diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index 2e3d77a03a..86a63e4dee 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -207,6 +207,17 @@ rtc_source_set("array_view") {
   ]
 }
 
+rtc_source_set("optional") {
+  sources = [
+    "optional.cc",
+    "optional.h",
+  ]
+  deps = [
+    ":array_view",
+    "../rtc_base:rtc_base_approved",
+  ]
+}
+
 rtc_source_set("libjingle_peerconnection_test_api") {
   testonly = true
   sources = [
@@ -266,6 +277,7 @@ if (rtc_include_tests) {
     }
     sources = [
       "array_view_unittest.cc",
+      "optional_unittest.cc",
       "ortc/mediadescription_unittest.cc",
       "ortc/sessiondescription_unittest.cc",
       "rtcerror_unittest.cc",
@@ -280,6 +292,7 @@ if (rtc_include_tests) {
     deps = [
       ":array_view",
       ":libjingle_peerconnection_api",
+      ":optional",
       ":ortc_api",
       "../rtc_base:rtc_base_approved",
       "../rtc_base:rtc_base_tests_utils",
diff --git a/webrtc/api/audio_codecs/BUILD.gn b/webrtc/api/audio_codecs/BUILD.gn
index 29238e65fe..04a5238f84 100644
--- a/webrtc/api/audio_codecs/BUILD.gn
+++ b/webrtc/api/audio_codecs/BUILD.gn
@@ -27,6 +27,7 @@ rtc_source_set("audio_codecs_api") {
   ]
   deps = [
     "..:array_view",
+    "..:optional",
     "../..:webrtc_common",
     "../../rtc_base:rtc_base_approved",
   ]
diff --git a/webrtc/api/audio_codecs/L16/BUILD.gn b/webrtc/api/audio_codecs/L16/BUILD.gn
index db61713c5d..8f06a8f332 100644
--- a/webrtc/api/audio_codecs/L16/BUILD.gn
+++ b/webrtc/api/audio_codecs/L16/BUILD.gn
@@ -19,6 +19,7 @@ rtc_static_library("audio_encoder_L16") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../..:optional",
     "../../..:webrtc_common",
     "../../../modules/audio_coding:pcm16b",
     "../../../rtc_base:rtc_base_approved",
@@ -32,6 +33,7 @@ rtc_static_library("audio_decoder_L16") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../..:optional",
     "../../..:webrtc_common",
     "../../../modules/audio_coding:pcm16b",
     "../../../rtc_base:rtc_base_approved",
diff --git a/webrtc/api/audio_codecs/L16/audio_decoder_L16.h b/webrtc/api/audio_codecs/L16/audio_decoder_L16.h
index 478da16abe..39b389a0ec 100644
--- a/webrtc/api/audio_codecs/L16/audio_decoder_L16.h
+++ b/webrtc/api/audio_codecs/L16/audio_decoder_L16.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/audio_codecs/L16/audio_encoder_L16.h b/webrtc/api/audio_codecs/L16/audio_encoder_L16.h
index e2e4019f21..3ac0423af9 100644
--- a/webrtc/api/audio_codecs/L16/audio_encoder_L16.h
+++ b/webrtc/api/audio_codecs/L16/audio_encoder_L16.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/audio_codecs/audio_decoder.h b/webrtc/api/audio_codecs/audio_decoder.h
index 0d15e6d29f..2d850fd461 100644
--- a/webrtc/api/audio_codecs/audio_decoder.h
+++ b/webrtc/api/audio_codecs/audio_decoder.h
@@ -15,9 +15,9 @@
 #include <vector>
 
 #include "webrtc/api/array_view.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/rtc_base/buffer.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/api/audio_codecs/audio_encoder.h b/webrtc/api/audio_codecs/audio_encoder.h
index 37a03d0317..7406d7d653 100644
--- a/webrtc/api/audio_codecs/audio_encoder.h
+++ b/webrtc/api/audio_codecs/audio_encoder.h
@@ -17,9 +17,9 @@
 #include <vector>
 
 #include "webrtc/api/array_view.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/rtc_base/buffer.h"
 #include "webrtc/rtc_base/deprecation.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/api/audio_codecs/audio_format.h b/webrtc/api/audio_codecs/audio_format.h
index c442972788..ab825bb960 100644
--- a/webrtc/api/audio_codecs/audio_format.h
+++ b/webrtc/api/audio_codecs/audio_format.h
@@ -16,7 +16,7 @@
 #include <string>
 #include <utility>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/audio_codecs/g711/BUILD.gn b/webrtc/api/audio_codecs/g711/BUILD.gn
index fbf9248683..aa86490a73 100644
--- a/webrtc/api/audio_codecs/g711/BUILD.gn
+++ b/webrtc/api/audio_codecs/g711/BUILD.gn
@@ -19,6 +19,7 @@ rtc_static_library("audio_encoder_g711") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../..:optional",
     "../../..:webrtc_common",
     "../../../modules/audio_coding:g711",
     "../../../rtc_base:rtc_base_approved",
@@ -32,6 +33,7 @@ rtc_static_library("audio_decoder_g711") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../..:optional",
     "../../..:webrtc_common",
     "../../../modules/audio_coding:g711",
     "../../../rtc_base:rtc_base_approved",
diff --git a/webrtc/api/audio_codecs/g711/audio_decoder_g711.h b/webrtc/api/audio_codecs/g711/audio_decoder_g711.h
index b9650c95a6..2824c32fd1 100644
--- a/webrtc/api/audio_codecs/g711/audio_decoder_g711.h
+++ b/webrtc/api/audio_codecs/g711/audio_decoder_g711.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/audio_codecs/g711/audio_encoder_g711.h b/webrtc/api/audio_codecs/g711/audio_encoder_g711.h
index a22650dd44..57501002c1 100644
--- a/webrtc/api/audio_codecs/g711/audio_encoder_g711.h
+++ b/webrtc/api/audio_codecs/g711/audio_encoder_g711.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/audio_codecs/g722/BUILD.gn b/webrtc/api/audio_codecs/g722/BUILD.gn
index f3108e7b21..5af7e5c223 100644
--- a/webrtc/api/audio_codecs/g722/BUILD.gn
+++ b/webrtc/api/audio_codecs/g722/BUILD.gn
@@ -26,6 +26,7 @@ rtc_static_library("audio_encoder_g722") {
   deps = [
     ":audio_encoder_g722_config",
     "..:audio_codecs_api",
+    "../..:optional",
     "../../..:webrtc_common",
     "../../../modules/audio_coding:g722",
     "../../../rtc_base:rtc_base_approved",
@@ -39,6 +40,7 @@ rtc_static_library("audio_decoder_g722") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../..:optional",
     "../../..:webrtc_common",
     "../../../modules/audio_coding:g722",
     "../../../rtc_base:rtc_base_approved",
diff --git a/webrtc/api/audio_codecs/g722/audio_decoder_g722.h b/webrtc/api/audio_codecs/g722/audio_decoder_g722.h
index 9b063c2bf0..cf0dfa9602 100644
--- a/webrtc/api/audio_codecs/g722/audio_decoder_g722.h
+++ b/webrtc/api/audio_codecs/g722/audio_decoder_g722.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/audio_codecs/g722/audio_encoder_g722.h b/webrtc/api/audio_codecs/g722/audio_encoder_g722.h
index 458a6c5a79..37cb968ee3 100644
--- a/webrtc/api/audio_codecs/g722/audio_encoder_g722.h
+++ b/webrtc/api/audio_codecs/g722/audio_encoder_g722.h
@@ -17,7 +17,7 @@
 #include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/audio_codecs/g722/audio_encoder_g722_config.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/audio_codecs/ilbc/BUILD.gn b/webrtc/api/audio_codecs/ilbc/BUILD.gn
index ab9681a3e5..0f5f80dfe2 100644
--- a/webrtc/api/audio_codecs/ilbc/BUILD.gn
+++ b/webrtc/api/audio_codecs/ilbc/BUILD.gn
@@ -26,6 +26,7 @@ rtc_static_library("audio_encoder_ilbc") {
   deps = [
     ":audio_encoder_ilbc_config",
     "..:audio_codecs_api",
+    "../..:optional",
     "../../..:webrtc_common",
     "../../../modules/audio_coding:ilbc",
     "../../../rtc_base:rtc_base_approved",
@@ -39,6 +40,7 @@ rtc_static_library("audio_decoder_ilbc") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../..:optional",
     "../../..:webrtc_common",
     "../../../modules/audio_coding:ilbc",
     "../../../rtc_base:rtc_base_approved",
diff --git a/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h b/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h
index 65a67e2311..343308df9a 100644
--- a/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h
+++ b/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h b/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h
index 6f7a9f6c62..a30d1092e8 100644
--- a/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h
+++ b/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h
@@ -17,7 +17,7 @@
 #include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/audio_codecs/isac/BUILD.gn b/webrtc/api/audio_codecs/isac/BUILD.gn
index 3340e28a06..b22eb95525 100644
--- a/webrtc/api/audio_codecs/isac/BUILD.gn
+++ b/webrtc/api/audio_codecs/isac/BUILD.gn
@@ -19,6 +19,7 @@ rtc_static_library("audio_encoder_isac_fix") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../..:optional",
     "../../..:webrtc_common",
     "../../../modules/audio_coding:isac_fix",
     "../../../rtc_base:rtc_base_approved",
@@ -32,6 +33,7 @@ rtc_static_library("audio_decoder_isac_fix") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../..:optional",
     "../../..:webrtc_common",
     "../../../modules/audio_coding:isac_fix",
     "../../../rtc_base:rtc_base_approved",
@@ -45,6 +47,7 @@ rtc_static_library("audio_encoder_isac_float") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../..:optional",
     "../../..:webrtc_common",
     "../../../modules/audio_coding:isac",
     "../../../rtc_base:rtc_base_approved",
@@ -58,6 +61,7 @@ rtc_static_library("audio_decoder_isac_float") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../..:optional",
     "../../..:webrtc_common",
     "../../../modules/audio_coding:isac",
     "../../../rtc_base:rtc_base_approved",
diff --git a/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h b/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h
index 71e7cc41f2..7662002a84 100644
--- a/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h
+++ b/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h b/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h
index c0dc880dae..f0232e9209 100644
--- a/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h
+++ b/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h b/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h
index 5f0e8fe4b5..ce4a1d01b3 100644
--- a/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h
+++ b/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h b/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h
index 35bc94b464..244e59b11d 100644
--- a/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h
+++ b/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/audio_codecs/opus/BUILD.gn b/webrtc/api/audio_codecs/opus/BUILD.gn
index 658d151a1d..faa5956dc5 100644
--- a/webrtc/api/audio_codecs/opus/BUILD.gn
+++ b/webrtc/api/audio_codecs/opus/BUILD.gn
@@ -18,6 +18,7 @@ rtc_static_library("audio_encoder_opus_config") {
     "audio_encoder_opus_config.h",
   ]
   deps = [
+    "../..:optional",
     "../../../rtc_base:rtc_base_approved",
   ]
   defines = []
@@ -51,6 +52,7 @@ rtc_static_library("audio_decoder_opus") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../..:optional",
     "../../..:webrtc_common",
     "../../../modules/audio_coding:webrtc_opus",
     "../../../rtc_base:rtc_base_approved",
diff --git a/webrtc/api/audio_codecs/opus/audio_decoder_opus.h b/webrtc/api/audio_codecs/opus/audio_decoder_opus.h
index 6a9d6da57a..272503da94 100644
--- a/webrtc/api/audio_codecs/opus/audio_decoder_opus.h
+++ b/webrtc/api/audio_codecs/opus/audio_decoder_opus.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h b/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
index c3fe17b39c..d07f7848e7 100644
--- a/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
+++ b/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
@@ -15,7 +15,7 @@
 
 #include <vector>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/mediaconstraintsinterface.h b/webrtc/api/mediaconstraintsinterface.h
index 72fdd97866..eb89b3ee52 100644
--- a/webrtc/api/mediaconstraintsinterface.h
+++ b/webrtc/api/mediaconstraintsinterface.h
@@ -23,8 +23,8 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/peerconnectioninterface.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/mediastreaminterface.h b/webrtc/api/mediastreaminterface.h
index aaf5b62aa4..5d6544716d 100644
--- a/webrtc/api/mediastreaminterface.h
+++ b/webrtc/api/mediastreaminterface.h
@@ -22,8 +22,8 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/video/video_frame.h"
-#include "webrtc/rtc_base/optional.h"
 // TODO(zhihuang): Remove unrelated headers once downstream applications stop
 // relying on them; they were previously transitively included by
 // mediachannel.h, which is no longer a dependency of this file.
diff --git a/webrtc/rtc_base/optional.cc b/webrtc/api/optional.cc
similarity index 85%
rename from webrtc/rtc_base/optional.cc
rename to webrtc/api/optional.cc
index 4b41a2d7fb..ec8696661f 100644
--- a/webrtc/rtc_base/optional.cc
+++ b/webrtc/api/optional.cc
@@ -8,14 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace rtc {
 namespace optional_internal {
 
 #if RTC_HAS_ASAN
 
-void* FunctionThatDoesNothingImpl(void* x) { return x; }
+void* FunctionThatDoesNothingImpl(void* x) {
+  return x;
+}
 
