Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/

Bug: webrtc:13555, webrtc:13082
Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35771}
diff --git a/modules/audio_coding/acm2/acm_receive_test.h b/modules/audio_coding/acm2/acm_receive_test.h
index 6349c63..2095ef9 100644
--- a/modules/audio_coding/acm2/acm_receive_test.h
+++ b/modules/audio_coding/acm2/acm_receive_test.h
@@ -18,7 +18,6 @@
 
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/scoped_refptr.h"
-#include "rtc_base/constructor_magic.h"
 #include "system_wrappers/include/clock.h"
 
 namespace webrtc {
@@ -45,6 +44,9 @@
                        rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
   virtual ~AcmReceiveTestOldApi();
 
+  AcmReceiveTestOldApi(const AcmReceiveTestOldApi&) = delete;
+  AcmReceiveTestOldApi& operator=(const AcmReceiveTestOldApi&) = delete;
+
   // Registers the codecs with default parameters from ACM.
   void RegisterDefaultCodecs();
 
@@ -67,8 +69,6 @@
   AudioSink* audio_sink_;
   int output_freq_hz_;
   NumOutputChannels exptected_output_channels_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AcmReceiveTestOldApi);
 };
 
 // This test toggles the output frequency every `toggle_period_ms`. The test
diff --git a/modules/audio_coding/acm2/acm_send_test.h b/modules/audio_coding/acm2/acm_send_test.h
index 0c82415..b14cb80 100644
--- a/modules/audio_coding/acm2/acm_send_test.h
+++ b/modules/audio_coding/acm2/acm_send_test.h
@@ -17,7 +17,6 @@
 #include "api/audio/audio_frame.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_coding/neteq/tools/packet_source.h"
-#include "rtc_base/constructor_magic.h"
 #include "system_wrappers/include/clock.h"
 
 namespace webrtc {
@@ -35,6 +34,9 @@
                     int test_duration_ms);
   ~AcmSendTestOldApi() override;
 
+  AcmSendTestOldApi(const AcmSendTestOldApi&) = delete;
+  AcmSendTestOldApi& operator=(const AcmSendTestOldApi&) = delete;
+
   // Registers the send codec. Returns true on success, false otherwise.
   bool RegisterCodec(const char* payload_name,
                      int sampling_freq_hz,
@@ -81,8 +83,6 @@
   uint16_t sequence_number_;
   std::vector<uint8_t> last_payload_vec_;
   bool data_to_send_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi);
 };
 
 }  // namespace test
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
index 1c91fa1..664e76b 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
@@ -21,7 +21,6 @@
 #include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
 #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
 #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -44,6 +43,9 @@
 
   ~AudioNetworkAdaptorImpl() override;
 
+  AudioNetworkAdaptorImpl(const AudioNetworkAdaptorImpl&) = delete;
+  AudioNetworkAdaptorImpl& operator=(const AudioNetworkAdaptorImpl&) = delete;
+
   void SetUplinkBandwidth(int uplink_bandwidth_bps) override;
 
   void SetUplinkPacketLossFraction(float uplink_packet_loss_fraction) override;
@@ -80,8 +82,6 @@
   absl::optional<AudioEncoderRuntimeConfig> prev_config_;
 
   ANAStats stats_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioNetworkAdaptorImpl);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller.h b/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
index 41bfbd1..c103214 100644
--- a/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
+++ b/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
@@ -16,7 +16,6 @@
 #include "absl/types/optional.h"
 #include "modules/audio_coding/audio_network_adaptor/controller.h"
 #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace audio_network_adaptor {
@@ -39,6 +38,9 @@
 
   ~BitrateController() override;
 
+  BitrateController(const BitrateController&) = delete;
+  BitrateController& operator=(const BitrateController&) = delete;
+
   void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
 
   void MakeDecision(AudioEncoderRuntimeConfig* config) override;
@@ -49,7 +51,6 @@
   int frame_length_ms_;
   absl::optional<int> target_audio_bitrate_bps_;
   absl::optional<size_t> overhead_bytes_per_packet_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(BitrateController);
 };
 
 }  // namespace audio_network_adaptor
diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller.h b/modules/audio_coding/audio_network_adaptor/channel_controller.h
index f211f40..3cd4bb7 100644
--- a/modules/audio_coding/audio_network_adaptor/channel_controller.h
+++ b/modules/audio_coding/audio_network_adaptor/channel_controller.h
@@ -16,7 +16,6 @@
 #include "absl/types/optional.h"
 #include "modules/audio_coding/audio_network_adaptor/controller.h"
 #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -41,6 +40,9 @@
 
   ~ChannelController() override;
 
+  ChannelController(const ChannelController&) = delete;
+  ChannelController& operator=(const ChannelController&) = delete;
+
   void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
 
   void MakeDecision(AudioEncoderRuntimeConfig* config) override;
@@ -49,7 +51,6 @@
   const Config config_;
   size_t channels_to_encode_;
   absl::optional<int> uplink_bandwidth_bps_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(ChannelController);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.h b/modules/audio_coding/audio_network_adaptor/controller_manager.h
index c168ebc..f7d7b34 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.h
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager.h
@@ -17,7 +17,6 @@
 #include <vector>
 
 #include "modules/audio_coding/audio_network_adaptor/controller.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -80,6 +79,9 @@
 
   ~ControllerManagerImpl() override;
 
+  ControllerManagerImpl(const ControllerManagerImpl&) = delete;
+  ControllerManagerImpl& operator=(const ControllerManagerImpl&) = delete;
+
   // Sort controllers based on their significance.
   std::vector<Controller*> GetSortedControllers(
       const Controller::NetworkMetrics& metrics) override;
@@ -114,8 +116,6 @@
   // `scoring_points_` saves the scoring points of various
   // controllers.
   std::map<const Controller*, ScoringPoint> controller_scoring_points_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(ControllerManagerImpl);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller.h b/modules/audio_coding/audio_network_adaptor/dtx_controller.h
index 83fdf3d..b8a8e47 100644
--- a/modules/audio_coding/audio_network_adaptor/dtx_controller.h
+++ b/modules/audio_coding/audio_network_adaptor/dtx_controller.h
@@ -14,7 +14,6 @@
 #include "absl/types/optional.h"
 #include "modules/audio_coding/audio_network_adaptor/controller.h"
 #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -35,6 +34,9 @@
 
   ~DtxController() override;
 
+  DtxController(const DtxController&) = delete;
+  DtxController& operator=(const DtxController&) = delete;
+
   void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
 
   void MakeDecision(AudioEncoderRuntimeConfig* config) override;
@@ -43,7 +45,6 @@
   const Config config_;
   bool dtx_enabled_;
   absl::optional<int> uplink_bandwidth_bps_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(DtxController);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer.h b/modules/audio_coding/audio_network_adaptor/event_log_writer.h
index c5e57e6..a147311 100644
--- a/modules/audio_coding/audio_network_adaptor/event_log_writer.h
+++ b/modules/audio_coding/audio_network_adaptor/event_log_writer.h
@@ -12,7 +12,6 @@
 #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
 
 #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 class RtcEventLog;
@@ -24,6 +23,10 @@
                  float min_bitrate_change_fraction,
                  float min_packet_loss_change_fraction);
   ~EventLogWriter();
+
+  EventLogWriter(const EventLogWriter&) = delete;
+  EventLogWriter& operator=(const EventLogWriter&) = delete;
+
   void MaybeLogEncoderConfig(const AudioEncoderRuntimeConfig& config);
 
  private:
@@ -34,7 +37,6 @@
   const float min_bitrate_change_fraction_;
   const float min_packet_loss_change_fraction_;
   AudioEncoderRuntimeConfig last_logged_config_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(EventLogWriter);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
index 85d235e..0c57ad1 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
@@ -18,7 +18,6 @@
 #include "modules/audio_coding/audio_network_adaptor/controller.h"
 #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
 #include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -53,6 +52,9 @@
 
   ~FecControllerPlrBased() override;
 
+  FecControllerPlrBased(const FecControllerPlrBased&) = delete;
+  FecControllerPlrBased& operator=(const FecControllerPlrBased&) = delete;
+
   void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
 
   void MakeDecision(AudioEncoderRuntimeConfig* config) override;
@@ -65,8 +67,6 @@
   bool fec_enabled_;
   absl::optional<int> uplink_bandwidth_bps_;
   const std::unique_ptr<SmoothingFilter> packet_loss_smoother_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(FecControllerPlrBased);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller.h b/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
index 74a787e..04693f8 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
@@ -19,7 +19,6 @@
 #include "absl/types/optional.h"
 #include "modules/audio_coding/audio_network_adaptor/controller.h"
 #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -62,6 +61,9 @@
 
   ~FrameLengthController() override;
 
+  FrameLengthController(const FrameLengthController&) = delete;
+  FrameLengthController& operator=(const FrameLengthController&) = delete;
+
   void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
 
   void MakeDecision(AudioEncoderRuntimeConfig* config) override;
@@ -84,8 +86,6 @@
   // True if the previous frame length decision was an increase, otherwise
   // false.
   bool prev_decision_increase_ = false;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(FrameLengthController);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
index 6185918..3fa42cb 100644
--- a/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
+++ b/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
@@ -19,7 +19,6 @@
 #include "api/audio_codecs/audio_decoder.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -28,6 +27,10 @@
   explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) {
     RTC_DCHECK_GE(num_channels, 1);
   }
+
+  AudioDecoderPcmU(const AudioDecoderPcmU&) = delete;
+  AudioDecoderPcmU& operator=(const AudioDecoderPcmU&) = delete;
+
   void Reset() override;
   std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
                                         uint32_t timestamp) override;
@@ -44,7 +47,6 @@
 
  private:
   const size_t num_channels_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmU);
 };
 
 class AudioDecoderPcmA final : public AudioDecoder {
@@ -52,6 +54,10 @@
   explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) {
     RTC_DCHECK_GE(num_channels, 1);
   }
+
+  AudioDecoderPcmA(const AudioDecoderPcmA&) = delete;
+  AudioDecoderPcmA& operator=(const AudioDecoderPcmA&) = delete;
+
   void Reset() override;
   std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
                                         uint32_t timestamp) override;
@@ -68,7 +74,6 @@
 
  private:
   const size_t num_channels_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmA);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
index c4413f5..d50be4b 100644
--- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
+++ b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -17,7 +17,6 @@
 #include "absl/types/optional.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/units/time_delta.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -83,6 +82,9 @@
   explicit AudioEncoderPcmA(const Config& config)
       : AudioEncoderPcm(config, kSampleRateHz) {}
 
+  AudioEncoderPcmA(const AudioEncoderPcmA&) = delete;
+  AudioEncoderPcmA& operator=(const AudioEncoderPcmA&) = delete;
+
  protected:
   size_t EncodeCall(const int16_t* audio,
                     size_t input_len,
@@ -94,7 +96,6 @@
 
  private:
   static const int kSampleRateHz = 8000;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmA);
 };
 
 class AudioEncoderPcmU final : public AudioEncoderPcm {
@@ -106,6 +107,9 @@
   explicit AudioEncoderPcmU(const Config& config)
       : AudioEncoderPcm(config, kSampleRateHz) {}
 
+  AudioEncoderPcmU(const AudioEncoderPcmU&) = delete;
+  AudioEncoderPcmU& operator=(const AudioEncoderPcmU&) = delete;
+
  protected:
   size_t EncodeCall(const int16_t* audio,
                     size_t input_len,
@@ -117,7 +121,6 @@
 
  private:
   static const int kSampleRateHz = 8000;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmU);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/g722/audio_decoder_g722.h b/modules/audio_coding/codecs/g722/audio_decoder_g722.h
index eeca139..39e9e63 100644
--- a/modules/audio_coding/codecs/g722/audio_decoder_g722.h
+++ b/modules/audio_coding/codecs/g722/audio_decoder_g722.h
@@ -12,7 +12,6 @@
 #define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
 
 #include "api/audio_codecs/audio_decoder.h"
-#include "rtc_base/constructor_magic.h"
 
 typedef struct WebRtcG722DecInst G722DecInst;
 
@@ -22,6 +21,10 @@
  public:
   AudioDecoderG722Impl();
   ~AudioDecoderG722Impl() override;
+
+  AudioDecoderG722Impl(const AudioDecoderG722Impl&) = delete;
+  AudioDecoderG722Impl& operator=(const AudioDecoderG722Impl&) = delete;
+
   bool HasDecodePlc() const override;
   void Reset() override;
   std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
@@ -39,13 +42,17 @@
 
  private:
   G722DecInst* dec_state_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722Impl);
 };
 
 class AudioDecoderG722StereoImpl final : public AudioDecoder {
  public:
   AudioDecoderG722StereoImpl();
   ~AudioDecoderG722StereoImpl() override;
+
+  AudioDecoderG722StereoImpl(const AudioDecoderG722StereoImpl&) = delete;
+  AudioDecoderG722StereoImpl& operator=(const AudioDecoderG722StereoImpl&) =
+      delete;
+
   void Reset() override;
   std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
                                         uint32_t timestamp) override;
@@ -71,7 +78,6 @@
 
   G722DecInst* dec_state_left_;
   G722DecInst* dec_state_right_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722StereoImpl);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/modules/audio_coding/codecs/g722/audio_encoder_g722.h
index c836503..a932aa8 100644
--- a/modules/audio_coding/codecs/g722/audio_encoder_g722.h
+++ b/modules/audio_coding/codecs/g722/audio_encoder_g722.h
@@ -20,7 +20,6 @@
 #include "api/units/time_delta.h"
 #include "modules/audio_coding/codecs/g722/g722_interface.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -29,6 +28,9 @@
   AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type);
   ~AudioEncoderG722Impl() override;
 
+  AudioEncoderG722Impl(const AudioEncoderG722Impl&) = delete;
+  AudioEncoderG722Impl& operator=(const AudioEncoderG722Impl&) = delete;
+
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   int RtpTimestampRateHz() const override;
@@ -63,7 +65,6 @@
   uint32_t first_timestamp_in_buffer_;
   const std::unique_ptr<EncoderState[]> encoders_;
   rtc::Buffer interleave_buffer_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722Impl);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h b/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
index c2d62ed..46ba755 100644
--- a/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
+++ b/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
@@ -18,7 +18,6 @@
 
 #include "api/audio_codecs/audio_decoder.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/constructor_magic.h"
 
 typedef struct iLBC_decinst_t_ IlbcDecoderInstance;
 
@@ -28,6 +27,10 @@
  public:
   AudioDecoderIlbcImpl();
   ~AudioDecoderIlbcImpl() override;
+
+  AudioDecoderIlbcImpl(const AudioDecoderIlbcImpl&) = delete;
+  AudioDecoderIlbcImpl& operator=(const AudioDecoderIlbcImpl&) = delete;
+
   bool HasDecodePlc() const override;
   size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
   void Reset() override;
@@ -45,7 +48,6 @@
 
  private:
   IlbcDecoderInstance* dec_state_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIlbcImpl);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
index 05a900e..c8dfa2c 100644
--- a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
+++ b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
@@ -21,7 +21,6 @@
 #include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
 #include "api/units/time_delta.h"
 #include "modules/audio_coding/codecs/ilbc/ilbc.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -30,6 +29,9 @@
   AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, int payload_type);
   ~AudioEncoderIlbcImpl() override;
 
