mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-25 04:08:57 +08:00
初步支持simulcast流的接收和分流
This commit is contained in:
ziyue
parent
faa43adbbc
commit
dcb91e3b50
@ -408,14 +408,7 @@ void WebRtcTransportImp::onStartWebRTC() {
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info->offer_ssrc_rtx = m_offer->getRtxSSRC();
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info->plan_rtp = &m_answer.plan[0];;
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info->plan_rtx = m_answer.getRelatedRtxPlan(info->plan_rtp->pt);
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info->rtcp_context_recv = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, true);
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info->rtcp_context_send = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, false);
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info->receiver = std::make_shared<RtpReceiverImp>([info, this](RtpPacket::Ptr rtp) mutable {
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onSortedRtp(*info, std::move(rtp));
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});
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info->nack_ctx.setOnNack([info, this](const FCI_NACK &nack) mutable {
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onSendNack(*info, nack);
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});
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//send ssrc --> RtpPayloadInfo
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_rtp_info_ssrc[info->answer_ssrc_rtp] = std::make_pair(false, info);
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@ -599,9 +592,14 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
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auto rtx = it->second.first;
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if (!rtx) {
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auto &info = it->second.second;
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info->rtcp_context_recv->onRtcp(sr);
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auto rr = info->rtcp_context_recv->createRtcpRR(info->answer_ssrc_rtp, info->offer_ssrc_rtp);
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sendRtcpPacket(rr->data(), rr->size(), true);
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auto it = info->rtcp_context_recv.find(sr->ssrc);
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if (it != info->rtcp_context_recv.end()) {
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it->second->onRtcp(sr);
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auto rr = it->second->createRtcpRR(info->answer_ssrc_rtp, sr->ssrc);
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sendRtcpPacket(rr->data(), rr->size(), true);
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} else {
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WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
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}
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}
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} else {
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WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
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@ -677,7 +675,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
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///////////////////////////////////////////////////////////////////
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void WebRtcTransportImp::changeRtpExtId(const RtpPayloadInfo *info, const RtpHeader *header, bool is_recv, bool is_rtx) const{
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void WebRtcTransportImp::changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, bool is_rtx, string *rid_ptr) const{
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auto ext_map = RtpExt::getExtValue(header);
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for (auto &pr : ext_map) {
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if (is_recv) {
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@ -690,6 +688,35 @@ void WebRtcTransportImp::changeRtpExtId(const RtpPayloadInfo *info, const RtpHea
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pr.second.setType(it->second);
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//重新赋值ext id为 ext type，作为后面处理ext的统一中间类型
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pr.second.setExtId((uint8_t) it->second);
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switch(it->second){
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case RtpExtType::sdes_repaired_rtp_stream_id :
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case RtpExtType::sdes_rtp_stream_id : {
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auto ssrc = ntohl(header->ssrc);
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auto rid = it->second == RtpExtType::sdes_rtp_stream_id ? pr.second.getRtpStreamId() : pr.second.getRepairedRtpStreamId();
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//根据rid获取rtp或rtx的ssrc
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auto &ssrc_ref = is_rtx ? info.rid_ssrc[rid].second : info.rid_ssrc[rid].first;
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if (!ssrc_ref) {
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//ssrc未赋值，赋值
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ssrc_ref = ssrc;
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DebugL << (is_rtx ? "got rid of rtx:" : "got rid:") << rid << ", ssrc:" << ssrc;
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}
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if (is_rtx) {
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//rtx ssrc --> rtp ssrc
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auto &rtp_ssrc_ref = info.rtx_ssrc_to_rtp_ssrc[ssrc];
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if (!rtp_ssrc_ref && info.rid_ssrc[rid].first) {
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//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
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rtp_ssrc_ref = info.rid_ssrc[rid].first;
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DebugL << "got ssrc of rid:" << rid << ", [rtx-rtp]:" << ssrc << "-" << rtp_ssrc_ref;
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}
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}
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if (rid_ptr) {
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*rid_ptr = rid;
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}
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break;
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}
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default : break;
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}
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} else {
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pr.second.setType((RtpExtType) pr.first);
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auto it = _rtp_ext_type_to_id.find((RtpExtType) pr.first);
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@ -715,6 +742,8 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
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}
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RtpHeader *rtp = (RtpHeader *) buf;
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auto ssrc = ntohl(rtp->ssrc);
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//根据接收到的rtp的pt信息，找到该流的信息
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auto it = _rtp_info_pt.find(rtp->pt);
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if (it == _rtp_info_pt.end()) {
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@ -734,16 +763,34 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
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#endif
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if (!rtx) {
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//统计rtp接受情况，便于生成nack rtcp包
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info->nack_ctx.received(seq);
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info->nack_ctx[ssrc].received(seq);
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//时间戳转换成毫秒
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auto stamp_ms = ntohl(rtp->stamp) * uint64_t(1000) / info->plan_rtp->sample_rate;
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//统计rtp收到的情况，好做rr汇报
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info->rtcp_context_recv->onRtp(seq, stamp_ms, len);
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auto &ref = info->rtcp_context_recv[ssrc];
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if (!ref) {
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ref = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, true);
