest.cc",
|
|
"peerconnection_media_unittest.cc",
|
|
"peerconnection_rtp_unittest.cc",
|
|
"peerconnection_signaling_unittest.cc",
|
|
"peerconnectionendtoend_unittest.cc",
|
|
"peerconnectionfactory_unittest.cc",
|
|
"peerconnectioninterface_unittest.cc",
|
|
"peerconnectionwrapper.cc",
|
|
"peerconnectionwrapper.h",
|
|
"proxy_unittest.cc",
|
|
"rtcstats_integrationtest.cc",
|
|
"rtcstatscollector_unittest.cc",
|
|
"rtpsenderreceiver_unittest.cc",
|
|
"sctputils_unittest.cc",
|
|
"statscollector_unittest.cc",
|
|
"test/fakeaudiocapturemodule_unittest.cc",
|
|
"test/testsdpstrings.h",
|
|
"trackmediainfomap_unittest.cc",
|
|
"videocapturertracksource_unittest.cc",
|
|
"videotrack_unittest.cc",
|
|
"webrtcsdp_unittest.cc",
|
|
]
|
|
|
|
if (rtc_enable_sctp) {
|
|
defines = [ "HAVE_SCTP" ]
|
|
}
|
|
|
|
configs += [ ":peerconnection_unittests_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" ]
|
|
}
|
|
|
|
# TODO(jschuh): Bug 1348: fix this warning.
|
|
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
|
|
|
|
if (is_win) {
|
|
cflags = [
|
|
"/wd4245", # conversion from int to size_t, signed/unsigned mismatch.
|
|
"/wd4389", # signed/unsigned mismatch.
|
|
]
|
|
}
|
|
|
|
deps = []
|
|
if (is_android) {
|
|
deps += [ ":android_black_magic" ]
|
|
}
|
|
|
|
deps += [
|
|
":libjingle_peerconnection",
|
|
":pc_test_utils",
|
|
"..:webrtc_common",
|
|
"../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",
|
|
"../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",
|
|
"../system_wrappers:metrics_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",
|
|
]
|
|
}
|
|
}
|
|
}
|