Add sender controlled playout delay limits

webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index 0bf95b7fe4b14f10d60d4cea44504d5a2083ee0e..df20601554d42c46552dad27dec8215459fb5ec2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -248,7 +248,6 @@ int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
     cvo_mode = _rtpSender.ActivateCVORtpHeaderExtension();
   }
 
-  uint16_t rtp_header_length = _rtpSender.RTPHeaderLength();
   size_t payload_bytes_to_send = payloadSize;
   const uint8_t* data = payloadData;
 
@@ -265,15 +264,22 @@ int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
   while (!last) {
     uint8_t dataBuffer[IP_PACKET_SIZE] = {0};
     size_t payload_bytes_in_packet = 0;
+
+    // Write RTP header.
+    // Note that RTP header size is dynamically computed since there may be
+    // an optional RTP header extension.
+    uint16_t rtp_header_length = _rtpSender.BuildRTPheader(
+        dataBuffer, payloadType, false, captureTimeStamp, capture_time_ms);
+
     if (!packetizer->NextPacket(&dataBuffer[rtp_header_length],
                                 &payload_bytes_in_packet, &last)) {
       return -1;
     }
 
-    // Write RTP header.
     // Set marker bit true if this is the last packet in frame.
-    _rtpSender.BuildRTPheader(
-        dataBuffer, payloadType, last, captureTimeStamp, capture_time_ms);
+    if (last)
+      dataBuffer[1] |= kRtpMarkerBitMask;
+
     // According to
     // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
     // ts_126114v120700p.pdf Section 7.4.5: