Let Call::OnRecoveredPacket parse RTP header extensions.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 1366 matching lines @@
   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // calls on the worker thread. We should move towards always using a network
   // thread. Then this check can be enabled.
   // RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
-// TODO(brandtr): Update this member function when we support protecting
-// audio packets with FlexFEC.
 void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
-  ReadLockScoped read_lock(*receive_crit_);
   rtc::Optional<RtpPacketReceived> parsed_packet =
       ParseRtpPacket(packet, length, nullptr);
   if (!parsed_packet)
     return;
 
   parsed_packet->set_recovered(true);
 
+  ReadLockScoped read_lock(*receive_crit_);
+  auto it = receive_rtp_config_.find(parsed_packet->Ssrc());
+  if (it == receive_rtp_config_.end()) {

[brandtr, 2017/08/21 11:05:31]

This logic is now duplicated between Call::DeliverRtp and
Call::OnRecoveredPacket. Perhaps it's better to move it back to ParseRtpPacket? 

[nisse-webrtc, 2017/08/21 11:54:24]

If that makes things simpler, sure. Longer term, I'd like to move identification
of extensions to the RtpStreamReceiverController, with extensions configured per
transport rather than per stream, which as far as I understand is also closer to
how they are defined in the spec.

[brandtr, 2017/08/22 13:14:23]

I'll leave it as is, because it will not be simpler if I move this code to
Call::ParseRtpPacket. The reason is that Call::DeliverRtp returns different
error codes for different parsing errors.

+    LOG(LS_ERROR) << "receive_rtp_config_ lookup failed for ssrc "
+                  << parsed_packet->Ssrc();
+    // Destruction of the receive stream, including deregistering from the
+    // RtpDemuxer, is not protected by the |receive_crit_| lock. But
+    // deregistering in the |receive_rtp_config_| map is protected by that lock.
+    // So by not passing the packet on to demuxing in this case, we prevent
+    // incoming packets to be passed on via the demuxer to a receive stream
+    // which is being torned down.
+    return;
+  }
+  parsed_packet->IdentifyExtensions(it->second.extensions);
+
+  // TODO(brandtr): Update here when we support protecting audio packets too.
   video_receiver_controller_.OnRtpPacket(*parsed_packet);
 }
 
 void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
                                      MediaType media_type) {
   auto it = receive_rtp_config_.find(packet.Ssrc());
   bool use_send_side_bwe =
       (it != receive_rtp_config_.end()) && it->second.use_send_side_bwe;
 
   RTPHeader header;
@@ skipping 14 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc