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 <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32224}
diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn
index d2da445..6a7cc57 100644
--- a/api/transport/BUILD.gn
+++ b/api/transport/BUILD.gn
@@ -144,7 +144,6 @@
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
index eeb876c..32f8b60 100644
--- 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 @@
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 @@
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
index 7bc3183..12fcb9f 100644
--- 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 @@
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 @@
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
index 70c7208..5e76642 100644
--- 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 @@
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 @@
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
index 1643a29..5b885bd 100644
--- 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 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 @@
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
index e5f561e..e96d52a 100644
--- 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 @@
// 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 @@
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
index 0cbe24a..c0a89c4 100644
--- 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
index 2c9e15d..52a8f3d 100644
--- 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 @@
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 @@
// 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 @@
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 @@
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
index f805d6c..0d3c5fe 100644
--- 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 @@
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 @@
// 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 @@
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 @@
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
index 3354033..cdf7821 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -793,7 +793,6 @@
"../../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
index 83fdcef..dbb1882 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -25,10 +25,7 @@
"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 @@
"../../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
index a0e1fc2..d96abc9 100644
--- a/modules/audio_processing/aec3/echo_canceller3.cc
+++ b/modules/audio_processing/aec3/echo_canceller3.cc
@@ -564,6 +564,11 @@
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 @@
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
index 04d93e4..a02cfa3 100644
--- a/modules/audio_processing/aec3/echo_canceller3_unittest.cc
+++ b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
@@ -108,6 +108,13 @@
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(
@@ -124,9 +131,6 @@
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
@@ -134,6 +138,13 @@
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 @@
private:
std::deque<std::vector<std::vector<std::vector<float>>>>
received_render_blocks_;
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderTransportVerificationProcessor);
};
class EchoCanceller3Tester {
@@ -184,6 +194,10 @@
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 @@
const int fullband_frame_length_;
AudioBuffer capture_buffer_;
AudioBuffer render_buffer_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EchoCanceller3Tester);
};
std::string ProduceDebugText(int sample_rate_hz) {
diff --git a/modules/audio_processing/aec3/matched_filter.h b/modules/audio_processing/aec3/matched_filter.h
index 954e478..fa44eb2 100644
--- a/modules/audio_processing/aec3/matched_filter.h
+++ b/modules/audio_processing/aec3/matched_filter.h
@@ -17,7 +17,6 @@
#include "api/array_view.h"
#include "modules/audio_processing/aec3/aec3_common.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/system/arch.h"
namespace webrtc {
@@ -104,6 +103,10 @@
float smoothing,
float matching_filter_threshold);
+ MatchedFilter() = delete;
+ MatchedFilter(const MatchedFilter&) = delete;
+ MatchedFilter& operator=(const MatchedFilter&) = delete;
+
~MatchedFilter();
// Updates the correlation with the values in the capture buffer.
@@ -139,8 +142,6 @@
const float excitation_limit_;
const float smoothing_;
const float matching_filter_threshold_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MatchedFilter);
};
} // namespace webrtc
diff --git a/modules/audio_processing/aec3/matched_filter_lag_aggregator.h b/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
index 0cc7789..d48011e 100644
--- a/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
+++ b/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
@@ -17,7 +17,6 @@
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/delay_estimate.h"
#include "modules/audio_processing/aec3/matched_filter.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
@@ -31,6 +30,12 @@
ApmDataDumper* data_dumper,
size_t max_filter_lag,
const EchoCanceller3Config::Delay::DelaySelectionThresholds& thresholds);
+
+ MatchedFilterLagAggregator() = delete;
+ MatchedFilterLagAggregator(const MatchedFilterLagAggregator&) = delete;
+ MatchedFilterLagAggregator& operator=(const MatchedFilterLagAggregator&) =
+ delete;
+
~MatchedFilterLagAggregator();
// Resets the aggregator.
@@ -47,8 +52,6 @@
int histogram_data_index_ = 0;
bool significant_candidate_found_ = false;
const EchoCanceller3Config::Delay::DelaySelectionThresholds thresholds_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MatchedFilterLagAggregator);
};
} // namespace webrtc
diff --git a/modules/audio_processing/aec3/render_buffer.h b/modules/audio_processing/aec3/render_buffer.h
index 3422df3..b8be6f5 100644
--- a/modules/audio_processing/aec3/render_buffer.h
+++ b/modules/audio_processing/aec3/render_buffer.h
@@ -23,7 +23,6 @@
#include "modules/audio_processing/aec3/fft_data.h"
#include "modules/audio_processing/aec3/spectrum_buffer.h"
#include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
@@ -33,6 +32,11 @@
RenderBuffer(BlockBuffer* block_buffer,
SpectrumBuffer* spectrum_buffer,
FftBuffer* fft_buffer);
+
+ RenderBuffer() = delete;
+ RenderBuffer(const RenderBuffer&) = delete;
+ RenderBuffer& operator=(const RenderBuffer&) = delete;
+
~RenderBuffer();
// Get a block.
@@ -105,7 +109,6 @@
const SpectrumBuffer* const spectrum_buffer_;
const FftBuffer* const fft_buffer_;
bool render_activity_ = false;
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderBuffer);
};
} // namespace webrtc
diff --git a/modules/audio_processing/aec3/render_delay_controller.cc b/modules/audio_processing/aec3/render_delay_controller.cc
index c42d22b..3677085 100644
--- a/modules/audio_processing/aec3/render_delay_controller.cc
+++ b/modules/audio_processing/aec3/render_delay_controller.cc
@@ -25,7 +25,6 @@
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/atomic_ops.h"
#include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
@@ -36,6 +35,12 @@
RenderDelayControllerImpl(const EchoCanceller3Config& config,
int sample_rate_hz,
size_t num_capture_channels);
+
+ RenderDelayControllerImpl() = delete;
+ RenderDelayControllerImpl(const RenderDelayControllerImpl&) = delete;
+ RenderDelayControllerImpl& operator=(const RenderDelayControllerImpl&) =
+ delete;
+
~RenderDelayControllerImpl() override;
void Reset(bool reset_delay_confidence) override;
void LogRenderCall() override;
@@ -57,7 +62,6 @@
size_t capture_call_counter_ = 0;
int delay_change_counter_ = 0;
DelayEstimate::Quality last_delay_estimate_quality_;
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderDelayControllerImpl);
};
DelayEstimate ComputeBufferDelay(
diff --git a/modules/audio_processing/agc/BUILD.gn b/modules/audio_processing/agc/BUILD.gn
index e5b3980..8235456 100644
--- a/modules/audio_processing/agc/BUILD.gn
+++ b/modules/audio_processing/agc/BUILD.gn
@@ -29,7 +29,6 @@
"../../../rtc_base:checks",
"../../../rtc_base:gtest_prod",
"../../../rtc_base:logging",
- "../../../rtc_base:macromagic",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:safe_minmax",
"../../../system_wrappers:field_trial",
@@ -51,7 +50,6 @@
]
deps = [
"../../../rtc_base:checks",
- "../../../rtc_base:macromagic",
"../vad",
]
}
diff --git a/modules/audio_processing/agc2/down_sampler.h b/modules/audio_processing/agc2/down_sampler.h
index 61d1b00..be7cbb3 100644
--- a/modules/audio_processing/agc2/down_sampler.h
+++ b/modules/audio_processing/agc2/down_sampler.h
@@ -13,7 +13,6 @@
#include "api/array_view.h"
#include "modules/audio_processing/agc2/biquad_filter.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
@@ -22,6 +21,11 @@
class DownSampler {
public:
explicit DownSampler(ApmDataDumper* data_dumper);
+
+ DownSampler() = delete;
+ DownSampler(const DownSampler&) = delete;
+ DownSampler& operator=(const DownSampler&) = delete;
+
void Initialize(int sample_rate_hz);
void DownSample(rtc::ArrayView<const float> in, rtc::ArrayView<float> out);
@@ -31,8 +35,6 @@
int sample_rate_hz_;
int down_sampling_factor_;
BiQuadFilter low_pass_filter_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DownSampler);
};
} // namespace webrtc
diff --git a/modules/audio_processing/agc2/noise_spectrum_estimator.h b/modules/audio_processing/agc2/noise_spectrum_estimator.h
index b22f9bb..e9895f0 100644
--- a/modules/audio_processing/agc2/noise_spectrum_estimator.h
+++ b/modules/audio_processing/agc2/noise_spectrum_estimator.h
@@ -12,7 +12,6 @@
#define MODULES_AUDIO_PROCESSING_AGC2_NOISE_SPECTRUM_ESTIMATOR_H_
#include "api/array_view.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
@@ -21,6 +20,11 @@
class NoiseSpectrumEstimator {
public:
explicit NoiseSpectrumEstimator(ApmDataDumper* data_dumper);
+
+ NoiseSpectrumEstimator() = delete;
+ NoiseSpectrumEstimator(const NoiseSpectrumEstimator&) = delete;
+ NoiseSpectrumEstimator& operator=(const NoiseSpectrumEstimator&) = delete;
+
void Initialize();
void Update(rtc::ArrayView<const float> spectrum, bool first_update);
@@ -31,8 +35,6 @@
private:
ApmDataDumper* data_dumper_;
float noise_spectrum_[65];
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(NoiseSpectrumEstimator);
};
} // namespace webrtc
diff --git a/modules/audio_processing/agc2/signal_classifier.h b/modules/audio_processing/agc2/signal_classifier.h
index ae288ae..20cce92 100644
--- a/modules/audio_processing/agc2/signal_classifier.h
+++ b/modules/audio_processing/agc2/signal_classifier.h
@@ -18,7 +18,6 @@
#include "common_audio/third_party/ooura/fft_size_128/ooura_fft.h"
#include "modules/audio_processing/agc2/down_sampler.h"
#include "modules/audio_processing/agc2/noise_spectrum_estimator.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
@@ -30,6 +29,11 @@
enum class SignalType { kNonStationary, kStationary };
explicit SignalClassifier(ApmDataDumper* data_dumper);
+
+ SignalClassifier() = delete;
+ SignalClassifier(const SignalClassifier&) = delete;
+ SignalClassifier& operator=(const SignalClassifier&) = delete;
+
~SignalClassifier();
void Initialize(int sample_rate_hz);
@@ -39,6 +43,11 @@
class FrameExtender {
public:
FrameExtender(size_t frame_size, size_t extended_frame_size);
+
+ FrameExtender() = delete;
+ FrameExtender(const FrameExtender&) = delete;
+ FrameExtender& operator=(const FrameExtender&) = delete;
+
~FrameExtender();
void ExtendFrame(rtc::ArrayView<const float> x,
@@ -46,8 +55,6 @@
private:
std::vector<float> x_old_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FrameExtender);
};
ApmDataDumper* const data_dumper_;
@@ -59,7 +66,6 @@
int consistent_classification_counter_;
SignalType last_signal_type_;
const OouraFft ooura_fft_;
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SignalClassifier);
};
} // namespace webrtc
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 32401af..43c40ff 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -31,6 +31,7 @@
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/audio_processing/include/config.h"
#include "rtc_base/arraysize.h"
+#include "rtc_base/constructor_magic.h"
#include "rtc_base/deprecation.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/system/file_wrapper.h"
diff --git a/modules/audio_processing/include/config.h b/modules/audio_processing/include/config.h
index 8a24586..7fab178 100644
--- a/modules/audio_processing/include/config.h
+++ b/modules/audio_processing/include/config.h
@@ -13,7 +13,6 @@
#include <map>
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@@ -105,7 +104,6 @@
typedef std::map<ConfigOptionID, BaseOption*> OptionMap;
OptionMap options_;
- // RTC_DISALLOW_COPY_AND_ASSIGN
Config(const Config&);
void operator=(const Config&);
};
diff --git a/modules/audio_processing/logging/apm_data_dumper.h b/modules/audio_processing/logging/apm_data_dumper.h
index 17a5c87..1824fdd 100644
--- a/modules/audio_processing/logging/apm_data_dumper.h
+++ b/modules/audio_processing/logging/apm_data_dumper.h
@@ -26,7 +26,6 @@
#include "common_audio/wav_file.h"
#include "rtc_base/checks.h"
#endif
-#include "rtc_base/constructor_magic.h"
// Check to verify that the define is properly set.
