Remove RTC_DISALLOW_COPY_AND_ASSIGN more.

Bug: webrtc:13555, webrtc:13082
Change-Id: I9c07708108da0a26f5e228384fd56cef4d1540b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35749}
This commit is contained in:
Byoungchan Lee 2022-01-18 09:35:48 +09:00 committed by WebRTC LUCI CQ
parent cf9899c518
commit c065e739e2
28 changed files with 100 additions and 86 deletions

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@ -16,7 +16,6 @@
#include "api/audio/channel_layout.h"
#include "api/rtp_packet_infos.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -58,6 +57,9 @@ class AudioFrame {
AudioFrame();
AudioFrame(const AudioFrame&) = delete;
AudioFrame& operator=(const AudioFrame&) = delete;
// Resets all members to their default state.
void Reset();
// Same as Reset(), but leaves mute state unchanged. Muting a frame requires
@ -164,8 +166,6 @@ class AudioFrame {
// capture timestamp of a received frame is found in `packet_infos_`.
// This timestamp MUST be based on the same clock as rtc::TimeMillis().
absl::optional<int64_t> absolute_capture_timestamp_ms_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
};
} // namespace webrtc

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@ -20,7 +20,6 @@
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "rtc_base/buffer.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -37,6 +36,9 @@ class AudioDecoder {
AudioDecoder() = default;
virtual ~AudioDecoder() = default;
AudioDecoder(const AudioDecoder&) = delete;
AudioDecoder& operator=(const AudioDecoder&) = delete;
class EncodedAudioFrame {
public:
struct DecodeResult {
@ -187,9 +189,6 @@ class AudioDecoder {
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type);
private:
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
};
} // namespace webrtc

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@ -22,7 +22,6 @@
#include "api/candidate.h"
#include "api/jsep.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -64,6 +63,10 @@ class JsepCandidateCollection : public IceCandidateCollection {
// Move constructor is defined so that a vector of JsepCandidateCollections
// can be resized.
JsepCandidateCollection(JsepCandidateCollection&& o);
JsepCandidateCollection(const JsepCandidateCollection&) = delete;
JsepCandidateCollection& operator=(const JsepCandidateCollection&) = delete;
// Returns a copy of the candidate collection.
JsepCandidateCollection Clone() const;
size_t count() const override;
@ -80,8 +83,6 @@ class JsepCandidateCollection : public IceCandidateCollection {
private:
std::vector<std::unique_ptr<JsepIceCandidate>> candidates_;
RTC_DISALLOW_COPY_AND_ASSIGN(JsepCandidateCollection);
};
} // namespace webrtc

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@ -22,7 +22,6 @@
#include "api/candidate.h"
#include "api/jsep.h"
#include "api/jsep_ice_candidate.h"
#include "rtc_base/constructor_magic.h"
namespace cricket {
class SessionDescription;
@ -43,6 +42,9 @@ class JsepSessionDescription : public SessionDescriptionInterface {
absl::string_view session_version);
virtual ~JsepSessionDescription();
JsepSessionDescription(const JsepSessionDescription&) = delete;
JsepSessionDescription& operator=(const JsepSessionDescription&) = delete;
// Takes ownership of `description`.
bool Initialize(std::unique_ptr<cricket::SessionDescription> description,
const std::string& session_id,
@ -82,8 +84,6 @@ class JsepSessionDescription : public SessionDescriptionInterface {
bool GetMediasectionIndex(const IceCandidateInterface* candidate,
size_t* index);
int GetMediasectionIndex(const cricket::Candidate& candidate);
RTC_DISALLOW_COPY_AND_ASSIGN(JsepSessionDescription);
};
} // namespace webrtc

