Move StrongAlias to rtc_base

It's useful for other parts of WebRTC and there is no real reason why
it should be located in net/dcsctp.

Bug: None
Change-Id: Iccaed4e943e21ddaea8603182d693114b2da9f6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232606
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35055}

net/dcsctp/common/BUILD.gn

@@ -10,7 +10,7 @@ import("../../../webrtc.gni")
 
 rtc_source_set("internal_types") {
   deps = [
-    "../public:strong_alias",
+    "../../../rtc_base:rtc_base_approved",
     "../public:types",
   ]
   sources = [ "internal_types.h" ]

net/dcsctp/common/internal_types.h

@@ -13,31 +13,32 @@
 #include <functional>
 #include <utility>
 
-#include "net/dcsctp/public/strong_alias.h"
 #include "net/dcsctp/public/types.h"
+#include "rtc_base/strong_alias.h"
 
 namespace dcsctp {
 
 // Stream Sequence Number (SSN)
-using SSN = StrongAlias<class SSNTag, uint16_t>;
+using SSN = webrtc::StrongAlias<class SSNTag, uint16_t>;
 
 // Message Identifier (MID)
-using MID = StrongAlias<class MIDTag, uint32_t>;
+using MID = webrtc::StrongAlias<class MIDTag, uint32_t>;
 
 // Fragment Sequence Number (FSN)
-using FSN = StrongAlias<class FSNTag, uint32_t>;
+using FSN = webrtc::StrongAlias<class FSNTag, uint32_t>;
 
 // Transmission Sequence Number (TSN)
-using TSN = StrongAlias<class TSNTag, uint32_t>;
+using TSN = webrtc::StrongAlias<class TSNTag, uint32_t>;
 
 // Reconfiguration Request Sequence Number
-using ReconfigRequestSN = StrongAlias<class ReconfigRequestSNTag, uint32_t>;
+using ReconfigRequestSN =
+    webrtc::StrongAlias<class ReconfigRequestSNTag, uint32_t>;
 
 // Verification Tag, used for packet validation.
-using VerificationTag = StrongAlias<class VerificationTagTag, uint32_t>;
+using VerificationTag = webrtc::StrongAlias<class VerificationTagTag, uint32_t>;
 
 // Tie Tag, used as a nonce when connecting.
-using TieTag = StrongAlias<class TieTagTag, uint64_t>;
+using TieTag = webrtc::StrongAlias<class TieTagTag, uint64_t>;
 
 }  // namespace dcsctp
 #endif  // NET_DCSCTP_COMMON_INTERNAL_TYPES_H_

net/dcsctp/common/sequence_numbers.h

@@ -26,7 +26,7 @@ namespace dcsctp {
 // need to be unwrapped in order, as long as the difference to the previous one
 // is not larger than half the range of the wrapped sequence number.
 //
-// The WrappedType must be a StrongAlias type.
+// The WrappedType must be a webrtc::StrongAlias type.
 template <typename WrappedType>
 class UnwrappedSequenceNumber {
  public:

net/dcsctp/common/sequence_numbers_test.cc

@@ -14,7 +14,7 @@
 namespace dcsctp {
 namespace {
 
-using Wrapped = StrongAlias<class WrappedTag, uint16_t>;
+using Wrapped = webrtc::StrongAlias<class WrappedTag, uint16_t>;
 using TestSequence = UnwrappedSequenceNumber<Wrapped>;
 
 TEST(SequenceNumbersTest, SimpleUnwrapping) {

net/dcsctp/packet/chunk/data_common.h

@@ -24,7 +24,7 @@ namespace dcsctp {
 class AnyDataChunk : public Chunk {
  public:
   // Represents the "immediate ack" flag on DATA/I-DATA, from RFC7053.
-  using ImmediateAckFlag = StrongAlias<class ImmediateAckFlagTag, bool>;
+  using ImmediateAckFlag = webrtc::StrongAlias<class ImmediateAckFlagTag, bool>;
 
   // Data chunk options.
   // See https://tools.ietf.org/html/rfc4960#section-3.3.1

net/dcsctp/packet/data.h

@@ -34,11 +34,11 @@ namespace dcsctp {
 struct Data {
   // Indicates if a chunk is the first in a fragmented message and maps to the
   // "beginning" flag in DATA/I-DATA chunk.
-  using IsBeginning = StrongAlias<class IsBeginningTag, bool>;
+  using IsBeginning = webrtc::StrongAlias<class IsBeginningTag, bool>;
 
   // Indicates if a chunk is the last in a fragmented message  and maps to the
   // "end" flag in DATA/I-DATA chunk.
-  using IsEnd = StrongAlias<class IsEndTag, bool>;
+  using IsEnd = webrtc::StrongAlias<class IsEndTag, bool>;
 
   Data(StreamID stream_id,
        SSN ssn,

net/dcsctp/public/BUILD.gn

@@ -8,13 +8,9 @@
 
 import("../../../webrtc.gni")
 
-rtc_source_set("strong_alias") {
-  sources = [ "strong_alias.h" ]
-}
-
 rtc_source_set("types") {
   deps = [
-    ":strong_alias",
+    "../../../rtc_base:rtc_base_approved",
     "../../../api:array_view",
   ]
   sources = [
@@ -27,7 +23,7 @@ rtc_source_set("types") {
 
 rtc_source_set("socket") {
   deps = [
-    ":strong_alias",
+    "../../../rtc_base:rtc_base_approved",
     ":types",
     "../../../api:array_view",
     "../../../rtc_base",
@@ -91,17 +87,14 @@ if (rtc_include_tests) {
 
     deps = [
       ":mocks",
-      ":strong_alias",
       ":types",
       "../../../rtc_base:checks",
       "../../../rtc_base:gunit_helpers",
       "../../../rtc_base:rtc_base_approved",
-      "../../../rtc_base/containers:flat_map",
       "../../../test:test_support",
     ]
     sources = [
       "mock_dcsctp_socket_test.cc",
-      "strong_alias_test.cc",
       "types_test.cc",
     ]
   }

net/dcsctp/public/dcsctp_handover_state.h

@@ -14,7 +14,7 @@
 #include <string>
 #include <vector>
 
-#include "net/dcsctp/public/strong_alias.h"
+#include "rtc_base/strong_alias.h"
 
 namespace dcsctp {
 
@@ -101,15 +101,15 @@ enum class HandoverUnreadinessReason : uint32_t {
 // state in which a snapshot of the state can be made by
 // `GetHandoverStateAndClose()`.
 class HandoverReadinessStatus
-    : public StrongAlias<class HandoverReadinessStatusTag, uint32_t> {
+    : public webrtc::StrongAlias<class HandoverReadinessStatusTag, uint32_t> {
  public:
   // Constructs an empty `HandoverReadinessStatus` which represents ready state.
   constexpr HandoverReadinessStatus()
-      : StrongAlias<class HandoverReadinessStatusTag, uint32_t>(0) {}
+      : webrtc::StrongAlias<class HandoverReadinessStatusTag, uint32_t>(0) {}
   // Constructs status object that contains a single reason for not being
   // handover ready.
   constexpr explicit HandoverReadinessStatus(HandoverUnreadinessReason reason)
-      : StrongAlias<class HandoverReadinessStatusTag, uint32_t>(
+      : webrtc::StrongAlias<class HandoverReadinessStatusTag, uint32_t>(
             static_cast<uint32_t>(reason)) {}
 
   // Convenience methods

net/dcsctp/public/types.h

@@ -14,28 +14,28 @@
 #include <cstdint>
 #include <limits>
 
-#include "net/dcsctp/public/strong_alias.h"
+#include "rtc_base/strong_alias.h"
 
 namespace dcsctp {
 
 // Stream Identifier
-using StreamID = StrongAlias<class StreamIDTag, uint16_t>;
+using StreamID = webrtc::StrongAlias<class StreamIDTag, uint16_t>;
 
 // Payload Protocol Identifier (PPID)
-using PPID = StrongAlias<class PPIDTag, uint32_t>;
+using PPID = webrtc::StrongAlias<class PPIDTag, uint32_t>;
 
 // Timeout Identifier
-using TimeoutID = StrongAlias<class TimeoutTag, uint64_t>;
+using TimeoutID = webrtc::StrongAlias<class TimeoutTag, uint64_t>;
 
 // Indicates if a message is allowed to be received out-of-order compared to
 // other messages on the same stream.
-using IsUnordered = StrongAlias<class IsUnorderedTag, bool>;
+using IsUnordered = webrtc::StrongAlias<class IsUnorderedTag, bool>;
 
 // Duration, as milliseconds. Overflows after 24 days.
-class DurationMs : public StrongAlias<class DurationMsTag, int32_t> {
+class DurationMs : public webrtc::StrongAlias<class DurationMsTag, int32_t> {
  public:
   constexpr explicit DurationMs(const UnderlyingType& v)
-      : StrongAlias<class DurationMsTag, int32_t>(v) {}
+      : webrtc::StrongAlias<class DurationMsTag, int32_t>(v) {}
 
   // Convenience methods for working with time.
   constexpr DurationMs& operator+=(DurationMs d) {
@@ -72,10 +72,10 @@ constexpr inline int32_t operator/(DurationMs lhs, DurationMs rhs) {
 }
 
 // Represents time, in milliseconds since a client-defined epoch.
-class TimeMs : public StrongAlias<class TimeMsTag, int64_t> {
+class TimeMs : public webrtc::StrongAlias<class TimeMsTag, int64_t> {
  public:
   constexpr explicit TimeMs(const UnderlyingType& v)
-      : StrongAlias<class TimeMsTag, int64_t>(v) {}
+      : webrtc::StrongAlias<class TimeMsTag, int64_t>(v) {}
 
   // Convenience methods for working with time.
   constexpr TimeMs& operator+=(DurationMs d) {

net/dcsctp/timer/BUILD.gn

@@ -17,7 +17,6 @@ rtc_library("timer") {
     "../../../rtc_base/containers:flat_map",
     "../../../rtc_base/containers:flat_set",
     "../public:socket",
-    "../public:strong_alias",
     "../public:types",
   ]
   sources = [
@@ -41,7 +40,6 @@ rtc_library("task_queue_timeout") {
     "../../../rtc_base/task_utils:pending_task_safety_flag",
     "../../../rtc_base/task_utils:to_queued_task",
     "../public:socket",
-    "../public:strong_alias",
     "../public:types",
   ]
   sources = [

net/dcsctp/timer/timer.h

@@ -21,13 +21,13 @@
 
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
-#include "net/dcsctp/public/strong_alias.h"
 #include "net/dcsctp/public/timeout.h"
+#include "rtc_base/strong_alias.h"
 
 namespace dcsctp {
 
-using TimerID = StrongAlias<class TimerIDTag, uint32_t>;
+using TimerID = webrtc::StrongAlias<class TimerIDTag, uint32_t>;
-using TimerGeneration = StrongAlias<class TimerGenerationTag, uint32_t>;
+using TimerGeneration = webrtc::StrongAlias<class TimerGenerationTag, uint32_t>;
 
 enum class TimerBackoffAlgorithm {
   // The base duration will be used for any restart.

rtc_base/BUILD.gn

@@ -136,6 +136,7 @@ rtc_library("rtc_base_approved") {
     "rate_statistics.h",
     "rate_tracker.cc",
     "rate_tracker.h",
+    "strong_alias.h",
     "swap_queue.h",
     "timestamp_aligner.cc",
     "timestamp_aligner.h",
@@ -1371,6 +1372,7 @@ if (rtc_include_tests) {
         "string_utils_unittest.cc",
         "strings/string_builder_unittest.cc",
         "strings/string_format_unittest.cc",
+        "strong_alias_unittest.cc",
         "swap_queue_unittest.cc",
         "thread_annotations_unittest.cc",
         "time_utils_unittest.cc",
@@ -1404,6 +1406,7 @@ if (rtc_include_tests) {
         ":socket_server",
         ":stringutils",
         ":testclient",
+        "containers:flat_map",
         ":threading",
         "../api:array_view",
         "../api:scoped_refptr",

rtc_base/strong_alias.h

@@ -8,13 +8,13 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef NET_DCSCTP_PUBLIC_STRONG_ALIAS_H_
+#ifndef RTC_BASE_STRONG_ALIAS_H_
-#define NET_DCSCTP_PUBLIC_STRONG_ALIAS_H_
+#define RTC_BASE_STRONG_ALIAS_H_
 
 #include <type_traits>
 #include <utility>
 
-namespace dcsctp {
+namespace webrtc {
 
 // This is a copy of
 // https://source.chromium.org/chromium/chromium/src/+/main:base/types/strong_alias.h
@@ -71,6 +71,6 @@ class StrongAlias {
   UnderlyingType value_;
 };
 
-}  // namespace dcsctp
+}  // namespace webrtc
 
-#endif  // NET_DCSCTP_PUBLIC_STRONG_ALIAS_H_
+#endif  // RTC_BASE_STRONG_ALIAS_H_

rtc_base/strong_alias_unittest.cc

@@ -8,7 +8,7 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "net/dcsctp/public/strong_alias.h"
+#include "rtc_base/strong_alias.h"
 
 #include <cstdint>
 #include <map>
@@ -26,7 +26,7 @@
 // but adapted to use WebRTC's includes, remove unit tests that test the ostream
 // operator (it's removed in this port) and other adaptations to pass lint.
 
-namespace dcsctp {
+namespace webrtc {
 namespace {
 
 // For test correctnenss, it's important that these getters return lexically
@@ -359,4 +359,4 @@ TEST(StrongAliasTest, BooleansAreEvaluatedAsBooleans) {
   EXPECT_TRUE(*happy);
   EXPECT_FALSE(*sad);
 }
-}  // namespace dcsctp
+}  // namespace webrtc