#if !defined(WEBRTC_APM_DEBUG_DUMP) || \
@@ -52,6 +51,10 @@
// instances of the code.
explicit ApmDataDumper(int instance_index);
+ ApmDataDumper() = delete;
+ ApmDataDumper(const ApmDataDumper&) = delete;
+ ApmDataDumper& operator=(const ApmDataDumper&) = delete;
+
~ApmDataDumper();
// Activates or deactivate the dumping functionality.
@@ -277,7 +280,6 @@
int num_channels,
WavFile::SampleFormat format);
#endif
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ApmDataDumper);
};
} // namespace webrtc
diff --git a/modules/audio_processing/test/aec_dump_based_simulator.h b/modules/audio_processing/test/aec_dump_based_simulator.h
index 092b82b..c8d82e6 100644
--- a/modules/audio_processing/test/aec_dump_based_simulator.h
+++ b/modules/audio_processing/test/aec_dump_based_simulator.h
@@ -15,7 +15,6 @@
#include <string>
#include "modules/audio_processing/test/audio_processing_simulator.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/ignore_wundef.h"
RTC_PUSH_IGNORING_WUNDEF()
@@ -35,6 +34,11 @@
AecDumpBasedSimulator(const SimulationSettings& settings,
rtc::scoped_refptr<AudioProcessing> audio_processing,
std::unique_ptr<AudioProcessingBuilder> ap_builder);
+
+ AecDumpBasedSimulator() = delete;
+ AecDumpBasedSimulator(const AecDumpBasedSimulator&) = delete;
+ AecDumpBasedSimulator& operator=(const AecDumpBasedSimulator&) = delete;
+
~AecDumpBasedSimulator() override;
// Processes the messages in the aecdump file.
@@ -65,7 +69,6 @@
bool artificial_nearend_eof_reported_ = false;
InterfaceType interface_used_ = InterfaceType::kNotSpecified;
std::unique_ptr<std::ofstream> call_order_output_file_;
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AecDumpBasedSimulator);
};
} // namespace test
diff --git a/modules/audio_processing/test/audio_processing_simulator.h b/modules/audio_processing/test/audio_processing_simulator.h
index 8579f4b..63e644a 100644
--- a/modules/audio_processing/test/audio_processing_simulator.h
+++ b/modules/audio_processing/test/audio_processing_simulator.h
@@ -24,7 +24,6 @@
#include "modules/audio_processing/test/api_call_statistics.h"
#include "modules/audio_processing/test/fake_recording_device.h"
#include "modules/audio_processing/test/test_utils.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/time_utils.h"
@@ -153,6 +152,11 @@
AudioProcessingSimulator(const SimulationSettings& settings,
rtc::scoped_refptr<AudioProcessing> audio_processing,
std::unique_ptr<AudioProcessingBuilder> ap_builder);
+
+ AudioProcessingSimulator() = delete;
+ AudioProcessingSimulator(const AudioProcessingSimulator&) = delete;
+ AudioProcessingSimulator& operator=(const AudioProcessingSimulator&) = delete;
+
virtual ~AudioProcessingSimulator();
// Processes the data in the input.
@@ -222,8 +226,6 @@
FakeRecordingDevice fake_recording_device_;
TaskQueueForTest worker_queue_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
};
} // namespace test
diff --git a/modules/audio_processing/test/wav_based_simulator.h b/modules/audio_processing/test/wav_based_simulator.h
index 3adbe70..286ce1f 100644
--- a/modules/audio_processing/test/wav_based_simulator.h
+++ b/modules/audio_processing/test/wav_based_simulator.h
@@ -14,7 +14,6 @@
#include <vector>
#include "modules/audio_processing/test/audio_processing_simulator.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
namespace test {
@@ -25,6 +24,11 @@
WavBasedSimulator(const SimulationSettings& settings,
rtc::scoped_refptr<AudioProcessing> audio_processing,
std::unique_ptr<AudioProcessingBuilder> ap_builder);
+
+ WavBasedSimulator() = delete;
+ WavBasedSimulator(const WavBasedSimulator&) = delete;
+ WavBasedSimulator& operator=(const WavBasedSimulator&) = delete;
+
~WavBasedSimulator() override;
// Processes the WAV input.
@@ -46,8 +50,6 @@
const std::string& filename);
std::vector<SimulationEventType> call_chain_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WavBasedSimulator);
};
} // namespace test
diff --git a/modules/congestion_controller/goog_cc/BUILD.gn b/modules/congestion_controller/goog_cc/BUILD.gn
index 9bee7b2..00e00f4 100644
--- a/modules/congestion_controller/goog_cc/BUILD.gn
+++ b/modules/congestion_controller/goog_cc/BUILD.gn
@@ -44,7 +44,6 @@
"../../../logging:rtc_event_pacing",
"../../../rtc_base:checks",
"../../../rtc_base:logging",
- "../../../rtc_base:macromagic",
"../../../rtc_base/experiments:alr_experiment",
"../../../rtc_base/experiments:field_trial_parser",
"../../../rtc_base/experiments:rate_control_settings",
diff --git a/modules/congestion_controller/goog_cc/delay_based_bwe.h b/modules/congestion_controller/goog_cc/delay_based_bwe.h
index 25f5a3b..74650dc 100644
--- a/modules/congestion_controller/goog_cc/delay_based_bwe.h
+++ b/modules/congestion_controller/goog_cc/delay_based_bwe.h
@@ -26,7 +26,6 @@
#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/remote_bitrate_estimator/inter_arrival.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/experiments/struct_parameters_parser.h"
#include "rtc_base/race_checker.h"
@@ -78,6 +77,11 @@
explicit DelayBasedBwe(const WebRtcKeyValueConfig* key_value_config,
RtcEventLog* event_log,
NetworkStatePredictor* network_state_predictor);
+
+ DelayBasedBwe() = delete;
+ DelayBasedBwe(const DelayBasedBwe&) = delete;
+ DelayBasedBwe& operator=(const DelayBasedBwe&) = delete;
+
virtual ~DelayBasedBwe();
Result IncomingPacketFeedbackVector(
@@ -143,7 +147,6 @@
bool has_once_detected_overuse_;
BandwidthUsage prev_state_;
bool alr_limited_backoff_enabled_;
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DelayBasedBwe);
};
} // namespace webrtc
diff --git a/modules/congestion_controller/goog_cc/goog_cc_network_control.h b/modules/congestion_controller/goog_cc/goog_cc_network_control.h
index 1e4dcf6..6dd70c8 100644
--- a/modules/congestion_controller/goog_cc/goog_cc_network_control.h
+++ b/modules/congestion_controller/goog_cc/goog_cc_network_control.h
@@ -33,7 +33,6 @@
#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
#include "modules/congestion_controller/goog_cc/probe_controller.h"
#include "modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/experiments/rate_control_settings.h"
@@ -48,6 +47,11 @@
public:
GoogCcNetworkController(NetworkControllerConfig config,
GoogCcConfig goog_cc_config);
+
+ GoogCcNetworkController() = delete;
+ GoogCcNetworkController(const GoogCcNetworkController&) = delete;
+ GoogCcNetworkController& operator=(const GoogCcNetworkController&) = delete;
+
~GoogCcNetworkController() override;
// NetworkControllerInterface
@@ -137,8 +141,6 @@
bool previously_in_alr_ = false;
absl::optional<DataSize> current_data_window_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GoogCcNetworkController);
};
} // namespace webrtc
diff --git a/modules/congestion_controller/include/receive_side_congestion_controller.h b/modules/congestion_controller/include/receive_side_congestion_controller.h
index 6cd8be3..034f2e9 100644
--- a/modules/congestion_controller/include/receive_side_congestion_controller.h
+++ b/modules/congestion_controller/include/receive_side_congestion_controller.h
@@ -18,7 +18,6 @@
#include "api/transport/network_control.h"
#include "modules/include/module.h"
#include "modules/remote_bitrate_estimator/remote_estimator_proxy.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/synchronization/mutex.h"
namespace webrtc {
@@ -66,6 +65,11 @@
public:
WrappingBitrateEstimator(RemoteBitrateObserver* observer, Clock* clock);
+ WrappingBitrateEstimator() = delete;
+ WrappingBitrateEstimator(const WrappingBitrateEstimator&) = delete;
+ WrappingBitrateEstimator& operator=(const WrappingBitrateEstimator&) =
+ delete;
+
~WrappingBitrateEstimator() override;
void IncomingPacket(int64_t arrival_time_ms,
@@ -96,8 +100,6 @@
bool using_absolute_send_time_;
uint32_t packets_since_absolute_send_time_;
int min_bitrate_bps_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WrappingBitrateEstimator);
};
const FieldTrialBasedConfig field_trial_config_;
diff --git a/modules/desktop_capture/win/scoped_gdi_object.h b/modules/desktop_capture/win/scoped_gdi_object.h
index 56abe95..d3ac9b9 100644
--- a/modules/desktop_capture/win/scoped_gdi_object.h
+++ b/modules/desktop_capture/win/scoped_gdi_object.h
@@ -58,27 +58,29 @@
template <typename T>
class DeleteObjectTraits {
public:
+ DeleteObjectTraits() = delete;
+ DeleteObjectTraits(const DeleteObjectTraits&) = delete;
+ DeleteObjectTraits& operator=(const DeleteObjectTraits&) = delete;
+
// Closes the handle.
static void Close(T handle) {
if (handle)
DeleteObject(handle);
}
-
- private:
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DeleteObjectTraits);
};
// The traits class that uses DestroyCursor() to close a handle.
class DestroyCursorTraits {
public:
+ DestroyCursorTraits() = delete;
+ DestroyCursorTraits(const DestroyCursorTraits&) = delete;
+ DestroyCursorTraits& operator=(const DestroyCursorTraits&) = delete;
+
// Closes the handle.
static void Close(HCURSOR handle) {
if (handle)
DestroyCursor(handle);
}
-
- private:
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DestroyCursorTraits);
};
typedef ScopedGDIObject<HBITMAP, DeleteObjectTraits<HBITMAP> > ScopedBitmap;
diff --git a/modules/remote_bitrate_estimator/inter_arrival.h b/modules/remote_bitrate_estimator/inter_arrival.h
index 1d84970..dbc630f 100644
--- a/modules/remote_bitrate_estimator/inter_arrival.h
+++ b/modules/remote_bitrate_estimator/inter_arrival.h
@@ -14,8 +14,6 @@
#include <stddef.h>
#include <stdint.h>
-#include "rtc_base/constructor_magic.h"
-
namespace webrtc {
// Helper class to compute the inter-arrival time delta and the size delta
@@ -35,6 +33,10 @@
double timestamp_to_ms_coeff,
bool enable_burst_grouping);
+ InterArrival() = delete;
+ InterArrival(const InterArrival&) = delete;
+ InterArrival& operator=(const InterArrival&) = delete;
+
// This function returns true if a delta was computed, or false if the current
// group is still incomplete or if only one group has been completed.
// |timestamp| is the timestamp.
@@ -87,8 +89,6 @@
double timestamp_to_ms_coeff_;
bool burst_grouping_;
int num_consecutive_reordered_packets_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(InterArrival);
};
} // namespace webrtc
diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h b/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
index 2423363..f42a28f 100644
--- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
+++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
@@ -27,7 +27,6 @@
#include "modules/remote_bitrate_estimator/overuse_detector.h"
#include "modules/remote_bitrate_estimator/overuse_estimator.h"
#include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/rate_statistics.h"
#include "rtc_base/synchronization/mutex.h"
@@ -76,6 +75,13 @@
public:
RemoteBitrateEstimatorAbsSendTime(RemoteBitrateObserver* observer,
Clock* clock);
+
+ RemoteBitrateEstimatorAbsSendTime() = delete;
+ RemoteBitrateEstimatorAbsSendTime(const RemoteBitrateEstimatorAbsSendTime&) =
+ delete;
+ RemoteBitrateEstimatorAbsSendTime& operator=(
+ const RemoteBitrateEstimatorAbsSendTime&) = delete;
+
~RemoteBitrateEstimatorAbsSendTime() override;
void IncomingPacket(int64_t arrival_time_ms,
@@ -141,8 +147,6 @@
mutable Mutex mutex_;
Ssrcs ssrcs_ RTC_GUARDED_BY(&mutex_);
AimdRateControl remote_rate_ RTC_GUARDED_BY(&mutex_);
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RemoteBitrateEstimatorAbsSendTime);
};
} // namespace webrtc
diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
index 6da67e5..9fd2f9f 100644
--- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
+++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
@@ -21,7 +21,6 @@
#include "api/transport/field_trial_based_config.h"
#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/rate_statistics.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
@@ -35,6 +34,13 @@
public:
RemoteBitrateEstimatorSingleStream(RemoteBitrateObserver* observer,
Clock* clock);
+
+ RemoteBitrateEstimatorSingleStream() = delete;
+ RemoteBitrateEstimatorSingleStream(
+ const RemoteBitrateEstimatorSingleStream&) = delete;
+ RemoteBitrateEstimatorSingleStream& operator=(
+ const RemoteBitrateEstimatorSingleStream&) = delete;
+
~RemoteBitrateEstimatorSingleStream() override;
void IncomingPacket(int64_t arrival_time_ms,
@@ -74,8 +80,6 @@
int64_t last_process_time_;
int64_t process_interval_ms_ RTC_GUARDED_BY(mutex_);
bool uma_recorded_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RemoteBitrateEstimatorSingleStream);
};
} // namespace webrtc
diff --git a/modules/remote_bitrate_estimator/test/bwe_test_logging.h b/modules/remote_bitrate_estimator/test/bwe_test_logging.h
index 5a30da8..079a7f8 100644
--- a/modules/remote_bitrate_estimator/test/bwe_test_logging.h
+++ b/modules/remote_bitrate_estimator/test/bwe_test_logging.h
@@ -263,10 +263,11 @@
Context(uint32_t name, int64_t timestamp_ms, bool enabled);
Context(const std::string& name, int64_t timestamp_ms, bool enabled);
Context(const char* name, int64_t timestamp_ms, bool enabled);
- ~Context();
- private:
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Context);
+ Context() = delete;
+ Context(const Context&) = delete;
+ Context& operator=(const Context&) = delete;
+ ~Context();
};
static Logging* GetInstance();
diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc
index 5d2c9a2..61e6085 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -37,7 +37,6 @@
#include "modules/rtp_rtcp/source/time_util.h"
#include "modules/rtp_rtcp/source/tmmbr_help.h"
#include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/trace_event.h"
@@ -56,6 +55,10 @@
PacketContainer(Transport* transport, RtcEventLog* event_log)
: transport_(transport), event_log_(event_log) {}
+ PacketContainer() = delete;
+ PacketContainer(const PacketContainer&) = delete;
+ PacketContainer& operator=(const PacketContainer&) = delete;
+
size_t SendPackets(size_t max_payload_length) {
size_t bytes_sent = 0;
Build(max_payload_length, [&](rtc::ArrayView<const uint8_t> packet) {
@@ -72,8 +75,6 @@
private:
Transport* transport_;
RtcEventLog* const event_log_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(PacketContainer);
};
// Helper to put several RTCP packets into lower layer datagram RTCP packet.
diff --git a/modules/rtp_rtcp/source/rtcp_sender.h b/modules/rtp_rtcp/source/rtcp_sender.h
index bcdf15e..22b2bb7 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/modules/rtp_rtcp/source/rtcp_sender.h
@@ -31,7 +31,6 @@
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/random.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
@@ -65,6 +64,11 @@
};
explicit RTCPSender(const RtpRtcpInterface::Configuration& config);
+
+ RTCPSender() = delete;
+ RTCPSender(const RTCPSender&) = delete;
+ RTCPSender& operator=(const RTCPSender&) = delete;
+
virtual ~RTCPSender();
RtcpMode Status() const RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
@@ -308,8 +312,6 @@
const RtcpContext&);
// Map from RTCPPacketType to builder.
std::map<uint32_t, BuilderFunc> builders_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTCPSender);
};
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_packet_history.h b/modules/rtp_rtcp/source/rtp_packet_history.h
index f249116..4a2bf91 100644
--- a/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -19,7 +19,6 @@
#include "api/function_view.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
@@ -63,6 +62,11 @@
static constexpr int kPacketCullingDelayFactor = 3;
RtpPacketHistory(Clock* clock, bool enable_padding_prio);
+
+ RtpPacketHistory() = delete;
+ RtpPacketHistory(const RtpPacketHistory&) = delete;
+ RtpPacketHistory& operator=(const RtpPacketHistory&) = delete;
+
~RtpPacketHistory();
// Set/get storage mode. Note that setting the state will clear the history,
@@ -211,8 +215,6 @@
// Objects from |packet_history_| ordered by "most likely to be useful", used
// in GetPayloadPaddingPacket().
PacketPrioritySet padding_priority_ RTC_GUARDED_BY(lock_);
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpPacketHistory);
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index fd1a8da..1580259 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -29,7 +29,6 @@
#include "modules/rtp_rtcp/source/rtp_packet_history.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/deprecation.h"
#include "rtc_base/random.h"
#include "rtc_base/rate_statistics.h"
@@ -49,6 +48,10 @@
RtpPacketHistory* packet_history,
RtpPacketSender* packet_sender);
+ RTPSender() = delete;
+ RTPSender(const RTPSender&) = delete;
+ RTPSender& operator=(const RTPSender&) = delete;
+
~RTPSender();
void SetSendingMediaStatus(bool enabled) RTC_LOCKS_EXCLUDED(send_mutex_);
@@ -230,8 +233,6 @@
bool supports_bwe_extension_ RTC_GUARDED_BY(send_mutex_);
RateLimiter* const retransmission_rate_limiter_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
};
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.h b/modules/rtp_rtcp/source/rtp_sender_audio.h
index 0ac308d..57b9dd7 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -22,7 +22,6 @@
#include "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
#include "modules/rtp_rtcp/source/dtmf_queue.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/one_time_event.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
@@ -33,6 +32,11 @@
class RTPSenderAudio {
public:
RTPSenderAudio(Clock* clock, RTPSender* rtp_sender);
+
+ RTPSenderAudio() = delete;
+ RTPSenderAudio(const RTPSenderAudio&) = delete;
+ RTPSenderAudio& operator=(const RTPSenderAudio&) = delete;
+
~RTPSenderAudio();
int32_t RegisterAudioPayload(absl::string_view payload_name,
@@ -109,8 +113,6 @@
const FieldTrialBasedConfig field_trials_;
const bool include_capture_clock_offset_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSenderAudio);
};
} // namespace webrtc
diff --git a/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc b/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
index cd39e72..39c14e4 100644
--- a/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
+++ b/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
@@ -76,23 +76,26 @@
decoded_image_(decoded_image),
decode_time_ms_(decode_time_ms),
qp_(qp) {}
+
+ DecodedImageData() = delete;
+ DecodedImageData(const DecodedImageData&) = delete;
+ DecodedImageData& operator=(const DecodedImageData&) = delete;
+
const AlphaCodecStream stream_idx_;
VideoFrame decoded_image_;
const absl::optional<int32_t> decode_time_ms_;
const absl::optional<uint8_t> qp_;
-
- private:
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DecodedImageData);
};
struct MultiplexDecoderAdapter::AugmentingData {
AugmentingData(std::unique_ptr<uint8_t[]> augmenting_data, uint16_t data_size)
: data_(std::move(augmenting_data)), size_(data_size) {}
+ AugmentingData() = delete;
+ AugmentingData(const AugmentingData&) = delete;
+ AugmentingData& operator=(const AugmentingData&) = delete;
+
std::unique_ptr<uint8_t[]> data_;
const uint16_t size_;
-
- private:
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AugmentingData);
};
MultiplexDecoderAdapter::MultiplexDecoderAdapter(
diff --git a/modules/video_coding/frame_buffer2.h b/modules/video_coding/frame_buffer2.h
index c88ae89..746773d 100644
--- a/modules/video_coding/frame_buffer2.h
+++ b/modules/video_coding/frame_buffer2.h
@@ -23,7 +23,6 @@
#include "modules/video_coding/inter_frame_delay.h"
#include "modules/video_coding/jitter_estimator.h"
#include "modules/video_coding/utility/decoded_frames_history.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/event.h"
#include "rtc_base/experiments/rtt_mult_experiment.h"
#include "rtc_base/numerics/sequence_number_util.h"
@@ -50,6 +49,10 @@
VCMTiming* timing,
VCMReceiveStatisticsCallback* stats_callback);
+ FrameBuffer() = delete;
+ FrameBuffer(const FrameBuffer&) = delete;
+ FrameBuffer& operator=(const FrameBuffer&) = delete;
+
virtual ~FrameBuffer();
// Insert a frame into the frame buffer. Returns the picture id
@@ -188,8 +191,6 @@
// rtt_mult experiment settings.
const absl::optional<RttMultExperiment::Settings> rtt_mult_settings_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FrameBuffer);
};
} // namespace video_coding
diff --git a/pc/audio_track.h b/pc/audio_track.h
index f8ad471..a0388e8 100644
--- a/pc/audio_track.h
+++ b/pc/audio_track.h
@@ -16,7 +16,6 @@
#include "api/media_stream_interface.h"
#include "api/scoped_refptr.h"
#include "pc/media_stream_track.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/thread_checker.h"
namespace webrtc {
@@ -27,6 +26,11 @@
// Protected ctor to force use of factory method.
AudioTrack(const std::string& label,
const rtc::scoped_refptr<AudioSourceInterface>& source);
+
+ AudioTrack() = delete;
+ AudioTrack(const AudioTrack&) = delete;
+ AudioTrack& operator=(const AudioTrack&) = delete;
+
~AudioTrack() override;
public:
@@ -50,7 +54,6 @@
private:
const rtc::scoped_refptr<AudioSourceInterface> audio_source_;
rtc::ThreadChecker thread_checker_;
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTrack);
};
} // namespace webrtc
diff --git a/pc/ice_transport.h b/pc/ice_transport.h
index 69b69e4..c1529de 100644
--- a/pc/ice_transport.h
+++ b/pc/ice_transport.h
@@ -12,7 +12,6 @@
#define PC_ICE_TRANSPORT_H_
#include "api/ice_transport_interface.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_checker.h"
@@ -29,6 +28,10 @@
RTC_DCHECK(internal_);
}
+ IceTransportWithPointer() = delete;
+ IceTransportWithPointer(const IceTransportWithPointer&) = delete;
+ IceTransportWithPointer& operator=(const IceTransportWithPointer&) = delete;
+
cricket::IceTransportInternal* internal() override;
// This call will ensure that the pointer passed at construction is
// no longer in use by this object. Later calls to internal() will return
@@ -39,7 +42,6 @@
~IceTransportWithPointer() override;
private:
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(IceTransportWithPointer);
const rtc::Thread* creator_thread_;
cricket::IceTransportInternal* internal_ RTC_GUARDED_BY(creator_thread_);
};
diff --git a/pc/remote_audio_source.cc b/pc/remote_audio_source.cc
index 18a4ed2..8ae0612 100644
--- a/pc/remote_audio_source.cc
+++ b/pc/remote_audio_source.cc
@@ -18,7 +18,6 @@
#include "absl/algorithm/container.h"
#include "api/scoped_refptr.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/numerics/safe_conversions.h"
@@ -36,6 +35,11 @@
explicit AudioDataProxy(RemoteAudioSource* source) : source_(source) {
RTC_DCHECK(source);
}
+
+ AudioDataProxy() = delete;
+ AudioDataProxy(const AudioDataProxy&) = delete;
+ AudioDataProxy& operator=(const AudioDataProxy&) = delete;
+
~AudioDataProxy() override { source_->OnAudioChannelGone(); }
// AudioSinkInterface implementation.
@@ -45,8 +49,6 @@
private:
const rtc::scoped_refptr<RemoteAudioSource> source_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioDataProxy);
};
RemoteAudioSource::RemoteAudioSource(rtc::Thread* worker_thread)
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index e3b4581..e15206c 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -599,7 +599,6 @@
]
deps = [
":checks",
- ":macromagic",
":rtc_base_approved",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
@@ -1066,7 +1065,6 @@
deps = [
":criticalsection",
":gunit_helpers",
- ":macromagic",
":rtc_base",
":rtc_base_tests_utils",
":timeutils",
@@ -1150,7 +1148,6 @@
]
deps = [
":checks",
- ":macromagic",
":rtc_base_approved",
":rtc_event",
":rtc_task_queue",
diff --git a/rtc_base/constructor_magic.h b/rtc_base/constructor_magic.h
index e63c2ff..8d12a7b 100644
--- a/rtc_base/constructor_magic.h
+++ b/rtc_base/constructor_magic.h
@@ -11,24 +11,10 @@
#ifndef RTC_BASE_CONSTRUCTOR_MAGIC_H_
#define RTC_BASE_CONSTRUCTOR_MAGIC_H_
-// Put this in the declarations for a class to be unassignable.
-#define RTC_DISALLOW_ASSIGN(TypeName) \
- TypeName& operator=(const TypeName&) = delete
-
// A macro to disallow the copy constructor and operator= functions. This should
// be used in the declarations for a class.
#define RTC_DISALLOW_COPY_AND_ASSIGN(TypeName) \
TypeName(const TypeName&) = delete; \
- RTC_DISALLOW_ASSIGN(TypeName)
-
-// A macro to disallow all the implicit constructors, namely the default
-// constructor, copy constructor and operator= functions.
-//
-// This should be used in the declarations for a class that wants to prevent
-// anyone from instantiating it. This is especially useful for classes
-// containing only static methods.
-#define RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(TypeName) \
- TypeName() = delete; \
- RTC_DISALLOW_COPY_AND_ASSIGN(TypeName)
+ TypeName& operator=(const TypeName&) = delete
#endif // RTC_BASE_CONSTRUCTOR_MAGIC_H_
diff --git a/rtc_base/deprecated/signal_thread.h b/rtc_base/deprecated/signal_thread.h
index 65f96bc..10805ad 100644
--- a/rtc_base/deprecated/signal_thread.h
+++ b/rtc_base/deprecated/signal_thread.h
@@ -110,14 +110,17 @@
class Worker : public Thread {
public:
explicit Worker(DEPRECATED_SignalThread* parent);
+
+ Worker() = delete;
+ Worker(const Worker&) = delete;
+ Worker& operator=(const Worker&) = delete;
+
~Worker() override;
void Run() override;
bool IsProcessingMessagesForTesting() override;
private:
DEPRECATED_SignalThread* parent_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Worker);
};
class RTC_SCOPED_LOCKABLE EnterExit {
@@ -131,6 +134,11 @@
RTC_DCHECK_NE(0, t_->refcount_);
++t_->refcount_;
}
+
+ EnterExit() = delete;
+ EnterExit(const EnterExit&) = delete;
+ EnterExit& operator=(const EnterExit&) = delete;
+
~EnterExit() RTC_UNLOCK_FUNCTION() {
bool d = (0 == --t_->refcount_);
t_->cs_.Leave();
@@ -140,8 +148,6 @@
private:
DEPRECATED_SignalThread* t_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EnterExit);
};
void Run();
diff --git a/rtc_base/random.h b/rtc_base/random.h
index 93241a3..0e2d103 100644
--- a/rtc_base/random.h
+++ b/rtc_base/random.h
@@ -16,7 +16,6 @@
#include <limits>
#include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
@@ -35,6 +34,10 @@
// See also discussion here: https://codereview.webrtc.org/1623543002/
explicit Random(uint64_t seed);
+ Random() = delete;
+ Random(const Random&) = delete;
+ Random& operator=(const Random&) = delete;
+
// Return pseudo-random integer of the specified type.
// We need to limit the size to 32 bits to keep the output close to uniform.
template <typename T>
@@ -73,8 +76,6 @@
}
uint64_t state_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Random);
};
// Return pseudo-random number in the interval [0.0, 1.0).
diff --git a/rtc_base/rate_limiter.h b/rtc_base/rate_limiter.h
index 051ccf6..9bbe21f 100644
--- a/rtc_base/rate_limiter.h
+++ b/rtc_base/rate_limiter.h
@@ -14,7 +14,6 @@
#include <stddef.h>
#include <stdint.h>
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/rate_statistics.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
@@ -29,6 +28,11 @@
class RateLimiter {
public:
RateLimiter(Clock* clock, int64_t max_window_ms);
+
+ RateLimiter() = delete;
+ RateLimiter(const RateLimiter&) = delete;
+ RateLimiter& operator=(const RateLimiter&) = delete;
+
~RateLimiter();
// Try to use rate to send bytes. Returns true on success and if so updates
@@ -49,8 +53,6 @@
RateStatistics current_rate_ RTC_GUARDED_BY(lock_);
int64_t window_size_ms_ RTC_GUARDED_BY(lock_);
uint32_t max_rate_bps_ RTC_GUARDED_BY(lock_);
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RateLimiter);
};
} // namespace webrtc
diff --git a/rtc_base/weak_ptr.h b/rtc_base/weak_ptr.h
index 3e63a75..8b2ba09 100644
--- a/rtc_base/weak_ptr.h
+++ b/rtc_base/weak_ptr.h
@@ -241,6 +241,10 @@
public:
explicit WeakPtrFactory(T* ptr) : ptr_(ptr) {}
+ WeakPtrFactory() = delete;
+ WeakPtrFactory(const WeakPtrFactory&) = delete;
+ WeakPtrFactory& operator=(const WeakPtrFactory&) = delete;
+
~WeakPtrFactory() { ptr_ = nullptr; }
WeakPtr<T> GetWeakPtr() {
@@ -263,7 +267,6 @@
private:
internal::WeakReferenceOwner weak_reference_owner_;
T* ptr_;
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WeakPtrFactory);
};
} // namespace rtc
diff --git a/rtc_tools/rtp_generator/rtp_generator.h b/rtc_tools/rtp_generator/rtp_generator.h
index 6248c6a..a317bf7 100644
--- a/rtc_tools/rtp_generator/rtp_generator.h
+++ b/rtc_tools/rtp_generator/rtp_generator.h
@@ -27,7 +27,6 @@
#include "call/rtp_config.h"
#include "call/video_send_stream.h"
#include "media/engine/webrtc_video_engine.h"
-#include "rtc_base/constructor_magic.h"
#include "test/frame_generator_capturer.h"
#include "test/rtp_file_reader.h"
#include "test/rtp_file_writer.h"
@@ -79,6 +78,11 @@
public:
// Construct a new RtpGenerator using the specified options.
explicit RtpGenerator(const RtpGeneratorOptions& options);
+
+ RtpGenerator() = delete;
+ RtpGenerator(const RtpGenerator&) = delete;
+ RtpGenerator& operator=(const RtpGenerator&) = delete;
+
// Cleans up the VideoSendStream.
~RtpGenerator() override;
// Generates an rtp_dump that is written out to
@@ -113,9 +117,6 @@
std::vector<uint32_t> durations_ms_;
uint32_t start_ms_ = 0;
std::unique_ptr<TaskQueueFactory> task_queue_;
-
- // This object cannot be copied.
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpGenerator);
};
} // namespace webrtc
diff --git a/video/BUILD.gn b/video/BUILD.gn
index 96d0faa..b833022 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -241,7 +241,6 @@
"../rtc_base:checks",
"../rtc_base:criticalsection",
"../rtc_base:logging",
- "../rtc_base:macromagic",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_event",
"../rtc_base:rtc_numerics",
diff --git a/video/adaptation/BUILD.gn b/video/adaptation/BUILD.gn
index eab01cd..2a60ead 100644
--- a/video/adaptation/BUILD.gn
+++ b/video/adaptation/BUILD.gn
@@ -42,7 +42,6 @@
"../../modules/video_coding:video_coding_utility",
"../../rtc_base:checks",
"../../rtc_base:logging",
- "../../rtc_base:macromagic",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_event",
"../../rtc_base:rtc_numerics",
