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rtsp播放器支持rtcp

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This commit is contained in:
xiongziliang 2019-05-09 13:35:54 +08:00
parent 9d92f613e2
commit 304275a0e4
7 changed files with 174 additions and 40 deletions
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13
src/Rtsp/RtpReceiver.cpp
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@ -116,11 +116,12 @@ bool RtpReceiver::handleOneRtp(int iTrackidx,SdpTrack::Ptr &track, unsigned char
_aui32SeqOkCnt[iTrackidx] = 0;
_abSortStarted[iTrackidx] = true;
// WarnL << "包乱序或丢包:" << iTrackidx <<" " << rtppt.sequence << " " << _aui16LastSeq[iTrackidx];
if(_aui16LastSeq[iTrackidx] > rtppt.sequence && _aui16LastSeq[iTrackidx] - rtppt.sequence > 0x7FFF){
if(_aui16LastSeq[iTrackidx] > rtppt.sequence && _aui16LastSeq[iTrackidx] - rtppt.sequence > 0xFF){
//sequence回环，清空所有排序缓存
while (_amapRtpSort[iTrackidx].size()) {
POP_HEAD(iTrackidx)
}
++_clcyeCount[iTrackidx];
}
}else{
//正确序列的包
@ -157,6 +158,7 @@ void RtpReceiver::clear() {
CLEAR_ARR(_aui32SsrcErrorCnt)
CLEAR_ARR(_aui32SeqOkCnt)
CLEAR_ARR(_abSortStarted)
CLEAR_ARR(_clcyeCount)
_amapRtpSort[0].clear();
_amapRtpSort[1].clear();
@ -166,4 +168,13 @@ void RtpReceiver::setPoolSize(int size) {
_pktPool.setSize(size);
}
int RtpReceiver::getJitterSize(int iTrackidx){
return _amapRtpSort[iTrackidx].size();
}
int RtpReceiver::getCycleCount(int iTrackidx){
return _clcyeCount[iTrackidx];
}
}//namespace mediakit

3
src/Rtsp/RtpReceiver.h
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@ -64,11 +64,14 @@ protected:
virtual void onRtpSorted(const RtpPacket::Ptr &rtppt, int trackidx){}
void clear();
void setPoolSize(int size);
int getJitterSize(int iTrackidx);
int getCycleCount(int iTrackidx);
private:
uint32_t _aui32SsrcErrorCnt[2] = { 0, 0 };
/* RTP包排序所用参数 */
uint16_t _aui16LastSeq[2] = { 0 , 0 };
uint32_t _aui32SeqOkCnt[2] = { 0 , 0};
uint32_t _clcyeCount[2] = { 0 , 0};
bool _abSortStarted[2] = { 0 , 0};
map<uint16_t , RtpPacket::Ptr> _amapRtpSort[2];
RtspMediaSource::PoolType _pktPool;

1
src/Rtsp/Rtsp.h
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@ -113,6 +113,7 @@ public:
uint32_t octCount = 0;
//网络字节序
uint32_t timeStamp = 0;
uint32_t lastTimeStamp = 0;
};
string FindField(const char *buf, const char *start, const char *end, int bufSize = 0);

163
src/Rtsp/RtspPlayer.cpp
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@ -71,7 +71,6 @@ void RtspPlayer::teardown(){
CLEAR_ARR(_aiFistStamp);
CLEAR_ARR(_aiNowStamp);
_pBeatTimer.reset();
_pPlayTimer.reset();
_pRtpTimer.reset();
_iSeekTo = 0;
@ -369,17 +368,6 @@ void RtspPlayer::handleResSETUP(const Parser &parser, unsigned int uiTrackIndex)
return;
}
//所有setup命令发送完毕
//设置心跳包发送定时器
weak_ptr<RtspPlayer> weakSelf = dynamic_pointer_cast<RtspPlayer>(shared_from_this());
_pBeatTimer.reset(new Timer((*this)[kBeatIntervalMS].as<int>() / 1000.0, [weakSelf](){
auto strongSelf = weakSelf.lock();
if (!strongSelf){
return false;
}
strongSelf->sendRtcpPacket();
return true;
},getPoller()));
//发送play命令
pause(false);
}
@ -491,11 +479,132 @@ void RtspPlayer::onRtcpPacket(int iTrackidx, SdpTrack::Ptr &track, unsigned char
}
void RtspPlayer::sendRtcpPacket(){
//目前只实现了通过options命令实现心跳包
sendOptions();
#if 0
//改代码提取自FFmpeg，参考之
// Receiver Report
avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
avio_w8(pb, RTCP_RR);
avio_wb16(pb, 7); /* length in words - 1 */
// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
avio_wb32(pb, s->ssrc + 1);
avio_wb32(pb, s->ssrc); // server SSRC
// some placeholders we should really fill...
// RFC 1889/p64
extended_max = stats->cycles + stats->max_seq;
expected = extended_max - stats->base_seq;
lost = expected - stats->received;
lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
expected_interval = expected - stats->expected_prior;
stats->expected_prior = expected;
received_interval = stats->received - stats->received_prior;
stats->received_prior = stats->received;
lost_interval = expected_interval - received_interval;
if (expected_interval == 0 || lost_interval <= 0)
fraction = 0;
else
fraction = (lost_interval << 8) / expected_interval;
fraction = (fraction << 24) | lost;
avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
avio_wb32(pb, extended_max); /* max sequence received */
avio_wb32(pb, stats->jitter >> 4); /* jitter */
if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
avio_wb32(pb, 0); /* last SR timestamp */
avio_wb32(pb, 0); /* delay since last SR */
} else {
uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time,
65536, AV_TIME_BASE);
avio_wb32(pb, middle_32_bits); /* last SR timestamp */
avio_wb32(pb, delay_since_last); /* delay since last SR */
}
// CNAME
avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
avio_w8(pb, RTCP_SDES);
len = strlen(s->hostname);
avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
avio_wb32(pb, s->ssrc + 1);
avio_w8(pb, 0x01);
avio_w8(pb, len);
avio_write(pb, s->hostname, len);
avio_w8(pb, 0); /* END */
// padding
for (len = (7 + len) % 4; len % 4; len++)
avio_w8(pb, 0);
#endif
void RtspPlayer::sendReceiverReport(bool overTcp,int iTrackIndex){
static const char s_cname[] = "ZLMediaKitRtsp";
uint8_t aui8Rtcp[4 + 32 + 10 + sizeof(s_cname) + 1] = {0};
uint8_t *pui8Rtcp_RR = aui8Rtcp + 4, *pui8Rtcp_SDES = pui8Rtcp_RR + 32;
auto &track = _aTrackInfo[iTrackIndex];
auto &counter = _aRtcpCnt[iTrackIndex];
aui8Rtcp[0] = '$';
aui8Rtcp[1] = track->_interleaved + 1;
aui8Rtcp[2] = (sizeof(aui8Rtcp) - 4) >> 8;
aui8Rtcp[3] = (sizeof(aui8Rtcp) - 4) & 0xFF;
pui8Rtcp_RR[0] = 0x81;/* 1 report block */
pui8Rtcp_RR[1] = 0xC9;//RTCP_RR
pui8Rtcp_RR[2] = 0x00;
pui8Rtcp_RR[3] = 0x07;/* length in words - 1 */
uint32_t ssrc=htonl(track->_ssrc + 1);
// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
memcpy(&pui8Rtcp_RR[4], &ssrc, 4);
ssrc=htonl(track->_ssrc);
// server SSRC
memcpy(&pui8Rtcp_RR[8], &ssrc, 4);
//FIXME: 8 bits of fraction, 24 bits of total packets lost
pui8Rtcp_RR[12] = 0x00;
pui8Rtcp_RR[13] = 0x00;
pui8Rtcp_RR[14] = 0x00;
pui8Rtcp_RR[15] = 0x00;
//FIXME: max sequence received
int cycleCount = getCycleCount(iTrackIndex);
pui8Rtcp_RR[16] = cycleCount >> 8;
pui8Rtcp_RR[17] = cycleCount & 0xFF;
pui8Rtcp_RR[18] = counter.pktCnt >> 8;
pui8Rtcp_RR[19] = counter.pktCnt & 0xFF;
uint32_t jitter = htonl(getJitterSize(iTrackIndex));
//FIXME: jitter
memcpy(pui8Rtcp_RR + 20, &jitter , 4);
/* last SR timestamp */
memcpy(pui8Rtcp_RR + 24, &counter.lastTimeStamp, 4);
uint32_t msInc = htonl(ntohl(counter.timeStamp) - ntohl(counter.lastTimeStamp));
/* delay since last SR */
memcpy(pui8Rtcp_RR + 28, &msInc, 4);
// CNAME
pui8Rtcp_SDES[0] = 0x81;
pui8Rtcp_SDES[1] = 0xCA;
pui8Rtcp_SDES[2] = 0x00;
pui8Rtcp_SDES[3] = 0x06;
memcpy(&pui8Rtcp_SDES[4], &ssrc, 4);
pui8Rtcp_SDES[8] = 0x01;
pui8Rtcp_SDES[9] = 0x0f;
memcpy(&pui8Rtcp_SDES[10], s_cname, sizeof(s_cname));
pui8Rtcp_SDES[10 + sizeof(s_cname)] = 0x00;
if(overTcp){
send(obtainBuffer((char *) aui8Rtcp, sizeof(aui8Rtcp)));
}else if(_apRtcpSock[iTrackIndex]) {
_apRtcpSock[iTrackIndex]->send((char *) aui8Rtcp + 4, sizeof(aui8Rtcp) - 4);
}
}
void RtspPlayer::onRtpSorted(const RtpPacket::Ptr &rtppt, int trackidx){
//统计丢包率
if (_aui16FirstSeq[trackidx] == 0 || rtppt->sequence < _aui16FirstSeq[trackidx]) {
@ -608,9 +717,29 @@ void RtspPlayer::sendRtspRequest(const string &cmd, const string &url,const StrC
send(printer << "\r\n");
}
void RtspPlayer::onRecvRTP_l(const RtpPacket::Ptr &pRtppt, const SdpTrack::Ptr &track) {
void RtspPlayer::onRecvRTP_l(const RtpPacket::Ptr &pkt, const SdpTrack::Ptr &track) {
_rtpTicker.resetTime();
onRecvRTP(pRtppt,track);
onRecvRTP(pkt,track);
int iTrackIndex = getTrackIndexByInterleaved(pkt->interleaved);
if(iTrackIndex == -1){
return;
}
RtcpCounter &counter = _aRtcpCnt[iTrackIndex];
counter.pktCnt = pkt->sequence;
auto &ticker = _aRtcpTicker[iTrackIndex];
if (ticker.elapsedTime() > 5 * 1000) {
//send rtcp every 5 second
counter.lastTimeStamp = counter.timeStamp;
//直接保存网络字节序
memcpy(&counter.timeStamp, pkt->payload + 8 , 4);
if(counter.lastTimeStamp != 0){
sendReceiverReport(_eType == Rtsp::RTP_TCP,iTrackIndex);
ticker.resetTime();
}
}
}
void RtspPlayer::onPlayResult_l(const SockException &ex) {
WarnL << ex.getErrCode() << " " << ex.what();

