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

BUG=webrtc:5277
R=mflodman

Review URL: https://codereview.webrtc.org/1513303003

Cr-Commit-Position: refs/heads/master@{#11025}

22 files changed, 67 insertions, 56 deletions

diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 334d81172c..796be1304c 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -112,7 +112,10 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_CONST_METHOD0(SendingMedia,
       bool());
   MOCK_CONST_METHOD4(BitrateSent,
-      void(uint32_t* totalRate, uint32_t* videoRate, uint32_t* fecRate, uint32_t* nackRate));
+                     void(uint32_t* totalRate,
+                          uint32_t* videoRate,
+                          uint32_t* fecRate,
+                          uint32_t* nackRate));
   MOCK_METHOD1(RegisterVideoBitrateObserver, void(BitrateStatisticsObserver*));
   MOCK_CONST_METHOD0(GetVideoBitrateObserver, BitrateStatisticsObserver*(void));
   MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
@@ -176,7 +179,10 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_CONST_METHOD1(RemoteRTCPStat,
       int32_t(std::vector<RTCPReportBlock>* receiveBlocks));
   MOCK_METHOD4(SetRTCPApplicationSpecificData,
-      int32_t(const uint8_t subType, const uint32_t name, const uint8_t* data, const uint16_t length));
+               int32_t(const uint8_t subType,
+                       const uint32_t name,
+                       const uint8_t* data,
+                       const uint16_t length));
   MOCK_METHOD1(SetRTCPVoIPMetrics,
       int32_t(const RTCPVoIPMetric* VoIPMetric));
   MOCK_METHOD1(SetRtcpXrRrtrStatus,
@@ -221,8 +227,6 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_CONST_METHOD1(SendREDPayloadType, int32_t(int8_t* payloadType));
   MOCK_METHOD2(SetRTPAudioLevelIndicationStatus,
       int32_t(const bool enable, const uint8_t ID));
-  MOCK_CONST_METHOD2(GetRTPAudioLevelIndicationStatus,
-      int32_t(bool& enable, uint8_t& ID));
   MOCK_METHOD1(SetAudioLevel,
       int32_t(const uint8_t level_dBov));
   MOCK_METHOD1(SetTargetSendBitrate,
@@ -242,8 +246,6 @@ class MockRtpRtcp : public RtpRtcp {
       int32_t(const KeyFrameRequestMethod method));
   MOCK_METHOD0(RequestKeyFrame,
       int32_t());
-  MOCK_CONST_METHOD3(Version,
-      int32_t(char* version, uint32_t& remaining_buffer_in_bytes, uint32_t& position));
   MOCK_METHOD0(TimeUntilNextProcess,
         int64_t());
   MOCK_METHOD0(Process,
diff --git a/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg b/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg
new file mode 100644
index 0000000000..7edbb96e9b
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg
@@ -0,0 +1,8 @@
+#tmmbr_help is refactored in CL#1474693002
+exclude_files=tmmbr_help.*
+#rtp_utility is refactored in CL#1481773004
+exclude_files=rtp_utility.*
+#rtcp_utility planned to be removed when webrtc:5260 will be finished.
+exclude_files=rtcp_utility.*
+#rtcp_receiver/rtcp_receiver_help will be refactored more deeply as part of webrtc:5260
+exclude_files=rtcp_receiver.*
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
index b79f6ba2f4..6a63813f40 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
@@ -25,7 +25,7 @@ class CriticalSectionWrapper;
 
 class FecReceiverImpl : public FecReceiver {
  public:
-  FecReceiverImpl(RtpData* callback);
+  explicit FecReceiverImpl(RtpData* callback);
   virtual ~FecReceiverImpl();
 
   int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index e7d8fd71a5..e19c31bfec 100644
--- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -53,7 +53,7 @@ class VerifyingRtxReceiver : public NullRtpData {
 
 class TestRtpFeedback : public NullRtpFeedback {
  public:
-  TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
+  explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
   virtual ~TestRtpFeedback() {}
 
   void OnIncomingSSRCChanged(const uint32_t ssrc) override {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
index a639f7f7f5..87c0259b3e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -26,7 +26,8 @@ namespace {
 
