diff options
author | Byoungchan Lee <daniel.l@hpcnt.com> | 2022-01-21 09:49:39 +0900 |
---|---|---|
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | 2022-01-24 11:50:20 +0000 |
commit | 604fd2f1ab24a229b8b75fae6ac4fac433156acf (patch) | |
tree | 76719829133bb6d8f18226cc455b3e4f1cb37ff6 /modules | |
parent | ce6170fcdfa5654fc015e13934bccca4e8997878 (diff) | |
download | webrtc-604fd2f1ab24a229b8b75fae6ac4fac433156acf.tar.gz |
Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/
Bug: webrtc:13555, webrtc:13082
Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35771}
Diffstat (limited to 'modules')
127 files changed, 466 insertions, 393 deletions
diff --git a/modules/audio_coding/acm2/acm_receive_test.h b/modules/audio_coding/acm2/acm_receive_test.h index 6349c6392a..2095ef9025 100644 --- a/modules/audio_coding/acm2/acm_receive_test.h +++ b/modules/audio_coding/acm2/acm_receive_test.h @@ -18,7 +18,6 @@ #include "api/audio_codecs/audio_decoder_factory.h" #include "api/scoped_refptr.h" -#include "rtc_base/constructor_magic.h" #include "system_wrappers/include/clock.h" namespace webrtc { @@ -45,6 +44,9 @@ class AcmReceiveTestOldApi { rtc::scoped_refptr<AudioDecoderFactory> decoder_factory); virtual ~AcmReceiveTestOldApi(); + AcmReceiveTestOldApi(const AcmReceiveTestOldApi&) = delete; + AcmReceiveTestOldApi& operator=(const AcmReceiveTestOldApi&) = delete; + // Registers the codecs with default parameters from ACM. void RegisterDefaultCodecs(); @@ -67,8 +69,6 @@ class AcmReceiveTestOldApi { AudioSink* audio_sink_; int output_freq_hz_; NumOutputChannels exptected_output_channels_; - - RTC_DISALLOW_COPY_AND_ASSIGN(AcmReceiveTestOldApi); }; // This test toggles the output frequency every `toggle_period_ms`. The test diff --git a/modules/audio_coding/acm2/acm_send_test.h b/modules/audio_coding/acm2/acm_send_test.h index 0c82415d11..b14cb80c6a 100644 --- a/modules/audio_coding/acm2/acm_send_test.h +++ b/modules/audio_coding/acm2/acm_send_test.h @@ -17,7 +17,6 @@ #include "api/audio/audio_frame.h" #include "modules/audio_coding/include/audio_coding_module.h" #include "modules/audio_coding/neteq/tools/packet_source.h" -#include "rtc_base/constructor_magic.h" #include "system_wrappers/include/clock.h" namespace webrtc { @@ -35,6 +34,9 @@ class AcmSendTestOldApi : public AudioPacketizationCallback, int test_duration_ms); ~AcmSendTestOldApi() override; + AcmSendTestOldApi(const AcmSendTestOldApi&) = delete; + AcmSendTestOldApi& operator=(const AcmSendTestOldApi&) = delete; + // Registers the send codec. Returns true on success, false otherwise. bool RegisterCodec(const char* payload_name, int sampling_freq_hz, @@ -81,8 +83,6 @@ class AcmSendTestOldApi : public AudioPacketizationCallback, uint16_t sequence_number_; std::vector<uint8_t> last_payload_vec_; bool data_to_send_; - - RTC_DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi); }; } // namespace test diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h index 1c91fa19a8..664e76bda5 100644 --- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h +++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h @@ -21,7 +21,6 @@ #include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h" #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -44,6 +43,9 @@ class AudioNetworkAdaptorImpl final : public AudioNetworkAdaptor { ~AudioNetworkAdaptorImpl() override; + AudioNetworkAdaptorImpl(const AudioNetworkAdaptorImpl&) = delete; + AudioNetworkAdaptorImpl& operator=(const AudioNetworkAdaptorImpl&) = delete; + void SetUplinkBandwidth(int uplink_bandwidth_bps) override; void SetUplinkPacketLossFraction(float uplink_packet_loss_fraction) override; @@ -80,8 +82,6 @@ class AudioNetworkAdaptorImpl final : public AudioNetworkAdaptor { absl::optional<AudioEncoderRuntimeConfig> prev_config_; ANAStats stats_; - - RTC_DISALLOW_COPY_AND_ASSIGN(AudioNetworkAdaptorImpl); }; } // namespace webrtc diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller.h b/modules/audio_coding/audio_network_adaptor/bitrate_controller.h index 41bfbd1c32..c1032146cc 100644 --- a/modules/audio_coding/audio_network_adaptor/bitrate_controller.h +++ b/modules/audio_coding/audio_network_adaptor/bitrate_controller.h @@ -16,7 +16,6 @@ #include "absl/types/optional.h" #include "modules/audio_coding/audio_network_adaptor/controller.h" #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { namespace audio_network_adaptor { @@ -39,6 +38,9 @@ class BitrateController final : public Controller { ~BitrateController() override; + BitrateController(const BitrateController&) = delete; + BitrateController& operator=(const BitrateController&) = delete; + void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override; void MakeDecision(AudioEncoderRuntimeConfig* config) override; @@ -49,7 +51,6 @@ class BitrateController final : public Controller { int frame_length_ms_; absl::optional<int> target_audio_bitrate_bps_; absl::optional<size_t> overhead_bytes_per_packet_; - RTC_DISALLOW_COPY_AND_ASSIGN(BitrateController); }; } // namespace audio_network_adaptor diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller.h b/modules/audio_coding/audio_network_adaptor/channel_controller.h index f211f40f17..3cd4bb7dec 100644 --- a/modules/audio_coding/audio_network_adaptor/channel_controller.h +++ b/modules/audio_coding/audio_network_adaptor/channel_controller.h @@ -16,7 +16,6 @@ #include "absl/types/optional.h" #include "modules/audio_coding/audio_network_adaptor/controller.h" #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -41,6 +40,9 @@ class ChannelController final : public Controller { ~ChannelController() override; + ChannelController(const ChannelController&) = delete; + ChannelController& operator=(const ChannelController&) = delete; + void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override; void MakeDecision(AudioEncoderRuntimeConfig* config) override; @@ -49,7 +51,6 @@ class ChannelController final : public Controller { const Config config_; size_t channels_to_encode_; absl::optional<int> uplink_bandwidth_bps_; - RTC_DISALLOW_COPY_AND_ASSIGN(ChannelController); }; } // namespace webrtc diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.h b/modules/audio_coding/audio_network_adaptor/controller_manager.h index c168ebc6ce..f7d7b34fb1 100644 --- a/modules/audio_coding/audio_network_adaptor/controller_manager.h +++ b/modules/audio_coding/audio_network_adaptor/controller_manager.h @@ -17,7 +17,6 @@ #include <vector> #include "modules/audio_coding/audio_network_adaptor/controller.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -80,6 +79,9 @@ class ControllerManagerImpl final : public ControllerManager { ~ControllerManagerImpl() override; + ControllerManagerImpl(const ControllerManagerImpl&) = delete; + ControllerManagerImpl& operator=(const ControllerManagerImpl&) = delete; + // Sort controllers based on their significance. std::vector<Controller*> GetSortedControllers( const Controller::NetworkMetrics& metrics) override; @@ -114,8 +116,6 @@ class ControllerManagerImpl final : public ControllerManager { // `scoring_points_` saves the scoring points of various // controllers. std::map<const Controller*, ScoringPoint> controller_scoring_points_; - - RTC_DISALLOW_COPY_AND_ASSIGN(ControllerManagerImpl); }; } // namespace webrtc diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller.h b/modules/audio_coding/audio_network_adaptor/dtx_controller.h index 83fdf3ddd7..b8a8e476e4 100644 --- a/modules/audio_coding/audio_network_adaptor/dtx_controller.h +++ b/modules/audio_coding/audio_network_adaptor/dtx_controller.h @@ -14,7 +14,6 @@ #include "absl/types/optional.h" #include "modules/audio_coding/audio_network_adaptor/controller.h" #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -35,6 +34,9 @@ class DtxController final : public Controller { ~DtxController() override; + DtxController(const DtxController&) = delete; + DtxController& operator=(const DtxController&) = delete; + void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override; void MakeDecision(AudioEncoderRuntimeConfig* config) override; @@ -43,7 +45,6 @@ class DtxController final : public Controller { const Config config_; bool dtx_enabled_; absl::optional<int> uplink_bandwidth_bps_; - RTC_DISALLOW_COPY_AND_ASSIGN(DtxController); }; } // namespace webrtc diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer.h b/modules/audio_coding/audio_network_adaptor/event_log_writer.h index c5e57e63e6..a147311fc7 100644 --- a/modules/audio_coding/audio_network_adaptor/event_log_writer.h +++ b/modules/audio_coding/audio_network_adaptor/event_log_writer.h @@ -12,7 +12,6 @@ #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_ #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { class RtcEventLog; @@ -24,6 +23,10 @@ class EventLogWriter final { float min_bitrate_change_fraction, float min_packet_loss_change_fraction); ~EventLogWriter(); + + EventLogWriter(const EventLogWriter&) = delete; + EventLogWriter& operator=(const EventLogWriter&) = delete; + void MaybeLogEncoderConfig(const AudioEncoderRuntimeConfig& config); private: @@ -34,7 +37,6 @@ class EventLogWriter final { const float min_bitrate_change_fraction_; const float min_packet_loss_change_fraction_; AudioEncoderRuntimeConfig last_logged_config_; - RTC_DISALLOW_COPY_AND_ASSIGN(EventLogWriter); }; } // namespace webrtc diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h index 85d235ed26..0c57ad1d1e 100644 --- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h +++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h @@ -18,7 +18,6 @@ #include "modules/audio_coding/audio_network_adaptor/controller.h" #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -53,6 +52,9 @@ class FecControllerPlrBased final : public Controller { ~FecControllerPlrBased() override; + FecControllerPlrBased(const FecControllerPlrBased&) = delete; + FecControllerPlrBased& operator=(const FecControllerPlrBased&) = delete; + void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override; void MakeDecision(AudioEncoderRuntimeConfig* config) override; @@ -65,8 +67,6 @@ class FecControllerPlrBased final : public Controller { bool fec_enabled_; absl::optional<int> uplink_bandwidth_bps_; const std::unique_ptr<SmoothingFilter> packet_loss_smoother_; - - RTC_DISALLOW_COPY_AND_ASSIGN(FecControllerPlrBased); }; } // namespace webrtc diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller.h b/modules/audio_coding/audio_network_adaptor/frame_length_controller.h index 74a787e1c1..04693f8db7 100644 --- a/modules/audio_coding/audio_network_adaptor/frame_length_controller.h +++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller.h @@ -19,7 +19,6 @@ #include "absl/types/optional.h" #include "modules/audio_coding/audio_network_adaptor/controller.h" #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -62,6 +61,9 @@ class FrameLengthController final : public Controller { ~FrameLengthController() override; + FrameLengthController(const FrameLengthController&) = delete; + FrameLengthController& operator=(const FrameLengthController&) = delete; + void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override; void MakeDecision(AudioEncoderRuntimeConfig* config) override; @@ -84,8 +86,6 @@ class FrameLengthController final : public Controller { // True if the previous frame length decision was an increase, otherwise // false. bool prev_decision_increase_ = false; - - RTC_DISALLOW_COPY_AND_ASSIGN(FrameLengthController); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/modules/audio_coding/codecs/g711/audio_decoder_pcm.h index 618591876d..3fa42cba30 100644 --- a/modules/audio_coding/codecs/g711/audio_decoder_pcm.h +++ b/modules/audio_coding/codecs/g711/audio_decoder_pcm.h @@ -19,7 +19,6 @@ #include "api/audio_codecs/audio_decoder.h" #include "rtc_base/buffer.h" #include "rtc_base/checks.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -28,6 +27,10 @@ class AudioDecoderPcmU final : public AudioDecoder { explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) { RTC_DCHECK_GE(num_channels, 1); } + + AudioDecoderPcmU(const AudioDecoderPcmU&) = delete; + AudioDecoderPcmU& operator=(const AudioDecoderPcmU&) = delete; + void Reset() override; std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, uint32_t timestamp) override; @@ -44,7 +47,6 @@ class AudioDecoderPcmU final : public AudioDecoder { private: const size_t num_channels_; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmU); }; class AudioDecoderPcmA final : public AudioDecoder { @@ -52,6 +54,10 @@ class AudioDecoderPcmA final : public AudioDecoder { explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) { RTC_DCHECK_GE(num_channels, 1); } + + AudioDecoderPcmA(const AudioDecoderPcmA&) = delete; + AudioDecoderPcmA& operator=(const AudioDecoderPcmA&) = delete; + void Reset() override; std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, uint32_t timestamp) override; @@ -68,7 +74,6 @@ class AudioDecoderPcmA final : public AudioDecoder { private: const size_t num_channels_; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmA); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h index c4413f50a4..d50be4b457 100644 --- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h +++ b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h @@ -17,7 +17,6 @@ #include "absl/types/optional.h" #include "api/audio_codecs/audio_encoder.h" #include "api/units/time_delta.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -83,6 +82,9 @@ class AudioEncoderPcmA final : public AudioEncoderPcm { explicit AudioEncoderPcmA(const Config& config) : AudioEncoderPcm(config, kSampleRateHz) {} + AudioEncoderPcmA(const AudioEncoderPcmA&) = delete; + AudioEncoderPcmA& operator=(const AudioEncoderPcmA&) = delete; + protected: size_t EncodeCall(const int16_t* audio, size_t input_len, @@ -94,7 +96,6 @@ class AudioEncoderPcmA final : public AudioEncoderPcm { private: static const int kSampleRateHz = 8000; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmA); }; class AudioEncoderPcmU final : public AudioEncoderPcm { @@ -106,6 +107,9 @@ class AudioEncoderPcmU final : public AudioEncoderPcm { explicit AudioEncoderPcmU(const Config& config) : AudioEncoderPcm(config, kSampleRateHz) {} + AudioEncoderPcmU(const AudioEncoderPcmU&) = delete; + AudioEncoderPcmU& operator=(const AudioEncoderPcmU&) = delete; + protected: size_t EncodeCall(const int16_t* audio, size_t input_len, @@ -117,7 +121,6 @@ class AudioEncoderPcmU final : public AudioEncoderPcm { private: static const int kSampleRateHz = 8000; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmU); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/g722/audio_decoder_g722.h b/modules/audio_coding/codecs/g722/audio_decoder_g722.h index eeca13975f..39e9e630be 100644 --- a/modules/audio_coding/codecs/g722/audio_decoder_g722.h +++ b/modules/audio_coding/codecs/g722/audio_decoder_g722.h @@ -12,7 +12,6 @@ #define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_ #include "api/audio_codecs/audio_decoder.h" -#include "rtc_base/constructor_magic.h" typedef struct WebRtcG722DecInst G722DecInst; @@ -22,6 +21,10 @@ class AudioDecoderG722Impl final : public AudioDecoder { public: AudioDecoderG722Impl(); ~AudioDecoderG722Impl() override; + + AudioDecoderG722Impl(const AudioDecoderG722Impl&) = delete; + AudioDecoderG722Impl& operator=(const AudioDecoderG722Impl&) = delete; + bool HasDecodePlc() const override; void Reset() override; std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, @@ -39,13 +42,17 @@ class AudioDecoderG722Impl final : public AudioDecoder { private: G722DecInst* dec_state_; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722Impl); }; class AudioDecoderG722StereoImpl final : public AudioDecoder { public: AudioDecoderG722StereoImpl(); ~AudioDecoderG722StereoImpl() override; + + AudioDecoderG722StereoImpl(const AudioDecoderG722StereoImpl&) = delete; + AudioDecoderG722StereoImpl& operator=(const AudioDecoderG722StereoImpl&) = + delete; + void Reset() override; std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, uint32_t timestamp) override; @@ -71,7 +78,6 @@ class AudioDecoderG722StereoImpl final : public AudioDecoder { G722DecInst* dec_state_left_; G722DecInst* dec_state_right_; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722StereoImpl); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/modules/audio_coding/codecs/g722/audio_encoder_g722.h index c836503f2b..a932aa8b7d 100644 --- a/modules/audio_coding/codecs/g722/audio_encoder_g722.h +++ b/modules/audio_coding/codecs/g722/audio_encoder_g722.h @@ -20,7 +20,6 @@ #include "api/units/time_delta.h" #include "modules/audio_coding/codecs/g722/g722_interface.h" #include "rtc_base/buffer.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -29,6 +28,9 @@ class AudioEncoderG722Impl final : public AudioEncoder { AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type); ~AudioEncoderG722Impl() override; + AudioEncoderG722Impl(const AudioEncoderG722Impl&) = delete; + AudioEncoderG722Impl& operator=(const AudioEncoderG722Impl&) = delete; + int SampleRateHz() const override; size_t NumChannels() const override; int RtpTimestampRateHz() const override; @@ -63,7 +65,6 @@ class AudioEncoderG722Impl final : public AudioEncoder { uint32_t first_timestamp_in_buffer_; const std::unique_ptr<EncoderState[]> encoders_; rtc::Buffer interleave_buffer_; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722Impl); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h b/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h index c2d62ed2d1..46ba755148 100644 --- a/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h +++ b/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h @@ -18,7 +18,6 @@ #include "api/audio_codecs/audio_decoder.h" #include "rtc_base/buffer.h" -#include "rtc_base/constructor_magic.h" typedef struct iLBC_decinst_t_ IlbcDecoderInstance; @@ -28,6 +27,10 @@ class AudioDecoderIlbcImpl final : public AudioDecoder { public: AudioDecoderIlbcImpl(); ~AudioDecoderIlbcImpl() override; + + AudioDecoderIlbcImpl(const AudioDecoderIlbcImpl&) = delete; + AudioDecoderIlbcImpl& operator=(const AudioDecoderIlbcImpl&) = delete; + bool HasDecodePlc() const override; size_t DecodePlc(size_t num_frames, int16_t* decoded) override; void Reset() override; @@ -45,7 +48,6 @@ class AudioDecoderIlbcImpl final : public AudioDecoder { private: IlbcDecoderInstance* dec_state_; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIlbcImpl); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h index 05a900e3c4..c8dfa2ca6d 100644 --- a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h +++ b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h @@ -21,7 +21,6 @@ #include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h" #include "api/units/time_delta.h" #include "modules/audio_coding/codecs/ilbc/ilbc.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -30,6 +29,9 @@ class AudioEncoderIlbcImpl final : public AudioEncoder { AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, int payload_type); ~AudioEncoderIlbcImpl() override; + AudioEncoderIlbcImpl(const AudioEncoderIlbcImpl&) = delete; + AudioEncoderIlbcImpl& operator=(const AudioEncoderIlbcImpl&) = delete; + int SampleRateHz() const override; size_t NumChannels() const override; size_t Num10MsFramesInNextPacket() const override; @@ -53,7 +55,6 @@ class AudioEncoderIlbcImpl final : public AudioEncoder { uint32_t first_timestamp_in_buffer_; int16_t input_buffer_[kMaxSamplesPerPacket]; IlbcEncoderInstance* encoder_; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIlbcImpl); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h b/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h index 23a302018f..aae708f295 100644 --- a/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h +++ b/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h @@ -16,7 +16,6 @@ #include "absl/types/optional.h" #include "api/audio_codecs/audio_decoder.h" #include "api/scoped_refptr.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -30,6 +29,9 @@ class AudioDecoderIsacT final : public AudioDecoder { explicit AudioDecoderIsacT(const Config& config); virtual ~AudioDecoderIsacT() override; + AudioDecoderIsacT(const AudioDecoderIsacT&) = delete; + AudioDecoderIsacT& operator=(const AudioDecoderIsacT&) = delete; + bool HasDecodePlc() const override; size_t DecodePlc(size_t num_frames, int16_t* decoded) override; void Reset() override; @@ -45,8 +47,6 @@ class AudioDecoderIsacT final : public AudioDecoder { private: typename T::instance_type* isac_state_; int sample_rate_hz_; - - RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h index 8bde0e34ad..c382ea076e 100644 --- a/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h +++ b/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h @@ -18,7 +18,6 @@ #include "api/audio_codecs/audio_encoder.h" #include "api/scoped_refptr.h" #include "api/units/time_delta.h" -#include "rtc_base/constructor_magic.h" #include "system_wrappers/include/field_trial.h" namespace webrtc { @@ -44,6 +43,9 @@ class AudioEncoderIsacT final : public AudioEncoder { explicit AudioEncoderIsacT(const Config& config); ~AudioEncoderIsacT() override; + AudioEncoderIsacT(const AudioEncoderIsacT&) = delete; + AudioEncoderIsacT& operator=(const AudioEncoderIsacT&) = delete; + int SampleRateHz() const override; size_t NumChannels() const override; size_t Num10MsFramesInNextPacket() const override; @@ -99,8 +101,6 @@ class AudioEncoderIsacT final : public AudioEncoder { // Start out with a reasonable default that we can use until we receive a real // value. DataSize overhead_per_packet_ = DataSize::Bytes(28); - - RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h index efc3f0dda8..2ff47a8a53 100644 --- a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h +++ b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h @@ -21,7 +21,6 @@ #include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h" #include "modules/audio_coding/codecs/opus/opus_interface.h" #include "rtc_base/buffer.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -32,6 +31,11 @@ class AudioDecoderMultiChannelOpusImpl final : public AudioDecoder { ~AudioDecoderMultiChannelOpusImpl() override; + AudioDecoderMultiChannelOpusImpl(const AudioDecoderMultiChannelOpusImpl&) = + delete; + AudioDecoderMultiChannelOpusImpl& operator=( + const AudioDecoderMultiChannelOpusImpl&) = delete; + std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, uint32_t timestamp) override; void Reset() override; @@ -63,7 +67,6 @@ class AudioDecoderMultiChannelOpusImpl final : public AudioDecoder { OpusDecInst* dec_state_; const AudioDecoderMultiChannelOpusConfig config_; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderMultiChannelOpusImpl); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/modules/audio_coding/codecs/opus/audio_decoder_opus.h index c79272284d..e8fd0440bc 100644 --- a/modules/audio_coding/codecs/opus/audio_decoder_opus.h +++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.h @@ -19,7 +19,6 @@ #include "api/audio_codecs/audio_decoder.h" #include "modules/audio_coding/codecs/opus/opus_interface.h" #include "rtc_base/buffer.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -29,6 +28,9 @@ class AudioDecoderOpusImpl final : public AudioDecoder { int sample_rate_hz = 48000); ~AudioDecoderOpusImpl() override; + AudioDecoderOpusImpl(const AudioDecoderOpusImpl&) = delete; + AudioDecoderOpusImpl& operator=(const AudioDecoderOpusImpl&) = delete; + std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, uint32_t timestamp) override; void Reset() override; @@ -55,7 +57,6 @@ class AudioDecoderOpusImpl final : public AudioDecoder { OpusDecInst* dec_state_; const size_t channels_; const int sample_rate_hz_; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpusImpl); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h index eadb4a6eb9..8a7210515c 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h +++ b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h @@ -21,7 +21,6 @@ #include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h" #include "api/units/time_delta.h" #include "modules/audio_coding/codecs/opus/opus_interface.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -34,6 +33,11 @@ class AudioEncoderMultiChannelOpusImpl final : public AudioEncoder { int payload_type); ~AudioEncoderMultiChannelOpusImpl() override; + AudioEncoderMultiChannelOpusImpl(const AudioEncoderMultiChannelOpusImpl&) = + delete; + AudioEncoderMultiChannelOpusImpl& operator=( + const AudioEncoderMultiChannelOpusImpl&) = delete; + // Static interface for use by BuiltinAudioEncoderFactory. static constexpr const char* GetPayloadName() { return "multiopus"; } static absl::optional<AudioCodecInfo> QueryAudioEncoder( @@ -81,7 +85,6 @@ class AudioEncoderMultiChannelOpusImpl final : public AudioEncoder { int next_frame_length_ms_; friend struct AudioEncoderMultiChannelOpus; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderMultiChannelOpusImpl); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/modules/audio_coding/codecs/opus/audio_encoder_opus.h index c7ee4f4523..14477cc317 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_opus.h +++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.h @@ -23,7 +23,6 @@ #include "common_audio/smoothing_filter.h" #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" #include "modules/audio_coding/codecs/opus/opus_interface.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -61,6 +60,9 @@ class AudioEncoderOpusImpl final : public AudioEncoder { AudioEncoderOpusImpl(int payload_type, const SdpAudioFormat& format); ~AudioEncoderOpusImpl() override; + AudioEncoderOpusImpl(const AudioEncoderOpusImpl&) = delete; + AudioEncoderOpusImpl& operator=(const AudioEncoderOpusImpl&) = delete; + int SampleRateHz() const override; size_t NumChannels() const override; int RtpTimestampRateHz() const override; @@ -175,7 +177,6 @@ class AudioEncoderOpusImpl final : public AudioEncoder { int consecutive_dtx_frames_; friend struct AudioEncoderOpus; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpusImpl); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h index f08c4a6298..6f50161d3f 100644 --- a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h +++ b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h @@ -18,13 +18,16 @@ #include "api/audio_codecs/audio_decoder.h" #include "rtc_base/buffer.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { class AudioDecoderPcm16B final : public AudioDecoder { public: AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels); + + AudioDecoderPcm16B(const AudioDecoderPcm16B&) = delete; + AudioDecoderPcm16B& operator=(const AudioDecoderPcm16B&) = delete; + void Reset() override; std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, uint32_t timestamp) override; @@ -42,7 +45,6 @@ class AudioDecoderPcm16B final : public AudioDecoder { private: const int sample_rate_hz_; const size_t num_channels_; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcm16B); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h index 71c757250a..c363b40b3f 100644 --- a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h +++ b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h @@ -12,7 +12,6 @@ #define MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_ #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -29,6 +28,9 @@ class AudioEncoderPcm16B final : public AudioEncoderPcm { explicit AudioEncoderPcm16B(const Config& config) : AudioEncoderPcm(config, config.sample_rate_hz) {} + AudioEncoderPcm16B(const AudioEncoderPcm16B&) = delete; + AudioEncoderPcm16B& operator=(const AudioEncoderPcm16B&) = delete; + protected: size_t EncodeCall(const int16_t* audio, size_t input_len, @@ -37,9 +39,6 @@ class AudioEncoderPcm16B final : public AudioEncoderPcm { size_t BytesPerSample() const override; AudioEncoder::CodecType GetCodecType() const override; - - private: - RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcm16B); }; } // namespace webrtc diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h index d5b1bf6868..d163193251 100644 --- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h +++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h @@ -23,7 +23,6 @@ #include "api/audio_codecs/audio_encoder.h" #include "api/units/time_delta.h" #include "rtc_base/buffer.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -47,6 +46,9 @@ class AudioEncoderCopyRed final : public AudioEncoder { ~AudioEncoderCopyRed() override; + AudioEncoderCopyRed(const AudioEncoderCopyRed&) = delete; + AudioEncoderCopyRed& operator=(const AudioEncoderCopyRed&) = delete; + int SampleRateHz() const override; size_t NumChannels() const override; int RtpTimestampRateHz() const override; @@ -92,8 +94,6 @@ class AudioEncoderCopyRed final : public AudioEncoder { size_t max_packet_length_; int red_payload_type_; std::list<std::pair<EncodedInfo, rtc::Buffer>> redundant_encodings_; - - RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCopyRed); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/accelerate.h b/modules/audio_coding/neteq/accelerate.h index e03f609ffb..01fe874d54 100644 --- a/modules/audio_coding/neteq/accelerate.h +++ b/modules/audio_coding/neteq/accelerate.h @@ -15,7 +15,6 @@ #include <stdint.h> #include "modules/audio_coding/neteq/time_stretch.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -33,6 +32,9 @@ class Accelerate : public TimeStretch { const BackgroundNoise& background_noise) : TimeStretch(sample_rate_hz, num_channels, background_noise) {} + Accelerate(const Accelerate&) = delete; + Accelerate& operator=(const Accelerate&) = delete; + // This method performs the actual Accelerate operation. The samples are // read from `input`, of length `input_length` elements, and are written to // `output`. The number of samples removed through time-stretching is @@ -62,9 +64,6 @@ class Accelerate : public TimeStretch { bool active_speech, bool fast_mode, AudioMultiVector* output) const override; - - private: - RTC_DISALLOW_COPY_AND_ASSIGN(Accelerate); }; struct AccelerateFactory { diff --git a/modules/audio_coding/neteq/audio_multi_vector.h b/modules/audio_coding/neteq/audio_multi_vector.h index 10179d7f07..715ec6dfc7 100644 --- a/modules/audio_coding/neteq/audio_multi_vector.h +++ b/modules/audio_coding/neteq/audio_multi_vector.h @@ -18,7 +18,6 @@ #include "api/array_view.h" #include "modules/audio_coding/neteq/audio_vector.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -34,6 +33,9 @@ class AudioMultiVector { virtual ~AudioMultiVector(); + AudioMultiVector(const AudioMultiVector&) = delete; + AudioMultiVector& operator=(const AudioMultiVector&) = delete; + // Deletes all values and make the vector empty. virtual void Clear(); @@ -130,9 +132,6 @@ class AudioMultiVector { protected: std::vector<AudioVector*> channels_; size_t num_channels_; - - private: - RTC_DISALLOW_COPY_AND_ASSIGN(AudioMultiVector); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/audio_vector.h b/modules/audio_coding/neteq/audio_vector.h index c722b56965..d68f3ec6be 100644 --- a/modules/audio_coding/neteq/audio_vector.h +++ b/modules/audio_coding/neteq/audio_vector.h @@ -17,7 +17,6 @@ #include <memory> #include "rtc_base/checks.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -31,6 +30,9 @@ class AudioVector { virtual ~AudioVector(); + AudioVector(const AudioVector&) = delete; + AudioVector& operator=(const AudioVector&) = delete; + // Deletes all values and make the vector empty. virtual void Clear(); @@ -164,8 +166,6 @@ class AudioVector { // The index of the sample after the last sample in `array_`. size_t end_index_; - - RTC_DISALLOW_COPY_AND_ASSIGN(AudioVector); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/background_noise.h b/modules/audio_coding/neteq/background_noise.h index 005b3766fc..8e6d5890a0 100644 --- a/modules/audio_coding/neteq/background_noise.h +++ b/modules/audio_coding/neteq/background_noise.h @@ -16,7 +16,6 @@ #include <memory> #include "api/array_view.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -34,6 +33,9 @@ class BackgroundNoise { explicit BackgroundNoise(size_t num_channels); virtual ~BackgroundNoise(); + BackgroundNoise(const BackgroundNoise&) = delete; + BackgroundNoise& operator=(const BackgroundNoise&) = delete; + void Reset(); // Updates the parameter estimates based on the signal currently in the @@ -130,8 +132,6 @@ class BackgroundNoise { size_t num_channels_; std::unique_ptr<ChannelParameters[]> channel_parameters_; bool initialized_; - - RTC_DISALLOW_COPY_AND_ASSIGN(BackgroundNoise); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/buffer_level_filter.h b/modules/audio_coding/neteq/buffer_level_filter.h index 94a37150e4..ced36da9c2 100644 --- a/modules/audio_coding/neteq/buffer_level_filter.h +++ b/modules/audio_coding/neteq/buffer_level_filter.h @@ -14,14 +14,16 @@ #include <stddef.h> #include <stdint.h> -#include "rtc_base/constructor_magic.h" - namespace webrtc { class BufferLevelFilter { public: BufferLevelFilter(); virtual ~BufferLevelFilter() {} + + BufferLevelFilter(const BufferLevelFilter&) = delete; + BufferLevelFilter& operator=(const BufferLevelFilter&) = delete; + virtual void Reset(); // Updates the filter. Current buffer size is `buffer_size_samples`. @@ -46,8 +48,6 @@ class BufferLevelFilter { private: int level_factor_; // Filter factor for the buffer level filter in Q8. int filtered_current_level_; // Filtered current buffer level in Q8. - - RTC_DISALLOW_COPY_AND_ASSIGN(BufferLevelFilter); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/comfort_noise.h b/modules/audio_coding/neteq/comfort_noise.h index 6419d397d3..31fcee31d0 100644 --- a/modules/audio_coding/neteq/comfort_noise.h +++ b/modules/audio_coding/neteq/comfort_noise.h @@ -13,8 +13,6 @@ #include <stddef.h> -#include "rtc_base/constructor_magic.h" - namespace webrtc { // Forward declarations. @@ -42,6 +40,9 @@ class ComfortNoise { decoder_database_(decoder_database), sync_buffer_(sync_buffer) {} + ComfortNoise(const ComfortNoise&) = delete; + ComfortNoise& operator=(const ComfortNoise&) = delete; + // Resets the state. Should be called before each new comfort noise period. void Reset(); @@ -65,7 +66,6 @@ class ComfortNoise { DecoderDatabase* decoder_database_; SyncBuffer* sync_buffer_; int internal_error_code_; - RTC_DISALLOW_COPY_AND_ASSIGN(ComfortNoise); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/decision_logic.h b/modules/audio_coding/neteq/decision_logic.h index 693f6169e4..a8571ade96 100644 --- a/modules/audio_coding/neteq/decision_logic.h +++ b/modules/audio_coding/neteq/decision_logic.h @@ -18,7 +18,6 @@ #include "api/neteq/tick_timer.h" #include "modules/audio_coding/neteq/buffer_level_filter.h" #include "modules/audio_coding/neteq/delay_manager.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/experiments/field_trial_parser.h" namespace webrtc { @@ -37,6 +36,9 @@ class DecisionLogic : public NetEqController { ~DecisionLogic() override; + DecisionLogic(const DecisionLogic&) = delete; + DecisionLogic& operator=(const DecisionLogic&) = delete; + // Resets object to a clean state. void Reset() override; @@ -192,8 +194,6 @@ class DecisionLogic : public NetEqController { FieldTrialParameter<bool> estimate_dtx_delay_; FieldTrialParameter<bool> time_stretch_cn_; FieldTrialConstrained<int> target_level_window_ms_; - - RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogic); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/decoder_database.h b/modules/audio_coding/neteq/decoder_database.h index a63a9cff18..6c2ce54039 100644 --- a/modules/audio_coding/neteq/decoder_database.h +++ b/modules/audio_coding/neteq/decoder_database.h @@ -20,7 +20,6 @@ #include "api/scoped_refptr.h" #include "modules/audio_coding/codecs/cng/webrtc_cng.h" #include "modules/audio_coding/neteq/packet.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -122,6 +121,9 @@ class DecoderDatabase { virtual ~DecoderDatabase(); + DecoderDatabase(const DecoderDatabase&) = delete; + DecoderDatabase& operator=(const DecoderDatabase&) = delete; + // Returns true if the database is empty. virtual bool Empty() const; @@ -208,8 +210,6 @@ class DecoderDatabase { mutable std::unique_ptr<ComfortNoiseDecoder> active_cng_decoder_; rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; const absl::optional<AudioCodecPairId> codec_pair_id_; - - RTC_DISALLOW_COPY_AND_ASSIGN(DecoderDatabase); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/delay_manager.h b/modules/audio_coding/neteq/delay_manager.h index 410aa94b61..56d108ad11 100644 --- a/modules/audio_coding/neteq/delay_manager.h +++ b/modules/audio_coding/neteq/delay_manager.h @@ -22,7 +22,6 @@ #include "modules/audio_coding/neteq/relative_arrival_delay_tracker.h" #include "modules/audio_coding/neteq/reorder_optimizer.h" #include "modules/audio_coding/neteq/underrun_optimizer.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -52,6 +51,9 @@ class DelayManager { virtual ~DelayManager(); + DelayManager(const DelayManager&) = delete; + DelayManager& operator=(const DelayManager&) = delete; + // Updates the delay manager with a new incoming packet, with `timestamp` from // the RTP header. This updates the statistics and a new target buffer level // is calculated. Returns the relative delay if it can be calculated. If @@ -111,9 +113,7 @@ class DelayManager { int maximum_delay_ms_; // Externally set maximum allowed delay. int packet_len_ms_ = 0; - int target_level_ms_; // Currently preferred buffer level. - - RTC_DISALLOW_COPY_AND_ASSIGN(DelayManager); + int target_level_ms_; // Currently preferred buffer level. }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/dsp_helper.h b/modules/audio_coding/neteq/dsp_helper.h index 7bdeba6ec0..4aead7df18 100644 --- a/modules/audio_coding/neteq/dsp_helper.h +++ b/modules/audio_coding/neteq/dsp_helper.h @@ -16,7 +16,6 @@ #include "modules/audio_coding/neteq/audio_multi_vector.h" #include "modules/audio_coding/neteq/audio_vector.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -150,11 +149,12 @@ class DspHelper { bool compensate_delay, int16_t* output); + DspHelper(const DspHelper&) = delete; + DspHelper& operator=(const DspHelper&) = delete; + private: // Table of constants used in method DspHelper::ParabolicFit(). static const int16_t kParabolaCoefficients[17][3]; - - RTC_DISALLOW_COPY_AND_ASSIGN(DspHelper); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/dtmf_buffer.h b/modules/audio_coding/neteq/dtmf_buffer.h index 9209cae864..62b751525c 100644 --- a/modules/audio_coding/neteq/dtmf_buffer.h +++ b/modules/audio_coding/neteq/dtmf_buffer.h @@ -16,8 +16,6 @@ #include <list> -#include "rtc_base/constructor_magic.h" - namespace webrtc { struct DtmfEvent { @@ -50,6 +48,9 @@ class DtmfBuffer { virtual ~DtmfBuffer(); + DtmfBuffer(const DtmfBuffer&) = delete; + DtmfBuffer& operator=(const DtmfBuffer&) = delete; + // Flushes the buffer. virtual void Flush(); @@ -97,8 +98,6 @@ class DtmfBuffer { static bool CompareEvents(const DtmfEvent& a, const DtmfEvent& b); DtmfList buffer_; - - RTC_DISALLOW_COPY_AND_ASSIGN(DtmfBuffer); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/dtmf_tone_generator.h b/modules/audio_coding/neteq/dtmf_tone_generator.h index 968bc7f8c7..35114f4f49 100644 --- a/modules/audio_coding/neteq/dtmf_tone_generator.h +++ b/modules/audio_coding/neteq/dtmf_tone_generator.h @@ -15,7 +15,6 @@ #include <stdint.h> #include "modules/audio_coding/neteq/audio_multi_vector.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -29,6 +28,10 @@ class DtmfToneGenerator { DtmfToneGenerator(); virtual ~DtmfToneGenerator() {} + + DtmfToneGenerator(const DtmfToneGenerator&) = delete; + DtmfToneGenerator& operator=(const DtmfToneGenerator&) = delete; + virtual int Init(int fs, int event, int attenuation); virtual void Reset(); virtual int Generate(size_t num_samples, AudioMultiVector* output); @@ -48,8 +51,6 @@ class DtmfToneGenerator { int amplitude_; // Amplitude for this event. int16_t sample_history1_[2]; // Last 2 samples for the 1st oscillator. int16_t sample_history2_[2]; // Last 2 samples for the 2nd oscillator. - - RTC_DISALLOW_COPY_AND_ASSIGN(DtmfToneGenerator); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/expand.h b/modules/audio_coding/neteq/expand.h index 2d22b11289..2e64583ec2 100644 --- a/modules/audio_coding/neteq/expand.h +++ b/modules/audio_coding/neteq/expand.h @@ -15,7 +15,6 @@ #include <memory> #include "modules/audio_coding/neteq/audio_vector.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -41,6 +40,9 @@ class Expand { virtual ~Expand(); + Expand(const Expand&) = delete; + Expand& operator=(const Expand&) = delete; + // Resets the object. virtual void Reset(); @@ -134,8 +136,6 @@ class Expand { bool stop_muting_; size_t expand_duration_samples_; std::unique_ptr<ChannelParameters[]> channel_parameters_; - - RTC_DISALLOW_COPY_AND_ASSIGN(Expand); }; struct ExpandFactory { diff --git a/modules/audio_coding/neteq/expand_uma_logger.h b/modules/audio_coding/neteq/expand_uma_logger.h index 246aaffd4f..a29d3532f3 100644 --- a/modules/audio_coding/neteq/expand_uma_logger.h +++ b/modules/audio_coding/neteq/expand_uma_logger.h @@ -17,7 +17,6 @@ #include "absl/types/optional.h" #include "api/neteq/tick_timer.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -36,6 +35,9 @@ class ExpandUmaLogger { ~ExpandUmaLogger(); + ExpandUmaLogger(const ExpandUmaLogger&) = delete; + ExpandUmaLogger& operator=(const ExpandUmaLogger&) = delete; + // In this call, value should be an incremental sample counter. The sample // rate must be strictly positive. void UpdateSampleCounter(uint64_t value, int sample_rate_hz); @@ -48,8 +50,6 @@ class ExpandUmaLogger { absl::optional<uint64_t> last_logged_value_; uint64_t last_value_ = 0; int sample_rate_hz_ = 0; - - RTC_DISALLOW_COPY_AND_ASSIGN(ExpandUmaLogger); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/merge.h b/modules/audio_coding/neteq/merge.h index 13aa31df8e..2f27106bfe 100644 --- a/modules/audio_coding/neteq/merge.h +++ b/modules/audio_coding/neteq/merge.h @@ -12,7 +12,6 @@ #define MODULES_AUDIO_CODING_NETEQ_MERGE_H_ #include "modules/audio_coding/neteq/audio_multi_vector.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -36,6 +35,9 @@ class Merge { SyncBuffer* sync_buffer); virtual ~Merge(); + Merge(const Merge&) = delete; + Merge& operator=(const Merge&) = delete; + // The main method to produce the audio data. The decoded data is supplied in // `input`, having `input_length` samples in total for all channels // (interleaved). The result is written to `output`. The number of channels @@ -93,8 +95,6 @@ class Merge { int16_t input_downsampled_[kInputDownsampLength]; AudioMultiVector expanded_; std::vector<int16_t> temp_data_; - - RTC_DISALLOW_COPY_AND_ASSIGN(Merge); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h index 2522e31a39..e2cd6c6054 100644 --- a/modules/audio_coding/neteq/neteq_impl.h +++ b/modules/audio_coding/neteq/neteq_impl.h @@ -29,7 +29,6 @@ #include "modules/audio_coding/neteq/packet.h" #include "modules/audio_coding/neteq/random_vector.h" #include "modules/audio_coding/neteq/statistics_calculator.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" @@ -124,6 +123,9 @@ class NetEqImpl : public webrtc::NetEq { ~NetEqImpl() override; + NetEqImpl(const NetEqImpl&) = delete; + NetEqImpl& operator=(const NetEqImpl&) = delete; + // Inserts a new packet into NetEq. Returns 0 on success, -1 on failure. int InsertPacket(const RTPHeader& rtp_header, rtc::ArrayView<const uint8_t> payload) override; @@ -399,9 +401,6 @@ class NetEqImpl : public webrtc::NetEq { ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(mutex_); bool no_time_stretching_ RTC_GUARDED_BY(mutex_); // Only used for test. rtc::BufferT<int16_t> concealment_audio_ RTC_GUARDED_BY(mutex_); - - private: - RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/normal.h b/modules/audio_coding/neteq/normal.h index 3607208f11..772293b605 100644 --- a/modules/audio_coding/neteq/normal.h +++ b/modules/audio_coding/neteq/normal.h @@ -17,7 +17,6 @@ #include "api/neteq/neteq.h" #include "modules/audio_coding/neteq/statistics_calculator.h" #include "rtc_base/checks.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/numerics/safe_conversions.h" namespace webrtc { @@ -49,6 +48,9 @@ class Normal { virtual ~Normal() {} + Normal(const Normal&) = delete; + Normal& operator=(const Normal&) = delete; + // Performs the "Normal" operation. The decoder data is supplied in `input`, // having `length` samples in total for all channels (interleaved). The // result is written to `output`. The number of channels allocated in @@ -68,8 +70,6 @@ class Normal { const size_t samples_per_ms_; const int16_t default_win_slope_Q14_; StatisticsCalculator* const statistics_; - - RTC_DISALLOW_COPY_AND_ASSIGN(Normal); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/packet_buffer.h b/modules/audio_coding/neteq/packet_buffer.h index 20a053323a..c6fb47ffbf 100644 --- a/modules/audio_coding/neteq/packet_buffer.h +++ b/modules/audio_coding/neteq/packet_buffer.h @@ -15,7 +15,6 @@ #include "modules/audio_coding/neteq/decoder_database.h" #include "modules/audio_coding/neteq/packet.h" #include "modules/include/module_common_types_public.h" // IsNewerTimestamp -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -51,6 +50,9 @@ class PacketBuffer { // Deletes all packets in the buffer before destroying the buffer. virtual ~PacketBuffer(); + PacketBuffer(const PacketBuffer&) = delete; + PacketBuffer& operator=(const PacketBuffer&) = delete; + // Flushes the buffer and deletes all packets in it. virtual void Flush(StatisticsCalculator* stats); @@ -173,7 +175,6 @@ class PacketBuffer { size_t max_number_of_packets_; PacketList buffer_; const TickTimer* tick_timer_; - RTC_DISALLOW_COPY_AND_ASSIGN(PacketBuffer); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/post_decode_vad.h b/modules/audio_coding/neteq/post_decode_vad.h index 3134d5f3a9..3bd91b9edb 100644 --- a/modules/audio_coding/neteq/post_decode_vad.h +++ b/modules/audio_coding/neteq/post_decode_vad.h @@ -16,7 +16,6 @@ #include "api/audio_codecs/audio_decoder.h" #include "common_audio/vad/include/webrtc_vad.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -31,6 +30,9 @@ class PostDecodeVad { virtual ~PostDecodeVad(); + PostDecodeVad(const PostDecodeVad&) = delete; + PostDecodeVad& operator=(const PostDecodeVad&) = delete; + // Enables post-decode VAD. void Enable(); @@ -63,8 +65,6 @@ class PostDecodeVad { bool active_speech_; int sid_interval_counter_; ::VadInst* vad_instance_; - - RTC_DISALLOW_COPY_AND_ASSIGN(PostDecodeVad); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/preemptive_expand.h b/modules/audio_coding/neteq/preemptive_expand.h index 708ebfd1bd..6338b993fd 100644 --- a/modules/audio_coding/neteq/preemptive_expand.h +++ b/modules/audio_coding/neteq/preemptive_expand.h @@ -15,7 +15,6 @@ #include <stdint.h> #include "modules/audio_coding/neteq/time_stretch.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -36,6 +35,9 @@ class PreemptiveExpand : public TimeStretch { old_data_length_per_channel_(0), overlap_samples_(overlap_samples) {} + PreemptiveExpand(const PreemptiveExpand&) = delete; + PreemptiveExpand& operator=(const PreemptiveExpand&) = delete; + // This method performs the actual PreemptiveExpand operation. The samples are // read from `input`, of length `input_length` elements, and are written to // `output`. The number of samples added through time-stretching is @@ -67,8 +69,6 @@ class PreemptiveExpand : public TimeStretch { private: size_t old_data_length_per_channel_; size_t overlap_samples_; - - RTC_DISALLOW_COPY_AND_ASSIGN(PreemptiveExpand); }; struct PreemptiveExpandFactory { diff --git a/modules/audio_coding/neteq/random_vector.h b/modules/audio_coding/neteq/random_vector.h index 1d3760055b..4a782f1116 100644 --- a/modules/audio_coding/neteq/random_vector.h +++ b/modules/audio_coding/neteq/random_vector.h @@ -14,8 +14,6 @@ #include <stddef.h> #include <stdint.h> -#include "rtc_base/constructor_magic.h" - namespace webrtc { // This class generates pseudo-random samples. @@ -26,6 +24,9 @@ class RandomVector { RandomVector() : seed_(777), seed_increment_(1) {} + RandomVector(const RandomVector&) = delete; + RandomVector& operator=(const RandomVector&) = delete; + void Reset(); void Generate(size_t length, int16_t* output); @@ -39,8 +40,6 @@ class RandomVector { private: uint32_t seed_; int16_t seed_increment_; - - RTC_DISALLOW_COPY_AND_ASSIGN(RandomVector); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/red_payload_splitter.h b/modules/audio_coding/neteq/red_payload_splitter.h index 55660913d5..2f48e4b7d4 100644 --- a/modules/audio_coding/neteq/red_payload_splitter.h +++ b/modules/audio_coding/neteq/red_payload_splitter.h @@ -12,7 +12,6 @@ #define MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_ #include "modules/audio_coding/neteq/packet.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -30,6 +29,9 @@ class RedPayloadSplitter { virtual ~RedPayloadSplitter() {} + RedPayloadSplitter(const RedPayloadSplitter&) = delete; + RedPayloadSplitter& operator=(const RedPayloadSplitter&) = delete; + // Splits each packet in `packet_list` into its separate RED payloads. Each // RED payload is packetized into a Packet. The original elements in // `packet_list` are properly deleted, and replaced by the new packets. @@ -43,9 +45,6 @@ class RedPayloadSplitter { // is accepted. Any packet with another payload type is discarded. virtual void CheckRedPayloads(PacketList* packet_list, const DecoderDatabase& decoder_database); - - private: - RTC_DISALLOW_COPY_AND_ASSIGN(RedPayloadSplitter); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/statistics_calculator.h b/modules/audio_coding/neteq/statistics_calculator.h index 5c3fb75d1b..269e6a09b2 100644 --- a/modules/audio_coding/neteq/statistics_calculator.h +++ b/modules/audio_coding/neteq/statistics_calculator.h @@ -15,7 +15,6 @@ #include <string> #include "api/neteq/neteq.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -28,6 +27,9 @@ class StatisticsCalculator { virtual ~StatisticsCalculator(); + StatisticsCalculator(const StatisticsCalculator&) = delete; + StatisticsCalculator& operator=(const StatisticsCalculator&) = delete; + // Resets most of the counters. void Reset(); @@ -197,8 +199,6 @@ class StatisticsCalculator { PeriodicUmaAverage excess_buffer_delay_; PeriodicUmaCount buffer_full_counter_; bool decoded_output_played_ = false; - - RTC_DISALLOW_COPY_AND_ASSIGN(StatisticsCalculator); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/sync_buffer.h b/modules/audio_coding/neteq/sync_buffer.h index 7d24730cb3..cf56c432e3 100644 --- a/modules/audio_coding/neteq/sync_buffer.h +++ b/modules/audio_coding/neteq/sync_buffer.h @@ -20,7 +20,6 @@ #include "modules/audio_coding/neteq/audio_multi_vector.h" #include "modules/audio_coding/neteq/audio_vector.h" #include "rtc_base/buffer.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -32,6 +31,9 @@ class SyncBuffer : public AudioMultiVector { end_timestamp_(0), dtmf_index_(0) {} + SyncBuffer(const SyncBuffer&) = delete; + SyncBuffer& operator=(const SyncBuffer&) = delete; + // Returns the number of samples yet to play out from the buffer. size_t FutureLength() const; @@ -102,8 +104,6 @@ class SyncBuffer : public AudioMultiVector { size_t next_index_; uint32_t end_timestamp_; // The timestamp of the last sample in the buffer. size_t dtmf_index_; // Index to the first non-DTMF sample in the buffer. - - RTC_DISALLOW_COPY_AND_ASSIGN(SyncBuffer); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/time_stretch.h b/modules/audio_coding/neteq/time_stretch.h index 998d080714..f0ddaebeca 100644 --- a/modules/audio_coding/neteq/time_stretch.h +++ b/modules/audio_coding/neteq/time_stretch.h @@ -14,7 +14,6 @@ #include <string.h> // memset, size_t #include "modules/audio_coding/neteq/audio_multi_vector.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -49,6 +48,9 @@ class TimeStretch { virtual ~TimeStretch() {} + TimeStretch(const TimeStretch&) = delete; + TimeStretch& operator=(const TimeStretch&) = delete; + // This method performs the processing common to both Accelerate and // PreemptiveExpand. ReturnCodes Process(const int16_t* input, @@ -105,8 +107,6 @@ class TimeStretch { int32_t vec2_energy, size_t peak_index, int scaling) const; - - RTC_DISALLOW_COPY_AND_ASSIGN(TimeStretch); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/timestamp_scaler.h b/modules/audio_coding/neteq/timestamp_scaler.h index 4d578fc433..f42ce7207a 100644 --- a/modules/audio_coding/neteq/timestamp_scaler.h +++ b/modules/audio_coding/neteq/timestamp_scaler.h @@ -12,7 +12,6 @@ #define MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_ #include "modules/audio_coding/neteq/packet.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -34,6 +33,9 @@ class TimestampScaler { virtual ~TimestampScaler() {} + TimestampScaler(const TimestampScaler&) = delete; + TimestampScaler& operator=(const TimestampScaler&) = delete; + // Start over. virtual void Reset(); @@ -59,8 +61,6 @@ class TimestampScaler { uint32_t external_ref_; uint32_t internal_ref_; const DecoderDatabase& decoder_database_; - - RTC_DISALLOW_COPY_AND_ASSIGN(TimestampScaler); }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/tools/audio_checksum.h b/modules/audio_coding/neteq/tools/audio_checksum.h index e4306fa036..9d6f3432c0 100644 --- a/modules/audio_coding/neteq/tools/audio_checksum.h +++ b/modules/audio_coding/neteq/tools/audio_checksum.h @@ -16,7 +16,6 @@ #include "modules/audio_coding/neteq/tools/audio_sink.h" #include "rtc_base/buffer.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/message_digest.h" #include "rtc_base/string_encode.h" #include "rtc_base/system/arch.h" @@ -31,6 +30,9 @@ class AudioChecksum : public AudioSink { checksum_result_(checksum_->Size()), finished_(false) {} + AudioChecksum(const AudioChecksum&) = delete; + AudioChecksum& operator=(const AudioChecksum&) = delete; + bool WriteArray(const int16_t* audio, size_t num_samples) override { if (finished_) return false; @@ -56,8 +58,6 @@ class AudioChecksum : public AudioSink { std::unique_ptr<rtc::MessageDigest> checksum_; rtc::Buffer checksum_result_; bool finished_; - - RTC_DISALLOW_COPY_AND_ASSIGN(AudioChecksum); }; } // namespace test diff --git a/modules/audio_coding/neteq/tools/audio_loop.h b/modules/audio_coding/neteq/tools/audio_loop.h index 25da463921..a73be2dd68 100644 --- a/modules/audio_coding/neteq/tools/audio_loop.h +++ b/modules/audio_coding/neteq/tools/audio_loop.h @@ -15,7 +15,6 @@ #include <string> #include "api/array_view.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { namespace test { @@ -29,6 +28,9 @@ class AudioLoop { virtual ~AudioLoop() {} + AudioLoop(const AudioLoop&) = delete; + AudioLoop& operator=(const AudioLoop&) = delete; + // Initializes the AudioLoop by reading from `file_name`. The loop will be no // longer than `max_loop_length_samples`, if the length of the file is // greater. Otherwise, the loop length is the same as the file length. @@ -47,8 +49,6 @@ class AudioLoop { size_t loop_length_samples_; size_t block_length_samples_; std::unique_ptr<int16_t[]> audio_array_; - - RTC_DISALLOW_COPY_AND_ASSIGN(AudioLoop); }; } // namespace test diff --git a/modules/audio_coding/neteq/tools/audio_sink.h b/modules/audio_coding/neteq/tools/audio_sink.h index cd6733b1d8..53729fa920 100644 --- a/modules/audio_coding/neteq/tools/audio_sink.h +++ b/modules/audio_coding/neteq/tools/audio_sink.h @@ -12,7 +12,6 @@ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_ #include "api/audio/audio_frame.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { namespace test { @@ -24,6 +23,9 @@ class AudioSink { AudioSink() {} virtual ~AudioSink() {} + AudioSink(const AudioSink&) = delete; + AudioSink& operator=(const AudioSink&) = delete; + // Writes `num_samples` from `audio` to the AudioSink. Returns true if // successful, otherwise false. virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0; @@ -34,9 +36,6 @@ class AudioSink { return WriteArray(audio_frame.data(), audio_frame.samples_per_channel_ * audio_frame.num_channels_); } - - private: - RTC_DISALLOW_COPY_AND_ASSIGN(AudioSink); }; // Forks the output audio to two AudioSink objects. @@ -45,23 +44,25 @@ class AudioSinkFork : public AudioSink { AudioSinkFork(AudioSink* left, AudioSink* right) : left_sink_(left), right_sink_(right) {} + AudioSinkFork(const AudioSinkFork&) = delete; + AudioSinkFork& operator=(const AudioSinkFork&) = delete; + bool WriteArray(const int16_t* audio, size_t num_samples) override; private: AudioSink* left_sink_; AudioSink* right_sink_; - - RTC_DISALLOW_COPY_AND_ASSIGN(AudioSinkFork); }; // An AudioSink implementation that does nothing. class VoidAudioSink : public AudioSink { public: VoidAudioSink() = default; - bool WriteArray(const int16_t* audio, size_t num_samples) override; - private: - RTC_DISALLOW_COPY_AND_ASSIGN(VoidAudioSink); + VoidAudioSink(const VoidAudioSink&) = delete; + VoidAudioSink& operator=(const VoidAudioSink&) = delete; + + bool WriteArray(const int16_t* audio, size_t num_samples) override; }; } // namespace test diff --git a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h index 6a79ce4d1f..ab4f5c2281 100644 --- a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h +++ b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h @@ -16,7 +16,6 @@ #include <string> #include "modules/audio_coding/neteq/tools/packet_source.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { namespace test { @@ -31,6 +30,9 @@ class ConstantPcmPacketSource : public PacketSource { int sample_rate_hz, int payload_type); + ConstantPcmPacketSource(const ConstantPcmPacketSource&) = delete; + ConstantPcmPacketSource& operator=(const ConstantPcmPacketSource&) = delete; + std::unique_ptr<Packet> NextPacket() override; private: @@ -46,8 +48,6 @@ class ConstantPcmPacketSource : public PacketSource { uint16_t seq_number_; uint32_t timestamp_; const uint32_t payload_ssrc_; - - RTC_DISALLOW_COPY_AND_ASSIGN(ConstantPcmPacketSource); }; } // namespace test diff --git a/modules/audio_coding/neteq/tools/input_audio_file.h b/modules/audio_coding/neteq/tools/input_audio_file.h index 010d8cccbd..c6e65a0c35 100644 --- a/modules/audio_coding/neteq/tools/input_audio_file.h +++ b/modules/audio_coding/neteq/tools/input_audio_file.h @@ -15,8 +15,6 @@ #include <string> -#include "rtc_base/constructor_magic.h" - namespace webrtc { namespace test { @@ -27,6 +25,9 @@ class InputAudioFile { virtual ~InputAudioFile(); + InputAudioFile(const InputAudioFile&) = delete; + InputAudioFile& operator=(const InputAudioFile&) = delete; + // Reads `samples` elements from source file to `destination`. Returns true // if the read was successful, otherwise false. If the file end is reached, // the file is rewound and reading continues from the beginning. @@ -52,7 +53,6 @@ class InputAudioFile { private: FILE* fp_; const bool loop_at_end_; - RTC_DISALLOW_COPY_AND_ASSIGN(InputAudioFile); }; } // namespace test diff --git a/modules/audio_coding/neteq/tools/output_audio_file.h b/modules/audio_coding/neteq/tools/output_audio_file.h index ad97722cbc..491cbd0420 100644 --- a/modules/audio_coding/neteq/tools/output_audio_file.h +++ b/modules/audio_coding/neteq/tools/output_audio_file.h @@ -16,7 +16,6 @@ #include <string> #include "modules/audio_coding/neteq/tools/audio_sink.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { namespace test { @@ -34,6 +33,9 @@ class OutputAudioFile : public AudioSink { fclose(out_file_); } + OutputAudioFile(const OutputAudioFile&) = delete; + OutputAudioFile& operator=(const OutputAudioFile&) = delete; + bool WriteArray(const int16_t* audio, size_t num_samples) override { RTC_DCHECK(out_file_); return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples; @@ -41,8 +43,6 @@ class OutputAudioFile : public AudioSink { private: FILE* out_file_; - - RTC_DISALLOW_COPY_AND_ASSIGN(OutputAudioFile); }; } // namespace test diff --git a/modules/audio_coding/neteq/tools/output_wav_file.h b/modules/audio_coding/neteq/tools/output_wav_file.h index ae2e9700fe..1485f4e911 100644 --- a/modules/audio_coding/neteq/tools/output_wav_file.h +++ b/modules/audio_coding/neteq/tools/output_wav_file.h @@ -15,7 +15,6 @@ #include "common_audio/wav_file.h" #include "modules/audio_coding/neteq/tools/audio_sink.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { namespace test { @@ -29,6 +28,9 @@ class OutputWavFile : public AudioSink { int num_channels = 1) : wav_writer_(file_name, sample_rate_hz, num_channels) {} + OutputWavFile(const OutputWavFile&) = delete; + OutputWavFile& operator=(const OutputWavFile&) = delete; + bool WriteArray(const int16_t* audio, size_t num_samples) override { wav_writer_.WriteSamples(audio, num_samples); return true; @@ -36,8 +38,6 @@ class OutputWavFile : public AudioSink { private: WavWriter wav_writer_; - - RTC_DISALLOW_COPY_AND_ASSIGN(OutputWavFile); }; } // namespace test diff --git a/modules/audio_coding/neteq/tools/packet.h b/modules/audio_coding/neteq/tools/packet.h index 92e5ee9c3d..96710907df 100644 --- a/modules/audio_coding/neteq/tools/packet.h +++ b/modules/audio_coding/neteq/tools/packet.h @@ -16,7 +16,6 @@ #include "api/array_view.h" #include "api/rtp_headers.h" #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/copy_on_write_buffer.h" namespace webrtc { @@ -54,6 +53,9 @@ class Packet { virtual ~Packet(); + Packet(const Packet&) = delete; + Packet& operator=(const Packet&) = delete; + // Parses the first bytes of the RTP payload, interpreting them as RED headers // according to RFC 2198. The headers will be inserted into `headers`. The // caller of the method assumes ownership of the objects in the list, and @@ -95,8 +97,6 @@ class Packet { size_t virtual_payload_length_bytes_ = 0; const double time_ms_; // Used to denote a packet's arrival time. const bool valid_header_; - - RTC_DISALLOW_COPY_AND_ASSIGN(Packet); }; } // namespace test diff --git a/modules/audio_coding/neteq/tools/packet_source.h b/modules/audio_coding/neteq/tools/packet_source.h index 975680f5a9..be1705cae1 100644 --- a/modules/audio_coding/neteq/tools/packet_source.h +++ b/modules/audio_coding/neteq/tools/packet_source.h @@ -15,7 +15,6 @@ #include <memory> #include "modules/audio_coding/neteq/tools/packet.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { namespace test { @@ -26,6 +25,9 @@ class PacketSource { PacketSource(); virtual ~PacketSource(); + PacketSource(const PacketSource&) = delete; + PacketSource& operator=(const PacketSource&) = delete; + // Returns next packet. Returns nullptr if the source is depleted, or if an // error occurred. virtual std::unique_ptr<Packet> NextPacket() = 0; @@ -34,9 +36,6 @@ class PacketSource { protected: std::bitset<128> filter_; // Payload type is 7 bits in the RFC. - - private: - RTC_DISALLOW_COPY_AND_ASSIGN(PacketSource); }; } // namespace test diff --git a/modules/audio_coding/neteq/tools/resample_input_audio_file.h b/modules/audio_coding/neteq/tools/resample_input_audio_file.h index 9106d5b769..497a4109df 100644 --- a/modules/audio_coding/neteq/tools/resample_input_audio_file.h +++ b/modules/audio_coding/neteq/tools/resample_input_audio_file.h @@ -15,7 +15,6 @@ #include "common_audio/resampler/include/resampler.h" #include "modules/audio_coding/neteq/tools/input_audio_file.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { namespace test { @@ -37,6 +36,9 @@ class ResampleInputAudioFile : public InputAudioFile { file_rate_hz_(file_rate_hz), output_rate_hz_(output_rate_hz) {} + ResampleInputAudioFile(const ResampleInputAudioFile&) = delete; + ResampleInputAudioFile& operator=(const ResampleInputAudioFile&) = delete; + bool Read(size_t samples, int output_rate_hz, int16_t* destination); bool Read(size_t samples, int16_t* destination) override; void set_output_rate_hz(int rate_hz); @@ -45,7 +47,6 @@ class ResampleInputAudioFile : public InputAudioFile { const int file_rate_hz_; int output_rate_hz_; Resampler resampler_; - RTC_DISALLOW_COPY_AND_ASSIGN(ResampleInputAudioFile); }; } // namespace test diff --git a/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/modules/audio_coding/neteq/tools/rtc_event_log_source.h index d4be2a7939..e2d0f61666 100644 --- a/modules/audio_coding/neteq/tools/rtc_event_log_source.h +++ b/modules/audio_coding/neteq/tools/rtc_event_log_source.h @@ -19,7 +19,6 @@ #include "logging/rtc_event_log/rtc_event_log_parser.h" #include "modules/audio_coding/neteq/tools/packet_source.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -43,6 +42,9 @@ class RtcEventLogSource : public PacketSource { virtual ~RtcEventLogSource(); + RtcEventLogSource(const RtcEventLogSource&) = delete; + RtcEventLogSource& operator=(const RtcEventLogSource&) = delete; + std::unique_ptr<Packet> NextPacket() override; // Returns the timestamp of the next audio output event, in milliseconds. The @@ -60,8 +62,6 @@ class RtcEventLogSource : public PacketSource { size_t rtp_packet_index_ = 0; std::vector<int64_t> audio_outputs_; size_t audio_output_index_ = 0; - - RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource); }; } // namespace test diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.h b/modules/audio_coding/neteq/tools/rtp_file_source.h index d6aab24abc..7e284aca45 100644 --- a/modules/audio_coding/neteq/tools/rtp_file_source.h +++ b/modules/audio_coding/neteq/tools/rtp_file_source.h @@ -19,7 +19,6 @@ #include "absl/types/optional.h" #include "modules/audio_coding/neteq/tools/packet_source.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -41,6 +40,9 @@ class RtpFileSource : public PacketSource { ~RtpFileSource() override; + RtpFileSource(const RtpFileSource&) = delete; + RtpFileSource& operator=(const RtpFileSource&) = delete; + // Registers an RTP header extension and binds it to `id`. virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id); @@ -58,8 +60,6 @@ class RtpFileSource : public PacketSource { std::unique_ptr<RtpFileReader> rtp_reader_; const absl::optional<uint32_t> ssrc_filter_; RtpHeaderExtensionMap rtp_header_extension_map_; - - RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource); }; } // namespace test diff --git a/modules/audio_coding/neteq/tools/rtp_generator.h b/modules/audio_coding/neteq/tools/rtp_generator.h index 6ca6e1bac7..2e615adec5 100644 --- a/modules/audio_coding/neteq/tools/rtp_generator.h +++ b/modules/audio_coding/neteq/tools/rtp_generator.h @@ -12,7 +12,6 @@ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_ #include "api/rtp_headers.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { namespace test { @@ -34,6 +33,9 @@ class RtpGenerator { virtual ~RtpGenerator() {} + RtpGenerator(const RtpGenerator&) = delete; + RtpGenerator& operator=(const RtpGenerator&) = delete; + // Writes the next RTP header to `rtp_header`, which will be of type // `payload_type`. Returns the send time for this packet (in ms). The value of // `payload_length_samples` determines the send time for the next packet. @@ -50,9 +52,6 @@ class RtpGenerator { const uint32_t ssrc_; const int samples_per_ms_; double drift_factor_; - - private: - RTC_DISALLOW_COPY_AND_ASSIGN(RtpGenerator); }; class TimestampJumpRtpGenerator : public RtpGenerator { @@ -66,6 +65,10 @@ class TimestampJumpRtpGenerator : public RtpGenerator { jump_from_timestamp_(jump_from_timestamp), jump_to_timestamp_(jump_to_timestamp) {} + TimestampJumpRtpGenerator(const TimestampJumpRtpGenerator&) = delete; + TimestampJumpRtpGenerator& operator=(const TimestampJumpRtpGenerator&) = + delete; + uint32_t GetRtpHeader(uint8_t payload_type, size_t payload_length_samples, RTPHeader* rtp_header) override; @@ -73,7 +76,6 @@ class TimestampJumpRtpGenerator : public RtpGenerator { private: uint32_t jump_from_timestamp_; uint32_t jump_to_timestamp_; - RTC_DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator); }; } // namespace test diff --git a/modules/audio_mixer/audio_mixer_impl.h b/modules/audio_mixer/audio_mixer_impl.h index 737fcbdc43..76b1131777 100644 --- a/modules/audio_mixer/audio_mixer_impl.h +++ b/modules/audio_mixer/audio_mixer_impl.h @@ -22,7 +22,6 @@ #include "api/scoped_refptr.h" #include "modules/audio_mixer/frame_combiner.h" #include "modules/audio_mixer/output_rate_calculator.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/race_checker.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" @@ -48,6 +47,9 @@ class AudioMixerImpl : public AudioMixer { ~AudioMixerImpl() override; + AudioMixerImpl(const AudioMixerImpl&) = delete; + AudioMixerImpl& operator=(const AudioMixerImpl&) = delete; + // AudioMixer functions bool AddSource(Source* audio_source) override; void RemoveSource(Source* audio_source) override; @@ -92,8 +94,6 @@ class AudioMixerImpl : public AudioMixer { // Component that handles actual adding of audio frames. FrameCombiner frame_combiner_; - - RTC_DISALLOW_COPY_AND_ASSIGN(AudioMixerImpl); }; } // namespace webrtc diff --git a/modules/audio_processing/aec3/aec3_fft.h b/modules/audio_processing/aec3/aec3_fft.h index 6f7fbe4d0e..c68de53963 100644 --- a/modules/audio_processing/aec3/aec3_fft.h +++ b/modules/audio_processing/aec3/aec3_fft.h @@ -18,7 +18,6 @@ #include "modules/audio_processing/aec3/aec3_common.h" #include "modules/audio_processing/aec3/fft_data.h" #include "rtc_base/checks.