modules/rtp_rtcp/include folder cleared of lint warnings

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 258 matching lines @@
       last_received_timestamp_ = 0;
       last_received_sequence_number_ = 0;
       last_received_frame_time_ms_ = -1;
 
       // Do we have a SSRC? Then the stream is restarted.
       if (ssrc_ != 0) {
         // Do we have the same codec? Then re-initialize coder.
         if (rtp_header.payloadType == last_received_payload_type) {
           re_initialize_decoder = true;
 
-          Payload* payload;
-          if (!rtp_payload_registry_->PayloadTypeToPayload(
-              rtp_header.payloadType, payload)) {
+          const Payload* payload = rtp_payload_registry_->PayloadTypeToPayload(
+              rtp_header.payloadType);
+          if (!payload) {
             return;
           }
-          assert(payload);
           payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
           strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
           if (payload->audio) {
             channels = payload->typeSpecific.Audio.channels;
             rate = payload->typeSpecific.Audio.rate;
           }
         }
       }
       ssrc_ = rtp_header.ssrc;
     }
@@ skipping 63 matching lines @@
 
       rtp_media_receiver_->CheckPayloadChanged(
         payload_type, specific_payload,
         &should_discard_changes);
 
       if (should_discard_changes) {
         is_red = false;
         return 0;
       }
 
-      Payload* payload;
-      if (!rtp_payload_registry_->PayloadTypeToPayload(payload_type, payload)) {
+      const Payload* payload =
+          rtp_payload_registry_->PayloadTypeToPayload(payload_type);
+      if (!payload) {
         // Not a registered payload type.
         return -1;
       }
-      assert(payload);
       payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
       strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
 
       rtp_payload_registry_->set_last_received_payload_type(payload_type);
 
       re_initialize_decoder = true;
 
       rtp_media_receiver_->SetLastMediaSpecificPayload(payload->typeSpecific);
       rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
 
@@ skipping 97 matching lines @@
     // implementations might have CSRC 0 as a valid value.
     if (num_csrcs_diff > 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(0, true);
     } else if (num_csrcs_diff < 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(0, false);
     }
   }
 }
 
 }  // namespace webrtc