Enable the -Wundef warning for clang

NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2358993004
Cr-Commit-Position: refs/heads/master@{#14425}

webrtc/BUILD.gn

@@ -179,6 +179,7 @@ config("common_config") {
         "-Wimplicit-fallthrough",
         "-Wthread-safety",
         "-Winconsistent-missing-override",
+        "-Wundef",
       ]
     }
   }

webrtc/DEPS

@@ -7,7 +7,6 @@ include_rules = [
   "+external/webrtc/webrtc",  # Android platform build.
   "+gflags",
   "+libyuv",
-  "+testing",
   "-webrtc",  # Has to be disabled; otherwise all dirs below will be allowed.
   # Individual headers that will be moved out of here, see webrtc:4243.
   "+webrtc/call.h",

webrtc/api/mediastream_unittest.cc

@@ -14,8 +14,8 @@
 #include "webrtc/api/mediastream.h"
 #include "webrtc/api/videotrack.h"
 #include "webrtc/api/test/fakevideotracksource.h"
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/refcount.h"
 

webrtc/api/peerconnectioninterface_unittest.cc

@@ -12,7 +12,7 @@
 #include <string>
 #include <utility>
 
-#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/test/gmock.h"
 #include "webrtc/api/audiotrack.h"
 #include "webrtc/api/jsepsessiondescription.h"
 #include "webrtc/api/mediastream.h"

webrtc/api/proxy_unittest.cc

@@ -13,7 +13,7 @@
 #include <memory>
 #include <string>
 
-#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/test/gmock.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/thread.h"

webrtc/api/rtpsenderreceiver_unittest.cc

@@ -12,8 +12,8 @@
 #include <string>
 #include <utility>
 
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/api/audiotrack.h"
 #include "webrtc/api/fakemediacontroller.h"
 #include "webrtc/api/localaudiosource.h"

webrtc/api/statscollector_unittest.cc

@@ -15,8 +15,8 @@
 
 #include "webrtc/api/statscollector.h"
 
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/api/mediastream.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/mediastreamtrack.h"

webrtc/api/test/androidtestinitializer.cc

@@ -12,12 +12,16 @@
 
 #include <pthread.h>
 
+#include "webrtc/base/ignore_wundef.h"
+
 // Note: this dependency is dangerous since it reaches into Chromium's base.
 // There's a risk of e.g. macro clashes. This file may only be used in tests.
 // Since we use Chromes build system for creating the gtest binary, this should
 // be fine.
+RTC_PUSH_IGNORING_WUNDEF()
 #include "base/android/context_utils.h"
 #include "base/android/jni_android.h"
+RTC_POP_IGNORING_WUNDEF()
 
 #include "webrtc/api/android/jni/classreferenceholder.h"
 #include "webrtc/api/android/jni/jni_helpers.h"

webrtc/api/test/mock_datachannel.h

@@ -12,7 +12,7 @@
 #define WEBRTC_API_TEST_MOCK_DATACHANNEL_H_
 
 #include "webrtc/api/datachannel.h"
-#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/test/gmock.h"
 
 namespace webrtc {
 

webrtc/api/test/mock_peerconnection.h

@@ -14,7 +14,7 @@
 #include <vector>
 
 #include "webrtc/api/peerconnection.h"
-#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/test/gmock.h"
 
 namespace webrtc {
 

webrtc/api/test/mock_webrtcsession.h

@@ -15,7 +15,7 @@
 #include <string>
 
 #include "webrtc/api/webrtcsession.h"
-#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/test/gmock.h"
 
 namespace webrtc {
 

webrtc/audio/audio_receive_stream_unittest.cc

@@ -11,7 +11,7 @@
 #include <string>
 #include <vector>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/conversion.h"

webrtc/audio/audio_send_stream_unittest.cc

@@ -11,7 +11,7 @@
 #include <string>
 #include <vector>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"

webrtc/audio/audio_state_unittest.cc

@@ -10,7 +10,7 @@
 
 #include <memory>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/test/mock_voice_engine.h"

webrtc/base/BUILD.gn

@@ -117,6 +117,7 @@ rtc_static_library("rtc_base_approved") {
     "file.h",
     "format_macros.h",
     "function_view.h",
+    "ignore_wundef.h",
     "location.cc",
     "location.h",
     "md5.cc",

webrtc/base/base.gyp

@@ -59,6 +59,7 @@
         'file.h',
         'format_macros.h',
         'function_view.h',
+        'ignore_wundef.h',
         'location.h',
         'location.cc',
         'md5.cc',

webrtc/base/basictypes.h

@@ -31,7 +31,7 @@
 
 #if !defined(RTC_ARCH_CPU_BIG_ENDIAN) && !defined(RTC_ARCH_CPU_LITTLE_ENDIAN)
 // x86, arm or GCC provided __BYTE_ORDER__ macros
-#if CPU_X86 || CPU_ARM ||  \
+#if defined(CPU_X86) || defined(CPU_ARM) ||                             \
   (defined(__BYTE_ORDER__) && __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__)
 #define RTC_ARCH_CPU_LITTLE_ENDIAN
 #elif defined(__BYTE_ORDER__) && __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__

webrtc/base/bitbuffer_unittest.cc

@@ -168,7 +168,7 @@ TEST(BitBufferTest, SetOffsetValues) {
 
   // Disable death test on Android because it relies on fork() and doesn't play
   // nicely.
-#if defined(GTEST_HAS_DEATH_TEST)
+#if GTEST_HAS_DEATH_TEST
 #if !defined(WEBRTC_ANDROID)
   // Passing a NULL out parameter is death.
   EXPECT_DEATH(buffer.GetCurrentOffset(&byte_offset, NULL), "");

webrtc/base/common.cc

@@ -13,7 +13,7 @@
 #include <stdio.h>
 #include <string.h>
 
-#if WEBRTC_WIN
+#ifdef WEBRTC_WIN
 #define WIN32_LEAN_AND_MEAN
 #include <windows.h>
 #endif  // WEBRTC_WIN 
@@ -33,7 +33,7 @@
 namespace rtc {
 
 void Break() {
-#if WEBRTC_WIN
+#ifdef WEBRTC_WIN
   ::DebugBreak();
 #else  // !WEBRTC_WIN 
   // On POSIX systems, SIGTRAP signals debuggers to break without killing the

webrtc/base/criticalsection.h

@@ -39,6 +39,8 @@
 
 #if (!defined(NDEBUG) || defined(DCHECK_ALWAYS_ON))
 #define CS_DEBUG_CHECKS 1
+#else
+#define CS_DEBUG_CHECKS 0
 #endif
 
 #if CS_DEBUG_CHECKS

webrtc/base/event.cc

@@ -86,7 +86,7 @@ bool Event::Wait(int milliseconds) {
     // Converting from seconds and microseconds (1e-6) plus
     // milliseconds (1e-3) to seconds and nanoseconds (1e-9).
 
-#if HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE
+#ifdef HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE
     // Use relative time version, which tends to be more efficient for
     // pthread implementations where provided (like on Android).
     ts.tv_sec = milliseconds / 1000;
@@ -109,7 +109,7 @@ bool Event::Wait(int milliseconds) {
   pthread_mutex_lock(&event_mutex_);
   if (milliseconds != kForever) {
     while (!event_status_ && error == 0) {
-#if HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE
+#ifdef HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE
       error = pthread_cond_timedwait_relative_np(
           &event_cond_, &event_mutex_, &ts);
 #else

webrtc/base/event_tracer_unittest.cc

@@ -10,7 +10,7 @@
 
 #include "webrtc/base/event_tracer.h"
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/system_wrappers/include/static_instance.h"
 

webrtc/base/gunit.h

@@ -15,7 +15,7 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread.h"
 #if defined(GTEST_RELATIVE_PATH)
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #else
 #include "testing/base/public/gunit.h"
 #endif

webrtc/base/httpclient.h

@@ -51,7 +51,7 @@ class SignalThread;
 // What to do:  Define STRICT_HTTP_ERROR=1 in your makefile.  Use HttpError in
 // your code (HttpErrorType should only be used for code that is shared
 // with groups which have not yet migrated).
-#if STRICT_HTTP_ERROR
+#if defined(STRICT_HTTP_ERROR) && STRICT_HTTP_ERROR
 typedef HttpError HttpErrorType;
 #else  // !STRICT_HTTP_ERROR
 typedef int HttpErrorType;

webrtc/base/httpcommon.cc

@@ -388,7 +388,7 @@ bool HttpDateToSeconds(const std::string& date, time_t* seconds) {
   tm *tm_for_timezone = localtime(&gmt);
   *seconds = gmt + tm_for_timezone->tm_gmtoff;
 #else
-#if _MSC_VER >= 1900
+#if defined(_MSC_VER) && _MSC_VER >= 1900
   long timezone = 0;
   _get_timezone(&timezone);
 #endif

webrtc/base/ignore_wundef.h

@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2016 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_BASE_IGNORE_WUNDEF_H_
+#define WEBRTC_BASE_IGNORE_WUNDEF_H_
+
+// If a header file uses #if on possibly undefined macros (and it's for some
+// reason not possible to just fix the header file), include it like this:
+//
+//   RTC_PUSH_IGNORING_WUNDEF()
+//   #include "misbehaving_header.h"
+//   RTC_POP_IGNORING_WUNDEF()
+//
+// This will cause the compiler to not emit -Wundef warnings for that file.
+
+#ifdef __clang__
+#define RTC_PUSH_IGNORING_WUNDEF() \
+  _Pragma("clang diagnostic push") \
+      _Pragma("clang diagnostic ignored \"-Wundef\"")
+#define RTC_POP_IGNORING_WUNDEF() _Pragma("clang diagnostic pop")
+#else
+#define RTC_PUSH_IGNORING_WUNDEF()
+#define RTC_POP_IGNORING_WUNDEF()
+#endif  // __clang__
+
+#endif  // WEBRTC_BASE_IGNORE_WUNDEF_H_

webrtc/base/logging.cc

@@ -150,7 +150,7 @@ LogMessage::LogMessage(const char* file,
       case ERRCTX_ERRNO:
         tmp << " " << strerror(err);
         break;
-#if WEBRTC_WIN
+#ifdef WEBRTC_WIN
       case ERRCTX_HRESULT: {
         char msgbuf[256];
         DWORD flags = FORMAT_MESSAGE_FROM_SYSTEM;

webrtc/base/logging.h

@@ -335,7 +335,7 @@ inline bool LogCheckLevel(LoggingSeverity sev) {
   LOG_GLE(sev)
 #define LAST_SYSTEM_ERROR \
   (::GetLastError())
-#elif __native_client__
+#elif defined(__native_client__) && __native_client__
 #define LOG_ERR_EX(sev, err) \
   LOG(sev)
 #define LOG_ERR(sev) \

webrtc/base/mod_ops_unittest.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/mod_ops.h"
 
 namespace webrtc {

webrtc/base/pathutils.cc

@@ -33,7 +33,7 @@ const char EXT_DELIM = '.';
 const char* const FOLDER_DELIMS = "/\\";
 
 // DEFAULT_FOLDER_DELIM is the preferred delimiter for this platform
-#if WEBRTC_WIN
+#ifdef WEBRTC_WIN
 const char DEFAULT_FOLDER_DELIM = '\\';
 #else  // !WEBRTC_WIN
 const char DEFAULT_FOLDER_DELIM = '/';

webrtc/base/platform_thread_unittest.cc

@@ -10,7 +10,7 @@
 
 #include "webrtc/base/platform_thread.h"
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 
 namespace rtc {

webrtc/base/proxydetect.cc

@@ -38,11 +38,15 @@
 #include "webrtc/base/pathutils.h"
 #include "webrtc/base/stringutils.h"
 
-#if defined(WEBRTC_WIN)
-#define _TRY_WINHTTP 1
 #define _TRY_JSPROXY 0
 #define _TRY_WM_FINDPROXY 0
+
+#if defined(WEBRTC_WIN)
+#define _TRY_WINHTTP 1
 #define _TRY_IE_LAN_SETTINGS 1
+#else
+#define _TRY_WINHTTP 0
+#define _TRY_IE_LAN_SETTINGS 0
 #endif  // WEBRTC_WIN
 
 // For all platforms try Firefox.

webrtc/base/random_unittest.cc

@@ -13,7 +13,7 @@
 #include <limits>
 #include <vector>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/mathutils.h"  // unsigned difference
 #include "webrtc/base/random.h"
 

webrtc/base/rate_limiter_unittest.cc

@@ -11,7 +11,7 @@
 #include <algorithm>
 #include <memory>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/base/event.h"
 #include "webrtc/base/platform_thread.h"

webrtc/base/rate_statistics_unittest.cc

@@ -10,7 +10,7 @@
 
 #include <algorithm>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/rate_statistics.h"
 
 namespace {

webrtc/base/sequenced_task_checker_unittest.cc

@@ -7,7 +7,7 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/platform_thread.h"
@@ -197,7 +197,7 @@ TEST(SequencedTaskCheckerTest, DetachFromTaskQueueAndUseOnThread) {
   EXPECT_TRUE(done_event.Wait(1000));
 }
 
-#if !NDEBUG || DCHECK_ALWAYS_ON
+#if !defined(NDEBUG) || DCHECK_ALWAYS_ON
 TEST(SequencedTaskCheckerTest, MethodNotAllowedOnDifferentThreadInDebug) {
   RunMethodOnDifferentThread(false);
 }
@@ -207,7 +207,7 @@ TEST(SequencedTaskCheckerTest, MethodAllowedOnDifferentThreadInRelease) {
 }
 #endif
 
-#if !NDEBUG || DCHECK_ALWAYS_ON
+#if !defined(NDEBUG) || DCHECK_ALWAYS_ON
 TEST(SequencedTaskCheckerTest, MethodNotAllowedOnDifferentTaskQueueInDebug) {
   RunMethodOnDifferentTaskQueue(false);
 }
@@ -217,7 +217,7 @@ TEST(SequencedTaskCheckerTest, MethodAllowedOnDifferentTaskQueueInRelease) {
 }
 #endif
 
-#if !NDEBUG || DCHECK_ALWAYS_ON
+#if !defined(NDEBUG) || DCHECK_ALWAYS_ON
 TEST(SequencedTaskCheckerTest, DetachFromTaskQueueInDebug) {
   DetachThenCallFromDifferentTaskQueue(false);
 }

webrtc/base/swap_queue_unittest.cc

@@ -12,7 +12,7 @@
 
 #include <vector>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/base/thread_checker_unittest.cc

@@ -12,7 +12,7 @@
 
 #include <memory>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/task_queue.h"

webrtc/call/bitrate_allocator_unittest.cc

@@ -12,8 +12,8 @@
 #include <memory>
 #include <vector>
 
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/call/bitrate_allocator.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 

webrtc/call/bitrate_estimator_tests.cc

@@ -12,7 +12,7 @@
 #include <memory>
 #include <string>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/api/call/audio_state.h"
 #include "webrtc/base/checks.h"

webrtc/call/call_perf_tests.cc

@@ -13,7 +13,7 @@
 #include <memory>
 #include <string>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"

webrtc/call/call_unittest.cc

@@ -11,7 +11,7 @@
 #include <list>
 #include <memory>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/api/call/audio_state.h"
 #include "webrtc/call.h"

webrtc/call/mock/mock_rtc_event_log.h

@@ -13,7 +13,7 @@
 
 #include <string>
 
-#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/test/gmock.h"
 
 #include "webrtc/call/rtc_event_log.h"
 

webrtc/call/packet_injection_tests.cc

@@ -10,7 +10,7 @@
 
 #include <memory>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/null_transport.h"

webrtc/call/rampup_tests.cc

@@ -10,7 +10,7 @@
 
 #include "webrtc/call/rampup_tests.h"
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/test/testsupport/perf_test.h"

webrtc/call/ringbuffer_unittest.cc

@@ -11,7 +11,7 @@
 
 #include <list>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/random.h"
 #include "webrtc/call/ringbuffer.h"
 

webrtc/call/rtc_event_log_helper_thread.h

@@ -19,6 +19,7 @@
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/event.h"
+#include "webrtc/base/ignore_wundef.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/swap_queue.h"
 #include "webrtc/call/ringbuffer.h"
@@ -27,11 +28,13 @@
 
 #ifdef ENABLE_RTC_EVENT_LOG
 // Files generated at build-time by the protobuf compiler.
+RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
 #else
 #include "webrtc/call/rtc_event_log.pb.h"
 #endif
+RTC_POP_IGNORING_WUNDEF()
 #endif
 
 #ifdef ENABLE_RTC_EVENT_LOG

webrtc/call/rtc_event_log_parser.h

@@ -13,16 +13,19 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/base/ignore_wundef.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 // Files generated at build-time by the protobuf compiler.
+RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
 #else
 #include "webrtc/call/rtc_event_log.pb.h"
 #endif
+RTC_POP_IGNORING_WUNDEF()
 
 namespace webrtc {
 

webrtc/call/rtc_event_log_unittest.cc

@@ -14,7 +14,7 @@
 #include <utility>
 #include <vector>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/random.h"

webrtc/call/rtc_event_log_unittest_helper.cc

@@ -14,7 +14,7 @@
 
 #include <string>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/test/test_suite.h"
 #include "webrtc/test/testsupport/fileutils.h"

webrtc/common_audio/audio_converter_unittest.cc

@@ -13,7 +13,7 @@
 #include <memory>
 #include <vector>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/common_audio/audio_converter.h"

webrtc/common_audio/audio_ring_buffer_unittest.cc

@@ -12,7 +12,7 @@
 
 #include "webrtc/common_audio/audio_ring_buffer.h"
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/channel_buffer.h"
 
 namespace webrtc {

webrtc/common_audio/audio_util_unittest.cc

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/typedefs.h"
 

webrtc/common_audio/blocker_unittest.cc

@@ -12,7 +12,7 @@
 
 #include "webrtc/common_audio/blocker.h"
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/arraysize.h"
 
 namespace {

webrtc/common_audio/channel_buffer_unittest.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/channel_buffer.h"
 
 namespace webrtc {

webrtc/common_audio/fir_filter_unittest.cc

@@ -14,7 +14,7 @@
 
 #include <memory>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 namespace {

webrtc/common_audio/lapped_transform_unittest.cc

@@ -14,7 +14,7 @@
 #include <cmath>
 #include <cstring>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 using std::complex;
 

webrtc/common_audio/real_fourier_unittest.cc

@@ -12,7 +12,7 @@
 
 #include <stdlib.h>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/real_fourier_openmax.h"
 #include "webrtc/common_audio/real_fourier_ooura.h"
 

webrtc/common_audio/resampler/push_resampler_unittest.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"  // force defintion of RTC_DCHECK_IS_ON
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 

webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc

@@ -12,8 +12,8 @@
 #include <cstring>
 #include <memory>
 
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"

webrtc/common_audio/resampler/resampler_unittest.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/common_audio/resampler/include/resampler.h"
 

webrtc/common_audio/resampler/sinc_resampler_unittest.cc

@@ -18,8 +18,8 @@
 
 #include <memory>
 
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common_audio/resampler/sinc_resampler.h"
 #include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"

webrtc/common_audio/ring_buffer_unittest.cc

@@ -16,7 +16,7 @@
 #include <algorithm>
 #include <memory>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/common_audio/signal_processing/real_fft_unittest.cc

@@ -12,7 +12,7 @@
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/typedefs.h"
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 namespace {

webrtc/common_audio/signal_processing/signal_processing_unittest.cc

@@ -11,7 +11,7 @@
 #include <algorithm>
 #include <sstream>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 static const size_t kVector16Size = 9;

webrtc/common_audio/sparse_fir_filter_unittest.cc

@@ -12,7 +12,7 @@
 
 #include "webrtc/common_audio/sparse_fir_filter.h"
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/common_audio/fir_filter.h"
 

webrtc/common_audio/vad/mock/mock_vad.h

@@ -13,7 +13,7 @@
 
 #include "webrtc/common_audio/vad/include/vad.h"
 
-#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/test/gmock.h"
 
 namespace webrtc {
 

webrtc/common_audio/vad/vad_core_unittest.cc

@@ -10,7 +10,7 @@
 
 #include <stdlib.h>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/vad/vad_unittest.h"
 #include "webrtc/typedefs.h"
 

webrtc/common_audio/vad/vad_filterbank_unittest.cc

@@ -10,7 +10,7 @@
 
 #include <stdlib.h>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/vad/vad_unittest.h"
 #include "webrtc/typedefs.h"
 

webrtc/common_audio/vad/vad_gmm_unittest.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/vad/vad_unittest.h"
 #include "webrtc/typedefs.h"
 

webrtc/common_audio/vad/vad_sp_unittest.cc

@@ -10,7 +10,7 @@
 
 #include <stdlib.h>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/vad/vad_unittest.h"
 #include "webrtc/typedefs.h"
 

webrtc/common_audio/vad/vad_unittest.cc

@@ -12,7 +12,7 @@
 
 #include <stdlib.h>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"

webrtc/common_audio/vad/vad_unittest.h

@@ -13,7 +13,7 @@
 
 #include <stddef.h>  // size_t
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/typedefs.h"
 

webrtc/common_audio/wav_file_unittest.cc

@@ -14,7 +14,7 @@
 #include <cmath>
 #include <limits>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/wav_header.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/test/testsupport/fileutils.h"

webrtc/common_audio/wav_header_unittest.cc

@@ -10,7 +10,7 @@
 
 #include <limits>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/common_audio/wav_header.h"
 
 namespace webrtc {

webrtc/common_audio/window_generator_unittest.cc

@@ -12,7 +12,7 @@
 
 #include <cstring>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {
 

webrtc/common_types.h

@@ -27,9 +27,9 @@
 #pragma warning(disable : 4351)
 #endif
 
-#ifdef WEBRTC_EXPORT
+#if defined(WEBRTC_EXPORT)
 #define WEBRTC_DLLEXPORT _declspec(dllexport)
-#elif WEBRTC_DLL
+#elif defined(WEBRTC_DLL)
 #define WEBRTC_DLLEXPORT _declspec(dllimport)
 #else
 #define WEBRTC_DLLEXPORT

webrtc/common_video/bitrate_adjuster_unittest.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/common_video/include/bitrate_adjuster.h"
 #include "webrtc/system_wrappers/include/clock.h"

webrtc/common_video/h264/pps_parser_unittest.cc

@@ -13,7 +13,7 @@
 #include <limits>
 #include <memory>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/base/bitbuffer.h"
 #include "webrtc/base/buffer.h"

webrtc/common_video/h264/sps_parser_unittest.cc

@@ -10,7 +10,7 @@
 
 #include "webrtc/common_video/h264/sps_parser.h"
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/bitbuffer.h"

webrtc/common_video/h264/sps_vui_rewriter_unittest.cc

@@ -10,7 +10,7 @@
 
 #include <vector>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/base/bitbuffer.h"
 #include "webrtc/base/buffer.h"

webrtc/common_video/i420_buffer_pool_unittest.cc

@@ -10,7 +10,7 @@
 
 #include <string>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/common_video/include/i420_buffer_pool.h"
 
 namespace webrtc {

webrtc/common_video/i420_video_frame_unittest.cc

@@ -11,7 +11,7 @@
 #include <math.h>
 #include <string.h>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/test/fake_texture_frame.h"
 #include "webrtc/test/frame_utils.h"

webrtc/common_video/libyuv/libyuv_unittest.cc

@@ -13,7 +13,7 @@
 
 #include <memory>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/video_frame.h"

webrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m

@@ -569,7 +569,7 @@ static __strong NSData *CRLFCRLF;
             [SSLOptions setValue:[NSNumber numberWithBool:NO] forKey:(__bridge id)kCFStreamSSLValidatesCertificateChain];
         }
         
-#if DEBUG
+#ifdef DEBUG
         [SSLOptions setValue:[NSNumber numberWithBool:NO] forKey:(__bridge id)kCFStreamSSLValidatesCertificateChain];
         NSLog(@"SocketRocket: In debug mode.  Allowing connection to any root cert");
 #endif

webrtc/media/base/test/mock_mediachannel.h

@@ -12,7 +12,7 @@
 #define WEBRTC_MEDIA_BASE_TEST_MOCK_MEDIACHANNEL_H_
 
 #include "webrtc/media/base/fakemediaengine.h"
-#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/test/gmock.h"
 
 namespace webrtc {
 

webrtc/media/engine/nullwebrtcvideoengine_unittest.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/media/engine/nullwebrtcvideoengine.h"
 #include "webrtc/media/engine/webrtcvoiceengine.h"

webrtc/media/engine/payload_type_mapper_unittest.cc

@@ -11,7 +11,7 @@
 #include <set>
 #include <string>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/media/engine/payload_type_mapper.h"
 
 namespace cricket {

webrtc/media/engine/simulcast_unittest.cc

@@ -10,7 +10,7 @@
 
 #include <string>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/media/engine/simulcast.h"
 
 namespace cricket {

webrtc/media/engine/webrtcmediaengine_unittest.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"

webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc

@@ -15,7 +15,7 @@
 
 #include <memory>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"

webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc

@@ -13,7 +13,7 @@
 #include <algorithm>  // std::min
 #include <memory>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"

webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc

@@ -14,7 +14,7 @@
 #include <stdio.h>
 #include <string.h>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc

@@ -13,7 +13,7 @@
 #include <memory>
 #include <vector>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/md5digest.h"
 #include "webrtc/base/platform_thread.h"

webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc

@@ -10,7 +10,7 @@
 
 #include <memory>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"

webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc

@@ -10,7 +10,7 @@
 
 #include <memory>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"

webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc

@@ -11,7 +11,7 @@
 #include <utility>
 #include <vector>
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h"

webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc

@@ -11,7 +11,7 @@
 #include <memory>
 
 #include "webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h"
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/gtest.h"
 
 namespace webrtc {