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Replace test_support_main by test_main and get rid of test_support_main_threaded_mac

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test_support_main_threaded_mac doesn't seem to be used. It looks like it was
last used about a year and a half ago, and was removed in
https://webrtc-codereview.appspot.com/55379004

BUG=webrtc:6424
R=kjellander@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2540693002
Cr-Commit-Position: refs/heads/master@{#15332}
This commit is contained in:
ehmaldonado
2016-11-30 06:12:01 -08:00
committed by Commit bot
parent 18f7c8d4df
commit 26bddb92f0
22 changed files with 32 additions and 309 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/common_audio/BUILD.gn
View File

@ -289,7 +289,7 @@ if (rtc_include_tests) {
deps = [
":common_audio",
"../test:test_support_main",
"../test:test_main",
"//testing/gmock",
"//testing/gtest",
]

1
webrtc/common_audio/resampler/sinc_resampler_unittest.cc
View File

@ -25,7 +25,6 @@
#include "webrtc/system_wrappers/include/stringize_macros.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/test_suite.h"
using testing::_;

2
webrtc/common_video/BUILD.gn
View File

@ -119,7 +119,7 @@ if (rtc_include_tests) {
deps = [
":common_video",
"../system_wrappers:system_wrappers",
"../test:test_support_main",
"../test:test_main",
"../test:video_test_common",
"//testing/gmock",
"//testing/gtest",

1
webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
View File

@ -28,7 +28,6 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/test_suite.h"
#include "webrtc/test/testsupport/fileutils.h"
// Files generated at build-time by the protobuf compiler.

1
webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
View File

@ -16,7 +16,6 @@
#include "webrtc/base/checks.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/test_suite.h"
#include "webrtc/test/testsupport/fileutils.h"
// Files generated at build-time by the protobuf compiler.

6
webrtc/modules/BUILD.gn
View File

@ -74,8 +74,8 @@ if (rtc_include_tests) {
"../modules/video_coding",
"../modules/video_coding:video_codecs_test_framework",
"../system_wrappers",
"../test:test_main",
"../test:test_support",
"../test:test_support_main",
"//testing/gmock",
"//testing/gtest",
]
@ -649,8 +649,8 @@ if (rtc_include_tests) {
"../system_wrappers",
"../test:rtp_test_utils",
"../test:test_common",
"../test:test_main",
"../test:test_support",
"../test:test_support_main",
"../test:video_test_common",
"audio_coding",
"audio_coding:acm_receive_test",
@ -750,7 +750,7 @@ if (rtc_include_tests) {
"..:webrtc_common",
"../base:rtc_base_approved",
"../test:test_common",
"../test:test_support_main",
"../test:test_main",
"remote_bitrate_estimator:bwe_simulator_lib",
"video_capture",
"//testing/gmock",

14
webrtc/modules/audio_coding/BUILD.gn
View File

@ -1075,7 +1075,7 @@ if (rtc_include_tests) {
":neteq",
":neteq_unittest_tools",
"../../common_audio",
"../../test:test_support_main",
"../../test:test_main",
"//testing/gtest",
]
@ -1173,7 +1173,7 @@ if (rtc_include_tests) {
":isac_fix",
":webrtc_opus",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support_main",
"../../test:test_main",
"../audio_processing",
"//testing/gtest",
]
@ -1451,7 +1451,7 @@ if (rtc_include_tests) {
":neteq_quality_test_support",
":neteq_unittest_tools",
":webrtc_opus",
"../../test:test_support_main",
"../../test:test_main",
"//testing/gtest",
"//third_party/gflags",
]
@ -1498,7 +1498,7 @@ if (rtc_include_tests) {
":neteq_quality_test_support",
":neteq_unittest_tools",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support_main",
"../../test:test_main",
"//testing/gtest",
"//third_party/gflags",
]
@ -1515,7 +1515,7 @@ if (rtc_include_tests) {
":isac_fix",
":neteq",
":neteq_quality_test_support",
"../../test:test_support_main",
"../../test:test_main",
"//testing/gtest",
"//third_party/gflags",
]
@ -1532,7 +1532,7 @@ if (rtc_include_tests) {
":g711",
":neteq",
":neteq_quality_test_support",
"../../test:test_support_main",
"../../test:test_main",
"//testing/gtest",
"//third_party/gflags",
]
@ -1678,8 +1678,8 @@ if (rtc_include_tests) {
":webrtc_opus",
"../../base:rtc_base_approved",
"../../common_audio",
"../../test:test_main",
"../../test:test_support",
"../../test:test_support_main",
"//testing/gtest",
]

