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

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index bd9314d5d768b6e2d30c5be2a4ef2287b9c35ef5..707b6013d0293367195662948a5dce2828eae834 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -642,11 +642,13 @@ size_t RTPSender::SendPadData(size_t bytes,
         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).
@@ -1079,7 +1081,22 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
     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 (using_transport_seq && transport_feedback_observer_) {

[sprang_webrtc, 2016/01/27 11:32:17]

Nit: Wrap in same if-block as if (using_transport_seq) { above
and just switch on transport_feedback_observer_.
Is that even a valid case btw? Should we just DCHECK on it instead?

[stefan-webrtc, 2016/01/27 11:39:41]

Done.

+    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