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

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.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 50 matching lines @@
   _packetSizeSamples = packetSizeSamples;
   return 0;
 }
 
 int32_t RTPSenderAudio::RegisterAudioPayload(
     const char payloadName[RTP_PAYLOAD_NAME_SIZE],
     const int8_t payloadType,
     const uint32_t frequency,
     const uint8_t channels,
     const uint32_t rate,
-    RtpUtility::Payload*& payload) {
+    RtpUtility::Payload** payload) {
   if (RtpUtility::StringCompare(payloadName, "cn", 2)) {
     CriticalSectionScoped cs(_sendAudioCritsect.get());
     //  we can have multiple CNG payload types
     switch (frequency) {
       case 8000:
         _cngNBPayloadType = payloadType;
         break;
       case 16000:
         _cngWBPayloadType = payloadType;
         break;
       case 32000:
         _cngSWBPayloadType = payloadType;
         break;
       case 48000:
         _cngFBPayloadType = payloadType;
         break;
       default:
         return -1;
     }
   } else if (RtpUtility::StringCompare(payloadName, "telephone-event", 15)) {
     CriticalSectionScoped cs(_sendAudioCritsect.get());
     // Don't add it to the list
     // we dont want to allow send with a DTMF payloadtype
     _dtmfPayloadType = payloadType;
     return 0;
     // The default timestamp rate is 8000 Hz, but other rates may be defined.
   }
-  payload = new RtpUtility::Payload;
-  payload->typeSpecific.Audio.frequency = frequency;
-  payload->typeSpecific.Audio.channels = channels;
-  payload->typeSpecific.Audio.rate = rate;
-  payload->audio = true;
-  payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = '\0';
-  strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+  *payload = new RtpUtility::Payload;
+  (*payload)->typeSpecific.Audio.frequency = frequency;
+  (*payload)->typeSpecific.Audio.channels = channels;
+  (*payload)->typeSpecific.Audio.rate = rate;
+  (*payload)->audio = true;
+  (*payload)->name[RTP_PAYLOAD_NAME_SIZE - 1] = '\0';
+  strncpy((*payload)->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
   return 0;
 }
 
 bool RTPSenderAudio::MarkerBit(FrameType frameType, int8_t payload_type) {
   CriticalSectionScoped cs(_sendAudioCritsect.get());
   // for audio true for first packet in a speech burst
   bool markerBit = false;
   if (_lastPayloadType != payload_type) {
     if (payload_type != -1 && (_cngNBPayloadType == payload_type ||
                                _cngWBPayloadType == payload_type ||
@@ skipping 261 matching lines @@
 int32_t RTPSenderAudio::SetRED(int8_t payloadType) {
   if (payloadType < -1) {
     return -1;
   }
   CriticalSectionScoped cs(_sendAudioCritsect.get());
   _REDPayloadType = payloadType;
   return 0;
 }
 
 // Get payload type for Redundant Audio Data RFC 2198
-int32_t RTPSenderAudio::RED(int8_t& payloadType) const {
+int32_t RTPSenderAudio::RED(int8_t* payloadType) const {
   CriticalSectionScoped cs(_sendAudioCritsect.get());
   if (_REDPayloadType == -1) {
     // not configured
     return -1;
   }
-  payloadType = _REDPayloadType;
+  *payloadType = _REDPayloadType;
   return 0;
 }
 
 // Send a TelephoneEvent tone using RFC 2833 (4733)
 int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key,
                                            uint16_t time_ms,
                                            uint8_t level) {
   {
     CriticalSectionScoped lock(_sendAudioCritsect.get());
     if (_dtmfPayloadType < 0) {
@@ skipping 51 matching lines @@
                          dtmfTimeStamp, "seqnum", _rtpSender->SequenceNumber());
     retVal = _rtpSender->SendToNetwork(
         dtmfbuffer, 4, 12, TickTime::MillisecondTimestamp(),
         kAllowRetransmission, RtpPacketSender::kHighPriority);
     sendCount--;
   } while (sendCount > 0 && retVal == 0);
 
   return retVal;
 }
 }  // namespace webrtc