RtpPacketReceiver base class and OnRtpPacket, with a pre-parsed RTP packet.

webrtc/video/rtp_stream_receiver.cc

Index: webrtc/video/rtp_stream_receiver.cc
diff --git a/webrtc/video/rtp_stream_receiver.cc b/webrtc/video/rtp_stream_receiver.cc
index d49d0959f32fc826dd3da0cb26d7dda3fe1e0079..1d5aea3822d863d3fc3ba3e3486ef77685d274eb 100644
--- a/webrtc/video/rtp_stream_receiver.cc
+++ b/webrtc/video/rtp_stream_receiver.cc
@@ -94,6 +94,7 @@ RtpStreamReceiver::RtpStreamReceiver(
     VCMTiming* timing)
     : clock_(Clock::GetRealTimeClock()),
       config_(*config),
+      rtp_config_(config->rtp.extensions, config->rtp.transport_cc),
       video_receiver_(video_receiver),
       packet_router_(packet_router),
       remb_(remb),
@@ -242,6 +243,56 @@ RtpReceiver* RtpStreamReceiver::GetRtpReceiver() const {
   return rtp_receiver_.get();
 }
 
+bool RtpStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
+  {
+    rtc::CritScope lock(&receive_cs_);
+    if (!receiving_) {
+      return false;
+    }
+  }
+
+  RTPHeader header;
+  packet.GetHeader(&header);
+
+  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: " << header.ssrc << " with payload type: "
+         << static_cast<int>(header.payloadType) << ", timestamp: "
+         << header.timestamp << ", sequence number: " << header.sequenceNumber
+         << ", arrival time: " << packet.arrival_time_ms();
+      if (header.extension.hasTransmissionTimeOffset)
+        ss << ", toffset: " << header.extension.transmissionTimeOffset;
+      if (header.extension.hasAbsoluteSendTime)
+        ss << ", abs send time: " << header.extension.absoluteSendTime;
+      LOG(LS_INFO) << ss.str();
+      last_packet_log_ms_ = now_ms;
+    }
+  }
+
+  header.payload_type_frequency = kVideoPayloadTypeFrequency;
+
+  bool in_order = IsPacketInOrder(header);
+  rtp_payload_registry_.SetIncomingPayloadType(header);
+  // TODO(nisse): Is .data() and .size() right? Strip headers or not?

[stefan-webrtc, 2017/02/09 13:40:06]

Yes, this looks correct, as the full packet is needed for FEC recovery etc in
ParseAndHandleEncapsulatingHeader. If not for that we could strip the header
here already.

+  bool ret = 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));
+  return ret;
+}
+
+const RtpPacketReceiver::RtpConfig& RtpStreamReceiver::rtp_config() const {
+  return rtp_config_;
+}
+
+
 int32_t RtpStreamReceiver::OnReceivedPayloadData(
     const uint8_t* payload_data,
     size_t payload_size,
@@ -316,59 +367,6 @@ void RtpStreamReceiver::OnIncomingSSRCChanged(const uint32_t ssrc) {
   rtp_rtcp_->SetRemoteSSRC(ssrc);
 }
 
-bool RtpStreamReceiver::DeliverRtp(const uint8_t* rtp_packet,
-                                   size_t rtp_packet_length,
-                                   const PacketTime& packet_time) {
-  {
-    rtc::CritScope lock(&receive_cs_);
-    if (!receiving_) {
-      return false;
-    }
-  }
-
-  RTPHeader header;
-  if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length,
-                                 &header)) {
-    return false;
-  }
-  int64_t arrival_time_ms;
-  int64_t now_ms = clock_->TimeInMilliseconds();
-  if (packet_time.timestamp != -1)
-    arrival_time_ms = (packet_time.timestamp + 500) / 1000;
-  else
-    arrival_time_ms = now_ms;
-
-  {
-    // 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: " << header.ssrc << " with payload type: "
-         << static_cast<int>(header.payloadType) << ", timestamp: "
-         << header.timestamp << ", sequence number: " << header.sequenceNumber
-         << ", arrival time: " << arrival_time_ms;
-      if (header.extension.hasTransmissionTimeOffset)
-        ss << ", toffset: " << header.extension.transmissionTimeOffset;
-      if (header.extension.hasAbsoluteSendTime)
-        ss << ", abs send time: " << header.extension.absoluteSendTime;
-      LOG(LS_INFO) << ss.str();
-      last_packet_log_ms_ = now_ms;
-    }
-  }
-
-  header.payload_type_frequency = kVideoPayloadTypeFrequency;
-
-  bool in_order = IsPacketInOrder(header);
-  rtp_payload_registry_.SetIncomingPayloadType(header);
-  bool ret = ReceivePacket(rtp_packet, rtp_packet_length, 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, rtp_packet_length, IsPacketRetransmitted(header, in_order));
-  return ret;
-}
-
 int32_t RtpStreamReceiver::RequestKeyFrame() {
   return rtp_rtcp_->RequestKeyFrame();
 }