Remove RTC_DISALLOW_COPY_AND_ASSIGN more. Bug: webrtc:13555, webrtc:13082 Change-Id: I9c07708108da0a26f5e228384fd56cef4d1540b3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247300 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#35749}

diff --git a/api/audio/audio_frame.h b/api/audio/audio_frame.h
index 628a1ec..d5dcb5f 100644
--- a/api/audio/audio_frame.h
+++ b/api/audio/audio_frame.h

@@ -16,7 +16,6 @@
 
 #include "api/audio/channel_layout.h"
 #include "api/rtp_packet_infos.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -58,6 +57,9 @@
 
   AudioFrame();
 
+  AudioFrame(const AudioFrame&) = delete;
+  AudioFrame& operator=(const AudioFrame&) = delete;
+
   // Resets all members to their default state.
   void Reset();
   // Same as Reset(), but leaves mute state unchanged. Muting a frame requires
@@ -164,8 +166,6 @@
   // capture timestamp of a received frame is found in `packet_infos_`.
   // This timestamp MUST be based on the same clock as rtc::TimeMillis().
   absl::optional<int64_t> absolute_capture_timestamp_ms_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
 };
 
 }  // namespace webrtc

diff --git a/api/audio_codecs/audio_decoder.h b/api/audio_codecs/audio_decoder.h
index 336e384..4113874 100644
--- a/api/audio_codecs/audio_decoder.h
+++ b/api/audio_codecs/audio_decoder.h

@@ -20,7 +20,6 @@
 #include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -37,6 +36,9 @@
   AudioDecoder() = default;
   virtual ~AudioDecoder() = default;
 
+  AudioDecoder(const AudioDecoder&) = delete;
+  AudioDecoder& operator=(const AudioDecoder&) = delete;
+
   class EncodedAudioFrame {
    public:
     struct DecodeResult {
@@ -187,9 +189,6 @@
                                       int sample_rate_hz,
                                       int16_t* decoded,
                                       SpeechType* speech_type);
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
 };
 
 }  // namespace webrtc

diff --git a/api/jsep_ice_candidate.h b/api/jsep_ice_candidate.h
index 40e2783..8f47a10 100644
--- a/api/jsep_ice_candidate.h
+++ b/api/jsep_ice_candidate.h

@@ -22,7 +22,6 @@
 
 #include "api/candidate.h"
 #include "api/jsep.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -64,6 +63,10 @@
   // Move constructor is defined so that a vector of JsepCandidateCollections
   // can be resized.
   JsepCandidateCollection(JsepCandidateCollection&& o);
+
+  JsepCandidateCollection(const JsepCandidateCollection&) = delete;
+  JsepCandidateCollection& operator=(const JsepCandidateCollection&) = delete;
+
   // Returns a copy of the candidate collection.
   JsepCandidateCollection Clone() const;
   size_t count() const override;
@@ -80,8 +83,6 @@
 
  private:
   std::vector<std::unique_ptr<JsepIceCandidate>> candidates_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(JsepCandidateCollection);
 };
 
 }  // namespace webrtc

diff --git a/api/jsep_session_description.h b/api/jsep_session_description.h
index a4300eb..0b65734 100644
--- a/api/jsep_session_description.h
+++ b/api/jsep_session_description.h

@@ -22,7 +22,6 @@
 #include "api/candidate.h"
 #include "api/jsep.h"
 #include "api/jsep_ice_candidate.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace cricket {
 class SessionDescription;
@@ -43,6 +42,9 @@
       absl::string_view session_version);
   virtual ~JsepSessionDescription();
 
+  JsepSessionDescription(const JsepSessionDescription&) = delete;
+  JsepSessionDescription& operator=(const JsepSessionDescription&) = delete;
+
   // Takes ownership of `description`.
   bool Initialize(std::unique_ptr<cricket::SessionDescription> description,
                   const std::string& session_id,
@@ -82,8 +84,6 @@
   bool GetMediasectionIndex(const IceCandidateInterface* candidate,
                             size_t* index);
   int GetMediasectionIndex(const cricket::Candidate& candidate);
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(JsepSessionDescription);
 };
 
