修改rtp pt时，确保不影响其他播放器

webrtc/SrtpSession.cpp

@@ -213,7 +213,7 @@ namespace RTC
 		}
 	}
 
-	bool SrtpSession::EncryptRtp(const uint8_t** data, size_t* len)
+	bool SrtpSession::EncryptRtp(const uint8_t** data, size_t* len, uint8_t pt)
 	{
 		MS_TRACE();
 
@@ -226,6 +226,7 @@ namespace RTC
 		}
 
 		std::memcpy(EncryptBuffer, *data, *len);
+        EncryptBuffer[1] |= (pt & 0x7F);
 
 		srtp_err_status_t err =
 		  srtp_protect(this->session, static_cast<void*>(EncryptBuffer), reinterpret_cast<int*>(len));

webrtc/SrtpSession.hpp

@@ -46,7 +46,7 @@ namespace RTC
 		~SrtpSession();
 
 	public:
-		bool EncryptRtp(const uint8_t** data, size_t* len);
+		bool EncryptRtp(const uint8_t** data, size_t* len, uint8_t pt);
 		bool DecryptSrtp(uint8_t* data, size_t* len);
 		bool EncryptRtcp(const uint8_t** data, size_t* len);
 		bool DecryptSrtcp(uint8_t* data, size_t* len);

webrtc/WebRtcTransport.cpp

@@ -179,11 +179,11 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *
     }
 }
 
-void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush) {
+void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt) {
     const uint8_t *p = (uint8_t *) buf;
     bool ret = false;
     if (_srtp_session_send) {
-        ret = _srtp_session_send->EncryptRtp(&p, &len);
+        ret = _srtp_session_send->EncryptRtp(&p, &len, pt);
     }
     if (ret) {
         onSendSockData((char *) p, len, flush);
@@ -467,16 +467,12 @@ void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const Rtp
 }
 
 void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){
-    if (!_send_rtp_pt[rtp->type]) {
+    auto &pt = _send_rtp_pt[rtp->type];
+    if (!pt) {
         //忽略，对方不支持该编码类型
         return;
     }
-    auto tmp = rtp->getHeader()->pt;
-    //设置pt
-    rtp->getHeader()->pt = _send_rtp_pt[rtp->type];
-    sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush);
+    sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, pt);
     //统计rtp发送情况，好做sr汇报
-    _rtp_receiver[_send_rtp_pt[rtp->type]].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
-    //还原pt
-    rtp->getHeader()->pt = tmp;
+    _rtp_receiver[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
 }

webrtc/WebRtcTransport.h

@@ -45,7 +45,7 @@ public:
      * @param buf rtcp内容
      * @param len rtcp长度
      */
-    void sendRtpPacket(char *buf, size_t len, bool flush);
+    void sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt);
     void sendRtcpPacket(char *buf, size_t len, bool flush);
 
 protected: