Add a test which breaks if libwebrtc.a don't pull in the right symbols.
It's impossible to ensure we are pulling in everything people
reasonably believe is used, but it should be a good chunk of it.
I don't plan to actually run this test on the bots, it's enough if
it is built (which it should, because I add it to the default set
of things to build).
Bug: webrtc:11027
Change-Id: I186936eeb450d2f63b3a5bed13189e84d5b3fb76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157175
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29518}
diff --git a/BUILD.gn b/BUILD.gn
index 31cf726..f7eca78 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -73,6 +73,9 @@
} else {
deps += [ "modules/video_capture:video_capture_tests" ]
}
+ if (!is_android && !is_ios) {
+ deps += [ ":webrtc_lib_link_test" ]
+ }
if (rtc_enable_protobuf) {
deps += [
"audio:low_bandwidth_audio_test",
@@ -389,8 +392,11 @@
if (!build_with_chromium) {
# Target to build all the WebRTC production code.
rtc_static_library("webrtc") {
- # Only the root target should depend on this.
- visibility = [ "//:default" ]
+ # Only the root target and the test should depend on this.
+ visibility = [
+ "//:default",
+ "//:webrtc_lib_link_test",
+ ]
sources = []
complete_static_lib = true
@@ -459,6 +465,23 @@
deps += [ "logging:rtc_event_log_proto" ]
}
}
+
+ if (rtc_include_tests && !is_android && !is_ios) {
+ # Note: This test can't work on mobile because the test runner machinery
+ # on those platforms depend on abseil, which will link-clash with libwebrtc.
+ rtc_test("webrtc_lib_link_test") {
+ testonly = true
+
+ sources = [
+ "webrtc_lib_link_test.cc",
+ ]
+ deps = [
+ ":webrtc",
+ "//test:test_main",
+ "//testing/gtest",
+ ]
+ }
+ }
}
rtc_source_set("webrtc_common") {
diff --git a/DEPS b/DEPS
index 0dd43c2..839a927 100644
--- a/DEPS
+++ b/DEPS
@@ -2280,3 +2280,11 @@
# Abseil flags are allowed in tests and tools.
"+absl/flags",
]
+
+specific_include_rules = {
+ "webrtc_lib_link_test\.cc": [
+ "+media/engine",
+ "+modules/audio_device",
+ "+modules/audio_processing",
+ ]
+}
diff --git a/webrtc_lib_link_test.cc b/webrtc_lib_link_test.cc
new file mode 100644
index 0000000..afd787f
--- /dev/null
+++ b/webrtc_lib_link_test.cc
@@ -0,0 +1,87 @@
+/*
+ * Copyright 2019 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.
+ */
+
+#include "api/audio_codecs/audio_decoder_factory_template.h"
+#include "api/audio_codecs/audio_encoder_factory_template.h"
+#include "api/audio_codecs/opus/audio_decoder_opus.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "api/call/call_factory_interface.h"
+#include "api/create_peerconnection_factory.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_event_log/rtc_event_log_factory.h"
+#include "api/stats/rtcstats_objects.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/video_codecs/builtin_video_decoder_factory.h"
+#include "api/video_codecs/builtin_video_encoder_factory.h"
+#include "media/engine/webrtc_media_engine.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+cricket::MediaEngineDependencies CreateSomeMediaDeps(
+ TaskQueueFactory* task_queue_factory) {
+ cricket::MediaEngineDependencies media_deps;
+ media_deps.task_queue_factory = task_queue_factory;
+ media_deps.adm = AudioDeviceModule::CreateForTest(
+ AudioDeviceModule::kDummyAudio, task_queue_factory);
+ media_deps.audio_encoder_factory =
+ webrtc::CreateAudioEncoderFactory<webrtc::AudioEncoderOpus>();
+ media_deps.audio_decoder_factory =
+ webrtc::CreateAudioDecoderFactory<webrtc::AudioDecoderOpus>();
+ media_deps.video_encoder_factory = CreateBuiltinVideoEncoderFactory();
+ media_deps.video_decoder_factory = webrtc::CreateBuiltinVideoDecoderFactory();
+ media_deps.audio_processing = webrtc::AudioProcessingBuilder().Create();
+ return media_deps;
+}
+
+// This test should pull in as much of WebRTC as possible to make sure most
+// commonly used symbols are actually in libwebrtc.a.
+webrtc::PeerConnectionFactoryDependencies CreateSomePcfDeps() {
+ webrtc::PeerConnectionFactoryDependencies pcf_deps;
+ pcf_deps.task_queue_factory = CreateDefaultTaskQueueFactory();
+ pcf_deps.signaling_thread = rtc::Thread::Current();
+ pcf_deps.network_thread = rtc::Thread::Current();
+ pcf_deps.worker_thread = rtc::Thread::Current();
+ pcf_deps.call_factory = webrtc::CreateCallFactory();
+ pcf_deps.event_log_factory = std::make_unique<webrtc::RtcEventLogFactory>(
+ pcf_deps.task_queue_factory.get());
+ auto media_deps = CreateSomeMediaDeps(pcf_deps.task_queue_factory.get());
+ pcf_deps.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
+ return pcf_deps;
+}
+
+TEST(WebRTCLinkTest, TestCreatingAPeerConnectionViaModularFactory) {
+ auto pcf_deps = CreateSomePcfDeps();
+ auto peer_connection_factory =
+ webrtc::CreateModularPeerConnectionFactory(std::move(pcf_deps));
+ webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
+ auto peer_connection = peer_connection_factory->CreatePeerConnection(
+ rtc_config, nullptr, nullptr, nullptr);
+ ASSERT_EQ(peer_connection.get(), nullptr)
+ << "Should fail, we're not setting things up right";
+}
+
+TEST(WebRTCLinkTest, TestCreatingViaPCFactory) {
+ auto task_queue_factory = CreateDefaultTaskQueueFactory();
+ auto media_deps = CreateSomeMediaDeps(task_queue_factory.get());
+
+ auto peer_connection_factory = webrtc::CreatePeerConnectionFactory(
+ rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
+ std::move(media_deps.adm), std::move(media_deps.audio_encoder_factory),
+ std::move(media_deps.audio_decoder_factory),
+ std::move(media_deps.video_encoder_factory),
+ std::move(media_deps.video_decoder_factory), nullptr, nullptr);
+ ASSERT_NE(peer_connection_factory.get(), nullptr);
+}
+
+} // namespace webrtc