 }  // namespace webrtc

diff --git a/api/ref_counted_base.h b/api/ref_counted_base.h
index 931cb20..f20228b 100644
--- a/api/ref_counted_base.h
+++ b/api/ref_counted_base.h

@@ -12,7 +12,6 @@
 
 #include <type_traits>
 
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/ref_counter.h"
 
 namespace rtc {
@@ -21,6 +20,9 @@
  public:
   RefCountedBase() = default;
 
+  RefCountedBase(const RefCountedBase&) = delete;
+  RefCountedBase& operator=(const RefCountedBase&) = delete;
+
   void AddRef() const { ref_count_.IncRef(); }
   RefCountReleaseStatus Release() const {
     const auto status = ref_count_.DecRef();
@@ -39,8 +41,6 @@
 
  private:
   mutable webrtc::webrtc_impl::RefCounter ref_count_{0};
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(RefCountedBase);
 };
 
 // Template based version of `RefCountedBase` for simple implementations that do
@@ -61,6 +61,9 @@
  public:
   RefCountedNonVirtual() = default;
 
+  RefCountedNonVirtual(const RefCountedNonVirtual&) = delete;
+  RefCountedNonVirtual& operator=(const RefCountedNonVirtual&) = delete;
+
   void AddRef() const { ref_count_.IncRef(); }
   RefCountReleaseStatus Release() const {
     // If you run into this assert, T has virtual methods. There are two
@@ -88,8 +91,6 @@
 
  private:
   mutable webrtc::webrtc_impl::RefCounter ref_count_{0};
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(RefCountedNonVirtual);
 };
 
 }  // namespace rtc

diff --git a/api/stats_types.h b/api/stats_types.h
index b7cb8ef..c3e4451 100644
--- a/api/stats_types.h
+++ b/api/stats_types.h

@@ -22,7 +22,6 @@
 
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/ref_count.h"
 #include "rtc_base/system/rtc_export.h"
 
@@ -288,6 +287,9 @@
 
     ~Value();
 
+    Value(const Value&) = delete;
+    Value& operator=(const Value&) = delete;
+
     // Support ref counting. Note that for performance reasons, we
     // don't use thread safe operations. Therefore, all operations
     // affecting the ref count (in practice, creation and copying of
@@ -358,8 +360,6 @@
       const char* static_string_;
       Id* id_;
     } value_;
-
-    RTC_DISALLOW_COPY_AND_ASSIGN(Value);
   };
 
   typedef rtc::scoped_refptr<Value> ValuePtr;
@@ -369,6 +369,9 @@
   explicit StatsReport(const Id& id);
   ~StatsReport();
 
+  StatsReport(const StatsReport&) = delete;
+  StatsReport& operator=(const StatsReport&) = delete;
+
   // Factory functions for various types of stats IDs.
   static Id NewBandwidthEstimationId();
   static Id NewTypedId(StatsType type, const std::string& id);
@@ -408,8 +411,6 @@
   const Id id_;
   double timestamp_;  // Time since 1970-01-01T00:00:00Z in milliseconds.
   Values values_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(StatsReport);
 };
 
 // Typedef for an array of const StatsReport pointers.

diff --git a/call/call.cc b/call/call.cc
index d83d5fb..d34b9d3 100644
--- a/call/call.cc
+++ b/call/call.cc

@@ -51,7 +51,6 @@
 #include "modules/utility/include/process_thread.h"
 #include "modules/video_coding/fec_controller_default.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
@@ -209,6 +208,9 @@
        TaskQueueFactory* task_queue_factory);
   ~Call() override;
 
+  Call(const Call&) = delete;
+  Call& operator=(const Call&) = delete;
+
   // Implements webrtc::Call.
   PacketReceiver* Receiver() override;
 
@@ -469,8 +471,6 @@
   RTC_NO_UNIQUE_ADDRESS SequenceChecker sent_packet_sequence_checker_;
   absl::optional<rtc::SentPacket> last_sent_packet_
       RTC_GUARDED_BY(sent_packet_sequence_checker_);
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 

diff --git a/call/fake_network_pipe.h b/call/fake_network_pipe.h
index fadae33..be72e91 100644
--- a/call/fake_network_pipe.h
+++ b/call/fake_network_pipe.h

@@ -23,7 +23,6 @@
 #include "api/test/simulated_network.h"
 #include "call/call.h"
 #include "call/simulated_packet_receiver.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
 
@@ -109,6 +108,9 @@
 
   ~FakeNetworkPipe() override;
 
+  FakeNetworkPipe(const FakeNetworkPipe&) = delete;
+  FakeNetworkPipe& operator=(const FakeNetworkPipe&) = delete;
+
   void SetClockOffset(int64_t offset_ms);
 
   // Must not be called in parallel with DeliverPacket or Process.
@@ -228,8 +230,6 @@
   int64_t last_log_time_us_;
 
   std::map<Transport*, size_t> active_transports_ RTC_GUARDED_BY(config_lock_);
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
 };
 
 }  // namespace webrtc

diff --git a/call/rtp_bitrate_configurator.h b/call/rtp_bitrate_configurator.h
index 7ad83f8..5cb779a 100644
--- a/call/rtp_bitrate_configurator.h
+++ b/call/rtp_bitrate_configurator.h

@@ -14,7 +14,6 @@
 #include "absl/types/optional.h"
 #include "api/transport/bitrate_settings.h"
 #include "api/units/data_rate.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -24,6 +23,10 @@
  public:
   explicit RtpBitrateConfigurator(const BitrateConstraints& bitrate_config);
   ~RtpBitrateConfigurator();
+
+  RtpBitrateConfigurator(const RtpBitrateConfigurator&) = delete;
+  RtpBitrateConfigurator& operator=(const RtpBitrateConfigurator&) = delete;
+
   BitrateConstraints GetConfig() const;
 
   // The greater min and smaller max set by this and SetClientBitratePreferences
@@ -68,8 +71,6 @@
 
   // Bandwidth cap applied for relayed calls.
   DataRate max_bitrate_over_relay_ = DataRate::PlusInfinity();
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(RtpBitrateConfigurator);
 };
 }  // namespace webrtc
 

diff --git a/call/rtp_transport_controller_send.h b/call/rtp_transport_controller_send.h
index 62af78c..e5ff162 100644
--- a/call/rtp_transport_controller_send.h
+++ b/call/rtp_transport_controller_send.h

@@ -32,7 +32,6 @@
 #include "modules/pacing/rtp_packet_pacer.h"
 #include "modules/pacing/task_queue_paced_sender.h"
 #include "modules/utility/include/process_thread.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/network_route.h"
 #include "rtc_base/race_checker.h"
 #include "rtc_base/task_queue.h"
@@ -63,6 +62,10 @@
       const WebRtcKeyValueConfig* trials);
   ~RtpTransportControllerSend() override;
 
+  RtpTransportControllerSend(const RtpTransportControllerSend&) = delete;
+  RtpTransportControllerSend& operator=(const RtpTransportControllerSend&) =
+      delete;
+
   // TODO(tommi): Change to std::unique_ptr<>.
   RtpVideoSenderInterface* CreateRtpVideoSender(
       const std::map<uint32_t, RtpState>& suspended_ssrcs,
@@ -215,7 +218,6 @@
   // `task_queue_` is defined last to ensure all pending tasks are cancelled
   // and deleted before any other members.
   rtc::TaskQueue task_queue_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(RtpTransportControllerSend);
 };
 
 }  // namespace webrtc

diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
index 25ef20f..9832246 100644
--- a/call/rtp_video_sender.h
+++ b/call/rtp_video_sender.h

@@ -35,7 +35,6 @@
 #include "modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/rate_limiter.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
@@ -89,6 +88,9 @@
       rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
   ~RtpVideoSender() override;
 
+  RtpVideoSender(const RtpVideoSender&) = delete;
+  RtpVideoSender& operator=(const RtpVideoSender&) = delete;
+
   // RtpVideoSender will only route packets if being active, all packets will be
   // dropped otherwise.
   void SetActive(bool active) RTC_LOCKS_EXCLUDED(mutex_) override;
@@ -209,8 +211,6 @@
   // This map is set at construction time and never changed, but it's
   // non-trivial to make it properly const.
   std::map<uint32_t, RtpRtcpInterface*> ssrc_to_rtp_module_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(RtpVideoSender);
 };
 
 }  // namespace webrtc

diff --git a/logging/rtc_event_log/encoder/bit_writer.h b/logging/rtc_event_log/encoder/bit_writer.h
index 85340c3..421e7c4 100644
--- a/logging/rtc_event_log/encoder/bit_writer.h
+++ b/logging/rtc_event_log/encoder/bit_writer.h

@@ -20,7 +20,6 @@
 #include "absl/strings/string_view.h"
 #include "rtc_base/bit_buffer.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -36,6 +35,9 @@
     RTC_DCHECK_GT(byte_count, 0);
   }
 
+  BitWriter(const BitWriter&) = delete;
+  BitWriter& operator=(const BitWriter&) = delete;
+
   void WriteBits(uint64_t val, size_t bit_count);
 
   void WriteBits(absl::string_view input);
@@ -52,8 +54,6 @@
   // to go anywhere near the limit, though, so this is good enough.
   size_t written_bits_;
   bool valid_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(BitWriter);
 };
 
 }  // namespace webrtc

diff --git a/logging/rtc_event_log/encoder/delta_encoding.cc b/logging/rtc_event_log/encoder/delta_encoding.cc
index a96d3a7..3a2bee1 100644
--- a/logging/rtc_event_log/encoder/delta_encoding.cc
+++ b/logging/rtc_event_log/encoder/delta_encoding.cc

@@ -21,7 +21,6 @@
 #include "rtc_base/bit_buffer.h"
 #include "rtc_base/bitstream_reader.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
 
@@ -187,6 +186,9 @@
       absl::optional<uint64_t> base,
       const std::vector<absl::optional<uint64_t>>& values);
 
+  FixedLengthDeltaEncoder(const FixedLengthDeltaEncoder&) = delete;
+  FixedLengthDeltaEncoder& operator=(const FixedLengthDeltaEncoder&) = delete;
+
  private:
   // Calculate min/max values of unsigned/signed deltas, given the bit width
   // of all the values in the series.
@@ -249,8 +251,6 @@
   // ctor has finished running when this is constructed, so that the lower
   // bound on the buffer size would be guaranteed correct.
   std::unique_ptr<BitWriter> writer_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(FixedLengthDeltaEncoder);
 };
 
 // TODO(eladalon): Reduce the number of passes.
@@ -566,6 +566,9 @@
       absl::optional<uint64_t> base,
       size_t num_of_deltas);
 
+  FixedLengthDeltaDecoder(const FixedLengthDeltaDecoder&) = delete;
+  FixedLengthDeltaDecoder& operator=(const FixedLengthDeltaDecoder&) = delete;
+
  private:
   // Reads the encoding header in `input` and returns a FixedLengthDeltaDecoder
   // with the corresponding configuration, that can be used to decode the
@@ -619,8 +622,6 @@
 
   // The number of values to be known to be decoded.
   const size_t num_of_deltas_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(FixedLengthDeltaDecoder);
 };
 
 bool FixedLengthDeltaDecoder::IsSuitableDecoderFor(const std::string& input) {

diff --git a/modules/video_coding/utility/ivf_file_reader.h b/modules/video_coding/utility/ivf_file_reader.h
index 5e0634f..cc64d4c 100644
--- a/modules/video_coding/utility/ivf_file_reader.h
+++ b/modules/video_coding/utility/ivf_file_reader.h

@@ -17,6 +17,7 @@
 #include "absl/types/optional.h"
 #include "api/video/encoded_image.h"
 #include "api/video_codecs/video_codec.h"
+#include "rtc_base/constructor_magic.h"
 #include "rtc_base/system/file_wrapper.h"
 
 namespace webrtc {

diff --git a/p2p/base/async_stun_tcp_socket.h b/p2p/base/async_stun_tcp_socket.h
index eb4eef7..f0df42b 100644
--- a/p2p/base/async_stun_tcp_socket.h
+++ b/p2p/base/async_stun_tcp_socket.h

@@ -15,7 +15,6 @@
 
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/async_tcp_socket.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/socket.h"
 #include "rtc_base/socket_address.h"
 
@@ -32,6 +31,9 @@
 
   explicit AsyncStunTCPSocket(rtc::Socket* socket);
 
+  AsyncStunTCPSocket(const AsyncStunTCPSocket&) = delete;
+  AsyncStunTCPSocket& operator=(const AsyncStunTCPSocket&) = delete;
+
   int Send(const void* pv,
            size_t cb,
            const rtc::PacketOptions& options) override;
@@ -42,8 +44,6 @@
   // This method also returns the number of padding bytes needed/added to the
   // turn message. `pad_bytes` should be used only when `is_turn` is true.
   size_t GetExpectedLength(const void* data, size_t len, int* pad_bytes);
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AsyncStunTCPSocket);
 };
 
 }  // namespace cricket

diff --git a/p2p/base/dtls_transport.h b/p2p/base/dtls_transport.h
index edfa889..d503a92 100644
--- a/p2p/base/dtls_transport.h
+++ b/p2p/base/dtls_transport.h

@@ -22,7 +22,6 @@
 #include "p2p/base/ice_transport_internal.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/buffer_queue.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/stream.h"
 #include "rtc_base/strings/string_builder.h"
@@ -40,6 +39,9 @@
  public:
   explicit StreamInterfaceChannel(IceTransportInternal* ice_transport);
 
+  StreamInterfaceChannel(const StreamInterfaceChannel&) = delete;
+  StreamInterfaceChannel& operator=(const StreamInterfaceChannel&) = delete;
+
   // Push in a packet; this gets pulled out from Read().
   bool OnPacketReceived(const char* data, size_t size);
 
@@ -60,8 +62,6 @@
   IceTransportInternal* const ice_transport_;  // owned by DtlsTransport
   rtc::StreamState state_ RTC_GUARDED_BY(sequence_checker_);
   rtc::BufferQueue packets_ RTC_GUARDED_BY(sequence_checker_);
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(StreamInterfaceChannel);
 };
 
 // This class provides a DTLS SSLStreamAdapter inside a TransportChannel-style
@@ -110,6 +110,9 @@
 
   ~DtlsTransport() override;
 
+  DtlsTransport(const DtlsTransport&) = delete;
+  DtlsTransport& operator=(const DtlsTransport&) = delete;
+
   webrtc::DtlsTransportState dtls_state() const override;
   const std::string& transport_name() const override;
   int component() const override;
@@ -248,8 +251,6 @@
   bool writable_ = false;
 
   webrtc::RtcEventLog* const event_log_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(DtlsTransport);
 };
 
 }  // namespace cricket

diff --git a/p2p/base/dtls_transport_internal.h b/p2p/base/dtls_transport_internal.h
index 0b26a7f..24c682f 100644
--- a/p2p/base/dtls_transport_internal.h
+++ b/p2p/base/dtls_transport_internal.h

@@ -25,7 +25,6 @@
 #include "p2p/base/ice_transport_internal.h"
 #include "p2p/base/packet_transport_internal.h"
 #include "rtc_base/callback_list.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/ssl_certificate.h"
 #include "rtc_base/ssl_fingerprint.h"
 #include "rtc_base/ssl_stream_adapter.h"
@@ -48,6 +47,9 @@
  public:
   ~DtlsTransportInternal() override;
 
+  DtlsTransportInternal(const DtlsTransportInternal&) = delete;
+  DtlsTransportInternal& operator=(const DtlsTransportInternal&) = delete;
+
   virtual webrtc::DtlsTransportState dtls_state() const = 0;
 
   virtual int component() const = 0;
@@ -135,7 +137,6 @@
   DtlsTransportInternal();
 
  private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(DtlsTransportInternal);
   webrtc::CallbackList<const rtc::SSLHandshakeError>
       dtls_handshake_error_callback_list_;
   webrtc::CallbackList<DtlsTransportInternal*, const webrtc::DtlsTransportState>

diff --git a/p2p/base/p2p_transport_channel.h b/p2p/base/p2p_transport_channel.h
index 28248e7..7f5fb05 100644
--- a/p2p/base/p2p_transport_channel.h
+++ b/p2p/base/p2p_transport_channel.h

