diff --git a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
index 1e2ae0db84..8f58e028de 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
@@ -54,7 +54,7 @@ class RTPPayloadStrategy {
 class RTPPayloadRegistry {
  public:
   // The registry takes ownership of the strategy.
-  RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy);
+  explicit RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy);
   ~RTPPayloadRegistry();
 
   int32_t RegisterReceivePayload(
@@ -110,8 +110,16 @@ class RTPPayloadRegistry {
 
   int GetPayloadTypeFrequency(uint8_t payload_type) const;
 
+  // DEPRECATED. Use PayloadTypeToPayload below that returns const Payload*
+  // instead of taking output parameter.
+  // TODO(danilchap): Remove this when all callers have been updated.
   bool PayloadTypeToPayload(const uint8_t payload_type,
-                            RtpUtility::Payload*& payload) const;
+                            RtpUtility::Payload*& payload) const {  // NOLINT
+    payload =
+        const_cast<RtpUtility::Payload*>(PayloadTypeToPayload(payload_type));
+    return payload != nullptr;
+  }
+  const RtpUtility::Payload* PayloadTypeToPayload(uint8_t payload_type) const;
 
   void ResetLastReceivedPayloadTypes() {
     CriticalSectionScoped cs(crit_sect_.get());
@@ -147,7 +155,7 @@ class RTPPayloadRegistry {
   int8_t last_received_media_payload_type() const {
     CriticalSectionScoped cs(crit_sect_.get());
     return last_received_media_payload_type_;
-  };
+  }
 
   bool use_rtx_payload_mapping_on_restore() const {
     CriticalSectionScoped cs(crit_sect_.get());
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
index a65f573337..6a7022a94c 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -581,9 +581,13 @@ class RtpRtcp : public Module {
     *
     *   return -1 on failure else 0
     */
-     virtual int32_t SendREDPayloadType(
-         int8_t& payloadType) const = 0;
-
+    // DEPRECATED. Use SendREDPayloadType below that takes output parameter
+    // by pointer instead of by reference.
+    // TODO(danilchap): Remove this when all callers have been updated.
+    int32_t SendREDPayloadType(int8_t& payloadType) const {  // NOLINT
+      return SendREDPayloadType(&payloadType);
+    }
+    virtual int32_t SendREDPayloadType(int8_t* payload_type) const = 0;
      /*
      * Store the audio level in dBov for header-extension-for-audio-level-
      * indication.
@@ -615,10 +619,17 @@ class RtpRtcp : public Module {
     /*
     *   Get generic FEC setting
     */
-    virtual void GenericFECStatus(bool& enable,
-                                     uint8_t& payloadTypeRED,
-                                     uint8_t& payloadTypeFEC) = 0;
-
+    // DEPRECATED. Use GenericFECStatus below that takes output parameters
+    // by pointers instead of by references.
+    // TODO(danilchap): Remove this when all callers have been updated.
+    void GenericFECStatus(bool& enable,               // NOLINT
+                          uint8_t& payloadTypeRED,    // NOLINT
+                          uint8_t& payloadTypeFEC) {  // NOLINT
+      GenericFECStatus(&enable, &payloadTypeRED, &payloadTypeFEC);
+    }
+    virtual void GenericFECStatus(bool* enable,
+                                  uint8_t* payload_type_red,
+                                  uint8_t* payload_type_fec) = 0;
 
     virtual int32_t SetFecParameters(
         const FecProtectionParams* delta_params,
@@ -639,4 +650,4 @@ class RtpRtcp : public Module {
     virtual int32_t RequestKeyFrame() = 0;
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 6373de28e7..d2ac62e845 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -33,31 +33,23 @@ const int kVideoPayloadTypeFrequency = 90000;
 // Minimum RTP header size in bytes.
 const uint8_t kRtpHeaderSize = 12;
 
-struct AudioPayload
-{
+struct AudioPayload {
     uint32_t    frequency;
     uint8_t     channels;
     uint32_t    rate;
 };
 
-struct VideoPayload
-{
+struct VideoPayload {
     RtpVideoCodecTypes   videoCodecType;
     uint32_t       maxRate;
 };
 