 #endif
 
diff --git a/webrtc/api/optional.h b/webrtc/api/optional.h
new file mode 100644
index 0000000000..de1ec764bf
--- /dev/null
+++ b/webrtc/api/optional.h
@@ -0,0 +1,407 @@
+/*
+ *  Copyright 2015 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_API_OPTIONAL_H_
+#define WEBRTC_API_OPTIONAL_H_
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+#ifdef UNIT_TEST
+#include <iomanip>
+#include <ostream>
+#endif  // UNIT_TEST
+
+#include "webrtc/api/array_view.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/sanitizer.h"
+
+namespace rtc {
+
+namespace optional_internal {
+
+#if RTC_HAS_ASAN
+
+// This is a non-inlined function. The optimizer can't see inside it.  It
+// prevents the compiler from generating optimized code that reads value_ even
+// if it is unset. Although safe, this causes memory sanitizers to complain.
+void* FunctionThatDoesNothingImpl(void*);
+
+template <typename T>
+inline T* FunctionThatDoesNothing(T* x) {
+  return reinterpret_cast<T*>(
+      FunctionThatDoesNothingImpl(reinterpret_cast<void*>(x)));
+}
+
+#else
+
+template <typename T>
+inline T* FunctionThatDoesNothing(T* x) {
+  return x;
+}
+
+#endif
+
+}  // namespace optional_internal
+
+// Simple std::optional-wannabe. It either contains a T or not.
+//
+// A moved-from Optional<T> may only be destroyed, and assigned to if T allows
+// being assigned to after having been moved from. Specifically, you may not
+// assume that it just doesn't contain a value anymore.
+//
+// Examples of good places to use Optional:
+//
+// - As a class or struct member, when the member doesn't always have a value:
+//     struct Prisoner {
+//       std::string name;
+//       Optional<int> cell_number;  // Empty if not currently incarcerated.
+//     };
+//
+// - As a return value for functions that may fail to return a value on all
+//   allowed inputs. For example, a function that searches an array might
+//   return an Optional<size_t> (the index where it found the element, or
+//   nothing if it didn't find it); and a function that parses numbers might
+//   return Optional<double> (the parsed number, or nothing if parsing failed).
+//
+// Examples of bad places to use Optional:
+//
+// - As a return value for functions that may fail because of disallowed
+//   inputs. For example, a string length function should not return
+//   Optional<size_t> so that it can return nothing in case the caller passed
+//   it a null pointer; the function should probably use RTC_[D]CHECK instead,
+//   and return plain size_t.
+//
+// - As a return value for functions that may fail to return a value on all
+//   allowed inputs, but need to tell the caller what went wrong. Returning
+//   Optional<double> when parsing a single number as in the example above
+//   might make sense, but any larger parse job is probably going to need to
+//   tell the caller what the problem was, not just that there was one.
+//
+// - As a non-mutable function argument. When you want to pass a value of a
+//   type T that can fail to be there, const T* is almost always both fastest
+//   and cleanest. (If you're *sure* that the the caller will always already
+//   have an Optional<T>, const Optional<T>& is slightly faster than const T*,
+//   but this is a micro-optimization. In general, stick to const T*.)
+//
+// TODO(kwiberg): Get rid of this class when the standard library has
+// std::optional (and we're allowed to use it).
+template <typename T>
+class Optional final {
+ public:
+  // Construct an empty Optional.
+  Optional() : has_value_(false), empty_('\0') { PoisonValue(); }
+
+  // Construct an Optional that contains a value.
+  explicit Optional(const T& value) : has_value_(true) {
+    new (&value_) T(value);
+  }
+  explicit Optional(T&& value) : has_value_(true) {
+    new (&value_) T(std::move(value));
+  }
+
+  // Copy constructor: copies the value from m if it has one.
+  Optional(const Optional& m) : has_value_(m.has_value_) {
+    if (has_value_)
+      new (&value_) T(m.value_);
+    else
+      PoisonValue();
+  }
+
+  // Move constructor: if m has a value, moves the value from m, leaving m
+  // still in a state where it has a value, but a moved-from one (the
+  // properties of which depends on T; the only general guarantee is that we
+  // can destroy m).
+  Optional(Optional&& m) : has_value_(m.has_value_) {
+    if (has_value_)
+      new (&value_) T(std::move(m.value_));
+    else
+      PoisonValue();
+  }
+
+  ~Optional() {
+    if (has_value_)
+      value_.~T();
+    else
+      UnpoisonValue();
+  }
+
+  // Copy assignment. Uses T's copy assignment if both sides have a value, T's
+  // copy constructor if only the right-hand side has a value.
+  Optional& operator=(const Optional& m) {
+    if (m.has_value_) {
+      if (has_value_) {
+        value_ = m.value_;  // T's copy assignment.
+      } else {
+        UnpoisonValue();
+        new (&value_) T(m.value_);  // T's copy constructor.
+        has_value_ = true;
+      }
+    } else {
+      reset();
+    }
+    return *this;
+  }
+
+  // Move assignment. Uses T's move assignment if both sides have a value, T's
+  // move constructor if only the right-hand side has a value. The state of m
+  // after it's been moved from is as for the move constructor.
+  Optional& operator=(Optional&& m) {
+    if (m.has_value_) {
+      if (has_value_) {
+        value_ = std::move(m.value_);  // T's move assignment.
+      } else {
+        UnpoisonValue();
+        new (&value_) T(std::move(m.value_));  // T's move constructor.
+        has_value_ = true;
+      }
+    } else {
+      reset();
+    }
+    return *this;
+  }
+
+  // Swap the values if both m1 and m2 have values; move the value if only one
+  // of them has one.
+  friend void swap(Optional& m1, Optional& m2) {
+    if (m1.has_value_) {
+      if (m2.has_value_) {
+        // Both have values: swap.
+        using std::swap;
+        swap(m1.value_, m2.value_);
+      } else {
+        // Only m1 has a value: move it to m2.
+        m2.UnpoisonValue();
+        new (&m2.value_) T(std::move(m1.value_));
+        m1.value_.~T();  // Destroy the moved-from value.
+        m1.has_value_ = false;
+        m2.has_value_ = true;
+        m1.PoisonValue();
+      }
+    } else if (m2.has_value_) {
+      // Only m2 has a value: move it to m1.
+      m1.UnpoisonValue();
+      new (&m1.value_) T(std::move(m2.value_));
+      m2.value_.~T();  // Destroy the moved-from value.
+      m1.has_value_ = true;
+      m2.has_value_ = false;
+      m2.PoisonValue();
+    }
+  }
+
+  // Destroy any contained value. Has no effect if we have no value.
+  void reset() {
+    if (!has_value_)
+      return;
+    value_.~T();
+    has_value_ = false;
+    PoisonValue();
+  }
+
+  template <class... Args>
+  void emplace(Args&&... args) {
+    if (has_value_)
+      value_.~T();
+    else
+      UnpoisonValue();
+    new (&value_) T(std::forward<Args>(args)...);
+    has_value_ = true;
+  }
+
+  // Conversion to bool to test if we have a value.
+  explicit operator bool() const { return has_value_; }
+  bool has_value() const { return has_value_; }
+
+  // Dereferencing. Only allowed if we have a value.
+  const T* operator->() const {
+    RTC_DCHECK(has_value_);
+    return &value_;
+  }
+  T* operator->() {
+    RTC_DCHECK(has_value_);
+    return &value_;
+  }
+  const T& operator*() const {
+    RTC_DCHECK(has_value_);
+    return value_;
+  }
+  T& operator*() {
+    RTC_DCHECK(has_value_);
+    return value_;
+  }
+  const T& value() const {
+    RTC_DCHECK(has_value_);
+    return value_;
+  }
+  T& value() {
+    RTC_DCHECK(has_value_);
+    return value_;
+  }
+
+  // Dereference with a default value in case we don't have a value.
+  const T& value_or(const T& default_val) const {
+    // The no-op call prevents the compiler from generating optimized code that
+    // reads value_ even if !has_value_, but only if FunctionThatDoesNothing is
+    // not completely inlined; see its declaration.).
+    return has_value_ ? *optional_internal::FunctionThatDoesNothing(&value_)
+                      : default_val;
+  }
+
+  // Dereference and move value.
+  T MoveValue() {
+    RTC_DCHECK(has_value_);
+    return std::move(value_);
+  }
+
+  // Equality tests. Two Optionals are equal if they contain equivalent values,
+  // or if they're both empty.
+  friend bool operator==(const Optional& m1, const Optional& m2) {
+    return m1.has_value_ && m2.has_value_ ? m1.value_ == m2.value_
+                                          : m1.has_value_ == m2.has_value_;
+  }
+  friend bool operator==(const Optional& opt, const T& value) {
+    return opt.has_value_ && opt.value_ == value;
+  }
+  friend bool operator==(const T& value, const Optional& opt) {
+    return opt.has_value_ && value == opt.value_;
+  }
+
+  friend bool operator!=(const Optional& m1, const Optional& m2) {
+    return m1.has_value_ && m2.has_value_ ? m1.value_ != m2.value_
+                                          : m1.has_value_ != m2.has_value_;
+  }
+  friend bool operator!=(const Optional& opt, const T& value) {
+    return !opt.has_value_ || opt.value_ != value;
+  }
+  friend bool operator!=(const T& value, const Optional& opt) {
+    return !opt.has_value_ || value != opt.value_;
+  }
+
+ private:
+  // Tell sanitizers that value_ shouldn't be touched.
+  void PoisonValue() {
+    rtc::AsanPoison(rtc::MakeArrayView(&value_, 1));
+    rtc::MsanMarkUninitialized(rtc::MakeArrayView(&value_, 1));
+  }
+
+  // Tell sanitizers that value_ is OK to touch again.
+  void UnpoisonValue() { rtc::AsanUnpoison(rtc::MakeArrayView(&value_, 1)); }
+
+  bool has_value_;  // True iff value_ contains a live value.
+  union {
+    // empty_ exists only to make it possible to initialize the union, even when
+    // it doesn't contain any data. If the union goes uninitialized, it may
+    // trigger compiler warnings.
+    char empty_;
+    // By placing value_ in a union, we get to manage its construction and
+    // destruction manually: the Optional constructors won't automatically
+    // construct it, and the Optional destructor won't automatically destroy
+    // it. Basically, this just allocates a properly sized and aligned block of
+    // memory in which we can manually put a T with placement new.
+    T value_;
+  };
+};
+
+#ifdef UNIT_TEST
+namespace optional_internal {
+
+// Checks if there's a valid PrintTo(const T&, std::ostream*) call for T.
+template <typename T>
+struct HasPrintTo {
+ private:
+  struct No {};
+
+  template <typename T2>
+  static auto Test(const T2& obj)
+      -> decltype(PrintTo(obj, std::declval<std::ostream*>()));
+
+  template <typename>
+  static No Test(...);
+
+ public:
+  static constexpr bool value =
+      !std::is_same<decltype(Test<T>(std::declval<const T&>())), No>::value;
+};
+
+// Checks if there's a valid operator<<(std::ostream&, const T&) call for T.
+template <typename T>
+struct HasOstreamOperator {
+ private:
+  struct No {};
+
+  template <typename T2>
+  static auto Test(const T2& obj)
+      -> decltype(std::declval<std::ostream&>() << obj);
+
+  template <typename>
+  static No Test(...);
+
+ public:
+  static constexpr bool value =
+      !std::is_same<decltype(Test<T>(std::declval<const T&>())), No>::value;
+};
+
+// Prefer using PrintTo to print the object.
+template <typename T>
+typename std::enable_if<HasPrintTo<T>::value, void>::type OptionalPrintToHelper(
+    const T& value,
+    std::ostream* os) {
+  PrintTo(value, os);
+}
+
+// Fall back to operator<<(std::ostream&, ...) if it exists.
+template <typename T>
+typename std::enable_if<HasOstreamOperator<T>::value && !HasPrintTo<T>::value,
+                        void>::type
+OptionalPrintToHelper(const T& value, std::ostream* os) {
+  *os << value;
+}
+
+inline void OptionalPrintObjectBytes(const unsigned char* bytes,
+                                     size_t size,
+                                     std::ostream* os) {
+  *os << "<optional with " << size << "-byte object [";
+  for (size_t i = 0; i != size; ++i) {
+    *os << (i == 0 ? "" : ((i & 1) ? "-" : " "));
+    *os << std::hex << std::setw(2) << std::setfill('0')
+        << static_cast<int>(bytes[i]);
+  }
+  *os << "]>";
+}
+
+// As a final back-up, just print the contents of the objcets byte-wise.
+template <typename T>
+typename std::enable_if<!HasOstreamOperator<T>::value && !HasPrintTo<T>::value,
+                        void>::type
+OptionalPrintToHelper(const T& value, std::ostream* os) {
+  OptionalPrintObjectBytes(reinterpret_cast<const unsigned char*>(&value),
+                           sizeof(value), os);
+}
+
+}  // namespace optional_internal
+
+// PrintTo is used by gtest to print out the results of tests. We want to ensure
+// the object contained in an Optional can be printed out if it's set, while
+// avoiding touching the object's storage if it is undefined.
+template <typename T>
+void PrintTo(const rtc::Optional<T>& opt, std::ostream* os) {
+  if (opt) {
+    optional_internal::OptionalPrintToHelper(*opt, os);
+  } else {
+    *os << "<empty optional>";
+  }
+}
+
+#endif  // UNIT_TEST
+
+}  // namespace rtc
+
+#endif  // WEBRTC_API_OPTIONAL_H_
diff --git a/webrtc/rtc_base/optional_unittest.cc b/webrtc/api/optional_unittest.cc
similarity index 98%
rename from webrtc/rtc_base/optional_unittest.cc
rename to webrtc/api/optional_unittest.cc
index 8f2f7fd1f0..f8f9cf300a 100644
--- a/webrtc/rtc_base/optional_unittest.cc
+++ b/webrtc/api/optional_unittest.cc
@@ -14,8 +14,8 @@
 #include <utility>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/rtc_base/gunit.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace rtc {
 
@@ -37,14 +37,12 @@ void PrintTo(const MyPrintableType& mpt, std::ostream* os) {
   *os << "The value is " << mpt.value;
 }
 