1
webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
View File

@ -20,7 +20,6 @@
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/test_suite.h"
#include "webrtc/test/testsupport/fileutils.h"
namespace webrtc {

2
webrtc/modules/audio_device/BUILD.gn
View File

@ -282,8 +282,8 @@ if (rtc_include_tests && !is_ios) {
":audio_device",
"../..:webrtc_common",
"../../system_wrappers",
"../../test:test_main",
"../../test:test_support",
"../../test:test_support_main",
"../rtp_rtcp",
"../utility",
"//testing/gtest",

2
webrtc/modules/rtp_rtcp/BUILD.gn
View File

@ -203,7 +203,7 @@ if (rtc_include_tests) {
deps = [
":rtp_rtcp",
"//testing/gtest",
"//webrtc/test:test_support_main",
"//webrtc/test:test_main",
]
} # test_packet_masks_metrics
}

2
webrtc/modules/video_capture/BUILD.gn
View File

@ -187,7 +187,7 @@ if (!build_with_chromium) {
"../utility",
"//testing/gtest",
]
deps += [ "//webrtc/test:test_support_main" ]
deps += [ "//webrtc/test:test_main" ]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).

2
webrtc/system_wrappers/BUILD.gn
View File

@ -196,7 +196,7 @@ if (rtc_include_tests) {
deps = [
":metrics_default",
":system_wrappers",
"../test:test_support_main",
"../test:test_main",
"//testing/gtest",
]

45
webrtc/test/BUILD.gn
View File

@ -22,9 +22,9 @@ group("test") {
":field_trial",
":rtp_test_utils",
":test_common",
":test_main",
":test_renderer",
":test_support",
":test_support_main",
":test_support_unittests",
":video_test_common",
]
@ -98,6 +98,7 @@ rtc_source_set("test_main") {
":field_trial",
":test_support",
"../system_wrappers:metrics_default",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
]
@ -163,46 +164,6 @@ rtc_source_set("test_support") {
}
}
# Depend on this target when you want to have test_support but also the
# main method needed for gtest to execute!
rtc_source_set("test_support_main") {
testonly = true
sources = [
"run_all_unittests.cc",
"test_suite.cc",
"test_suite.h",
]
deps = [
":field_trial",
":test_support",
"../system_wrappers:metrics_default",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
]
}
# Depend on this target when you want to have test_support and a special
# main for mac which will run your test on a worker thread and consume
# events on the main thread. Useful if you want to access a webcam.
# This main will provide all the scaffolding and objective-c black magic
# for you. All you need to do is to implement a function in the
# run_threaded_main_mac.h file (ImplementThisToRunYourTest).
rtc_source_set("test_support_main_threaded_mac") {
testonly = true
sources = [
"testsupport/mac/run_threaded_main_mac.h",
"testsupport/mac/run_threaded_main_mac.mm",
]
deps = [
":test_support",
]
}
rtc_source_set("run_test") {
testonly = true
sources = [
@ -276,7 +237,7 @@ rtc_test("test_support_unittests") {
deps += [
":test_common",
":test_support_main",
":test_main",
"../modules/video_capture",
"//testing/gmock",
"//testing/gtest",

16
webrtc/test/run_all_unittests.cc
View File

@ -1,16 +0,0 @@
/*
* Copyright (c) 2011 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 "webrtc/test/test_suite.h"
int main(int argc, char** argv) {
webrtc::test::TestSuite test_suite(argc, argv);
return test_suite.Run();
}

