Replace test_support_main by test_main and get rid of test_support_main_threaded_mac

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}

webrtc/common_audio/BUILD.gn

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

webrtc/common_audio/resampler/sinc_resampler_unittest.cc

@@ -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::_;
 

webrtc/common_video/BUILD.gn

@@ -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",

webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc

@@ -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.

webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc

@@ -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.

webrtc/modules/BUILD.gn

@@ -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",

webrtc/modules/audio_coding/BUILD.gn

@@ -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",
     ]
 

webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc

@@ -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 {

webrtc/modules/audio_device/BUILD.gn

@@ -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",

webrtc/modules/rtp_rtcp/BUILD.gn

@@ -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
 }

webrtc/modules/video_capture/BUILD.gn

@@ -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).

webrtc/system_wrappers/BUILD.gn

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

webrtc/test/BUILD.gn

@@ -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",

webrtc/test/run_all_unittests.cc

@@ -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();
-}

webrtc/test/test_main.cc

@@ -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();
 }

webrtc/test/test_suite.cc

@@ -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

webrtc/test/test_suite.h

@@ -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_

webrtc/test/testsupport/mac/run_threaded_main_mac.h

@@ -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);

webrtc/test/testsupport/mac/run_threaded_main_mac.mm

@@ -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;
-}

webrtc/tools/BUILD.gn

@@ -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",
     ]
 

webrtc/voice_engine/BUILD.gn

@@ -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) {

webrtc/voice_engine/test/auto_test/standard/codec_test.cc

@@ -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"