+  AudioEncoderIlbcImpl(const AudioEncoderIlbcImpl&) = delete;
+  AudioEncoderIlbcImpl& operator=(const AudioEncoderIlbcImpl&) = delete;
+
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
@@ -53,7 +55,6 @@
   uint32_t first_timestamp_in_buffer_;
   int16_t input_buffer_[kMaxSamplesPerPacket];
   IlbcEncoderInstance* encoder_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIlbcImpl);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h b/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
index 23a3020..aae708f 100644
--- a/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
+++ b/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
@@ -16,7 +16,6 @@
 #include "absl/types/optional.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/scoped_refptr.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -30,6 +29,9 @@
   explicit AudioDecoderIsacT(const Config& config);
   virtual ~AudioDecoderIsacT() override;
 
+  AudioDecoderIsacT(const AudioDecoderIsacT&) = delete;
+  AudioDecoderIsacT& operator=(const AudioDecoderIsacT&) = delete;
+
   bool HasDecodePlc() const override;
   size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
   void Reset() override;
@@ -45,8 +47,6 @@
  private:
   typename T::instance_type* isac_state_;
   int sample_rate_hz_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index 8bde0e3..c382ea0 100644
--- a/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -18,7 +18,6 @@
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/scoped_refptr.h"
 #include "api/units/time_delta.h"
-#include "rtc_base/constructor_magic.h"
 #include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
@@ -44,6 +43,9 @@
   explicit AudioEncoderIsacT(const Config& config);
   ~AudioEncoderIsacT() override;
 
+  AudioEncoderIsacT(const AudioEncoderIsacT&) = delete;
+  AudioEncoderIsacT& operator=(const AudioEncoderIsacT&) = delete;
+
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
@@ -99,8 +101,6 @@
   // Start out with a reasonable default that we can use until we receive a real
   // value.
   DataSize overhead_per_packet_ = DataSize::Bytes(28);
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
index efc3f0d..2ff47a8 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
+++ b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
@@ -21,7 +21,6 @@
 #include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h"
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -32,6 +31,11 @@
 
   ~AudioDecoderMultiChannelOpusImpl() override;
 
+  AudioDecoderMultiChannelOpusImpl(const AudioDecoderMultiChannelOpusImpl&) =
+      delete;
+  AudioDecoderMultiChannelOpusImpl& operator=(
+      const AudioDecoderMultiChannelOpusImpl&) = delete;
+
   std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
                                         uint32_t timestamp) override;
   void Reset() override;
@@ -63,7 +67,6 @@
 
   OpusDecInst* dec_state_;
   const AudioDecoderMultiChannelOpusConfig config_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderMultiChannelOpusImpl);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/modules/audio_coding/codecs/opus/audio_decoder_opus.h
index c792722..e8fd044 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus.h
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.h
@@ -19,7 +19,6 @@
 #include "api/audio_codecs/audio_decoder.h"
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -29,6 +28,9 @@
                                 int sample_rate_hz = 48000);
   ~AudioDecoderOpusImpl() override;
 
+  AudioDecoderOpusImpl(const AudioDecoderOpusImpl&) = delete;
+  AudioDecoderOpusImpl& operator=(const AudioDecoderOpusImpl&) = delete;
+
   std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
                                         uint32_t timestamp) override;
   void Reset() override;
@@ -55,7 +57,6 @@
   OpusDecInst* dec_state_;
   const size_t channels_;
   const int sample_rate_hz_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpusImpl);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
index eadb4a6..8a72105 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
+++ b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
@@ -21,7 +21,6 @@
 #include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
 #include "api/units/time_delta.h"
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -34,6 +33,11 @@
       int payload_type);
   ~AudioEncoderMultiChannelOpusImpl() override;
 
+  AudioEncoderMultiChannelOpusImpl(const AudioEncoderMultiChannelOpusImpl&) =
+      delete;
+  AudioEncoderMultiChannelOpusImpl& operator=(
+      const AudioEncoderMultiChannelOpusImpl&) = delete;
+
   // Static interface for use by BuiltinAudioEncoderFactory.
   static constexpr const char* GetPayloadName() { return "multiopus"; }
   static absl::optional<AudioCodecInfo> QueryAudioEncoder(
@@ -81,7 +85,6 @@
   int next_frame_length_ms_;
 
   friend struct AudioEncoderMultiChannelOpus;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderMultiChannelOpusImpl);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index c7ee4f4..14477cc 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -23,7 +23,6 @@
 #include "common_audio/smoothing_filter.h"
 #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -61,6 +60,9 @@
   AudioEncoderOpusImpl(int payload_type, const SdpAudioFormat& format);
   ~AudioEncoderOpusImpl() override;
 
+  AudioEncoderOpusImpl(const AudioEncoderOpusImpl&) = delete;
+  AudioEncoderOpusImpl& operator=(const AudioEncoderOpusImpl&) = delete;
+
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   int RtpTimestampRateHz() const override;
@@ -175,7 +177,6 @@
   int consecutive_dtx_frames_;
 
   friend struct AudioEncoderOpus;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpusImpl);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
index f08c4a6..6f50161 100644
--- a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
+++ b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
@@ -18,13 +18,16 @@
 
 #include "api/audio_codecs/audio_decoder.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
 class AudioDecoderPcm16B final : public AudioDecoder {
  public:
   AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels);
+
+  AudioDecoderPcm16B(const AudioDecoderPcm16B&) = delete;
+  AudioDecoderPcm16B& operator=(const AudioDecoderPcm16B&) = delete;
+
   void Reset() override;
   std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
                                         uint32_t timestamp) override;
@@ -42,7 +45,6 @@
  private:
   const int sample_rate_hz_;
   const size_t num_channels_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcm16B);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
index 71c7572..c363b40 100644
--- a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
+++ b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
@@ -12,7 +12,6 @@
 #define MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
 
 #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -29,6 +28,9 @@
   explicit AudioEncoderPcm16B(const Config& config)
       : AudioEncoderPcm(config, config.sample_rate_hz) {}
 
+  AudioEncoderPcm16B(const AudioEncoderPcm16B&) = delete;
+  AudioEncoderPcm16B& operator=(const AudioEncoderPcm16B&) = delete;
+
  protected:
   size_t EncodeCall(const int16_t* audio,
                     size_t input_len,
@@ -37,9 +39,6 @@
   size_t BytesPerSample() const override;
 
   AudioEncoder::CodecType GetCodecType() const override;
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcm16B);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index d5b1bf6..d163193 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -23,7 +23,6 @@
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/units/time_delta.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -47,6 +46,9 @@
 
   ~AudioEncoderCopyRed() override;
 
+  AudioEncoderCopyRed(const AudioEncoderCopyRed&) = delete;
+  AudioEncoderCopyRed& operator=(const AudioEncoderCopyRed&) = delete;
+
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   int RtpTimestampRateHz() const override;
@@ -92,8 +94,6 @@
   size_t max_packet_length_;
   int red_payload_type_;
   std::list<std::pair<EncodedInfo, rtc::Buffer>> redundant_encodings_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCopyRed);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/accelerate.h b/modules/audio_coding/neteq/accelerate.h
index e03f609..01fe874 100644
--- a/modules/audio_coding/neteq/accelerate.h
+++ b/modules/audio_coding/neteq/accelerate.h
@@ -15,7 +15,6 @@
 #include <stdint.h>
 
 #include "modules/audio_coding/neteq/time_stretch.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -33,6 +32,9 @@
              const BackgroundNoise& background_noise)
       : TimeStretch(sample_rate_hz, num_channels, background_noise) {}
 
