diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 9c4972e2..3ff65047 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/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(); - _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(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 { - onNack(*info, nack); - }); + auto m_offer = getSdp(SdpType::offer).getMedia(m_answer.type); + auto info = std::make_shared(); + + 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(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); + _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(); - 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(); + 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); diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index f9ca0d31..21b09372 100644 --- a/webrtc/WebRtcTransport.h +++ b/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; - - 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 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 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 _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 > _rtp_info_pt; //根据rtcp的ssrc获取相关信息 - unordered_map _rtp_info_ssrc; + unordered_map > _rtp_info_ssrc; //发送rtp时需要修改rtp ext id map _rtp_ext_type_to_id; //接收rtp时需要修改rtp ext id unordered_map _rtp_ext_id_to_type; -}; - - - - - - - - - - - - - - - - - - +}; \ No newline at end of file