// // ZoneMinder RTP Source Class Implementation, $Date$, $Revision$ // Copyright (C) 2001-2008 Philip Coombes // // This program is free software; you can redistribute it and/or // modify it under the terms of the GNU General Public License // as published by the Free Software Foundation; either version 2 // of the License, or (at your option) any later version. // // This program is distributed in the hope that it will be useful, // but WITHOUT ANY WARRANTY; without even the implied warranty of // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the // GNU General Public License for more details. // // You should have received a copy of the GNU General Public License // along with this program; if not, write to the Free Software // Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. // #include "zm_rtp_source.h" #include "zm_time.h" #include "zm_rtp_data.h" #include #if HAVE_LIBAVCODEC RtpSource::RtpSource( int id, const std::string &localHost, int localPortBase, const std::string &remoteHost, int remotePortBase, uint32_t ssrc, uint16_t seq, uint32_t rtpClock, uint32_t rtpTime, _AVCODECID codecId ) : mId( id ), mSsrc( ssrc ), mLocalHost( localHost ), mRemoteHost( remoteHost ), mRtpClock( rtpClock ), mCodecId( codecId ), mFrame( 65536 ), mFrameCount( 0 ), mFrameGood( true ), mFrameReady( false ), mFrameProcessed( false ) { char hostname[256] = ""; gethostname( hostname, sizeof(hostname) ); mCname = stringtf( "zm-%d@%s", mId, hostname ); Debug( 3, "RTP CName = %s", mCname.c_str() ); init( seq ); mMaxSeq = seq - 1; mProbation = MIN_SEQUENTIAL; mLocalPortChans[0] = localPortBase; mLocalPortChans[1] = localPortBase+1; mRemotePortChans[0] = remotePortBase; mRemotePortChans[1] = remotePortBase+1; mRtpFactor = mRtpClock; mBaseTimeReal = tvNow(); mBaseTimeNtp = tvZero(); mBaseTimeRtp = rtpTime; mLastSrTimeReal = tvZero(); mLastSrTimeNtp = tvZero(); mLastSrTimeRtp = 0; if(mCodecId != AV_CODEC_ID_H264 && mCodecId != AV_CODEC_ID_MPEG4) Warning( "The device is using a codec that may not be supported. Do not be surprised if things don't work." ); } void RtpSource::init( uint16_t seq ) { Debug( 3, "Initialising sequence" ); mBaseSeq = seq; mMaxSeq = seq; mBadSeq = RTP_SEQ_MOD + 1; // so seq == mBadSeq is false mCycles = 0; mReceivedPackets = 0; mReceivedPrior = 0; mExpectedPrior = 0; // other initialization mJitter = 0; mTransit = 0; } bool RtpSource::updateSeq( uint16_t seq ) { uint16_t uDelta = seq - mMaxSeq; // Source is not valid until MIN_SEQUENTIAL packets with // sequential sequence numbers have been received. Debug( 5, "Seq: %d", seq ); if ( mProbation) { // packet is in sequence if ( seq == mMaxSeq + 1) { Debug( 3, "Sequence in probation %d, in sequence", mProbation ); mProbation--; mMaxSeq = seq; if ( mProbation == 0 ) { init( seq ); mReceivedPackets++; return( true ); } } else { Warning( "Sequence in probation %d, out of sequence", mProbation ); mProbation = MIN_SEQUENTIAL - 1; mMaxSeq = seq; return( false ); } return( true ); } else if ( uDelta < MAX_DROPOUT ) { if ( uDelta == 1 ) { Debug( 4, "Packet in sequence, gap %d", uDelta ); } else { Warning( "Packet in sequence, gap %d", uDelta ); } // in order, with permissible gap if ( seq < mMaxSeq ) { // Sequence number wrapped - count another 64K cycle. mCycles += RTP_SEQ_MOD; } mMaxSeq = seq; } else if ( uDelta <= RTP_SEQ_MOD - MAX_MISORDER ) { Warning( "Packet out of