+  Accelerate(const Accelerate&) = delete;
+  Accelerate& operator=(const Accelerate&) = delete;
+
   // This method performs the actual Accelerate operation. The samples are
   // read from `input`, of length `input_length` elements, and are written to
   // `output`. The number of samples removed through time-stretching is
@@ -62,9 +64,6 @@
                                       bool active_speech,
                                       bool fast_mode,
                                       AudioMultiVector* output) const override;
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(Accelerate);
 };
 
 struct AccelerateFactory {
diff --git a/modules/audio_coding/neteq/audio_multi_vector.h b/modules/audio_coding/neteq/audio_multi_vector.h
index 10179d7..715ec6d 100644
--- a/modules/audio_coding/neteq/audio_multi_vector.h
+++ b/modules/audio_coding/neteq/audio_multi_vector.h
@@ -18,7 +18,6 @@
 
 #include "api/array_view.h"
 #include "modules/audio_coding/neteq/audio_vector.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -34,6 +33,9 @@
 
   virtual ~AudioMultiVector();
 
+  AudioMultiVector(const AudioMultiVector&) = delete;
+  AudioMultiVector& operator=(const AudioMultiVector&) = delete;
+
   // Deletes all values and make the vector empty.
   virtual void Clear();
 
@@ -130,9 +132,6 @@
  protected:
   std::vector<AudioVector*> channels_;
   size_t num_channels_;
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioMultiVector);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/audio_vector.h b/modules/audio_coding/neteq/audio_vector.h
index c722b56..d68f3ec 100644
--- a/modules/audio_coding/neteq/audio_vector.h
+++ b/modules/audio_coding/neteq/audio_vector.h
@@ -17,7 +17,6 @@
 #include <memory>
 
 #include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -31,6 +30,9 @@
 
   virtual ~AudioVector();
 
+  AudioVector(const AudioVector&) = delete;
+  AudioVector& operator=(const AudioVector&) = delete;
+
   // Deletes all values and make the vector empty.
   virtual void Clear();
 
@@ -164,8 +166,6 @@
 
   // The index of the sample after the last sample in `array_`.
   size_t end_index_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioVector);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/background_noise.h b/modules/audio_coding/neteq/background_noise.h
index 005b376..8e6d589 100644
--- a/modules/audio_coding/neteq/background_noise.h
+++ b/modules/audio_coding/neteq/background_noise.h
@@ -16,7 +16,6 @@
 #include <memory>
 
 #include "api/array_view.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -34,6 +33,9 @@
   explicit BackgroundNoise(size_t num_channels);
   virtual ~BackgroundNoise();
 
+  BackgroundNoise(const BackgroundNoise&) = delete;
+  BackgroundNoise& operator=(const BackgroundNoise&) = delete;
+
   void Reset();
 
   // Updates the parameter estimates based on the signal currently in the
@@ -130,8 +132,6 @@
   size_t num_channels_;
   std::unique_ptr<ChannelParameters[]> channel_parameters_;
   bool initialized_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(BackgroundNoise);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/buffer_level_filter.h b/modules/audio_coding/neteq/buffer_level_filter.h
index 94a3715..ced36da 100644
--- a/modules/audio_coding/neteq/buffer_level_filter.h
+++ b/modules/audio_coding/neteq/buffer_level_filter.h
@@ -14,14 +14,16 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "rtc_base/constructor_magic.h"
-
 namespace webrtc {
 
 class BufferLevelFilter {
  public:
   BufferLevelFilter();
   virtual ~BufferLevelFilter() {}
+
+  BufferLevelFilter(const BufferLevelFilter&) = delete;
+  BufferLevelFilter& operator=(const BufferLevelFilter&) = delete;
+
   virtual void Reset();
 
   // Updates the filter. Current buffer size is `buffer_size_samples`.
@@ -46,8 +48,6 @@
  private:
   int level_factor_;  // Filter factor for the buffer level filter in Q8.
   int filtered_current_level_;  // Filtered current buffer level in Q8.
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(BufferLevelFilter);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/comfort_noise.h b/modules/audio_coding/neteq/comfort_noise.h
index 6419d39..31fcee3 100644
--- a/modules/audio_coding/neteq/comfort_noise.h
+++ b/modules/audio_coding/neteq/comfort_noise.h
@@ -13,8 +13,6 @@
 
 #include <stddef.h>
 
-#include "rtc_base/constructor_magic.h"
-
 namespace webrtc {
 
 // Forward declarations.
@@ -42,6 +40,9 @@
         decoder_database_(decoder_database),
         sync_buffer_(sync_buffer) {}
 
+  ComfortNoise(const ComfortNoise&) = delete;
+  ComfortNoise& operator=(const ComfortNoise&) = delete;
+
   // Resets the state. Should be called before each new comfort noise period.
   void Reset();
 
@@ -65,7 +66,6 @@
   DecoderDatabase* decoder_database_;
   SyncBuffer* sync_buffer_;
   int internal_error_code_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(ComfortNoise);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/decision_logic.h b/modules/audio_coding/neteq/decision_logic.h
index 693f616..a8571ad 100644
--- a/modules/audio_coding/neteq/decision_logic.h
+++ b/modules/audio_coding/neteq/decision_logic.h
@@ -18,7 +18,6 @@
 #include "api/neteq/tick_timer.h"
 #include "modules/audio_coding/neteq/buffer_level_filter.h"
 #include "modules/audio_coding/neteq/delay_manager.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/experiments/field_trial_parser.h"
 
 namespace webrtc {
@@ -37,6 +36,9 @@
 
   ~DecisionLogic() override;
 
+  DecisionLogic(const DecisionLogic&) = delete;
+  DecisionLogic& operator=(const DecisionLogic&) = delete;
+
   // Resets object to a clean state.
   void Reset() override;
 
@@ -192,8 +194,6 @@
   FieldTrialParameter<bool> estimate_dtx_delay_;
   FieldTrialParameter<bool> time_stretch_cn_;
   FieldTrialConstrained<int> target_level_window_ms_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogic);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/decoder_database.h b/modules/audio_coding/neteq/decoder_database.h
index a63a9cf..6c2ce54 100644
--- a/modules/audio_coding/neteq/decoder_database.h
+++ b/modules/audio_coding/neteq/decoder_database.h
@@ -20,7 +20,6 @@
 #include "api/scoped_refptr.h"
 #include "modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "modules/audio_coding/neteq/packet.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -122,6 +121,9 @@
 
   virtual ~DecoderDatabase();
 
+  DecoderDatabase(const DecoderDatabase&) = delete;
+  DecoderDatabase& operator=(const DecoderDatabase&) = delete;
+
   // Returns true if the database is empty.
   virtual bool Empty() const;
 
@@ -208,8 +210,6 @@
   mutable std::unique_ptr<ComfortNoiseDecoder> active_cng_decoder_;
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
   const absl::optional<AudioCodecPairId> codec_pair_id_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(DecoderDatabase);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/delay_manager.h b/modules/audio_coding/neteq/delay_manager.h
index 410aa94..56d108a 100644
--- a/modules/audio_coding/neteq/delay_manager.h
+++ b/modules/audio_coding/neteq/delay_manager.h
@@ -22,7 +22,6 @@
 #include "modules/audio_coding/neteq/relative_arrival_delay_tracker.h"
 #include "modules/audio_coding/neteq/reorder_optimizer.h"
 #include "modules/audio_coding/neteq/underrun_optimizer.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -52,6 +51,9 @@
 
   virtual ~DelayManager();
 
+  DelayManager(const DelayManager&) = delete;
+  DelayManager& operator=(const DelayManager&) = delete;
+
   // Updates the delay manager with a new incoming packet, with `timestamp` from
   // the RTP header. This updates the statistics and a new target buffer level
   // is calculated. Returns the relative delay if it can be calculated. If
@@ -111,9 +113,7 @@
   int maximum_delay_ms_;            // Externally set maximum allowed delay.
 
