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ZLMediaKit
Commit d4ff84e4 by xiongziliang
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完善ssrc相关处理

parent f6eb84b4
正在显示 包含 83 行增加 和 97 行删除
webrtc/WebRtcTransport.cpp
......@@ -395,51 +395,45 @@ bool WebRtcTransportImp::canRecvRtp() const{
return _push_src && (sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly);
}
const RtcSession& WebRtcTransportImp::getSdpWithSSRC() const{
auto &offer = getSdp(SdpType::answer);
if (offer.haveSSRC()) {
return offer;
}
auto &answer = getSdp(SdpType::offer);
CHECK(answer.haveSSRC());
return answer;
}
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_with_ssrc = getSdpWithSSRC().getMedia(m_answer.type);
for (auto &plan_answer : m_answer.plan) {
//获取offer端rtp的ssrc和pt相关信息
auto info = std::make_shared<RtpPayloadInfo>();
_rtp_info_pt.emplace(plan_answer.pt, info);
info->media = m_with_ssrc;
info->is_common_rtp = getCodecId(plan_answer.codec) != CodecInvalid;
if (info->is_common_rtp) {
//rtp
_rtp_info_ssrc[info->media->rtp_rtx_ssrc[0].ssrc] = info;
info->plan_rtp = &plan_answer;
info->plan_rtx = m_answer.getRelatedRtxPlan(plan_answer.pt);
} else {
//rtx
if (info->media->rtp_rtx_ssrc.size() > 1) {
_rtp_info_ssrc[info->media->rtp_rtx_ssrc[1].ssrc] = info;
}
info->plan_rtp = m_answer.getPlan(atoi(plan_answer.getFmtp("apt").data()));
info->plan_rtx = &plan_answer;
}
info->rtcp_context_recv = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, true);
info->rtcp_context_send = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, false);
info->receiver = std::make_shared<RtpReceiverImp>([info, this](RtpPacket::Ptr rtp) mutable {
onSortedRtp(*info, std::move(rtp));
});
info->nack_ctx.setOnNack([info, this](const FCI_NACK &nack) mutable {
onNack(*info, nack);
});
auto m_offer = getSdp(SdpType::offer).getMedia(m_answer.type);
auto info = std::make_shared<RtpPayloadInfo>();
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_recv = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, true);
info->rtcp_context_send = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, false);
info->receiver = std::make_shared<RtpReceiverImp>([info, this](RtpPacket::Ptr rtp) mutable {
onSortedRtp(*info, std::move(rtp));
});
info->nack_ctx.setOnNack([info, this](const FCI_NACK &nack) mutable {
onSendNack(*info, nack);
});
//send ssrc --> RtpPayloadInfo
_rtp_info_ssrc[info->answer_ssrc_rtp] = std::make_pair(false, info);
_rtp_info_ssrc[info->answer_ssrc_rtx] = std::make_pair(true, info);
//recv ssrc --> RtpPayloadInfo
_rtp_info_ssrc[info->offer_ssrc_rtp] = std::make_pair(false, info);;
_rtp_info_ssrc[info->offer_ssrc_rtx] = std::make_pair(true, info);;
//rtp pt --> RtpPayloadInfo
_rtp_info_pt.emplace(info->plan_rtp->pt, std::make_pair(false, info));
if (info->plan_rtx) {
//rtx pt --> RtpPayloadInfo
_rtp_info_pt.emplace(info->plan_rtx->pt, std::make_pair(true, info));
}
if (m_answer.type != TrackApplication) {
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
for (auto &ext : m_answer.extmap) {
for (auto &ext : m_offer->extmap) {
auto ext_type = RtpExt::getExtType(ext.ext);
_rtp_ext_id_to_type.emplace(ext.id, ext_type);
_rtp_ext_type_to_id.emplace(ext_type, ext.id);
......@@ -475,7 +469,7 @@ void WebRtcTransportImp::onStartWebRTC() {
auto it = _rtp_info_pt.find(m.plan[0].pt);
CHECK(it != _rtp_info_pt.end());
//记录发送rtp时约定的信息，届时发送rtp时需要修改pt和ssrc
_send_rtp_info[m.type] = it->second;
_send_rtp_info[m.type] = it->second.second;
}
}
}
......@@ -593,9 +587,13 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
RtcpSR *sr = (RtcpSR *) rtcp;
auto it = _rtp_info_ssrc.find(sr->ssrc);
if (it != _rtp_info_ssrc.end()) {
it->second->rtcp_context_recv->onRtcp(sr);
auto rr = it->second->rtcp_context_recv->createRtcpRR(sr->items.ssrc, sr->ssrc);
sendRtcpPacket(rr->data(), rr->size(), true);
auto rtx = it->second.first;
if (!rtx) {
auto &info = it->second.second;
