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重写rtcp框架

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This commit is contained in:
xiongziliang
2021-01-31 19:18:17 +08:00
parent 629c39685b
commit 5c6560f55d
16 changed files with 1617 additions and 337 deletions
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261
src/Rtsp/RtspPlayer.cpp
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@@ -15,9 +15,8 @@
#include "Common/config.h"
#include "RtspPlayer.h"
#include "Util/MD5.h"
#include "Util/util.h"
#include "Util/base64.h"
#include "Network/sockutil.h"
#include "Rtcp/Rtcp.h"
using namespace toolkit;
using namespace mediakit::Client;
@@ -53,13 +52,10 @@ void RtspPlayer::teardown(){
_session_id.clear();
_content_base.clear();
RtpReceiver::clear();
_rtcp_context.clear();
CLEAR_ARR(_rtp_sock);
CLEAR_ARR(_rtcp_sock);
CLEAR_ARR(_rtp_seq_start)
CLEAR_ARR(_rtp_recv_count)
CLEAR_ARR(_rtp_recv_count)
CLEAR_ARR(_rtp_seq_now)
_play_check_timer.reset();
_rtp_check_timer.reset();
@@ -119,7 +115,12 @@ void RtspPlayer::onRecv(const Buffer::Ptr& buf) {
_rtp_recv_ticker.resetTime();
return;
}
input(buf->data(), buf->size());
try {
input(buf->data(), buf->size());
} catch (exception &e) {
SockException ex(Err_other, e.what());
onPlayResult_l(ex, !_play_check_timer);
}
}
void RtspPlayer::onErr(const SockException &ex) {
@@ -196,15 +197,15 @@ void RtspPlayer::handleResDESCRIBE(const Parser& parser) {
SdpParser sdpParser(parser.Content());
//解析sdp
_sdp_track = sdpParser.getAvailableTrack();
auto title = sdpParser.getTrack(TrackTitle);
if (_sdp_track.empty()) {
throw std::runtime_error("无有效的Sdp Track");
}
if (!onCheckSDP(sdpParser.toString())) {
throw std::runtime_error("onCheckSDP faied");
}
for (auto &track : _sdp_track) {
_rtcp_context.emplace_back(std::make_shared<RtcpContext>(track->_samplerate));
}
sendSetup(0);
}
@@ -331,7 +332,7 @@ void RtspPlayer::handleResSETUP(const Parser &parser, unsigned int track_idx) {
return;
}
strongSelf->handleOneRtp(track_idx, strongSelf->_sdp_track[track_idx]->_type,
strongSelf->_sdp_track[track_idx]->_samplerate, (unsigned char *) buf->data(), buf->size());
strongSelf->_sdp_track[track_idx]->_samplerate, (uint8_t *) buf->data(), buf->size());
});
if(pRtcpSockRef) {
@@ -345,7 +346,7 @@ void RtspPlayer::handleResSETUP(const Parser &parser, unsigned int track_idx) {
WarnL << "收到其他地址的rtcp数据:" << SockUtil::inet_ntoa(((struct sockaddr_in *) addr)->sin_addr);
return;
}
strongSelf->onRtcpPacket(track_idx, strongSelf->_sdp_track[track_idx], (unsigned char *) buf->data(), buf->size());
strongSelf->onRtcpPacket(track_idx, strongSelf->_sdp_track[track_idx], (uint8_t *) buf->data(), buf->size());
});
}
}
@@ -477,174 +478,44 @@ void RtspPlayer::onRtpPacket(const char *data, size_t len) {
uint8_t interleaved = data[1];
if(interleaved %2 == 0){
trackIdx = getTrackIndexByInterleaved(interleaved);
handleOneRtp(trackIdx, _sdp_track[trackIdx]->_type, _sdp_track[trackIdx]->_samplerate, (unsigned char *)data + 4, len - 4);
handleOneRtp(trackIdx, _sdp_track[trackIdx]->_type, _sdp_track[trackIdx]->_samplerate, (uint8_t *)data + 4, len - 4);
}else{
trackIdx = getTrackIndexByInterleaved(interleaved - 1);
onRtcpPacket(trackIdx, _sdp_track[trackIdx], (unsigned char *) data + 4, len - 4);
onRtcpPacket(trackIdx, _sdp_track[trackIdx], (uint8_t *) data + 4, len - 4);
}
}
//此处预留rtcp处理函数
void RtspPlayer::onRtcpPacket(int track_idx, SdpTrack::Ptr &track, unsigned char *data, size_t len){}
#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 over_tcp, int track_idx){
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 = _sdp_track[track_idx];
auto &counter = _rtcp_counter[track_idx];
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 */
auto track_ssrc = track->_ssrc ? track->_ssrc : getSSRC(track_idx);
// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
uint32_t ssrc = htonl(track_ssrc + 1);
memcpy(&pui8Rtcp_RR[4], &ssrc, 4);
// server SSRC
ssrc = htonl(track_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
uint16_t cycleCount = (uint16_t)getCycleCount(track_idx);
pui8Rtcp_RR[16] = (cycleCount >> 8) & 0xFF;
pui8Rtcp_RR[17] = cycleCount & 0xFF;
pui8Rtcp_RR[18] = counter.pktCnt >> 8;
pui8Rtcp_RR[19] = counter.pktCnt & 0xFF;
uint32_t jitter = htonl((uint32_t)getJitterSize(track_idx));
//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 (over_tcp) {
send(obtainBuffer((char *) aui8Rtcp, sizeof(aui8Rtcp)));
} else if (_rtcp_sock[track_idx]) {
_rtcp_sock[track_idx]->send((char *) aui8Rtcp + 4, sizeof(aui8Rtcp) - 4);
void RtspPlayer::onRtcpPacket(int track_idx, SdpTrack::Ptr &track, uint8_t *data, size_t len){
auto rtcp_arr = RtcpHeader::loadFromBytes((char *) data, len);
for (auto &rtcp : rtcp_arr) {
_rtcp_context[track_idx]->onRtcp(rtcp);
}
}
void RtspPlayer::onRtpSorted(const RtpPacket::Ptr &rtppt, int trackidx){
//统计丢包率
if (_rtp_seq_start[trackidx] == 0 || rtppt->sequence < _rtp_seq_start[trackidx]) {
_rtp_seq_start[trackidx] = rtppt->sequence;
_rtp_recv_count[trackidx] = 0;
}
_rtp_recv_count[trackidx] ++;
_rtp_seq_now[trackidx] = rtppt->sequence;
_stamp[trackidx] = rtppt->timeStamp;
//计算相对时间戳
onRecvRTP_l(rtppt, _sdp_track[trackidx]);
_rtp_recv_ticker.resetTime();
onRecvRTP(rtppt, _sdp_track[trackidx]);
}
float RtspPlayer::getPacketLossRate(TrackType type) const{
size_t lost = 0, expected = 0;
try {
auto track_idx = getTrackIndexByTrackType(type);
if (_rtp_seq_now[track_idx] - _rtp_seq_start[track_idx] + 1 == 0) {
return 0;
}
return (float)(1.0f - (double) _rtp_recv_count[track_idx] / (_rtp_seq_now[track_idx] - _rtp_seq_start[track_idx] + 1));
auto ctx = _rtcp_context[track_idx];
lost = ctx->getLost();
expected = ctx->getExpectedPackets();
} catch (...) {
uint64_t totalRecv = 0;
uint64_t totalSend = 0;
for (unsigned int i = 0; i < _sdp_track.size(); i++) {
totalRecv += _rtp_recv_count[i];
totalSend += (_rtp_seq_now[i] - _rtp_seq_start[i] + 1);
for (auto &ctx : _rtcp_context) {
lost += ctx->getLost();
expected += ctx->getExpectedPackets();
}
if (totalSend == 0) {
return 0;
}
return (float)(1.0f - (double) totalRecv / totalSend);
}
if (!expected) {
return 0;
}
return (float) (double(lost) / double(expected));
}
uint32_t RtspPlayer::getProgressMilliSecond() const{
@@ -720,35 +591,51 @@ void RtspPlayer::sendRtspRequest(const string &cmd, const string &url,const StrC
SockSender::send(std::move(printer));
}
void RtspPlayer::onRecvRTP_l(const RtpPacket::Ptr &rtp, const SdpTrack::Ptr &track) {
_rtp_recv_ticker.resetTime();
onRecvRTP(rtp, track);
void RtspPlayer::onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_idx){
auto &rtcp_ctx = _rtcp_context[track_idx];
rtcp_ctx->onRtp(rtp->sequence, rtp->timeStamp, rtp->size() - 4);
int track_idx = getTrackIndexByTrackType(rtp->type);
RtcpCounter &counter = _rtcp_counter[track_idx];
counter.pktCnt = rtp->sequence;
auto &ticker = _rtcp_send_ticker[track_idx];
if (ticker.elapsedTime() > 3 * 1000) {
//每3秒发送一次心跳，rtcp与rtsp信令轮流心跳，该特性用于兼容issue:642
if (_send_rtcp) {
counter.lastTimeStamp = counter.timeStamp;
//直接保存网络字节序
memcpy(&counter.timeStamp, rtp->data() + 8, 4);
if (counter.lastTimeStamp != 0) {
sendReceiverReport(_rtp_type == Rtsp::RTP_TCP, track_idx);
ticker.resetTime();
_send_rtcp = false;
}
} else {
//有些rtsp服务器需要rtcp保活，有些需要发送信令保活
if (track_idx == 0) {
//只需要发送一次心跳信令包
sendKeepAlive();
ticker.resetTime();
_send_rtcp = true;
}
}
if (ticker.elapsedTime() < 3 * 1000) {
//时间未到
return;
}
auto &rtcp_flag = _send_rtcp[track_idx];
//每3秒发送一次心跳，rtcp与rtsp信令轮流心跳，该特性用于兼容issue:642
//有些rtsp服务器需要rtcp保活，有些需要发送信令保活
//发送信令保活
if (!rtcp_flag) {
if (track_idx == 0) {
sendKeepAlive();
}
ticker.resetTime();
//下次发送rtcp保活
rtcp_flag = true;
return;
}
//发送rtcp
static auto send_rtcp = [](RtspPlayer *thiz, int index, Buffer::Ptr ptr) {
if (thiz->_rtp_type == Rtsp::RTP_TCP) {
auto &track = thiz->_sdp_track[index];
thiz->send(makeRtpOverTcpPrefix((uint16_t) (ptr->size()), track->_interleaved + 1));
thiz->send(std::move(ptr));
} else {
thiz->_rtcp_sock[index]->send(std::move(ptr));
}
};
auto rtcp = rtcp_ctx->createRtcpRR(rtp->ssrc + 1, rtp->ssrc);
auto rtcp_sdes = RtcpSdes::create({SERVER_NAME});
rtcp_sdes->items.type = (uint8_t) SdesType::RTCP_SDES_CNAME;
rtcp_sdes->items.ssrc = htonl(rtp->ssrc);
send_rtcp(this, track_idx, std::move(rtcp));
send_rtcp(this, track_idx, RtcpHeader::toBuffer(rtcp_sdes));
ticker.resetTime();
//下次发送信令保活
rtcp_flag = false;
}
void RtspPlayer::onPlayResult_l(const SockException &ex , bool handshake_done) {