-std::ostream& operator<<(std::ostream& os,
-                         const MyPrintableType& mpt) {
+std::ostream& operator<<(std::ostream& os, const MyPrintableType& mpt) {
   os << mpt.value;
   return os;
 }
 
-std::ostream& operator<<(std::ostream& os,
-                         const MyOstreamPrintableType& mpt) {
+std::ostream& operator<<(std::ostream& os, const MyOstreamPrintableType& mpt) {
   os << mpt.value;
   return os;
 }
diff --git a/webrtc/api/ortc/mediadescription.h b/webrtc/api/ortc/mediadescription.h
index efee6629a1..d638dfc433 100644
--- a/webrtc/api/ortc/mediadescription.h
+++ b/webrtc/api/ortc/mediadescription.h
@@ -15,8 +15,8 @@
 #include <utility>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/media/base/cryptoparams.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/ortc/rtptransportinterface.h b/webrtc/api/ortc/rtptransportinterface.h
index 165daada6b..1fdc12caab 100644
--- a/webrtc/api/ortc/rtptransportinterface.h
+++ b/webrtc/api/ortc/rtptransportinterface.h
@@ -13,10 +13,10 @@
 
 #include <string>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/ortc/packettransportinterface.h"
 #include "webrtc/api/rtcerror.h"
 #include "webrtc/common_types.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/rtpparameters.h b/webrtc/api/rtpparameters.h
index 46f71395ce..ce61b15f75 100644
--- a/webrtc/api/rtpparameters.h
+++ b/webrtc/api/rtpparameters.h
@@ -16,7 +16,7 @@
 #include <vector>
 
 #include "webrtc/api/mediatypes.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/video_codecs/BUILD.gn b/webrtc/api/video_codecs/BUILD.gn
index 7fad43d8f0..be7194e283 100644
--- a/webrtc/api/video_codecs/BUILD.gn
+++ b/webrtc/api/video_codecs/BUILD.gn
@@ -20,6 +20,7 @@ rtc_source_set("video_codecs_api") {
   ]
 
   deps = [
+    "..:optional",
     "..:video_frame_api",
     "../..:webrtc_common",
     "../../common_video",
diff --git a/webrtc/api/video_codecs/video_encoder.h b/webrtc/api/video_codecs/video_encoder.h
index 7a319d7216..4ec6e45f00 100644
--- a/webrtc/api/video_codecs/video_encoder.h
+++ b/webrtc/api/video_codecs/video_encoder.h
@@ -15,11 +15,11 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/video/video_frame.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/include/video_frame.h"
 #include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
index 2c62584ece..42b74ffcd6 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -37,6 +37,7 @@ rtc_static_library("audio") {
     "..:webrtc_common",
     "../api:audio_mixer_api",
     "../api:call_api",
+    "../api:optional",
     "../api/audio_codecs:audio_codecs_api",
     "../api/audio_codecs:builtin_audio_encoder_factory",
     "../call:call_interfaces",
diff --git a/webrtc/audio/time_interval.h b/webrtc/audio/time_interval.h
index 069127d641..448184359f 100644
--- a/webrtc/audio/time_interval.h
+++ b/webrtc/audio/time_interval.h
@@ -13,7 +13,7 @@
 
 #include <stdint.h>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 580d606f59..706d502b21 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -26,6 +26,7 @@ rtc_source_set("call_interfaces") {
     "..:webrtc_common",
     "../api:audio_mixer_api",
     "../api:libjingle_peerconnection_api",
+    "../api:optional",
     "../api:transport_api",
     "../api/audio_codecs:audio_codecs_api",
     "../rtc_base:rtc_base",
@@ -68,6 +69,7 @@ rtc_source_set("rtp_receiver") {
     ":rtp_interfaces",
     "..:webrtc_common",
     "../api:array_view",
+    "../api:optional",
     "../modules/rtp_rtcp",
     "../rtc_base:rtc_base_approved",
   ]
@@ -114,6 +116,7 @@ rtc_static_library("call") {
     ":rtp_sender",
     ":video_stream_api",
     "..:webrtc_common",
+    "../api:optional",
     "../api:transport_api",
     "../audio",
     "../logging:rtc_event_log_api",
@@ -144,6 +147,7 @@ rtc_source_set("video_stream_api") {
     ":rtp_interfaces",
     "../:webrtc_common",
     "../api:libjingle_peerconnection_api",
+    "../api:optional",
     "../api:transport_api",
     "../common_video:common_video",
     "../rtc_base:rtc_base_approved",
diff --git a/webrtc/call/audio_receive_stream.h b/webrtc/call/audio_receive_stream.h
index 018ad35fb7..6de62dffa6 100644
--- a/webrtc/call/audio_receive_stream.h
+++ b/webrtc/call/audio_receive_stream.h
@@ -18,11 +18,11 @@
 
 #include "webrtc/api/audio_codecs/audio_decoder_factory.h"
 #include "webrtc/api/call/transport.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/call/rtp_config.h"
 #include "webrtc/common_types.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/scoped_ref_ptr.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/call/audio_send_stream.h b/webrtc/call/audio_send_stream.h
index 8720249c4b..4f7f8c0f84 100644
--- a/webrtc/call/audio_send_stream.h
+++ b/webrtc/call/audio_send_stream.h
@@ -18,9 +18,9 @@
 #include "webrtc/api/audio_codecs/audio_encoder_factory.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/call/transport.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/call/rtp_config.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/scoped_ref_ptr.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index edb9bf3106..343b3ef41f 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -16,6 +16,7 @@
 #include <utility>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
@@ -40,7 +41,6 @@
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/location.h"
 #include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/rtc_base/sequenced_task_checker.h"
 #include "webrtc/rtc_base/task_queue.h"
diff --git a/webrtc/call/rtp_rtcp_demuxer_helper.h b/webrtc/call/rtp_rtcp_demuxer_helper.h
index 02d08aacba..0a4767d84a 100644
--- a/webrtc/call/rtp_rtcp_demuxer_helper.h
+++ b/webrtc/call/rtp_rtcp_demuxer_helper.h
@@ -16,8 +16,8 @@
 #include <utility>
 
 #include "webrtc/api/array_view.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/rtc_base/basictypes.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/call/syncable.h b/webrtc/call/syncable.h
index 734d20ec06..2692a8a09c 100644
--- a/webrtc/call/syncable.h
+++ b/webrtc/call/syncable.h
@@ -16,7 +16,7 @@
 
 #include <stdint.h>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/call/video_config.h b/webrtc/call/video_config.h
index cdff513263..4a98d01405 100644
--- a/webrtc/call/video_config.h
+++ b/webrtc/call/video_config.h
@@ -14,9 +14,9 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/rtc_base/basictypes.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/refcount.h"
 #include "webrtc/rtc_base/scoped_ref_ptr.h"
 #include "webrtc/typedefs.h"
diff --git a/webrtc/common_audio/BUILD.gn b/webrtc/common_audio/BUILD.gn
index a51a83c3a7..8cfd642d97 100644
--- a/webrtc/common_audio/BUILD.gn
+++ b/webrtc/common_audio/BUILD.gn
@@ -63,6 +63,7 @@ rtc_static_library("common_audio") {
   deps = [
     ":sinc_resampler",
     "..:webrtc_common",
+    "../api:optional",
     "../rtc_base:gtest_prod",
     "../rtc_base:rtc_base_approved",
     "../system_wrappers",
diff --git a/webrtc/common_audio/smoothing_filter.h b/webrtc/common_audio/smoothing_filter.h
index c59842ec8c..15ea3d1e55 100644
--- a/webrtc/common_audio/smoothing_filter.h
+++ b/webrtc/common_audio/smoothing_filter.h
@@ -11,8 +11,8 @@
 #ifndef WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_
 #define WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_
 
+#include "webrtc/api/optional.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index 229d858c92..49b42ac376 100644
--- a/webrtc/common_types.h
+++ b/webrtc/common_types.h
@@ -18,12 +18,12 @@
 #include <vector>
 
 #include "webrtc/api/array_view.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/api/video/video_content_type.h"
 #include "webrtc/api/video/video_rotation.h"
 #include "webrtc/api/video/video_timing.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/deprecation.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/typedefs.h"
 
 #if defined(_MSC_VER)
diff --git a/webrtc/common_video/BUILD.gn b/webrtc/common_video/BUILD.gn
index 93906ef51b..f0e262f23b 100644
--- a/webrtc/common_video/BUILD.gn
+++ b/webrtc/common_video/BUILD.gn
@@ -57,6 +57,7 @@ rtc_static_library("common_video") {
 
   deps = [
     "..:webrtc_common",
+    "../api:optional",
     "../media:rtc_h264_profile_id",
     "../modules:module_api",
     "../rtc_base:rtc_base",
diff --git a/webrtc/common_video/h264/h264_bitstream_parser.h b/webrtc/common_video/h264/h264_bitstream_parser.h
index 136c90317f..145700f509 100644
--- a/webrtc/common_video/h264/h264_bitstream_parser.h
+++ b/webrtc/common_video/h264/h264_bitstream_parser.h
@@ -13,9 +13,9 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/common_video/h264/pps_parser.h"
 #include "webrtc/common_video/h264/sps_parser.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace rtc {
 class BitBufferWriter;
diff --git a/webrtc/common_video/h264/pps_parser.h b/webrtc/common_video/h264/pps_parser.h
index 941b99e71e..b981d22a3b 100644
--- a/webrtc/common_video/h264/pps_parser.h
+++ b/webrtc/common_video/h264/pps_parser.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_COMMON_VIDEO_H264_PPS_PARSER_H_
 #define WEBRTC_COMMON_VIDEO_H264_PPS_PARSER_H_
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace rtc {
 class BitBuffer;
diff --git a/webrtc/common_video/h264/sps_parser.h b/webrtc/common_video/h264/sps_parser.h
index ea14f79202..77ad25d83f 100644
--- a/webrtc/common_video/h264/sps_parser.h
+++ b/webrtc/common_video/h264/sps_parser.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_COMMON_VIDEO_H264_SPS_PARSER_H_
 #define WEBRTC_COMMON_VIDEO_H264_SPS_PARSER_H_
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace rtc {
 class BitBuffer;
diff --git a/webrtc/common_video/h264/sps_vui_rewriter.h b/webrtc/common_video/h264/sps_vui_rewriter.h
index 6eff1abcea..7875117722 100644
--- a/webrtc/common_video/h264/sps_vui_rewriter.h
+++ b/webrtc/common_video/h264/sps_vui_rewriter.h
@@ -12,9 +12,9 @@
 #ifndef WEBRTC_COMMON_VIDEO_H264_SPS_VUI_REWRITER_H_
 #define WEBRTC_COMMON_VIDEO_H264_SPS_VUI_REWRITER_H_
 
+#include "webrtc/api/optional.h"
 #include "webrtc/common_video/h264/sps_parser.h"
 #include "webrtc/rtc_base/buffer.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace rtc {
 class BitBuffer;
diff --git a/webrtc/common_video/video_render_frames.h b/webrtc/common_video/video_render_frames.h
index 4f043e16ba..fe4d4f0b07 100644
--- a/webrtc/common_video/video_render_frames.h
+++ b/webrtc/common_video/video_render_frames.h
@@ -15,8 +15,8 @@
 
 #include <list>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/video/video_frame.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn
index 0ed8fc2663..25aa1e07a7 100644
--- a/webrtc/media/BUILD.gn
+++ b/webrtc/media/BUILD.gn
@@ -45,6 +45,7 @@ rtc_source_set("rtc_h264_profile_id") {
 
   deps = [
     "..:webrtc_common",
+    "../api:optional",
     "../rtc_base:rtc_base",
     "../rtc_base:rtc_base_approved",
   ]
@@ -227,6 +228,7 @@ rtc_static_library("rtc_audio_video") {
     "..:webrtc_common",
     "../api:call_api",
     "../api:libjingle_peerconnection_api",
+    "../api:optional",
     "../api:transport_api",
     "../api:video_frame_api",
     "../api/audio_codecs:audio_codecs_api",
diff --git a/webrtc/media/base/h264_profile_level_id.h b/webrtc/media/base/h264_profile_level_id.h
index 7b1d8c5c80..acbb84e194 100644
--- a/webrtc/media/base/h264_profile_level_id.h
+++ b/webrtc/media/base/h264_profile_level_id.h
@@ -14,8 +14,8 @@
 #include <map>
 #include <string>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/common_types.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 namespace H264 {
diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
index 5aae309feb..ef245a133a 100644
--- a/webrtc/media/base/mediachannel.h
+++ b/webrtc/media/base/mediachannel.h
@@ -15,6 +15,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/api/video/video_timing.h"
@@ -30,7 +31,6 @@
 #include "webrtc/rtc_base/dscp.h"
 #include "webrtc/rtc_base/logging.h"
 #include "webrtc/rtc_base/networkroute.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/sigslot.h"
 #include "webrtc/rtc_base/socket.h"
 #include "webrtc/rtc_base/window.h"
diff --git a/webrtc/media/base/videoadapter.cc b/webrtc/media/base/videoadapter.cc
index 73a5638bb3..45868b67d9 100644
--- a/webrtc/media/base/videoadapter.cc
+++ b/webrtc/media/base/videoadapter.cc
@@ -15,12 +15,12 @@
 #include <cstdlib>
 #include <limits>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/videocommon.h"
 #include "webrtc/rtc_base/arraysize.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace {
 struct Fraction {
diff --git a/webrtc/media/base/videoadapter.h b/webrtc/media/base/videoadapter.h
index 135f411b0a..61de4b5980 100644
--- a/webrtc/media/base/videoadapter.h
+++ b/webrtc/media/base/videoadapter.h
@@ -11,10 +11,10 @@
 #ifndef WEBRTC_MEDIA_BASE_VIDEOADAPTER_H_
 #define WEBRTC_MEDIA_BASE_VIDEOADAPTER_H_
 