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}
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ref->onRtp(seq, stamp_ms, len);
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//修改ext id至统一
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changeRtpExtId(*info, rtp, true, false);
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}
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//修改ext id至统一
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changeRtpExtId(info.get(), rtp, true, rtx);
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//解析并排序rtp
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info->receiver->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len);
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auto &ref = info->receiver[ssrc];
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if (!ref) {
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ref = std::make_shared<RtpReceiverImp>([info, this](RtpPacket::Ptr rtp) mutable {
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onSortedRtp(*info, std::move(rtp));
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});
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info->nack_ctx[ssrc].setOnNack([info, this, ssrc](const FCI_NACK &nack) mutable {
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onSendNack(*info, nack, ssrc);
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});
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//recv simulcast ssrc --> RtpPayloadInfo
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_rtp_info_ssrc[ssrc] = std::make_pair(false, info);
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InfoL << "receive rtp of ssrc:" << ssrc;
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}
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ref->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len);
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return;
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}
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@ -754,11 +801,22 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
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if (size < 2) {
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return;
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}
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//修改ext id至统一
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changeRtpExtId(*info, rtp, true, true);
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//前两个字节是原始的rtp的seq
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auto origin_seq = payload[0] << 8 | payload[1];
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InfoL << "received rtx rtp: " << origin_seq;
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rtp->seq = htons(origin_seq);
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rtp->ssrc = htonl(info->offer_ssrc_rtp);
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if (info->offer_ssrc_rtp) {
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//非simulcast
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rtp->ssrc = htonl(info->offer_ssrc_rtp);
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TraceL << "received rtx rtp,ssrc: " << ssrc << ", seq:" << origin_seq;
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} else {
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//todo simulcast下，辅码流通过rtx传输？
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//simulcast情况下，根据rtx的ssrc查找rtp的ssrc
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rtp->ssrc = htonl(info->rtx_ssrc_to_rtp_ssrc[ntohl(rtp->ssrc)]);
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}
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rtp->pt = info->plan_rtp->pt;
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memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf);
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buf += 2;
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@ -766,10 +824,11 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
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onRtp_l(buf, len, true);
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}
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void WebRtcTransportImp::onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack) {
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void WebRtcTransportImp::onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack, uint32_t ssrc) {
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auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_NACK, &nack, FCI_NACK::kSize);
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rtcp->ssrc = htons(info.answer_ssrc_rtp);
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rtcp->ssrc_media = htonl(info.offer_ssrc_rtp);
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rtcp->ssrc_media = htonl(ssrc);
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DebugL << htonl(ssrc) << " " << nack.getPid();
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sendRtcpPacket((char *) rtcp.get(), rtcp->getSize(), true);
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}
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@ -826,12 +885,12 @@ void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t &len, void *
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if (!pr->first || !pr->second->plan_rtx) {
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//普通的rtp,或者不支持rtx, 修改目标pt和ssrc
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changeRtpExtId(pr->second, header, false, false);
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changeRtpExtId(*pr->second, header, false, false);
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header->pt = pr->second->plan_rtp->pt;
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header->ssrc = htonl(pr->second->answer_ssrc_rtp);
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} else {
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//重传的rtp, rtx
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changeRtpExtId(pr->second, header, false, true);
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changeRtpExtId(*pr->second, header, false, true);
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header->pt = pr->second->plan_rtx->pt;
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if (pr->second->answer_ssrc_rtx) {
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//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
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@ -350,15 +350,17 @@ private:
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uint32_t answer_ssrc_rtx = 0;
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const RtcMedia *media;
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NackList nack_list;
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NackContext nack_ctx;
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RtcpContext::Ptr rtcp_context_recv;
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RtcpContext::Ptr rtcp_context_send;
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std::shared_ptr<RtpReceiverImp> receiver;
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unordered_map<string/*rid*/, std::pair<uint32_t/*rtp ssrc*/, uint32_t/*rtx ssrc*/> > rid_ssrc;
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unordered_map<uint32_t/*rtx ssrc*/, uint32_t/*rtp ssrc*/> rtx_ssrc_to_rtp_ssrc;
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unordered_map<uint32_t/*simulcast ssrc*/, NackContext> nack_ctx;
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unordered_map<uint32_t/*simulcast ssrc*/, RtcpContext::Ptr> rtcp_context_recv;
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unordered_map<uint32_t/*simulcast ssrc*/, std::shared_ptr<RtpReceiverImp> > receiver;
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};
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void onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp);
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void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack);
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void changeRtpExtId(const RtpPayloadInfo *info, const RtpHeader *header, bool is_recv, bool is_rtx = false) const;
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void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack, uint32_t ssrc);
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void changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, bool is_rtx = false, string *rid_ptr = nullptr) const;
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private:
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uint16_t _rtx_seq[2] = {0, 0};
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