Replace AudioReceiveStream::DeliverRtp with OnRtpPacket.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index ec8053768d794bdc814b0a290a2d8b434b3d0ef1..0a62a470c724bcc7100dfd8a21822059373e6568 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -1198,12 +1198,9 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
     if (it != audio_receive_ssrcs_.end()) {
       received_bytes_per_second_counter_.Add(static_cast<int>(length));
       received_audio_bytes_per_second_counter_.Add(static_cast<int>(length));
-      auto status = it->second->DeliverRtp(packet, length, packet_time)
-                        ? DELIVERY_OK
-                        : DELIVERY_PACKET_ERROR;
-      if (status == DELIVERY_OK)
-        event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
-      return status;
+      it->second->OnRtpPacket(*parsed_packet);
+      event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
+      return DELIVERY_OK;
     }
   }
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {