Delete class NullRtpData and function NullObjectRtpData

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc

 /*
  *  Copyright (c) 2012 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 18 matching lines @@
 using RtpUtility::Payload;
 
 // Only return the sources in the last 10 seconds.
 const int64_t kGetSourcesTimeoutMs = 10000;
 
 RtpReceiver* RtpReceiver::CreateVideoReceiver(
     Clock* clock,
     RtpData* incoming_payload_callback,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry) {
-  if (!incoming_payload_callback)
-    incoming_payload_callback = NullObjectRtpData();
+  RTC_DCHECK(incoming_payload_callback != nullptr);
   if (!incoming_messages_callback)
     incoming_messages_callback = NullObjectRtpFeedback();
   return new RtpReceiverImpl(
       clock, incoming_messages_callback, rtp_payload_registry,
       RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
 }
 
 RtpReceiver* RtpReceiver::CreateAudioReceiver(
     Clock* clock,
     RtpData* incoming_payload_callback,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry) {
-  if (!incoming_payload_callback)
-    incoming_payload_callback = NullObjectRtpData();
+  RTC_DCHECK(incoming_payload_callback != nullptr);
   if (!incoming_messages_callback)
     incoming_messages_callback = NullObjectRtpFeedback();
   return new RtpReceiverImpl(
       clock, incoming_messages_callback, rtp_payload_registry,
       RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback));
 }
 
 RtpReceiverImpl::RtpReceiverImpl(Clock* clock,
                                  RtpFeedback* incoming_messages_callback,
                                  RTPPayloadRegistry* rtp_payload_registry,
@@ skipping 481 matching lines @@
   for (vec_it = ssrc_sources_.begin(); vec_it != ssrc_sources_.end();
        ++vec_it) {
     if ((now_ms - vec_it->timestamp_ms()) <= kGetSourcesTimeoutMs) {
       break;
     }
   }
   ssrc_sources_.erase(ssrc_sources_.begin(), vec_it);
 }
 
 }  // namespace webrtc