zoneminder/src/zm_rtp_source.cpp

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//
// 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 <arpa/inet.h>
#if HAVE_LIBAVCODEC
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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)
{
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char hostname[256] = "";
gethostname(hostname, sizeof(hostname));
mCname = stringtf("zm-%d@%s", mId, hostname);
Debug(3, "RTP CName = %s", mCname.c_str());
init(seq);
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mMaxSeq = seq - 1;
mProbation = MIN_SEQUENTIAL;
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mLocalPortChans[0] = localPortBase;
mLocalPortChans[1] = localPortBase+1;
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mRemotePortChans[0] = remotePortBase;
mRemotePortChans[1] = remotePortBase+1;
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mRtpFactor = mRtpClock;
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mBaseTimeReal = tvNow();
mBaseTimeNtp = tvZero();
mBaseTimeRtp = rtpTime;
mLastSrTimeReal = tvZero();
mLastSrTimeNtp = tvZero();
mLastSrTimeRtp = 0;
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if ( mCodecId != AV_CODEC_ID_H264 && mCodecId != AV_CODEC_ID_MPEG4 )
Warning("The device is using a codec (%d) that may not be supported. Do not be surprised if things don't work.", mCodecId);
}
void RtpSource::init(uint16_t seq) {
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Debug(3, "Initialising sequence");
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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) {
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uint16_t uDelta = seq - mMaxSeq;
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// Source is not valid until MIN_SEQUENTIAL packets with
// sequential sequence numbers have been received.
Debug(5, "Seq: %d", seq);
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if ( mProbation) {
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// packet is in sequence
if ( seq == mMaxSeq + 1 ) {
Debug(3, "Sequence in probation %d, in sequence", mProbation);
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mProbation--;
mMaxSeq = seq;
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if ( mProbation == 0 ) {
init(seq);
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mReceivedPackets++;
return true;
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}
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} else {
Warning("Sequence in probation %d, out of sequence", mProbation);
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mProbation = MIN_SEQUENTIAL - 1;
mMaxSeq = seq;
return false;
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}
return true;
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} else if ( uDelta < MAX_DROPOUT ) {
if ( uDelta == 1 ) {
Debug(4, "Packet in sequence, gap %d", uDelta);
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} else {
Warning("Packet in sequence, gap %d", uDelta);
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}
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// in order, with permissible gap
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if ( seq < mMaxSeq ) {
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// Sequence number wrapped - count another 64K cycle.
mCycles += RTP_SEQ_MOD;
}
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mMaxSeq = seq;
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} else if ( uDelta <= RTP_SEQ_MOD - MAX_MISORDER ) {
Warning("Packet out of sequence, gap %d", uDelta);
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// the sequence number made a very large jump
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if ( seq == mBadSeq ) {
Debug(3, "Restarting sequence");
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// 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);
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} else {
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mBadSeq = (seq + 1) & (RTP_SEQ_MOD-1);
return false;
}
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} else {
Warning("Packet duplicate or reordered, gap %d", uDelta);
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// duplicate or reordered packet
return false;
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}
mReceivedPackets++;
return( uDelta==1?true:false );
}
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void RtpSource::updateJitter( const RtpDataHeader *header ) {
if ( mRtpFactor > 0 ) {
uint32_t localTimeRtp = mBaseTimeRtp + uint32_t(tvDiffSec(mBaseTimeReal) * mRtpFactor);
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uint32_t packetTransit = localTimeRtp - ntohl(header->timestampN);
Debug(5, "Delta rtp = %.6f\n"
"Local RTP time = %x",
"Packet RTP time = %x",
"Packet transit RTP time = %x",
tvDiffSec(mBaseTimeReal),
localTimeRtp,
ntohl(header->timestampN),
packetTransit);
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if ( mTransit > 0 ) {
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// Jitter
int d = packetTransit - mTransit;
Debug(5, "Jitter D = %d", d);
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if ( d < 0 )
d = -d;
//mJitter += (1./16.) * ((double)d - mJitter);
mJitter += d - ((mJitter + 8) >> 4);
}
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mTransit = packetTransit;
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} else {
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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);
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if ( mBaseTimeNtp.tv_sec == 0 ) {
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mBaseTimeReal = tvNow();
mBaseTimeNtp = ntpTime;
mBaseTimeRtp = rtpTime;
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} else if ( !mRtpClock ) {
Debug(5, "lastSrNtpTime: %ld.%06ld, rtpTime: %x"
