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Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/

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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}
This commit is contained in:
Byoungchan Lee
2022-01-21 09:49:39 +09:00
committed by WebRTC LUCI CQ
parent ce6170fcdf
commit 604fd2f1ab
127 changed files with 466 additions and 393 deletions
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6
modules/rtp_rtcp/source/rtp_format_h264.h
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@ -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_