Clean up dependencies of peerconnection_unittest.
There were a number of unused includes and undeclared
dependencies. I removed the includes that were causing
problems and added dependencies for the includes that
turned out to be needed.
Bug: webrtc:7239,webrtc:6828
Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284
Reviewed-on: https://webrtc-review.googlesource.com/1185
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19858}diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 09c58ab..8c6b5a6 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -373,7 +373,6 @@
}
rtc_test("peerconnection_unittests") {
- check_includes = false # TODO(kjellander): Remove (bugs.webrtc.org/6828)
testonly = true
sources = [
"datachannel_unittest.cc",
@@ -435,15 +434,7 @@
deps = []
if (is_android) {
- sources += [
- "test/androidtestinitializer.cc",
- "test/androidtestinitializer.h",
- ]
- deps += [
- "../sdk/android:libjingle_peerconnection_java",
- "../sdk/android:libjingle_peerconnection_jni",
- "//testing/android/native_test:native_test_support",
- ]
+ deps += [ ":android_black_magic" ]
}
deps += [
@@ -451,19 +442,55 @@
":pc_test_utils",
"..:webrtc_common",
"../api:fakemetricsobserver",
+ "../api:libjingle_peerconnection_test_api",
+ "../api:rtc_stats_api",
+ "../api/audio_codecs:builtin_audio_decoder_factory",
+ "../api/audio_codecs:builtin_audio_encoder_factory",
+ "../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_approved",
"../rtc_base:rtc_base_tests_main",
"../rtc_base:rtc_base_tests_utils",
"../system_wrappers:metrics_default",
"../test:audio_codec_mocks",
- "//testing/gmock",
+ "../test:test_support",
]
if (is_android) {
- deps += [ "//testing/android/native_test:native_test_support" ]
+ 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",
+ ]
+ }
+ }
}
diff --git a/pc/dtmfsender_unittest.cc b/pc/dtmfsender_unittest.cc
index 21343f6..a5d102b 100644
--- a/pc/dtmfsender_unittest.cc
+++ b/pc/dtmfsender_unittest.cc
@@ -15,10 +15,10 @@
#include <string>
#include <vector>
+
#include "pc/audiotrack.h"
#include "rtc_base/fakeclock.h"
#include "rtc_base/gunit.h"
-#include "rtc_base/logging.h"
#include "rtc_base/timeutils.h"
using webrtc::AudioTrackInterface;
@@ -39,7 +39,6 @@
// Implements DtmfSenderObserverInterface.
void OnToneChange(const std::string& tone) override {
- LOG(LS_VERBOSE) << "FakeDtmfObserver::OnToneChange '" << tone << "'.";
tones_.push_back(tone);
if (tone.empty()) {
completed_ = true;
@@ -89,9 +88,6 @@
}
last_insert_dtmf_call_ = rtc::TimeMillis();
- LOG(LS_VERBOSE) << "FakeDtmfProvider::InsertDtmf code=" << code
- << " duration=" << duration
- << " gap=" << gap << ".";
dtmf_info_queue_.push_back(DtmfInfo(code, duration, gap));
return true;
}
diff --git a/pc/jsepsessiondescription_unittest.cc b/pc/jsepsessiondescription_unittest.cc
index 473e7a0..781b596 100644
--- a/pc/jsepsessiondescription_unittest.cc
+++ b/pc/jsepsessiondescription_unittest.cc
@@ -19,8 +19,6 @@
#include "p2p/base/sessiondescription.h"
#include "pc/mediasession.h"
#include "rtc_base/gunit.h"
-#include "rtc_base/helpers.h"
-#include "rtc_base/ssladapter.h"
#include "rtc_base/stringencode.h"
using webrtc::IceCandidateCollection;
diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index 57f9e73..fa99d38 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -42,13 +42,8 @@
#include "pc/test/fakertccertificategenerator.h"
#include "pc/test/fakevideotrackrenderer.h"
#include "pc/test/mockpeerconnectionobservers.h"
-#include "rtc_base/asyncinvoker.h"
#include "rtc_base/fakenetwork.h"
#include "rtc_base/gunit.h"
-#include "rtc_base/helpers.h"
-#include "rtc_base/ssladapter.h"
-#include "rtc_base/sslstreamadapter.h"
-#include "rtc_base/thread.h"
#include "rtc_base/virtualsocketserver.h"
using cricket::ContentInfo;
diff --git a/pc/peerconnectionendtoend_unittest.cc b/pc/peerconnectionendtoend_unittest.cc
index 1dca5ff..b151264 100644
--- a/pc/peerconnectionendtoend_unittest.cc
+++ b/pc/peerconnectionendtoend_unittest.cc
@@ -15,11 +15,9 @@
#include "rtc_base/gunit.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