 class TestTransport : public Transport {
  public:
-  TestTransport(RTCPReceiver* rtcp_receiver) : rtcp_receiver_(rtcp_receiver) {}
+  explicit TestTransport(RTCPReceiver* rtcp_receiver)
+      : rtcp_receiver_(rtcp_receiver) {}
 
   bool SendRtp(const uint8_t* /*data*/,
                size_t /*len*/,
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index e1721394f8..d65b04c8ab 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -746,7 +746,7 @@ bool RTCPReceiver::UpdateRTCPReceiveInformationTimers() {
   return updateBoundingSet;
 }
 
-int32_t RTCPReceiver::BoundingSet(bool &tmmbrOwner, TMMBRSet* boundingSetRec) {
+int32_t RTCPReceiver::BoundingSet(bool* tmmbrOwner, TMMBRSet* boundingSetRec) {
   CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
 
   std::map<uint32_t, RTCPReceiveInformation*>::iterator receiveInfoIt =
@@ -766,7 +766,7 @@ int32_t RTCPReceiver::BoundingSet(bool &tmmbrOwner, TMMBRSet* boundingSetRec) {
         i++) {
       if(receiveInfo->TmmbnBoundingSet.Ssrc(i) == main_ssrc_) {
         // owner of bounding set
-        tmmbrOwner = true;
+        *tmmbrOwner = true;
       }
       boundingSetRec->SetEntry(i,
                                receiveInfo->TmmbnBoundingSet.Tmmbr(i),
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index a22fd81b85..24861bd49e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -103,7 +103,7 @@ public:
 
     bool UpdateRTCPReceiveInformationTimers();
 
-    int32_t BoundingSet(bool &tmmbrOwner, TMMBRSet* boundingSetRec);
+    int32_t BoundingSet(bool* tmmbrOwner, TMMBRSet* boundingSetRec);
 
     int32_t UpdateTMMBR();
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index ea62fcb0d6..15f9388b57 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -645,7 +645,7 @@ rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildTMMBR(
   // will accuire criticalSectionRTCPReceiver_ is a potental deadlock but
   // since RTCPreceiver is not doing the reverse we should be fine
   int32_t lengthOfBoundingSet =
-      ctx.feedback_state_.module->BoundingSet(tmmbrOwner, candidateSet);
+      ctx.feedback_state_.module->BoundingSet(&tmmbrOwner, candidateSet);
 
   if (lengthOfBoundingSet > 0) {
     for (int32_t i = 0; i < lengthOfBoundingSet; i++) {
@@ -1077,7 +1077,7 @@ bool RTCPSender::AllVolatileFlagsConsumed() const {
 bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) {
   class Sender : public rtcp::RtcpPacket::PacketReadyCallback {
    public:
-    Sender(Transport* transport)
+    explicit Sender(Transport* transport)
         : transport_(transport), send_failure_(false) {}
 
     void OnPacketReady(uint8_t* data, size_t length) override {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
index 37966de6c2..342e38a1f2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
@@ -28,7 +28,7 @@ const size_t kVideoRotationLength = 2;
 const size_t kTransportSequenceNumberLength = 3;
 
 struct HeaderExtension {
-  HeaderExtension(RTPExtensionType extension_type)
+  explicit HeaderExtension(RTPExtensionType extension_type)
       : type(extension_type), length(0), active(true) {
     Init();
   }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
index 3d4c09ab95..8e1a732b19 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -29,7 +29,7 @@ static const size_t kMaxHistoryCapacity = 9600;
 
 class RTPPacketHistory {
  public:
-  RTPPacketHistory(Clock* clock);
+  explicit RTPPacketHistory(Clock* clock);
   ~RTPPacketHistory();
 
   void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index bdc05c6f7b..fb9b206cd1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -170,7 +170,7 @@ bool RtpReceiverImpl::IncomingRtpPacket(
   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.
@@ -320,7 +320,7 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
 // 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;
 
@@ -338,7 +338,7 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
       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,
@@ -360,7 +360,7 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
         &should_discard_changes);
 
       if (should_discard_changes) {
-        is_red = false;
+        *is_red = false;
         return 0;
       }
 