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@ -12,7 +12,6 @@
#include <type_traits>
#include "rtc_base/constructor_magic.h"
#include "rtc_base/ref_counter.h"
namespace rtc {
@ -21,6 +20,9 @@ class RefCountedBase {
public:
RefCountedBase() = default;
RefCountedBase(const RefCountedBase&) = delete;
RefCountedBase& operator=(const RefCountedBase&) = delete;
void AddRef() const { ref_count_.IncRef(); }
RefCountReleaseStatus Release() const {
const auto status = ref_count_.DecRef();
@ -39,8 +41,6 @@ class RefCountedBase {
private:
mutable webrtc::webrtc_impl::RefCounter ref_count_{0};
RTC_DISALLOW_COPY_AND_ASSIGN(RefCountedBase);
};
// Template based version of `RefCountedBase` for simple implementations that do
@ -61,6 +61,9 @@ class RefCountedNonVirtual {
public:
RefCountedNonVirtual() = default;
RefCountedNonVirtual(const RefCountedNonVirtual&) = delete;
RefCountedNonVirtual& operator=(const RefCountedNonVirtual&) = delete;
void AddRef() const { ref_count_.IncRef(); }
RefCountReleaseStatus Release() const {
// If you run into this assert, T has virtual methods. There are two
@ -88,8 +91,6 @@ class RefCountedNonVirtual {
private:
mutable webrtc::webrtc_impl::RefCounter ref_count_{0};
RTC_DISALLOW_COPY_AND_ASSIGN(RefCountedNonVirtual);
};
} // namespace rtc

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@ -22,7 +22,6 @@
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
@ -288,6 +287,9 @@ class RTC_EXPORT StatsReport {
~Value();
Value(const Value&) = delete;
Value& operator=(const Value&) = delete;
// Support ref counting. Note that for performance reasons, we
// don't use thread safe operations. Therefore, all operations
// affecting the ref count (in practice, creation and copying of
@ -358,8 +360,6 @@ class RTC_EXPORT StatsReport {
const char* static_string_;
Id* id_;
} value_;
RTC_DISALLOW_COPY_AND_ASSIGN(Value);
};
typedef rtc::scoped_refptr<Value> ValuePtr;
@ -369,6 +369,9 @@ class RTC_EXPORT StatsReport {
explicit StatsReport(const Id& id);
~StatsReport();
StatsReport(const StatsReport&) = delete;
StatsReport& operator=(const StatsReport&) = delete;
// Factory functions for various types of stats IDs.
static Id NewBandwidthEstimationId();
static Id NewTypedId(StatsType type, const std::string& id);
@ -408,8 +411,6 @@ class RTC_EXPORT StatsReport {
const Id id_;
double timestamp_; // Time since 1970-01-01T00:00:00Z in milliseconds.
Values values_;
RTC_DISALLOW_COPY_AND_ASSIGN(StatsReport);
};
// Typedef for an array of const StatsReport pointers.

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@ -51,7 +51,6 @@
#include "modules/utility/include/process_thread.h"
#include "modules/video_coding/fec_controller_default.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
@ -209,6 +208,9 @@ class Call final : public webrtc::Call,
TaskQueueFactory* task_queue_factory);
~Call() override;
Call(const Call&) = delete;
Call& operator=(const Call&) = delete;
// Implements webrtc::Call.
PacketReceiver* Receiver() override;
@ -469,8 +471,6 @@ class Call final : public webrtc::Call,
RTC_NO_UNIQUE_ADDRESS SequenceChecker sent_packet_sequence_checker_;
absl::optional<rtc::SentPacket> last_sent_packet_
RTC_GUARDED_BY(sent_packet_sequence_checker_);
RTC_DISALLOW_COPY_AND_ASSIGN(Call);
};
} // namespace internal

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@ -23,7 +23,6 @@
#include "api/test/simulated_network.h"
#include "call/call.h"
#include "call/simulated_packet_receiver.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
@ -109,6 +108,9 @@ class FakeNetworkPipe : public SimulatedPacketReceiverInterface {
~FakeNetworkPipe() override;
FakeNetworkPipe(const FakeNetworkPipe&) = delete;
FakeNetworkPipe& operator=(const FakeNetworkPipe&) = delete;
void SetClockOffset(int64_t offset_ms);
// Must not be called in parallel with DeliverPacket or Process.
@ -228,8 +230,6 @@ class FakeNetworkPipe : public SimulatedPacketReceiverInterface {
int64_t last_log_time_us_;
std::map<Transport*, size_t> active_transports_ RTC_GUARDED_BY(config_lock_);
RTC_DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
};
} // namespace webrtc

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@ -14,7 +14,6 @@
#include "absl/types/optional.h"
#include "api/transport/bitrate_settings.h"
#include "api/units/data_rate.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -24,6 +23,10 @@ class RtpBitrateConfigurator {
public:
explicit RtpBitrateConfigurator(const BitrateConstraints& bitrate_config);
~RtpBitrateConfigurator();
RtpBitrateConfigurator(const RtpBitrateConfigurator&) = delete;
RtpBitrateConfigurator& operator=(const RtpBitrateConfigurator&) = delete;
BitrateConstraints GetConfig() const;
// The greater min and smaller max set by this and SetClientBitratePreferences
@ -68,8 +71,6 @@ class RtpBitrateConfigurator {
// Bandwidth cap applied for relayed calls.
DataRate max_bitrate_over_relay_ = DataRate::PlusInfinity();
RTC_DISALLOW_COPY_AND_ASSIGN(RtpBitrateConfigurator);
};
} // namespace webrtc

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@ -32,7 +32,6 @@
#include "modules/pacing/rtp_packet_pacer.h"
#include "modules/pacing/task_queue_paced_sender.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/network_route.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/task_queue.h"
@ -63,6 +62,10 @@ class RtpTransportControllerSend final
const WebRtcKeyValueConfig* trials);
~RtpTransportControllerSend() override;
RtpTransportControllerSend(const RtpTransportControllerSend&) = delete;
RtpTransportControllerSend& operator=(const RtpTransportControllerSend&) =
delete;
// TODO(tommi): Change to std::unique_ptr<>.
RtpVideoSenderInterface* CreateRtpVideoSender(
const std::map<uint32_t, RtpState>& suspended_ssrcs,
@ -215,7 +218,6 @@ class RtpTransportControllerSend final
// `task_queue_` is defined last to ensure all pending tasks are cancelled
// and deleted before any other members.
rtc::TaskQueue task_queue_;
RTC_DISALLOW_COPY_AND_ASSIGN(RtpTransportControllerSend);
};
} // namespace webrtc

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@ -35,7 +35,6 @@
#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
@ -89,6 +88,9 @@ class RtpVideoSender : public RtpVideoSenderInterface,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
~RtpVideoSender() override;
RtpVideoSender(const RtpVideoSender&) = delete;
RtpVideoSender& operator=(const RtpVideoSender&) = delete;
// RtpVideoSender will only route packets if being active, all packets will be
// dropped otherwise.
void SetActive(bool active) RTC_LOCKS_EXCLUDED(mutex_) override;
@ -209,8 +211,6 @@ class RtpVideoSender : public RtpVideoSenderInterface,
// This map is set at construction time and never changed, but it's
// non-trivial to make it properly const.
std::map<uint32_t, RtpRtcpInterface*> ssrc_to_rtp_module_;
RTC_DISALLOW_COPY_AND_ASSIGN(RtpVideoSender);
};
} // namespace webrtc

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@ -20,7 +20,6 @@
#include "absl/strings/string_view.h"
#include "rtc_base/bit_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -36,6 +35,9 @@ class BitWriter final {
RTC_DCHECK_GT(byte_count, 0);
}
BitWriter(const BitWriter&) = delete;
BitWriter& operator=(const BitWriter&) = delete;
void WriteBits(uint64_t val, size_t bit_count);
void WriteBits(absl::string_view input);
@ -52,8 +54,6 @@ class BitWriter final {
// to go anywhere near the limit, though, so this is good enough.
size_t written_bits_;
bool valid_;
RTC_DISALLOW_COPY_AND_ASSIGN(BitWriter);
};
} // namespace webrtc

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@ -21,7 +21,6 @@
#include "rtc_base/bit_buffer.h"
#include "rtc_base/bitstream_reader.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
@ -187,6 +186,9 @@ class FixedLengthDeltaEncoder final {
absl::optional<uint64_t> base,
const std::vector<absl::optional<uint64_t>>& values);
FixedLengthDeltaEncoder(const FixedLengthDeltaEncoder&) = delete;
FixedLengthDeltaEncoder& operator=(const FixedLengthDeltaEncoder&) = delete;
private:
// Calculate min/max values of unsigned/signed deltas, given the bit width
// of all the values in the series.
@ -249,8 +251,6 @@ class FixedLengthDeltaEncoder final {
// ctor has finished running when this is constructed, so that the lower
// bound on the buffer size would be guaranteed correct.
std::unique_ptr<BitWriter> writer_;
RTC_DISALLOW_COPY_AND_ASSIGN(FixedLengthDeltaEncoder);
};
// TODO(eladalon): Reduce the number of passes.
@ -566,6 +566,9 @@ class FixedLengthDeltaDecoder final {
absl::optional<uint64_t> base,
size_t num_of_deltas);
FixedLengthDeltaDecoder(const FixedLengthDeltaDecoder&) = delete;
FixedLengthDeltaDecoder& operator=(const FixedLengthDeltaDecoder&) = delete;
private:
// Reads the encoding header in `input` and returns a FixedLengthDeltaDecoder
// with the corresponding configuration, that can be used to decode the
@ -619,8 +622,6 @@ class FixedLengthDeltaDecoder final {
// The number of values to be known to be decoded.
const size_t num_of_deltas_;
RTC_DISALLOW_COPY_AND_ASSIGN(FixedLengthDeltaDecoder);
};
bool FixedLengthDeltaDecoder::IsSuitableDecoderFor(const std::string& input) {

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@ -17,6 +17,7 @@
#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 {

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@ -15,7 +15,6 @@
#include "rtc_base/async_packet_socket.h"
#include "rtc_base/async_tcp_socket.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/socket.h"
#include "rtc_base/socket_address.h"
@ -32,6 +31,9 @@ class AsyncStunTCPSocket : public rtc::AsyncTCPSocketBase {
explicit AsyncStunTCPSocket(rtc::Socket* socket);
AsyncStunTCPSocket(const AsyncStunTCPSocket&) = delete;
AsyncStunTCPSocket& operator=(const AsyncStunTCPSocket&) = delete;
int Send(const void* pv,
size_t cb,
const rtc::PacketOptions& options) override;
@ -42,8 +44,6 @@ class AsyncStunTCPSocket : public rtc::AsyncTCPSocketBase {
// This method also returns the number of padding bytes needed/added to the
// turn message. `pad_bytes` should be used only when `is_turn` is true.
size_t GetExpectedLength(const void* data, size_t len, int* pad_bytes);
RTC_DISALLOW_COPY_AND_ASSIGN(AsyncStunTCPSocket);
};
} // namespace cricket

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@ -22,7 +22,6 @@
#include "p2p/base/ice_transport_internal.h"
#include "rtc_base/buffer.h"
#include "rtc_base/buffer_queue.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/stream.h"
#include "rtc_base/strings/string_builder.h"
@ -40,6 +39,9 @@ class StreamInterfaceChannel : public rtc::StreamInterface {
public:
explicit StreamInterfaceChannel(IceTransportInternal* ice_transport);
StreamInterfaceChannel(const StreamInterfaceChannel&) = delete;
StreamInterfaceChannel& operator=(const StreamInterfaceChannel&) = delete;
// Push in a packet; this gets pulled out from Read().
bool OnPacketReceived(const char* data, size_t size);
@ -60,8 +62,6 @@ class StreamInterfaceChannel : public rtc::StreamInterface {
IceTransportInternal* const ice_transport_; // owned by DtlsTransport
rtc::StreamState state_ RTC_GUARDED_BY(sequence_checker_);
rtc::BufferQueue packets_ RTC_GUARDED_BY(sequence_checker_);
RTC_DISALLOW_COPY_AND_ASSIGN(StreamInterfaceChannel);
};
// This class provides a DTLS SSLStreamAdapter inside a TransportChannel-style
@ -110,6 +110,9 @@ class DtlsTransport : public DtlsTransportInternal {
~DtlsTransport() override;
DtlsTransport(const DtlsTransport&) = delete;
DtlsTransport& operator=(const DtlsTransport&) = delete;
webrtc::DtlsTransportState dtls_state() const override;
const std::string& transport_name() const override;
int component() const override;
@ -248,8 +251,6 @@ class DtlsTransport : public DtlsTransportInternal {
bool writable_ = false;
webrtc::RtcEventLog* const event_log_;
RTC_DISALLOW_COPY_AND_ASSIGN(DtlsTransport);
};
} // namespace cricket

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@ -25,7 +25,6 @@
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/packet_transport_internal.h"
#include "rtc_base/callback_list.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/ssl_certificate.h"
#include "rtc_base/ssl_fingerprint.h"
#include "rtc_base/ssl_stream_adapter.h"
@ -48,6 +47,9 @@ class DtlsTransportInternal : public rtc::PacketTransportInternal {
public:
~DtlsTransportInternal() override;
DtlsTransportInternal(const DtlsTransportInternal&) = delete;
DtlsTransportInternal& operator=(const DtlsTransportInternal&) = delete;
virtual webrtc::DtlsTransportState dtls_state() const = 0;
virtual int component() const = 0;
@ -135,7 +137,6 @@ class DtlsTransportInternal : public rtc::PacketTransportInternal {
DtlsTransportInternal();
private:
RTC_DISALLOW_COPY_AND_ASSIGN(DtlsTransportInternal);
webrtc::CallbackList<const rtc::SSLHandshakeError>
dtls_handshake_error_callback_list_;
webrtc::CallbackList<DtlsTransportInternal*, const webrtc::DtlsTransportState>

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@ -56,7 +56,6 @@
#include "p2p/base/transport_description.h"
#include "rtc_base/async_packet_socket.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/dscp.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
@ -124,6 +123,9 @@ class RTC_EXPORT P2PTransportChannel : public IceTransportInternal {
IceControllerFactoryInterface* ice_controller_factory = nullptr);
~P2PTransportChannel() override;
P2PTransportChannel(const P2PTransportChannel&) = delete;
P2PTransportChannel& operator=(const P2PTransportChannel&) = delete;
// From TransportChannelImpl:
IceTransportState GetState() const override;
webrtc::IceTransportState GetIceTransportState() const override;
@ -494,8 +496,6 @@ class RTC_EXPORT P2PTransportChannel : public IceTransportInternal {
int64_t last_data_received_ms_ = 0;
IceFieldTrials field_trials_;
RTC_DISALLOW_COPY_AND_ASSIGN(P2PTransportChannel);
};
} // namespace cricket

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@ -21,7 +21,6 @@
#include "rtc_base/async_packet_socket.h"
#include "rtc_base/async_resolver_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/helpers.h"
#include "rtc_base/logging.h"
#include "rtc_base/task_utils/to_queued_task.h"
@ -69,6 +68,9 @@ class StunProber::Requester : public sigslot::has_slots<> {
const std::vector<rtc::SocketAddress>& server_ips);
~Requester() override;
Requester(const Requester&) = delete;
Requester& operator=(const Requester&) = delete;
// There is no callback for SendStunRequest as the underneath socket send is
// expected to be completed immediately. Otherwise, it'll skip this request
// and move to the next one.
@ -105,8 +107,6 @@ class StunProber::Requester : public sigslot::has_slots<> {
int16_t num_response_received_ = 0;
webrtc::SequenceChecker& thread_checker_;
RTC_DISALLOW_COPY_AND_ASSIGN(Requester);
};
StunProber::Requester::Requester(

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@ -17,7 +17,6 @@
#include "api/sequence_checker.h"
#include "rtc_base/byte_buffer.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/ip_address.h"
#include "rtc_base/network.h"
#include "rtc_base/socket_address.h"
@ -101,6 +100,9 @@ class RTC_EXPORT StunProber : public sigslot::has_slots<> {
const rtc::NetworkManager::NetworkList& networks);
~StunProber() override;
StunProber(const StunProber&) = delete;
StunProber& operator=(const StunProber&) = delete;
// Begin performing the probe test against the `servers`. If
// `shared_socket_mode` is false, each request will be done with a new socket.
// Otherwise, a unique socket will be used for a single round of requests
@ -241,8 +243,6 @@ class RTC_EXPORT StunProber : public sigslot::has_slots<> {
rtc::NetworkManager::NetworkList networks_;
webrtc::ScopedTaskSafety task_safety_;
RTC_DISALLOW_COPY_AND_ASSIGN(StunProber);
};
} // namespace stunprober

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@ -18,7 +18,6 @@
#include "api/dtmf_sender_interface.h"
#include "api/scoped_refptr.h"
#include "pc/proxy.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/location.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/task_utils/pending_task_safety_flag.h"
@ -72,6 +71,9 @@ class DtmfSender : public DtmfSenderInterface, public sigslot::has_slots<> {
DtmfSender(rtc::Thread* signaling_thread, DtmfProviderInterface* provider);
virtual ~DtmfSender();
DtmfSender(const DtmfSender&) = delete;
DtmfSender& operator=(const DtmfSender&) = delete;
private:
DtmfSender();
@ -96,8 +98,6 @@ class DtmfSender : public DtmfSenderInterface, public sigslot::has_slots<> {
// For cancelling the tasks which feed the DTMF provider one tone at a time.
rtc::scoped_refptr<PendingTaskSafetyFlag> safety_flag_ RTC_GUARDED_BY(
signaling_thread_) RTC_PT_GUARDED_BY(signaling_thread_) = nullptr;
RTC_DISALLOW_COPY_AND_ASSIGN(DtmfSender);
};
// Define proxy for DtmfSenderInterface.

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@ -44,7 +44,6 @@
#include "pc/srtp_transport.h"
#include "pc/transport_stats.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/ssl_fingerprint.h"
#include "rtc_base/ssl_stream_adapter.h"
@ -106,6 +105,9 @@ class JsepTransport {
~JsepTransport();
JsepTransport(const JsepTransport&) = delete;
JsepTransport& operator=(const JsepTransport&) = delete;
// Returns the MID of this transport. This is only used for logging.
const std::string& mid() const { return mid_; }
@ -326,8 +328,6 @@ class JsepTransport {
// `rtcp_dtls_transport_` is destroyed. The JsepTransportController will
// receive the callback and update the aggregate transport states.
std::function<void()> rtcp_mux_active_callback_;
RTC_DISALLOW_COPY_AND_ASSIGN(JsepTransport);
};
} // namespace cricket

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@ -58,7 +58,6 @@
#include "pc/transport_stats.h"
#include "rtc_base/callback_list.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/helpers.h"
#include "rtc_base/ref_counted_object.h"
@ -150,6 +149,9 @@ class JsepTransportController : public sigslot::has_slots<> {
Config config);
virtual ~JsepTransportController();
JsepTransportController(const JsepTransportController&) = delete;
JsepTransportController& operator=(const JsepTransportController&) = delete;
// The main method to be called; applies a description at the transport
// level, creating/destroying transport objects as needed and updating their
// properties. This includes RTP, DTLS, and ICE (but not SCTP). At least not
@ -478,8 +480,6 @@ class JsepTransportController : public sigslot::has_slots<> {
rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
BundleManager bundles_;
RTC_DISALLOW_COPY_AND_ASSIGN(JsepTransportController);
};
} // namespace webrtc

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@ -15,7 +15,6 @@
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/synchronization/mutex.h"
// Forward declaration to avoid pulling in libsrtp headers here
@ -35,6 +34,9 @@ class SrtpSession {
SrtpSession();
~SrtpSession();
SrtpSession(const SrtpSession&) = delete;
SrtpSession& operator=(const SrtpSession&) = delete;
// Configures the session for sending data using the specified
// cipher-suite and key. Receiving must be done by a separate session.
bool SetSend(int cs,
@ -141,7 +143,6 @@ class SrtpSession {
bool external_auth_enabled_ = false;
int decryption_failure_count_ = 0;
bool dump_plain_rtp_ = false;
RTC_DISALLOW_COPY_AND_ASSIGN(SrtpSession);
};
} // namespace cricket

View file

@ -20,7 +20,6 @@
#include "api/video/video_source_interface.h"
#include "media/base/video_broadcaster.h"
#include "pc/video_track_source.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/thread_annotations.h"
@ -45,6 +44,9 @@ class VideoRtpTrackSource : public VideoTrackSource {
explicit VideoRtpTrackSource(Callback* callback);
VideoRtpTrackSource(const VideoRtpTrackSource&) = delete;
VideoRtpTrackSource& operator=(const VideoRtpTrackSource&) = delete;
// Call before the object implementing Callback finishes it's destructor. No
// more callbacks will be fired after completion. Must be called on the
// worker thread
@ -83,8 +85,6 @@ class VideoRtpTrackSource : public VideoTrackSource {
std::vector<rtc::VideoSinkInterface<RecordableEncodedFrame>*> encoded_sinks_
RTC_GUARDED_BY(mu_);
Callback* callback_ RTC_GUARDED_BY(worker_sequence_checker_);
RTC_DISALLOW_COPY_AND_ASSIGN(VideoRtpTrackSource);
};
} // namespace webrtc

View file

@ -26,7 +26,6 @@
#include "pc/channel_manager.h"
#include "pc/media_session.h"
#include "pc/sdp_state_provider.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/message_handler.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/rtc_certificate_generator.h"
@ -92,6 +91,11 @@ class WebRtcSessionDescriptionFactory : public rtc::MessageHandler,
on_certificate_ready);
virtual ~WebRtcSessionDescriptionFactory();
WebRtcSessionDescriptionFactory(const WebRtcSessionDescriptionFactory&) =
delete;
WebRtcSessionDescriptionFactory& operator=(
const WebRtcSessionDescriptionFactory&) = delete;
static void CopyCandidatesFromSessionDescription(
const SessionDescriptionInterface* source_desc,
const std::string& content_name,
@ -159,8 +163,6 @@ class WebRtcSessionDescriptionFactory : public rtc::MessageHandler,
std::function<void(const rtc::scoped_refptr<rtc::RTCCertificate>&)>
on_certificate_ready_;
RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSessionDescriptionFactory);
};
} // namespace webrtc

View file

@ -11,7 +11,6 @@
#include <algorithm>
#include "rtc_base/constructor_magic.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
@ -35,6 +34,9 @@ class RtcHistogram {
RTC_DCHECK_GT(bucket_count, 0);
}
RtcHistogram(const RtcHistogram&) = delete;
RtcHistogram& operator=(const RtcHistogram&) = delete;
void Add(int sample) {
sample = std::min(sample, max_);
sample = std::max(sample, min_ - 1); // Underflow bucket.
@ -99,8 +101,6 @@ class RtcHistogram {
const int min_;
const int max_;
SampleInfo info_ RTC_GUARDED_BY(mutex_);
RTC_DISALLOW_COPY_AND_ASSIGN(RtcHistogram);
};
class RtcHistogramMap {
@ -108,6 +108,9 @@ class RtcHistogramMap {
RtcHistogramMap() {}
~RtcHistogramMap() {}
RtcHistogramMap(const RtcHistogramMap&) = delete;
RtcHistogramMap& operator=(const RtcHistogramMap&) = delete;
Histogram* GetCountsHistogram(const std::string& name,
int min,
int max,
@ -178,8 +181,6 @@ class RtcHistogramMap {
mutable Mutex mutex_;
std::map<std::string, std::unique_ptr<RtcHistogram>> map_
RTC_GUARDED_BY(mutex_);
RTC_DISALLOW_COPY_AND_ASSIGN(RtcHistogramMap);
};
// RtcHistogramMap is allocated upon call to Enable().

View file

@ -26,6 +26,7 @@
#include "api/test/network_emulation_manager.h"
#include "api/test/simulated_network.h"
#include "api/units/timestamp.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network.h"
#include "rtc_base/network_constants.h"