12
webrtc/test/test_main.cc
View File

@ -12,8 +12,12 @@
#include "webrtc/base/logging.h"
#include "webrtc/system_wrappers/include/metrics_default.h"
#include "webrtc/test/field_trial.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/trace_to_stderr.h"
DEFINE_bool(logs, false, "print logs to stderr");
DEFINE_string(force_fieldtrials, "",
"Field trials control experimental feature code which can be forced. "
@ -21,7 +25,7 @@ DEFINE_string(force_fieldtrials, "",
" will assign the group Enable to field trial WebRTC-FooFeature.");
int main(int argc, char* argv[]) {
::testing::InitGoogleTest(&argc, argv);
::testing::InitGoogleMock(&argc, argv);
// Default to LS_INFO, even for release builds to provide better test logging.
// TODO(pbos): Consider adding a command-line override.
@ -36,5 +40,11 @@ int main(int argc, char* argv[]) {
webrtc::test::SetExecutablePath(argv[0]);
webrtc::test::InitFieldTrialsFromString(FLAGS_force_fieldtrials);
webrtc::metrics::Enable();
rtc::LogMessage::SetLogToStderr(FLAGS_logs);
std::unique_ptr<webrtc::test::TraceToStderr> trace_to_stderr;
if (FLAGS_logs)
trace_to_stderr.reset(new webrtc::test::TraceToStderr);
return RUN_ALL_TESTS();
}

64
webrtc/test/test_suite.cc
View File

@ -1,64 +0,0 @@
/*
* Copyright (c) 2011 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 "webrtc/test/test_suite.h"
#include "gflags/gflags.h"
#include "webrtc/base/logging.h"
#include "webrtc/system_wrappers/include/metrics_default.h"
#include "webrtc/test/field_trial.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/trace_to_stderr.h"
DEFINE_bool(logs, false, "print logs to stderr");
DEFINE_string(force_fieldtrials, "",
"Field trials control experimental feature code which can be forced. "
"E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
" will assign the group Enable to field trial WebRTC-FooFeature.");
namespace webrtc {
namespace test {
TestSuite::TestSuite(int argc, char** argv) {
SetExecutablePath(argv[0]);
testing::InitGoogleMock(&argc, argv); // Runs InitGoogleTest() internally.
// AllowCommandLineParsing allows us to ignore flags passed on to us by
// Chromium build bots without having to explicitly disable them.
google::AllowCommandLineReparsing();
google::ParseCommandLineFlags(&argc, &argv, true);
webrtc::test::InitFieldTrialsFromString(FLAGS_force_fieldtrials);
webrtc::metrics::Enable();
}
TestSuite::~TestSuite() {
}
int TestSuite::Run() {
Initialize();
int result = RUN_ALL_TESTS();
Shutdown();
return result;
}
void TestSuite::Initialize() {
rtc::LogMessage::SetLogToStderr(FLAGS_logs);
if (FLAGS_logs)
trace_to_stderr_.reset(new TraceToStderr);
}
void TestSuite::Shutdown() {
}
} // namespace test
} // namespace webrtc

51
webrtc/test/test_suite.h
View File

@ -1,51 +0,0 @@
/*
* Copyright (c) 2011 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.
*/
#ifndef WEBRTC_TEST_TEST_SUITE_H_
#define WEBRTC_TEST_TEST_SUITE_H_
// Derived from Chromium's src/base/test/test_suite.h.
// Defines a basic test suite framework for running gtest based tests. You can
// instantiate this class in your main function and call its Run method to run
// any gtest based tests that are linked into your executable.
#include <memory>
#include "webrtc/base/constructormagic.h"
namespace webrtc {
namespace test {
class TraceToStderr;
class TestSuite {
public:
TestSuite(int argc, char** argv);
virtual ~TestSuite();
int Run();
protected:
// Override these for custom initialization and shutdown handling. Use these
// instead of putting complex code in your constructor/destructor.
virtual void Initialize();
virtual void Shutdown();
RTC_DISALLOW_COPY_AND_ASSIGN(TestSuite);
private:
std::unique_ptr<TraceToStderr> trace_to_stderr_;
};
} // namespace test
} // namespace webrtc
#endif // TEST_TEST_SUITE_H_