-union PayloadUnion
-{
+union PayloadUnion {
     AudioPayload Audio;
     VideoPayload Video;
 };
 
-enum RTPAliveType
-{
-    kRtpDead   = 0,
-    kRtpNoRtp = 1,
-    kRtpAlive  = 2
-};
+enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 };
 
 enum ProtectionType {
   kUnprotectedPacket,
@@ -78,10 +70,7 @@ enum RTPExtensionType {
   kRtpExtensionTransportSequenceNumber,
 };
 
-enum RTCPAppSubTypes
-{
-    kAppSubtypeBwe     = 0x00
-};
+enum RTCPAppSubTypes { kAppSubtypeBwe = 0x00 };
 
 // TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up.
 enum RTCPPacketType : uint32_t {
@@ -109,17 +98,9 @@ enum RTCPPacketType : uint32_t {
 
 enum KeyFrameRequestMethod { kKeyFrameReqPliRtcp, kKeyFrameReqFirRtcp };
 
-enum RtpRtcpPacketType
-{
-    kPacketRtp        = 0,
-    kPacketKeepAlive = 1
-};
+enum RtpRtcpPacketType { kPacketRtp = 0, kPacketKeepAlive = 1 };
 
-enum NACKMethod
-{
-    kNackOff      = 0,
-    kNackRtcp     = 2
-};
+enum NACKMethod { kNackOff = 0, kNackRtcp = 2 };
 
 enum RetransmissionMode : uint8_t {
   kRetransmitOff = 0x0,
@@ -138,8 +119,7 @@ enum RtxMode {
 
 const size_t kRtxHeaderSize = 2;
 
-struct RTCPSenderInfo
-{
+struct RTCPSenderInfo {
     uint32_t NTPseconds;
     uint32_t NTPfraction;
     uint32_t RTPtimeStamp;
@@ -206,40 +186,36 @@ struct RtpState {
   bool media_has_been_sent;
 };
 
-class RtpData
-{
-public:
-    virtual ~RtpData() {}
+class RtpData {
+ public:
+  virtual ~RtpData() {}
 
-    virtual int32_t OnReceivedPayloadData(
-        const uint8_t* payloadData,
-        const size_t payloadSize,
-        const WebRtcRTPHeader* rtpHeader) = 0;
+  virtual int32_t OnReceivedPayloadData(const uint8_t* payloadData,
+                                        const size_t payloadSize,
+                                        const WebRtcRTPHeader* rtpHeader) = 0;
 
-    virtual bool OnRecoveredPacket(const uint8_t* packet,
-                                   size_t packet_length) = 0;
+  virtual bool OnRecoveredPacket(const uint8_t* packet,
+                                 size_t packet_length) = 0;
 };
 
-class RtpFeedback
-{
-public:
-    virtual ~RtpFeedback() {}
+class RtpFeedback {
+ public:
+  virtual ~RtpFeedback() {}
 
-    // Receiving payload change or SSRC change. (return success!)
-    /*
-    *   channels    - number of channels in codec (1 = mono, 2 = stereo)
-    */
-    virtual int32_t OnInitializeDecoder(
-        const int8_t payloadType,
-        const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-        const int frequency,
-        const uint8_t channels,
-        const uint32_t rate) = 0;
+  // Receiving payload change or SSRC change. (return success!)
+  /*
+  *   channels    - number of channels in codec (1 = mono, 2 = stereo)
+  */
+  virtual int32_t OnInitializeDecoder(
+      const int8_t payloadType,
+      const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+      const int frequency,
+      const uint8_t channels,
+      const uint32_t rate) = 0;
 
-    virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0;
+  virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0;
 
-    virtual void OnIncomingCSRCChanged(const uint32_t CSRC,
-                                       const bool added) = 0;
+  virtual void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) = 0;
 };
 
 class RtpAudioFeedback {
@@ -346,7 +322,7 @@ class RtcpRttStats {
 
   virtual int64_t LastProcessedRtt() const = 0;
 
-  virtual ~RtcpRttStats() {};
+  virtual ~RtcpRttStats() {}
 };
 
 // Null object version of RtpFeedback.
@@ -437,4 +413,4 @@ class TransportSequenceNumberAllocator {
 };
 
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 04abb31bfe..334d81172c 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -218,8 +218,7 @@ class MockRtpRtcp : public RtpRtcp {
       int32_t(const uint8_t key, const uint16_t time_ms, const uint8_t level));
   MOCK_METHOD1(SetSendREDPayloadType,
       int32_t(const int8_t payloadType));
-  MOCK_CONST_METHOD1(SendREDPayloadType,
-      int32_t(int8_t& payloadType));
+  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,
@@ -233,9 +232,9 @@ class MockRtpRtcp : public RtpRtcp {
                     const uint8_t payload_type_red,
                     const uint8_t payload_type_fec));
   MOCK_METHOD3(GenericFECStatus,
-               void(bool& enable,
-                    uint8_t& payloadTypeRED,
-                    uint8_t& payloadTypeFEC));
+               void(bool* enable,
+                    uint8_t* payloadTypeRED,
+                    uint8_t* payloadTypeFEC));
   MOCK_METHOD2(SetFecParameters,
       int32_t(const FecProtectionParams* delta_params,
               const FecProtectionParams* key_params));
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
index a5bfd71664..f7cdc4c7e3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -343,17 +343,16 @@ bool RTPPayloadRegistry::GetPayloadSpecifics(uint8_t payload_type,
 
 int RTPPayloadRegistry::GetPayloadTypeFrequency(
     uint8_t payload_type) const {
-  RtpUtility::Payload* payload;
-  if (!PayloadTypeToPayload(payload_type, payload)) {
+  const RtpUtility::Payload* payload = PayloadTypeToPayload(payload_type);
+  if (!payload) {
     return -1;
   }
   CriticalSectionScoped cs(crit_sect_.get());
   return rtp_payload_strategy_->GetPayloadTypeFrequency(*payload);
 }
 
-bool RTPPayloadRegistry::PayloadTypeToPayload(
-    const uint8_t payload_type,
-    RtpUtility::Payload*& payload) const {
+const RtpUtility::Payload* RTPPayloadRegistry::PayloadTypeToPayload(
+    uint8_t payload_type) const {
   CriticalSectionScoped cs(crit_sect_.get());
 
   RtpUtility::PayloadTypeMap::const_iterator it =
@@ -361,11 +360,10 @@ bool RTPPayloadRegistry::PayloadTypeToPayload(
 
   // Check that this is a registered payload type.
   if (it == payload_type_map_.end()) {
-    return false;
+    return nullptr;
   }
 
-  payload = it->second;
-  return true;
+  return it->second;
 }
 
 void RTPPayloadRegistry::SetIncomingPayloadType(const RTPHeader& header) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
index 1fd51a7385..f19c3daacd 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
@@ -74,9 +74,9 @@ TEST_F(RtpPayloadRegistryTest, RegistersAndRemembersPayloadsUntilDeregistered) {
 
   EXPECT_TRUE(new_payload_created) << "A new payload WAS created.";
 
-  RtpUtility::Payload* retrieved_payload = NULL;
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type,
-                                                          retrieved_payload));
+  const RtpUtility::Payload* retrieved_payload =
+      rtp_payload_registry_->PayloadTypeToPayload(payload_type);
+  EXPECT_TRUE(retrieved_payload);
 
   // We should get back the exact pointer to the payload returned by the
   // payload strategy.
@@ -84,8 +84,7 @@ TEST_F(RtpPayloadRegistryTest, RegistersAndRemembersPayloadsUntilDeregistered) {
 
   // Now forget about it and verify it's gone.
   EXPECT_EQ(0, rtp_payload_registry_->DeRegisterReceivePayload(payload_type));
-  EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload(payload_type,
-                                                           retrieved_payload));
+  EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload(payload_type));
 }
 
 TEST_F(RtpPayloadRegistryTest, AudioRedWorkProperly) {
@@ -106,10 +105,9 @@ TEST_F(RtpPayloadRegistryTest, AudioRedWorkProperly) {
 
   EXPECT_EQ(kRedPayloadType, rtp_payload_registry_->red_payload_type());
 
-  RtpUtility::Payload* retrieved_payload = NULL;
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(kRedPayloadType,
-                                                          retrieved_payload));
-  ASSERT_TRUE(retrieved_payload);
+  const RtpUtility::Payload* retrieved_payload =
+      rtp_payload_registry_->PayloadTypeToPayload(kRedPayloadType);
+  EXPECT_TRUE(retrieved_payload);
   EXPECT_TRUE(retrieved_payload->audio);
   EXPECT_STRCASEEQ("red", retrieved_payload->name);
 
@@ -142,12 +140,13 @@ TEST_F(RtpPayloadRegistryTest,
       << "With a different payload type is fine though.";
 
   // Ensure both payloads are preserved.
-  RtpUtility::Payload* retrieved_payload = NULL;
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type,
-                                                          retrieved_payload));
+  const RtpUtility::Payload* retrieved_payload =
+      rtp_payload_registry_->PayloadTypeToPayload(payload_type);
+  EXPECT_TRUE(retrieved_payload);
   EXPECT_EQ(first_payload_on_heap, retrieved_payload);
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type - 1,
-                                                          retrieved_payload));
+  retrieved_payload =
+      rtp_payload_registry_->PayloadTypeToPayload(payload_type - 1);
+  EXPECT_TRUE(retrieved_payload);
   EXPECT_EQ(second_payload_on_heap, retrieved_payload);
 
   // Ok, update the rate for one of the codecs. If either the incoming rate or
@@ -181,13 +180,10 @@ TEST_F(RtpPayloadRegistryTest,
                    kTypicalPayloadName, payload_type - 1, kTypicalFrequency,
                    kTypicalChannels, kTypicalRate, &ignored));
 
-  RtpUtility::Payload* retrieved_payload = NULL;
-  EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload(payload_type,
-                                                           retrieved_payload))
+  EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload(payload_type))
       << "The first payload should be "
          "deregistered because the only thing that differs is payload type.";
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type - 1,
-                                                          retrieved_payload))
+  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type - 1))
       << "The second payload should still be registered though.";
 
   // Now ensure non-compatible codecs aren't removed.
@@ -198,11 +194,9 @@ TEST_F(RtpPayloadRegistryTest,
                    kTypicalPayloadName, payload_type + 1, kTypicalFrequency,
                    kTypicalChannels, kTypicalRate, &ignored));
 
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type - 1,
-                                                          retrieved_payload))
+  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type - 1))
       << "Not compatible; both payloads should be kept.";
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type + 1,
-                                                          retrieved_payload))
+  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type + 1))
       << "Not compatible; both payloads should be kept.";
 }
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index bd7bd1b82e..bdc05c6f7b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -276,12 +276,11 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
         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) {
@@ -365,12 +364,12 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
         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);
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 3d20bab31f..30b1b96d93 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -800,9 +800,8 @@ int32_t ModuleRtpRtcpImpl::SetSendREDPayloadType(
 }
 
 // Get payload type for Redundant Audio Data RFC 2198.
-int32_t ModuleRtpRtcpImpl::SendREDPayloadType(
-    int8_t& payload_type) const {
-  return rtp_sender_.RED(&payload_type);
+int32_t ModuleRtpRtcpImpl::SendREDPayloadType(int8_t* payload_type) const {
+  return rtp_sender_.RED(payload_type);
 }
 
 void ModuleRtpRtcpImpl::SetTargetSendBitrate(uint32_t bitrate_bps) {
@@ -838,11 +837,10 @@ void ModuleRtpRtcpImpl::SetGenericFECStatus(
   rtp_sender_.SetGenericFECStatus(enable, payload_type_red, payload_type_fec);
 }
 
-void ModuleRtpRtcpImpl::GenericFECStatus(bool& enable,
-                                         uint8_t& payload_type_red,
-                                         uint8_t& payload_type_fec) {
-  rtp_sender_.GenericFECStatus(&enable, &payload_type_red,
-                                      &payload_type_fec);
+void ModuleRtpRtcpImpl::GenericFECStatus(bool* enable,
+                                         uint8_t* payload_type_red,
+                                         uint8_t* payload_type_fec) {
+  rtp_sender_.GenericFECStatus(enable, payload_type_red, payload_type_fec);
 }
 
 int32_t ModuleRtpRtcpImpl::SetFecParameters(
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index a65289c708..24cbbe4079 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -260,7 +260,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   int32_t SetSendREDPayloadType(int8_t payload_type) override;
 
   // Get payload type for Redundant Audio Data RFC 2198.
-  int32_t SendREDPayloadType(int8_t& payload_type) const override;
+  int32_t SendREDPayloadType(int8_t* payload_type) const override;
 
   // Store the audio level in d_bov for header-extension-for-audio-level-
   // indication.
@@ -282,9 +282,9 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
                            uint8_t payload_type_red,
                            uint8_t payload_type_fec) override;
 
-  void GenericFECStatus(bool& enable,
-                        uint8_t& payload_type_red,
-                        uint8_t& payload_type_fec) override;
+  void GenericFECStatus(bool* enable,
+                        uint8_t* payload_type_red,
+                        uint8_t* payload_type_fec) override;
 
   int32_t SetFecParameters(const FecProtectionParams* delta_params,
                            const FecProtectionParams* key_params) override;
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 36456f125e..5fd940dbcc 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -241,7 +241,7 @@ TEST_F(RtpRtcpAudioTest, RED) {
 
   EXPECT_EQ(0, module1->SetSendREDPayloadType(voice_codec.pltype));
   int8_t red = 0;
-  EXPECT_EQ(0, module1->SendREDPayloadType(red));
+  EXPECT_EQ(0, module1->SendREDPayloadType(&red));
   EXPECT_EQ(voice_codec.pltype, red);
   EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
       voice_codec.plname,
@@ -278,7 +278,7 @@ TEST_F(RtpRtcpAudioTest, RED) {
                                          &fragmentation));
 
   EXPECT_EQ(0, module1->SetSendREDPayloadType(-1));
-  EXPECT_EQ(-1, module1->SendREDPayloadType(red));
+  EXPECT_EQ(-1, module1->SendREDPayloadType(&red));
 }
 
 TEST_F(RtpRtcpAudioTest, DTMF) {
diff --git a/webrtc/video/vie_channel.cc b/webrtc/video/vie_channel.cc
index 54bc72bd16..6e2406a9bd 100644
--- a/webrtc/video/vie_channel.cc
+++ b/webrtc/video/vie_channel.cc
@@ -263,8 +263,8 @@ void ViEChannel::UpdateHistograms() {
       bool fec_enabled = false;
       uint8_t pltype_red;
       uint8_t pltype_fec;
-      rtp_rtcp_modules_[0]->GenericFECStatus(fec_enabled, pltype_red,
-                                             pltype_fec);
+      rtp_rtcp_modules_[0]->GenericFECStatus(&fec_enabled, &pltype_red,
+                                             &pltype_fec);
       if (fec_enabled) {
         RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateSentInKbps",
                                    static_cast<int>(rtp_rtx.fec.TotalBytes() *
@@ -531,7 +531,7 @@ bool ViEChannel::IsSendingFecEnabled() {
   uint8_t pltype_fec = 0;
 
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
-    rtp_rtcp->GenericFECStatus(fec_enabled, pltype_red, pltype_fec);
+    rtp_rtcp->GenericFECStatus(&fec_enabled, &pltype_red, &pltype_fec);
     if (fec_enabled)
       return true;
   }
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 37dc3b685b..18f3408276 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -3269,9 +3269,8 @@ Channel::GetREDStatus(bool& enabled, int& redPayloadtype)
     enabled = audio_coding_->REDStatus();
     if (enabled)
     {
-        int8_t payloadType(0);
-        if (_rtpRtcpModule->SendREDPayloadType(payloadType) != 0)
-        {
+      int8_t payloadType = 0;
+      if (_rtpRtcpModule->SendREDPayloadType(&payloadType) != 0) {
             _engineStatisticsPtr->SetLastError(
                 VE_RTP_RTCP_MODULE_ERROR, kTraceError,
                 "GetREDStatus() failed to retrieve RED PT from RTP/RTCP "