初步支持simulcast流的接收和分流

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<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);
@@ -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 it = info->rtcp_context_recv.find(sr->ssrc);
-                        auto rr = info->rtcp_context_recv->createRtcpRR(info->answer_ssrc_rtp, info->offer_ssrc_rtp);
+                        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<RtcpContext>(info->plan_rtp->sample_rate, true);
             }
+            ref->onRtp(seq, stamp_ms, len);
             //修改ext id至统一
-        changeRtpExtId(info.get(), rtp, true, rtx);
+            changeRtpExtId(*info, rtp, true, false);
+        }
+
         //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<RtpReceiverImp>([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);
+    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

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<RtpReceiverImp> receiver;
+        unordered_map<string/*rid*/, std::pair<uint32_t/*rtp ssrc*/, uint32_t/*rtx ssrc*/> > rid_ssrc;
+        unordered_map<uint32_t/*rtx ssrc*/, uint32_t/*rtp ssrc*/> rtx_ssrc_to_rtp_ssrc;
+        unordered_map<uint32_t/*simulcast ssrc*/, NackContext> nack_ctx;
+        unordered_map<uint32_t/*simulcast ssrc*/, RtcpContext::Ptr> rtcp_context_recv;
+        unordered_map<uint32_t/*simulcast ssrc*/, std::shared_ptr<RtpReceiverImp> > receiver;
     };
 
     void onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp);
-    void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack);
+    void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack, uint32_t ssrc);
-    void changeRtpExtId(const RtpPayloadInfo *info, const RtpHeader *header, bool is_recv, bool is_rtx = false) const;
+    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};