22
webrtc/test/testsupport/mac/run_threaded_main_mac.h
View File

@ -1,22 +0,0 @@
/*
* Copyright (c) 2012 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.
*/
/**
* This file and its corresponding .mm file implement a main function on Mac.
* It's useful if you need to access a webcam in your Mac application. The code
* forks a worker thread which runs the below ImplementThisToRunYourTest
* function, and uses the main thread to pump messages. That way we can run our
* code in a regular sequential fashion and still pump events, which are
* necessary to access the webcam for instance.
*/
// Implement this method to do whatever you want to do in the worker thread.
// The argc and argv variables are the unmodified command line from main.
int ImplementThisToRunYourTest(int argc, char** argv);

90
webrtc/test/testsupport/mac/run_threaded_main_mac.mm
View File

@ -1,90 +0,0 @@
/*
* Copyright (c) 2012 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/test/testsupport/mac/run_threaded_main_mac.h"
#import <Cocoa/Cocoa.h>
// This class passes parameter from main to the worked thread and back.
@interface AutoTestInWorkerThread : NSObject {
int argc_;
char** argv_;
int result_;
bool done_;
}
- (void)setDone:(bool)done;
- (bool)done;
- (void)setArgc:(int)argc argv:(char**)argv;
- (int) result;
- (void)runTest:(NSObject*)ignored;
@end
@implementation AutoTestInWorkerThread
- (void)setDone:(bool)done {
done_ = done;
}
- (bool)done {
return done_;
}
- (void)setArgc:(int)argc argv:(char**)argv {
argc_ = argc;
argv_ = argv;
}
- (void)runTest:(NSObject*)ignored {
NSAutoreleasePool* pool = [[NSAutoreleasePool alloc] init];
result_ = ImplementThisToRunYourTest(argc_, argv_);
done_ = true;
[pool release];
return;
}
- (int)result {
return result_;
}
@end
int main(int argc, char * argv[]) {
NSAutoreleasePool* pool = [[NSAutoreleasePool alloc] init];
[NSApplication sharedApplication];
int result = 0;
AutoTestInWorkerThread* tests = [[AutoTestInWorkerThread alloc] init];
[tests setArgc:argc argv:argv];
[tests setDone:false];
[NSThread detachNewThreadSelector:@selector(runTest:)
toTarget:tests
withObject:nil];
NSRunLoop* main_run_loop = [NSRunLoop mainRunLoop];
NSDate *loop_until = [NSDate dateWithTimeIntervalSinceNow:0.1];
bool runloop_ok = true;
while (![tests done] && runloop_ok) {
runloop_ok = [main_run_loop runMode:NSDefaultRunLoopMode
beforeDate:loop_until];
loop_until = [NSDate dateWithTimeIntervalSinceNow:0.1];
}
result = [tests result];
[pool release];
return result;
}

2
webrtc/tools/BUILD.gn
View File

@ -271,7 +271,7 @@ if (rtc_include_tests) {
":command_line_parser",
":frame_editing_lib",
":video_quality_analysis",
"../test:test_support_main",
"../test:test_main",
"//testing/gtest",
]

2
webrtc/voice_engine/BUILD.gn
View File

@ -199,7 +199,7 @@ if (rtc_include_tests) {
"//webrtc/modules/rtp_rtcp",
"//webrtc/modules/utility",
"//webrtc/system_wrappers",
"//webrtc/test:test_support_main",
"//webrtc/test:test_main",
]
if (is_android) {

1
webrtc/voice_engine/test/auto_test/standard/codec_test.cc
View File

@ -11,7 +11,6 @@
#include <stdio.h>
#include <string>
#include "webrtc/test/test_suite.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
#include "webrtc/voice_engine/voice_engine_defines.h"