mirror of
https://github.com/mollyim/webrtc.git
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76ba7dbc4f
This reverts commit 50686460ca.
Reason for revert: breaking downstream
Original change's description:
> Replace RTPHeader memset with assignment from a fresh object.
>
> Since RTPHeader contains std::string members, memset results in
> undefined behavior, with memory leaks being the best case.
>
> Bug: chromium:945598
> Change-Id: I5c04e6b2fb08816fc036abfbb6ba7aaf19469687
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129720
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27282}
TBR=danilchap@webrtc.org,nisse@webrtc.org
Change-Id: Ifc69b6df7cd2ecd8d1b3dd5186e933ad12b53a8e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:945598
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129722
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27285}
102 lines
3.3 KiB
C++
102 lines
3.3 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/rtp_rtcp/include/rtp_header_parser.h"
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#include <string.h>
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "modules/rtp_rtcp/source/rtp_utility.h"
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#include "rtc_base/critical_section.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class RtpHeaderParserImpl : public RtpHeaderParser {
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public:
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RtpHeaderParserImpl();
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~RtpHeaderParserImpl() override = default;
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bool Parse(const uint8_t* packet,
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size_t length,
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RTPHeader* header) const override;
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bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id) override;
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bool RegisterRtpHeaderExtension(RtpExtension extension) override;
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bool DeregisterRtpHeaderExtension(RTPExtensionType type) override;
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bool DeregisterRtpHeaderExtension(RtpExtension extension) override;
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private:
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rtc::CriticalSection critical_section_;
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RtpHeaderExtensionMap rtp_header_extension_map_
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RTC_GUARDED_BY(critical_section_);
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};
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RtpHeaderParser* RtpHeaderParser::Create() {
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return new RtpHeaderParserImpl;
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}
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RtpHeaderParserImpl::RtpHeaderParserImpl() {}
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bool RtpHeaderParser::IsRtcp(const uint8_t* packet, size_t length) {
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RtpUtility::RtpHeaderParser rtp_parser(packet, length);
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return rtp_parser.RTCP();
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}
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absl::optional<uint32_t> RtpHeaderParser::GetSsrc(const uint8_t* packet,
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size_t length) {
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RtpUtility::RtpHeaderParser rtp_parser(packet, length);
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RTPHeader header;
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if (rtp_parser.Parse(&header, nullptr)) {
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return header.ssrc;
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}
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return absl::nullopt;
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}
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bool RtpHeaderParserImpl::Parse(const uint8_t* packet,
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size_t length,
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RTPHeader* header) const {
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RtpUtility::RtpHeaderParser rtp_parser(packet, length);
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memset(header, 0, sizeof(*header));
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RtpHeaderExtensionMap map;
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{
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rtc::CritScope cs(&critical_section_);
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map = rtp_header_extension_map_;
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}
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const bool valid_rtpheader = rtp_parser.Parse(header, &map);
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if (!valid_rtpheader) {
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return false;
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}
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return true;
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}
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bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RtpExtension extension) {
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rtc::CritScope cs(&critical_section_);
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return rtp_header_extension_map_.RegisterByUri(extension.id, extension.uri);
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}
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bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RTPExtensionType type,
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uint8_t id) {
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rtc::CritScope cs(&critical_section_);
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return rtp_header_extension_map_.RegisterByType(id, type);
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}
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bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RtpExtension extension) {
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rtc::CritScope cs(&critical_section_);
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return rtp_header_extension_map_.Deregister(
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rtp_header_extension_map_.GetType(extension.id));
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}
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bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RTPExtensionType type) {
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rtc::CritScope cs(&critical_section_);
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return rtp_header_extension_map_.Deregister(type) == 0;
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}
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} // namespace webrtc
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