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -30,6 +29,9 @@ class Aec3Fft { Aec3Fft(); + Aec3Fft(const Aec3Fft&) = delete; + Aec3Fft& operator=(const Aec3Fft&) = delete; + // Computes the FFT. Note that both the input and output are modified. void Fft(std::array<float, kFftLength>* x, FftData* X) const { RTC_DCHECK(x); @@ -66,8 +68,6 @@ class Aec3Fft { private: const OouraFft ooura_fft_; - - RTC_DISALLOW_COPY_AND_ASSIGN(Aec3Fft); }; } // namespace webrtc diff --git a/modules/audio_processing/aec3/block_processor_metrics.h b/modules/audio_processing/aec3/block_processor_metrics.h index 4ba053683b..a70d0dac5b 100644 --- a/modules/audio_processing/aec3/block_processor_metrics.h +++ b/modules/audio_processing/aec3/block_processor_metrics.h @@ -11,8 +11,6 @@ #ifndef MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_METRICS_H_ #define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_METRICS_H_ -#include "rtc_base/constructor_magic.h" - namespace webrtc { // Handles the reporting of metrics for the block_processor. @@ -20,6 +18,9 @@ class BlockProcessorMetrics { public: BlockProcessorMetrics() = default; + BlockProcessorMetrics(const BlockProcessorMetrics&) = delete; + BlockProcessorMetrics& operator=(const BlockProcessorMetrics&) = delete; + // Updates the metric with new capture data. void UpdateCapture(bool underrun); @@ -38,8 +39,6 @@ class BlockProcessorMetrics { int render_buffer_underruns_ = 0; int render_buffer_overruns_ = 0; int buffer_render_calls_ = 0; - - RTC_DISALLOW_COPY_AND_ASSIGN(BlockProcessorMetrics); }; } // namespace webrtc diff --git a/modules/audio_processing/aec3/decimator.h b/modules/audio_processing/aec3/decimator.h index 3ccd292f08..dbff3d9fff 100644 --- a/modules/audio_processing/aec3/decimator.h +++ b/modules/audio_processing/aec3/decimator.h @@ -17,7 +17,6 @@ #include "api/array_view.h" #include "modules/audio_processing/aec3/aec3_common.h" #include "modules/audio_processing/utility/cascaded_biquad_filter.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -26,6 +25,9 @@ class Decimator { public: explicit Decimator(size_t down_sampling_factor); + Decimator(const Decimator&) = delete; + Decimator& operator=(const Decimator&) = delete; + // Downsamples the signal. void Decimate(rtc::ArrayView<const float> in, rtc::ArrayView<float> out); @@ -33,8 +35,6 @@ class Decimator { const size_t down_sampling_factor_; CascadedBiQuadFilter anti_aliasing_filter_; CascadedBiQuadFilter noise_reduction_filter_; - - RTC_DISALLOW_COPY_AND_ASSIGN(Decimator); }; } // namespace webrtc diff --git a/modules/audio_processing/aec3/echo_path_delay_estimator.h b/modules/audio_processing/aec3/echo_path_delay_estimator.h index 6c8c21282e..d8f97757bb 100644 --- a/modules/audio_processing/aec3/echo_path_delay_estimator.h +++ b/modules/audio_processing/aec3/echo_path_delay_estimator.h @@ -21,7 +21,6 @@ #include "modules/audio_processing/aec3/delay_estimate.h" #include "modules/audio_processing/aec3/matched_filter.h" #include "modules/audio_processing/aec3/matched_filter_lag_aggregator.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -37,6 +36,9 @@ class EchoPathDelayEstimator { size_t num_capture_channels); ~EchoPathDelayEstimator(); + EchoPathDelayEstimator(const EchoPathDelayEstimator&) = delete; + EchoPathDelayEstimator& operator=(const EchoPathDelayEstimator&) = delete; + // Resets the estimation. If the delay confidence is reset, the reset behavior // is as if the call is restarted. void Reset(bool reset_delay_confidence); @@ -71,8 +73,6 @@ class EchoPathDelayEstimator { // Internal reset method with more granularity. void Reset(bool reset_lag_aggregator, bool reset_delay_confidence); - - RTC_DISALLOW_COPY_AND_ASSIGN(EchoPathDelayEstimator); }; } // namespace webrtc diff --git a/modules/audio_processing/aec3/echo_remover_metrics.h b/modules/audio_processing/aec3/echo_remover_metrics.h index c3d8e20da1..aec8084d78 100644 --- a/modules/audio_processing/aec3/echo_remover_metrics.h +++ b/modules/audio_processing/aec3/echo_remover_metrics.h @@ -15,7 +15,6 @@ #include "modules/audio_processing/aec3/aec3_common.h" #include "modules/audio_processing/aec3/aec_state.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -34,6 +33,9 @@ class EchoRemoverMetrics { EchoRemoverMetrics(); + EchoRemoverMetrics(const EchoRemoverMetrics&) = delete; + EchoRemoverMetrics& operator=(const EchoRemoverMetrics&) = delete; + // Updates the metric with new data. void Update( const AecState& aec_state, @@ -52,8 +54,6 @@ class EchoRemoverMetrics { DbMetric erle_time_domain_; bool saturated_capture_ = false; bool metrics_reported_ = false; - - RTC_DISALLOW_COPY_AND_ASSIGN(EchoRemoverMetrics); }; namespace aec3 { diff --git a/modules/audio_processing/aec3/erl_estimator.h b/modules/audio_processing/aec3/erl_estimator.h index 89bf6ace36..639a52c561 100644 --- a/modules/audio_processing/aec3/erl_estimator.h +++ b/modules/audio_processing/aec3/erl_estimator.h @@ -18,7 +18,6 @@ #include "api/array_view.h" #include "modules/audio_processing/aec3/aec3_common.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -28,6 +27,9 @@ class ErlEstimator { explicit ErlEstimator(size_t startup_phase_length_blocks_); ~ErlEstimator(); + ErlEstimator(const ErlEstimator&) = delete; + ErlEstimator& operator=(const ErlEstimator&) = delete; + // Resets the ERL estimation. void Reset(); @@ -49,7 +51,6 @@ class ErlEstimator { float erl_time_domain_; int hold_counter_time_domain_; size_t blocks_since_reset_ = 0; - RTC_DISALLOW_COPY_AND_ASSIGN(ErlEstimator); }; } // namespace webrtc diff --git a/modules/audio_processing/aec3/render_delay_controller_metrics.h b/modules/audio_processing/aec3/render_delay_controller_metrics.h index 8c527a142e..309122d80d 100644 --- a/modules/audio_processing/aec3/render_delay_controller_metrics.h +++ b/modules/audio_processing/aec3/render_delay_controller_metrics.h @@ -15,7 +15,6 @@ #include "absl/types/optional.h" #include "modules/audio_processing/aec3/clockdrift_detector.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -24,6 +23,10 @@ class RenderDelayControllerMetrics { public: RenderDelayControllerMetrics(); + RenderDelayControllerMetrics(const RenderDelayControllerMetrics&) = delete; + RenderDelayControllerMetrics& operator=(const RenderDelayControllerMetrics&) = + delete; + // Updates the metric with new data. void Update(absl::optional<size_t> delay_samples, size_t buffer_delay_blocks, @@ -46,8 +49,6 @@ class RenderDelayControllerMetrics { bool metrics_reported_ = false; bool initial_update = true; int skew_shift_count_ = 0; - - RTC_DISALLOW_COPY_AND_ASSIGN(RenderDelayControllerMetrics); }; } // namespace webrtc diff --git a/modules/audio_processing/aec3/render_signal_analyzer.h b/modules/audio_processing/aec3/render_signal_analyzer.h index c7a3d8b7a0..2e4aaa4ba7 100644 --- a/modules/audio_processing/aec3/render_signal_analyzer.h +++ b/modules/audio_processing/aec3/render_signal_analyzer.h @@ -20,7 +20,6 @@ #include "modules/audio_processing/aec3/aec3_common.h" #include "modules/audio_processing/aec3/render_buffer.h" #include "rtc_base/checks.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -30,6 +29,9 @@ class RenderSignalAnalyzer { explicit RenderSignalAnalyzer(const EchoCanceller3Config& config); ~RenderSignalAnalyzer(); + RenderSignalAnalyzer(const RenderSignalAnalyzer&) = delete; + RenderSignalAnalyzer& operator=(const RenderSignalAnalyzer&) = delete; + // Updates the render signal analysis with the most recent render signal. void Update(const RenderBuffer& render_buffer, const absl::optional<size_t>& delay_partitions); @@ -53,8 +55,6 @@ class RenderSignalAnalyzer { std::array<size_t, kFftLengthBy2 - 1> narrow_band_counters_; absl::optional<int> narrow_peak_band_; size_t narrow_peak_counter_; - - RTC_DISALLOW_COPY_AND_ASSIGN(RenderSignalAnalyzer); }; } // namespace webrtc diff --git a/modules/audio_processing/aec3/suppression_filter.h b/modules/audio_processing/aec3/suppression_filter.h index dcf2292c7f..375bfda5a7 100644 --- a/modules/audio_processing/aec3/suppression_filter.h +++ b/modules/audio_processing/aec3/suppression_filter.h @@ -17,7 +17,6 @@ #include "modules/audio_processing/aec3/aec3_common.h" #include "modules/audio_processing/aec3/aec3_fft.h" #include "modules/audio_processing/aec3/fft_data.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -27,6 +26,10 @@ class SuppressionFilter { int sample_rate_hz, size_t num_capture_channels_); ~SuppressionFilter(); + + SuppressionFilter(const SuppressionFilter&) = delete; + SuppressionFilter& operator=(const SuppressionFilter&) = delete; + void ApplyGain(rtc::ArrayView<const FftData> comfort_noise, rtc::ArrayView<const FftData> comfort_noise_high_bands, const std::array<float, kFftLengthBy2Plus1>& suppression_gain, @@ -40,7 +43,6 @@ class SuppressionFilter { const size_t num_capture_channels_; const Aec3Fft fft_; std::vector<std::vector<std::array<float, kFftLengthBy2>>> e_output_old_; - RTC_DISALLOW_COPY_AND_ASSIGN(SuppressionFilter); }; } // namespace webrtc diff --git a/modules/audio_processing/aec3/suppression_gain.h b/modules/audio_processing/aec3/suppression_gain.h index 7c4a1c9f7d..c8e13f7cf4 100644 --- a/modules/audio_processing/aec3/suppression_gain.h +++ b/modules/audio_processing/aec3/suppression_gain.h @@ -25,7 +25,6 @@ #include "modules/audio_processing/aec3/nearend_detector.h" #include "modules/audio_processing/aec3/render_signal_analyzer.h" #include "modules/audio_processing/logging/apm_data_dumper.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -36,6 +35,10 @@ class SuppressionGain { int sample_rate_hz, size_t num_capture_channels); ~SuppressionGain(); + + SuppressionGain(const SuppressionGain&) = delete; + SuppressionGain& operator=(const SuppressionGain&) = delete; + void GetGain( rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> nearend_spectrum, @@ -134,8 +137,6 @@ class SuppressionGain { // echo spectrum. const bool use_unbounded_echo_spectrum_; std::unique_ptr<NearendDetector> dominant_nearend_detector_; - - RTC_DISALLOW_COPY_AND_ASSIGN(SuppressionGain); }; } // namespace webrtc diff --git a/modules/audio_processing/agc2/fixed_digital_level_estimator.h b/modules/audio_processing/agc2/fixed_digital_level_estimator.h index d96aedaf9e..d26b55950c 100644 --- a/modules/audio_processing/agc2/fixed_digital_level_estimator.h +++ b/modules/audio_processing/agc2/fixed_digital_level_estimator.h @@ -16,7 +16,6 @@ #include "modules/audio_processing/agc2/agc2_common.h" #include "modules/audio_processing/include/audio_frame_view.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -34,6 +33,10 @@ class FixedDigitalLevelEstimator { FixedDigitalLevelEstimator(int sample_rate_hz, ApmDataDumper* apm_data_dumper); + FixedDigitalLevelEstimator(const FixedDigitalLevelEstimator&) = delete; + FixedDigitalLevelEstimator& operator=(const FixedDigitalLevelEstimator&) = + delete; + // The input is assumed to be in FloatS16 format. Scaled input will // produce similarly scaled output. A frame of with kFrameDurationMs // ms of audio produces a level estimates in the same scale. The @@ -57,8 +60,6 @@ class FixedDigitalLevelEstimator { float filter_state_level_; int samples_in_frame_; int samples_in_sub_frame_; - - RTC_DISALLOW_COPY_AND_ASSIGN(FixedDigitalLevelEstimator); }; } // namespace webrtc diff --git a/modules/audio_processing/agc2/interpolated_gain_curve.h b/modules/audio_processing/agc2/interpolated_gain_curve.h index af993204ce..b1a5cf473b 100644 --- a/modules/audio_processing/agc2/interpolated_gain_curve.h +++ b/modules/audio_processing/agc2/interpolated_gain_curve.h @@ -15,7 +15,6 @@ #include <string> #include "modules/audio_processing/agc2/agc2_common.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/gtest_prod_util.h" #include "system_wrappers/include/metrics.h" @@ -64,6 +63,9 @@ class InterpolatedGainCurve { const std::string& histogram_name_prefix); ~InterpolatedGainCurve(); + InterpolatedGainCurve(const InterpolatedGainCurve&) = delete; + InterpolatedGainCurve& operator=(const InterpolatedGainCurve&) = delete; + Stats get_stats() const { return stats_; } // Given a non-negative input level (linear scale), a scalar factor to apply @@ -143,8 +145,6 @@ class InterpolatedGainCurve { // Stats. mutable Stats stats_; - - RTC_DISALLOW_COPY_AND_ASSIGN(InterpolatedGainCurve); }; } // namespace webrtc diff --git a/modules/audio_processing/echo_control_mobile_impl.cc b/modules/audio_processing/echo_control_mobile_impl.cc index 667d6bfecb..fa5cb8ffec 100644 --- a/modules/audio_processing/echo_control_mobile_impl.cc +++ b/modules/audio_processing/echo_control_mobile_impl.cc @@ -18,7 +18,6 @@ #include "modules/audio_processing/audio_buffer.h" #include "modules/audio_processing/include/audio_processing.h" #include "rtc_base/checks.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -85,6 +84,9 @@ class EchoControlMobileImpl::Canceller { WebRtcAecm_Free(state_); } + Canceller(const Canceller&) = delete; + Canceller& operator=(const Canceller&) = delete; + void* state() { RTC_DCHECK(state_); return state_; @@ -98,7 +100,6 @@ class EchoControlMobileImpl::Canceller { private: void* state_; - RTC_DISALLOW_COPY_AND_ASSIGN(Canceller); }; EchoControlMobileImpl::EchoControlMobileImpl() diff --git a/modules/audio_processing/test/conversational_speech/multiend_call.h b/modules/audio_processing/test/conversational_speech/multiend_call.h index 5b6300f0f1..693f00edd9 100644 --- a/modules/audio_processing/test/conversational_speech/multiend_call.h +++ b/modules/audio_processing/test/conversational_speech/multiend_call.h @@ -24,7 +24,6 @@ #include "modules/audio_processing/test/conversational_speech/timing.h" #include "modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h" #include "modules/audio_processing/test/conversational_speech/wavreader_interface.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { namespace test { @@ -57,6 +56,9 @@ class MultiEndCall { std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory); ~MultiEndCall(); + MultiEndCall(const MultiEndCall&) = delete; + MultiEndCall& operator=(const MultiEndCall&) = delete; + const std::set<std::string>& speaker_names() const { return speaker_names_; } const std::map<std::string, std::unique_ptr<WavReaderInterface>>& audiotrack_readers() const { @@ -92,8 +94,6 @@ class MultiEndCall { int sample_rate_hz_; size_t total_duration_samples_; std::vector<SpeakingTurn> speaking_turns_; - - RTC_DISALLOW_COPY_AND_ASSIGN(MultiEndCall); }; } // namespace conversational_speech diff --git a/modules/audio_processing/test/test_utils.h b/modules/audio_processing/test/test_utils.h index 30674cb143..aa132118fb 100644 --- a/modules/audio_processing/test/test_utils.h +++ b/modules/audio_processing/test/test_utils.h @@ -23,7 +23,6 @@ #include "common_audio/channel_buffer.h" #include "common_audio/wav_file.h" #include "modules/audio_processing/include/audio_processing.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -35,13 +34,14 @@ class RawFile final { explicit RawFile(const std::string& filename); ~RawFile(); + RawFile(const RawFile&) = delete; + RawFile& operator=(const RawFile&) = delete; + void WriteSamples(const int16_t* samples, size_t num_samples); void WriteSamples(const float* samples, size_t num_samples); private: FILE* file_handle_; - - RTC_DISALLOW_COPY_AND_ASSIGN(RawFile); }; // Encapsulates samples and metadata for an integer frame. @@ -78,6 +78,9 @@ class ChannelBufferWavReader final { explicit ChannelBufferWavReader(std::unique_ptr<WavReader> file); ~ChannelBufferWavReader(); + ChannelBufferWavReader(const ChannelBufferWavReader&) = delete; + ChannelBufferWavReader& operator=(const ChannelBufferWavReader&) = delete; + // Reads data from the file according to the `buffer` format. Returns false if // a full buffer can't be read from the file. bool Read(ChannelBuffer<float>* buffer); @@ -85,8 +88,6 @@ class ChannelBufferWavReader final { private: std::unique_ptr<WavReader> file_; std::vector<float> interleaved_; - - RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavReader); }; // Writes ChannelBuffers to a provided WavWriter. @@ -95,13 +96,14 @@ class ChannelBufferWavWriter final { explicit ChannelBufferWavWriter(std::unique_ptr<WavWriter> file); ~ChannelBufferWavWriter(); + ChannelBufferWavWriter(const ChannelBufferWavWriter&) = delete; + ChannelBufferWavWriter& operator=(const ChannelBufferWavWriter&) = delete; + void Write(const ChannelBuffer<float>& buffer); private: std::unique_ptr<WavWriter> file_; std::vector<float> interleaved_; - - RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavWriter); }; // Takes a pointer to a vector. Allows appending the samples of channel buffers diff --git a/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h b/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h index 927cf9b0cb..474d2970df 100644 --- a/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h +++ b/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h @@ -22,7 +22,6 @@ #include "api/transport/network_types.h" #include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h" #include "modules/congestion_controller/goog_cc/delay_based_bwe.h" -#include "rtc_base/constructor_magic.h" #include "system_wrappers/include/clock.h" #include "test/field_trial.h" #include "test/gtest.h" @@ -54,6 +53,9 @@ class RtpStream { RtpStream(int fps, int bitrate_bps); + RtpStream(const RtpStream&) = delete; + RtpStream& operator=(const RtpStream&) = delete; + // Generates a new frame for this stream. If called too soon after the // previous frame, no frame will be generated. The frame is split into // packets. @@ -74,8 +76,6 @@ class RtpStream { int fps_; int bitrate_bps_; int64_t next_rtp_time_; - - RTC_DISALLOW_COPY_AND_ASSIGN(RtpStream); }; class StreamGenerator { @@ -83,6 +83,9 @@ class StreamGenerator { StreamGenerator(int capacity, int64_t time_now); ~StreamGenerator(); + StreamGenerator(const StreamGenerator&) = delete; + StreamGenerator& operator=(const StreamGenerator&) = delete; + // Add a new stream. void AddStream(RtpStream* stream); @@ -108,8 +111,6 @@ class StreamGenerator { int64_t prev_arrival_time_us_; // All streams being transmitted on this simulated channel. std::vector<std::unique_ptr<RtpStream>> streams_; - - RTC_DISALLOW_COPY_AND_ASSIGN(StreamGenerator); }; } // namespace test diff --git a/modules/congestion_controller/goog_cc/delay_increase_detector_interface.h b/modules/congestion_controller/goog_cc/delay_increase_detector_interface.h index eaadb0d124..fc12cff7d5 100644 --- a/modules/congestion_controller/goog_cc/delay_increase_detector_interface.h +++ b/modules/congestion_controller/goog_cc/delay_increase_detector_interface.h @@ -13,7 +13,6 @@ #include <stdint.h> #include "api/network_state_predictor.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -22,6 +21,11 @@ class DelayIncreaseDetectorInterface { DelayIncreaseDetectorInterface() {} virtual ~DelayIncreaseDetectorInterface() {} + DelayIncreaseDetectorInterface(const DelayIncreaseDetectorInterface&) = + delete; + DelayIncreaseDetectorInterface& operator=( + const DelayIncreaseDetectorInterface&) = delete; + // Update the detector with a new sample. The deltas should represent deltas // between timestamp groups as defined by the InterArrival class. virtual void Update(double recv_delta_ms, @@ -32,8 +36,6 @@ class DelayIncreaseDetectorInterface { bool calculated_deltas) = 0; virtual BandwidthUsage State() const = 0; - - RTC_DISALLOW_COPY_AND_ASSIGN(DelayIncreaseDetectorInterface); }; } // namespace webrtc diff --git a/modules/congestion_controller/goog_cc/probe_controller.h b/modules/congestion_controller/goog_cc/probe_controller.h index 7f24ff98c8..d0f1458ece 100644 --- a/modules/congestion_controller/goog_cc/probe_controller.h +++ b/modules/congestion_controller/goog_cc/probe_controller.h @@ -22,7 +22,6 @@ #include "api/transport/network_control.h" #include "api/transport/webrtc_key_value_config.h" #include "api/units/data_rate.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/experiments/field_trial_parser.h" namespace webrtc { @@ -63,6 +62,9 @@ class ProbeController { RtcEventLog* event_log); ~ProbeController(); + ProbeController(const ProbeController&) = delete; + ProbeController& operator=(const ProbeController&) = delete; + ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> SetBitrates( int64_t min_bitrate_bps, int64_t start_bitrate_bps, @@ -143,8 +145,6 @@ class ProbeController { int32_t next_probe_cluster_id_ = 1; ProbeControllerConfig config_; - - RTC_DISALLOW_COPY_AND_ASSIGN(ProbeController); }; } // namespace webrtc diff --git a/modules/congestion_controller/goog_cc/trendline_estimator.h b/modules/congestion_controller/goog_cc/trendline_estimator.h index 75b971d187..6fd442498b 100644 --- a/modules/congestion_controller/goog_cc/trendline_estimator.h +++ b/modules/congestion_controller/goog_cc/trendline_estimator.h @@ -20,7 +20,6 @@ #include "api/network_state_predictor.h" #include "api/transport/webrtc_key_value_config.h" #include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/experiments/struct_parameters_parser.h" namespace webrtc { @@ -57,6 +56,9 @@ class TrendlineEstimator : public DelayIncreaseDetectorInterface { ~TrendlineEstimator() override; + TrendlineEstimator(const TrendlineEstimator&) = delete; + TrendlineEstimator& operator=(const TrendlineEstimator&) = delete; + // Update the estimator with a new sample. The deltas should represent deltas // between timestamp groups as defined by the InterArrival class. void Update(double recv_delta_ms, @@ -118,8 +120,6 @@ class TrendlineEstimator : public DelayIncreaseDetectorInterface { BandwidthUsage hypothesis_; BandwidthUsage hypothesis_predicted_; NetworkStatePredictor* network_state_predictor_; - - RTC_DISALLOW_COPY_AND_ASSIGN(TrendlineEstimator); }; } // namespace webrtc diff --git a/modules/congestion_controller/rtp/control_handler.h b/modules/congestion_controller/rtp/control_handler.h index 1da6463219..16ffc32a44 100644 --- a/modules/congestion_controller/rtp/control_handler.h +++ b/modules/congestion_controller/rtp/control_handler.h @@ -19,7 +19,6 @@ #include "api/units/data_size.h" #include "api/units/time_delta.h" #include "modules/pacing/paced_sender.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/system/no_unique_address.h" namespace webrtc { @@ -33,6 +32,9 @@ class CongestionControlHandler { CongestionControlHandler(); ~CongestionControlHandler(); + CongestionControlHandler(const CongestionControlHandler&) = delete; + CongestionControlHandler& operator=(const CongestionControlHandler&) = delete; + void SetTargetRate(TargetTransferRate new_target_rate); void SetNetworkAvailability(bool network_available); void SetPacerQueue(TimeDelta expected_queue_time); @@ -48,7 +50,6 @@ class CongestionControlHandler { int64_t pacer_expected_queue_ms_ = 0; RTC_NO_UNIQUE_ADDRESS SequenceChecker sequenced_checker_; - RTC_DISALLOW_COPY_AND_ASSIGN(CongestionControlHandler); }; } // namespace webrtc #endif // MODULES_CONGESTION_CONTROLLER_RTP_CONTROL_HANDLER_H_ diff --git a/modules/desktop_capture/cropped_desktop_frame.cc b/modules/desktop_capture/cropped_desktop_frame.cc index 3c5c9949b7..54488b7d62 100644 --- a/modules/desktop_capture/cropped_desktop_frame.cc +++ b/modules/desktop_capture/cropped_desktop_frame.cc @@ -15,7 +15,6 @@ #include "modules/desktop_capture/desktop_region.h" #include "rtc_base/checks.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -25,10 +24,11 @@ class CroppedDesktopFrame : public DesktopFrame { CroppedDesktopFrame(std::unique_ptr<DesktopFrame> frame, const DesktopRect& rect); + CroppedDesktopFrame(const CroppedDesktopFrame&) = delete; + CroppedDesktopFrame& operator=(const CroppedDesktopFrame&) = delete; + private: const std::unique_ptr<DesktopFrame> frame_; - - RTC_DISALLOW_COPY_AND_ASSIGN(CroppedDesktopFrame); }; std::unique_ptr<DesktopFrame> CreateCroppedDesktopFrame( diff --git a/modules/desktop_capture/desktop_and_cursor_composer.cc b/modules/desktop_capture/desktop_and_cursor_composer.cc index 7be8982abe..7ca0af038c 100644 --- a/modules/desktop_capture/desktop_and_cursor_composer.cc +++ b/modules/desktop_capture/desktop_and_cursor_composer.cc @@ -21,7 +21,6 @@ #include "modules/desktop_capture/mouse_cursor.h" #include "modules/desktop_capture/mouse_cursor_monitor.h" #include "rtc_base/checks.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -72,6 +71,9 @@ class DesktopFrameWithCursor : public DesktopFrame { bool cursor_changed); ~DesktopFrameWithCursor() override; + DesktopFrameWithCursor(const DesktopFrameWithCursor&) = delete; + DesktopFrameWithCursor& operator=(const DesktopFrameWithCursor&) = delete; + DesktopRect cursor_rect() const { return cursor_rect_; } private: @@ -80,8 +82,6 @@ class DesktopFrameWithCursor : public DesktopFrame { DesktopVector restore_position_; std::unique_ptr<DesktopFrame> restore_frame_; DesktopRect cursor_rect_; - - RTC_DISALLOW_COPY_AND_ASSIGN(DesktopFrameWithCursor); }; DesktopFrameWithCursor::DesktopFrameWithCursor( diff --git a/modules/desktop_capture/desktop_and_cursor_composer.h b/modules/desktop_capture/desktop_and_cursor_composer.h index a02705810c..edb764d168 100644 --- a/modules/desktop_capture/desktop_and_cursor_composer.h +++ b/modules/desktop_capture/desktop_and_cursor_composer.h @@ -21,7 +21,6 @@ #include "modules/desktop_capture/mouse_cursor.h" #include "modules/desktop_capture/mouse_cursor_monitor.h" #include "modules/desktop_capture/shared_memory.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/system/rtc_export.h" namespace webrtc { @@ -41,6 +40,9 @@ class RTC_EXPORT DesktopAndCursorComposer ~DesktopAndCursorComposer() override; + DesktopAndCursorComposer(const DesktopAndCursorComposer&) = delete; + DesktopAndCursorComposer& operator=(const DesktopAndCursorComposer&) = delete; + // Creates a new composer that relies on an external source for cursor shape // and position information via the MouseCursorMonitor::Callback interface. static std::unique_ptr<DesktopAndCursorComposer> @@ -84,8 +86,6 @@ class RTC_EXPORT DesktopAndCursorComposer DesktopVector cursor_position_; DesktopRect previous_cursor_rect_; bool cursor_changed_ = false; - - RTC_DISALLOW_COPY_AND_ASSIGN(DesktopAndCursorComposer); }; } // namespace webrtc diff --git a/modules/desktop_capture/desktop_frame.h b/modules/desktop_capture/desktop_frame.h index 9c53b836cb..3ee1867e70 100644 --- a/modules/desktop_capture/desktop_frame.h +++ b/modules/desktop_capture/desktop_frame.h @@ -19,7 +19,6 @@ #include "modules/desktop_capture/desktop_geometry.h" #include "modules/desktop_capture/desktop_region.h" #include "modules/desktop_capture/shared_memory.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/system/rtc_export.h" namespace webrtc { @@ -34,6 +33,9 @@ class RTC_EXPORT DesktopFrame { virtual ~DesktopFrame(); + DesktopFrame(const DesktopFrame&) = delete; + DesktopFrame& operator=(const DesktopFrame&) = delete; + // Returns the rectangle in full desktop coordinates to indicate it covers // the area of top_left() to top_letf() + size() / scale_factor(). DesktopRect rect() const; @@ -163,8 +165,6 @@ class RTC_EXPORT DesktopFrame { int64_t capture_time_ms_; uint32_t capturer_id_; std::vector<uint8_t> icc_profile_; - - RTC_DISALLOW_COPY_AND_ASSIGN(DesktopFrame); }; // A DesktopFrame that stores data in the heap. @@ -175,12 +175,12 @@ class RTC_EXPORT BasicDesktopFrame : public DesktopFrame { ~BasicDesktopFrame() override; + BasicDesktopFrame(const BasicDesktopFrame&) = delete; + BasicDesktopFrame& operator=(const BasicDesktopFrame&) = delete; + // Creates a BasicDesktopFrame that contains copy of `frame`. // TODO(zijiehe): Return std::unique_ptr<DesktopFrame> static DesktopFrame* CopyOf(const DesktopFrame& frame); - - private: - RTC_DISALLOW_COPY_AND_ASSIGN(BasicDesktopFrame); }; // A DesktopFrame that stores data in shared memory. @@ -206,6 +206,9 @@ class RTC_EXPORT SharedMemoryDesktopFrame : public DesktopFrame { ~SharedMemoryDesktopFrame() override; + SharedMemoryDesktopFrame(const SharedMemoryDesktopFrame&) = delete; + SharedMemoryDesktopFrame& operator=(const SharedMemoryDesktopFrame&) = delete; + private: // Avoid unexpected order of parameter evaluation. // Executing both std::unique_ptr<T>::operator->() and @@ -217,8 +220,6 @@ class RTC_EXPORT SharedMemoryDesktopFrame : public DesktopFrame { SharedMemoryDesktopFrame(DesktopRect rect, int stride, SharedMemory* shared_memory); - - RTC_DISALLOW_COPY_AND_ASSIGN(SharedMemoryDesktopFrame); }; } // namespace webrtc diff --git a/modules/desktop_capture/full_screen_application_handler.h b/modules/desktop_capture/full_screen_application_handler.h index 849cb2c761..b7e097a474 100644 --- a/modules/desktop_capture/full_screen_application_handler.h +++ b/modules/desktop_capture/full_screen_application_handler.h @@ -12,8 +12,8 @@ #define MODULES_DESKTOP_CAPTURE_FULL_SCREEN_APPLICATION_HANDLER_H_ #include <memory> + #include "modules/desktop_capture/desktop_capturer.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -25,6 +25,10 @@ class FullScreenApplicationHandler { public: virtual ~FullScreenApplicationHandler() {} + FullScreenApplicationHandler(const FullScreenApplicationHandler&) = delete; + FullScreenApplicationHandler& operator=(const FullScreenApplicationHandler&) = + delete; + explicit FullScreenApplicationHandler(DesktopCapturer::SourceId sourceId); // Returns the full-screen window in place of the original window if all the @@ -39,8 +43,6 @@ class FullScreenApplicationHandler { private: const DesktopCapturer::SourceId source_id_; - - RTC_DISALLOW_COPY_AND_ASSIGN(FullScreenApplicationHandler); }; } // namespace webrtc diff --git a/modules/desktop_capture/full_screen_window_detector.h b/modules/desktop_capture/full_screen_window_detector.h index ca30d95de4..998b720d90 100644 --- a/modules/desktop_capture/full_screen_window_detector.h +++ b/modules/desktop_capture/full_screen_window_detector.h @@ -12,12 +12,12 @@ #define MODULES_DESKTOP_CAPTURE_FULL_SCREEN_WINDOW_DETECTOR_H_ #include <memory> + #include "api/function_view.h" #include "api/ref_counted_base.h" #include "api/scoped_refptr.h" #include "modules/desktop_capture/desktop_capturer.h" #include "modules/desktop_capture/full_screen_application_handler.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -42,6 +42,9 @@ class FullScreenWindowDetector FullScreenWindowDetector( ApplicationHandlerFactory application_handler_factory); + FullScreenWindowDetector(const FullScreenWindowDetector&) = delete; + FullScreenWindowDetector& operator=(const FullScreenWindowDetector&) = delete; + // Returns the full-screen window in place of the original window if all the // criteria provided by FullScreenApplicationHandler are met, or 0 if no such // window found. @@ -73,7 +76,6 @@ class FullScreenWindowDetector DesktopCapturer::SourceId no_handler_source_id_; DesktopCapturer::SourceList window_list_; - RTC_DISALLOW_COPY_AND_ASSIGN(FullScreenWindowDetector); }; } // namespace webrtc diff --git a/modules/desktop_capture/mac/desktop_configuration_monitor.h b/modules/desktop_capture/mac/desktop_configuration_monitor.h index aa0ebfbacc..747295a538 100644 --- a/modules/desktop_capture/mac/desktop_configuration_monitor.h +++ b/modules/desktop_capture/mac/desktop_configuration_monitor.h @@ -18,7 +18,6 @@ #include "api/ref_counted_base.h" #include "modules/desktop_capture/mac/desktop_configuration.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/synchronization/mutex.h" namespace webrtc { @@ -31,6 +30,10 @@ class DesktopConfigurationMonitor final DesktopConfigurationMonitor(); ~DesktopConfigurationMonitor(); + DesktopConfigurationMonitor(const DesktopConfigurationMonitor&) = delete; + DesktopConfigurationMonitor& operator=(const DesktopConfigurationMonitor&) = + delete; + // Returns the current desktop configuration. MacDesktopConfiguration desktop_configuration(); @@ -45,8 +48,6 @@ class DesktopConfigurationMonitor final MacDesktopConfiguration desktop_configuration_ RTC_GUARDED_BY(&desktop_configuration_lock_); std::set<CGDirectDisplayID> reconfiguring_displays_; - - RTC_DISALLOW_COPY_AND_ASSIGN(DesktopConfigurationMonitor); }; } // namespace webrtc diff --git a/modules/desktop_capture/mac/desktop_frame_cgimage.h b/modules/desktop_capture/mac/desktop_frame_cgimage.h index 87804382f2..d6279f9b36 100644 --- a/modules/desktop_capture/mac/desktop_frame_cgimage.h +++ b/modules/desktop_capture/mac/desktop_frame_cgimage.h @@ -35,6 +35,9 @@ class DesktopFrameCGImage final : public DesktopFrame { ~DesktopFrameCGImage() override; + DesktopFrameCGImage(const DesktopFrameCGImage&) = delete; + DesktopFrameCGImage& operator=(const DesktopFrameCGImage&) = delete; + private: static std::unique_ptr<DesktopFrameCGImage> CreateFromCGImage( rtc::ScopedCFTypeRef<CGImageRef> cg_image); @@ -48,8 +51,6 @@ class DesktopFrameCGImage final : public DesktopFrame { const rtc::ScopedCFTypeRef<CGImageRef> cg_image_; const rtc::ScopedCFTypeRef<CFDataRef> cg_data_; - - RTC_DISALLOW_COPY_AND_ASSIGN(DesktopFrameCGImage); }; } // namespace webrtc diff --git a/modules/desktop_capture/mac/desktop_frame_iosurface.h b/modules/desktop_capture/mac/desktop_frame_iosurface.h index f79197aa90..73da0f693c 100644 --- a/modules/desktop_capture/mac/desktop_frame_iosurface.h +++ b/modules/desktop_capture/mac/desktop_frame_iosurface.h @@ -30,13 +30,14 @@ class DesktopFrameIOSurface final : public DesktopFrame { ~DesktopFrameIOSurface() override; + DesktopFrameIOSurface(const DesktopFrameIOSurface&) = delete; + DesktopFrameIOSurface& operator=(const DesktopFrameIOSurface&) = delete; + private: // This constructor expects `io_surface` to hold a non-null IOSurfaceRef. explicit DesktopFrameIOSurface(rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface); const rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface_; - - RTC_DISALLOW_COPY_AND_ASSIGN(DesktopFrameIOSurface); }; } // namespace webrtc diff --git a/modules/desktop_capture/mac/desktop_frame_provider.h b/modules/desktop_capture/mac/desktop_frame_provider.h index bb45d1ed54..aad28d2f30 100644 --- a/modules/desktop_capture/mac/desktop_frame_provider.h +++ b/modules/desktop_capture/mac/desktop_frame_provider.h @@ -28,6 +28,9 @@ class DesktopFrameProvider { explicit DesktopFrameProvider(bool allow_iosurface); ~DesktopFrameProvider(); + DesktopFrameProvider(const DesktopFrameProvider&) = delete; + DesktopFrameProvider& operator=(const DesktopFrameProvider&) = delete; + // The caller takes ownership of the returned desktop frame. Otherwise // returns null if `display_id` is invalid or not ready. Note that this // function does not remove the frame from the internal container. Caller @@ -49,8 +52,6 @@ class DesktopFrameProvider { // Most recent IOSurface that contains a capture of matching display. std::map<CGDirectDisplayID, std::unique_ptr<SharedDesktopFrame>> io_surfaces_; - - RTC_DISALLOW_COPY_AND_ASSIGN(DesktopFrameProvider); }; } // namespace webrtc diff --git a/modules/desktop_capture/mac/screen_capturer_mac.h b/modules/desktop_capture/mac/screen_capturer_mac.h index 68b8655b1c..d9a5966efa 100644 --- a/modules/desktop_capture/mac/screen_capturer_mac.h +++ b/modules/desktop_capture/mac/screen_capturer_mac.h @@ -42,6 +42,9 @@ class ScreenCapturerMac final : public DesktopCapturer { bool allow_iosurface); ~ScreenCapturerMac() override; + ScreenCapturerMac(const ScreenCapturerMac&) = delete; + ScreenCapturerMac& operator=(const ScreenCapturerMac&) = delete; + // TODO(julien.isorce): Remove Init() or make it private. bool Init(); @@ -111,8 +114,6 @@ class ScreenCapturerMac final : public DesktopCapturer { // Start, CaptureFrame and destructor have to called in the same thread. SequenceChecker thread_checker_; - - RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCapturerMac); }; } // namespace webrtc diff --git a/modules/desktop_capture/mock_desktop_capturer_callback.h b/modules/desktop_capture/mock_desktop_capturer_callback.h index 6530dc5542..bb15ceaf4a 100644 --- a/modules/desktop_capture/mock_desktop_capturer_callback.h +++ b/modules/desktop_capture/mock_desktop_capturer_callback.h @@ -22,15 +22,16 @@ class MockDesktopCapturerCallback : public DesktopCapturer::Callback { MockDesktopCapturerCallback(); ~MockDesktopCapturerCallback() override; + MockDesktopCapturerCallback(const MockDesktopCapturerCallback&) = delete; + MockDesktopCapturerCallback& operator=(const MockDesktopCapturerCallback&) = + delete; + MOCK_METHOD(void, OnCaptureResultPtr, (DesktopCapturer::Result result, std::unique_ptr<DesktopFrame>* frame)); void OnCaptureResult(DesktopCapturer::Result result, std::unique_ptr<DesktopFrame> frame) final; - - private: - RTC_DISALLOW_COPY_AND_ASSIGN(MockDesktopCapturerCallback); }; } // namespace webrtc diff --git a/modules/desktop_capture/mouse_cursor.h b/modules/desktop_capture/mouse_cursor.h index 0a74140b4b..2dd793179b 100644 --- a/modules/desktop_capture/mouse_cursor.h +++ b/modules/desktop_capture/mouse_cursor.h @@ -15,7 +15,6 @@ #include "modules/desktop_capture/desktop_frame.h" #include "modules/desktop_capture/desktop_geometry.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/system/rtc_export.h" namespace webrtc { @@ -29,6 +28,9 @@ class RTC_EXPORT MouseCursor { ~MouseCursor(); + MouseCursor(const MouseCursor&) = delete; + MouseCursor& operator=(const MouseCursor&) = delete; + static MouseCursor* CopyOf(const MouseCursor& cursor); void set_image(DesktopFrame* image) { image_.reset(image); } @@ -40,8 +42,6 @@ class RTC_EXPORT MouseCursor { private: std::unique_ptr<DesktopFrame> image_; DesktopVector hotspot_; - - RTC_DISALLOW_COPY_AND_ASSIGN(MouseCursor); }; } // namespace webrtc diff --git a/modules/desktop_capture/screen_capture_frame_queue.h b/modules/desktop_capture/screen_capture_frame_queue.h index a92bac68d8..f54b8b606a 100644 --- a/modules/desktop_capture/screen_capture_frame_queue.h +++ b/modules/desktop_capture/screen_capture_frame_queue.h @@ -13,7 +13,6 @@ #include <memory> -#include "rtc_base/constructor_magic.h" // TODO(zijiehe): These headers are not used in this file, but to avoid build // break in remoting/host. We should add headers in each individual files. #include "modules/desktop_capture/desktop_frame.h" // Remove @@ -40,6 +39,9 @@ class ScreenCaptureFrameQueue { ScreenCaptureFrameQueue() : current_(0) {} ~ScreenCaptureFrameQueue() = default; + ScreenCaptureFrameQueue(const ScreenCaptureFrameQueue&) = delete; + ScreenCaptureFrameQueue& operator=(const ScreenCaptureFrameQueue&) = delete; + // Moves to the next frame in the queue, moving the 'current' frame to become // the 'previous' one. void MoveToNextFrame() { current_ = (current_ + 1) % kQueueLength; } @@ -71,8 +73,6 @@ class ScreenCaptureFrameQueue { static const int kQueueLength = 2; std::unique_ptr<FrameType> frames_[kQueueLength]; - - RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCaptureFrameQueue); }; } // namespace webrtc diff --git a/modules/desktop_capture/screen_capturer_helper.h b/modules/desktop_capture/screen_capturer_helper.h index f7c447aac7..cd7fa689c0 100644 --- a/modules/desktop_capture/screen_capturer_helper.h +++ b/modules/desktop_capture/screen_capturer_helper.h @@ -15,7 +15,6 @@ #include "modules/desktop_capture/desktop_geometry.h" #include "modules/desktop_capture/desktop_region.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" @@ -30,6 +29,9 @@ class ScreenCapturerHelper { ScreenCapturerHelper() = default; ~ScreenCapturerHelper() = default; + ScreenCapturerHelper(const ScreenCapturerHelper&) = delete; + ScreenCapturerHelper& operator=(const ScreenCapturerHelper&) = delete; + // Clear out the invalid region. void ClearInvalidRegion(); @@ -82,8 +84,6 @@ class ScreenCapturerHelper { // expanded. // If the value is <= 0, then the invalid region is not expanded to a grid. int log_grid_size_ = 0; - - RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCapturerHelper); }; } // namespace webrtc diff --git a/modules/desktop_capture/screen_capturer_unittest.cc b/modules/desktop_capture/screen_capturer_unittest.cc index ba6b8bfe3d..8f5fe631f1 100644 --- a/modules/desktop_capture/screen_capturer_unittest.cc +++ b/modules/desktop_capture/screen_capturer_unittest.cc @@ -15,7 +15,6 @@ #include "modules/desktop_capture/desktop_frame.h" #include "modules/desktop_capture/desktop_region.h" #include "modules/desktop_capture/mock_desktop_capturer_callback.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/logging.h" #include "test/gmock.h" #include "test/gtest.h" @@ -76,9 +75,11 @@ class FakeSharedMemory : public SharedMemory { : SharedMemory(buffer, size, 0, kTestSharedMemoryId), buffer_(buffer) {} ~FakeSharedMemory() override { delete[] buffer_; } + FakeSharedMemory(const FakeSharedMemory&) = delete; + FakeSharedMemory& operator=(const FakeSharedMemory&) = delete; + private: char* buffer_; - RTC_DISALLOW_COPY_AND_ASSIGN(FakeSharedMemory); }; class FakeSharedMemoryFactory : public SharedMemoryFactory { @@ -86,13 +87,13 @@ class FakeSharedMemoryFactory : public SharedMemoryFactory { FakeSharedMemoryFactory() {} ~FakeSharedMemoryFactory() override {} + FakeSharedMemoryFactory(const FakeSharedMemoryFactory&) = delete; + FakeSharedMemoryFactory& operator=(const FakeSharedMemoryFactory&) = delete; + std::unique_ptr<SharedMemory> CreateSharedMemory(size_t size) override { return std::unique_ptr<SharedMemory>( new FakeSharedMemory(new char[size], size)); } - - private: - RTC_DISALLOW_COPY_AND_ASSIGN(FakeSharedMemoryFactory); }; ACTION_P(SaveUniquePtrArg, dest) { diff --git a/modules/desktop_capture/shared_desktop_frame.h b/modules/desktop_capture/shared_desktop_frame.h index 29f9306b39..c6f52247f4 100644 --- a/modules/desktop_capture/shared_desktop_frame.h +++ b/modules/desktop_capture/shared_desktop_frame.h @@ -15,7 +15,6 @@ #include "api/scoped_refptr.h" #include "modules/desktop_capture/desktop_frame.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/ref_counted_object.h" #include "rtc_base/system/rtc_export.h" @@ -27,6 +26,9 @@ class RTC_EXPORT SharedDesktopFrame final : public DesktopFrame { public: ~SharedDesktopFrame() override; + SharedDesktopFrame(const SharedDesktopFrame&) = delete; + SharedDesktopFrame& operator=(const SharedDesktopFrame&) = delete; + static std::unique_ptr<SharedDesktopFrame> Wrap( std::unique_ptr<DesktopFrame> desktop_frame); @@ -56,8 +58,6 @@ class RTC_EXPORT SharedDesktopFrame final : public DesktopFrame { SharedDesktopFrame(rtc::scoped_refptr<Core> core); const rtc::scoped_refptr<Core> core_; - - RTC_DISALLOW_COPY_AND_ASSIGN(SharedDesktopFrame); }; } // namespace webrtc diff --git a/modules/desktop_capture/shared_memory.h b/modules/desktop_capture/shared_memory.h index 49a9252c1b..a7add4447b 100644 --- a/modules/desktop_capture/shared_memory.h +++ b/modules/desktop_capture/shared_memory.h @@ -21,7 +21,6 @@ typedef void* HANDLE; #include <memory> -#include "rtc_base/constructor_magic.h" #include "rtc_base/system/rtc_export.h" namespace webrtc { @@ -54,6 +53,9 @@ class RTC_EXPORT SharedMemory { virtual ~SharedMemory() {} + SharedMemory(const SharedMemory&) = delete; + SharedMemory& operator=(const SharedMemory&) = delete; + protected: SharedMemory(void* data, size_t size, Handle handle, int id); @@ -61,9 +63,6 @@ class RTC_EXPORT SharedMemory { const size_t size_; const Handle handle_; const int id_; - - private: - RTC_DISALLOW_COPY_AND_ASSIGN(SharedMemory); }; // Interface used to create SharedMemory instances. @@ -72,10 +71,10 @@ class SharedMemoryFactory { SharedMemoryFactory() {} virtual ~SharedMemoryFactory() {} - virtual std::unique_ptr<SharedMemory> CreateSharedMemory(size_t size) = 0; + SharedMemoryFactory(const SharedMemoryFactory&) = delete; + SharedMemoryFactory& operator=(const SharedMemoryFactory&) = delete; - private: - RTC_DISALLOW_COPY_AND_ASSIGN(SharedMemoryFactory); + virtual std::unique_ptr<SharedMemory> CreateSharedMemory(size_t size) = 0; }; } // namespace webrtc diff --git a/modules/desktop_capture/win/screen_capturer_win_directx.h b/modules/desktop_capture/win/screen_capturer_win_directx.h index 378f9a7ea8..2d0fce6f90 100644 --- a/modules/desktop_capture/win/screen_capturer_win_directx.h +++ b/modules/desktop_capture/win/screen_capturer_win_directx.h @@ -23,6 +23,7 @@ #include "modules/desktop_capture/screen_capture_frame_queue.h" #include "modules/desktop_capture/win/dxgi_duplicator_controller.h" #include "modules/desktop_capture/win/dxgi_frame.h" +#include "rtc_base/constructor_magic.h" #include "rtc_base/system/rtc_export.h" namespace webrtc { diff --git a/modules/desktop_capture/window_capturer_mac.mm b/modules/desktop_capture/window_capturer_mac.mm index de15d65ac5..f0b413b0a6 100644 --- a/modules/desktop_capture/window_capturer_mac.mm +++ b/modules/desktop_capture/window_capturer_mac.mm @@ -24,7 +24,6 @@ #include "modules/desktop_capture/mac/window_list_utils.h" #include "modules/desktop_capture/window_finder_mac.h" #include "rtc_base/checks.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/logging.h" #include "rtc_base/trace_event.h" @@ -52,6 +51,9 @@ class WindowCapturerMac : public DesktopCapturer { rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor); ~WindowCapturerMac() override; + WindowCapturerMac(const WindowCapturerMac&) = delete; + WindowCapturerMac& operator=(const WindowCapturerMac&) = delete; + // DesktopCapturer interface. void Start(Callback* callback) override; void CaptureFrame() override; @@ -71,8 +73,6 @@ class WindowCapturerMac : public DesktopCapturer { const rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor_; WindowFinderMac window_finder_; - - RTC_DISALLOW_COPY_AND_ASSIGN(WindowCapturerMac); }; WindowCapturerMac::WindowCapturerMac( diff --git a/modules/pacing/packet_router.h b/modules/pacing/packet_router.h index 9958a50b6e..11d8979052 100644 --- a/modules/pacing/packet_router.h +++ b/modules/pacing/packet_router.h @@ -25,7 +25,6 @@ #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtcp_packet.h" #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" @@ -44,6 +43,9 @@ class PacketRouter : public PacingController::PacketSender { explicit PacketRouter(uint16_t start_transport_seq); ~PacketRouter() override; + PacketRouter(const PacketRouter&) = delete; + PacketRouter& operator=(const PacketRouter&) = delete; + void AddSendRtpModule(RtpRtcpInterface* rtp_module, bool remb_candidate); void RemoveSendRtpModule(RtpRtcpInterface* rtp_module); @@ -107,8 +109,6 @@ class PacketRouter : public PacingController::PacketSender { // process thread is gone. std::vector<std::unique_ptr<RtpPacketToSend>> pending_fec_packets_ RTC_GUARDED_BY(modules_mutex_); - - RTC_DISALLOW_COPY_AND_ASSIGN(PacketRouter); }; } // namespace webrtc #endif // MODULES_PACING_PACKET_ROUTER_H_ diff --git a/modules/remote_bitrate_estimator/overuse_detector.h b/modules/remote_bitrate_estimator/overuse_detector.h index d1c6aa8d30..179e290c21 100644 --- a/modules/remote_bitrate_estimator/overuse_detector.h +++ b/modules/remote_bitrate_estimator/overuse_detector.h @@ -14,7 +14,6 @@ #include "api/network_state_predictor.h" #include "api/transport/webrtc_key_value_config.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -26,6 +25,9 @@ class OveruseDetector { explicit OveruseDetector(const WebRtcKeyValueConfig* key_value_config); virtual ~OveruseDetector(); + OveruseDetector(const OveruseDetector&) = delete; + OveruseDetector& operator=(const OveruseDetector&) = delete; + // Update the detection state based on the estimated inter-arrival time delta // offset. `timestamp_delta` is the delta between the last timestamp which the // estimated offset is based on and the last timestamp on which the last @@ -54,8 +56,6 @@ class OveruseDetector { double time_over_using_; int overuse_counter_; BandwidthUsage hypothesis_; - - RTC_DISALLOW_COPY_AND_ASSIGN(OveruseDetector); }; } // namespace webrtc diff --git a/modules/remote_bitrate_estimator/overuse_estimator.h b/modules/remote_bitrate_estimator/overuse_estimator.h index d023b36d89..c021f00da7 100644 --- a/modules/remote_bitrate_estimator/overuse_estimator.h +++ b/modules/remote_bitrate_estimator/overuse_estimator.h @@ -15,7 +15,6 @@ #include <deque> #include "api/network_state_predictor.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -39,6 +38,9 @@ class OveruseEstimator { explicit OveruseEstimator(const OverUseDetectorOptions& options); ~OveruseEstimator(); + OveruseEstimator(const OveruseEstimator&) = delete; + OveruseEstimator& operator=(const OveruseEstimator&) = delete; + // Update the estimator with a new sample. The deltas should represent deltas // between timestamp groups as defined by the InterArrival class. // `current_hypothesis` should be the hypothesis of the over-use detector at @@ -75,8 +77,6 @@ class OveruseEstimator { double avg_noise_; double var_noise_; std::deque<double> ts_delta_hist_; - - RTC_DISALLOW_COPY_AND_ASSIGN(OveruseEstimator); }; } // namespace webrtc diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc b/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc index 10dd99cfd9..d8ef23cc92 100644 --- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc +++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc @@ -11,7 +11,6 @@ #include "modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h" #include "modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h" -#include "rtc_base/constructor_magic.h" #include "test/gtest.h" namespace webrtc { @@ -20,13 +19,18 @@ class RemoteBitrateEstimatorAbsSendTimeTest : public RemoteBitrateEstimatorTest { public: RemoteBitrateEstimatorAbsSendTimeTest() {} + + RemoteBitrateEstimatorAbsSendTimeTest( + const RemoteBitrateEstimatorAbsSendTimeTest&) = delete; + RemoteBitrateEstimatorAbsSendTimeTest& operator=( + const RemoteBitrateEstimatorAbsSendTimeTest&) = delete; + virtual void SetUp() { bitrate_estimator_.reset(new RemoteBitrateEstimatorAbsSendTime( bitrate_observer_.get(), &clock_)); } protected: - RTC_DISALLOW_COPY_AND_ASSIGN(RemoteBitrateEstimatorAbsSendTimeTest); }; TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, InitialBehavior) { diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc index d0a7b22505..64ef39d935 100644 --- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc +++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc @@ -11,7 +11,6 @@ #include "modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h" #include "modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h" -#include "rtc_base/constructor_magic.h" #include "test/gtest.h" namespace webrtc { @@ -19,13 +18,18 @@ namespace webrtc { class RemoteBitrateEstimatorSingleTest : public RemoteBitrateEstimatorTest { public: RemoteBitrateEstimatorSingleTest() {} + + RemoteBitrateEstimatorSingleTest(const RemoteBitrateEstimatorSingleTest&) = + delete; + RemoteBitrateEstimatorSingleTest& operator=( + const RemoteBitrateEstimatorSingleTest&) = delete; + virtual void SetUp() { bitrate_estimator_.reset(new RemoteBitrateEstimatorSingleStream( bitrate_observer_.get(), &clock_)); } protected: - RTC_DISALLOW_COPY_AND_ASSIGN(RemoteBitrateEstimatorSingleTest); }; TEST_F(RemoteBitrateEstimatorSingleTest, InitialBehavior) { diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h b/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h index db4e899fbe..a3b1cfdb34 100644 --- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h +++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h @@ -18,7 +18,6 @@ #include <vector> #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" -#include "rtc_base/constructor_magic.h" #include "system_wrappers/include/clock.h" #include "test/gtest.h" @@ -71,6 +70,10 @@ class RtpStream { uint32_t frequency, uint32_t timestamp_offset, int64_t rtcp_receive_time); + + RtpStream(const RtpStream&) = delete; + RtpStream& operator=(const RtpStream&) = delete; + void set_rtp_timestamp_offset(uint32_t offset); // Generates a new frame for this stream. If called too soon after the @@ -104,8 +107,6 @@ class RtpStream { int64_t next_rtcp_time_; uint32_t rtp_timestamp_offset_; const double kNtpFracPerMs; - - RTC_DISALLOW_COPY_AND_ASSIGN(RtpStream); }; class StreamGenerator { @@ -116,6 +117,9 @@ class StreamGenerator { ~StreamGenerator(); + StreamGenerator(const StreamGenerator&) = delete; + StreamGenerator& operator=(const StreamGenerator&) = delete; + // Add a new stream. void AddStream(RtpStream* stream); @@ -142,8 +146,6 @@ class StreamGenerator { int64_t prev_arrival_time_us_; // All streams being transmitted on this simulated channel. StreamMap streams_; - - RTC_DISALLOW_COPY_AND_ASSIGN(StreamGenerator); }; } // namespace testing @@ -152,6 +154,10 @@ class RemoteBitrateEstimatorTest : public ::testing::Test { RemoteBitrateEstimatorTest(); virtual ~RemoteBitrateEstimatorTest(); + RemoteBitrateEstimatorTest(const RemoteBitrateEstimatorTest&) = delete; + RemoteBitrateEstimatorTest& operator=(const RemoteBitrateEstimatorTest&) = + delete; + protected: virtual void SetUp() = 0; @@ -213,8 +219,6 @@ class RemoteBitrateEstimatorTest : public ::testing::Test { std::unique_ptr<RemoteBitrateEstimator> bitrate_estimator_; std::unique_ptr<testing::StreamGenerator> stream_generator_; int64_t arrival_time_offset_ms_; - - RTC_DISALLOW_COPY_AND_ASSIGN(RemoteBitrateEstimatorTest); }; } // namespace webrtc diff --git a/modules/rtp_rtcp/include/remote_ntp_time_estimator.h b/modules/rtp_rtcp/include/remote_ntp_time_estimator.h index 5734a50e14..f31503dc4e 100644 --- a/modules/rtp_rtcp/include/remote_ntp_time_estimator.h +++ b/modules/rtp_rtcp/include/remote_ntp_time_estimator.h @@ -14,7 +14,6 @@ #include <stdint.h> #include "absl/types/optional.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/numerics/moving_median_filter.h" #include "system_wrappers/include/rtp_to_ntp_estimator.h" @@ -32,6 +31,9 @@ class RemoteNtpTimeEstimator { ~RemoteNtpTimeEstimator(); + RemoteNtpTimeEstimator(const RemoteNtpTimeEstimator&) = delete; + RemoteNtpTimeEstimator& operator=(const RemoteNtpTimeEstimator&) = delete; + // Updates the estimator with round trip time `rtt`, NTP seconds `ntp_secs`, // NTP fraction `ntp_frac` and RTP timestamp `rtp_timestamp`. bool UpdateRtcpTimestamp(int64_t rtt, @@ -52,7 +54,6 @@ class RemoteNtpTimeEstimator { MovingMedianFilter<int64_t> ntp_clocks_offset_estimator_; RtpToNtpEstimator rtp_to_ntp_; int64_t last_timing_log_ms_; - RTC_DISALLOW_COPY_AND_ASSIGN(RemoteNtpTimeEstimator); }; } // namespace webrtc diff --git a/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h b/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h index 8bee600692..d98dbd088d 100644 --- a/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h +++ b/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h @@ -16,7 +16,6 @@ #include <vector> #include "modules/rtp_rtcp/source/rtcp_packet.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { namespace rtcp { @@ -26,6 +25,9 @@ class CompoundPacket : public RtcpPacket { CompoundPacket(); ~CompoundPacket() override; + CompoundPacket(const CompoundPacket&) = delete; + CompoundPacket& operator=(const CompoundPacket&) = delete; + void Append(std::unique_ptr<RtcpPacket> packet); // Size of this packet in bytes (i.e. total size of nested packets). @@ -38,9 +40,6 @@ class CompoundPacket : public RtcpPacket { protected: std::vector<std::unique_ptr<RtcpPacket>> appended_packets_; - - private: - RTC_DISALLOW_COPY_AND_ASSIGN(CompoundPacket); }; } // namespace rtcp diff --git a/modules/rtp_rtcp/source/rtp_format_h264.h b/modules/rtp_rtcp/source/rtp_format_h264.h index f658594243..283beacb19 100644 --- a/modules/rtp_rtcp/source/rtp_format_h264.h +++ b/modules/rtp_rtcp/source/rtp_format_h264.h @@ -23,7 +23,6 @@ #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" #include "modules/video_coding/codecs/h264/include/h264_globals.h" #include "rtc_base/buffer.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -37,6 +36,9 @@ class RtpPacketizerH264 : public RtpPacketizer { ~RtpPacketizerH264() override; + RtpPacketizerH264(const RtpPacketizerH264&) = delete; + RtpPacketizerH264& operator=(const RtpPacketizerH264&) = delete; + size_t NumPackets() const override; // Get the next payload with H264 payload header. @@ -82,8 +84,6 @@ class RtpPacketizerH264 : public RtpPacketizer { size_t num_packets_left_; std::deque<rtc::ArrayView<const uint8_t>> input_fragments_; std::queue<PacketUnit> packets_; - - RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264); }; } // namespace webrtc #endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ diff --git a/modules/rtp_rtcp/source/rtp_format_video_generic.h b/modules/rtp_rtcp/source/rtp_format_video_generic.h index 5acd691163..fd44bd1980 100644 --- a/modules/rtp_rtcp/source/rtp_format_video_generic.h +++ b/modules/rtp_rtcp/source/rtp_format_video_generic.h @@ -16,7 +16,6 @@ #include "api/array_view.h" #include "modules/rtp_rtcp/source/rtp_format.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -48,6 +47,9 @@ class RtpPacketizerGeneric : public RtpPacketizer { ~RtpPacketizerGeneric() override; + RtpPacketizerGeneric(const RtpPacketizerGeneric&) = delete; + RtpPacketizerGeneric& operator=(const RtpPacketizerGeneric&) = delete; + size_t NumPackets() const override; // Get the next payload. @@ -64,8 +66,6 @@ class RtpPacketizerGeneric : public RtpPacketizer { rtc::ArrayView<const uint8_t> remaining_payload_; std::vector<int> payload_sizes_; std::vector<int>::const_iterator current_packet_; - - RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric); }; } // namespace webrtc #endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_ diff --git a/modules/rtp_rtcp/source/rtp_format_vp8.h b/modules/rtp_rtcp/source/rtp_format_vp8.h index 21009280e4..d1f569a946 100644 --- a/modules/rtp_rtcp/source/rtp_format_vp8.h +++ b/modules/rtp_rtcp/source/rtp_format_vp8.h @@ -35,7 +35,6 @@ #include "modules/rtp_rtcp/source/rtp_format.h" #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" #include "modules/video_coding/codecs/vp8/include/vp8_globals.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -50,6 +49,9 @@ class RtpPacketizerVp8 : public RtpPacketizer { ~RtpPacketizerVp8() override; + RtpPacketizerVp8(const RtpPacketizerVp8&) = delete; + RtpPacketizerVp8& operator=(const RtpPacketizerVp8&) = delete; + size_t NumPackets() const override; // Get the next payload with VP8 payload header. @@ -66,8 +68,6 @@ class RtpPacketizerVp8 : public RtpPacketizer { rtc::ArrayView<const uint8_t> remaining_payload_; std::vector<int> payload_sizes_; std::vector<int>::const_iterator current_packet_; - - RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerVp8); }; } // namespace webrtc diff --git a/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h b/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h index 916d6577f1..3ecaa476da 100644 --- a/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h +++ b/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h @@ -21,7 +21,6 @@ #include "modules/rtp_rtcp/source/rtp_format_vp8.h" #include "modules/video_coding/codecs/vp8/include/vp8_globals.h" #include "rtc_base/buffer.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -29,6 +28,10 @@ class RtpFormatVp8TestHelper { public: RtpFormatVp8TestHelper(const RTPVideoHeaderVP8* hdr, size_t payload_len); ~RtpFormatVp8TestHelper(); + + RtpFormatVp8TestHelper(const RtpFormatVp8TestHelper&) = delete; + RtpFormatVp8TestHelper& operator=(const RtpFormatVp8TestHelper&) = delete; + void GetAllPacketsAndCheck(RtpPacketizerVp8* packetizer, rtc::ArrayView<const size_t> expected_sizes); @@ -46,8 +49,6 @@ class RtpFormatVp8TestHelper { const RTPVideoHeaderVP8* const hdr_info_; rtc::Buffer payload_; - - RTC_DISALLOW_COPY_AND_ASSIGN(RtpFormatVp8TestHelper); }; } // namespace webrtc diff --git a/modules/rtp_rtcp/source/rtp_format_vp9.h b/modules/rtp_rtcp/source/rtp_format_vp9.h index 02458aea6a..3cf4dd56e5 100644 --- a/modules/rtp_rtcp/source/rtp_format_vp9.h +++ b/modules/rtp_rtcp/source/rtp_format_vp9.h @@ -30,7 +30,6 @@ #include "modules/rtp_rtcp/source/rtp_format.h" #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" #include "modules/video_coding/codecs/vp9/include/vp9_globals.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -43,6 +42,9 @@ class RtpPacketizerVp9 : public RtpPacketizer { ~RtpPacketizerVp9() override; + RtpPacketizerVp9(const RtpPacketizerVp9&) = delete; + RtpPacketizerVp9& operator=(const RtpPacketizerVp9&) = delete; + size_t NumPackets() const override; // Gets the next payload with VP9 payload header. @@ -64,8 +66,6 @@ class RtpPacketizerVp9 : public RtpPacketizer { rtc::ArrayView<const uint8_t> remaining_payload_; std::vector<int> payload_sizes_; std::vector<int>::const_iterator current_packet_; - - RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerVp9); }; } // namespace webrtc diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h index f3cb8d2c86..a411b237a0 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h +++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h @@ -27,7 +27,6 @@ #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" #include "modules/rtp_rtcp/source/rtp_sequence_number_map.h" #include "modules/rtp_rtcp/source/video_fec_generator.h" -#include "rtc_base/constructor_magic.h" #include "system_wrappers/include/ntp_time.h" namespace webrtc { @@ -47,6 +46,9 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface { Configuration() = default; Configuration(Configuration&& rhs) = default; + Configuration(const Configuration&) = delete; + Configuration& operator=(const Configuration&) = delete; + // True for a audio version of the RTP/RTCP module object false will create // a video version. bool audio = false; @@ -145,9 +147,6 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface { // Estimate RTT as non-sender as described in // https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5 bool non_sender_rtt_measurement = false; - - private: - RTC_DISALLOW_COPY_AND_ASSIGN(Configuration); }; // Stats for RTCP sender reports (SR) for a specific SSRC. diff --git a/modules/video_coding/codecs/test/videoprocessor.h b/modules/video_coding/codecs/test/videoprocessor.h index eafe492870..595a4f12b0 100644 --- a/modules/video_coding/codecs/test/videoprocessor.h +++ b/modules/video_coding/codecs/test/videoprocessor.h @@ -37,7 +37,6 @@ #include "modules/video_coding/utility/ivf_file_writer.h" #include "rtc_base/buffer.h" #include "rtc_base/checks.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/system/no_unique_address.h" #include "rtc_base/thread_annotations.h" #include "test/testsupport/frame_reader.h" @@ -70,6 +69,9 @@ class VideoProcessor { FrameWriterList* decoded_frame_writers); ~VideoProcessor(); + VideoProcessor(const VideoProcessor&) = delete; + VideoProcessor& operator=(const VideoProcessor&) = delete; + // Reads a frame and sends it to the encoder. When the encode callback // is received, the encoded frame is buffered. After encoding is finished // buffered frame is sent to decoder. Quality evaluation is done in @@ -270,8 +272,6 @@ class VideoProcessor { // This class must be operated on a TaskQueue. RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_; - - RTC_DISALLOW_COPY_AND_ASSIGN(VideoProcessor); }; } // namespace test diff --git a/modules/video_coding/fec_controller_default.h b/modules/video_coding/fec_controller_default.h index 6b9e8eb8e5..cca1658a99 100644 --- a/modules/video_coding/fec_controller_default.h +++ b/modules/video_coding/fec_controller_default.h @@ -19,7 +19,6 @@ #include "api/fec_controller.h" #include "modules/video_coding/media_opt_util.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" #include "system_wrappers/include/clock.h" @@ -32,6 +31,10 @@ class FecControllerDefault : public FecController { VCMProtectionCallback* protection_callback); explicit FecControllerDefault(Clock* clock); ~FecControllerDefault() override; + + FecControllerDefault(const FecControllerDefault&) = delete; + FecControllerDefault& operator=(const FecControllerDefault&) = delete; + void SetProtectionCallback( VCMProtectionCallback* protection_callback) override; void SetProtectionMethod(bool enable_fec, bool enable_nack) override; @@ -58,7 +61,7 @@ class FecControllerDefault : public FecController { std::unique_ptr<media_optimization::VCMLossProtectionLogic> loss_prot_logic_ RTC_GUARDED_BY(mutex_); size_t max_payload_size_ RTC_GUARDED_BY(mutex_); - RTC_DISALLOW_COPY_AND_ASSIGN(FecControllerDefault); + const float overhead_threshold_; }; diff --git a/modules/video_coding/h264_sprop_parameter_sets.h b/modules/video_coding/h264_sprop_parameter_sets.h index dbf27ef034..8a32f31cc0 100644 --- a/modules/video_coding/h264_sprop_parameter_sets.h +++ b/modules/video_coding/h264_sprop_parameter_sets.h @@ -15,13 +15,15 @@ #include <string> #include <vector> -#include "rtc_base/constructor_magic.h" - namespace webrtc { class H264SpropParameterSets { public: H264SpropParameterSets() {} + + H264SpropParameterSets(const H264SpropParameterSets&) = delete; + H264SpropParameterSets& operator=(const H264SpropParameterSets&) = delete; + bool DecodeSprop(const std::string& sprop); const std::vector<uint8_t>& sps_nalu() { return sps_; } const std::vector<uint8_t>& pps_nalu() { return pps_; } @@ -29,7 +31,6 @@ class H264SpropParameterSets { private: std::vector<uint8_t> sps_; std::vector<uint8_t> pps_; - RTC_DISALLOW_COPY_AND_ASSIGN(H264SpropParameterSets); }; } // namespace webrtc diff --git a/modules/video_coding/jitter_buffer.h b/modules/video_coding/jitter_buffer.h index 23f19bfa38..df7581a87e 100644 --- a/modules/video_coding/jitter_buffer.h +++ b/modules/video_coding/jitter_buffer.h @@ -26,7 +26,6 @@ #include "modules/video_coding/inter_frame_delay.h" #include "modules/video_coding/jitter_buffer_common.h" #include "modules/video_coding/jitter_estimator.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" @@ -74,6 +73,9 @@ class VCMJitterBuffer { ~VCMJitterBuffer(); + VCMJitterBuffer(const VCMJitterBuffer&) = delete; + VCMJitterBuffer& operator=(const VCMJitterBuffer&) = delete; + // Initializes and starts jitter buffer. void Start(); @@ -264,8 +266,6 @@ class VCMJitterBuffer { // average_packets_per_frame converges fast if we have fewer than this many // frames. int frame_counter_; - - RTC_DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer); }; } // namespace webrtc diff --git a/modules/video_coding/test/stream_generator.h b/modules/video_coding/test/stream_generator.h index f542bab3fc..ddb23ebb76 100644 --- a/modules/video_coding/test/stream_generator.h +++ b/modules/video_coding/test/stream_generator.h @@ -15,7 +15,6 @@ #include <list> #include "modules/video_coding/packet.h" -#include "rtc_base/constructor_magic.h" namespace webrtc { @@ -29,6 +28,10 @@ const int kDefaultFramePeriodMs = 1000 / kDefaultFrameRate; class StreamGenerator { public: StreamGenerator(uint16_t start_seq_num, int64_t current_time); + + StreamGenerator(const StreamGenerator&) = delete; + StreamGenerator& operator=(const StreamGenerator&) = delete; + void Init(uint16_t start_seq_num, int64_t current_time); // `time_ms` denotes the timestamp you want to put on the frame, and the unit @@ -64,8 +67,6 @@ class StreamGenerator { uint16_t sequence_number_; int64_t start_time_; uint8_t packet_buffer_[kMaxPacketSize]; - - RTC_DISALLOW_COPY_AND_ASSIGN(StreamGenerator); }; } // namespace webrtc diff --git a/modules/video_coding/utility/ivf_file_reader.h b/modules/video_coding/utility/ivf_file_reader.h index cc64d4c5c1..75f2e3ac8c 100644 --- a/modules/video_coding/utility/ivf_file_reader.h +++ b/modules/video_coding/utility/ivf_file_reader.h @@ -17,7 +17,6 @@ #include "absl/types/optional.h" #include "api/video/encoded_image.h" #include "api/video_codecs/video_codec.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/system/file_wrapper.h" namespace webrtc { @@ -27,6 +26,10 @@ class IvfFileReader { // Creates IvfFileReader. Returns nullptr if error acquired. static std::unique_ptr<IvfFileReader> Create(FileWrapper file); ~IvfFileReader(); + + IvfFileReader(const IvfFileReader&) = delete; + IvfFileReader& operator=(const IvfFileReader&) = delete; + // Reinitializes reader. Returns false if any error acquired. bool Reset(); @@ -72,8 +75,6 @@ class IvfFileReader { absl::optional<FrameHeader> next_frame_header_; bool has_error_; - - RTC_DISALLOW_COPY_AND_ASSIGN(IvfFileReader); }; } // namespace webrtc diff --git a/modules/video_coding/utility/ivf_file_writer.h b/modules/video_coding/utility/ivf_file_writer.h index 874f60adfc..b53459b5de 100644 --- a/modules/video_coding/utility/ivf_file_writer.h +++ b/modules/video_coding/utility/ivf_file_writer.h @@ -18,7 +18,6 @@ #include "api/video/encoded_image.h" #include "api/video/video_codec_type.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/system/file_wrapper.h" #include "rtc_base/time_utils.h" @@ -34,6 +33,9 @@ class IvfFileWriter { size_t byte_limit); ~IvfFileWriter(); + IvfFileWriter(const IvfFileWriter&) = delete; + IvfFileWriter& operator=(const IvfFileWriter&) = delete; + bool WriteFrame(const EncodedImage& encoded_image, VideoCodecType codec_type); bool Close(); @@ -57,8 +59,6 @@ class IvfFileWriter { bool using_capture_timestamps_; rtc::TimestampWrapAroundHandler wrap_handler_; FileWrapper file_; - - RTC_DISALLOW_COPY_AND_ASSIGN(IvfFileWriter); }; } // namespace webrtc diff --git a/modules/video_coding/utility/simulcast_rate_allocator.h b/modules/video_coding/utility/simulcast_rate_allocator.h index 9b2f9696e6..6f93dbde74 100644 --- a/modules/video_coding/utility/simulcast_rate_allocator.h +++ b/modules/video_coding/utility/simulcast_rate_allocator.h @@ -19,7 +19,6 @@ #include "api/video/video_bitrate_allocation.h" #include "api/video/video_bitrate_allocator.h" #include "api/video_codecs/video_codec.h" -#include "rtc_base/constructor_magic.h" #include "rtc_base/experiments/rate_control_settings.h" #include "rtc_base/experiments/stable_target_rate_experiment.h" @@ -30,6 +29,9 @@ class SimulcastRateAllocator : public VideoBitrateAllocator { explicit SimulcastRateAllocator(const VideoCodec& codec); ~SimulcastRateAllocator() override; + SimulcastRateAllocator(const SimulcastRateAllocator&) = delete; + SimulcastRateAllocator& operator=(const SimulcastRateAllocator&) = delete; + VideoBitrateAllocation Allocate( VideoBitrateAllocationParameters parameters) override; const VideoCodec& GetCodec() const; @@ -61,8 +63,6 @@ class SimulcastRateAllocator : public VideoBitrateAllocator { const RateControlSettings rate_control_settings_; std::vector<bool> stream_enabled_; bool legacy_conference_mode_; - - RTC_DISALLOW_COPY_AND_ASSIGN(SimulcastRateAllocator); }; } // namespace webrtc |