sequence, gap %d", uDelta ); // the sequence number made a very large jump if ( seq == mBadSeq ) { Debug( 3, "Restarting sequence" ); // Two sequential packets -- assume that the other side // restarted without telling us so just re-sync // (i.e., pretend this was the first packet). init( seq ); } else { mBadSeq = (seq + 1) & (RTP_SEQ_MOD-1); return( false ); } } else { Warning( "Packet duplicate or reordered, gap %d", uDelta ); // duplicate or reordered packet return( false ); } mReceivedPackets++; return( uDelta==1?true:false ); } void RtpSource::updateJitter( const RtpDataHeader *header ) { if ( mRtpFactor > 0 ) { Debug( 5, "Delta rtp = %.6f", tvDiffSec( mBaseTimeReal ) ); uint32_t localTimeRtp = mBaseTimeRtp + uint32_t( tvDiffSec( mBaseTimeReal ) * mRtpFactor ); Debug( 5, "Local RTP time = %x", localTimeRtp ); Debug( 5, "Packet RTP time = %x", ntohl(header->timestampN) ); uint32_t packetTransit = localTimeRtp - ntohl(header->timestampN); Debug( 5, "Packet transit RTP time = %x", packetTransit ); if ( mTransit > 0 ) { // Jitter int d = packetTransit - mTransit; Debug( 5, "Jitter D = %d", d ); if ( d < 0 ) d = -d; //mJitter += (1./16.) * ((double)d - mJitter); mJitter += d - ((mJitter + 8) >> 4); } mTransit = packetTransit; } else { mJitter = 0; } Debug( 5, "RTP Jitter: %d", mJitter ); } void RtpSource::updateRtcpData( uint32_t ntpTimeSecs, uint32_t ntpTimeFrac, uint32_t rtpTime ) { struct timeval ntpTime = tvMake( ntpTimeSecs, suseconds_t((USEC_PER_SEC*(ntpTimeFrac>>16))/(1<<16)) ); Debug( 5, "ntpTime: %ld.%06ld, rtpTime: %x", ntpTime.tv_sec, ntpTime.tv_usec, rtpTime ); if ( mBaseTimeNtp.tv_sec == 0 ) { mBaseTimeReal = tvNow(); mBaseTimeNtp = ntpTime; mBaseTimeRtp = rtpTime; } else if ( !mRtpClock ) { Debug( 5, "lastSrNtpTime: %ld.%06ld, rtpTime: %x", mLastSrTimeNtp.tv_sec, mLastSrTimeNtp.tv_usec, rtpTime ); Debug( 5, "ntpTime: %ld.%06ld, rtpTime: %x", ntpTime.tv_sec, ntpTime.tv_usec, rtpTime ); double diffNtpTime = tvDiffSec( mBaseTimeNtp, ntpTime ); uint32_t diffRtpTime = rtpTime - mBaseTimeRtp; //Debug( 5, "Real-diff: %.6f", diffRealTime ); Debug( 5, "NTP-diff: %.6f", diffNtpTime ); Debug( 5, "RTP-diff: %d", diffRtpTime ); mRtpFactor = (uint32_t)(diffRtpTime / diffNtpTime); Debug( 5, "RTPfactor: %d", mRtpFactor ); } mLastSrTimeNtpSecs = ntpTimeSecs; mLastSrTimeNtpFrac = ntpTimeFrac; mLastSrTimeNtp = ntpTime; mLastSrTimeRtp = rtpTime; } void RtpSource::updateRtcpStats() { uint32_t extendedMax = mCycles + mMaxSeq; mExpectedPackets = extendedMax - mBaseSeq + 1; Debug( 5, "Expected packets = %d", mExpectedPackets ); // The number of packets lost is defined to be the number of packets // expected less the number of packets actually received: mLostPackets = mExpectedPackets - mReceivedPackets; Debug( 5, "Lost packets = %d", mLostPackets ); uint32_t expectedInterval = mExpectedPackets - mExpectedPrior; Debug( 5, "Expected interval = %d", expectedInterval ); mExpectedPrior = mExpectedPackets; uint32_t receivedInterval = mReceivedPackets - mReceivedPrior; Debug( 5, "Received interval = %d", receivedInterval ); mReceivedPrior = mReceivedPackets; uint32_t lostInterval = expectedInterval - receivedInterval; Debug( 5, "Lost interval = %d", lostInterval ); if ( expectedInterval == 0 || lostInterval <= 0 ) mLostFraction = 0; else mLostFraction = (lostInterval << 8) / expectedInterval; Debug( 5, "Lost fraction = %d", mLostFraction ); } bool RtpSource::handlePacket( const unsigned char *packet, size_t packetLen ) { const RtpDataHeader *rtpHeader; rtpHeader = (RtpDataHeader *)packet; int rtpHeaderSize = 12 + rtpHeader->cc * 4; // No need to check for nal type as non fragmented packets already have 001 start sequence appended bool h264FragmentEnd = (mCodecId == AV_CODEC_ID_H264) && (packet[rtpHeaderSize+1] & 0x40); // M stands for Marker, it is the 8th bit // The interpretation of the marker is defined by a profile. It is intended // to allow significant events such as frame boundaries to be marked in the // packet stream. A profile may define additional marker bits or specify // that there is no marker bit by changing the number of bits in the payload type field. bool thisM = rtpHeader->m || h264FragmentEnd; if ( updateSeq( ntohs(rtpHeader->seqN) ) ) { Hexdump( 4, packet+rtpHeaderSize, 16 ); if ( mFrameGood ) { int extraHeader = 0; if( mCodecId == AV_CODEC_ID_H264 ) { int nalType = (packet[rtpHeaderSize] & 0x1f); Debug( 3, "Have H264 frame: nal type is %d", nalType ); switch (nalType) { case 24: // STAP-A { extraHeader = 2; break; } case 25: // STAP-B case 26: // MTAP-16 case 27: // MTAP-24 { extraHeader = 3; break; } // FU-A and FU-B case 28: case 29: { // Is this NAL the first NAL in fragmentation sequence if ( packet[rtpHeaderSize+1] & 0x80 ) { // Now we will form new header of frame mFrame.append( "\x0\x0\x1\x0", 4 ); // Reconstruct NAL header from FU headers *(mFrame+3) = (packet[rtpHeaderSize+1] & 0x1f) | (packet[rtpHeaderSize] & 0xe0); } extraHeader = 2; break; } default: { Debug(3, "Unhandled nalType %d", nalType ); } } // Append NAL frame start code if ( !mFrame.size() ) mFrame.append( "\x0\x0\x1", 3 ); } mFrame.append( packet+rtpHeaderSize+extraHeader, packetLen-rtpHeaderSize-extraHeader ); } else { Debug( 3, "NOT H264 frame: type is %d", mCodecId ); } Hexdump( 4, mFrame.head(), 16 ); if ( thisM ) { if ( mFrameGood ) { Debug( 3, "Got new frame %d, %d bytes", mFrameCount, mFrame.size() ); mFrameProcessed.setValueImmediate( false ); mFrameReady.updateValueSignal( true ); if ( !mFrameProcessed.getValueImmediate() ) { // What is the point of this for loop? Is it just me, or will it call getUpdatedValue once or twice? Could it not be better written as // if ( ! mFrameProcessed.getUpdatedValue( 1 ) && mFrameProcessed.getUpdatedValue( 1 ) ) return false; for ( int count = 0; !mFrameProcessed.getUpdatedValue( 1 ); count++ ) if( count > 1 ) return( false ); } mFrameCount++; } else { Warning( "Discarding incomplete frame %d, %d bytes", mFrameCount, mFrame.size() ); } mFrame.clear(); } } else { if ( mFrame.size() ) { Warning( "Discarding partial frame %d, %d bytes", mFrameCount, mFrame.size() ); } else { Warning( "Discarding frame %d", mFrameCount ); } mFrameGood = false; mFrame.clear(); } if ( thisM ) { mFrameGood = true; prevM = true; } else prevM = false; updateJitter( rtpHeader ); return( true ); } bool RtpSource::getFrame( Buffer &buffer ) { Debug( 3, "Getting frame" ); if ( !mFrameReady.getValueImmediate() ) { // Allow for a couple of spurious returns for ( int count = 0; !mFrameReady.getUpdatedValue( 1 ); count++ ) if ( count > 1 ) return( false ); } buffer = mFrame; mFrameReady.setValueImmediate( false ); mFrameProcessed.updateValueSignal( true ); Debug( 4, "Copied %d bytes", buffer.size() ); return( true ); } #endif // HAVE_LIBAVCODEC