Add path for recovered packets from internal::Call to RtpStreamReceiver.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index dd08d770ee850c0ff17d10bd732a97a7cc3a5495..e3a86a9bd193948fdcfaad8392180548d5e1ff51 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -18,6 +18,7 @@
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/scoped_voe_interface.h"
+#include "webrtc/base/basictypes.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/logging.h"
@@ -32,6 +33,7 @@
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/utility/include/process_thread.h"
@@ -57,6 +59,7 @@ namespace internal {
 
 class Call : public webrtc::Call,
              public PacketReceiver,
+             public RecoveredPacketReceiver,
              public CongestionController::Observer,
              public BitrateAllocator::LimitObserver {
  public:
@@ -91,6 +94,9 @@ class Call : public webrtc::Call,
                                size_t length,
                                const PacketTime& packet_time) override;
 
+  // Implements RecoveredPacketReceiver.
+  bool OnRecoveredPacket(const uint8_t* packet, size_t length) override;
+
   void SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
 
@@ -948,5 +954,16 @@ PacketReceiver::DeliveryStatus Call::DeliverPacket(
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
+// TODO(brandtr): Update this member function when we support protecting
+// audio packets with FlexFEC.
+bool Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
+  uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
+  ReadLockScoped read_lock(*receive_crit_);
+  auto it = video_receive_ssrcs_.find(ssrc);
+  if (it != video_receive_ssrcs_.end())

[stefan-webrtc, 2016/10/10 15:42:02]

nit: I prefer the opposite, i.e., that errors are handled in the if statements
and the correct path ends the method.

if (it == ...end())
  return false;
return it->..OnRecoverdPacket(...);

Feel free to disagree though. :)

[brandtr, 2016/10/10 23:05:19]

I think your suggestion makes it neater, so I've changed the code accordingly.

I modeled the original version of this function after DeliverRtp (see e.g.
https://cs.chromium.org/chromium/src/third_party/webrtc/call/call.cc?l=909), but
since that function tries multiple delivery paths before failing, I guess the
opposite order makes sense there.

+    return it->second->OnRecoveredPacket(packet, length);
+  return false;
+}
+
 }  // namespace internal
 }  // namespace webrtc