/* * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #include "test/rtp_header_parser.h" #include #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "modules/rtp_rtcp/source/rtp_utility.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" namespace webrtc { class RtpHeaderParserImpl : public RtpHeaderParser { public: RtpHeaderParserImpl(); ~RtpHeaderParserImpl() override = default; bool Parse(const uint8_t* packet, size_t length, RTPHeader* header) const override; bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id) override; bool RegisterRtpHeaderExtension(RtpExtension extension) override; bool DeregisterRtpHeaderExtension(RTPExtensionType type) override; bool DeregisterRtpHeaderExtension(RtpExtension extension) override; private: mutable Mutex mutex_; RtpHeaderExtensionMap rtp_header_extension_map_ RTC_GUARDED_BY(mutex_); }; std::unique_ptr RtpHeaderParser::CreateForTest() { return std::make_unique(); } RtpHeaderParserImpl::RtpHeaderParserImpl() {} bool RtpHeaderParser::IsRtcp(const uint8_t* packet, size_t length) { RtpUtility::RtpHeaderParser rtp_parser(packet, length); return rtp_parser.RTCP(); } absl::optional RtpHeaderParser::GetSsrc(const uint8_t* packet, size_t length) { RtpUtility::RtpHeaderParser rtp_parser(packet, length); RTPHeader header; if (rtp_parser.Parse(&header, nullptr)) { return header.ssrc; } return absl::nullopt; } bool RtpHeaderParserImpl::Parse(const uint8_t* packet, size_t length, RTPHeader* header) const { RtpUtility::RtpHeaderParser rtp_parser(packet, length); *header = RTPHeader(); RtpHeaderExtensionMap map; { MutexLock lock(&mutex_); map = rtp_header_extension_map_; } const bool valid_rtpheader = rtp_parser.Parse(header, &map); if (!valid_rtpheader) { return false; } return true; } bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RtpExtension extension) { MutexLock lock(&mutex_); return rtp_header_extension_map_.RegisterByUri(extension.id, extension.uri); } bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id) { MutexLock lock(&mutex_); return rtp_header_extension_map_.RegisterByType(id, type); } bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RtpExtension extension) { MutexLock lock(&mutex_); return rtp_header_extension_map_.Deregister( rtp_header_extension_map_.GetType(extension.id)); } bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RTPExtensionType type) { MutexLock lock(&mutex_); return rtp_header_extension_map_.Deregister(type) == 0; } } // namespace webrtc