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317a1f09ed
WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
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 "test/rtp_header_parser.h"
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#include <memory>
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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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std::unique_ptr<RtpHeaderParser> RtpHeaderParser::CreateForTest() {
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return std::make_unique<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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*header = RTPHeader();
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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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