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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
Download Patch File Download Diff File Expand all files Collapse all files

5
modules/rtp_rtcp/include/remote_ntp_time_estimator.h
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@ -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

7
modules/rtp_rtcp/source/rtcp_packet/compound_packet.h
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@ -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

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_

6
modules/rtp_rtcp/source/rtp_format_video_generic.h
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@ -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_

6
modules/rtp_rtcp/source/rtp_format_vp8.h
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@ -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

7
modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
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@ -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

6
modules/rtp_rtcp/source/rtp_format_vp9.h
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@ -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

7
modules/rtp_rtcp/source/rtp_rtcp_interface.h
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@ -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.