commit caba2d2a370cb6b5e67c881ecfa57fdac7411de8
author pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Wed May 14 13:57:12 2014
committer pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Wed May 14 13:57:12 2014
tree 651d470261967b7173218f1eb1fa3009c76169a0
parent 581e2172af690941d8630895380ce67dab53b31d

Add DeliveryStatus enum to DeliverPacket().

Allows signalling why packet delivery failed. Especially enables
signaling that delivery fails because the incoming packet had an unknown
SSRC. This allows an application to react and create receivers for the
new streams.

R=mflodman@webrtc.org
BUG=3228

Review URL: https://webrtc-codereview.appspot.com/12289005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6150 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/video/call.cc

diff --git a/webrtc/video/call.cc b/webrtc/video/call.cc
index a327654..6adb0fc 100644
--- a/webrtc/video/call.cc
+++ b/webrtc/video/call.cc
@@ -86,13 +86,14 @@
   virtual uint32_t SendBitrateEstimate() OVERRIDE;
   virtual uint32_t ReceiveBitrateEstimate() OVERRIDE;
 
-  virtual bool DeliverPacket(const uint8_t* packet, size_t length) OVERRIDE;
+  virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
+                                       size_t length) OVERRIDE;
 
  private:
-  bool DeliverRtcp(const uint8_t* packet, size_t length);
-  bool DeliverRtp(const RTPHeader& header,
-                  const uint8_t* packet,
-                  size_t length);
+  DeliveryStatus DeliverRtcp(const uint8_t* packet, size_t length);
+  DeliveryStatus DeliverRtp(const RTPHeader& header,
+                            const uint8_t* packet,
+                            size_t length);
 
   Call::Config config_;
 
@@ -278,9 +279,12 @@
   return 0;
 }
 
-bool Call::DeliverRtcp(const uint8_t* packet, size_t length) {
+Call::PacketReceiver::DeliveryStatus Call::DeliverRtcp(const uint8_t* packet,
+                                                       size_t length) {
   // TODO(pbos): Figure out what channel needs it actually.
   //             Do NOT broadcast! Also make sure it's a valid packet.
+  //             Return DELIVERY_UNKNOWN_SSRC if it can be determined that
+  //             there's no receiver of the packet.
   bool rtcp_delivered = false;
   {
     ReadLockScoped read_lock(*receive_lock_);
@@ -303,30 +307,33 @@
         rtcp_delivered = true;
     }
   }
-  return rtcp_delivered;
+  return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
 
-bool Call::DeliverRtp(const RTPHeader& header,
-                      const uint8_t* packet,
-                      size_t length) {
+Call::PacketReceiver::DeliveryStatus Call::DeliverRtp(const RTPHeader& header,
+                                                      const uint8_t* packet,
+                                                      size_t length) {
   ReadLockScoped read_lock(*receive_lock_);
   std::map<uint32_t, VideoReceiveStream*>::iterator it =
       receive_ssrcs_.find(header.ssrc);
-  if (it == receive_ssrcs_.end()) {
-    // TODO(pbos): Log some warning, SSRC without receiver.
-    return false;
-  }
-  return it->second->DeliverRtp(static_cast<const uint8_t*>(packet), length);
+
+  if (it == receive_ssrcs_.end())
+    return DELIVERY_UNKNOWN_SSRC;
+
+  return it->second->DeliverRtp(static_cast<const uint8_t*>(packet), length)
+             ? DELIVERY_OK
+             : DELIVERY_PACKET_ERROR;
 }
 
-bool Call::DeliverPacket(const uint8_t* packet, size_t length) {
+Call::PacketReceiver::DeliveryStatus Call::DeliverPacket(const uint8_t* packet,
+                                                         size_t length) {
   // TODO(pbos): ExtensionMap if there are extensions.
   if (RtpHeaderParser::IsRtcp(packet, static_cast<int>(length)))
     return DeliverRtcp(packet, length);
 
   RTPHeader rtp_header;
   if (!rtp_header_parser_->Parse(packet, static_cast<int>(length), &rtp_header))
-    return false;
+    return DELIVERY_PACKET_ERROR;
 
   return DeliverRtp(rtp_header, packet, length);
 }