修复rtp seq重复导致rtp nack重传列队异常的bug

webrtc/Nack.cpp

@@ -16,22 +16,22 @@ using namespace mediakit;
 
 static constexpr uint32_t kMaxNackMS = 10 * 1000;
 
-void NackList::push_back(RtpPacket::Ptr rtp) {
+void NackList::pushBack(RtpPacket::Ptr rtp) {
     auto seq = rtp->getSeq();
     _nack_cache_seq.emplace_back(seq);
     _nack_cache_pkt.emplace(seq, std::move(rtp));
-    while (get_cache_ms() > kMaxNackMS) {
+    while (getCacheMS() > kMaxNackMS) {
         //需要清除部分nack缓存
-        pop_front();
+        popFront();
     }
 }
 
-void NackList::for_each_nack(const FCI_NACK &nack, const function<void(const RtpPacket::Ptr &rtp)> &func) {
+void NackList::forEach(const FCI_NACK &nack, const function<void(const RtpPacket::Ptr &rtp)> &func) {
     auto seq = nack.getPid();
     for (auto bit : nack.getBitArray()) {
         if (bit) {
             //丢包
-            RtpPacket::Ptr *ptr = get_rtp(seq);
+            RtpPacket::Ptr *ptr = getRtp(seq);
             if (ptr) {
                 func(*ptr);
             }
@@ -40,7 +40,7 @@ void NackList::for_each_nack(const FCI_NACK &nack, const function<void(const Rtp
     }
 }
 
-void NackList::pop_front() {
+void NackList::popFront() {
     if (_nack_cache_seq.empty()) {
         return;
     }
@@ -48,7 +48,7 @@ void NackList::pop_front() {
     _nack_cache_seq.pop_front();
 }
 
-RtpPacket::Ptr *NackList::get_rtp(uint16_t seq) {
+RtpPacket::Ptr *NackList::getRtp(uint16_t seq) {
     auto it = _nack_cache_pkt.find(seq);
     if (it == _nack_cache_pkt.end()) {
         return nullptr;
@@ -56,19 +56,35 @@ RtpPacket::Ptr *NackList::get_rtp(uint16_t seq) {
     return &it->second;
 }
 
-uint32_t NackList::get_cache_ms() {
-    if (_nack_cache_seq.size() < 2) {
+uint32_t NackList::getCacheMS() {
+    auto back_stamp = getRtpStamp(_nack_cache_seq.back());
+    if (back_stamp == -1) {
+        _nack_cache_seq.pop_back();
         return 0;
     }
-    uint32_t back = _nack_cache_pkt[_nack_cache_seq.back()]->getStampMS(false);
-    uint32_t front = _nack_cache_pkt[_nack_cache_seq.front()]->getStampMS(false);
-    if (back >= front) {
-        return back - front;
+
+    auto front_stamp = getRtpStamp(_nack_cache_seq.front());
+    if (front_stamp == -1) {
+        _nack_cache_seq.pop_front();
+        return 0;
+    }
+
+    if (back_stamp >= front_stamp) {
+        return back_stamp - front_stamp;
     }
     //很有可能回环了
-    return back + (UINT32_MAX - front);
+    return back_stamp + (UINT32_MAX - front_stamp);
+}
+
+int64_t NackList::getRtpStamp(uint16_t seq) {
+    auto it = _nack_cache_pkt.find(seq);
+    if (it == _nack_cache_pkt.end()) {
+        return -1;
+    }
+    return it->second->getStampMS(false);
 }
 
+
 ////////////////////////////////////////////////////////////////////////////////////////////////
 
 void NackContext::received(uint16_t seq, bool is_rtx) {

webrtc/Nack.h

@@ -20,13 +20,14 @@ public:
     NackList() = default;
     ~NackList() = default;
 
-    void push_back(mediakit::RtpPacket::Ptr rtp);
-    void for_each_nack(const mediakit::FCI_NACK &nack, const std::function<void(const mediakit::RtpPacket::Ptr &rtp)> &cb);
+    void pushBack(mediakit::RtpPacket::Ptr rtp);
+    void forEach(const mediakit::FCI_NACK &nack, const std::function<void(const mediakit::RtpPacket::Ptr &rtp)> &cb);
 
 private:
-    void pop_front();
-    uint32_t get_cache_ms();
-    mediakit::RtpPacket::Ptr *get_rtp(uint16_t seq);
+    void popFront();
+    uint32_t getCacheMS();
+    int64_t getRtpStamp(uint16_t seq);
+    mediakit::RtpPacket::Ptr *getRtp(uint16_t seq);
 
 private:
     std::deque<uint16_t> _nack_cache_seq;

webrtc/WebRtcTransport.cpp

@@ -652,7 +652,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
                         }
                         auto &track = it->second;
                         auto &fci = fb->getFci<FCI_NACK>();
-                        track->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) {
+                        track->nack_list.forEach(fci, [&](const RtpPacket::Ptr &rtp) {
                             //rtp重传
                             onSendRtp(rtp, true, true);
                         });
@@ -807,7 +807,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r
     if (!rtx) {
         //统计rtp发送情况，好做sr汇报
         track->rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStamp(), rtp->ntp_stamp, rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
-        track->nack_list.push_back(rtp);
+        track->nack_list.pushBack(rtp);
 #if 0
         //此处模拟发送丢包
         if (rtp->type == TrackVideo && rtp->getSeq() % 100 == 0) {