Renamed all RTP logs to Verbose

include/rtc/rtp.hpp

@@ -97,7 +97,7 @@ public:
 
         // Middle 32 bits of NTP Timestamp
         //		  this->lastReport = lastSR_NTP >> 16u;
-        setNTPOfSR(uint32_t(lastSR_NTP));
+        setNTPOfSR(uint64_t(lastSR_NTP));
         setDelaySinceSR(uint32_t(lastSR_DELAY));
 
         // The delay, expressed in units of 1/65536 seconds
@@ -142,7 +142,7 @@ public:
     inline uint32_t getDelaySinceSR() const { return ntohl(_delaySinceLastReport); }
 
     inline void log() const {
-        PLOG_DEBUG << "RTCP report block: "
+        PLOG_VERBOSE << "RTCP report block: "
                    << "ssrc="
                    << ntohl(ssrc)
                    // TODO: Implement these reports
@@ -168,7 +168,7 @@ public:
     inline uint16_t length() const { return ntohs(_length); }
 
     inline void setPayloadType(uint8_t type) { _payloadType = type; }
-    inline void setReportCount(uint8_t count) { _first = (_first & 0b11100000) | (count & 0b00011111); }
+    inline void setReportCount(uint8_t count) { _first = (_first & 0b11100000u) | (count & 0b00011111u); }
     inline void setLength(uint16_t length) { _length = htons(length); }
 
     inline void prepareHeader(uint8_t payloadType, uint8_t reportCount, uint16_t length) {
@@ -179,7 +179,7 @@ public:
     }
 
     inline void log() const {
-        PLOG_DEBUG << "RTCP header: "
+        PLOG_VERBOSE << "RTCP header: "
                    << "version=" << unsigned(version()) << ", padding=" << padding()
                    << ", reportCount=" << unsigned(reportCount())
                    << ", payloadType=" << unsigned(payloadType()) << ", length=" << length();
@@ -209,7 +209,7 @@ struct RTCP_FB_HEADER {
 
     void log() {
         header.log();
-        PLOG_DEBUG << "FB: " << " packet sender: " << getPacketSenderSSRC() << " media source: " << getMediaSourceSSRC();
+        PLOG_VERBOSE << "FB: " << " packet sender: " << getPacketSenderSSRC() << " media source: " << getMediaSourceSSRC();
     }
 };
 
@@ -251,7 +251,7 @@ public:
 
     inline void log() const {
         header.log();
-        PLOG_DEBUG << "RTCP SR: "
+        PLOG_VERBOSE << "RTCP SR: "
                    << " SSRC=" << ntohl(senderSSRC) << ", NTP_TS=" << ntpTimestamp()
                    << ", RTP_TS=" << rtpTimestamp() << ", packetCount=" << packetCount()
                    << ", octetCount=" << octetCount();
@@ -294,7 +294,7 @@ public:
 
     inline void log() const {
         header.log();
-        PLOG_DEBUG << "RTCP RR: "
+        PLOG_VERBOSE << "RTCP RR: "
                    << " SSRC=" << ntohl(senderSSRC);
 
         for (unsigned i = 0; i < unsigned(header.reportCount()); i++) {

src/dtlssrtptransport.cpp

@@ -90,6 +90,9 @@ bool DtlsSrtpTransport::sendMedia(message_ptr message) {
 	if (size < 8)
 		throw std::runtime_error("RTP/RTCP packet too short");
 
+//    return outgoing(message);
+//    return DtlsTransport::send(message);
+
 	// srtp_protect() and srtp_protect_rtcp() assume that they can write SRTP_MAX_TRAILER_LEN (for
 	// the authentication tag) into the location in memory immediately following the RTP packet.
 	message->resize(size + SRTP_MAX_TRAILER_LEN);

src/rtcp.cpp

@@ -97,7 +97,7 @@ void RtcpReceivingSession::pushREMB(unsigned int bitrate) {
 	auto remb = reinterpret_cast<RTCP_REMB *>(msg->data());
 	remb->preparePacket(mSsrc, 1, bitrate);
 	remb->setSsrc(0, mSsrc);
-	//remb->log();
+//	remb->log();
 
     send(msg);
 }

src/track.cpp

@@ -136,6 +136,18 @@ void Track::incoming(message_ptr message) {
 
 void Track::setRtcpHandler(std::shared_ptr<RtcpHandler> handler) {
 	mRtcpHandler = std::move(handler);
+	mRtcpHandler->onOutgoing([&](const rtc::message_ptr& message) {
+        #if RTC_ENABLE_MEDIA
+                auto transport = mDtlsSrtpTransport.lock();
+                if (!transport)
+                    throw std::runtime_error("Track transport is not open");
+
+                return transport->sendMedia(message);
+        #else
+                PLOG_WARNING << "Ignoring track send (not compiled with SRTP support)";
+            return false;
+        #endif
+	});
 }
 
 } // namespace rtc