info->rtcp_context_recv->onRtcp(sr);
auto rr = info->rtcp_context_recv->createRtcpRR(info->answer_ssrc_rtp, info->offer_ssrc_rtp);
sendRtcpPacket(rr->data(), rr->size(), true);
}
} else {
WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
}
......@@ -608,8 +606,12 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
for (auto item : rr->getItemList()) {
auto it = _rtp_info_ssrc.find(item->ssrc);
if (it != _rtp_info_ssrc.end()) {
auto sr = it->second->rtcp_context_send->createRtcpSR(item->ssrc);
sendRtcpPacket(sr->data(), sr->size(), true);
auto rtx = it->second.first;
if (!rtx) {
auto &info = it->second.second;
auto sr = info->rtcp_context_send->createRtcpSR(info->answer_ssrc_rtp);
sendRtcpPacket(sr->data(), sr->size(), true);
}
} else {
WarnL << "未识别的rr rtcp包:" << rtcp->dumpString();
}
......@@ -625,10 +627,6 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
WarnL << "未识别的bye rtcp包:" << rtcp->dumpString();
continue;
}
_rtp_info_pt.erase(it->second->plan_rtp->pt);
if (it->second->plan_rtx) {
_rtp_info_pt.erase(it->second->plan_rtx->pt);
}
_rtp_info_ssrc.erase(it);
}
onShutdown(SockException(Err_eof, "rtcp bye message received"));
......@@ -648,11 +646,15 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
WarnL << "未识别的 rtcp包:" << rtcp->dumpString();
return;
}
auto &fci = fb->getFci<FCI_NACK>();
it->second->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) {
//rtp重传
onSendRtp(rtp, true, true);
});
auto rtx = it->second.first;
if (!rtx) {
auto &info = it->second.second;
auto &fci = fb->getFci<FCI_NACK>();
info->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) {
//rtp重传
onSendRtp(rtp, true, true);
});
}
break;
}
default: break;
......@@ -704,8 +706,8 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
WarnL;
return;
}
auto &info = it->second;
if (info->is_common_rtp) {
auto &info = it->second.second;
if (!it->second.first) {
//这是普通的rtp数据
auto seq = ntohs(rtp->seq);
#if 0
......@@ -741,7 +743,7 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
auto origin_seq = payload[0] << 8 | payload[1];
InfoL << "received rtx rtp: " << origin_seq;
rtp->seq = htons(origin_seq);
rtp->ssrc = htonl(info->media->rtp_rtx_ssrc[0].ssrc);
rtp->ssrc = htonl(info->offer_ssrc_rtp);
rtp->pt = info->plan_rtp->pt;
memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf);
buf += 2;
......@@ -749,10 +751,10 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
onRtp_l(buf, len, true);
}
void WebRtcTransportImp::onNack(RtpPayloadInfo &info, const FCI_NACK &nack) {
void WebRtcTransportImp::onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack) {
auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_NACK, &nack, FCI_NACK::kSize);
rtcp->ssrc = htons(0);
rtcp->ssrc_media = htonl(info.media->rtp_rtx_ssrc[0].ssrc);
rtcp->ssrc = htons(info.answer_ssrc_rtp);
rtcp->ssrc_media = htonl(info.offer_ssrc_rtp);
sendRtcpPacket((char *) rtcp.get(), rtcp->getSize(), true);
}
......@@ -790,7 +792,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r
info->nack_list.push_back(rtp);
#if 0
//此处模拟发送丢包
if(rtp->getSeq() % 100 == 0){
if (rtp->type == TrackVideo && rtp->getSeq() % 100 == 0) {
DebugL << "send dropped:" << rtp->getSeq();
return;
}
......@@ -811,13 +813,13 @@ void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t &len, void *
if (!pr->first || !pr->second->plan_rtx) {
//普通的rtp,或者不支持rtx, 修改目标pt和ssrc
header->pt = pr->second->plan_rtp->pt;
header->ssrc = htonl(pr->second->media->rtp_rtx_ssrc[0].ssrc);
header->ssrc = htonl(pr->second->answer_ssrc_rtp);
} else {
//重传的rtp, rtx
header->pt = pr->second->plan_rtx->pt;
if (pr->second->media->rtp_rtx_ssrc.size() > 1) {
//有rtx单独的ssrc
header->ssrc = htonl(pr->second->media->rtp_rtx_ssrc[1].ssrc);
if (pr->second->answer_ssrc_rtx) {
//有rtx单独的ssrc,有些情况下，浏览器支持rtx，但是未指定rtx单独的ssrc
header->ssrc = htonl(pr->second->answer_ssrc_rtx);
}
auto origin_seq = ntohs(header->seq);
......
webrtc/WebRtcTransport.h
......@@ -338,25 +338,26 @@ private:
SdpAttrCandidate::Ptr getIceCandidate() const;
bool canSendRtp() const;
bool canRecvRtp() const;
const RtcSession& getSdpWithSSRC() const;
class RtpPayloadInfo {
public:
using Ptr = std::shared_ptr<RtpPayloadInfo>;
bool is_common_rtp;
const RtcCodecPlan *plan_rtp;
const RtcCodecPlan *plan_rtx;
uint32_t offer_ssrc_rtp = 0;
uint32_t offer_ssrc_rtx = 0;
uint32_t answer_ssrc_rtp = 0;
uint32_t answer_ssrc_rtx = 0;
const RtcMedia *media;
std::shared_ptr<RtpReceiverImp> receiver;
RtcpContext::Ptr rtcp_context_recv;
RtcpContext::Ptr rtcp_context_send;
NackList nack_list;
NackContext nack_ctx;
RtcpContext::Ptr rtcp_context_recv;
RtcpContext::Ptr rtcp_context_send;
std::shared_ptr<RtpReceiverImp> receiver;
};
void onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp);
void onNack(RtpPayloadInfo &info, const FCI_NACK &nack);
void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack);
private:
//用掉的总流量
......@@ -371,8 +372,6 @@ private:
Ticker _alive_ticker;
//pli rtcp计时器
Ticker _pli_ticker;
//记录协商的发送rtp的pt和ssrc
RtpPayloadInfo::Ptr _send_rtp_info[2];
//复合udp端口，接收一切rtp与rtcp
Socket::Ptr _socket;
//推流的rtsp源
......@@ -381,30 +380,14 @@ private:
RtspMediaSource::Ptr _play_src;
//播放rtsp源的reader对象
RtspMediaSource::RingType::RingReader::Ptr _reader;
//根据rtp的pt获取相关信息
unordered_map<uint8_t/*pt*/, RtpPayloadInfo::Ptr> _rtp_info_pt;
//根据发送rtp的track类型获取相关信息
RtpPayloadInfo::Ptr _send_rtp_info[2];
//根据接收rtp的pt获取相关信息
unordered_map<uint8_t/*pt*/, std::pair<bool/*is rtx*/,RtpPayloadInfo::Ptr> > _rtp_info_pt;
//根据rtcp的ssrc获取相关信息
unordered_map<uint32_t/*ssrc*/, RtpPayloadInfo::Ptr> _rtp_info_ssrc;
unordered_map<uint32_t/*ssrc*/, std::pair<bool/*is rtx*/,RtpPayloadInfo::Ptr> > _rtp_info_ssrc;
//发送rtp时需要修改rtp ext id
map<RtpExtType, uint8_t> _rtp_ext_type_to_id;
//接收rtp时需要修改rtp ext id
unordered_map<uint8_t, RtpExtType> _rtp_ext_id_to_type;
};
};
\ No newline at end of file
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