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This commit is contained in:
xiongguangjie
2021-06-25 21:24:53 +08:00
parent e7e7906ecd 8c670c652e
commit 6d8d64caf6
19 changed files with 639 additions and 484 deletions
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357
webrtc/WebRtcTransport.cpp
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@@ -148,7 +148,6 @@ void WebRtcTransport::sendRtcpRemb(uint32_t ssrc, size_t bit_rate) {
fb->ssrc = htonl(0);
fb->ssrc_media = htonl(ssrc);
sendRtcpPacket((char *) fb.get(), fb->getSize(), true);
TraceL << ssrc << " " << bit_rate;
}
void WebRtcTransport::sendRtcpPli(uint32_t ssrc) {
@@ -255,7 +254,8 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *
if (_srtp_session_recv->DecryptSrtp((uint8_t *) buf, &len)) {
onRtp(buf, len);
} else {
WarnL;
RtpHeader *rtp = (RtpHeader *) buf;
WarnL << "srtp_unprotect rtp failed, pt:" << (int)rtp->pt;
}
return;
}
@@ -399,37 +399,49 @@ 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<RtpPayloadInfo>();
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>(info->plan_rtp->sample_rate, 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 --> 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);
//send ssrc --> MediaTrack
_ssrc_to_track[track->answer_ssrc_rtp] = track;
_ssrc_to_track[track->answer_ssrc_rtx] = track;
//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);;
//recv ssrc --> MediaTrack
_ssrc_to_track[track->offer_ssrc_rtp] = track;
_ssrc_to_track[track->offer_ssrc_rtx] = track;
//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));
//rtp pt --> MediaTrack
_pt_to_track.emplace(track->plan_rtp->pt, std::make_pair(false, track));
if (track->plan_rtx) {
//rtx pt --> MediaTrack
_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
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);
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] = track;
InfoL << "get rtp, pt:" << (int) pt << ", ssrc:" << ssrc << ", rid:" << rid;
});
int index = 0;
for (auto &ssrc : m_offer->rtp_ssrc_sim) {
//记录ssrc对应的MediaTrack
_ssrc_to_track[ssrc.ssrc] = track;
if (m_offer->rtp_rids.size() > index) {
//支持firefox的simulcast, 提前映射好ssrc和rid的关系
track->rtp_ext_ctx->setRid(ssrc.ssrc, m_offer->rtp_rids[index]);
}
++index;
}
}
}
@@ -466,10 +478,10 @@ void WebRtcTransportImp::onStartWebRTC() {
}
auto rtsp_media = rtsp_send_sdp.getMedia(m.type);
if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) {
auto it = _rtp_info_pt.find(m.plan[0].pt);
CHECK(it != _rtp_info_pt.end());
auto it = _pt_to_track.find(m.plan[0].pt);
CHECK(it != _pt_to_track.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.second;
_type_to_track[m.type] = it->second.second;
}
}
}
@@ -558,27 +570,45 @@ SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{
///////////////////////////////////////////////////////////////////
class RtpReceiverImp : public RtpReceiver {
class RtpChannel : public RtpTrackImp {
public:
RtpReceiverImp( function<void(RtpPacket::Ptr rtp)> cb){
_on_sort = std::move(cb);
RtpChannel(RtpTrackImp::OnSorted cb, function<void(const FCI_NACK &nack)> on_nack) {
setOnSorted(std::move(cb));
_nack_ctx.setOnNack(std::move(on_nack));
}
~RtpReceiverImp() override = default;
~RtpChannel() override = default;
bool inputRtp(TrackType type, int samplerate, uint8_t *ptr, size_t len){
return handleOneRtp((int) type, type, samplerate, ptr, len);
bool inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t len, bool is_rtx){
if (!is_rtx) {
RtpHeader *rtp = (RtpHeader *) ptr;
auto seq = ntohs(rtp->seq);
//统计rtp接受情况，便于生成nack rtcp包
_nack_ctx.received(seq);
//统计rtp收到的情况，好做rr汇报
_rtcp_context.onRtp(seq, ntohl(rtp->stamp), len);
}
return RtpTrack::inputRtp(type, sample_rate, ptr, len);
}
protected:
void onRtpSorted(RtpPacket::Ptr rtp, int track_index) override {
_on_sort(std::move(rtp));
Buffer::Ptr createRtcpRR(RtcpHeader *sr, uint32_t ssrc) {
_rtcp_context.onRtcp(sr);
return _rtcp_context.createRtcpRR(ssrc, getSSRC());
}
private:
function<void(RtpPacket::Ptr rtp)> _on_sort;
NackContext _nack_ctx;
RtcpContext _rtcp_context{true};
};
std::shared_ptr<RtpChannel> MediaTrack::getRtpChannel(uint32_t ssrc) const{
auto it_chn = rtp_channel.find(rtp_ext_ctx->getRid(ssrc));
if (it_chn == rtp_channel.end()) {
return nullptr;
}
return it_chn->second;
}
void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
_bytes_usage += len;
auto rtcps = RtcpHeader::loadFromBytes((char *) buf, len);
@@ -587,19 +617,15 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
case RtcpType::RTCP_SR : {
//对方汇报rtp发送情况
RtcpSR *sr = (RtcpSR *) rtcp;
auto it = _rtp_info_ssrc.find(sr->ssrc);
if (it != _rtp_info_ssrc.end()) {
auto rtx = it->second.first;
if (!rtx) {
auto &info = it->second.second;
auto it = info->rtcp_context_recv.find(sr->ssrc);
if (it != info->rtcp_context_recv.end()) {
it->second->onRtcp(sr);
auto rr = it->second->createRtcpRR(info->answer_ssrc_rtp, sr->ssrc);
sendRtcpPacket(rr->data(), rr->size(), true);
} else {
WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
}
auto it = _ssrc_to_track.find(sr->ssrc);
if (it != _ssrc_to_track.end()) {
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, track->answer_ssrc_rtp);
sendRtcpPacket(rr->data(), rr->size(), true);
}
} else {
WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
@@ -611,14 +637,11 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
//对方汇报rtp接收情况
RtcpRR *rr = (RtcpRR *) rtcp;
for (auto item : rr->getItemList()) {
auto it = _rtp_info_ssrc.find(item->ssrc);
if (it != _rtp_info_ssrc.end()) {
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);
}
auto it = _ssrc_to_track.find(item->ssrc);
if (it != _ssrc_to_track.end()) {
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();
}
@@ -629,12 +652,12 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
//对方汇报停止发送rtp
RtcpBye *bye = (RtcpBye *) rtcp;
for (auto ssrc : bye->getSSRC()) {
auto it = _rtp_info_ssrc.find(*ssrc);
if (it == _rtp_info_ssrc.end()) {
auto it = _ssrc_to_track.find(*ssrc);
if (it == _ssrc_to_track.end()) {
WarnL << "未识别的bye rtcp包:" << rtcp->dumpString();
continue;
}
_rtp_info_ssrc.erase(it);
_ssrc_to_track.erase(it);
}
onShutdown(SockException(Err_eof, "rtcp bye message received"));
break;
@@ -648,20 +671,17 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
switch ((RTPFBType) rtcp->report_count) {
case RTPFBType::RTCP_RTPFB_NACK : {
RtcpFB *fb = (RtcpFB *) rtcp;
auto it = _rtp_info_ssrc.find(fb->ssrc_media);
if (it == _rtp_info_ssrc.end()) {
auto it = _ssrc_to_track.find(fb->ssrc_media);
if (it == _ssrc_to_track.end()) {
WarnL << "未识别的 rtcp包:" << rtcp->dumpString();
return;
}
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);
});
}
auto &track = it->second;
auto &fci = fb->getFci<FCI_NACK>();
track->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) {
//rtp重传
onSendRtp(rtp, true, true);
});
break;
}
default: break;
@@ -675,122 +695,55 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, bool is_rtx, string *rid_ptr) const{
auto ext_map = RtpExt::getExtValue(header);
for (auto &pr : ext_map) {
if (is_recv) {
auto it = _rtp_ext_id_to_type.find(pr.first);
if (it == _rtp_ext_id_to_type.end()) {
WarnL << "接收rtp时,忽略不识别的rtp ext, id=" << (int) pr.first;
pr.second.clearExt();
continue;
}
pr.second.setType(it->second);
//重新赋值ext id为 ext type，作为后面处理ext的统一中间类型
pr.second.setExtId((uint8_t) it->second);
switch(it->second){
case RtpExtType::sdes_repaired_rtp_stream_id :
case RtpExtType::sdes_rtp_stream_id : {
auto ssrc = ntohl(header->ssrc);
auto rid = it->second == RtpExtType::sdes_rtp_stream_id ? pr.second.getRtpStreamId() : pr.second.getRepairedRtpStreamId();
//根据rid获取rtp或rtx的ssrc
auto &ssrc_ref = is_rtx ? info.rid_ssrc[rid].second : info.rid_ssrc[rid].first;
if (!ssrc_ref) {
//ssrc未赋值，赋值
ssrc_ref = ssrc;
DebugL << (is_rtx ? "got rid of rtx:" : "got rid:") << rid << ", ssrc:" << ssrc;
}
if (is_rtx) {
//rtx ssrc --> rtp ssrc
auto &rtp_ssrc_ref = info.rtx_ssrc_to_rtp_ssrc[ssrc];
if (!rtp_ssrc_ref && info.rid_ssrc[rid].first) {
//未找到rtx到rtp ssrc的映射关系，且已经获取rtp的ssrc，那么设置映射关系
rtp_ssrc_ref = info.rid_ssrc[rid].first;
DebugL << "got ssrc of rid:" << rid << ", [rtx-rtp]:" << ssrc << "-" << rtp_ssrc_ref;
}
}
if (rid_ptr) {
*rid_ptr = rid;
}
break;
}
default : break;
}
} else {
pr.second.setType((RtpExtType) pr.first);
auto it = _rtp_ext_type_to_id.find((RtpExtType) pr.first);
if (it == _rtp_ext_type_to_id.end()) {
WarnL << "发送rtp时, 忽略不被客户端支持rtp ext:" << pr.second.dumpString();
pr.second.clearExt();
continue;
}
//重新赋值ext id为客户端sdp声明的类型
pr.second.setExtId(it->second);
}
}
void WebRtcTransportImp::createRtpChannel(const string &rid, uint32_t ssrc, const MediaTrack::Ptr &track) {
//rid --> RtpReceiverImp
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:" << track->plan_rtp->codec;
}
void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
onRtp_l(buf, len, false);
}
void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
if (!rtx) {
_bytes_usage += len;
_alive_ticker.resetTime();
}
_bytes_usage += len;
_alive_ticker.resetTime();
RtpHeader *rtp = (RtpHeader *) buf;
auto ssrc = ntohl(rtp->ssrc);
//根据接收到的rtp的pt信息，找到该流的信息
auto it = _rtp_info_pt.find(rtp->pt);
if (it == _rtp_info_pt.end()) {
WarnL;
auto it = _pt_to_track.find(rtp->pt);
if (it == _pt_to_track.end()) {
WarnL << "unknown rtp pt:" << (int)rtp->pt;
return;
}
auto &info = it->second.second;
if (!it->second.first) {
bool is_rtx = it->second.first;
auto ssrc = ntohl(rtp->ssrc);
auto &track = it->second.second;
//修改ext id至统一
string rid;
track->rtp_ext_ctx->changeRtpExtId(rtp, true, &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:" << track->plan_rtp->codec << ", seq:" << ntohs(rtp->seq);
return;
}
createRtpChannel(rid, ssrc, track);
}
if (!is_rtx) {
//这是普通的rtp数据
auto seq = ntohs(rtp->seq);
#if 0
if (!rtx && info->media->type == TrackVideo && seq % 100 == 0) {
auto seq = ntohs(rtp->seq);
if (track->media->type == TrackVideo && seq % 100 == 0) {
//此处模拟接受丢包
DebugL << "recv dropped:" << seq;
return;
}
#endif
auto &ref = info->receiver[ssrc];
if (!rtx) {
//统计rtp接受情况，便于生成nack rtcp包
info->nack_ctx[ssrc].received(seq);
//时间戳转换成毫秒
auto stamp_ms = ntohl(rtp->stamp) * uint64_t(1000) / info->plan_rtp->sample_rate;
//统计rtp收到的情况，好做rr汇报
auto &cxt_ref = info->rtcp_context_recv[ssrc];
if (!cxt_ref) {
cxt_ref = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, true);
}
cxt_ref->onRtp(seq, stamp_ms, len);
//修改ext id至统一
string rid;
changeRtpExtId(*info, rtp, true, false, &rid);
if (!ref) {
ref = std::make_shared<RtpReceiverImp>([info, this, rid](RtpPacket::Ptr rtp) mutable {
onSortedRtp(*info, rid, std::move(rtp));
});
info->nack_ctx[ssrc].setOnNack([info, this, ssrc](const FCI_NACK &nack) mutable {
onSendNack(*info, nack, ssrc);
});
//recv simulcast ssrc --> RtpPayloadInfo
_rtp_info_ssrc[ssrc] = std::make_pair(false, info);
InfoL << "receive rtp of ssrc:" << ssrc;
}
}
//解析并排序rtp
if(!ref){
InfoL << "ignore no rtp receiver of ssrc:" << ssrc<<" is rtx:"<<rtx;
@@ -809,31 +762,22 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
return;
}
//修改ext id至统一
changeRtpExtId(*info, rtp, true, true);
//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 = track->plan_rtp->pt;
rtp->seq = htons(origin_seq);
if (info->offer_ssrc_rtp) {
//非simulcast或音频
rtp->ssrc = htonl(info->offer_ssrc_rtp);
TraceL << "received rtx rtp,ssrc: " << ssrc << ", seq:" << origin_seq << ", pt:" << (int)rtp->pt;
} else {
//todo simulcast下，辅码流通过rtx传输？
//simulcast情况下，根据rtx的ssrc查找rtp的ssrc
rtp->ssrc = htonl(info->rtx_ssrc_to_rtp_ssrc[ntohl(rtp->ssrc)]);
}
rtp->pt = info->plan_rtp->pt;
rtp->ssrc = htonl(ref->getSSRC());
memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf);
buf += 2;
len -= 2;
onRtp_l(buf, len, true);
ref->inputRtp(track->media->type, track->plan_rtp->sample_rate, (uint8_t *) buf, len, true);
}
void WebRtcTransportImp::onSendNack(RtpPayloadInfo &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);
@@ -841,8 +785,8 @@ void WebRtcTransportImp::onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack,
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &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());
@@ -879,42 +823,41 @@ void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &info, const string &rid, Rt
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx){
auto &info = _send_rtp_info[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(), rtp->getStampMS(), 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) {
DebugL << "send dropped:" << rtp->getSeq();
return;
}
#endif
} else {
WarnL << "send rtx rtp:" << rtp->getSeq();
}
pair<bool/*rtx*/, RtpPayloadInfo *> 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;
}
void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t &len, void *ctx) {
auto pr = (pair<bool/*rtx*/, RtpPayloadInfo *> *) ctx;
auto pr = (pair<bool/*rtx*/, MediaTrack *> *) ctx;
auto header = (RtpHeader *) buf;
if (!pr->first || !pr->second->plan_rtx) {
//普通的rtp,或者不支持rtx, 修改目标pt和ssrc
changeRtpExtId(*pr->second, header, false, false);
pr->second->rtp_ext_ctx->changeRtpExtId(header, false);
header->pt = pr->second->plan_rtp->pt;
header->ssrc = htonl(pr->second->answer_ssrc_rtp);
} else {
//重传的rtp, rtx
changeRtpExtId(*pr->second, header, false, true);
pr->second->rtp_ext_ctx->changeRtpExtId(header, false);
header->pt = pr->second->plan_rtx->pt;
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
@@ -949,7 +892,7 @@ void WebRtcTransportImp::onShutdown(const SockException &ex){
bool WebRtcTransportImp::close(MediaSource &sender, bool force) {
//此回调在其他线程触发
if(!_push_src || (!force && _push_src->totalReaderCount())){
if (!force && totalReaderCount(sender)) {
return false;
}
string err = StrPrinter << "close media:" << sender.getSchema() << "/" << sender.getVhost() << "/" << sender.getApp() << "/" << sender.getId() << " " << force;
@@ -958,7 +901,11 @@ bool WebRtcTransportImp::close(MediaSource &sender, bool force) {
}
int WebRtcTransportImp::totalReaderCount(MediaSource &sender) {
return _push_src ? _push_src->totalReaderCount() : sender.readerCount();
auto total_count = 0;
for (auto &src : _push_src_simulcast) {
total_count += src.second->totalReaderCount();
}
return total_count;
}
MediaOriginType WebRtcTransportImp::getOriginType(MediaSource &sender) const {