Add transport sequence number on the non-pacer path of the rtp sender.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 624 matching lines @@
       if (rtx_ == kRtxOff) {
         // Without RTX we can't send padding in the middle of frames.
         if (!last_packet_marker_bit_)
           return 0;
         ssrc = ssrc_;
         sequence_number = sequence_number_;
         ++sequence_number_;
         payload_type = payload_type_;
         over_rtx = false;
       } else {
-        // Without abs-send-time a media packet must be sent before padding so
-        // that the timestamps used for estimation are correct.
-        if (!media_has_been_sent_ && !rtp_header_extension_map_.IsRegistered(
-            kRtpExtensionAbsoluteSendTime))
+        // Without abs-send-time or transport sequence number a media packet
+        // must be sent before padding so that the timestamps used for
+        // estimation are correct.
+        if (!media_has_been_sent_ &&
+            !(rtp_header_extension_map_.IsRegistered(
+                  kRtpExtensionAbsoluteSendTime) ||
+              using_transport_seq)) {
           return 0;
+        }
         // Only change change the timestamp of padding packets sent over RTX.
         // Padding only packets over RTP has to be sent as part of a media
         // frame (and therefore the same timestamp).
         if (last_timestamp_time_ms_ > 0) {
           timestamp +=
               (clock_->TimeInMilliseconds() - last_timestamp_time_ms_) * 90;
           capture_time_ms +=
               (clock_->TimeInMilliseconds() - last_timestamp_time_ms_);
         }
         ssrc = ssrc_rtx_;
@@ skipping 412 matching lines @@
       TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                                "PacedSend", corrected_time_ms,
                                "capture_time_ms", corrected_time_ms);
     }
     return 0;
   }
   if (capture_time_ms > 0) {
     UpdateDelayStatistics(capture_time_ms, now_ms);
   }
 
-  bool sent = SendPacketToNetwork(buffer, length, PacketOptions());
+  // TODO(sprang): Potentially too much overhead in IsRegistered()?
+  bool using_transport_seq = rtp_header_extension_map_.IsRegistered(
+                                 kRtpExtensionTransportSequenceNumber) &&
+                             transport_sequence_number_allocator_;
+
+  PacketOptions options;
+  if (using_transport_seq) {
+    options.packet_id =
+        UpdateTransportSequenceNumber(buffer, length, rtp_header);
+    if (transport_feedback_observer_) {
+      transport_feedback_observer_->AddPacket(options.packet_id, length, true);
+    }
+  }
+
+  bool sent = SendPacketToNetwork(buffer, length, options);
 
   // Mark the packet as sent in the history even if send failed. Dropping a
   // packet here should be treated as any other packet drop so we should be
   // ready for a retransmission.
   packet_history_.SetSent(rtp_header.sequenceNumber);
 
   if (!sent)
     return -1;
 
   {
@@ skipping 812 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc