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Remove RTC_DISALLOW_COPY_AND_ASSIGN from rtc_base/

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Bug: webrtc:13555, webrtc:13082
Change-Id: I406b7f04497562866ea3329e97c5adc96e927b6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245680
Reviewed-by: Harald Alvestrand <hta@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@{#35691}
This commit is contained in:
Byoungchan Lee
2022-01-12 05:24:58 +09:00
committed by WebRTC LUCI CQ
parent ca15fcd37e
commit 14af7622a7
43 changed files with 203 additions and 151 deletions
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7
rtc_base/async_packet_socket.h
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@ -13,7 +13,6 @@
#include <vector>
#include "rtc_base/constructor_magic.h"
#include "rtc_base/dscp.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/socket.h"
@ -69,6 +68,9 @@ class RTC_EXPORT AsyncPacketSocket : public sigslot::has_slots<> {
AsyncPacketSocket();
~AsyncPacketSocket() override;
AsyncPacketSocket(const AsyncPacketSocket&) = delete;
AsyncPacketSocket& operator=(const AsyncPacketSocket&) = delete;
// Returns current local address. Address may be set to null if the
// socket is not bound yet (GetState() returns STATE_BINDING).
virtual SocketAddress GetLocalAddress() const = 0;
@ -127,9 +129,6 @@ class RTC_EXPORT AsyncPacketSocket : public sigslot::has_slots<> {
// Emitted for client TCP sockets when state is changed from
// CONNECTED to CLOSED.
sigslot::signal2<AsyncPacketSocket*, int> SignalClose;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(AsyncPacketSocket);
};
// Listen socket, producing an AsyncPacketSocket when a peer connects.