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