d1003d74b2
This test creates a one way audio and video call, allows for bandwidth estimation to ramp up and then runs the call for 10 seconds. The average bandwidth estimate over this time is recorded as a perf metric. This is done at the PeerConnection level with the intention to catch regressions related to ICE configurations. Stats are taken from PeerConnection for BWE, and the network simulation is done with a VirtualSocketServer. Bug: webrtc:7668 Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db Reviewed-on: https://webrtc-review.googlesource.com/78361 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23758}
600 lines
18 KiB
Plaintext
600 lines
18 KiB
Plaintext
# 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") {
|
|
deps = [
|
|
":rtc_pc",
|
|
]
|
|
}
|
|
|
|
config("rtc_pc_config") {
|
|
defines = []
|
|
if (rtc_enable_sctp) {
|
|
defines += [ "HAVE_SCTP" ]
|
|
}
|
|
}
|
|
|
|
rtc_static_library("rtc_pc_base") {
|
|
visibility = [ "*" ]
|
|
defines = []
|
|
sources = [
|
|
"channel.cc",
|
|
"channel.h",
|
|
"channelmanager.cc",
|
|
"channelmanager.h",
|
|
"dtlssrtptransport.cc",
|
|
"dtlssrtptransport.h",
|
|
"externalhmac.cc",
|
|
"externalhmac.h",
|
|
"jseptransport.cc",
|
|
"jseptransport.h",
|
|
"jseptransportcontroller.cc",
|
|
"jseptransportcontroller.h",
|
|
"mediasession.cc",
|
|
"mediasession.h",
|
|
"rtcpmuxfilter.cc",
|
|
"rtcpmuxfilter.h",
|
|
"rtpmediautils.cc",
|
|
"rtpmediautils.h",
|
|
"rtptransport.cc",
|
|
"rtptransport.h",
|
|
"rtptransportinternal.h",
|
|
"rtptransportinternaladapter.h",
|
|
"sessiondescription.cc",
|
|
"sessiondescription.h",
|
|
"srtpfilter.cc",
|
|
"srtpfilter.h",
|
|
"srtpsession.cc",
|
|
"srtpsession.h",
|
|
"srtptransport.cc",
|
|
"srtptransport.h",
|
|
"transportstats.cc",
|
|
"transportstats.h",
|
|
]
|
|
|
|
deps = [
|
|
"..:webrtc_common",
|
|
"../api:array_view",
|
|
"../api:call_api",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:ortc_api",
|
|
"../api/video:video_frame",
|
|
"../call:rtp_interfaces",
|
|
"../call:rtp_receiver",
|
|
"../common_video:common_video",
|
|
"../logging:rtc_event_log_api",
|
|
"../media:rtc_data",
|
|
"../media:rtc_h264_profile_id",
|
|
"../media:rtc_media_base",
|
|
"../modules/rtp_rtcp:rtp_rtcp_format",
|
|
"../p2p:rtc_p2p",
|
|
"../rtc_base:base64",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:rtc_base",
|
|
"../rtc_base:rtc_task_queue",
|
|
"../rtc_base:stringutils",
|
|
"../system_wrappers:metrics_api",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
|
|
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") {
|
|
visibility = [ "*" ]
|
|
allow_poison = [
|
|
"audio_codecs", # TODO(bugs.webrtc.org/8396): Remove.
|
|
"software_video_codecs", # TODO(bugs.webrtc.org/7925): Remove.
|
|
]
|
|
deps = [
|
|
":rtc_pc_base",
|
|
"../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") {
|
|
visibility = [ "*" ]
|
|
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",
|
|
"peerconnectioninternal.h",
|
|
"remoteaudiosource.cc",
|
|
"remoteaudiosource.h",
|
|
"rtcstatscollector.cc",
|
|
"rtcstatscollector.h",
|
|
"rtcstatstraversal.cc",
|
|
"rtcstatstraversal.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:fec_controller_api",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:rtc_stats_api",
|
|
"../api/video:video_frame",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../call:call_interfaces",
|
|
"../common_video:common_video",
|
|
"../logging:ice_log",
|
|
"../logging:rtc_event_log_api",
|
|
"../logging:rtc_event_log_impl_output",
|
|
"../media:rtc_data",
|
|
"../media:rtc_media_base",
|
|
"../modules/congestion_controller/bbr",
|
|
"../p2p:rtc_p2p",
|
|
"../rtc_base:base64",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:rtc_base",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:stringutils",
|
|
"../rtc_base/experiments:congestion_controller_experiment",
|
|
"../stats",
|
|
"../system_wrappers",
|
|
"../system_wrappers:field_trial_api",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
}
|
|
|
|
# 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:callfactory_api",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api/audio:audio_mixer_api",
|
|
"../api/audio_codecs:audio_codecs_api",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../call",
|
|
"../call:call_interfaces",
|
|
"../logging:rtc_event_log_api",
|
|
"../logging:rtc_event_log_impl_base",
|
|
"../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") {
|
|
visibility = [ "*" ]
|
|
allow_poison = [
|
|
"audio_codecs", # TODO(bugs.webrtc.org/8396): Remove.
|
|
"software_video_codecs", # TODO(bugs.webrtc.org/7925): Remove.
