/* * Copyright (c) 2012 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. */ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_ #include // size_t, ptrdiff_t #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" #include "webrtc/typedefs.h" namespace webrtc { const uint8_t kRtpMarkerBitMask = 0x80; RtpData* NullObjectRtpData(); RtpFeedback* NullObjectRtpFeedback(); RtpAudioFeedback* NullObjectRtpAudioFeedback(); ReceiveStatistics* NullObjectReceiveStatistics(); namespace RtpUtility { // January 1970, in NTP seconds. const uint32_t NTP_JAN_1970 = 2208988800UL; // Magic NTP fractional unit. const double NTP_FRAC = 4.294967296E+9; struct Payload { char name[RTP_PAYLOAD_NAME_SIZE]; bool audio; PayloadUnion typeSpecific; }; typedef std::map PayloadTypeMap; // Return the current RTP timestamp from the NTP timestamp // returned by the specified clock. uint32_t GetCurrentRTP(Clock* clock, uint32_t freq); // Return the current RTP absolute timestamp. uint32_t ConvertNTPTimeToRTP(uint32_t NTPsec, uint32_t NTPfrac, uint32_t freq); uint32_t pow2(uint8_t exp); // Returns true if |newTimestamp| is older than |existingTimestamp|. // |wrapped| will be set to true if there has been a wraparound between the // two timestamps. bool OldTimestamp(uint32_t newTimestamp, uint32_t existingTimestamp, bool* wrapped); bool StringCompare(const char* str1, const char* str2, const uint32_t length); // Round up to the nearest size that is a multiple of 4. size_t Word32Align(size_t size); class RtpHeaderParser { public: RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength); ~RtpHeaderParser(); bool RTCP() const; bool ParseRtcp(RTPHeader* header) const; bool Parse(RTPHeader& parsedPacket, RtpHeaderExtensionMap* ptrExtensionMap = NULL) const; private: void ParseOneByteExtensionHeader( RTPHeader& parsedPacket, const RtpHeaderExtensionMap* ptrExtensionMap, const uint8_t* ptrRTPDataExtensionEnd, const uint8_t* ptr) const; uint8_t ParsePaddingBytes( const uint8_t* ptrRTPDataExtensionEnd, const uint8_t* ptr) const; const uint8_t* const _ptrRTPDataBegin; const uint8_t* const _ptrRTPDataEnd; }; } // namespace RtpUtility } // namespace webrtc #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_