   int packet_len_ms_ = 0;
-  int target_level_ms_;       // Currently preferred buffer level.
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(DelayManager);
+  int target_level_ms_;  // Currently preferred buffer level.
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/dsp_helper.h b/modules/audio_coding/neteq/dsp_helper.h
index 7bdeba6..4aead7d 100644
--- a/modules/audio_coding/neteq/dsp_helper.h
+++ b/modules/audio_coding/neteq/dsp_helper.h
@@ -16,7 +16,6 @@
 
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
 #include "modules/audio_coding/neteq/audio_vector.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -150,11 +149,12 @@
                               bool compensate_delay,
                               int16_t* output);
 
+  DspHelper(const DspHelper&) = delete;
+  DspHelper& operator=(const DspHelper&) = delete;
+
  private:
   // Table of constants used in method DspHelper::ParabolicFit().
   static const int16_t kParabolaCoefficients[17][3];
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(DspHelper);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/dtmf_buffer.h b/modules/audio_coding/neteq/dtmf_buffer.h
index 9209cae..62b7515 100644
--- a/modules/audio_coding/neteq/dtmf_buffer.h
+++ b/modules/audio_coding/neteq/dtmf_buffer.h
@@ -16,8 +16,6 @@
 
 #include <list>
 
-#include "rtc_base/constructor_magic.h"
-
 namespace webrtc {
 
 struct DtmfEvent {
@@ -50,6 +48,9 @@
 
   virtual ~DtmfBuffer();
 
+  DtmfBuffer(const DtmfBuffer&) = delete;
+  DtmfBuffer& operator=(const DtmfBuffer&) = delete;
+
   // Flushes the buffer.
   virtual void Flush();
 
@@ -97,8 +98,6 @@
   static bool CompareEvents(const DtmfEvent& a, const DtmfEvent& b);
 
   DtmfList buffer_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(DtmfBuffer);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/dtmf_tone_generator.h b/modules/audio_coding/neteq/dtmf_tone_generator.h
index 968bc7f..35114f4 100644
--- a/modules/audio_coding/neteq/dtmf_tone_generator.h
+++ b/modules/audio_coding/neteq/dtmf_tone_generator.h
@@ -15,7 +15,6 @@
 #include <stdint.h>
 
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -29,6 +28,10 @@
 
   DtmfToneGenerator();
   virtual ~DtmfToneGenerator() {}
+
+  DtmfToneGenerator(const DtmfToneGenerator&) = delete;
+  DtmfToneGenerator& operator=(const DtmfToneGenerator&) = delete;
+
   virtual int Init(int fs, int event, int attenuation);
   virtual void Reset();
   virtual int Generate(size_t num_samples, AudioMultiVector* output);
@@ -48,8 +51,6 @@
   int amplitude_;               // Amplitude for this event.
   int16_t sample_history1_[2];  // Last 2 samples for the 1st oscillator.
   int16_t sample_history2_[2];  // Last 2 samples for the 2nd oscillator.
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(DtmfToneGenerator);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/expand.h b/modules/audio_coding/neteq/expand.h
index 2d22b11..2e64583 100644
--- a/modules/audio_coding/neteq/expand.h
+++ b/modules/audio_coding/neteq/expand.h
@@ -15,7 +15,6 @@
 #include <memory>
 
 #include "modules/audio_coding/neteq/audio_vector.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -41,6 +40,9 @@
 
   virtual ~Expand();
 
+  Expand(const Expand&) = delete;
+  Expand& operator=(const Expand&) = delete;
+
   // Resets the object.
   virtual void Reset();
 
@@ -134,8 +136,6 @@
   bool stop_muting_;
   size_t expand_duration_samples_;
   std::unique_ptr<ChannelParameters[]> channel_parameters_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Expand);
 };
 
 struct ExpandFactory {
diff --git a/modules/audio_coding/neteq/expand_uma_logger.h b/modules/audio_coding/neteq/expand_uma_logger.h
index 246aaff..a29d353 100644
--- a/modules/audio_coding/neteq/expand_uma_logger.h
+++ b/modules/audio_coding/neteq/expand_uma_logger.h
@@ -17,7 +17,6 @@
 
 #include "absl/types/optional.h"
 #include "api/neteq/tick_timer.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -36,6 +35,9 @@
 
   ~ExpandUmaLogger();
 
+  ExpandUmaLogger(const ExpandUmaLogger&) = delete;
+  ExpandUmaLogger& operator=(const ExpandUmaLogger&) = delete;
+
   // In this call, value should be an incremental sample counter. The sample
   // rate must be strictly positive.
   void UpdateSampleCounter(uint64_t value, int sample_rate_hz);
@@ -48,8 +50,6 @@
   absl::optional<uint64_t> last_logged_value_;
   uint64_t last_value_ = 0;
   int sample_rate_hz_ = 0;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(ExpandUmaLogger);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/merge.h b/modules/audio_coding/neteq/merge.h
index 13aa31d..2f27106 100644
--- a/modules/audio_coding/neteq/merge.h
+++ b/modules/audio_coding/neteq/merge.h
@@ -12,7 +12,6 @@
 #define MODULES_AUDIO_CODING_NETEQ_MERGE_H_
 
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -36,6 +35,9 @@
         SyncBuffer* sync_buffer);
   virtual ~Merge();
 
+  Merge(const Merge&) = delete;
+  Merge& operator=(const Merge&) = delete;
+
   // The main method to produce the audio data. The decoded data is supplied in
   // `input`, having `input_length` samples in total for all channels
   // (interleaved). The result is written to `output`. The number of channels
@@ -93,8 +95,6 @@
   int16_t input_downsampled_[kInputDownsampLength];
   AudioMultiVector expanded_;
   std::vector<int16_t> temp_data_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Merge);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index 2522e31..e2cd6c6 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -29,7 +29,6 @@
 #include "modules/audio_coding/neteq/packet.h"
 #include "modules/audio_coding/neteq/random_vector.h"
 #include "modules/audio_coding/neteq/statistics_calculator.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
 
@@ -124,6 +123,9 @@
 
   ~NetEqImpl() override;
 
+  NetEqImpl(const NetEqImpl&) = delete;
+  NetEqImpl& operator=(const NetEqImpl&) = delete;
+
   // Inserts a new packet into NetEq. Returns 0 on success, -1 on failure.
   int InsertPacket(const RTPHeader& rtp_header,
                    rtc::ArrayView<const uint8_t> payload) override;
@@ -399,9 +401,6 @@
   ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(mutex_);
   bool no_time_stretching_ RTC_GUARDED_BY(mutex_);  // Only used for test.
   rtc::BufferT<int16_t> concealment_audio_ RTC_GUARDED_BY(mutex_);
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/normal.h b/modules/audio_coding/neteq/normal.h
index 3607208..772293b 100644
--- a/modules/audio_coding/neteq/normal.h
+++ b/modules/audio_coding/neteq/normal.h
@@ -17,7 +17,6 @@
 #include "api/neteq/neteq.h"
 #include "modules/audio_coding/neteq/statistics_calculator.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/numerics/safe_conversions.h"
 
 namespace webrtc {
@@ -49,6 +48,9 @@
 
   virtual ~Normal() {}
 
+  Normal(const Normal&) = delete;
+  Normal& operator=(const Normal&) = delete;
+
   // Performs the "Normal" operation. The decoder data is supplied in `input`,
   // having `length` samples in total for all channels (interleaved). The
   // result is written to `output`. The number of channels allocated in
@@ -68,8 +70,6 @@
   const size_t samples_per_ms_;
   const int16_t default_win_slope_Q14_;
   StatisticsCalculator* const statistics_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Normal);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/packet_buffer.h b/modules/audio_coding/neteq/packet_buffer.h
index 20a0533..c6fb47f 100644
--- a/modules/audio_coding/neteq/packet_buffer.h
+++ b/modules/audio_coding/neteq/packet_buffer.h
@@ -15,7 +15,6 @@
 #include "modules/audio_coding/neteq/decoder_database.h"
 #include "modules/audio_coding/neteq/packet.h"
 #include "modules/include/module_common_types_public.h"  // IsNewerTimestamp
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -51,6 +50,9 @@
   // Deletes all packets in the buffer before destroying the buffer.
   virtual ~PacketBuffer();
 