+#include "webrtc/api/optional.h"
 #include "webrtc/media/base/videocommon.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace cricket {
 
diff --git a/webrtc/media/base/videosourceinterface.h b/webrtc/media/base/videosourceinterface.h
index 24b3bed5b6..551a6ea12b 100644
--- a/webrtc/media/base/videosourceinterface.h
+++ b/webrtc/media/base/videosourceinterface.h
@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MEDIA_BASE_VIDEOSOURCEINTERFACE_H_
 #define WEBRTC_MEDIA_BASE_VIDEOSOURCEINTERFACE_H_
 
+#include "webrtc/api/optional.h"
 #include "webrtc/media/base/videosinkinterface.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace rtc {
 
diff --git a/webrtc/media/engine/payload_type_mapper.h b/webrtc/media/engine/payload_type_mapper.h
index 1a74f21008..128f557ae7 100644
--- a/webrtc/media/engine/payload_type_mapper.h
+++ b/webrtc/media/engine/payload_type_mapper.h
@@ -15,8 +15,8 @@
 #include <set>
 
 #include "webrtc/api/audio_codecs/audio_format.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/media/base/codec.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace cricket {
 
diff --git a/webrtc/media/engine/webrtcvideoengine.h b/webrtc/media/engine/webrtcvideoengine.h
index 3224d0027f..7710c043d1 100644
--- a/webrtc/media/engine/webrtcvideoengine.h
+++ b/webrtc/media/engine/webrtcvideoengine.h
@@ -18,6 +18,7 @@
 #include <vector>
 
 #include "webrtc/api/call/transport.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/api/video/video_frame.h"
 #include "webrtc/call/call.h"
 #include "webrtc/call/flexfec_receive_stream.h"
@@ -31,7 +32,6 @@
 #include "webrtc/rtc_base/asyncinvoker.h"
 #include "webrtc/rtc_base/criticalsection.h"
 #include "webrtc/rtc_base/networkroute.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/thread_annotations.h"
 #include "webrtc/rtc_base/thread_checker.h"
 
diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn
index 359bd8c05b..dd8e54b68b 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -36,6 +36,7 @@ rtc_source_set("module_api") {
   ]
   deps = [
     "..:webrtc_common",
+    "../api:optional",
     "../api:video_frame_api",
     "../rtc_base:rtc_base_approved",
   ]
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 8d92c46719..4003db1b35 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -47,6 +47,7 @@ rtc_static_library("audio_format_conversion") {
   deps = [
     "../..:webrtc_common",
     "../../api:array_view",
+    "../../api:optional",
     "../../api/audio_codecs:audio_codecs_api",
     "../../rtc_base:rtc_base_approved",
   ]
@@ -61,6 +62,7 @@ rtc_static_library("rent_a_codec") {
   ]
   deps = [
            "../../api:array_view",
+           "../../api:optional",
            "../../api/audio_codecs:audio_codecs_api",
            "../..:webrtc_common",
            "../../rtc_base:protobuf_utils",
@@ -132,6 +134,7 @@ rtc_static_library("audio_coding") {
            ":neteq",
            ":rent_a_codec",
            "../../rtc_base:rtc_base_approved",
+           "../../api:optional",
            "../../logging:rtc_event_log_api",
          ]
   defines = audio_coding_defines
@@ -814,6 +817,7 @@ rtc_static_library("webrtc_opus") {
   deps = [
     ":audio_network_adaptor",
     "../..:webrtc_common",
+    "../../api:optional",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/audio_codecs/opus:audio_encoder_opus_config",
     "../../common_audio",
@@ -908,6 +912,7 @@ rtc_static_library("audio_network_adaptor") {
 
   deps = [
     "../..:webrtc_common",
+    "../../api:optional",
     "../../common_audio",
     "../../logging:rtc_event_log_api",
     "../../rtc_base:protobuf_utils",
@@ -934,6 +939,7 @@ rtc_source_set("neteq_decoder_enum") {
     "neteq/neteq_decoder_enum.h",
   ]
   deps = [
+    "../../api:optional",
     "../../api/audio_codecs:audio_codecs_api",
     "../../rtc_base:rtc_base_approved",
   ]
@@ -1023,6 +1029,7 @@ rtc_static_library("neteq") {
     ":pcm16b",
     "..:module_api",
     "../..:webrtc_common",
+    "../../api:optional",
     "../../api/audio_codecs:audio_codecs_api",
     "../../common_audio",
     "../../rtc_base:gtest_prod",
@@ -1082,6 +1089,7 @@ rtc_source_set("neteq_tools_minimal") {
     ":neteq",
     "..:module_api",
     "../..:webrtc_common",
+    "../../api:optional",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/audio_codecs:builtin_audio_decoder_factory",
     "../../rtc_base:rtc_base_approved",
@@ -1166,6 +1174,7 @@ rtc_source_set("neteq_tools") {
   deps = [
     "../..:webrtc_common",
     "../../api:array_view",
+    "../../api:optional",
     "../../api/audio_codecs:audio_codecs_api",
     "../../common_audio",
     "../../rtc_base:rtc_base_approved",
@@ -1290,6 +1299,7 @@ if (rtc_include_tests) {
       ":pcm16b_c",
       "..:module_api",
       "../..:webrtc_common",
+      "../../api:optional",
       "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:system_wrappers",
@@ -1396,6 +1406,7 @@ if (rtc_include_tests) {
       ":audio_format_conversion",
       "..:module_api",
       "../../:webrtc_common",
+      "../../api:optional",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers",
       "../../system_wrappers:system_wrappers_default",
@@ -1425,6 +1436,7 @@ if (rtc_include_tests) {
       ":audio_format_conversion",
       "..:module_api",
       "../../:webrtc_common",
+      "../../api:optional",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers",
       "../../system_wrappers:system_wrappers_default",
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index 5acb013288..ebeecec7f9 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -17,6 +17,7 @@
 #include <vector>
 
 #include "webrtc/api/array_view.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/acm2/call_statistics.h"
@@ -24,7 +25,6 @@
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/thread_annotations.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/audio_coding/acm2/codec_manager.h b/webrtc/modules/audio_coding/acm2/codec_manager.h
index 3d45b335bc..b5c8d3e53c 100644
--- a/webrtc/modules/audio_coding/acm2/codec_manager.h
+++ b/webrtc/modules/audio_coding/acm2/codec_manager.h
@@ -13,12 +13,12 @@
 
 #include <map>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/thread_checker.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/acm2/rent_a_codec.h b/webrtc/modules/audio_coding/acm2/rent_a_codec.h
index 3dd9dc8209..6dcf793d04 100644
--- a/webrtc/modules/audio_coding/acm2/rent_a_codec.h
+++ b/webrtc/modules/audio_coding/acm2/rent_a_codec.h
@@ -18,10 +18,10 @@
 #include "webrtc/api/array_view.h"
 #include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/api/audio_codecs/audio_encoder.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/neteq/neteq_decoder_enum.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/scoped_ref_ptr.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
index ab9cfb706f..38c2c2122a 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
index 6d03730a17..fa742e69f6 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
 #define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc b/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
index 17f8d9623e..1e53bf35fc 100644
--- a/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
@@ -13,8 +13,8 @@
 #include <string.h>
 
 #include "webrtc/api/array_view.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/safe_conversions.h"
 #include "webrtc/rtc_base/sanitizer.h"
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
index 387b34b5da..66f021a87c 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
@@ -14,9 +14,9 @@
 #include <vector>
 
 #include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index f78cf1e39c..588f268e7f 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -19,11 +19,11 @@
 #include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/common_audio/smoothing_filter.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/protobuf_utils.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
index 44eec75666..4ceade25d0 100644
--- a/webrtc/modules/audio_coding/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -17,13 +17,13 @@
 
 #include "webrtc/api/audio_codecs/audio_decoder_factory.h"
 #include "webrtc/api/audio_codecs/audio_encoder.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/rtc_base/deprecation.h"
 #include "webrtc/rtc_base/function_view.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index 56c30e5f2d..9f5af7cea5 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -16,10 +16,10 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/scoped_ref_ptr.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/audio_coding/neteq/neteq_decoder_enum.h b/webrtc/modules/audio_coding/neteq/neteq_decoder_enum.h
index a36034ccdf..e108ccfc41 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_decoder_enum.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_decoder_enum.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_DECODER_ENUM_H_
 
 #include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index f4b014adac..fea1682f74 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -14,6 +14,7 @@
 #include <memory>
 #include <string>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
 #include "webrtc/modules/audio_coding/neteq/defines.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
@@ -25,7 +26,6 @@
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/thread_annotations.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/audio_coding/neteq/packet_buffer.h b/webrtc/modules/audio_coding/neteq/packet_buffer.h
index afd5f04454..5c2499c6b9 100644
--- a/webrtc/modules/audio_coding/neteq/packet_buffer.h
+++ b/webrtc/modules/audio_coding/neteq/packet_buffer.h
@@ -11,10 +11,10 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
index d293ecde21..09c2e789f0 100644
--- a/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
+++ b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
@@ -15,8 +15,8 @@
 
 #include "webrtc/api/array_view.h"
 #include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 namespace test {
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h b/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
index b652bf2290..e142a3b0c3 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
@@ -16,9 +16,9 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_test.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_input.h b/webrtc/modules/audio_coding/neteq/tools/neteq_input.h
index 1e0889336a..08f554bd4e 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_input.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_input.h
@@ -15,11 +15,11 @@
 #include <memory>
 #include <string>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
 #include "webrtc/rtc_base/buffer.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 namespace test {
diff --git a/webrtc/modules/audio_coding/test/PCMFile.h b/webrtc/modules/audio_coding/test/PCMFile.h
index fd1c53a10b..3576dc67d8 100644
--- a/webrtc/modules/audio_coding/test/PCMFile.h
+++ b/webrtc/modules/audio_coding/test/PCMFile.h
@@ -16,8 +16,8 @@
 
 #include <string>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn
index b9716aea94..2db993b7b3 100644
--- a/webrtc/modules/audio_device/BUILD.gn
+++ b/webrtc/modules/audio_device/BUILD.gn
@@ -283,6 +283,7 @@ if (rtc_include_tests) {
       ":audio_device",
       ":mock_audio_device",
       "../../api:array_view",
+      "../../api:optional",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:system_wrappers",
       "../../test:test_support",
diff --git a/webrtc/modules/audio_device/audio_device_unittest.cc b/webrtc/modules/audio_device/audio_device_unittest.cc
index 67e9f4732f..c65f139192 100644
--- a/webrtc/modules/audio_device/audio_device_unittest.cc
+++ b/webrtc/modules/audio_device/audio_device_unittest.cc
@@ -13,6 +13,7 @@
 #include <numeric>
 
 #include "webrtc/api/array_view.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_device/audio_device_impl.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_device/include/mock_audio_transport.h"
@@ -20,7 +21,6 @@
 #include "webrtc/rtc_base/criticalsection.h"
 #include "webrtc/rtc_base/event.h"
 #include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/race_checker.h"
 #include "webrtc/rtc_base/safe_conversions.h"
 #include "webrtc/rtc_base/scoped_ref_ptr.h"
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index e0719b4315..4887250d88 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -238,6 +238,7 @@ rtc_static_library("audio_processing") {
     "..:module_api",
     "../..:webrtc_common",
     "../../api:array_view",
+    "../../api:optional",
     "../../audio/utility:audio_frame_operations",
     "../../rtc_base:gtest_prod",
     "../../rtc_base:protobuf_utils",
@@ -553,6 +554,7 @@ if (rtc_include_tests) {
       "..:module_api",
       "../..:webrtc_common",
       "../../api:array_view",
+      "../../api:optional",
       "../../common_audio:common_audio",
       "../../rtc_base:gtest_prod",
       "../../rtc_base:protobuf_utils",
@@ -747,6 +749,7 @@ if (rtc_include_tests) {
         ":audioproc_debug_proto",
         ":audioproc_protobuf_utils",
         ":audioproc_test_utils",
+        "../../api:optional",
         "../../common_audio:common_audio",
         "../../rtc_base:protobuf_utils",
         "../../rtc_base:rtc_base_approved",
@@ -778,6 +781,7 @@ if (rtc_include_tests) {
       ":audio_processing",
       "..:module_api",
       "../../api:array_view",
+      "../../api:optional",
       "../../common_audio",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:system_wrappers",
diff --git a/webrtc/modules/audio_processing/aec3/aec_state.h b/webrtc/modules/audio_processing/aec3/aec_state.h
index 8dc45304f1..38ed3b047b 100644
--- a/webrtc/modules/audio_processing/aec3/aec_state.h
+++ b/webrtc/modules/audio_processing/aec3/aec_state.h
@@ -16,6 +16,7 @@
 #include <vector>
 
 #include "webrtc/api/array_view.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/aec3/aec3_common.h"
 #include "webrtc/modules/audio_processing/aec3/echo_path_variability.h"
 #include "webrtc/modules/audio_processing/aec3/erl_estimator.h"
@@ -23,7 +24,6 @@
 #include "webrtc/modules/audio_processing/aec3/render_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/aec3/block_processor.cc b/webrtc/modules/audio_processing/aec3/block_processor.cc
index 536335e8a2..7bb9e15129 100644
--- a/webrtc/modules/audio_processing/aec3/block_processor.cc
+++ b/webrtc/modules/audio_processing/aec3/block_processor.cc
@@ -9,13 +9,13 @@
  */
 #include "webrtc/modules/audio_processing/aec3/block_processor.h"
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/aec3/aec3_common.h"
 #include "webrtc/modules/audio_processing/aec3/block_processor_metrics.h"
 #include "webrtc/modules/audio_processing/aec3/echo_path_variability.h"
 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
 #include "webrtc/rtc_base/atomicops.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 namespace {
diff --git a/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h b/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h
index 4b536b99a9..7ff1cb33a5 100644
--- a/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h
+++ b/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h
@@ -13,13 +13,13 @@
 
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/aec3/decimator_by_4.h"
 #include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h"
 #include "webrtc/modules/audio_processing/aec3/matched_filter.h"
 #include "webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/aec3/echo_remover.h b/webrtc/modules/audio_processing/aec3/echo_remover.h
index ce43fa8a72..8a28729ba6 100644
--- a/webrtc/modules/audio_processing/aec3/echo_remover.h
+++ b/webrtc/modules/audio_processing/aec3/echo_remover.h
@@ -13,10 +13,10 @@
 
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/aec3/echo_path_variability.h"
 #include "webrtc/modules/audio_processing/aec3/render_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/aec3/matched_filter.h b/webrtc/modules/audio_processing/aec3/matched_filter.h
index 0702a124e8..a200af1261 100644
--- a/webrtc/modules/audio_processing/aec3/matched_filter.h
+++ b/webrtc/modules/audio_processing/aec3/matched_filter.h
@@ -15,10 +15,10 @@
 #include <memory>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/aec3/aec3_common.h"
 #include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 namespace aec3 {
diff --git a/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h b/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
index 983ae3b7f3..d490f0d45d 100644
--- a/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
+++ b/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
@@ -13,9 +13,9 @@
 
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/aec3/matched_filter.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/aec3/mock/mock_echo_remover.h b/webrtc/modules/audio_processing/aec3/mock/mock_echo_remover.h
index 008fca669f..8d2aca992d 100644
--- a/webrtc/modules/audio_processing/aec3/mock/mock_echo_remover.h
+++ b/webrtc/modules/audio_processing/aec3/mock/mock_echo_remover.h
@@ -13,10 +13,10 @@
 
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/aec3/echo_path_variability.h"
 #include "webrtc/modules/audio_processing/aec3/echo_remover.h"
 #include "webrtc/modules/audio_processing/aec3/render_buffer.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/test/gmock.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h b/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h
index 46a59dc201..64daef5a45 100644
--- a/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h
+++ b/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h
@@ -12,9 +12,9 @@
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_CONTROLLER_H_
 
 #include "webrtc/api/array_view.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h"
 #include "webrtc/modules/audio_processing/aec3/render_delay_controller.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/test/gmock.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_processing/aec3/render_delay_controller.h b/webrtc/modules/audio_processing/aec3/render_delay_controller.h
index c1e2229d66..e840178cb6 100644
--- a/webrtc/modules/audio_processing/aec3/render_delay_controller.h
+++ b/webrtc/modules/audio_processing/aec3/render_delay_controller.h
@@ -12,11 +12,11 @@
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
 
 #include "webrtc/api/array_view.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h"
 #include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h b/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h
index 0cef897c34..625f30f04a 100644
--- a/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h
+++ b/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h
@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_
 
+#include "webrtc/api/optional.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc b/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
index 5f24fe3d7f..bb39e77471 100644
--- a/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
+++ b/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
@@ -9,8 +9,8 @@
  */
 
 #include "webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/aec3/aec3_common.h"
-#include "webrtc/rtc_base/optional.h"
 
 #include "webrtc/test/gtest.h"
 
diff --git a/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h b/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
index b0b56cde9e..48ce81e9fc 100644
--- a/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
+++ b/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
@@ -14,10 +14,10 @@
 #include <array>
 #include <memory>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/aec3/aec3_common.h"
 #include "webrtc/modules/audio_processing/aec3/render_buffer.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/beamformer/array_util.h b/webrtc/modules/audio_processing/beamformer/array_util.h
index bfb9dfdb65..aa18e71931 100644
--- a/webrtc/modules/audio_processing/beamformer/array_util.h
+++ b/webrtc/modules/audio_processing/beamformer/array_util.h
@@ -14,7 +14,7 @@
 #include <cmath>
 #include <vector>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/echo_detector/circular_buffer.h b/webrtc/modules/audio_processing/echo_detector/circular_buffer.h
index 9fadb0d66f..cf58e33840 100644
--- a/webrtc/modules/audio_processing/echo_detector/circular_buffer.h
+++ b/webrtc/modules/audio_processing/echo_detector/circular_buffer.h
@@ -13,7 +13,7 @@
 
 #include <vector>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc
index e5760183f0..f882d10cd6 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -10,11 +10,11 @@
 
 #include "webrtc/modules/audio_processing/gain_control_impl.h"
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/level_controller/level_controller.h b/webrtc/modules/audio_processing/level_controller/level_controller.h
index e04620f19c..9aa50852df 100644
--- a/webrtc/modules/audio_processing/level_controller/level_controller.h
+++ b/webrtc/modules/audio_processing/level_controller/level_controller.h
@@ -14,6 +14,7 @@
 #include <memory>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/level_controller/gain_applier.h"
 #include "webrtc/modules/audio_processing/level_controller/gain_selector.h"
@@ -22,7 +23,6 @@
 #include "webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.h"
 #include "webrtc/modules/audio_processing/level_controller/signal_classifier.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc b/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc
index b7571e92d5..d37b58011b 100644
--- a/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc
+++ b/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc
@@ -11,12 +11,12 @@
 #include <vector>
 
 #include "webrtc/api/array_view.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/level_controller/level_controller.h"
 #include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
 #include "webrtc/modules/audio_processing/test/bitexactness_tools.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/test/gtest.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_processing/rms_level.h b/webrtc/modules/audio_processing/rms_level.h
index 4ce28a4957..e0df754f56 100644
--- a/webrtc/modules/audio_processing/rms_level.h
+++ b/webrtc/modules/audio_processing/rms_level.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
 
 #include "webrtc/api/array_view.h"
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.h b/webrtc/modules/audio_processing/test/audio_processing_simulator.h
index 1ae465cbef..8a2295f0cd 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.h
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.h
@@ -17,11 +17,11 @@
 #include <memory>
 #include <string>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/task_queue.h"
 #include "webrtc/rtc_base/timeutils.h"
 
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn b/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
index 1f24c0982a..1d492bf644 100644
--- a/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
+++ b/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
@@ -65,6 +65,7 @@ rtc_source_set("unittest") {
     ":lib",
     "../../../..:webrtc_common",
     "../../../../api:array_view",
+    "../../../../api:optional",
     "../../../../common_audio",
     "../../../../rtc_base:rtc_base_approved",
     "../../../../test:test_support",
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc b/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
index 9a52e5cc1f..e0aa43de23 100644
--- a/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
+++ b/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
@@ -42,6 +42,7 @@
 #include <memory>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/test/conversational_speech/config.h"
 #include "webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h"
@@ -50,7 +51,6 @@
 #include "webrtc/modules/audio_processing/test/conversational_speech/timing.h"
 #include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h"
 #include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/pathutils.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
diff --git a/webrtc/modules/audio_processing/test/performance_timer.h b/webrtc/modules/audio_processing/test/performance_timer.h
index 3e75e3f5f5..9377f4ae77 100644
--- a/webrtc/modules/audio_processing/test/performance_timer.h
+++ b/webrtc/modules/audio_processing/test/performance_timer.h
@@ -13,7 +13,7 @@
 
 #include <vector>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/congestion_controller/BUILD.gn b/webrtc/modules/congestion_controller/BUILD.gn
index 8c3f35be14..be0afd2dfd 100644
--- a/webrtc/modules/congestion_controller/BUILD.gn
+++ b/webrtc/modules/congestion_controller/BUILD.gn
@@ -49,6 +49,7 @@ rtc_static_library("congestion_controller") {
   deps = [
     "..:module_api",
     "../..:webrtc_common",
+    "../../api:optional",
     "../../logging:rtc_event_log_api",
     "../../rtc_base:rtc_base",
     "../../rtc_base:rtc_base_approved",
diff --git a/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h b/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h
index bab5d93ce5..491acb67c3 100644
--- a/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h
+++ b/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h
@@ -14,8 +14,8 @@
 #include <memory>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/congestion_controller/bitrate_estimator.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/congestion_controller/bitrate_estimator.h b/webrtc/modules/congestion_controller/bitrate_estimator.h
index f27d0630ef..ad7bef5fc2 100644
--- a/webrtc/modules/congestion_controller/bitrate_estimator.h
+++ b/webrtc/modules/congestion_controller/bitrate_estimator.h
@@ -13,7 +13,7 @@
 
 #include <vector>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/congestion_controller/trendline_estimator.cc b/webrtc/modules/congestion_controller/trendline_estimator.cc
index 4d25b45bba..e51d4fb137 100644
--- a/webrtc/modules/congestion_controller/trendline_estimator.cc
+++ b/webrtc/modules/congestion_controller/trendline_estimator.cc
@@ -12,9 +12,9 @@
 
 #include <algorithm>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/include/module_common_types.h b/webrtc/modules/include/module_common_types.h
index 1ee428deed..a3ac2bbc56 100644
--- a/webrtc/modules/include/module_common_types.h
+++ b/webrtc/modules/include/module_common_types.h
@@ -17,6 +17,7 @@
 #include <algorithm>
 #include <limits>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/video/video_rotation.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/video_coding/codecs/h264/include/h264_globals.h"
@@ -24,7 +25,6 @@
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/deprecation.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/safe_conversions.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/pacing/BUILD.gn b/webrtc/modules/pacing/BUILD.gn
index 7884c01e55..4882d4f714 100644
--- a/webrtc/modules/pacing/BUILD.gn
+++ b/webrtc/modules/pacing/BUILD.gn
@@ -31,6 +31,7 @@ rtc_static_library("pacing") {
   deps = [
     "..:module_api",
     "../../:webrtc_common",
+    "../../api:optional",
     "../../logging:rtc_event_log_api",
     "../../rtc_base:rtc_base_approved",
     "../../system_wrappers",
diff --git a/webrtc/modules/pacing/alr_detector.h b/webrtc/modules/pacing/alr_detector.h
index 1e124d99e9..0d43fbb8f2 100644
--- a/webrtc/modules/pacing/alr_detector.h
+++ b/webrtc/modules/pacing/alr_detector.h
@@ -11,10 +11,10 @@
 #ifndef WEBRTC_MODULES_PACING_ALR_DETECTOR_H_
 #define WEBRTC_MODULES_PACING_ALR_DETECTOR_H_
 
+#include "webrtc/api/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/pacing/interval_budget.h"
 #include "webrtc/modules/pacing/paced_sender.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/rate_statistics.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/pacing/paced_sender.h b/webrtc/modules/pacing/paced_sender.h
index 74b87804de..a9c6233b27 100644
--- a/webrtc/modules/pacing/paced_sender.h
+++ b/webrtc/modules/pacing/paced_sender.h
@@ -15,9 +15,9 @@
 #include <memory>
 #include <set>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/pacing/pacer.h"
 #include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/thread_annotations.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/remote_bitrate_estimator/BUILD.gn b/webrtc/modules/remote_bitrate_estimator/BUILD.gn
index 8bcbe7b334..72e6f1de12 100644
--- a/webrtc/modules/remote_bitrate_estimator/BUILD.gn
+++ b/webrtc/modules/remote_bitrate_estimator/BUILD.gn
@@ -140,6 +140,7 @@ if (rtc_include_tests) {
       ":remote_bitrate_estimator",
       "..:module_api",
       "../..:webrtc_common",
+      "../../api:optional",
       "../../rtc_base:gtest_prod",
       "../../rtc_base:rtc_base",
       "../../rtc_base:rtc_base_approved",
diff --git a/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h b/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
index 21f9e57cad..426dbb1b6c 100644
--- a/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
+++ b/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
 #define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/typedefs.h"
 
 #define BWE_MAX(a, b) ((a) > (b) ? (a) : (b))
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.h b/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.h
index 6f4d831bc2..cf61c5747e 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.h
@@ -17,9 +17,9 @@
 #include <memory>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/modules/video_coding/sequence_number_util.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/random.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h b/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h
index 105a748d63..3194d81c2a 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h
@@ -14,7 +14,7 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.h"
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h b/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h
index f45437eee5..82451bd242 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h
@@ -16,7 +16,7 @@
 #include <limits>
 #include <list>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 namespace testing {
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
index 8d17d1a799..2095aadb28 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -169,6 +169,7 @@ rtc_static_library("rtp_rtcp") {
     "../..:webrtc_common",
     "../../api:array_view",
     "../../api:libjingle_peerconnection_api",
+    "../../api:optional",
     "../../api:transport_api",
     "../../api/audio_codecs:audio_codecs_api",
     "../../common_video",
@@ -222,6 +223,7 @@ rtc_source_set("mock_rtp_rtcp") {
   deps = [
     ":rtp_rtcp",
     "..:module_api",
+    "../../api:optional",
     "../../rtc_base:rtc_base_approved",
     "../../test:test_support",
   ]
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
index 427229472c..25d904672a 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -16,13 +16,13 @@
 #include <utility>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/deprecation.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 54d84d595d..1d45463484 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -15,12 +15,12 @@
 #include <utility>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/test/gmock.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h
index aed304f74b..33a23de57a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h
@@ -13,12 +13,12 @@
 
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 namespace rtcp {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index 7b3ad496dd..ae06a17d8e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -19,6 +19,7 @@
 #include <vector>
 
