diff --git a/BUILD.gn b/BUILD.gn
index 698e4cff70..adb78df424 100644
--- a/BUILD.gn
+++ b/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" ]
   }
diff --git a/rtc_base/synchronization/BUILD.gn b/rtc_base/synchronization/BUILD.gn
index b1942e7815..84ed31ff5a 100644
--- a/rtc_base/synchronization/BUILD.gn
+++ b/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") {
diff --git a/rtc_base/synchronization/mutex.h b/rtc_base/synchronization/mutex.h
index bfcadc2139..7980dae9d7 100644
--- a/rtc_base/synchronization/mutex.h
+++ b/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)
diff --git a/webrtc.gni b/webrtc.gni
index d222bc7ec9..bf4d0a00b3 100644
--- a/webrtc.gni
+++ b/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