|
|
]
|
|
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 = [
|
|
"channel_unittest.cc",
|
|
"channelmanager_unittest.cc",
|
|
"dtlssrtptransport_unittest.cc",
|
|
"jseptransport_unittest.cc",
|
|
"jseptransportcontroller_unittest.cc",
|
|
"mediasession_unittest.cc",
|
|
"rtcpmuxfilter_unittest.cc",
|
|
"rtptransport_unittest.cc",
|
|
"rtptransporttestutil.h",
|
|
"srtpfilter_unittest.cc",
|
|
"srtpsession_unittest.cc",
|
|
"srtptestutil.h",
|
|
"srtptransport_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",
|
|
":rtc_pc_base",
|
|
"../api:array_view",
|
|
"../api:fakemetricsobserver",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../call:rtp_interfaces",
|
|
"../logging:rtc_event_log_api",
|
|
"../media:rtc_media_base",
|
|
"../media:rtc_media_tests_utils",
|
|
"../modules/rtp_rtcp:rtp_rtcp_format",
|
|
"../p2p:p2p_test_utils",
|
|
"../p2p:rtc_p2p",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:rtc_base",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_base_tests_main",
|
|
"../rtc_base:rtc_base_tests_utils",
|
|
"../system_wrappers:metrics_default",
|
|
"../system_wrappers:runtime_enabled_features_default",
|
|
"../test:test_support",
|
|
]
|
|
|
|
if (rtc_build_libsrtp) {
|
|
deps += [ "//third_party/libsrtp" ]
|
|
}
|
|
|
|
if (is_android) {
|
|
deps += [ "//testing/android/native_test:native_test_support" ]
|
|
}
|
|
}
|
|
|
|
rtc_source_set("peerconnection_perf_tests") {
|
|
testonly = true
|
|
sources = [
|
|
"peerconnection_rampup_tests.cc",
|
|
"peerconnectionwrapper.cc",
|
|
"peerconnectionwrapper.h",
|
|
]
|
|
deps = [
|
|
":pc_test_utils",
|
|
"../api:fakemetricsobserver",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:libjingle_peerconnection_test_api",
|
|
"../api:rtc_stats_api",
|
|
"../api/audio_codecs:builtin_audio_decoder_factory",
|
|
"../api/audio_codecs:builtin_audio_encoder_factory",
|
|
"../api/video_codecs:builtin_video_decoder_factory",
|
|
"../api/video_codecs:builtin_video_encoder_factory",
|
|
"../media:rtc_media_tests_utils",
|
|
"../p2p:p2p_test_utils",
|
|
"../p2p:rtc_p2p",
|
|
"../pc:peerconnection",
|
|
"../rtc_base:rtc_base",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_base_tests_utils",
|
|
"../test:perf_test",
|
|
"../test:test_support",
|
|
]
|
|
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("pc_test_utils") {
|
|
testonly = true
|
|
sources = [
|
|
"test/fakeaudiocapturemodule.cc",
|
|
"test/fakeaudiocapturemodule.h",
|
|
"test/fakedatachannelprovider.h",
|
|
"test/fakepeerconnectionbase.h",
|
|
"test/fakepeerconnectionforstats.h",
|
|
"test/fakeperiodicvideosource.h",
|
|
"test/fakeperiodicvideotracksource.h",
|
|
"test/fakertccertificategenerator.h",
|
|
"test/fakesctptransport.h",
|
|
"test/fakevideotrackrenderer.h",
|
|
"test/fakevideotracksource.h",
|
|
"test/framegeneratorcapturervideotracksource.h",
|
|
"test/mock_datachannel.h",
|
|
"test/mock_peerconnection.h",
|
|
"test/mock_rtpreceiverinternal.h",
|
|
"test/mock_rtpsenderinternal.h",
|
|
"test/mockpeerconnectionobservers.h",
|
|
"test/peerconnectiontestwrapper.cc",
|
|
"test/peerconnectiontestwrapper.h",
|
|
"test/rtcstatsobtainer.h",
|
|
"test/testsdpstrings.h",
|
|
]
|
|
|
|
deps = [
|
|
":libjingle_peerconnection",
|
|
":peerconnection",
|
|
":rtc_pc_base",
|
|
"..:webrtc_common",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:libjingle_peerconnection_test_api",
|
|
"../api:rtc_stats_api",
|
|
"../api/video:video_frame",
|
|
"../api/video_codecs:builtin_video_decoder_factory",
|
|
"../api/video_codecs:builtin_video_encoder_factory",
|
|
"../call:call_interfaces",
|
|
"../logging:rtc_event_log_api",
|
|
"../media:rtc_data",
|
|
"../media:rtc_media",
|
|
"../media:rtc_media_base",
|
|
"../media:rtc_media_tests_utils",
|
|
"../modules/audio_device:audio_device",
|
|
"../modules/audio_processing:audio_processing",
|
|
"../p2p:p2p_test_utils",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:rtc_base",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_base_tests_utils",
|
|
"../rtc_base:rtc_task_queue",
|
|
"../test:test_support",
|
|
"../test:video_test_common",
|
|
]
|
|
|
|
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_histogram_unittest.cc",
|
|
"peerconnection_ice_unittest.cc",
|
|
"peerconnection_integrationtest.cc",
|
|
"peerconnection_jsep_unittest.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",
|
|
"rtcstatstraversal_unittest.cc",
|
|
"rtpmediautils_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" ]
|
|
}
|
|
|
|
deps = [
|
|
":peerconnection",
|
|
":rtc_pc_base",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:mock_rtp",
|
|
"../api/units:time_delta",
|
|
"../logging:fake_rtc_event_log",
|
|
"../rtc_base:base64",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:stringutils",
|
|
"../test:fileutils",
|
|
]
|
|
if (is_android) {
|
|
deps += [ ":android_black_magic" ]
|
|
}
|
|
|
|
deps += [
|
|
":libjingle_peerconnection",
|
|
":pc_test_utils",
|
|
"..:webrtc_common",
|
|
"../api:callfactory_api",
|
|
"../api:fakemetricsobserver",
|
|
"../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",
|
|
"../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",
|
|
]
|
|
}
|
|
}
|
|
}
|