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ZLMediaKit
Commit edf30a04 by ziyue
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优化代码

parent 89ef1836
正在显示 包含 52 行增加 和 52 行删除
webrtc/WebRtcTransport.cpp
......@@ -399,37 +399,37 @@ void WebRtcTransportImp::onStartWebRTC() {
//获取ssrc和pt相关信息,届时收到rtp和rtcp时分别可以根据pt和ssrc找到相关的信息
for (auto &m_answer : getSdp(SdpType::answer).media) {
auto m_offer = getSdp(SdpType::offer).getMedia(m_answer.type);
auto info = std::make_shared<MediaTrack>();
auto track = std::make_shared<MediaTrack>();
info->media = &m_answer;
info->answer_ssrc_rtp = m_answer.getRtpSSRC();
info->answer_ssrc_rtx = m_answer.getRtxSSRC();
info->offer_ssrc_rtp = m_offer->getRtpSSRC();
info->offer_ssrc_rtx = m_offer->getRtxSSRC();
info->plan_rtp = &m_answer.plan[0];;
info->plan_rtx = m_answer.getRelatedRtxPlan(info->plan_rtp->pt);
info->rtcp_context_send = std::make_shared<RtcpContext>(false);
track->media = &m_answer;
track->answer_ssrc_rtp = m_answer.getRtpSSRC();
track->answer_ssrc_rtx = m_answer.getRtxSSRC();
track->offer_ssrc_rtp = m_offer->getRtpSSRC();
track->offer_ssrc_rtx = m_offer->getRtxSSRC();
track->plan_rtp = &m_answer.plan[0];;
track->plan_rtx = m_answer.getRelatedRtxPlan(track->plan_rtp->pt);
track->rtcp_context_send = std::make_shared<RtcpContext>(false);
//send ssrc --> MediaTrack
_ssrc_to_track[info->answer_ssrc_rtp] = info;
_ssrc_to_track[info->answer_ssrc_rtx] = info;
_ssrc_to_track[track->answer_ssrc_rtp] = track;
_ssrc_to_track[track->answer_ssrc_rtx] = track;
//recv ssrc --> MediaTrack
_ssrc_to_track[info->offer_ssrc_rtp] = info;
_ssrc_to_track[info->offer_ssrc_rtx] = info;
_ssrc_to_track[track->offer_ssrc_rtp] = track;
_ssrc_to_track[track->offer_ssrc_rtx] = track;
//rtp pt --> MediaTrack
_pt_to_track.emplace(info->plan_rtp->pt, std::make_pair(false, info));
if (info->plan_rtx) {
_pt_to_track.emplace(track->plan_rtp->pt, std::make_pair(false, track));
if (track->plan_rtx) {
//rtx pt --> MediaTrack
_pt_to_track.emplace(info->plan_rtx->pt, std::make_pair(true, info));
_pt_to_track.emplace(track->plan_rtx->pt, std::make_pair(true, track));
}
if (m_offer->type != TrackApplication) {
//记录rtp ext类型与id的关系，方便接收或发送rtp时修改rtp ext id
info->rtp_ext_ctx = std::make_shared<RtpExtContext>(*m_offer);
info->rtp_ext_ctx->setOnGetRtp([this, info](uint8_t pt, uint32_t ssrc, const string &rid) {
track->rtp_ext_ctx = std::make_shared<RtpExtContext>(*m_offer);
track->rtp_ext_ctx->setOnGetRtp([this, track](uint8_t pt, uint32_t ssrc, const string &rid) {
//ssrc --> MediaTrack
_ssrc_to_track[ssrc] = info;
_ssrc_to_track[ssrc] = track;
InfoL << "get rtp, pt:" << (int) pt << ", ssrc:" << ssrc << ", rid:" << rid;
});
}
......@@ -608,12 +608,12 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
RtcpSR *sr = (RtcpSR *) rtcp;
auto it = _ssrc_to_track.find(sr->ssrc);
if (it != _ssrc_to_track.end()) {
auto &info = it->second;
auto rtp_chn = info->getRtpChannel(sr->ssrc);
auto &track = it->second;
auto rtp_chn = track->getRtpChannel(sr->ssrc);
if(!rtp_chn){
WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
} else {
auto rr = rtp_chn->createRtcpRR(sr, info->answer_ssrc_rtp);
auto rr = rtp_chn->createRtcpRR(sr, track->answer_ssrc_rtp);
sendRtcpPacket(rr->data(), rr->size(), true);
}
} else {
......@@ -628,8 +628,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
for (auto item : rr->getItemList()) {
auto it = _ssrc_to_track.find(item->ssrc);
if (it != _ssrc_to_track.end()) {
auto &info = it->second;
auto sr = info->rtcp_context_send->createRtcpSR(info->answer_ssrc_rtp);
auto &track = it->second;
auto sr = track->rtcp_context_send->createRtcpSR(track->answer_ssrc_rtp);
sendRtcpPacket(sr->data(), sr->size(), true);
} else {
WarnL << "未识别的rr rtcp包:" << rtcp->dumpString();
......@@ -665,9 +665,9 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
WarnL << "未识别的 rtcp包:" << rtcp->dumpString();
return;
}
auto &info = it->second;
auto &track = it->second;
auto &fci = fb->getFci<FCI_NACK>();
info->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) {
track->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) {
//rtp重传
onSendRtp(rtp, true, true);
});
......@@ -684,15 +684,15 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::createRtpChannel(const string &rid, uint32_t ssrc, const MediaTrack::Ptr &info) {
void WebRtcTransportImp::createRtpChannel(const string &rid, uint32_t ssrc, const MediaTrack::Ptr &track) {
//rid --> RtpReceiverImp
auto &ref = info->rtp_channel[rid];
ref = std::make_shared<RtpChannel>([info, this, rid](RtpPacket::Ptr rtp) mutable {
onSortedRtp(*info, rid, std::move(rtp));
}, [info, this, ssrc](const FCI_NACK &nack) mutable {
onSendNack(*info, nack, ssrc);
auto &ref = track->rtp_channel[rid];
ref = std::make_shared<RtpChannel>([track, this, rid](RtpPacket::Ptr rtp) mutable {
onSortedRtp(*track, rid, std::move(rtp));
}, [track, this, ssrc](const FCI_NACK &nack) mutable {
onSendNack(*track, nack, ssrc);
});
InfoL << "create rtp receiver of ssrc:" << ssrc << ", rid:" << rid << ", codec:" << info->plan_rtp->codec;
InfoL << "create rtp receiver of ssrc:" << ssrc << ", rid:" << rid << ", codec:" << track->plan_rtp->codec;
}
void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
......@@ -708,33 +708,33 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
}
bool is_rtx = it->second.first;
auto ssrc = ntohl(rtp->ssrc);
auto &info = it->second.second;
auto &track = it->second.second;
//修改ext id至统一
string rid;
info->rtp_ext_ctx->changeRtpExtId(rtp, true, &rid);
track->rtp_ext_ctx->changeRtpExtId(rtp, true, &rid);
auto &ref = info->rtp_channel[rid];
auto &ref = track->rtp_channel[rid];
if (!ref) {
if (is_rtx) {
//再接收到对应的rtp前，丢弃rtx包
WarnL << "unknown rtx rtp, rid:" << rid << ", ssrc:" << ssrc << ", codec:" << info->plan_rtp->codec << ", seq:" << ntohs(rtp->seq);
WarnL << "unknown rtx rtp, rid:" << rid << ", ssrc:" << ssrc << ", codec:" << track->plan_rtp->codec << ", seq:" << ntohs(rtp->seq);
return;
}
createRtpChannel(rid, ssrc, info);
createRtpChannel(rid, ssrc, track);
}
if (!is_rtx) {
//这是普通的rtp数据
#if 0
auto seq = ntohs(rtp->seq);
if (info->media->type == TrackVideo && seq % 100 == 0) {
if (track->media->type == TrackVideo && seq % 100 == 0) {
//此处模拟接受丢包
return;
}
#endif
//解析并排序rtp
ref->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len, false);
ref->inputRtp(track->media->type, track->plan_rtp->sample_rate, (uint8_t *) buf, len, false);
return;
}
......@@ -749,19 +749,19 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
//前两个字节是原始的rtp的seq
auto origin_seq = payload[0] << 8 | payload[1];
//rtx 转换为 rtp
rtp->pt = info->plan_rtp->pt;
rtp->pt = track->plan_rtp->pt;
rtp->seq = htons(origin_seq);
rtp->ssrc = htonl(ref->getSSRC());
memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf);
buf += 2;
len -= 2;
ref->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len, true);
ref->inputRtp(track->media->type, track->plan_rtp->sample_rate, (uint8_t *) buf, len, true);
}
void WebRtcTransportImp::onSendNack(MediaTrack &info, const FCI_NACK &nack, uint32_t ssrc) {
void WebRtcTransportImp::onSendNack(MediaTrack &track, const FCI_NACK &nack, uint32_t ssrc) {
auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_NACK, &nack, FCI_NACK::kSize);
rtcp->ssrc = htons(info.answer_ssrc_rtp);
rtcp->ssrc = htons(track.answer_ssrc_rtp);
rtcp->ssrc_media = htonl(ssrc);
DebugL << htonl(ssrc) << " " << nack.getPid();
sendRtcpPacket((char *) rtcp.get(), rtcp->getSize(), true);
......@@ -769,8 +769,8 @@ void WebRtcTransportImp::onSendNack(MediaTrack &info, const FCI_NACK &nack, uint
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::onSortedRtp(MediaTrack &info, const string &rid, RtpPacket::Ptr rtp) {
if (info.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) {
void WebRtcTransportImp::onSortedRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp) {
if (track.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) {
//定期发送pli请求关键帧，方便非rtc等协议
_pli_ticker.resetTime();
sendRtcpPli(rtp->getSSRC());
......@@ -807,15 +807,15 @@ void WebRtcTransportImp::onSortedRtp(MediaTrack &info, const string &rid, RtpPac
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx){
auto &info = _type_to_track[rtp->type];
if (!info) {
auto &track = _type_to_track[rtp->type];
if (!track) {
//忽略，对方不支持该编码类型
return;
}
if (!rtx) {
//统计rtp发送情况，好做sr汇报
info->rtcp_context_send->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
info->nack_list.push_back(rtp);
track->rtcp_context_send->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
track->nack_list.push_back(rtp);
#if 0
//此处模拟发送丢包
if (rtp->type == TrackVideo && rtp->getSeq() % 100 == 0) {
......@@ -825,7 +825,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r
} else {
WarnL << "send rtx rtp:" << rtp->getSeq();
}
pair<bool/*rtx*/, MediaTrack *> ctx{rtx, info.get()};
pair<bool/*rtx*/, MediaTrack *> ctx{rtx, track.get()};
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, &ctx);
_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
}
......
webrtc/WebRtcTransport.h
......@@ -216,7 +216,7 @@ private:
void onSortedRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp);
void onSendNack(MediaTrack &track, const FCI_NACK &nack, uint32_t ssrc);
void createRtpChannel(const string &rid, uint32_t ssrc, const MediaTrack::Ptr &info);
void createRtpChannel(const string &rid, uint32_t ssrc, const MediaTrack::Ptr &track);
private:
uint16_t _rtx_seq[2] = {0, 0};
......
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