Move optional.h to webrtc/api/

We use Optional in our public API, so its header should be in
webrtc/api/.

BUG=webrtc:8205

Review-Url: https://codereview.webrtc.org/3011943002
Cr-Commit-Position: refs/heads/master@{#19693}

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",

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",
   ]

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",

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 {
 

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 {
 

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 {

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 {

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 {
 

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",

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 {
 

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 {
 

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",

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 {
 

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 {
 

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",

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 {
 

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 {
 

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",

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 {
 

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 {
 

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 {
 

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 {
 

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",

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 {
 

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 {
 

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 {
 

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.

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
 

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_

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,
+std::ostream& operator<<(std::ostream& os, const MyPrintableType& mpt) {
-                         const MyPrintableType& mpt) {
   os << mpt.value;
   return os;
 }
 
-std::ostream& operator<<(std::ostream& os,
+std::ostream& operator<<(std::ostream& os, const MyOstreamPrintableType& mpt) {
-                         const MyOstreamPrintableType& mpt) {
   os << mpt.value;
   return os;
 }

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 {
 

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 {
 

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 {
 

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",

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 {

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",

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 {
 

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",

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"
 

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"
 

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"

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 {
 

webrtc/call/syncable.h

@@ -16,7 +16,7 @@
 
 #include <stdint.h>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 

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"

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",

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 {

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)

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",

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;

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;

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;

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;

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 {
 

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",

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 {

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"

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 {

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 {
 

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 {
 

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 {
 

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"
 

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",
   ]

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",

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"
 

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 {

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"
 

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 {
 

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 {
 

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"
 

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 {

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 {

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"
 

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"
 

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 {
 

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"
 

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 {

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 {

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 {

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 {

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 {

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",

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"

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",

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 {
 

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 {

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 {
 

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 {
 

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 {

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 {
 

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 {

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 {

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 {
 

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 {
 

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"
 

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 {
 

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 {
 

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 {
 

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 {
 

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 {
 

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 {