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..812dbf1f55398377798991eb7d7cf0047eef5527 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -198,7 +198,7 @@ size_t RTPSenderVideo::FECPacketOverhead() const {
     // from an FEC viewpoint, they are part of the payload to be protected.
     // (The base RTP header is already protected by the FEC header.)
     return ForwardErrorCorrection::PacketOverhead() + REDForFECHeaderLength +
-           (_rtpSender.RTPHeaderLength() - kRtpHeaderSize);
+           (_rtpSender.RtpHeaderMaxLength() - kRtpHeaderSize);
   }
   return 0;
 }
@@ -248,7 +248,7 @@ int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
     cvo_mode = _rtpSender.ActivateCVORtpHeaderExtension();
   }
 
-  uint16_t rtp_header_length = _rtpSender.RTPHeaderLength();
+  int rtp_header_length = _rtpSender.RtpHeaderCurrentLength();
   size_t payload_bytes_to_send = payloadSize;
   const uint8_t* data = payloadData;
 
@@ -265,15 +265,19 @@ int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
   while (!last) {
     uint8_t dataBuffer[IP_PACKET_SIZE] = {0};
     size_t payload_bytes_in_packet = 0;
-    if (!packetizer->NextPacket(&dataBuffer[rtp_header_length],
-                                &payload_bytes_in_packet, &last)) {
+
+    // Note that RTP header size is dynamically computed since there may be
+    // an optional RTP header extension.
+    if (!packetizer->NextPacket(
+            &dataBuffer[_rtpSender.RtpHeaderCurrentLength()],
+            &payload_bytes_in_packet, &last)) {
       return -1;
     }
 
     // Write RTP header.
-    // Set marker bit true if this is the last packet in frame.
     _rtpSender.BuildRTPheader(

[danilchap, 2016/06/02 12:12:34]

playout_delay_oracle_ is not protected by crit_
That means that extension might be changed after payload is filled, but before
this call to BuildRTPheader. BuildRTPheader then will use less bytes which would
calculate payload offset incorrectly.

May be would be clearer to protect playout_delay_oracle with crit_,
or set extension Active/Inactive, so that less mechanics could be used for
optional extensions.

Either way ensure rtp_header_length is unchanged during this function.


Do not try to be too nice about this code: I plan to change this code with
RtpPacket shortly (and probably replace inactive extension concept with
something less confusing).

[Irfan, 2016/06/02 18:15:53]

Thanks for pointing this out. With the updated change the playout delay
extension can get active or inactive only before each call to this method, so
this way we do not have the issue of it changing while preparing the header.

         dataBuffer, payloadType, last, captureTimeStamp, capture_time_ms);
+
     // According to
     // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
     // ts_126114v120700p.pdf Section 7.4.5: