Get rid of unnecessary cast of FlexfecReceiveStreamImpl to FlexfecReceiveStream

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 844 matching lines @@
   // TODO(brandtr): Store config in RtcEventLog here.
 
   return receive_stream;
 }
 
 void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyFlexfecReceiveStream");
   RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
 
   RTC_DCHECK(receive_stream != nullptr);
-  // There exist no other derived classes of FlexfecReceiveStream,
-  // so this downcast is safe.
-  FlexfecReceiveStreamImpl* receive_stream_impl =
-      static_cast<FlexfecReceiveStreamImpl*>(receive_stream);
   {
     WriteLockScoped write_lock(*receive_crit_);
 
-    const FlexfecReceiveStream::Config& config =
-        receive_stream_impl->GetConfig();
+    const FlexfecReceiveStream::Config& config = receive_stream->GetConfig();
     uint32_t ssrc = config.remote_ssrc;
     receive_rtp_config_.erase(ssrc);
 
     // Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be
     // destroyed.
     receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config))
         ->RemoveStream(ssrc);
   }
 
-  delete receive_stream_impl;
+  delete receive_stream;
 }
 
 Call::Stats Call::GetStats() const {
   // TODO(solenberg): Some test cases in EndToEndTest use this from a different
   // thread. Re-enable once that is fixed.
   // RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   Stats stats;
   // Fetch available send/receive bitrates.
   uint32_t send_bandwidth = 0;
   transport_send_->send_side_cc()->GetBitrateController()->AvailableBandwidth(
@@ skipping 489 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc