diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 15ffe15e..bdcf48fc 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -408,14 +408,7 @@ void WebRtcTransportImp::onStartWebRTC() { 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(info->plan_rtp->sample_rate, true); info->rtcp_context_send = std::make_shared(info->plan_rtp->sample_rate, false); - info->receiver = std::make_shared([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); @@ -599,9 +592,14 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { 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); + 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(); + } } } else { WarnL << "未识别的sr rtcp包:" << rtcp->dumpString(); @@ -677,7 +675,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { /////////////////////////////////////////////////////////////////// -void WebRtcTransportImp::changeRtpExtId(const RtpPayloadInfo *info, const RtpHeader *header, bool is_recv, bool is_rtx) const{ +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) { @@ -690,6 +688,35 @@ void WebRtcTransportImp::changeRtpExtId(const RtpPayloadInfo *info, const RtpHea 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); @@ -715,6 +742,8 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { } 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()) { @@ -734,16 +763,34 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { #endif if (!rtx) { //统计rtp接受情况，便于生成nack rtcp包 - info->nack_ctx.received(seq); + 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 - info->rtcp_context_recv->onRtp(seq, stamp_ms, len); + auto &ref = info->rtcp_context_recv[ssrc]; + if (!ref) { + ref = std::make_shared(info->plan_rtp->sample_rate, true); + } + ref->onRtp(seq, stamp_ms, len); + //修改ext id至统一 + changeRtpExtId(*info, rtp, true, false); } - //修改ext id至统一 - changeRtpExtId(info.get(), rtp, true, rtx); + //解析并排序rtp - info->receiver->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len); + auto &ref = info->receiver[ssrc]; + if (!ref) { + ref = std::make_shared([info, this](RtpPacket::Ptr rtp) mutable { + onSortedRtp(*info, 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; + } + + ref->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len); return; } @@ -754,11 +801,22 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { if (size < 2) { 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]; - InfoL << "received rtx rtp: " << origin_seq; rtp->seq = htons(origin_seq); - rtp->ssrc = htonl(info->offer_ssrc_rtp); + if (info->offer_ssrc_rtp) { + //非simulcast + rtp->ssrc = htonl(info->offer_ssrc_rtp); + TraceL << "received rtx rtp,ssrc: " << ssrc << ", seq:" << origin_seq; + } 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; memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf); buf += 2; @@ -766,10 +824,11 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { onRtp_l(buf, len, true); } -void WebRtcTransportImp::onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack) { +void WebRtcTransportImp::onSendNack(RtpPayloadInfo &info, 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_media = htonl(info.offer_ssrc_rtp); + rtcp->ssrc_media = htonl(ssrc); + DebugL << htonl(ssrc) << " " << nack.getPid(); sendRtcpPacket((char *) rtcp.get(), rtcp->getSize(), true); } @@ -826,12 +885,12 @@ 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 - changeRtpExtId(pr->second, header, false, false); + changeRtpExtId(*pr->second, header, false, 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); + changeRtpExtId(*pr->second, header, false, true); 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 diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 7f0dbb6b..a309e6bd 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -350,15 +350,17 @@ private: uint32_t answer_ssrc_rtx = 0; const RtcMedia *media; NackList nack_list; - NackContext nack_ctx; - RtcpContext::Ptr rtcp_context_recv; RtcpContext::Ptr rtcp_context_send; - std::shared_ptr receiver; + unordered_map > rid_ssrc; + unordered_map rtx_ssrc_to_rtp_ssrc; + unordered_map nack_ctx; + unordered_map rtcp_context_recv; + unordered_map > receiver; }; void onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp); - void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack); - void changeRtpExtId(const RtpPayloadInfo *info, const RtpHeader *header, bool is_recv, bool is_rtx = false) const; + void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack, uint32_t ssrc); + void changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, bool is_rtx = false, string *rid_ptr = nullptr) const; private: uint16_t _rtx_seq[2] = {0, 0};