Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket.

webrtc/video/rtp_stream_receiver.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 257 matching lines @@
   } else {
     uint8_t* data = new uint8_t[packet.sizeBytes];
     memcpy(data, packet.dataPtr, packet.sizeBytes);
     packet.dataPtr = data;
   }
 
   packet_buffer_->InsertPacket(&packet);
   return 0;
 }
 
+// TODO(nisse): Try to delete this method. Obstacles: It is used by
+// ParseAndHandleEncapsulatingHeader, for handling Rtx packets. And
+// it's part of the RtpData interface which we implement.
 bool RtpStreamReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
                                           size_t rtp_packet_length) {
   RTPHeader header;
   if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
     return false;
   }
   header.payload_type_frequency = kVideoPayloadTypeFrequency;
   bool in_order = IsPacketInOrder(header);
   return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
 }
 
 // TODO(pbos): Remove as soon as audio can handle a changing payload type
 // without this callback.
 int32_t RtpStreamReceiver::OnInitializeDecoder(
     const int8_t payload_type,
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const int frequency,
     const size_t channels,
     const uint32_t rate) {
   RTC_NOTREACHED();
   return 0;
 }
 
 void RtpStreamReceiver::OnIncomingSSRCChanged(const uint32_t ssrc) {
   rtp_rtcp_->SetRemoteSSRC(ssrc);
 }
 
+// This method handles both regular RTP packets and packets recovered
+// via FlexFEC.
 void RtpStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
   {
     rtc::CritScope lock(&receive_cs_);
     if (!receiving_) {
       return;
     }
   }
+  if (!packet.recovered()) {
+    int64_t now_ms = clock_->TimeInMilliseconds();
 
-  int64_t now_ms = clock_->TimeInMilliseconds();
-
-  {
-    // Periodically log the RTP header of incoming packets.
-    rtc::CritScope lock(&receive_cs_);
-    if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
-      std::stringstream ss;
-      ss << "Packet received on SSRC: " << packet.Ssrc()
-         << " with payload type: " << static_cast<int>(packet.PayloadType())
-         << ", timestamp: " << packet.Timestamp()
-         << ", sequence number: " << packet.SequenceNumber()
-         << ", arrival time: " << packet.arrival_time_ms();
-      int32_t time_offset;
-      if (packet.GetExtension<TransmissionOffset>(&time_offset)) {
-        ss << ", toffset: " << time_offset;
+    {
+      // Periodically log the RTP header of incoming packets.
+      rtc::CritScope lock(&receive_cs_);
+      if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
+        std::stringstream ss;
+        ss << "Packet received on SSRC: " << packet.Ssrc()
+           << " with payload type: " << static_cast<int>(packet.PayloadType())
+           << ", timestamp: " << packet.Timestamp()
+           << ", sequence number: " << packet.SequenceNumber()
+           << ", arrival time: " << packet.arrival_time_ms();
+        int32_t time_offset;
+        if (packet.GetExtension<TransmissionOffset>(&time_offset)) {
+          ss << ", toffset: " << time_offset;
+        }
+        uint32_t send_time;
+        if (packet.GetExtension<AbsoluteSendTime>(&send_time)) {
+          ss << ", abs send time: " << send_time;
+        }
+        LOG(LS_INFO) << ss.str();
+        last_packet_log_ms_ = now_ms;
       }
-      uint32_t send_time;
-      if (packet.GetExtension<AbsoluteSendTime>(&send_time)) {
-        ss << ", abs send time: " << send_time;
-      }
-      LOG(LS_INFO) << ss.str();
-      last_packet_log_ms_ = now_ms;
     }

[sprang_webrtc, 2017/05/11 08:48:58]

nit: can we clean up the extra scopes for locking here? I don't think it's
justified to have one for checking if we're receiving above, then release it
only to (potentially) grab it again after checking the time....

[nisse-webrtc, 2017/05/11 09:04:01]

Done.

   }
 
   // TODO(nisse): Delete use of GetHeader, but needs refactoring of
   // ReceivePacket and IncomingPacket methods below.
   RTPHeader header;
   packet.GetHeader(&header);
 
   header.payload_type_frequency = kVideoPayloadTypeFrequency;
 
   bool in_order = IsPacketInOrder(header);
-  rtp_payload_registry_.SetIncomingPayloadType(header);
+  if (!packet.recovered()) {
+    // TODO(nisse): Why isn't this done for recovered packets?

[sprang_webrtc, 2017/05/11 08:48:58]

Will the header here have the actual payload type, or will it have ulpfec or red
or rtx some other payload type we don't want to register for this ssrc?
Anyway, that logic might be better put in
rtp_payload_registry_.SetIncomingPayloadType() since that's where rtx is
filtered out...

[nisse-webrtc, 2017/05/11 09:04:01]

No idea. At the moment I don't think rtx or ulpfec enter this code path.

+    rtp_payload_registry_.SetIncomingPayloadType(header);
+  }
   ReceivePacket(packet.data(), packet.size(), header, in_order);
   // Update receive statistics after ReceivePacket.
   // Receive statistics will be reset if the payload type changes (make sure
   // that the first packet is included in the stats).
-  rtp_receive_statistics_->IncomingPacket(
-      header, packet.size(), IsPacketRetransmitted(header, in_order));
+  if (!packet.recovered()) {
+    // TODO(nisse): We should pass a recovered flag to stats, to aid
+    // fixing bug bugs.webrtc.org/6339.
+    rtp_receive_statistics_->IncomingPacket(
+        header, packet.size(), IsPacketRetransmitted(header, in_order));
+  }
 }
 
 int32_t RtpStreamReceiver::RequestKeyFrame() {
   return rtp_rtcp_->RequestKeyFrame();
 }
 
 bool RtpStreamReceiver::IsUlpfecEnabled() const {
   return config_.rtp.ulpfec.ulpfec_payload_type != -1;
 }
 
@@ skipping 286 matching lines @@
     return;
 
   if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str()))
     return;
 
   tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(),
                              sprop_decoder.pps_nalu());
 }
 
 }  // namespace webrtc