@@ -56,7 +56,6 @@
 #include "p2p/base/transport_description.h"
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/dscp.h"
 #include "rtc_base/network/sent_packet.h"
 #include "rtc_base/network_route.h"
@@ -124,6 +123,9 @@
       IceControllerFactoryInterface* ice_controller_factory = nullptr);
   ~P2PTransportChannel() override;
 
+  P2PTransportChannel(const P2PTransportChannel&) = delete;
+  P2PTransportChannel& operator=(const P2PTransportChannel&) = delete;
+
   // From TransportChannelImpl:
   IceTransportState GetState() const override;
   webrtc::IceTransportState GetIceTransportState() const override;
@@ -494,8 +496,6 @@
   int64_t last_data_received_ms_ = 0;
 
   IceFieldTrials field_trials_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(P2PTransportChannel);
 };
 
 }  // namespace cricket

diff --git a/p2p/stunprober/stun_prober.cc b/p2p/stunprober/stun_prober.cc
index 4195230..efe0fbd 100644
--- a/p2p/stunprober/stun_prober.cc
+++ b/p2p/stunprober/stun_prober.cc

@@ -21,7 +21,6 @@
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/async_resolver_interface.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/helpers.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/task_utils/to_queued_task.h"
@@ -69,6 +68,9 @@
             const std::vector<rtc::SocketAddress>& server_ips);
   ~Requester() override;
 
+  Requester(const Requester&) = delete;
+  Requester& operator=(const Requester&) = delete;
+
   // There is no callback for SendStunRequest as the underneath socket send is
   // expected to be completed immediately. Otherwise, it'll skip this request
   // and move to the next one.
@@ -105,8 +107,6 @@
   int16_t num_response_received_ = 0;
 
   webrtc::SequenceChecker& thread_checker_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Requester);
 };
 
 StunProber::Requester::Requester(

diff --git a/p2p/stunprober/stun_prober.h b/p2p/stunprober/stun_prober.h
index fe2f14c..b562394 100644
--- a/p2p/stunprober/stun_prober.h
+++ b/p2p/stunprober/stun_prober.h

@@ -17,7 +17,6 @@
 
 #include "api/sequence_checker.h"
 #include "rtc_base/byte_buffer.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/ip_address.h"
 #include "rtc_base/network.h"
 #include "rtc_base/socket_address.h"
@@ -101,6 +100,9 @@
              const rtc::NetworkManager::NetworkList& networks);
   ~StunProber() override;
 
+  StunProber(const StunProber&) = delete;
+  StunProber& operator=(const StunProber&) = delete;
+
   // Begin performing the probe test against the `servers`. If
   // `shared_socket_mode` is false, each request will be done with a new socket.
   // Otherwise, a unique socket will be used for a single round of requests
@@ -241,8 +243,6 @@
   rtc::NetworkManager::NetworkList networks_;
 
   webrtc::ScopedTaskSafety task_safety_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(StunProber);
 };
 
 }  // namespace stunprober

diff --git a/pc/dtmf_sender.h b/pc/dtmf_sender.h
index a208b10..915d987 100644
--- a/pc/dtmf_sender.h
+++ b/pc/dtmf_sender.h

@@ -18,7 +18,6 @@
 #include "api/dtmf_sender_interface.h"
 #include "api/scoped_refptr.h"
 #include "pc/proxy.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/location.h"
 #include "rtc_base/ref_count.h"
 #include "rtc_base/task_utils/pending_task_safety_flag.h"
@@ -72,6 +71,9 @@
   DtmfSender(rtc::Thread* signaling_thread, DtmfProviderInterface* provider);
   virtual ~DtmfSender();
 
+  DtmfSender(const DtmfSender&) = delete;
+  DtmfSender& operator=(const DtmfSender&) = delete;
+
  private:
   DtmfSender();
 
@@ -96,8 +98,6 @@
   // For cancelling the tasks which feed the DTMF provider one tone at a time.
   rtc::scoped_refptr<PendingTaskSafetyFlag> safety_flag_ RTC_GUARDED_BY(
       signaling_thread_) RTC_PT_GUARDED_BY(signaling_thread_) = nullptr;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(DtmfSender);
 };
 
 // Define proxy for DtmfSenderInterface.

diff --git a/pc/jsep_transport.h b/pc/jsep_transport.h
index e3e929b..93604a1 100644
--- a/pc/jsep_transport.h
+++ b/pc/jsep_transport.h

@@ -44,7 +44,6 @@
 #include "pc/srtp_transport.h"
 #include "pc/transport_stats.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/rtc_certificate.h"
 #include "rtc_base/ssl_fingerprint.h"
 #include "rtc_base/ssl_stream_adapter.h"
@@ -106,6 +105,9 @@
 
   ~JsepTransport();
 
+  JsepTransport(const JsepTransport&) = delete;
+  JsepTransport& operator=(const JsepTransport&) = delete;
+
   // Returns the MID of this transport. This is only used for logging.
   const std::string& mid() const { return mid_; }
 
@@ -326,8 +328,6 @@
   // `rtcp_dtls_transport_` is destroyed. The JsepTransportController will
   // receive the callback and update the aggregate transport states.
   std::function<void()> rtcp_mux_active_callback_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(JsepTransport);
 };
 
 }  // namespace cricket

diff --git a/pc/jsep_transport_controller.h b/pc/jsep_transport_controller.h
index fb42009..4e06566 100644
--- a/pc/jsep_transport_controller.h
+++ b/pc/jsep_transport_controller.h

@@ -58,7 +58,6 @@
 #include "pc/transport_stats.h"
 #include "rtc_base/callback_list.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/helpers.h"
 #include "rtc_base/ref_counted_object.h"
@@ -150,6 +149,9 @@
       Config config);
   virtual ~JsepTransportController();
 
+  JsepTransportController(const JsepTransportController&) = delete;
+  JsepTransportController& operator=(const JsepTransportController&) = delete;
+
   // The main method to be called; applies a description at the transport
   // level, creating/destroying transport objects as needed and updating their
   // properties. This includes RTP, DTLS, and ICE (but not SCTP). At least not
@@ -478,8 +480,6 @@
   rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
 
   BundleManager bundles_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(JsepTransportController);
 };
 
 }  // namespace webrtc

diff --git a/pc/srtp_session.h b/pc/srtp_session.h
index 89fab0d..f1b6a52 100644
--- a/pc/srtp_session.h
+++ b/pc/srtp_session.h

@@ -15,7 +15,6 @@
 
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/synchronization/mutex.h"
 
 // Forward declaration to avoid pulling in libsrtp headers here
@@ -35,6 +34,9 @@
   SrtpSession();
   ~SrtpSession();
 
+  SrtpSession(const SrtpSession&) = delete;
+  SrtpSession& operator=(const SrtpSession&) = delete;
+
   // Configures the session for sending data using the specified
   // cipher-suite and key. Receiving must be done by a separate session.
   bool SetSend(int cs,
@@ -141,7 +143,6 @@
   bool external_auth_enabled_ = false;
   int decryption_failure_count_ = 0;
   bool dump_plain_rtp_ = false;
-  RTC_DISALLOW_COPY_AND_ASSIGN(SrtpSession);
 };
 
 }  // namespace cricket

diff --git a/pc/video_rtp_track_source.h b/pc/video_rtp_track_source.h
index 23a7cd2..a9e43f6 100644
--- a/pc/video_rtp_track_source.h
+++ b/pc/video_rtp_track_source.h

@@ -20,7 +20,6 @@
 #include "api/video/video_source_interface.h"
 #include "media/base/video_broadcaster.h"
 #include "pc/video_track_source.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/system/no_unique_address.h"
 #include "rtc_base/thread_annotations.h"
@@ -45,6 +44,9 @@
 
   explicit VideoRtpTrackSource(Callback* callback);
 
