Mutex: remove Abseil static initializer.

The change adds conditional inclusion of mutex_abseil.h from mutex.h
and conditional referencing of
//third_party/abseil-cpp/absl/synchronization
which introduces a static initializer.

https://webrtc-review.googlesource.com/c/src/+/176230 introduced a
static initializer which broke the Chromium autoroll,
https://chromium-review.googlesource.com/c/chromium/src/+/2230887.
Example failure:
https://ci.chromium.org/p/chromium/builders/try/android-lollipop-arm-rel/34133

TBR=karl@webrtc.org

No-Try: True
Bug: webrtc:11567
Change-Id: Id78af798f34d5d6beaf9f6e0150e6b3ddd31ff4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176513
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31451}

BUILD.gn

@@ -265,6 +265,10 @@ config("common_config") {
     defines += [ "WEBRTC_USE_H264" ]
   }
 
+  if (rtc_use_absl_mutex) {
+    defines += [ "WEBRTC_ABSL_MUTEX" ]
+  }
+
   if (rtc_disable_logging) {
     defines += [ "RTC_DISABLE_LOGGING" ]
   }

rtc_base/synchronization/BUILD.gn

@@ -25,18 +25,23 @@ rtc_library("mutex") {
   sources = [
     "mutex.cc",
     "mutex.h",
-    "mutex_abseil.h",
     "mutex_critical_section.h",
     "mutex_pthread.h",
   ]
+  if (rtc_use_absl_mutex) {
+    sources += [ "mutex_abseil.h" ]
+  }
+
   deps = [
     ":yield",
     "..:checks",
     "..:macromagic",
     "../system:unused",
     "//third_party/abseil-cpp/absl/base:core_headers",
-    "//third_party/abseil-cpp/absl/synchronization",
   ]
+  if (rtc_use_absl_mutex) {
+    deps += [ "//third_party/abseil-cpp/absl/synchronization" ]
+  }
 }
 
 rtc_library("rw_lock_wrapper") {

rtc_base/synchronization/mutex.h

@@ -19,7 +19,7 @@
 #include "rtc_base/thread_annotations.h"
 
 #if defined(WEBRTC_ABSL_MUTEX)
-#include "rtc_base/synchronization/mutex_abseil.h"
+#include "rtc_base/synchronization/mutex_abseil.h"  // nogncheck
 #elif defined(WEBRTC_WIN)
 #include "rtc_base/synchronization/mutex_critical_section.h"
 #elif defined(WEBRTC_POSIX)

webrtc.gni

@@ -155,6 +155,9 @@ declare_args() {
   rtc_use_h264 =
       proprietary_codecs && !is_android && !is_ios && !(is_win && !is_clang)
 
+  # Enable this flag to make webrtc::Mutex be implemented by absl::Mutex.
+  rtc_use_absl_mutex = false
+
   # By default, use normal platform audio support or dummy audio, but don't
   # use file-based audio playout and record.
   rtc_use_dummy_audio_file_devices = false