hbos db346a7cbe RTCStatsIntegrationTest added.
This is an integration test using peerconnectiontestwrapper.h to set up
and end to end test using a real PeerConnection implementation. These
tests will complement rtcstatscollector_unittest.cc which collects all
stats using mocks.

The integration test is set up so that all stats types are returned by
GetStats and verifies that expected dictionary members are defined. The
test could in the future be updated to include sanity checks for the
values of members. There is a sanity check that references to other
stats dictionaries yield existing stats of the appropriate type, but
other than that members are only tested for if they are defined not.

StatsCallback of rtcstatscollector_unittest.cc is moved so that it can
be reused and renamed to RTCStatsObtainer.

TODO: Audio stream track stats members are missing in the test. Find out
if this is because of a real problem or because of testing without real
devices. Do this before closing crbug.com/627816.

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2521663002
Cr-Commit-Position: refs/heads/master@{#15287}
2016-11-29 09:57:08 +00:00

541 lines
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# Copyright (c) 2015 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("../build/webrtc.gni")
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
group("api") {
public_deps = [
":libjingle_peerconnection",
]
if (is_android && !build_with_chromium) {
public_deps += [
":libjingle_peerconnection_java",
":libjingle_peerconnection_so",
]
}
}
rtc_source_set("call_api") {
sources = [
"call/audio_receive_stream.h",
"call/audio_send_stream.cc",
"call/audio_send_stream.h",
"call/audio_sink.h",
"call/audio_state.h",
"call/flexfec_receive_stream.h",
]
deps = [
# TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
":audio_mixer_api",
":transport_api",
"..:webrtc_common",
"../base:rtc_base_approved",
"../modules/audio_coding:audio_encoder_interface",
]
}
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("libjingle_peerconnection") {
cflags = []
sources = [
"audiotrack.cc",
"audiotrack.h",
"datachannel.cc",
"datachannel.h",
"datachannelinterface.h",
"dtmfsender.cc",
"dtmfsender.h",
"dtmfsenderinterface.h",
"jsep.h",
"jsepicecandidate.cc",
"jsepicecandidate.h",
"jsepsessiondescription.cc",
"jsepsessiondescription.h",
"localaudiosource.cc",
"localaudiosource.h",
"mediaconstraintsinterface.cc",
"mediaconstraintsinterface.h",
"mediacontroller.cc",
"mediacontroller.h",
"mediastream.cc",
"mediastream.h",
"mediastreaminterface.h",
"mediastreamobserver.cc",
"mediastreamobserver.h",
"mediastreamproxy.h",
"mediastreamtrack.h",
"mediastreamtrackproxy.h",
"notifier.h",
"peerconnection.cc",
"peerconnection.h",
"peerconnectionfactory.cc",
"peerconnectionfactory.h",
"peerconnectionfactoryproxy.h",
"peerconnectioninterface.h",
"peerconnectionproxy.h",
"proxy.h",
"remoteaudiosource.cc",
"remoteaudiosource.h",
"rtcstatscollector.cc",
"rtcstatscollector.h",
"rtpparameters.h",
"rtpreceiver.cc",
"rtpreceiver.h",
"rtpreceiverinterface.h",
"rtpsender.cc",
"rtpsender.h",
"rtpsenderinterface.h",
"sctputils.cc",
"sctputils.h",
"statscollector.cc",
"statscollector.h",
"statstypes.cc",
"statstypes.h",
"streamcollection.h",
"videocapturertracksource.cc",
"videocapturertracksource.h",
"videosourceproxy.h",
"videotrack.cc",
"videotrack.h",
"videotracksource.cc",
"videotracksource.h",
"webrtcsdp.cc",
"webrtcsdp.h",
"webrtcsession.cc",
"webrtcsession.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 = [
":call_api",
":rtc_stats_api",
"../call",
"../media",
"../pc",
"../stats",
]
if (rtc_use_quic) {
sources += [
"quicdatachannel.cc",
"quicdatachannel.h",
"quicdatatransport.cc",
"quicdatatransport.h",
]
deps += [ "//third_party/libquic" ]
public_deps = [
"//third_party/libquic",
]
}
}
# Exclude the targets below from the Chromium build since they cannot be built
# due to incompability with Chromium's logging implementation.
if (is_android && !build_with_chromium) {
config("libjingle_peerconnection_jni_warnings_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.
if (!is_win) {
cflags = [
"-Wno-sign-compare",
"-Wno-unused-variable",
]
}
}
rtc_static_library("libjingle_peerconnection_jni") {
sources = [
"android/jni/androidhistogram_jni.cc",
"android/jni/androidmediacodeccommon.h",
"android/jni/androidmediadecoder_jni.cc",
"android/jni/androidmediadecoder_jni.h",
"android/jni/androidmediaencoder_jni.cc",
"android/jni/androidmediaencoder_jni.h",
"android/jni/androidnetworkmonitor_jni.cc",
"android/jni/androidnetworkmonitor_jni.h",
"android/jni/androidvideotracksource.cc",
"android/jni/androidvideotracksource.h",
"android/jni/androidvideotracksource_jni.cc",
"android/jni/classreferenceholder.cc",
"android/jni/classreferenceholder.h",
"android/jni/jni_helpers.cc",
"android/jni/jni_helpers.h",
"android/jni/native_handle_impl.cc",
"android/jni/native_handle_impl.h",
"android/jni/peerconnection_jni.cc",
"android/jni/surfacetexturehelper_jni.cc",
"android/jni/surfacetexturehelper_jni.h",
]
configs += [ ":libjingle_peerconnection_jni_warnings_config" ]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [
"//build/config/clang:extra_warnings",
"//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 = [
":libjingle_peerconnection",
]
if (rtc_build_libyuv) {
deps += [ "$rtc_libyuv_dir" ]
public_deps = [
"$rtc_libyuv_dir",
]
} else {
# Need to add a directory normally exported by libyuv.
include_dirs = [ "$rtc_libyuv_dir/include" ]
}
}
rtc_static_library("libjingle_peerconnection_metrics_default_jni") {
sources = [
"android/jni/androidmetrics_jni.cc",
]
configs += [ ":libjingle_peerconnection_jni_warnings_config" ]
deps = [
":libjingle_peerconnection",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
]
}
rtc_shared_library("libjingle_peerconnection_so") {
sources = [
"android/jni/jni_onload.cc",
]
suppressed_configs += [ "//build/config/android:hide_native_jni_exports" ]
deps = [
":libjingle_peerconnection",
":libjingle_peerconnection_jni",
":libjingle_peerconnection_metrics_default_jni",
]
output_extension = "so"
}
}
# Classes that don't require jni can be used in Chromium.
if (is_android) {
android_library("libjingle_peerconnection_java") {
java_files = [
"../modules/audio_device/android/java/src/org/webrtc/voiceengine/BuildInfo.java",
"../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java",
"../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java",
"../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java",
"../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java",
"../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java",
"android/java/src/org/webrtc/AudioSource.java",
"android/java/src/org/webrtc/AudioTrack.java",
"android/java/src/org/webrtc/CallSessionFileRotatingLogSink.java",
"android/java/src/org/webrtc/Camera1Capturer.java",
"android/java/src/org/webrtc/Camera1Enumerator.java",
"android/java/src/org/webrtc/Camera1Session.java",
"android/java/src/org/webrtc/Camera2Capturer.java",
"android/java/src/org/webrtc/Camera2Enumerator.java",
"android/java/src/org/webrtc/Camera2Session.java",
"android/java/src/org/webrtc/CameraCapturer.java",
"android/java/src/org/webrtc/CameraEnumerationAndroid.java",
"android/java/src/org/webrtc/CameraEnumerator.java",
"android/java/src/org/webrtc/CameraSession.java",
"android/java/src/org/webrtc/CameraVideoCapturer.java",
"android/java/src/org/webrtc/DataChannel.java",
"android/java/src/org/webrtc/EglBase.java",
"android/java/src/org/webrtc/EglBase10.java",
"android/java/src/org/webrtc/EglBase14.java",
"android/java/src/org/webrtc/EglRenderer.java",
"android/java/src/org/webrtc/FileVideoCapturer.java",
"android/java/src/org/webrtc/GlRectDrawer.java",
"android/java/src/org/webrtc/GlShader.java",
"android/java/src/org/webrtc/GlTextureFrameBuffer.java",
"android/java/src/org/webrtc/GlUtil.java",
"android/java/src/org/webrtc/Histogram.java",
"android/java/src/org/webrtc/IceCandidate.java",
"android/java/src/org/webrtc/MediaCodecVideoDecoder.java",
"android/java/src/org/webrtc/MediaCodecVideoEncoder.java",
"android/java/src/org/webrtc/MediaConstraints.java",
"android/java/src/org/webrtc/MediaSource.java",
"android/java/src/org/webrtc/MediaStream.java",
"android/java/src/org/webrtc/MediaStreamTrack.java",
"android/java/src/org/webrtc/NetworkMonitor.java",
"android/java/src/org/webrtc/NetworkMonitorAutoDetect.java",
"android/java/src/org/webrtc/PeerConnection.java",
"android/java/src/org/webrtc/PeerConnectionFactory.java",
"android/java/src/org/webrtc/RendererCommon.java",
"android/java/src/org/webrtc/RtpParameters.java",
"android/java/src/org/webrtc/RtpReceiver.java",
"android/java/src/org/webrtc/RtpSender.java",
"android/java/src/org/webrtc/ScreenCapturerAndroid.java",
"android/java/src/org/webrtc/SdpObserver.java",
"android/java/src/org/webrtc/SessionDescription.java",
"android/java/src/org/webrtc/StatsObserver.java",
"android/java/src/org/webrtc/StatsReport.java",
"android/java/src/org/webrtc/SurfaceTextureHelper.java",
"android/java/src/org/webrtc/SurfaceViewRenderer.java",
"android/java/src/org/webrtc/VideoCapturer.java",
"android/java/src/org/webrtc/VideoCapturerAndroid.java",
"android/java/src/org/webrtc/VideoFileRenderer.java",
"android/java/src/org/webrtc/VideoRenderer.java",
"android/java/src/org/webrtc/VideoRendererGui.java",
"android/java/src/org/webrtc/VideoSource.java",
"android/java/src/org/webrtc/VideoTrack.java",
"android/java/src/org/webrtc/YuvConverter.java",
]
deps = [
"../base:base_java",
]
}
android_library("libjingle_peerconnection_metrics_default_java") {
java_files = [ "android/java/src/org/webrtc/Metrics.java" ]
deps = [
"../base:base_java",
]
}
}
rtc_source_set("rtc_stats_api") {
cflags = []
sources = [
"stats/rtcstats.h",
"stats/rtcstats_objects.h",
"stats/rtcstatsreport.h",
]
deps = [
"../base:rtc_base_approved",
]
}
rtc_source_set("audio_mixer_api") {
sources = [
"audio/audio_mixer.h",
]
deps = [
"../base:rtc_base_approved",
]
}
rtc_source_set("transport_api") {
sources = [
"call/transport.h",
]
}
if (rtc_include_tests) {
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",
"-Wno-unused-function",
]
}
if (!is_win) {
cflags = [ "-Wno-sign-compare" ]
}
}
rtc_test("peerconnection_unittests") {
testonly = true
sources = [
"datachannel_unittest.cc",
"dtmfsender_unittest.cc",
"fakemetricsobserver.cc",
"fakemetricsobserver.h",
"jsepsessiondescription_unittest.cc",
"localaudiosource_unittest.cc",
"mediaconstraintsinterface_unittest.cc",
"mediastream_unittest.cc",
"peerconnection_unittest.cc",
"peerconnectionendtoend_unittest.cc",
"peerconnectionfactory_unittest.cc",
"peerconnectioninterface_unittest.cc",
"proxy_unittest.cc",
"rtcstats_integrationtest.cc",
"rtcstatscollector_unittest.cc",
"rtpsenderreceiver_unittest.cc",
"sctputils_unittest.cc",
"statscollector_unittest.cc",
"test/fakeaudiocapturemodule.cc",
"test/fakeaudiocapturemodule.h",
"test/fakeaudiocapturemodule_unittest.cc",
"test/fakeconstraints.h",
"test/fakedatachannelprovider.h",
"test/fakeperiodicvideocapturer.h",
"test/fakertccertificategenerator.h",
"test/fakevideotrackrenderer.h",
"test/mock_datachannel.h",
"test/mock_peerconnection.h",
"test/mock_webrtcsession.h",
"test/mockpeerconnectionobservers.h",
"test/peerconnectiontestwrapper.cc",
"test/peerconnectiontestwrapper.h",
"test/rtcstatsobtainer.h",
"test/testsdpstrings.h",
"videocapturertracksource_unittest.cc",
"videotrack_unittest.cc",
"webrtcsdp_unittest.cc",
"webrtcsession_unittest.cc",
]
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.
]
}
if (rtc_use_quic) {
public_deps = [
"//third_party/libquic",
]
sources += [
"quicdatachannel_unittest.cc",
"quicdatatransport_unittest.cc",
]
}
deps = []
if (is_android) {
sources += [
"test/androidtestinitializer.cc",
"test/androidtestinitializer.h",
]
deps += [
":libjingle_peerconnection_java",
":libjingle_peerconnection_jni",
"//testing/android/native_test:native_test_support",
]
}
deps += [
":libjingle_peerconnection",
"..:webrtc_common",
"../base:rtc_base_tests_utils",
"../media:rtc_unittest_main",
"../pc:rtc_pc",
"../system_wrappers:metrics_default",
"//testing/gmock",
]
if (is_android) {
deps += [ "//testing/android/native_test:native_test_support" ]
shard_timeout = 900
}
}
rtc_source_set("mock_audio_mixer") {
testonly = true
sources = [
"test/mock_audio_mixer.h",
]
public_deps = [
":audio_mixer_api",
]
deps = [
"//testing/gmock",
]
}
if (is_android) {
instrumentation_test_apk("libjingle_peerconnection_android_unittest") {
apk_name = "libjingle_peerconnection_android_unittest"
android_manifest = "androidtests/AndroidManifest.xml"
java_files = [
"androidtests/src/org/webrtc/Camera1CapturerUsingByteBufferTest.java",
"androidtests/src/org/webrtc/Camera1CapturerUsingTextureTest.java",
"androidtests/src/org/webrtc/Camera2CapturerTest.java",
"androidtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java",
"androidtests/src/org/webrtc/EglRendererTest.java",
"androidtests/src/org/webrtc/GlRectDrawerTest.java",
"androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java",
"androidtests/src/org/webrtc/NetworkMonitorTest.java",
"androidtests/src/org/webrtc/PeerConnectionTest.java",
"androidtests/src/org/webrtc/RendererCommonTest.java",
"androidtests/src/org/webrtc/SurfaceTextureHelperTest.java",
"androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java",
"androidtests/src/org/webrtc/WebRtcJniBootTest.java",
]
deps = [
":libjingle_peerconnection_java",
":libjingle_peerconnection_metrics_default_java",
"../base:base_java",
"//base:base_java",
]
shared_libraries = [ ":libjingle_peerconnection_so" ]
}
}
}