+  PacketBuffer(const PacketBuffer&) = delete;
+  PacketBuffer& operator=(const PacketBuffer&) = delete;
+
   // Flushes the buffer and deletes all packets in it.
   virtual void Flush(StatisticsCalculator* stats);
 
@@ -173,7 +175,6 @@
   size_t max_number_of_packets_;
   PacketList buffer_;
   const TickTimer* tick_timer_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(PacketBuffer);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/post_decode_vad.h b/modules/audio_coding/neteq/post_decode_vad.h
index 3134d5f..3bd91b9 100644
--- a/modules/audio_coding/neteq/post_decode_vad.h
+++ b/modules/audio_coding/neteq/post_decode_vad.h
@@ -16,7 +16,6 @@
 
 #include "api/audio_codecs/audio_decoder.h"
 #include "common_audio/vad/include/webrtc_vad.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -31,6 +30,9 @@
 
   virtual ~PostDecodeVad();
 
+  PostDecodeVad(const PostDecodeVad&) = delete;
+  PostDecodeVad& operator=(const PostDecodeVad&) = delete;
+
   // Enables post-decode VAD.
   void Enable();
 
@@ -63,8 +65,6 @@
   bool active_speech_;
   int sid_interval_counter_;
   ::VadInst* vad_instance_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(PostDecodeVad);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/preemptive_expand.h b/modules/audio_coding/neteq/preemptive_expand.h
index 708ebfd..6338b99 100644
--- a/modules/audio_coding/neteq/preemptive_expand.h
+++ b/modules/audio_coding/neteq/preemptive_expand.h
@@ -15,7 +15,6 @@
 #include <stdint.h>
 
 #include "modules/audio_coding/neteq/time_stretch.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -36,6 +35,9 @@
         old_data_length_per_channel_(0),
         overlap_samples_(overlap_samples) {}
 
+  PreemptiveExpand(const PreemptiveExpand&) = delete;
+  PreemptiveExpand& operator=(const PreemptiveExpand&) = delete;
+
   // This method performs the actual PreemptiveExpand operation. The samples are
   // read from `input`, of length `input_length` elements, and are written to
   // `output`. The number of samples added through time-stretching is
@@ -67,8 +69,6 @@
  private:
   size_t old_data_length_per_channel_;
   size_t overlap_samples_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(PreemptiveExpand);
 };
 
 struct PreemptiveExpandFactory {
diff --git a/modules/audio_coding/neteq/random_vector.h b/modules/audio_coding/neteq/random_vector.h
index 1d37600..4a782f1 100644
--- a/modules/audio_coding/neteq/random_vector.h
+++ b/modules/audio_coding/neteq/random_vector.h
@@ -14,8 +14,6 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "rtc_base/constructor_magic.h"
-
 namespace webrtc {
 
 // This class generates pseudo-random samples.
@@ -26,6 +24,9 @@
 
   RandomVector() : seed_(777), seed_increment_(1) {}
 
+  RandomVector(const RandomVector&) = delete;
+  RandomVector& operator=(const RandomVector&) = delete;
+
   void Reset();
 
   void Generate(size_t length, int16_t* output);
@@ -39,8 +40,6 @@
  private:
   uint32_t seed_;
   int16_t seed_increment_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(RandomVector);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/red_payload_splitter.h b/modules/audio_coding/neteq/red_payload_splitter.h
index 5566091..2f48e4b 100644
--- a/modules/audio_coding/neteq/red_payload_splitter.h
+++ b/modules/audio_coding/neteq/red_payload_splitter.h
@@ -12,7 +12,6 @@
 #define MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_
 
 #include "modules/audio_coding/neteq/packet.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -30,6 +29,9 @@
 
   virtual ~RedPayloadSplitter() {}
 
+  RedPayloadSplitter(const RedPayloadSplitter&) = delete;
+  RedPayloadSplitter& operator=(const RedPayloadSplitter&) = delete;
+
   // Splits each packet in `packet_list` into its separate RED payloads. Each
   // RED payload is packetized into a Packet. The original elements in
   // `packet_list` are properly deleted, and replaced by the new packets.
@@ -43,9 +45,6 @@
   // is accepted. Any packet with another payload type is discarded.
   virtual void CheckRedPayloads(PacketList* packet_list,
                                 const DecoderDatabase& decoder_database);
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(RedPayloadSplitter);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/statistics_calculator.h b/modules/audio_coding/neteq/statistics_calculator.h
index 5c3fb75..269e6a0 100644
--- a/modules/audio_coding/neteq/statistics_calculator.h
+++ b/modules/audio_coding/neteq/statistics_calculator.h
@@ -15,7 +15,6 @@
 #include <string>
 
 #include "api/neteq/neteq.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -28,6 +27,9 @@
 
   virtual ~StatisticsCalculator();
 
+  StatisticsCalculator(const StatisticsCalculator&) = delete;
+  StatisticsCalculator& operator=(const StatisticsCalculator&) = delete;
+
   // Resets most of the counters.
   void Reset();
 
@@ -197,8 +199,6 @@
   PeriodicUmaAverage excess_buffer_delay_;
   PeriodicUmaCount buffer_full_counter_;
   bool decoded_output_played_ = false;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(StatisticsCalculator);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/sync_buffer.h b/modules/audio_coding/neteq/sync_buffer.h
index 7d24730..cf56c43 100644
--- a/modules/audio_coding/neteq/sync_buffer.h
+++ b/modules/audio_coding/neteq/sync_buffer.h
@@ -20,7 +20,6 @@
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
 #include "modules/audio_coding/neteq/audio_vector.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -32,6 +31,9 @@
         end_timestamp_(0),
         dtmf_index_(0) {}
 
+  SyncBuffer(const SyncBuffer&) = delete;
+  SyncBuffer& operator=(const SyncBuffer&) = delete;
+
   // Returns the number of samples yet to play out from the buffer.
   size_t FutureLength() const;
 
@@ -102,8 +104,6 @@
   size_t next_index_;
   uint32_t end_timestamp_;  // The timestamp of the last sample in the buffer.
   size_t dtmf_index_;       // Index to the first non-DTMF sample in the buffer.
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(SyncBuffer);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/time_stretch.h b/modules/audio_coding/neteq/time_stretch.h
index 998d080..f0ddaeb 100644
--- a/modules/audio_coding/neteq/time_stretch.h
+++ b/modules/audio_coding/neteq/time_stretch.h
@@ -14,7 +14,6 @@
 #include <string.h>  // memset, size_t
 
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -49,6 +48,9 @@
 
   virtual ~TimeStretch() {}
 
+  TimeStretch(const TimeStretch&) = delete;
+  TimeStretch& operator=(const TimeStretch&) = delete;
+
   // This method performs the processing common to both Accelerate and
   // PreemptiveExpand.
   ReturnCodes Process(const int16_t* input,
@@ -105,8 +107,6 @@
                        int32_t vec2_energy,
                        size_t peak_index,
                        int scaling) const;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(TimeStretch);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/timestamp_scaler.h b/modules/audio_coding/neteq/timestamp_scaler.h
index 4d578fc..f42ce72 100644
--- a/modules/audio_coding/neteq/timestamp_scaler.h
+++ b/modules/audio_coding/neteq/timestamp_scaler.h
@@ -12,7 +12,6 @@
 #define MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
 
 #include "modules/audio_coding/neteq/packet.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -34,6 +33,9 @@
 
   virtual ~TimestampScaler() {}
 
+  TimestampScaler(const TimestampScaler&) = delete;
+  TimestampScaler& operator=(const TimestampScaler&) = delete;
+
   // Start over.
   virtual void Reset();
 
@@ -59,8 +61,6 @@
   uint32_t external_ref_;
   uint32_t internal_ref_;
   const DecoderDatabase& decoder_database_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(TimestampScaler);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/tools/audio_checksum.h b/modules/audio_coding/neteq/tools/audio_checksum.h
index e4306fa..9d6f343 100644
--- a/modules/audio_coding/neteq/tools/audio_checksum.h
+++ b/modules/audio_coding/neteq/tools/audio_checksum.h
@@ -16,7 +16,6 @@
 
 #include "modules/audio_coding/neteq/tools/audio_sink.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/message_digest.h"
 #include "rtc_base/string_encode.h"
 #include "rtc_base/system/arch.h"
@@ -31,6 +30,9 @@
         checksum_result_(checksum_->Size()),
         finished_(false) {}
 
+  AudioChecksum(const AudioChecksum&) = delete;
+  AudioChecksum& operator=(const AudioChecksum&) = delete;
+
   bool WriteArray(const int16_t* audio, size_t num_samples) override {
     if (finished_)
       return false;
@@ -56,8 +58,6 @@
   std::unique_ptr<rtc::MessageDigest> checksum_;
   rtc::Buffer checksum_result_;
   bool finished_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioChecksum);
 };
 
 }  // namespace test
diff --git a/modules/audio_coding/neteq/tools/audio_loop.h b/modules/audio_coding/neteq/tools/audio_loop.h
index 25da463..a73be2d 100644
--- a/modules/audio_coding/neteq/tools/audio_loop.h
+++ b/modules/audio_coding/neteq/tools/audio_loop.h
@@ -15,7 +15,6 @@
 #include <string>
 
 #include "api/array_view.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -29,6 +28,9 @@
 
   virtual ~AudioLoop() {}
 
+  AudioLoop(const AudioLoop&) = delete;
+  AudioLoop& operator=(const AudioLoop&) = delete;
+
   // Initializes the AudioLoop by reading from `file_name`. The loop will be no
   // longer than `max_loop_length_samples`, if the length of the file is
   // greater. Otherwise, the loop length is the same as the file length.
@@ -47,8 +49,6 @@
   size_t loop_length_samples_;
   size_t block_length_samples_;
   std::unique_ptr<int16_t[]> audio_array_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioLoop);
 };
 
 }  // namespace test
diff --git a/modules/audio_coding/neteq/tools/audio_sink.h b/modules/audio_coding/neteq/tools/audio_sink.h
index cd6733b..53729fa 100644
--- a/modules/audio_coding/neteq/tools/audio_sink.h
+++ b/modules/audio_coding/neteq/tools/audio_sink.h
@@ -12,7 +12,6 @@
 #define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
 
 #include "api/audio/audio_frame.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -24,6 +23,9 @@
   AudioSink() {}
   virtual ~AudioSink() {}
 
+  AudioSink(const AudioSink&) = delete;
+  AudioSink& operator=(const AudioSink&) = delete;
+
   // Writes `num_samples` from `audio` to the AudioSink. Returns true if
   // successful, otherwise false.
   virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0;
@@ -34,9 +36,6 @@
     return WriteArray(audio_frame.data(), audio_frame.samples_per_channel_ *
                                               audio_frame.num_channels_);
   }
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioSink);
 };
 
 // Forks the output audio to two AudioSink objects.
@@ -45,23 +44,25 @@
   AudioSinkFork(AudioSink* left, AudioSink* right)
       : left_sink_(left), right_sink_(right) {}
 
+  AudioSinkFork(const AudioSinkFork&) = delete;
+  AudioSinkFork& operator=(const AudioSinkFork&) = delete;
+
   bool WriteArray(const int16_t* audio, size_t num_samples) override;
 
  private:
   AudioSink* left_sink_;
   AudioSink* right_sink_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioSinkFork);
 };
 
 // An AudioSink implementation that does nothing.
 class VoidAudioSink : public AudioSink {
  public:
   VoidAudioSink() = default;
-  bool WriteArray(const int16_t* audio, size_t num_samples) override;
 
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(VoidAudioSink);
+  VoidAudioSink(const VoidAudioSink&) = delete;
+  VoidAudioSink& operator=(const VoidAudioSink&) = delete;
+
+  bool WriteArray(const int16_t* audio, size_t num_samples) override;
 };
 
 }  // namespace test
diff --git a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
index 6a79ce4..ab4f5c2 100644
--- a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
+++ b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
@@ -16,7 +16,6 @@
 #include <string>
 
 #include "modules/audio_coding/neteq/tools/packet_source.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -31,6 +30,9 @@
                           int sample_rate_hz,
                           int payload_type);
 
+  ConstantPcmPacketSource(const ConstantPcmPacketSource&) = delete;
+  ConstantPcmPacketSource& operator=(const ConstantPcmPacketSource&) = delete;
+
   std::unique_ptr<Packet> NextPacket() override;
 
  private:
@@ -46,8 +48,6 @@
   uint16_t seq_number_;
   uint32_t timestamp_;
   const uint32_t payload_ssrc_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(ConstantPcmPacketSource);
 };
 
 }  // namespace test
diff --git a/modules/audio_coding/neteq/tools/input_audio_file.h b/modules/audio_coding/neteq/tools/input_audio_file.h
index 010d8cc..c6e65a0 100644
--- a/modules/audio_coding/neteq/tools/input_audio_file.h
+++ b/modules/audio_coding/neteq/tools/input_audio_file.h
@@ -15,8 +15,6 @@
 
 #include <string>
 
-#include "rtc_base/constructor_magic.h"
-
 namespace webrtc {
 namespace test {
 
@@ -27,6 +25,9 @@
 
   virtual ~InputAudioFile();
 
+  InputAudioFile(const InputAudioFile&) = delete;
+  InputAudioFile& operator=(const InputAudioFile&) = delete;
+
   // Reads `samples` elements from source file to `destination`. Returns true
   // if the read was successful, otherwise false. If the file end is reached,
   // the file is rewound and reading continues from the beginning.
@@ -52,7 +53,6 @@
  private:
   FILE* fp_;
   const bool loop_at_end_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(InputAudioFile);
 };
 
 }  // namespace test
diff --git a/modules/audio_coding/neteq/tools/output_audio_file.h b/modules/audio_coding/neteq/tools/output_audio_file.h
index ad97722..491cbd0 100644
--- a/modules/audio_coding/neteq/tools/output_audio_file.h
+++ b/modules/audio_coding/neteq/tools/output_audio_file.h
@@ -16,7 +16,6 @@
 #include <string>
 
 #include "modules/audio_coding/neteq/tools/audio_sink.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -34,6 +33,9 @@
       fclose(out_file_);
   }
 
+  OutputAudioFile(const OutputAudioFile&) = delete;
+  OutputAudioFile& operator=(const OutputAudioFile&) = delete;
+
   bool WriteArray(const int16_t* audio, size_t num_samples) override {
     RTC_DCHECK(out_file_);
     return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples;
@@ -41,8 +43,6 @@
 
  private:
   FILE* out_file_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(OutputAudioFile);
 };
 
 }  // namespace test
diff --git a/modules/audio_coding/neteq/tools/output_wav_file.h b/modules/audio_coding/neteq/tools/output_wav_file.h
index ae2e970..1485f4e 100644
--- a/modules/audio_coding/neteq/tools/output_wav_file.h
+++ b/modules/audio_coding/neteq/tools/output_wav_file.h
@@ -15,7 +15,6 @@
 
 #include "common_audio/wav_file.h"
 #include "modules/audio_coding/neteq/tools/audio_sink.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -29,6 +28,9 @@
                 int num_channels = 1)
       : wav_writer_(file_name, sample_rate_hz, num_channels) {}
 
+  OutputWavFile(const OutputWavFile&) = delete;
+  OutputWavFile& operator=(const OutputWavFile&) = delete;
+
   bool WriteArray(const int16_t* audio, size_t num_samples) override {
     wav_writer_.WriteSamples(audio, num_samples);
     return true;
@@ -36,8 +38,6 @@
 
  private:
   WavWriter wav_writer_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(OutputWavFile);
 };
 
 }  // namespace test
diff --git a/modules/audio_coding/neteq/tools/packet.h b/modules/audio_coding/neteq/tools/packet.h
index 92e5ee9..9671090 100644
--- a/modules/audio_coding/neteq/tools/packet.h
+++ b/modules/audio_coding/neteq/tools/packet.h
@@ -16,7 +16,6 @@
 #include "api/array_view.h"
 #include "api/rtp_headers.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/copy_on_write_buffer.h"
 
 namespace webrtc {
@@ -54,6 +53,9 @@
 
   virtual ~Packet();
 
+  Packet(const Packet&) = delete;
+  Packet& operator=(const Packet&) = delete;
+
   // Parses the first bytes of the RTP payload, interpreting them as RED headers
   // according to RFC 2198. The headers will be inserted into `headers`. The
   // caller of the method assumes ownership of the objects in the list, and
@@ -95,8 +97,6 @@
   size_t virtual_payload_length_bytes_ = 0;
   const double time_ms_;     // Used to denote a packet's arrival time.
   const bool valid_header_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Packet);
 };
 
 }  // namespace test
diff --git a/modules/audio_coding/neteq/tools/packet_source.h b/modules/audio_coding/neteq/tools/packet_source.h
index 975680f..be1705c 100644
--- a/modules/audio_coding/neteq/tools/packet_source.h
+++ b/modules/audio_coding/neteq/tools/packet_source.h
@@ -15,7 +15,6 @@
 #include <memory>
 
 #include "modules/audio_coding/neteq/tools/packet.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -26,6 +25,9 @@
   PacketSource();
   virtual ~PacketSource();
 
+  PacketSource(const PacketSource&) = delete;
+  PacketSource& operator=(const PacketSource&) = delete;
+
   // Returns next packet. Returns nullptr if the source is depleted, or if an
   // error occurred.
   virtual std::unique_ptr<Packet> NextPacket() = 0;
@@ -34,9 +36,6 @@
 
  protected:
   std::bitset<128> filter_;  // Payload type is 7 bits in the RFC.
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(PacketSource);
 };
 
 }  // namespace test
diff --git a/modules/audio_coding/neteq/tools/resample_input_audio_file.h b/modules/audio_coding/neteq/tools/resample_input_audio_file.h
index 9106d5b..497a410 100644
--- a/modules/audio_coding/neteq/tools/resample_input_audio_file.h
+++ b/modules/audio_coding/neteq/tools/resample_input_audio_file.h
@@ -15,7 +15,6 @@
 
 #include "common_audio/resampler/include/resampler.h"
 #include "modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -37,6 +36,9 @@
         file_rate_hz_(file_rate_hz),
         output_rate_hz_(output_rate_hz) {}
 
+  ResampleInputAudioFile(const ResampleInputAudioFile&) = delete;
+  ResampleInputAudioFile& operator=(const ResampleInputAudioFile&) = delete;
+
   bool Read(size_t samples, int output_rate_hz, int16_t* destination);
   bool Read(size_t samples, int16_t* destination) override;
   void set_output_rate_hz(int rate_hz);
@@ -45,7 +47,6 @@
   const int file_rate_hz_;
   int output_rate_hz_;
   Resampler resampler_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(ResampleInputAudioFile);
 };
 
 }  // namespace test
diff --git a/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/modules/audio_coding/neteq/tools/rtc_event_log_source.h
index d4be2a7..e2d0f61 100644
--- a/modules/audio_coding/neteq/tools/rtc_event_log_source.h
+++ b/modules/audio_coding/neteq/tools/rtc_event_log_source.h
@@ -19,7 +19,6 @@
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
 #include "modules/audio_coding/neteq/tools/packet_source.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -43,6 +42,9 @@
 
   virtual ~RtcEventLogSource();
 
+  RtcEventLogSource(const RtcEventLogSource&) = delete;
+  RtcEventLogSource& operator=(const RtcEventLogSource&) = delete;
+
   std::unique_ptr<Packet> NextPacket() override;
 
   // Returns the timestamp of the next audio output event, in milliseconds. The
@@ -60,8 +62,6 @@
   size_t rtp_packet_index_ = 0;
   std::vector<int64_t> audio_outputs_;
   size_t audio_output_index_ = 0;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
 };
 
 }  // namespace test
diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.h b/modules/audio_coding/neteq/tools/rtp_file_source.h
index d6aab24..7e284ac 100644
--- a/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -19,7 +19,6 @@
 #include "absl/types/optional.h"
 #include "modules/audio_coding/neteq/tools/packet_source.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -41,6 +40,9 @@
 
   ~RtpFileSource() override;
 
+  RtpFileSource(const RtpFileSource&) = delete;
+  RtpFileSource& operator=(const RtpFileSource&) = delete;
+
   // Registers an RTP header extension and binds it to `id`.
   virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
 
@@ -58,8 +60,6 @@
   std::unique_ptr<RtpFileReader> rtp_reader_;
   const absl::optional<uint32_t> ssrc_filter_;
   RtpHeaderExtensionMap rtp_header_extension_map_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
 };
 
 }  // namespace test
diff --git a/modules/audio_coding/neteq/tools/rtp_generator.h b/modules/audio_coding/neteq/tools/rtp_generator.h
index 6ca6e1b..2e615ad 100644
--- a/modules/audio_coding/neteq/tools/rtp_generator.h
+++ b/modules/audio_coding/neteq/tools/rtp_generator.h
@@ -12,7 +12,6 @@
 #define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
 
 #include "api/rtp_headers.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 namespace test {
@@ -34,6 +33,9 @@
 
   virtual ~RtpGenerator() {}
 
+  RtpGenerator(const RtpGenerator&) = delete;
+  RtpGenerator& operator=(const RtpGenerator&) = delete;
+
   // Writes the next RTP header to `rtp_header`, which will be of type
   // `payload_type`. Returns the send time for this packet (in ms). The value of
   // `payload_length_samples` determines the send time for the next packet.
@@ -50,9 +52,6 @@
   const uint32_t ssrc_;
   const int samples_per_ms_;
   double drift_factor_;
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
 };
 
 class TimestampJumpRtpGenerator : public RtpGenerator {
@@ -66,6 +65,10 @@
         jump_from_timestamp_(jump_from_timestamp),
         jump_to_timestamp_(jump_to_timestamp) {}
 
+  TimestampJumpRtpGenerator(const TimestampJumpRtpGenerator&) = delete;
+  TimestampJumpRtpGenerator& operator=(const TimestampJumpRtpGenerator&) =
+      delete;
+
   uint32_t GetRtpHeader(uint8_t payload_type,
                         size_t payload_length_samples,
                         RTPHeader* rtp_header) override;
@@ -73,7 +76,6 @@
  private:
   uint32_t jump_from_timestamp_;
   uint32_t jump_to_timestamp_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator);
 };
 
 }  // namespace test