 #include "webrtc/api/call/transport.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
@@ -30,7 +31,6 @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/random.h"
 #include "webrtc/rtc_base/thread_annotations.h"
 #include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 7a28be536d..aa9555076a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -16,6 +16,7 @@
 #include <utility>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
@@ -24,7 +25,6 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/rtc_base/criticalsection.h"
 #include "webrtc/rtc_base/gtest_prod_util.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 9c4fdc1b3d..f345274b9d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -18,6 +18,7 @@
 
 #include "webrtc/api/array_view.h"
 #include "webrtc/api/call/transport.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
@@ -29,7 +30,6 @@
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/criticalsection.h"
 #include "webrtc/rtc_base/deprecation.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/random.h"
 #include "webrtc/rtc_base/rate_statistics.h"
 #include "webrtc/rtc_base/thread_annotations.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
index e4a24052e5..f91fa69051 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -14,6 +14,7 @@
 #include <map>
 #include <memory>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -24,7 +25,6 @@
 #include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
 #include "webrtc/rtc_base/criticalsection.h"
 #include "webrtc/rtc_base/onetimeevent.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/rate_statistics.h"
 #include "webrtc/rtc_base/sequenced_task_checker.h"
 #include "webrtc/rtc_base/thread_annotations.h"
diff --git a/webrtc/modules/video_coding/BUILD.gn b/webrtc/modules/video_coding/BUILD.gn
index 3af09ea232..3bf221be17 100644
--- a/webrtc/modules/video_coding/BUILD.gn
+++ b/webrtc/modules/video_coding/BUILD.gn
@@ -96,6 +96,7 @@ rtc_static_library("video_coding") {
     ":webrtc_vp9",
     "..:module_api",
     "../..:webrtc_common",
+    "../../api:optional",
     "../../call:video_stream_api",
     "../../common_video",
     "../../rtc_base:rtc_base",
@@ -136,6 +137,7 @@ rtc_static_library("video_coding_utility") {
   deps = [
     "..:module_api",
     "../..:webrtc_common",
+    "../../api:optional",
     "../../api/video_codecs:video_codecs_api",
     "../../common_video",
     "../../modules/rtp_rtcp:rtp_rtcp",
@@ -232,6 +234,7 @@ rtc_static_library("webrtc_vp8") {
     ":video_coding_utility",
     "..:module_api",
     "../..:webrtc_common",
+    "../../api:optional",
     "../../api/video_codecs:video_codecs_api",
     "../../common_video",
     "../../rtc_base:rtc_base_approved",
@@ -438,6 +441,7 @@ if (rtc_include_tests) {
       ":webrtc_h264",
       ":webrtc_vp8",
       ":webrtc_vp9",
+      "../../api:optional",
       "../../api:video_frame_api",
       "../../common_video:common_video",
       "../../rtc_base:rtc_base_approved",
diff --git a/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h b/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
index 4762f872a7..985874b620 100644
--- a/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
+++ b/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
index 310fa3c543..3eb04be037 100644
--- a/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
@@ -12,13 +12,13 @@
 
 #include <memory>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/video/i420_buffer.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_coding/codecs/test/video_codec_test.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
 #include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/timeutils.h"
 #include "webrtc/test/field_trial.h"
 #include "webrtc/test/frame_utils.h"
diff --git a/webrtc/modules/video_coding/frame_object.h b/webrtc/modules/video_coding/frame_object.h
index 90c1c00564..69ab6f03c8 100644
--- a/webrtc/modules/video_coding/frame_object.h
+++ b/webrtc/modules/video_coding/frame_object.h
@@ -11,10 +11,10 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_OBJECT_H_
 #define WEBRTC_MODULES_VIDEO_CODING_FRAME_OBJECT_H_
 
+#include "webrtc/api/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/encoded_frame.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 namespace video_coding {
diff --git a/webrtc/modules/video_coding/generic_encoder.cc b/webrtc/modules/video_coding/generic_encoder.cc
index adc4365497..339e1b516f 100644
--- a/webrtc/modules/video_coding/generic_encoder.cc
+++ b/webrtc/modules/video_coding/generic_encoder.cc
@@ -12,13 +12,13 @@
 
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/video/i420_buffer.h"
 #include "webrtc/modules/pacing/alr_detector.h"
 #include "webrtc/modules/video_coding/encoded_frame.h"
 #include "webrtc/modules/video_coding/media_optimization.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/timeutils.h"
 #include "webrtc/rtc_base/trace_event.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
diff --git a/webrtc/modules/video_coding/sequence_number_util.h b/webrtc/modules/video_coding/sequence_number_util.h
index 05f33676d5..3e7d64a2f5 100644
--- a/webrtc/modules/video_coding/sequence_number_util.h
+++ b/webrtc/modules/video_coding/sequence_number_util.h
@@ -14,8 +14,8 @@
 #include <limits>
 #include <type_traits>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/rtc_base/mod_ops.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/safe_compare.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/video_coding/utility/moving_average.h b/webrtc/modules/video_coding/utility/moving_average.h
index bc31cd5213..0f2c4cc3c8 100644
--- a/webrtc/modules/video_coding/utility/moving_average.h
+++ b/webrtc/modules/video_coding/utility/moving_average.h
@@ -13,7 +13,7 @@
 
 #include <vector>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 class MovingAverage {
diff --git a/webrtc/modules/video_coding/utility/quality_scaler.h b/webrtc/modules/video_coding/utility/quality_scaler.h
index 690d1bc059..25d39cc986 100644
--- a/webrtc/modules/video_coding/utility/quality_scaler.h
+++ b/webrtc/modules/video_coding/utility/quality_scaler.h
@@ -13,10 +13,10 @@
 
 #include <utility>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/video_codecs/video_encoder.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/video_coding/utility/moving_average.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/sequenced_task_checker.h"
 
 namespace webrtc {
diff --git a/webrtc/ortc/BUILD.gn b/webrtc/ortc/BUILD.gn
index 199ee625c0..485771c9a4 100644
--- a/webrtc/ortc/BUILD.gn
+++ b/webrtc/ortc/BUILD.gn
@@ -33,6 +33,7 @@ rtc_static_library("ortc") {
   # PeerConnection code shares, so that ortc can depend on that instead of
   # libjingle_peerconnection.
   deps = [
+    "../api:optional",
     "../api/audio_codecs:builtin_audio_decoder_factory",
     "../api/audio_codecs:builtin_audio_encoder_factory",
     "../call:call_interfaces",
diff --git a/webrtc/ortc/rtpparametersconversion.h b/webrtc/ortc/rtpparametersconversion.h
index fe183da127..072aaf2264 100644
--- a/webrtc/ortc/rtpparametersconversion.h
+++ b/webrtc/ortc/rtpparametersconversion.h
@@ -14,11 +14,11 @@
 #include <memory>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/rtcerror.h"
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/media/base/codec.h"
 #include "webrtc/pc/mediasession.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/p2p/BUILD.gn b/webrtc/p2p/BUILD.gn
index 588f6bd1a3..9223f865b4 100644
--- a/webrtc/p2p/BUILD.gn
+++ b/webrtc/p2p/BUILD.gn
@@ -86,6 +86,7 @@ rtc_static_library("rtc_p2p") {
 
   deps = [
     "../api:libjingle_peerconnection_api",
+    "../api:optional",
     "../api:ortc_api",
     "../rtc_base:rtc_base",
     "../system_wrappers:field_trial_api",
diff --git a/webrtc/p2p/base/jseptransport.h b/webrtc/p2p/base/jseptransport.h
index aee07180c1..f680646e16 100644
--- a/webrtc/p2p/base/jseptransport.h
+++ b/webrtc/p2p/base/jseptransport.h
@@ -16,13 +16,13 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/p2p/base/candidate.h"
 #include "webrtc/p2p/base/p2pconstants.h"
 #include "webrtc/p2p/base/sessiondescription.h"
 #include "webrtc/p2p/base/transportinfo.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/messagequeue.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/rtccertificate.h"
 #include "webrtc/rtc_base/sigslot.h"
 #include "webrtc/rtc_base/sslstreamadapter.h"
diff --git a/webrtc/p2p/base/port.h b/webrtc/p2p/base/port.h
index 19391b52ba..76a3deaa49 100644
--- a/webrtc/p2p/base/port.h
+++ b/webrtc/p2p/base/port.h
@@ -17,6 +17,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/p2p/base/candidate.h"
 #include "webrtc/p2p/base/candidatepairinterface.h"
 #include "webrtc/p2p/base/jseptransport.h"
@@ -28,7 +29,6 @@
 #include "webrtc/rtc_base/asyncpacketsocket.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/network.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/proxyinfo.h"
 #include "webrtc/rtc_base/ratetracker.h"
 #include "webrtc/rtc_base/sigslot.h"
diff --git a/webrtc/p2p/base/udptransport.h b/webrtc/p2p/base/udptransport.h
index c3c9513bb9..823c88fcab 100644
--- a/webrtc/p2p/base/udptransport.h
+++ b/webrtc/p2p/base/udptransport.h
@@ -14,10 +14,10 @@
 #include <memory>
 #include <string>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/ortc/udptransportinterface.h"
 #include "webrtc/p2p/base/packettransportinternal.h"
 #include "webrtc/rtc_base/asyncpacketsocket.h"  // For PacketOptions.
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/thread_checker.h"
 
 namespace rtc {
diff --git a/webrtc/p2p/quic/quictransportchannel.h b/webrtc/p2p/quic/quictransportchannel.h
index 4c34675f20..76560e9f03 100644
--- a/webrtc/p2p/quic/quictransportchannel.h
+++ b/webrtc/p2p/quic/quictransportchannel.h
@@ -17,11 +17,11 @@
 
 #include "net/quic/quic_crypto_client_stream.h"
 #include "net/quic/quic_packet_writer.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/p2p/base/transportchannelimpl.h"
 #include "webrtc/p2p/quic/quicconnectionhelper.h"
 #include "webrtc/p2p/quic/quicsession.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace cricket {
 
diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn
index 6473cf68cc..09c58ab221 100644
--- a/webrtc/pc/BUILD.gn
+++ b/webrtc/pc/BUILD.gn
@@ -62,6 +62,7 @@ rtc_static_library("rtc_pc_base") {
     "..:webrtc_common",
     "../api:call_api",
     "../api:libjingle_peerconnection_api",
+    "../api:optional",
     "../api:ortc_api",
     "../media:rtc_data",
     "../media:rtc_h264_profile_id",
@@ -166,6 +167,7 @@ rtc_static_library("peerconnection") {
     ":rtc_pc_base",
     "..:webrtc_common",
     "../api:call_api",
+    "../api:optional",
     "../api:rtc_stats_api",
     "../api/video_codecs:video_codecs_api",
     "../call:call_interfaces",
diff --git a/webrtc/pc/mediasession.cc b/webrtc/pc/mediasession.cc
index 4d4744aeb0..2d9fd94b05 100644
--- a/webrtc/pc/mediasession.cc
+++ b/webrtc/pc/mediasession.cc
@@ -18,6 +18,7 @@
 #include <unordered_map>
 #include <utility>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/media/base/cryptoparams.h"
 #include "webrtc/media/base/h264_profile_level_id.h"
@@ -29,7 +30,6 @@
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/helpers.h"
 #include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/stringutils.h"
 
 namespace {
diff --git a/webrtc/pc/rtcstatscollector.h b/webrtc/pc/rtcstatscollector.h
index 980edd6078..5ba4bfd34f 100644
--- a/webrtc/pc/rtcstatscollector.h
+++ b/webrtc/pc/rtcstatscollector.h
@@ -16,6 +16,7 @@
 #include <set>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/stats/rtcstats_objects.h"
 #include "webrtc/api/stats/rtcstatscollectorcallback.h"
 #include "webrtc/api/stats/rtcstatsreport.h"
@@ -24,7 +25,6 @@
 #include "webrtc/pc/datachannel.h"
 #include "webrtc/pc/trackmediainfomap.h"
 #include "webrtc/rtc_base/asyncinvoker.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/refcount.h"
 #include "webrtc/rtc_base/scoped_ref_ptr.h"
 #include "webrtc/rtc_base/sigslot.h"
diff --git a/webrtc/pc/srtpfilter.h b/webrtc/pc/srtpfilter.h
index 619aaa37d5..2a90360860 100644
--- a/webrtc/pc/srtpfilter.h
+++ b/webrtc/pc/srtpfilter.h
@@ -17,13 +17,13 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/media/base/cryptoparams.h"
 #include "webrtc/p2p/base/sessiondescription.h"
 #include "webrtc/rtc_base/basictypes.h"
 #include "webrtc/rtc_base/buffer.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/sslstreamadapter.h"
 #include "webrtc/rtc_base/thread_checker.h"
 
diff --git a/webrtc/pc/webrtcsession.h b/webrtc/pc/webrtcsession.h
index 0f4d321eea..4d52a5c361 100644
--- a/webrtc/pc/webrtcsession.h
+++ b/webrtc/pc/webrtcsession.h
@@ -16,6 +16,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/statstypes.h"
 #include "webrtc/call/call.h"
@@ -24,7 +25,6 @@
 #include "webrtc/pc/datachannel.h"
 #include "webrtc/pc/mediasession.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/sigslot.h"
 #include "webrtc/rtc_base/sslidentity.h"
 #include "webrtc/rtc_base/thread.h"
diff --git a/webrtc/rtc_base/BUILD.gn b/webrtc/rtc_base/BUILD.gn
index 1914b27439..6676fdf2ea 100644
--- a/webrtc/rtc_base/BUILD.gn
+++ b/webrtc/rtc_base/BUILD.gn
@@ -137,8 +137,6 @@ rtc_static_library("rtc_base_approved") {
     "mod_ops.h",
     "moving_max_counter.h",
     "onetimeevent.h",
-    "optional.cc",
-    "optional.h",
     "pathutils.cc",
     "pathutils.h",
     "platform_file.cc",
@@ -436,6 +434,7 @@ rtc_static_library("rtc_base") {
   defines = []
   deps = [
     "..:webrtc_common",
+    "../api:optional",
   ]
   public_deps = [
     ":rtc_base_approved",
@@ -855,7 +854,6 @@ if (rtc_include_tests) {
       "mod_ops_unittest.cc",
       "moving_max_counter_unittest.cc",
       "onetimeevent_unittest.cc",
-      "optional_unittest.cc",
       "pathutils_unittest.cc",
       "platform_thread_unittest.cc",
       "random_unittest.cc",
@@ -1047,6 +1045,7 @@ if (rtc_include_tests) {
     deps = [
       ":rtc_base_tests_main",
       ":rtc_base_tests_utils",
+      "../api:optional",
       "../test:test_support",
     ]
     public_deps = [
diff --git a/webrtc/rtc_base/moving_max_counter.h b/webrtc/rtc_base/moving_max_counter.h
index 907e9cf298..190b7b85c2 100644
--- a/webrtc/rtc_base/moving_max_counter.h
+++ b/webrtc/rtc_base/moving_max_counter.h
@@ -17,9 +17,9 @@
 #include <limits>
 #include <utility>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace rtc {
 
diff --git a/webrtc/rtc_base/optional.h b/webrtc/rtc_base/optional.h
index ca3208e4d5..b70774684c 100644
--- a/webrtc/rtc_base/optional.h
+++ b/webrtc/rtc_base/optional.h
@@ -8,402 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+// This header is for backwards compatibility only, and will be removed soon.
+// Include webrtc/api/optional.h instead.
+
 #ifndef WEBRTC_RTC_BASE_OPTIONAL_H_
 #define WEBRTC_RTC_BASE_OPTIONAL_H_
 
