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

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index ec8053768d794bdc814b0a290a2d8b434b3d0ef1..e8c4c0cf54e9db3a164dc925a4e1701aa62c00ee 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -177,7 +177,7 @@ class Call : public webrtc::Call,
 
   rtc::Optional<RtpPacketReceived> ParseRtpPacket(const uint8_t* packet,
                                                   size_t length,
-                                                  const PacketTime& packet_time)
+                                                  const PacketTime* packet_time)
       SHARED_LOCKS_REQUIRED(receive_crit_);
 
   void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
@@ -398,7 +398,7 @@ Call::~Call() {
 rtc::Optional<RtpPacketReceived> Call::ParseRtpPacket(
     const uint8_t* packet,
     size_t length,
-    const PacketTime& packet_time) {
+    const PacketTime* packet_time) {
   RtpPacketReceived parsed_packet;
   if (!parsed_packet.Parse(packet, length))
     return rtc::Optional<RtpPacketReceived>();
@@ -408,8 +408,8 @@ rtc::Optional<RtpPacketReceived> Call::ParseRtpPacket(
     parsed_packet.IdentifyExtensions(it->second.extensions);
 
   int64_t arrival_time_ms;
-  if (packet_time.timestamp != -1) {
-    arrival_time_ms = (packet_time.timestamp + 500) / 1000;
+  if (packet_time && packet_time->timestamp != -1) {
+    arrival_time_ms = (packet_time->timestamp + 500) / 1000;
   } else {
     arrival_time_ms = clock_->TimeInMilliseconds();
   }
@@ -1184,7 +1184,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
   // TODO(nisse): We should parse the RTP header only here, and pass
   // on parsed_packet to the receive streams.
   rtc::Optional<RtpPacketReceived> parsed_packet =
-      ParseRtpPacket(packet, length, packet_time);
+      ParseRtpPacket(packet, length, &packet_time);
 
   if (!parsed_packet)
     return DELIVERY_PACKET_ERROR;
@@ -1254,12 +1254,20 @@ PacketReceiver::DeliveryStatus Call::DeliverPacket(
 // TODO(brandtr): Update this member function when we support protecting
 // audio packets with FlexFEC.
 bool Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {

[brandtr, 2017/02/15 12:20:50]

could the return value be turned into void here as well?

[nisse-webrtc, 2017/02/15 13:59:14]

That's a change of the RecoveredPacketReceiver interface. I'll do it.

The logic to keep calling OnRecoveredPacket again and again if it returns false
doesn't look particularly useful. 

-  uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
   ReadLockScoped read_lock(*receive_crit_);
-  auto it = video_receive_ssrcs_.find(ssrc);
+  rtc::Optional<RtpPacketReceived> parsed_packet =
+      ParseRtpPacket(packet, length, nullptr);
+  if (!parsed_packet)
+    return false;
+
+  parsed_packet->set_recovered(true);
+
+  auto it = video_receive_ssrcs_.find(parsed_packet->Ssrc());
   if (it == video_receive_ssrcs_.end())
     return false;
-  return it->second->OnRecoveredPacket(packet, length);
+
+  it->second->OnRtpPacket(*parsed_packet);
+  return true;
 }
 
 void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,