@@ -390,7 +390,7 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
       }
     } else {
       rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
-      is_red = false;
+      *is_red = false;
     }
   }  // End critsect.
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
index 4ec35315d2..eaa07d98bb 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -71,7 +71,7 @@ class RtpReceiverImpl : public RtpReceiver {
   void CheckCSRC(const WebRtcRTPHeader& rtp_header);
   int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
                               const int8_t first_payload_byte,
-                              bool& is_red,
+                              bool* is_red,
                               PayloadUnion* payload);
 
   Clock* clock_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index 0e0a34a4a4..0f7ad30e87 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -95,7 +95,7 @@ class RTPReceiverStrategy {
   // Note: Implementations may call the callback for other reasons than calls
   // to ParseRtpPacket, for instance if the implementation somehow recovers a
   // packet.
-  RTPReceiverStrategy(RtpData* data_callback);
+  explicit RTPReceiverStrategy(RtpData* data_callback);
 
   rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   PayloadUnion last_payload_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 30b1b96d93..450eed698e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -956,8 +956,8 @@ bool ModuleRtpRtcpImpl::UpdateRTCPReceiveInformationTimers() {
 }
 
 // Called from RTCPsender.
-int32_t ModuleRtpRtcpImpl::BoundingSet(bool& tmmbr_owner,
-                                       TMMBRSet*& bounding_set) {
+int32_t ModuleRtpRtcpImpl::BoundingSet(bool* tmmbr_owner,
+                                       TMMBRSet* bounding_set) {
   return rtcp_receiver_.BoundingSet(tmmbr_owner, bounding_set);
 }
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 24cbbe4079..04e09c1217 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -295,7 +295,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
 
   bool LastReceivedXrReferenceTimeInfo(RtcpReceiveTimeInfo* info) const;
 
-  virtual int32_t BoundingSet(bool& tmmbr_owner, TMMBRSet*& bounding_set_rec);
+  int32_t BoundingSet(bool* tmmbr_owner, TMMBRSet* bounding_set_rec);
 
   void BitrateSent(uint32_t* total_rate,
                    uint32_t* video_rate,
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index f975ededdc..405a78d8c5 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -97,7 +97,7 @@ class SendTransport : public Transport,
 
 class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
  public:
-  RtpRtcpModule(SimulatedClock* clock)
+  explicit RtpRtcpModule(SimulatedClock* clock)
       : receive_statistics_(ReceiveStatistics::Create(clock)) {
     RtpRtcp::Configuration config;
     config.audio = false;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 4e91a299bd..17fbbac791 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -326,11 +326,11 @@ int32_t RTPSender::RegisterPayload(
     return -1;
   }
   int32_t ret_val = 0;
-  RtpUtility::Payload* payload = NULL;
+  RtpUtility::Payload* payload = nullptr;
   if (audio_configured_) {
     // TODO(mflodman): Change to CreateAudioPayload and make static.
     ret_val = audio_->RegisterAudioPayload(payload_name, payload_number,
-                                           frequency, channels, rate, payload);
+                                           frequency, channels, rate, &payload);
   } else {
     payload = video_->CreateVideoPayload(payload_name, payload_number, rate);
   }
@@ -455,7 +455,7 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type,
   }
   if (audio_configured_) {
     int8_t red_pl_type = -1;
-    if (audio_->RED(red_pl_type) == 0) {
+    if (audio_->RED(&red_pl_type) == 0) {
       // We have configured RED.
       if (red_pl_type == payload_type) {
         // And it's a match...
@@ -1001,7 +1001,7 @@ bool RTPSender::IsFecPacket(const uint8_t* buffer,
   bool fec_enabled;
   uint8_t pt_red;
   uint8_t pt_fec;
-  video_->GenericFECStatus(fec_enabled, pt_red, pt_fec);
+  video_->GenericFECStatus(&fec_enabled, &pt_red, &pt_fec);
   return fec_enabled &&
       header.payloadType == pt_red &&
       buffer[header.headerLength] == pt_fec;
@@ -1778,7 +1778,7 @@ int32_t RTPSender::RED(int8_t *payload_type) const {
   if (!audio_configured_) {
     return -1;
   }
-  return audio_->RED(*payload_type);
+  return audio_->RED(payload_type);
 }
 
 RtpVideoCodecTypes RTPSender::VideoCodecType() const {
@@ -1804,7 +1804,7 @@ void RTPSender::GenericFECStatus(bool* enable,
                                     uint8_t* payload_type_red,
                                     uint8_t* payload_type_fec) const {
   RTC_DCHECK(!audio_configured_);
-  video_->GenericFECStatus(*enable, *payload_type_red, *payload_type_fec);
+  video_->GenericFECStatus(enable, payload_type_red, payload_type_fec);
 }
 
 int32_t RTPSender::SetFecParameters(
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 3bc861ccda..86407f9f40 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -68,7 +68,7 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
     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
@@ -96,13 +96,13 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
     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;
 }
 
@@ -384,13 +384,13 @@ int32_t RTPSenderAudio::SetRED(int8_t payloadType) {
 }
 
 // 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;
 }
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index 381bc13f97..a3cee5e707 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -31,7 +31,7 @@ class RTPSenderAudio : public DTMFqueue {
                                uint32_t frequency,
                                uint8_t channels,
                                uint32_t rate,
-                               RtpUtility::Payload*& payload);
+                               RtpUtility::Payload** payload);
 
   int32_t SendAudio(FrameType frameType,
                     int8_t payloadType,
@@ -58,7 +58,7 @@ class RTPSenderAudio : public DTMFqueue {
   int32_t SetRED(int8_t payloadType);
 
   // Get payload type for Redundant Audio Data RFC 2198
-  int32_t RED(int8_t& payloadType) const;
+  int32_t RED(int8_t* payloadType) const;
 
  protected:
   int32_t SendTelephoneEventPacket(
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 80967b2966..1ca7831ab2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -1281,9 +1281,9 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
   // For Telephone events, payload is not added to the registered payload list,
   // it will register only the payload used for audio stream.
   // Registering the payload again for audio stream with different payload name.
-  strcpy(payload_name, "payload_name");
+  const char kPayloadName[] = "payload_name";
   ASSERT_EQ(
-      0, rtp_sender_->RegisterPayload(payload_name, payload_type, 8000, 1, 0));
+      0, rtp_sender_->RegisterPayload(kPayloadName, payload_type, 8000, 1, 0));
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   // DTMF event key=9, duration=500 and attenuationdB=10
   rtp_sender_->SendTelephoneEvent(9, 500, 10);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index bec3623656..e4604f97c9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -190,13 +190,13 @@ void RTPSenderVideo::SetGenericFECStatus(const bool enable,
       kFecMaskRandom;
 }
 
-void RTPSenderVideo::GenericFECStatus(bool& enable,
-                                      uint8_t& payloadTypeRED,
-                                      uint8_t& payloadTypeFEC) const {
+void RTPSenderVideo::GenericFECStatus(bool* enable,
+                                      uint8_t* payloadTypeRED,
+                                      uint8_t* payloadTypeFEC) const {
   CriticalSectionScoped cs(crit_.get());
-  enable = fec_enabled_;
-  payloadTypeRED = red_payload_type_;
-  payloadTypeFEC = fec_payload_type_;
+  *enable = fec_enabled_;
+  *payloadTypeRED = red_payload_type_;
+  *payloadTypeFEC = fec_payload_type_;
 }
 
 size_t RTPSenderVideo::FECPacketOverhead() const {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
index 465240e581..e59321ab93 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -67,9 +67,9 @@ class RTPSenderVideo {
                            const uint8_t payloadTypeRED,
                            const uint8_t payloadTypeFEC);
 
-  void GenericFECStatus(bool& enable,
-                        uint8_t& payloadTypeRED,
-                        uint8_t& payloadTypeFEC) const;
+  void GenericFECStatus(bool* enable,
+                        uint8_t* payloadTypeRED,
+                        uint8_t* payloadTypeFEC) const;
 
   void SetFecParameters(const FecProtectionParams* delta_params,
                         const FecProtectionParams* key_params);