-#include "rtc_base/ssladapter.h"
-#include "rtc_base/sslstreamadapter.h"
#include "rtc_base/stringencode.h"
#include "rtc_base/stringutils.h"
-#include "rtc_base/thread.h"
+
#ifdef WEBRTC_ANDROID
#include "pc/test/androidtestinitializer.h"
#endif
diff --git a/pc/peerconnectionfactory_unittest.cc b/pc/peerconnectionfactory_unittest.cc
index 36553a7..5688d15 100644
--- a/pc/peerconnectionfactory_unittest.cc
+++ b/pc/peerconnectionfactory_unittest.cc
@@ -14,12 +14,10 @@
#include "api/mediastreaminterface.h"
#include "media/base/fakevideocapturer.h"
-#include "media/engine/webrtccommon.h"
-#include "media/engine/webrtcvoe.h"
#include "p2p/base/fakeportallocator.h"
#include "pc/peerconnectionfactory.h"
#include "rtc_base/gunit.h"
-#include "rtc_base/thread.h"
+
#ifdef WEBRTC_ANDROID
#include "pc/test/androidtestinitializer.h"
#endif
diff --git a/pc/peerconnectioninterface_unittest.cc b/pc/peerconnectioninterface_unittest.cc
index c1facac..a3cf315 100644
--- a/pc/peerconnectioninterface_unittest.cc
+++ b/pc/peerconnectioninterface_unittest.cc
@@ -40,10 +40,7 @@
#include "pc/videocapturertracksource.h"
#include "pc/videotrack.h"
#include "rtc_base/gunit.h"
-#include "rtc_base/ssladapter.h"
-#include "rtc_base/sslstreamadapter.h"
#include "rtc_base/stringutils.h"
-#include "rtc_base/thread.h"
#include "rtc_base/virtualsocketserver.h"
#include "test/gmock.h"
diff --git a/pc/proxy_unittest.cc b/pc/proxy_unittest.cc
index 00d2195..81f040a 100644
--- a/pc/proxy_unittest.cc
+++ b/pc/proxy_unittest.cc
@@ -15,7 +15,6 @@
#include "rtc_base/gunit.h"
#include "rtc_base/refcount.h"
-#include "rtc_base/thread.h"
#include "test/gmock.h"
using ::testing::_;
diff --git a/pc/rtcstatscollector_unittest.cc b/pc/rtcstatscollector_unittest.cc
index 48048c2..cb5a79f 100644
--- a/pc/rtcstatscollector_unittest.cc
+++ b/pc/rtcstatscollector_unittest.cc
@@ -22,7 +22,6 @@
#include "api/stats/rtcstatsreport.h"
#include "api/test/mock_rtpreceiver.h"
#include "api/test/mock_rtpsender.h"
-#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/base/fakemediaengine.h"
#include "media/base/test/mock_mediachannel.h"
#include "p2p/base/p2pconstants.h"
@@ -38,7 +37,6 @@
#include "rtc_base/fakesslidentity.h"
#include "rtc_base/gunit.h"
#include "rtc_base/logging.h"
-#include "rtc_base/socketaddress.h"
#include "rtc_base/thread_checker.h"
#include "rtc_base/timedelta.h"
#include "rtc_base/timeutils.h"
diff --git a/pc/rtpsenderreceiver_unittest.cc b/pc/rtpsenderreceiver_unittest.cc
index b4b28bd..5b0b59e 100644
--- a/pc/rtpsenderreceiver_unittest.cc
+++ b/pc/rtpsenderreceiver_unittest.cc
@@ -12,9 +12,7 @@
#include <string>
#include <utility>
-#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/base/fakemediaengine.h"
-#include "media/base/mediachannel.h"
#include "media/engine/fakewebrtccall.h"
#include "p2p/base/faketransportcontroller.h"
#include "pc/audiotrack.h"
@@ -29,7 +27,6 @@
#include "pc/videotrack.h"
#include "pc/videotracksource.h"
#include "rtc_base/gunit.h"
-#include "rtc_base/sigslot.h"
#include "test/gmock.h"
#include "test/gtest.h"
diff --git a/pc/statscollector_unittest.cc b/pc/statscollector_unittest.cc
index d2d34b5..6a1d827 100644
--- a/pc/statscollector_unittest.cc
+++ b/pc/statscollector_unittest.cc
@@ -16,7 +16,6 @@
#include "pc/statscollector.h"
#include "api/mediastreaminterface.h"
-#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/base/fakemediaengine.h"
#include "media/base/test/mock_mediachannel.h"
#include "pc/channelmanager.h"
@@ -32,7 +31,6 @@
#include "rtc_base/base64.h"
#include "rtc_base/fakesslidentity.h"
#include "rtc_base/gunit.h"
-#include "rtc_base/network.h"
#include "rtc_base/stringencode.h"
#include "test/gmock.h"
#include "test/gtest.h"
diff --git a/pc/test/androidtestinitializer.cc b/pc/test/androidtestinitializer.cc
index c9525f8..f39a3e7 100644
--- a/pc/test/androidtestinitializer.cc
+++ b/pc/test/androidtestinitializer.cc
@@ -12,6 +12,7 @@
#include <pthread.h>
+
#include "rtc_base/ignore_wundef.h"
// Note: this dependency is dangerous since it reaches into Chromium's base.
