[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs

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 152 matching lines @@
   const uint8_t* payload,
   size_t payload_length,
   PayloadUnion payload_specific,
   bool in_order) {
   // Trigger our callbacks.
   CheckSSRCChanged(rtp_header);
 
   int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0;
   bool is_red = false;
 
-  if (CheckPayloadChanged(rtp_header, first_payload_byte, is_red,
+  if (CheckPayloadChanged(rtp_header, first_payload_byte, &is_red,
                           &payload_specific) == -1) {
     if (payload_length == 0) {
       // OK, keep-alive packet.
       return true;
     }
     LOG(LS_WARNING) << "Receiving invalid payload type.";
     return false;
   }
 
   WebRtcRTPHeader webrtc_rtp_header;
@@ skipping 129 matching lines @@
 
 // Implementation note: must not hold critsect when called.
 // TODO(phoglund): Move as much as possible of this code path into the media
 // specific receivers. Basically this method goes through a lot of trouble to
 // compute something which is only used by the media specific parts later. If
 // this code path moves we can get rid of some of the rtp_receiver ->
 // media_specific interface (such as CheckPayloadChange, possibly get/set
 // last known payload).
 int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
                                              const int8_t first_payload_byte,
-                                             bool& is_red,
+                                             bool* is_red,
                                              PayloadUnion* specific_payload) {
   bool re_initialize_decoder = false;
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE];
   int8_t payload_type = rtp_header.payloadType;
 
   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
 
     int8_t last_received_payload_type =
         rtp_payload_registry_->last_received_payload_type();
     // TODO(holmer): Remove this code when RED parsing has been broken out from
     // RtpReceiverAudio.
     if (payload_type != last_received_payload_type) {
       if (rtp_payload_registry_->red_payload_type() == payload_type) {
         // Get the real codec payload type.
         payload_type = first_payload_byte & 0x7f;
-        is_red = true;
+        *is_red = true;
 
         if (rtp_payload_registry_->red_payload_type() == payload_type) {
           // Invalid payload type, traced by caller. If we proceeded here,
           // this would be set as |_last_received_payload_type|, and we would no
           // longer catch corrupt packets at this level.
           return -1;
         }
 
         // When we receive RED we need to check the real payload type.
         if (payload_type == last_received_payload_type) {
           rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
           return 0;
         }
       }
       bool should_discard_changes = false;
 
       rtp_media_receiver_->CheckPayloadChanged(
         payload_type, specific_payload,
         &should_discard_changes);
 
       if (should_discard_changes) {
-        is_red = false;
+        *is_red = false;
         return 0;
       }
 
       const Payload* payload =
           rtp_payload_registry_->PayloadTypeToPayload(payload_type);
       if (!payload) {
         // Not a registered payload type.
         return -1;
       }
       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);
 
       if (!payload->audio) {
         bool media_type_unchanged =
             rtp_payload_registry_->ReportMediaPayloadType(payload_type);
         if (media_type_unchanged) {
           // Only reset the decoder if the media codec type has changed.
           re_initialize_decoder = false;
         }
       }
     } else {
       rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
-      is_red = false;
+      *is_red = false;
     }
   }  // End critsect.
 
   if (re_initialize_decoder) {
     if (-1 ==
         rtp_media_receiver_->InvokeOnInitializeDecoder(
             cb_rtp_feedback_, payload_type, payload_name, *specific_payload)) {
       return -1;  // Wrong payload type.
     }
   }
@@ skipping 76 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