12
src/Rtsp/RtspPlayer.h
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@ -94,11 +94,6 @@ protected:
* @param uiLen
*/
virtual void onRtcpPacket(int iTrackidx, SdpTrack::Ptr &track, unsigned char *pucData, unsigned int uiLen);
/**
* 发送rtcp包维持心跳
*/
virtual void sendRtcpPacket();
private:
void onRecvRTP_l(const RtpPacket::Ptr &pRtppt, const SdpTrack::Ptr &track);
void onPlayResult_l(const SockException &ex);
@ -125,6 +120,7 @@ private:
void sendRtspRequest(const string &cmd, const string &url ,const StrCaseMap &header = StrCaseMap());
void sendRtspRequest(const string &cmd, const string &url ,const std::initializer_list<string> &header);
void sendReceiverReport(bool overTcp,int iTrackIndex);
private:
string _strUrl;
SdpAttr _sdpAttr;
@ -151,8 +147,6 @@ private:
Ticker _rtpTicker;
std::shared_ptr<Timer> _pPlayTimer;
std::shared_ptr<Timer> _pRtpTimer;
//心跳定时器
std::shared_ptr<Timer> _pBeatTimer;
//播放进度控制,单位毫秒
uint32_t _iSeekTo = 0;
@ -160,6 +154,10 @@ private:
//单位毫秒
uint32_t _aiFistStamp[2] = {0,0};
uint32_t _aiNowStamp[2] = {0,0};
//rtcp相关
RtcpCounter _aRtcpCnt[2]; //rtcp统计,trackid idx 为数组下标
Ticker _aRtcpTicker[2]; //rtcp发送时间,trackid idx 为数组下标
};
} /* namespace mediakit */

16
src/Rtsp/RtspSession.cpp
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@ -1190,7 +1190,6 @@ inline void RtspSession::sendRtpPacket(const RtpPacket::Ptr & pkt) {
break;
}
#ifdef RTSP_SEND_RTCP
int iTrackIndex = getTrackIndexByInterleaved(pkt->interleaved);
if(iTrackIndex == -1){
return;
@ -1206,21 +1205,19 @@ inline void RtspSession::sendRtpPacket(const RtpPacket::Ptr & pkt) {
memcpy(&counter.timeStamp, pkt->payload + 8 , 4);
sendSenderReport(_rtpType == Rtsp::RTP_TCP,iTrackIndex);
}
#endif
}
#ifdef RTSP_SEND_RTCP
inline void RtspSession::sendSenderReport(bool overTcp,int iTrackIndex) {
static const char server_name[] = "ZLMediaKitRtsp";
uint8_t aui8Rtcp[4 + 28 + 10 + sizeof(server_name) + 1] = {0};
static const char s_cname[] = "ZLMediaKitRtsp";
uint8_t aui8Rtcp[4 + 28 + 10 + sizeof(s_cname) + 1] = {0};
uint8_t *pui8Rtcp_SR = aui8Rtcp + 4, *pui8Rtcp_SDES = pui8Rtcp_SR + 28;
auto &track = _aTrackInfo[iTrackIndex];
auto &counter = _aRtcpCnt[iTrackIndex];
aui8Rtcp[0] = '$';
aui8Rtcp[1] = track->_interleaved + 1;
aui8Rtcp[2] = (sizeof(aui8Rtcp) - 4) / 256;
aui8Rtcp[3] = (sizeof(aui8Rtcp) - 4) % 256;
aui8Rtcp[2] = (sizeof(aui8Rtcp) - 4) >> 8;
aui8Rtcp[3] = (sizeof(aui8Rtcp) - 4) & 0xFF;
pui8Rtcp_SR[0] = 0x80;
pui8Rtcp_SR[1] = 0xC8;
@ -1260,8 +1257,8 @@ inline void RtspSession::sendSenderReport(bool overTcp,int iTrackIndex) {
pui8Rtcp_SDES[8] = 0x01;
pui8Rtcp_SDES[9] = 0x0f;
memcpy(&pui8Rtcp_SDES[10], server_name, sizeof(server_name));
pui8Rtcp_SDES[10 + sizeof(server_name)] = 0x00;
memcpy(&pui8Rtcp_SDES[10], s_cname, sizeof(s_cname));
pui8Rtcp_SDES[10 + sizeof(s_cname)] = 0x00;
if(overTcp){
send(obtainBuffer((char *) aui8Rtcp, sizeof(aui8Rtcp)));
@ -1269,7 +1266,6 @@ inline void RtspSession::sendSenderReport(bool overTcp,int iTrackIndex) {
_apRtcpSock[iTrackIndex]->send((char *) aui8Rtcp + 4, sizeof(aui8Rtcp) - 4);
}
}
#endif
}
/* namespace mediakit */

4
src/Rtsp/RtspSession.h
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@ -46,8 +46,6 @@
using namespace std;
using namespace toolkit;
#define RTSP_SEND_RTCP
namespace mediakit {
class RtspSession;
@ -200,11 +198,9 @@ private:
//rtsp推流相关
RtspToRtmpMediaSource::Ptr _pushSrc;
#ifdef RTSP_SEND_RTCP
RtcpCounter _aRtcpCnt[2]; //rtcp统计,trackid idx 为数组下标
Ticker _aRtcpTicker[2]; //rtcp发送时间,trackid idx 为数组下标
inline void sendSenderReport(bool overTcp,int iTrackIndex);
#endif
};
/**