Merge pull request #1152 from sbarrac/master

Added playout delay extension support

include/rtc/rtc.h

@@ -341,6 +341,9 @@ typedef struct {
 	// AV1 only
 	rtcObuPacketization obuPacketization; // OBU paketization for AV1 samples
 
+	uint8_t playoutDelayId;
+	uint16_t playoutDelayMin;
+	uint16_t playoutDelayMax;
 } rtcPacketizerInit;
 
 // Deprecated, do not use

include/rtc/rtppacketizationconfig.hpp

@@ -61,6 +61,14 @@ public:
 	uint8_t ridId = 0;
 	optional<std::string> rid;
 
+	// the negotiated ID of the playout delay header extension
+	// https://webrtc.googlesource.com/src/+/main/docs/native-code/rtp-hdrext/playout-delay/README.md
+	uint8_t playoutDelayId;
+
+	// Minimum/maxiumum playout delay, in 10ms intervals. A value of 10 would equal a 100ms delay
+	uint16_t playoutDelayMin;
+	uint16_t playoutDelayMax;
+
 	/// Construct RTP configuration used in packetization process
 	/// @param ssrc SSRC of source
 	/// @param cname CNAME of source

src/capi.cpp

@@ -275,6 +275,9 @@ createRtpPacketizationConfig(const rtcPacketizationHandlerInit *init) {
 	                                                       init->payloadType, init->clockRate);
 	config->sequenceNumber = init->sequenceNumber;
 	config->timestamp = init->timestamp;
+	config->playoutDelayId = init->playoutDelayId;
+	config->playoutDelayMin = init->playoutDelayMin;
+	config->playoutDelayMax = init->playoutDelayMax;
 	return config;
 }
 

src/rtppacketizer.cpp

@@ -31,6 +31,11 @@ message_ptr RtpPacketizer::packetize(shared_ptr<binary> payload, bool mark) {
 	if (setVideoRotation)
 		rtpExtHeaderSize += 2;
 
+	const bool setPlayoutDelay = (rtpConfig->playoutDelayId > 0 && rtpConfig->playoutDelayId < 15);
+
+	if (setPlayoutDelay)
+		rtpExtHeaderSize += 4;
+
 	if (rtpConfig->mid.has_value())
 		rtpExtHeaderSize += (1 + rtpConfig->mid->length());
 
@@ -85,6 +90,21 @@ message_ptr RtpPacketizer::packetize(shared_ptr<binary> payload, bool mark) {
 			    reinterpret_cast<const std::byte *>(rtpConfig->rid->c_str()),
 			    rtpConfig->rid->length());
 		}
+
+		if (setPlayoutDelay) {
+			uint16_t min = rtpConfig->playoutDelayMin & 0xFFF;
+			uint16_t max = rtpConfig->playoutDelayMax & 0xFFF;
+
+			// 12 bits for min + 12 bits for max
+			byte data[] = {
+				byte((min >> 4) & 0xFF), 
+				byte(((min & 0xF) << 4) | ((max >> 8) & 0xF)),
+				byte(max & 0xFF)
+			};
+
+			extHeader->writeOneByteHeader(offset, rtpConfig->playoutDelayId, data, 3);
+			offset += 4;
+		}
 	}
 
 	rtp->preparePacket();