-#include <algorithm>
-#include <memory>
-#include <utility>
-
-#ifdef UNIT_TEST
-#include <iomanip>
-#include <ostream>
-#endif  // UNIT_TEST
-
-#include "webrtc/api/array_view.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/sanitizer.h"
-
-namespace rtc {
-
-namespace optional_internal {
-
-#if RTC_HAS_ASAN
-
-// This is a non-inlined function. The optimizer can't see inside it.  It
-// prevents the compiler from generating optimized code that reads value_ even
-// if it is unset. Although safe, this causes memory sanitizers to complain.
-void* FunctionThatDoesNothingImpl(void*);
-
-template <typename T>
-inline T* FunctionThatDoesNothing(T* x) {
-  return reinterpret_cast<T*>(
-      FunctionThatDoesNothingImpl(reinterpret_cast<void*>(x)));
-}
-
-#else
-
-template <typename T>
-inline T* FunctionThatDoesNothing(T* x) { return x; }
-
-#endif
-
-}  // namespace optional_internal
-
-// Simple std::optional-wannabe. It either contains a T or not.
-//
-// A moved-from Optional<T> may only be destroyed, and assigned to if T allows
-// being assigned to after having been moved from. Specifically, you may not
-// assume that it just doesn't contain a value anymore.
-//
-// Examples of good places to use Optional:
-//
-// - As a class or struct member, when the member doesn't always have a value:
-//     struct Prisoner {
-//       std::string name;
-//       Optional<int> cell_number;  // Empty if not currently incarcerated.
-//     };
-//
-// - As a return value for functions that may fail to return a value on all
-//   allowed inputs. For example, a function that searches an array might
-//   return an Optional<size_t> (the index where it found the element, or
-//   nothing if it didn't find it); and a function that parses numbers might
-//   return Optional<double> (the parsed number, or nothing if parsing failed).
-//
-// Examples of bad places to use Optional:
-//
-// - As a return value for functions that may fail because of disallowed
-//   inputs. For example, a string length function should not return
-//   Optional<size_t> so that it can return nothing in case the caller passed
-//   it a null pointer; the function should probably use RTC_[D]CHECK instead,
-//   and return plain size_t.
-//
-// - As a return value for functions that may fail to return a value on all
-//   allowed inputs, but need to tell the caller what went wrong. Returning
-//   Optional<double> when parsing a single number as in the example above
-//   might make sense, but any larger parse job is probably going to need to
-//   tell the caller what the problem was, not just that there was one.
-//
-// - As a non-mutable function argument. When you want to pass a value of a
-//   type T that can fail to be there, const T* is almost always both fastest
-//   and cleanest. (If you're *sure* that the the caller will always already
-//   have an Optional<T>, const Optional<T>& is slightly faster than const T*,
-//   but this is a micro-optimization. In general, stick to const T*.)
-//
-// TODO(kwiberg): Get rid of this class when the standard library has
-// std::optional (and we're allowed to use it).
-template <typename T>
-class Optional final {
- public:
-  // Construct an empty Optional.
-  Optional() : has_value_(false), empty_('\0') {
-    PoisonValue();
-  }
-
-  // Construct an Optional that contains a value.
-  explicit Optional(const T& value) : has_value_(true) {
-    new (&value_) T(value);
-  }
-  explicit Optional(T&& value) : has_value_(true) {
-    new (&value_) T(std::move(value));
-  }
-
-  // Copy constructor: copies the value from m if it has one.
-  Optional(const Optional& m) : has_value_(m.has_value_) {
-    if (has_value_)
-      new (&value_) T(m.value_);
-    else
-      PoisonValue();
-  }
-
-  // Move constructor: if m has a value, moves the value from m, leaving m
-  // still in a state where it has a value, but a moved-from one (the
-  // properties of which depends on T; the only general guarantee is that we
-  // can destroy m).
-  Optional(Optional&& m) : has_value_(m.has_value_) {
-    if (has_value_)
-      new (&value_) T(std::move(m.value_));
-    else
-      PoisonValue();
-  }
-
-  ~Optional() {
-    if (has_value_)
-      value_.~T();
-    else
-      UnpoisonValue();
-  }
-
-  // Copy assignment. Uses T's copy assignment if both sides have a value, T's
-  // copy constructor if only the right-hand side has a value.
-  Optional& operator=(const Optional& m) {
-    if (m.has_value_) {
-      if (has_value_) {
-        value_ = m.value_;  // T's copy assignment.
-      } else {
-        UnpoisonValue();
-        new (&value_) T(m.value_);  // T's copy constructor.
-        has_value_ = true;
-      }
-    } else {
-      reset();
-    }
-    return *this;
-  }
-
-  // Move assignment. Uses T's move assignment if both sides have a value, T's
-  // move constructor if only the right-hand side has a value. The state of m
-  // after it's been moved from is as for the move constructor.
-  Optional& operator=(Optional&& m) {
-    if (m.has_value_) {
-      if (has_value_) {
-        value_ = std::move(m.value_);  // T's move assignment.
-      } else {
-        UnpoisonValue();
-        new (&value_) T(std::move(m.value_));  // T's move constructor.
-        has_value_ = true;
-      }
-    } else {
-      reset();
-    }
-    return *this;
-  }
-
-  // Swap the values if both m1 and m2 have values; move the value if only one
-  // of them has one.
-  friend void swap(Optional& m1, Optional& m2) {
-    if (m1.has_value_) {
-      if (m2.has_value_) {
-        // Both have values: swap.
-        using std::swap;
-        swap(m1.value_, m2.value_);
-      } else {
-        // Only m1 has a value: move it to m2.
-        m2.UnpoisonValue();
-        new (&m2.value_) T(std::move(m1.value_));
-        m1.value_.~T();  // Destroy the moved-from value.
-        m1.has_value_ = false;
-        m2.has_value_ = true;
-        m1.PoisonValue();
-      }
-    } else if (m2.has_value_) {
-      // Only m2 has a value: move it to m1.
-      m1.UnpoisonValue();
-      new (&m1.value_) T(std::move(m2.value_));
-      m2.value_.~T();  // Destroy the moved-from value.
-      m1.has_value_ = true;
-      m2.has_value_ = false;
-      m2.PoisonValue();
-    }
-  }
-
-  // Destroy any contained value. Has no effect if we have no value.
-  void reset() {
-    if (!has_value_)
-      return;
-    value_.~T();
-    has_value_ = false;
-    PoisonValue();
-  }
-
-  template <class... Args>
-  void emplace(Args&&... args) {
-    if (has_value_)
-      value_.~T();
-    else
-      UnpoisonValue();
-    new (&value_) T(std::forward<Args>(args)...);
-    has_value_ = true;
-  }
-
-  // Conversion to bool to test if we have a value.
-  explicit operator bool() const { return has_value_; }
-  bool has_value() const { return has_value_; }
-
-  // Dereferencing. Only allowed if we have a value.
-  const T* operator->() const {
-    RTC_DCHECK(has_value_);
-    return &value_;
-  }
-  T* operator->() {
-    RTC_DCHECK(has_value_);
-    return &value_;
-  }
-  const T& operator*() const {
-    RTC_DCHECK(has_value_);
-    return value_;
-  }
-  T& operator*() {
-    RTC_DCHECK(has_value_);
-    return value_;
-  }
-  const T& value() const {
-    RTC_DCHECK(has_value_);
-    return value_;
-  }
-  T& value() {
-    RTC_DCHECK(has_value_);
-    return value_;
-  }
-
-  // Dereference with a default value in case we don't have a value.
-  const T& value_or(const T& default_val) const {
-    // The no-op call prevents the compiler from generating optimized code that
-    // reads value_ even if !has_value_, but only if FunctionThatDoesNothing is
-    // not completely inlined; see its declaration.).
-    return has_value_ ? *optional_internal::FunctionThatDoesNothing(&value_)
-                      : default_val;
-  }
-
-  // Dereference and move value.
-  T MoveValue() {
-    RTC_DCHECK(has_value_);
-    return std::move(value_);
-  }
-
-  // Equality tests. Two Optionals are equal if they contain equivalent values,
-  // or if they're both empty.
-  friend bool operator==(const Optional& m1, const Optional& m2) {
-    return m1.has_value_ && m2.has_value_ ? m1.value_ == m2.value_
-                                          : m1.has_value_ == m2.has_value_;
-  }
-  friend bool operator==(const Optional& opt, const T& value) {
-    return opt.has_value_ && opt.value_ == value;
-  }
-  friend bool operator==(const T& value, const Optional& opt) {
-    return opt.has_value_ && value == opt.value_;
-  }
-
-  friend bool operator!=(const Optional& m1, const Optional& m2) {
-    return m1.has_value_ && m2.has_value_ ? m1.value_ != m2.value_
-                                          : m1.has_value_ != m2.has_value_;
-  }
-  friend bool operator!=(const Optional& opt, const T& value) {
-    return !opt.has_value_ || opt.value_ != value;
-  }
-  friend bool operator!=(const T& value, const Optional& opt) {
-    return !opt.has_value_ || value != opt.value_;
-  }
-
- private:
-  // Tell sanitizers that value_ shouldn't be touched.
-  void PoisonValue() {
-    rtc::AsanPoison(rtc::MakeArrayView(&value_, 1));
-    rtc::MsanMarkUninitialized(rtc::MakeArrayView(&value_, 1));
-  }
-
-  // Tell sanitizers that value_ is OK to touch again.
-  void UnpoisonValue() {
-    rtc::AsanUnpoison(rtc::MakeArrayView(&value_, 1));
-  }
-
-  bool has_value_;  // True iff value_ contains a live value.
-  union {
-    // empty_ exists only to make it possible to initialize the union, even when
-    // it doesn't contain any data. If the union goes uninitialized, it may
-    // trigger compiler warnings.
-    char empty_;
-    // By placing value_ in a union, we get to manage its construction and
-    // destruction manually: the Optional constructors won't automatically
-    // construct it, and the Optional destructor won't automatically destroy
-    // it. Basically, this just allocates a properly sized and aligned block of
-    // memory in which we can manually put a T with placement new.
-    T value_;
-  };
-};
-
-#ifdef UNIT_TEST
-namespace optional_internal {
-
-// Checks if there's a valid PrintTo(const T&, std::ostream*) call for T.
-template <typename T>
-struct HasPrintTo {
- private:
-  struct No {};
-
-  template <typename T2>
-  static auto Test(const T2& obj)
-      -> decltype(PrintTo(obj, std::declval<std::ostream*>()));
-
-  template <typename>
-  static No Test(...);
-
- public:
-  static constexpr bool value =
-      !std::is_same<decltype(Test<T>(std::declval<const T&>())), No>::value;
-};
-
-// Checks if there's a valid operator<<(std::ostream&, const T&) call for T.
-template <typename T>
-struct HasOstreamOperator {
- private:
-  struct No {};
-
-  template <typename T2>
-  static auto Test(const T2& obj)
-      -> decltype(std::declval<std::ostream&>() << obj);
-
-  template <typename>
-  static No Test(...);
-
- public:
-  static constexpr bool value =
-      !std::is_same<decltype(Test<T>(std::declval<const T&>())), No>::value;
-};
-
-// Prefer using PrintTo to print the object.
-template <typename T>
-typename std::enable_if<HasPrintTo<T>::value, void>::type OptionalPrintToHelper(
-    const T& value,
-    std::ostream* os) {
-  PrintTo(value, os);
-}
-
-// Fall back to operator<<(std::ostream&, ...) if it exists.
-template <typename T>
-typename std::enable_if<HasOstreamOperator<T>::value && !HasPrintTo<T>::value,
-                        void>::type
-OptionalPrintToHelper(const T& value, std::ostream* os) {
-  *os << value;
-}
-
-inline void OptionalPrintObjectBytes(const unsigned char* bytes,
-                                     size_t size,
-                                     std::ostream* os) {
-  *os << "<optional with " << size << "-byte object [";
-  for (size_t i = 0; i != size; ++i) {
-    *os << (i == 0 ? "" : ((i & 1) ? "-" : " "));
-    *os << std::hex << std::setw(2) << std::setfill('0')
-        << static_cast<int>(bytes[i]);
-  }
-  *os << "]>";
-}
-
-// As a final back-up, just print the contents of the objcets byte-wise.
-template <typename T>
-typename std::enable_if<!HasOstreamOperator<T>::value && !HasPrintTo<T>::value,
-                        void>::type
-OptionalPrintToHelper(const T& value, std::ostream* os) {
-  OptionalPrintObjectBytes(reinterpret_cast<const unsigned char*>(&value),
-                           sizeof(value), os);
-}
-
-}  // namespace optional_internal
-
-// PrintTo is used by gtest to print out the results of tests. We want to ensure
-// the object contained in an Optional can be printed out if it's set, while
-// avoiding touching the object's storage if it is undefined.
-template <typename T>
-void PrintTo(const rtc::Optional<T>& opt, std::ostream* os) {
-  if (opt) {
-    optional_internal::OptionalPrintToHelper(*opt, os);
-  } else {
-    *os << "<empty optional>";
-  }
-}
-
-#endif  // UNIT_TEST
-
-}  // namespace rtc
+#include "webrtc/api/optional.h"
 