@@ -22,8 +23,13 @@
#include "base/android/jni_android.h"
RTC_POP_IGNORING_WUNDEF()
+
#include "modules/utility/include/jvm_android.h"
#include "rtc_base/checks.h"
+
+// TODO(phoglund): This include is also to a target we can't really depend on.
+// We need to either break it out into a smaller target or find some way to
+// not use it.
#include "rtc_base/ssladapter.h"
namespace webrtc {
diff --git a/pc/test/fakeaudiocapturemodule_unittest.cc b/pc/test/fakeaudiocapturemodule_unittest.cc
index 6fc44a5..312c4e8 100644
--- a/pc/test/fakeaudiocapturemodule_unittest.cc
+++ b/pc/test/fakeaudiocapturemodule_unittest.cc
@@ -15,7 +15,6 @@
#include "rtc_base/criticalsection.h"
#include "rtc_base/gunit.h"
#include "rtc_base/scoped_ref_ptr.h"
-#include "rtc_base/thread.h"
using std::min;
diff --git a/pc/webrtcsdp_unittest.cc b/pc/webrtcsdp_unittest.cc
index 3acad29..eea8e01 100644
--- a/pc/webrtcsdp_unittest.cc
+++ b/pc/webrtcsdp_unittest.cc
@@ -16,16 +16,14 @@
#include "api/jsepsessiondescription.h"
#include "media/base/mediaconstants.h"
#include "media/engine/webrtcvideoengine.h"
-#include "modules/video_coding/codecs/h264/include/h264.h"
#include "p2p/base/p2pconstants.h"
#include "pc/mediasession.h"
#include "rtc_base/checks.h"
#include "rtc_base/gunit.h"
#include "rtc_base/logging.h"
-#include "rtc_base/messagedigest.h"
-#include "rtc_base/sslfingerprint.h"
#include "rtc_base/stringencode.h"
#include "rtc_base/stringutils.h"
+
#ifdef WEBRTC_ANDROID
#include "pc/test/androidtestinitializer.h"
#endif
diff --git a/pc/webrtcsession_unittest.cc b/pc/webrtcsession_unittest.cc
index 8c2331a..fe0049d 100644
--- a/pc/webrtcsession_unittest.cc
+++ b/pc/webrtcsession_unittest.cc
@@ -15,7 +15,6 @@
#include "api/fakemetricsobserver.h"
#include "api/jsepicecandidate.h"
#include "api/jsepsessiondescription.h"
-#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/base/fakemediaengine.h"
#include "media/base/fakevideorenderer.h"
#include "media/base/mediachannel.h"
@@ -40,12 +39,7 @@
#include "rtc_base/firewallsocketserver.h"
#include "rtc_base/gunit.h"
#include "rtc_base/logging.h"
-#include "rtc_base/network.h"
-#include "rtc_base/ssladapter.h"
-#include "rtc_base/sslidentity.h"
-#include "rtc_base/sslstreamadapter.h"
#include "rtc_base/stringutils.h"
-#include "rtc_base/thread.h"
#include "rtc_base/virtualsocketserver.h"
using cricket::FakeVoiceMediaChannel;
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 31848df..7a76b7e 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -777,6 +777,7 @@
"nattypes.h",
"proxyserver.cc",
"proxyserver.h",
+ "refcount.h",
"sha1.cc",
"sha1.h",
"sha1digest.cc",
