完善nack/srtp相关代码

webrtc/Nack.cpp

@@ -192,7 +192,7 @@ uint64_t NackContext::reSendNack() {
             it = _nack_send_status.erase(it);
             continue;
         }
-        if (now - it->second.update_stamp < 2 * _rtt) {
+        if (now - it->second.update_stamp < kNackIntervalRatio * _rtt) {
             //距离上次nack不足2倍的rtt，不用再发送nack
             ++it;
             continue;

webrtc/Nack.h

@@ -39,11 +39,13 @@ public:
     using Ptr = std::shared_ptr<NackContext>;
     using onNack = function<void(const FCI_NACK &nack)>;
     //最大保留的rtp丢包状态个数
-    static constexpr auto kNackMaxSize = 100;
+    static constexpr auto kNackMaxSize = 1024;
     //rtp丢包状态最长保留时间
     static constexpr auto kNackMaxMS = 3 * 1000;
     //nack最多请求重传10次
     static constexpr auto kNackMaxCount = 10;
+    //nack重传频率，rtt的倍数
+    static constexpr auto kNackIntervalRatio = 2.0f;
 
     NackContext() = default;
     ~NackContext() = default;

webrtc/WebRtcTransport.cpp

@@ -97,12 +97,7 @@ void WebRtcTransport::OnDtlsTransportConnected(
         std::string &remoteCert) {
     InfoL;
     _srtp_session_send = std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpLocalKey, srtpLocalKeyLen);
-
-    string srtpRemoteKey_str((char *) srtpRemoteKey, srtpRemoteKeyLen);
-    _srtp_session_recv_alloc = [srtpCryptoSuite, srtpRemoteKey_str]() {
-        return std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::INBOUND, srtpCryptoSuite,
-                                                  (uint8_t *) srtpRemoteKey_str.data(), srtpRemoteKey_str.size());
-    };
+    _srtp_session_recv = std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::INBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen);
     onStartWebRTC();
 }
 
@@ -255,19 +250,14 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *
         _dtls_transport->ProcessDtlsData((uint8_t *) buf, len);
         return;
     }
-    RtpHeader *rtp = (RtpHeader *) buf;
-    auto it = _srtp_session_recv.find(rtp->pt);
-    if (it == _srtp_session_recv.end()) {
-        it = _srtp_session_recv.emplace((uint8_t) rtp->pt, _srtp_session_recv_alloc()).first;
-    }
     if (is_rtp(buf)) {
-        if (it->second->DecryptSrtp((uint8_t *) buf, &len)) {
+        if (_srtp_session_recv->DecryptSrtp((uint8_t *) buf, &len)) {
             onRtp(buf, len);
         }
         return;
     }
     if (is_rtcp(buf)) {
-        if (it->second->DecryptSrtcp((uint8_t *) buf, &len)) {
+        if (_srtp_session_recv->DecryptSrtcp((uint8_t *) buf, &len)) {
             onRtcp(buf, len);
         }
         return;

webrtc/WebRtcTransport.h

@@ -122,10 +122,9 @@ private:
     std::shared_ptr<RTC::IceServer> _ice_server;
     std::shared_ptr<RTC::DtlsTransport> _dtls_transport;
     std::shared_ptr<RTC::SrtpSession> _srtp_session_send;
+    std::shared_ptr<RTC::SrtpSession> _srtp_session_recv;
     RtcSession::Ptr _offer_sdp;
     RtcSession::Ptr _answer_sdp;
-    function<std::shared_ptr<RTC::SrtpSession>() > _srtp_session_recv_alloc;
-    std::unordered_map<uint8_t /*pt*/, std::shared_ptr<RTC::SrtpSession> > _srtp_session_recv;
 };
 
 class RtpChannel;