 #endif  // WEBRTC_RTC_BASE_OPTIONAL_H_
diff --git a/webrtc/rtc_base/rate_statistics.h b/webrtc/rtc_base/rate_statistics.h
index 7c4daeaa6e..87f6c35bdf 100644
--- a/webrtc/rtc_base/rate_statistics.h
+++ b/webrtc/rtc_base/rate_statistics.h
@@ -13,7 +13,7 @@
 
 #include <memory>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/rtc_base/rtccertificategenerator.h b/webrtc/rtc_base/rtccertificategenerator.h
index 272a5d61f0..4e0437c58a 100644
--- a/webrtc/rtc_base/rtccertificategenerator.h
+++ b/webrtc/rtc_base/rtccertificategenerator.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_RTC_BASE_RTCCERTIFICATEGENERATOR_H_
 #define WEBRTC_RTC_BASE_RTCCERTIFICATEGENERATOR_H_
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/rtc_base/refcount.h"
 #include "webrtc/rtc_base/rtccertificate.h"
 #include "webrtc/rtc_base/scoped_ref_ptr.h"
diff --git a/webrtc/rtc_base/rtccertificategenerator_unittest.cc b/webrtc/rtc_base/rtccertificategenerator_unittest.cc
index df820d93cc..829a6ee9a3 100644
--- a/webrtc/rtc_base/rtccertificategenerator_unittest.cc
+++ b/webrtc/rtc_base/rtccertificategenerator_unittest.cc
@@ -12,10 +12,10 @@
 
 #include <memory>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/gunit.h"
 #include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/thread.h"
 
 namespace rtc {
diff --git a/webrtc/rtc_base/string_to_number.h b/webrtc/rtc_base/string_to_number.h
index c61d1ed73c..93bd850f63 100644
--- a/webrtc/rtc_base/string_to_number.h
+++ b/webrtc/rtc_base/string_to_number.h
@@ -14,7 +14,7 @@
 #include <string>
 #include <limits>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace rtc {
 
diff --git a/webrtc/rtc_tools/network_tester/BUILD.gn b/webrtc/rtc_tools/network_tester/BUILD.gn
index 02bbc3182b..cff453d808 100644
--- a/webrtc/rtc_tools/network_tester/BUILD.gn
+++ b/webrtc/rtc_tools/network_tester/BUILD.gn
@@ -41,6 +41,7 @@ if (rtc_enable_protobuf) {
     deps = [
       ":network_tester_config_proto",
       ":network_tester_packet_proto",
+      "../../api:optional",
       "../../p2p",
       "../../rtc_base:protobuf_utils",
       "../../rtc_base:rtc_base_approved",
diff --git a/webrtc/rtc_tools/network_tester/config_reader.h b/webrtc/rtc_tools/network_tester/config_reader.h
index 83fbdebc6a..ae4348f483 100644
--- a/webrtc/rtc_tools/network_tester/config_reader.h
+++ b/webrtc/rtc_tools/network_tester/config_reader.h
@@ -14,8 +14,8 @@
 #include <fstream>
 #include <string>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 
 #include "webrtc/rtc_base/ignore_wundef.h"
 
diff --git a/webrtc/sdk/BUILD.gn b/webrtc/sdk/BUILD.gn
index ff97082229..086ce49f47 100644
--- a/webrtc/sdk/BUILD.gn
+++ b/webrtc/sdk/BUILD.gn
@@ -183,6 +183,7 @@ if (is_ios || is_mac) {
         ":common_objc",
         ":videotracksource_objc",
         "../api:libjingle_peerconnection_api",
+        "../api:optional",
         "../common_video",
         "../media:rtc_media_base",
         "../rtc_base:rtc_base",
diff --git a/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.mm b/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.mm
index 900f2840ea..14301ba3d5 100644
--- a/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.mm
+++ b/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.mm
@@ -20,7 +20,7 @@
 #import "RTCShader.h"
 #import "WebRTC/RTCLogging.h"
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 static const int kYTextureUnit = 0;
 static const int kUTextureUnit = 1;
diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn
index 0858468e5c..9ec02902d9 100644
--- a/webrtc/test/BUILD.gn
+++ b/webrtc/test/BUILD.gn
@@ -60,6 +60,7 @@ rtc_source_set("video_test_common") {
 
   deps = [
     "..:webrtc_common",
+    "../api:optional",
     "../api/video_codecs:video_codecs_api",
     "../call:video_stream_api",
     "../common_video",
@@ -375,6 +376,7 @@ rtc_source_set("fileutils") {
   ]
   deps = [
     "..:webrtc_common",
+    "../api:optional",
     "../rtc_base:rtc_base_approved",
   ]
   if (is_ios) {
@@ -437,6 +439,7 @@ rtc_source_set("fileutils_unittests") {
   deps = [
     ":fileutils",
     ":test_support",
+    "../api:optional",
     "../rtc_base:rtc_base_approved",
     "//testing/gmock",
     "//testing/gtest",
diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn
index e9d771cc73..6035cd93af 100644
--- a/webrtc/test/fuzzers/BUILD.gn
+++ b/webrtc/test/fuzzers/BUILD.gn
@@ -207,6 +207,7 @@ rtc_static_library("audio_decoder_fuzzer") {
   ]
   deps = [
     "../..:webrtc_common",
+    "../../api:optional",
     "../../api/audio_codecs:audio_codecs_api",
     "../../modules/rtp_rtcp",
     "../../rtc_base:rtc_base_approved",
@@ -369,6 +370,7 @@ webrtc_fuzzer_test("audio_processing_fuzzer") {
     "audio_processing_fuzzer_configs.cc",
   ]
   deps = [
+    "../../api:optional",
     "../../modules:module_api",
     "../../modules/audio_processing",
     "../../rtc_base:rtc_base_approved",
diff --git a/webrtc/test/fuzzers/audio_decoder_fuzzer.cc b/webrtc/test/fuzzers/audio_decoder_fuzzer.cc
index ff4478dca9..441b12b5f2 100644
--- a/webrtc/test/fuzzers/audio_decoder_fuzzer.cc
+++ b/webrtc/test/fuzzers/audio_decoder_fuzzer.cc
@@ -13,9 +13,9 @@
 #include <limits>
 
 #include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 namespace {
diff --git a/webrtc/test/fuzzers/audio_processing_fuzzer_configs.cc b/webrtc/test/fuzzers/audio_processing_fuzzer_configs.cc
index f9814ec306..21810ad83a 100644
--- a/webrtc/test/fuzzers/audio_processing_fuzzer_configs.cc
+++ b/webrtc/test/fuzzers/audio_processing_fuzzer_configs.cc
@@ -11,7 +11,7 @@
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/test/fuzzers/audio_processing_fuzzer.h"
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/test/testsupport/fileutils.h b/webrtc/test/testsupport/fileutils.h
index 570d7b4d20..5ea7d31705 100644
--- a/webrtc/test/testsupport/fileutils.h
+++ b/webrtc/test/testsupport/fileutils.h
@@ -16,7 +16,7 @@
 #include <string>
 #include <vector>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 namespace test {
diff --git a/webrtc/test/testsupport/fileutils_unittest.cc b/webrtc/test/testsupport/fileutils_unittest.cc
index bdabb8470a..fb780be56e 100644
--- a/webrtc/test/testsupport/fileutils_unittest.cc
+++ b/webrtc/test/testsupport/fileutils_unittest.cc
@@ -17,8 +17,8 @@
 #include <list>
 #include <string>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/pathutils.h"
 #include "webrtc/test/gtest.h"
 
diff --git a/webrtc/test/video_capturer.h b/webrtc/test/video_capturer.h
index 2535370f8e..deeb5c8c2a 100644
--- a/webrtc/test/video_capturer.h
+++ b/webrtc/test/video_capturer.h
@@ -14,12 +14,12 @@
 
 #include <memory>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/video/i420_buffer.h"
 #include "webrtc/api/video/video_frame.h"
 #include "webrtc/media/base/videoadapter.h"
 #include "webrtc/media/base/videosourceinterface.h"
 #include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace cricket {
 class VideoAdapter;
diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn
index c93ef8cc40..7756ff416d 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -55,6 +55,7 @@ rtc_static_library("video") {
 
   deps = [
     "..:webrtc_common",
+    "../api:optional",
     "../api:transport_api",
     "../api/video_codecs:video_codecs_api",
     "../call:call_interfaces",
@@ -93,6 +94,7 @@ if (rtc_include_tests) {
       "video_quality_test.h",
     ]
     deps = [
+      "../api:optional",
       "../call:call_interfaces",
       "../common_video",
       "../logging:rtc_event_log_api",
@@ -265,6 +267,7 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":video",
+      "../api:optional",
       "../api:video_frame_api",
       "../api/video_codecs:video_codecs_api",
       "../call:call_interfaces",
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index a30b08a320..644319cf79 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -15,6 +15,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/api/video_codecs/video_encoder.h"
 #include "webrtc/call/call.h"
 #include "webrtc/common_video/include/frame_callback.h"
@@ -38,7 +39,6 @@
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/event.h"
 #include "webrtc/rtc_base/file.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/rtc_base/random.h"
 #include "webrtc/rtc_base/rate_limiter.h"
diff --git a/webrtc/video/overuse_frame_detector.h b/webrtc/video/overuse_frame_detector.h
index 8bf1a78aed..a5eef51020 100644
--- a/webrtc/video/overuse_frame_detector.h
+++ b/webrtc/video/overuse_frame_detector.h
@@ -14,10 +14,10 @@
 #include <list>
 #include <memory>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/video_coding/utility/quality_scaler.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/numerics/exp_filter.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/sequenced_task_checker.h"
 #include "webrtc/rtc_base/task_queue.h"
 #include "webrtc/rtc_base/thread_annotations.h"
diff --git a/webrtc/video/quality_threshold.h b/webrtc/video/quality_threshold.h
index ee13f8b810..a142d4e716 100644
--- a/webrtc/video/quality_threshold.h
+++ b/webrtc/video/quality_threshold.h
@@ -13,7 +13,7 @@
 
 #include <memory>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
diff --git a/webrtc/video/receive_statistics_proxy.h b/webrtc/video/receive_statistics_proxy.h
index 140cdbd407..45424d980f 100644
--- a/webrtc/video/receive_statistics_proxy.h
+++ b/webrtc/video/receive_statistics_proxy.h
@@ -14,13 +14,13 @@
 #include <map>
 #include <string>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/call/video_receive_stream.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/rtc_base/criticalsection.h"
 #include "webrtc/rtc_base/moving_max_counter.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/rate_statistics.h"
 #include "webrtc/rtc_base/ratetracker.h"
 #include "webrtc/rtc_base/thread_annotations.h"
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 2aa6295ccf..8224377627 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -18,6 +18,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/call/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
@@ -37,7 +38,6 @@
 #include "webrtc/rtc_base/format_macros.h"
 #include "webrtc/rtc_base/logging.h"
 #include "webrtc/rtc_base/memory_usage.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/pathutils.h"
 #include "webrtc/rtc_base/platform_file.h"
 #include "webrtc/rtc_base/ptr_util.h"
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index 6750c9e0de..74c90134b9 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -16,6 +16,7 @@
 #include <string>
 #include <utility>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/call/rtp_stream_receiver_controller_interface.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/h264/profile_level_id.h"
@@ -31,7 +32,6 @@
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/location.h"
 #include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/trace_event.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
index 9862579c85..809333b850 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -139,6 +139,7 @@ rtc_static_library("voice_engine") {
     "../api:audio_mixer_api",
     "../api:call_api",
     "../api:libjingle_peerconnection_api",
+    "../api:optional",
     "../api:transport_api",
     "../api/audio_codecs:audio_codecs_api",
     "../api/audio_codecs:builtin_audio_decoder_factory",
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 884845ef83..0d71a41d82 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -16,6 +16,7 @@
 #include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/call/audio_sink.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
@@ -29,7 +30,6 @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/rtc_base/criticalsection.h"
 #include "webrtc/rtc_base/event.h"
-#include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/thread_checker.h"
 #include "webrtc/voice_engine/audio_level.h"
 #include "webrtc/voice_engine/file_player.h"
diff --git a/webrtc/voice_engine/transport_feedback_packet_loss_tracker.h b/webrtc/voice_engine/transport_feedback_packet_loss_tracker.h
index 987ba25832..8168216f69 100644
--- a/webrtc/voice_engine/transport_feedback_packet_loss_tracker.h
+++ b/webrtc/voice_engine/transport_feedback_packet_loss_tracker.h
@@ -13,8 +13,8 @@
 
 #include <map>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {