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Delete macros RTC_DISALLOW_ASSIGN and RTC_DISALLOW_IMPLICIT_CONSTRUCTORS

Browse source 
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}
This commit is contained in:
Niels Möller 2020-09-29 09:46:21 +02:00 committed by Commit Bot
parent 46be80d349
commit de95329daa
55 changed files with 256 additions and 172 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

1
api/transport/BUILD.gn
View file

@ -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",

8
audio/audio_receive_stream.h
View file

@ -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

8
audio/audio_send_stream.h
View file

@ -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

8
audio/audio_state.h
View file

@ -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

8
audio/audio_transport_impl.h
View file

@ -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

8
common_audio/smoothing_filter.h
View file

@ -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

9
common_audio/window_generator.h
View file

@ -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

13
media/engine/webrtc_voice_engine.cc
View file

@ -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(

15
media/engine/webrtc_voice_engine.h
View file

@ -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

1
modules/audio_coding/BUILD.gn
View file

@ -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",

6
modules/audio_processing/BUILD.gn
View file

@ -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",

6
modules/audio_processing/aec3/echo_canceller3.cc
View file

@ -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(

24
modules/audio_processing/aec3/echo_canceller3_unittest.cc
View file

@ -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(
@ -124,9 +131,6 @@ 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
@ -134,6 +138,13 @@ class CaptureTransportVerificationProcessor : public BlockProcessor {
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) {

7
modules/audio_processing/aec3/matched_filter.h
View file

@ -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 @@ class MatchedFilter {
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 @@ class MatchedFilter {
const float excitation_limit_;
const float smoothing_;
const float matching_filter_threshold_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MatchedFilter);
};
} // namespace webrtc

9
modules/audio_processing/aec3/matched_filter_lag_aggregator.h
View file

@ -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 @@ class MatchedFilterLagAggregator {
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 @@ class MatchedFilterLagAggregator {
int histogram_data_index_ = 0;
bool significant_candidate_found_ = false;
const EchoCanceller3Config::Delay::DelaySelectionThresholds thresholds_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MatchedFilterLagAggregator);
};
} // namespace webrtc

7
modules/audio_processing/aec3/render_buffer.h
View file

@ -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 @@ class RenderBuffer {
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 @@ class RenderBuffer {
const SpectrumBuffer* const spectrum_buffer_;
const FftBuffer* const fft_buffer_;
bool render_activity_ = false;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderBuffer);
};
} // namespace webrtc

8
modules/audio_processing/aec3/render_delay_controller.cc
View file

@ -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 @@ class RenderDelayControllerImpl final : public RenderDelayController {
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 @@ class RenderDelayControllerImpl final : public RenderDelayController {
size_t capture_call_counter_ = 0;
int delay_change_counter_ = 0;
DelayEstimate::Quality last_delay_estimate_quality_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderDelayControllerImpl);
};
DelayEstimate ComputeBufferDelay(

2
modules/audio_processing/agc/BUILD.gn
View file

@ -29,7 +29,6 @@ rtc_library("agc") {
"../../../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 @@ rtc_library("level_estimation") {
]
deps = [
"../../../rtc_base:checks",
"../../../rtc_base:macromagic",
"../vad",
]
}

8
modules/audio_processing/agc2/down_sampler.h
View file

@ -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 ApmDataDumper;
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 @@ class DownSampler {
int sample_rate_hz_;
int down_sampling_factor_;
BiQuadFilter low_pass_filter_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DownSampler);
};
} // namespace webrtc

8
modules/audio_processing/agc2/noise_spectrum_estimator.h
View file

@ -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 ApmDataDumper;
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 @@ class NoiseSpectrumEstimator {
private:
ApmDataDumper* data_dumper_;
float noise_spectrum_[65];
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(NoiseSpectrumEstimator);
};
} // namespace webrtc

14
modules/audio_processing/agc2/signal_classifier.h
View file

@ -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 @@ class SignalClassifier {
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 SignalClassifier {
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 @@ class SignalClassifier {
private:
std::vector<float> x_old_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FrameExtender);
};
ApmDataDumper* const data_dumper_;
@ -59,7 +66,6 @@ class SignalClassifier {
int consistent_classification_counter_;
SignalType last_signal_type_;
const OouraFft ooura_fft_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SignalClassifier);
};
} // namespace webrtc

1
modules/audio_processing/include/audio_processing.h
View file

@ -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"

2
modules/audio_processing/include/config.h
View file

@ -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 @@ class RTC_EXPORT Config {
typedef std::map<ConfigOptionID, BaseOption*> OptionMap;
OptionMap options_;
// RTC_DISALLOW_COPY_AND_ASSIGN
Config(const Config&);
void operator=(const Config&);
};

6
modules/audio_processing/logging/apm_data_dumper.h
View file

@ -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 @@ class ApmDataDumper {
// 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 @@ class ApmDataDumper {
int num_channels,
WavFile::SampleFormat format);
#endif
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ApmDataDumper);
};
} // namespace webrtc

7
modules/audio_processing/test/aec_dump_based_simulator.h
View file

@ -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 @@ class AecDumpBasedSimulator final : public AudioProcessingSimulator {
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 @@ class AecDumpBasedSimulator final : public AudioProcessingSimulator {
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

8
modules/audio_processing/test/audio_processing_simulator.h
View file

@ -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 @@ class AudioProcessingSimulator {
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 @@ class AudioProcessingSimulator {
FakeRecordingDevice fake_recording_device_;
TaskQueueForTest worker_queue_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
};
} // namespace test

8
modules/audio_processing/test/wav_based_simulator.h
View file

@ -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 @@ class WavBasedSimulator final : public AudioProcessingSimulator {
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 @@ class WavBasedSimulator final : public AudioProcessingSimulator {
const std::string& filename);
std::vector<SimulationEventType> call_chain_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WavBasedSimulator);
};
} // namespace test

1
modules/congestion_controller/goog_cc/BUILD.gn
View file

@ -44,7 +44,6 @@ rtc_library("goog_cc") {
"../../../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",

7
modules/congestion_controller/goog_cc/delay_based_bwe.h
View file

@ -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 @@ class DelayBasedBwe {
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 @@ class DelayBasedBwe {
bool has_once_detected_overuse_;
BandwidthUsage prev_state_;
bool alr_limited_backoff_enabled_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DelayBasedBwe);
};
} // namespace webrtc

8
modules/congestion_controller/goog_cc/goog_cc_network_control.h
View file

@ -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 @@ class GoogCcNetworkController : public NetworkControllerInterface {
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 @@ class GoogCcNetworkController : public NetworkControllerInterface {
bool previously_in_alr_ = false;
absl::optional<DataSize> current_data_window_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GoogCcNetworkController);
};
} // namespace webrtc

8
modules/congestion_controller/include/receive_side_congestion_controller.h
View file

@ -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 @@ class ReceiveSideCongestionController : public CallStatsObserver,
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 @@ class ReceiveSideCongestionController : public CallStatsObserver,
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_;

14
modules/desktop_capture/win/scoped_gdi_object.h
View file

@ -58,27 +58,29 @@ class ScopedGDIObject {
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;

8
modules/remote_bitrate_estimator/inter_arrival.h
View file

@ -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 @@ class InterArrival {
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 @@ class InterArrival {
double timestamp_to_ms_coeff_;
bool burst_grouping_;
int num_consecutive_reordered_packets_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(InterArrival);
};
} // namespace webrtc

10
modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
View file

@ -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 @@ class RemoteBitrateEstimatorAbsSendTime : public RemoteBitrateEstimator {
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 @@ class RemoteBitrateEstimatorAbsSendTime : public RemoteBitrateEstimator {
mutable Mutex mutex_;
Ssrcs ssrcs_ RTC_GUARDED_BY(&mutex_);
AimdRateControl remote_rate_ RTC_GUARDED_BY(&mutex_);
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RemoteBitrateEstimatorAbsSendTime);
};
} // namespace webrtc

10
modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
View file

@ -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 @@ class RemoteBitrateEstimatorSingleStream : public RemoteBitrateEstimator {
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 @@ class RemoteBitrateEstimatorSingleStream : public RemoteBitrateEstimator {
int64_t last_process_time_;
int64_t process_interval_ms_ RTC_GUARDED_BY(mutex_);
bool uma_recorded_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RemoteBitrateEstimatorSingleStream);
};
} // namespace webrtc

7
modules/remote_bitrate_estimator/test/bwe_test_logging.h
View file

@ -263,10 +263,11 @@ class Logging {
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();

7
modules/rtp_rtcp/source/rtcp_sender.cc
View file

@ -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 @@ class PacketContainer : public rtcp::CompoundPacket {
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 @@ class PacketContainer : public rtcp::CompoundPacket {
private:
Transport* transport_;
RtcEventLog* const event_log_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(PacketContainer);
};
// Helper to put several RTCP packets into lower layer datagram RTCP packet.

8
modules/rtp_rtcp/source/rtcp_sender.h
View file

@ -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 @@ class RTCPSender final {
};
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 @@ class RTCPSender final {
const RtcpContext&);
// Map from RTCPPacketType to builder.
std::map<uint32_t, BuilderFunc> builders_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTCPSender);
};
} // namespace webrtc

8
modules/rtp_rtcp/source/rtp_packet_history.h
View file

@ -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 @@ class RtpPacketHistory {
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 @@ class RtpPacketHistory {
// 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_

7
modules/rtp_rtcp/source/rtp_sender.h
View file

@ -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 @@ class RTPSender {
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 @@ class RTPSender {
bool supports_bwe_extension_ RTC_GUARDED_BY(send_mutex_);
RateLimiter* const retransmission_rate_limiter_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
};
} // namespace webrtc

8
modules/rtp_rtcp/source/rtp_sender_audio.h
View file

@ -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 @@ namespace webrtc {
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 @@ class RTPSenderAudio {
const FieldTrialBasedConfig field_trials_;
const bool include_capture_clock_offset_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSenderAudio);
};
} // namespace webrtc

15
modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
View file

@ -76,23 +76,26 @@ struct MultiplexDecoderAdapter::DecodedImageData {
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(

7
modules/video_coding/frame_buffer2.h
View file

@ -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 @@ class FrameBuffer {
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 @@ class FrameBuffer {
// rtt_mult experiment settings.
const absl::optional<RttMultExperiment::Settings> rtt_mult_settings_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FrameBuffer);
};
} // namespace video_coding

7
pc/audio_track.h
View file

@ -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 @@ class AudioTrack : public MediaStreamTrack<AudioTrackInterface>,
// 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 @@ class AudioTrack : public MediaStreamTrack<AudioTrackInterface>,
private:
const rtc::scoped_refptr<AudioSourceInterface> audio_source_;
rtc::ThreadChecker thread_checker_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTrack);
};
} // namespace webrtc

6
pc/ice_transport.h
View file

@ -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 @@ class IceTransportWithPointer : public IceTransportInterface {
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 @@ class IceTransportWithPointer : public IceTransportInterface {
~IceTransportWithPointer() override;
private:
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(IceTransportWithPointer);
const rtc::Thread* creator_thread_;
cricket::IceTransportInternal* internal_ RTC_GUARDED_BY(creator_thread_);
};

8
pc/remote_audio_source.cc
View file

@ -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 @@ class RemoteAudioSource::AudioDataProxy : public AudioSinkInterface {
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 @@ class RemoteAudioSource::AudioDataProxy : public AudioSinkInterface {
private:
const rtc::scoped_refptr<RemoteAudioSource> source_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioDataProxy);
};
RemoteAudioSource::RemoteAudioSource(rtc::Thread* worker_thread)

3
rtc_base/BUILD.gn
View file

@ -599,7 +599,6 @@ rtc_library("rtc_numerics") {
]
deps = [
":checks",
":macromagic",
":rtc_base_approved",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
@ -1066,7 +1065,6 @@ rtc_library("testclient") {
deps = [
":criticalsection",
":gunit_helpers",
":macromagic",
":rtc_base",
":rtc_base_tests_utils",
":timeutils",
@ -1150,7 +1148,6 @@ rtc_library("task_queue_for_test") {
]
deps = [
":checks",
":macromagic",
":rtc_base_approved",
":rtc_event",
":rtc_task_queue",

16
rtc_base/constructor_magic.h
View file

@ -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_

14
rtc_base/deprecated/signal_thread.h
View file

@ -110,14 +110,17 @@ class DEPRECATED_SignalThread : public sigslot::has_slots<>,
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 @@ class DEPRECATED_SignalThread : public sigslot::has_slots<>,
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 @@ class DEPRECATED_SignalThread : public sigslot::has_slots<>,
private:
DEPRECATED_SignalThread* t_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EnterExit);
};
void Run();

7
rtc_base/random.h
View file

@ -16,7 +16,6 @@
#include <limits>
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -35,6 +34,10 @@ class Random {
// 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 @@ class Random {
}
uint64_t state_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Random);
};
// Return pseudo-random number in the interval [0.0, 1.0).

8
rtc_base/rate_limiter.h
View file

@ -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 Clock;
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 @@ class RateLimiter {
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

5
rtc_base/weak_ptr.h
View file

@ -241,6 +241,10 @@ class WeakPtrFactory {
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 @@ class WeakPtrFactory {
private:
internal::WeakReferenceOwner weak_reference_owner_;
T* ptr_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WeakPtrFactory);
};
} // namespace rtc

9
rtc_tools/rtp_generator/rtp_generator.h
View file

@ -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 @@ class RtpGenerator final : public webrtc::Transport {
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 @@ class RtpGenerator final : public webrtc::Transport {
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

1
video/BUILD.gn
View file

@ -241,7 +241,6 @@ rtc_library("video_stream_encoder_impl") {
"../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",

1
video/adaptation/BUILD.gn
View file

@ -42,7 +42,6 @@ rtc_library("video_adaptation") {
"../../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",