+  VideoRtpTrackSource(const VideoRtpTrackSource&) = delete;
+  VideoRtpTrackSource& operator=(const VideoRtpTrackSource&) = delete;
+
   // Call before the object implementing Callback finishes it's destructor. No
   // more callbacks will be fired after completion. Must be called on the
   // worker thread
@@ -83,8 +85,6 @@
   std::vector<rtc::VideoSinkInterface<RecordableEncodedFrame>*> encoded_sinks_
       RTC_GUARDED_BY(mu_);
   Callback* callback_ RTC_GUARDED_BY(worker_sequence_checker_);
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(VideoRtpTrackSource);
 };
 
 }  // namespace webrtc

diff --git a/pc/webrtc_session_description_factory.h b/pc/webrtc_session_description_factory.h
index 8e80fb5..efa208f 100644
--- a/pc/webrtc_session_description_factory.h
+++ b/pc/webrtc_session_description_factory.h

@@ -26,7 +26,6 @@
 #include "pc/channel_manager.h"
 #include "pc/media_session.h"
 #include "pc/sdp_state_provider.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/message_handler.h"
 #include "rtc_base/rtc_certificate.h"
 #include "rtc_base/rtc_certificate_generator.h"
@@ -92,6 +91,11 @@
           on_certificate_ready);
   virtual ~WebRtcSessionDescriptionFactory();
 
+  WebRtcSessionDescriptionFactory(const WebRtcSessionDescriptionFactory&) =
+      delete;
+  WebRtcSessionDescriptionFactory& operator=(
+      const WebRtcSessionDescriptionFactory&) = delete;
+
   static void CopyCandidatesFromSessionDescription(
       const SessionDescriptionInterface* source_desc,
       const std::string& content_name,
@@ -159,8 +163,6 @@
 
   std::function<void(const rtc::scoped_refptr<rtc::RTCCertificate>&)>
       on_certificate_ready_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSessionDescriptionFactory);
 };
 }  // namespace webrtc
 

diff --git a/system_wrappers/source/metrics.cc b/system_wrappers/source/metrics.cc
index b14eef4..8c9cf0c 100644
--- a/system_wrappers/source/metrics.cc
+++ b/system_wrappers/source/metrics.cc

@@ -11,7 +11,6 @@
 
 #include <algorithm>
 
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
 
@@ -35,6 +34,9 @@
     RTC_DCHECK_GT(bucket_count, 0);
   }
 
+  RtcHistogram(const RtcHistogram&) = delete;
+  RtcHistogram& operator=(const RtcHistogram&) = delete;
+
   void Add(int sample) {
     sample = std::min(sample, max_);
     sample = std::max(sample, min_ - 1);  // Underflow bucket.
@@ -99,8 +101,6 @@
   const int min_;
   const int max_;
   SampleInfo info_ RTC_GUARDED_BY(mutex_);
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(RtcHistogram);
 };
 
 class RtcHistogramMap {
@@ -108,6 +108,9 @@
   RtcHistogramMap() {}
   ~RtcHistogramMap() {}
 
+  RtcHistogramMap(const RtcHistogramMap&) = delete;
+  RtcHistogramMap& operator=(const RtcHistogramMap&) = delete;
+
   Histogram* GetCountsHistogram(const std::string& name,
                                 int min,
                                 int max,
@@ -178,8 +181,6 @@
   mutable Mutex mutex_;
   std::map<std::string, std::unique_ptr<RtcHistogram>> map_
       RTC_GUARDED_BY(mutex_);
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(RtcHistogramMap);
 };
 
 // RtcHistogramMap is allocated upon call to Enable().

diff --git a/test/network/network_emulation.h b/test/network/network_emulation.h
index e60deaf..d10e9a8 100644
--- a/test/network/network_emulation.h
+++ b/test/network/network_emulation.h

@@ -26,6 +26,7 @@
 #include "api/test/network_emulation_manager.h"
 #include "api/test/simulated_network.h"
 #include "api/units/timestamp.h"
+#include "rtc_base/constructor_magic.h"
 #include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/network.h"
 #include "rtc_base/network_constants.h"