"ntpTime: %ld.%06ld, rtpTime: %x",
mLastSrTimeNtp.tv_sec, mLastSrTimeNtp.tv_usec, rtpTime,
ntpTime.tv_sec, ntpTime.tv_usec, rtpTime);
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double diffNtpTime = tvDiffSec( mBaseTimeNtp, ntpTime );
uint32_t diffRtpTime = rtpTime - mBaseTimeRtp;
mRtpFactor = (uint32_t)(diffRtpTime / diffNtpTime);
Debug( 5, "NTP-diff: %.6f RTP-diff: %d RTPfactor: %d",
diffNtpTime, diffRtpTime, mRtpFactor);
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}
mLastSrTimeNtpSecs = ntpTimeSecs;
mLastSrTimeNtpFrac = ntpTimeFrac;
mLastSrTimeNtp = ntpTime;
mLastSrTimeRtp = rtpTime;
}
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void RtpSource::updateRtcpStats() {
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uint32_t extendedMax = mCycles + mMaxSeq;
mExpectedPackets = extendedMax - mBaseSeq + 1;
// The number of packets lost is defined to be the number of packets
// expected less the number of packets actually received:
mLostPackets = mExpectedPackets - mReceivedPackets;
uint32_t expectedInterval = mExpectedPackets - mExpectedPrior;
mExpectedPrior = mExpectedPackets;
uint32_t receivedInterval = mReceivedPackets - mReceivedPrior;
mReceivedPrior = mReceivedPackets;
uint32_t lostInterval = expectedInterval - receivedInterval;
if ( expectedInterval == 0 || lostInterval <= 0 )
mLostFraction = 0;
else
mLostFraction = (lostInterval << 8) / expectedInterval;
Debug(5,
"Expected packets = %d\n",
"Lost packets = %d\n",
"Expected interval = %d\n",
"Received interval = %d\n",
"Lost interval = %d\n",
"Lost fraction = %d\n",
mExpectedPackets,
mLostPackets,
expectedInterval,
receivedInterval,
lostInterval,
mLostFraction);
}
bool RtpSource::handlePacket(const unsigned char *packet, size_t packetLen) {
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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);
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if ( mFrameGood ) {
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int extraHeader = 0;
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if ( mCodecId == AV_CODEC_ID_H264 ) {
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int nalType = (packet[rtpHeaderSize] & 0x1f);
Debug(3, "Have H264 frame: nal type is %d", nalType);
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switch (nalType) {
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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
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if ( packet[rtpHeaderSize+1] & 0x80 ) {
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// 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;
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default:
Debug(3, "Unhandled nalType %d", nalType);
}
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// Append NAL frame start code
if ( !mFrame.size() )
mFrame.append("\x0\x0\x1", 3);
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} // end if H264
mFrame.append(packet+rtpHeaderSize+extraHeader,
packetLen-rtpHeaderSize-extraHeader);
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} else {
Debug(3, "NOT H264 frame: type is %d", mCodecId);
}
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Hexdump(4, mFrame.head(), 16);
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if ( thisM ) {
if ( mFrameGood ) {
Debug(3, "Got new frame %d, %d bytes", mFrameCount, mFrame.size());
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mFrameProcessed.setValueImmediate(false);
mFrameReady.updateValueSignal(true);
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if ( !mFrameProcessed.getValueImmediate() ) {
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// 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;
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for ( int count = 0; !mFrameProcessed.getUpdatedValue(1); count++ )
if ( count > 1 )
return false;
}
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mFrameCount++;
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} else {
Warning("Discarding incomplete frame %d, %d bytes", mFrameCount, mFrame.size());
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}
mFrame.clear();
}
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} else {
if ( mFrame.size() ) {
Warning("Discarding partial frame %d, %d bytes", mFrameCount, mFrame.size());
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} else {
Warning("Discarding frame %d", mFrameCount);
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}
mFrameGood = false;
mFrame.clear();
}
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if ( thisM ) {
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mFrameGood = true;
prevM = true;
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} else
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prevM = false;
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updateJitter(rtpHeader);
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return true;
}
bool RtpSource::getFrame(Buffer &buffer) {
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if ( !mFrameReady.getValueImmediate() ) {
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Debug(3, "Getting frame but not ready");
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// Allow for a couple of spurious returns
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for ( int count = 0; !mFrameReady.getUpdatedValue(1); count++ ) {
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if ( count > 1 )
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return false;
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}
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}
buffer = mFrame;
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mFrameReady.setValueImmediate(false);
mFrameProcessed.updateValueSignal(true);
Debug(4, "Copied %d bytes", buffer.size());
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return true;
}
#endif // HAVE_LIBAVCODEC