Delete macros RTC_DISALLOW_ASSIGN and RTC_DISALLOW_IMPLICIT_CONSTRUCTORS

The former was unused, the latter is replaced with the explicit C++11 deletions. The related RTC_DISALLOW_COPY_AND_ASSIGN is left for now, it is used in a lot more places. Bug: None Change-Id: I49503e7f2b9ff43c6285f8695833479bbc18c380 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185500 Reviewed-by: Karl Wiberg <[email protected]> Commit-Queue: Niels Moller <[email protected]> Cr-Commit-Position: refs/heads/master@{#32224}
cdx08222028 · Sep 29, 2020 · de95329 · de95329
1 parent 46be80d
commit de95329
Show file tree

Hide file tree

Showing 55 changed files with 256 additions and 172 deletions.
diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn
@@ -144,7 +144,6 @@ if (rtc_include_tests) {
     deps = [
       ":stun_types",
       "../../rtc_base",
-      "../../rtc_base:macromagic",
       "../../rtc_base:rtc_base_approved",
       "../../test:test_support",
       "//testing/gtest",

diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
@@ -21,7 +21,6 @@
 #include "call/audio_receive_stream.h"
 #include "call/syncable.h"
 #include "modules/rtp_rtcp/source/source_tracker.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/thread_checker.h"
 #include "system_wrappers/include/clock.h"
 
@@ -61,6 +60,11 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
       const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
       webrtc::RtcEventLog* event_log,
       std::unique_ptr<voe::ChannelReceiveInterface> channel_receive);
+
+  AudioReceiveStream() = delete;
+  AudioReceiveStream(const AudioReceiveStream&) = delete;
+  AudioReceiveStream& operator=(const AudioReceiveStream&) = delete;
+
   ~AudioReceiveStream() override;
 
   // webrtc::AudioReceiveStream implementation.
@@ -113,8 +117,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;
 
   std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_;
-
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
 };
 }  // namespace internal
 }  // namespace webrtc

diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
@@ -21,7 +21,6 @@
 #include "call/audio_state.h"
 #include "call/bitrate_allocator.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/experiments/struct_parameters_parser.h"
 #include "rtc_base/race_checker.h"
 #include "rtc_base/synchronization/mutex.h"
@@ -75,6 +74,11 @@ class AudioSendStream final : public webrtc::AudioSendStream,
                   RtcEventLog* event_log,
                   const absl::optional<RtpState>& suspended_rtp_state,
                   std::unique_ptr<voe::ChannelSendInterface> channel_send);
+
+  AudioSendStream() = delete;
+  AudioSendStream(const AudioSendStream&) = delete;
+  AudioSendStream& operator=(const AudioSendStream&) = delete;
+
   ~AudioSendStream() override;
 
   // webrtc::AudioSendStream implementation.
@@ -206,8 +210,6 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   size_t total_packet_overhead_bytes_ RTC_GUARDED_BY(worker_queue_) = 0;
   absl::optional<std::pair<TimeDelta, TimeDelta>> frame_length_range_
       RTC_GUARDED_BY(worker_queue_);
-
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc

diff --git a/audio/audio_state.h b/audio/audio_state.h
@@ -18,7 +18,6 @@
 #include "audio/audio_transport_impl.h"
 #include "audio/null_audio_poller.h"
 #include "call/audio_state.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/ref_count.h"
 #include "rtc_base/thread_checker.h"
 
@@ -32,6 +31,11 @@ namespace internal {
 class AudioState : public webrtc::AudioState {
  public:
   explicit AudioState(const AudioState::Config& config);
+
+  AudioState() = delete;
+  AudioState(const AudioState&) = delete;
+  AudioState& operator=(const AudioState&) = delete;
+
   ~AudioState() override;
 
   AudioProcessing* audio_processing() override;
@@ -82,8 +86,6 @@ class AudioState : public webrtc::AudioState {
     size_t num_channels = 0;
   };
   std::map<webrtc::AudioSendStream*, StreamProperties> sending_streams_;
-
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState);
 };
 }  // namespace internal
 }  // namespace webrtc

diff --git a/audio/audio_transport_impl.h b/audio/audio_transport_impl.h
@@ -19,7 +19,6 @@
 #include "modules/audio_device/include/audio_device.h"
 #include "modules/audio_processing/include/audio_processing.h"
 #include "modules/audio_processing/typing_detection.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
 
@@ -30,6 +29,11 @@ class AudioSender;
 class AudioTransportImpl : public AudioTransport {
  public:
   AudioTransportImpl(AudioMixer* mixer, AudioProcessing* audio_processing);
+
+  AudioTransportImpl() = delete;
+  AudioTransportImpl(const AudioTransportImpl&) = delete;
+  AudioTransportImpl& operator=(const AudioTransportImpl&) = delete;
+
   ~AudioTransportImpl() override;
 
   int32_t RecordedDataIsAvailable(const void* audioSamples,
@@ -85,8 +89,6 @@ class AudioTransportImpl : public AudioTransport {
   AudioFrame mixed_frame_;
   // Converts mixed audio to the audio device output rate.
   PushResampler<int16_t> render_resampler_;
-
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTransportImpl);
 };
 }  // namespace webrtc
 

diff --git a/common_audio/smoothing_filter.h b/common_audio/smoothing_filter.h
@@ -14,7 +14,6 @@
 #include <stdint.h>
 
 #include "absl/types/optional.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -42,6 +41,11 @@ class SmoothingFilterImpl final : public SmoothingFilter {
   // will be set to |init_time_ms| first and can be changed through
   // |SetTimeConstantMs|.
   explicit SmoothingFilterImpl(int init_time_ms);
+
+  SmoothingFilterImpl() = delete;
+  SmoothingFilterImpl(const SmoothingFilterImpl&) = delete;
+  SmoothingFilterImpl& operator=(const SmoothingFilterImpl&) = delete;
+
   ~SmoothingFilterImpl() override;
 
   void AddSample(float sample) override;
@@ -64,8 +68,6 @@ class SmoothingFilterImpl final : public SmoothingFilter {
   float alpha_;
   float state_;
   int64_t last_state_time_ms_;
-
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SmoothingFilterImpl);
 };
 
 }  // namespace webrtc

diff --git a/common_audio/window_generator.h b/common_audio/window_generator.h
@@ -13,18 +13,17 @@
 
 #include <stddef.h>
 
-#include "rtc_base/constructor_magic.h"
-
 namespace webrtc {
 
 // Helper class with generators for various signal transform windows.
 class WindowGenerator {
  public:
+  WindowGenerator() = delete;
+  WindowGenerator(const WindowGenerator&) = delete;
+  WindowGenerator& operator=(const WindowGenerator&) = delete;
+
   static void Hanning(int length, float* window);
   static void KaiserBesselDerived(float alpha, size_t length, float* window);
-
- private:
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WindowGenerator);
 };
 
 }  // namespace webrtc

diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
@@ -33,7 +33,6 @@
 #include "modules/audio_processing/include/audio_processing.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/byte_order.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/experiments/field_trial_parser.h"
 #include "rtc_base/experiments/field_trial_units.h"
 #include "rtc_base/experiments/struct_parameters_parser.h"
@@ -802,6 +801,10 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     stream_ = call_->CreateAudioSendStream(config_);
   }
 
+  WebRtcAudioSendStream() = delete;
+  WebRtcAudioSendStream(const WebRtcAudioSendStream&) = delete;
+  WebRtcAudioSendStream& operator=(const WebRtcAudioSendStream&) = delete;
+
   ~WebRtcAudioSendStream() override {
     RTC_DCHECK(worker_thread_checker_.IsCurrent());
     ClearSource();
@@ -1143,8 +1146,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
   // TODO(webrtc:11717): Remove this once audio_network_adaptor in AudioOptions
   // has been removed.
   absl::optional<std::string> audio_network_adaptor_config_from_options_;
-
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
 };
 
 class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
@@ -1193,6 +1194,10 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
     RecreateAudioReceiveStream();
   }
 
+  WebRtcAudioReceiveStream() = delete;
+  WebRtcAudioReceiveStream(const WebRtcAudioReceiveStream&) = delete;
+  WebRtcAudioReceiveStream& operator=(const WebRtcAudioReceiveStream&) = delete;
+
   ~WebRtcAudioReceiveStream() {
     RTC_DCHECK(worker_thread_checker_.IsCurrent());
     call_->DestroyAudioReceiveStream(stream_);
@@ -1356,8 +1361,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
   bool playout_ = false;
   float output_volume_ = 1.0;
   std::unique_ptr<webrtc::AudioSinkInterface> raw_audio_sink_;
-
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
 };
 
 WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(

diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
@@ -26,7 +26,6 @@
 #include "media/base/media_engine.h"
 #include "media/base/rtp_utils.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/network_route.h"
 #include "rtc_base/task_queue.h"
 #include "rtc_base/thread_checker.h"
@@ -52,6 +51,11 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
       rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
       rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing,
       const webrtc::WebRtcKeyValueConfig& trials);
+
+  WebRtcVoiceEngine() = delete;
+  WebRtcVoiceEngine(const WebRtcVoiceEngine&) = delete;
+  WebRtcVoiceEngine& operator=(const WebRtcVoiceEngine&) = delete;
+
   ~WebRtcVoiceEngine() override;
 
   // Does initialization that needs to occur on the worker thread.
@@ -133,8 +137,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
   // redundancy for opus audio.
   const bool audio_red_for_opus_trial_enabled_;
   const bool minimized_remsampling_on_mobile_trial_enabled_;
-
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceEngine);
 };
 
 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
@@ -147,6 +149,11 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
                           const AudioOptions& options,
                           const webrtc::CryptoOptions& crypto_options,
                           webrtc::Call* call);
+
+  WebRtcVoiceMediaChannel() = delete;
+  WebRtcVoiceMediaChannel(const WebRtcVoiceMediaChannel&) = delete;
+  WebRtcVoiceMediaChannel& operator=(const WebRtcVoiceMediaChannel&) = delete;
+
   ~WebRtcVoiceMediaChannel() override;
 
   const AudioOptions& options() const { return options_; }
@@ -339,8 +346,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
       unsignaled_frame_decryptor_;
 
   const bool audio_red_for_opus_trial_enabled_;
-
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 

diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
@@ -793,7 +793,6 @@ rtc_library("webrtc_multiopus") {
     "../../api/units:time_delta",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
-    "../../rtc_base:macromagic",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base:safe_minmax",
     "../../rtc_base:stringutils",

diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
@@ -25,10 +25,7 @@ rtc_library("config") {
     "include/config.cc",
     "include/config.h",
   ]
-  deps = [
-    "../../rtc_base:macromagic",
-    "../../rtc_base/system:rtc_export",
-  ]
+  deps = [ "../../rtc_base/system:rtc_export" ]
 }
 
 rtc_library("api") {
@@ -47,7 +44,6 @@ rtc_library("api") {
     "../../api/audio:audio_frame_api",
     "../../api/audio:echo_control",
     "../../rtc_base:deprecation",
-    "../../rtc_base:macromagic",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/system:arch",
     "../../rtc_base/system:file_wrapper",

diff --git a/modules/audio_processing/aec3/echo_canceller3.cc b/modules/audio_processing/aec3/echo_canceller3.cc
@@ -564,6 +564,11 @@ class EchoCanceller3::RenderWriter {
                          Aec3RenderQueueItemVerifier>* render_transfer_queue,
                size_t num_bands,
                size_t num_channels);
+
+  RenderWriter() = delete;
+  RenderWriter(const RenderWriter&) = delete;
+  RenderWriter& operator=(const RenderWriter&) = delete;
+
   ~RenderWriter();
   void Insert(const AudioBuffer& input);
 
@@ -575,7 +580,6 @@ class EchoCanceller3::RenderWriter {
   std::vector<std::vector<std::vector<float>>> render_queue_input_frame_;
   SwapQueue<std::vector<std::vector<std::vector<float>>>,
             Aec3RenderQueueItemVerifier>* render_transfer_queue_;
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderWriter);
 };
 
 EchoCanceller3::RenderWriter::RenderWriter(

diff --git a/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
@@ -108,6 +108,13 @@ bool VerifyOutputFrameBitexactness(rtc::ArrayView<const float> reference,
 class CaptureTransportVerificationProcessor : public BlockProcessor {
  public:
   explicit CaptureTransportVerificationProcessor(size_t num_bands) {}
+
+  CaptureTransportVerificationProcessor() = delete;
+  CaptureTransportVerificationProcessor(
+      const CaptureTransportVerificationProcessor&) = delete;
+  CaptureTransportVerificationProcessor& operator=(
+      const CaptureTransportVerificationProcessor&) = delete;
+
   ~CaptureTransportVerificationProcessor() override = default;
 
   void ProcessCapture(
@@ -124,16 +131,20 @@ class CaptureTransportVerificationProcessor : public BlockProcessor {
   void GetMetrics(EchoControl::Metrics* metrics) const override {}
 
   void SetAudioBufferDelay(int delay_ms) override {}
-
- private:
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(CaptureTransportVerificationProcessor);
 };
 
 // Class for testing that the render data is properly received by the block
 // processor.
 class RenderTransportVerificationProcessor : public BlockProcessor {
  public:
   explicit RenderTransportVerificationProcessor(size_t num_bands) {}
+
+  RenderTransportVerificationProcessor() = delete;
+  RenderTransportVerificationProcessor(
+      const RenderTransportVerificationProcessor&) = delete;
+  RenderTransportVerificationProcessor& operator=(
+      const RenderTransportVerificationProcessor&) = delete;
+
   ~RenderTransportVerificationProcessor() override = default;
 
   void ProcessCapture(
@@ -161,7 +172,6 @@ class RenderTransportVerificationProcessor : public BlockProcessor {
  private:
   std::deque<std::vector<std::vector<std::vector<float>>>>
       received_render_blocks_;
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderTransportVerificationProcessor);
 };
 
 class EchoCanceller3Tester {
@@ -184,6 +194,10 @@ class EchoCanceller3Tester {
                        fullband_frame_length_ * 100,
                        1) {}
 
+  EchoCanceller3Tester() = delete;
+  EchoCanceller3Tester(const EchoCanceller3Tester&) = delete;
+  EchoCanceller3Tester& operator=(const EchoCanceller3Tester&) = delete;
+
   // Verifies that the capture data is properly received by the block processor
   // and that the processor data is properly passed to the EchoCanceller3
   // output.
@@ -602,8 +616,6 @@ class EchoCanceller3Tester {
   const int fullband_frame_length_;
   AudioBuffer capture_buffer_;
   AudioBuffer render_buffer_;
-
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EchoCanceller3Tester);
 };
 
 std::string ProduceDebugText(int sample_rate_hz) {