diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index e54a29a..adfc0d5 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -24,21 +24,20 @@
                           const uint8_t* payloadData,
                           size_t payloadSize,
                           const RTPFragmentationHeader* fragmentation) {
-  WebRtcRTPHeader rtpInfo;
+  RTPHeader rtp_header;
   int32_t status;
   size_t payloadDataSize = payloadSize;
 
-  rtpInfo.header.markerBit = false;
-  rtpInfo.header.ssrc = 0;
-  rtpInfo.header.sequenceNumber =
+  rtp_header.markerBit = false;
+  rtp_header.ssrc = 0;
+  rtp_header.sequenceNumber =
       (external_sequence_number_ < 0)
           ? _seqNo++
           : static_cast<uint16_t>(external_sequence_number_);
-  rtpInfo.header.payloadType = payloadType;
-  rtpInfo.header.timestamp =
-      (external_send_timestamp_ < 0)
-          ? timeStamp
-          : static_cast<uint32_t>(external_send_timestamp_);
+  rtp_header.payloadType = payloadType;
+  rtp_header.timestamp = (external_send_timestamp_ < 0)
+                             ? timeStamp
+                             : static_cast<uint32_t>(external_send_timestamp_);
 
   if (frameType == kEmptyFrame) {
     // When frame is empty, we should not transmit it. The frame size of the
@@ -75,16 +74,16 @@
       memcpy(_payloadData, payloadData + fragmentation->fragmentationOffset[0],
              fragmentation->fragmentationLength[0]);
       payloadDataSize = fragmentation->fragmentationLength[0];
-      rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0];
+      rtp_header.payloadType = fragmentation->fragmentationPlType[0];
     }
   } else {
     memcpy(_payloadData, payloadData, payloadDataSize);
     if (_isStereo) {
       if (_leftChannel) {
-        memcpy(&_rtpInfo, &rtpInfo, sizeof(WebRtcRTPHeader));
+        _rtp_header = rtp_header;
         _leftChannel = false;
       } else {
-        memcpy(&rtpInfo, &_rtpInfo, sizeof(WebRtcRTPHeader));
+        rtp_header = _rtp_header;
         _leftChannel = true;
       }
     }
@@ -96,7 +95,7 @@
   }
 
   if (!_isStereo) {
-    CalcStatistics(rtpInfo, payloadSize);
+    CalcStatistics(rtp_header, payloadSize);
   }
   _useLastFrameSize = false;
   _lastInTimestamp = timeStamp;
@@ -116,16 +115,16 @@
     return 0;
   }
 
-  status = _receiverACM->IncomingPacket(_payloadData, payloadDataSize,
-                                        rtpInfo.header);
+  status =
+      _receiverACM->IncomingPacket(_payloadData, payloadDataSize, rtp_header);
 
   return status;
 }
 
 // TODO(turajs): rewite this method.
-void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
+void Channel::CalcStatistics(const RTPHeader& rtp_header, size_t payloadSize) {
   int n;
-  if ((rtpInfo.header.payloadType != _lastPayloadType) &&
+  if ((rtp_header.payloadType != _lastPayloadType) &&
       (_lastPayloadType != -1)) {
     // payload-type is changed.
     // we have to terminate the calculations on the previous payload type
@@ -138,12 +137,12 @@
       }
     }
   }
-  _lastPayloadType = rtpInfo.header.payloadType;
+  _lastPayloadType = rtp_header.payloadType;
 
   bool newPayload = true;
   ACMTestPayloadStats* currentPayloadStr = NULL;
   for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
-    if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) {
+    if (rtp_header.payloadType == _payloadStats[n].payloadType) {
       newPayload = false;
       currentPayloadStr = &_payloadStats[n];
       break;
@@ -154,7 +153,7 @@
     if (!currentPayloadStr->newPacket) {
       if (!_useLastFrameSize) {
         _lastFrameSizeSample =
-            (uint32_t)((uint32_t)rtpInfo.header.timestamp -
+            (uint32_t)((uint32_t)rtp_header.timestamp -
                        (uint32_t)currentPayloadStr->lastTimestamp);
       }
       assert(_lastFrameSizeSample > 0);
@@ -194,13 +193,13 @@
             currentPayloadStr->lastPayloadLenByte;
       }
       // store the current values for the next time
-      currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
+      currentPayloadStr->lastTimestamp = rtp_header.timestamp;
       currentPayloadStr->lastPayloadLenByte = payloadSize;
     } else {
       currentPayloadStr->newPacket = false;
       currentPayloadStr->lastPayloadLenByte = payloadSize;
-      currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
-      currentPayloadStr->payloadType = rtpInfo.header.payloadType;
+      currentPayloadStr->lastTimestamp = rtp_header.timestamp;
+      currentPayloadStr->payloadType = rtp_header.payloadType;
       memset(currentPayloadStr->frameSizeStats, 0,
              MAX_NUM_FRAMESIZES * sizeof(ACMTestFrameSizeStats));
     }
@@ -212,8 +211,8 @@
     // first packet
     _payloadStats[n].newPacket = false;
     _payloadStats[n].lastPayloadLenByte = payloadSize;
-    _payloadStats[n].lastTimestamp = rtpInfo.header.timestamp;
-    _payloadStats[n].payloadType = rtpInfo.header.payloadType;
+    _payloadStats[n].lastTimestamp = rtp_header.timestamp;
+    _payloadStats[n].payloadType = rtp_header.payloadType;
     memset(_payloadStats[n].frameSizeStats, 0,
            MAX_NUM_FRAMESIZES * sizeof(ACMTestFrameSizeStats));
   }
diff --git a/modules/audio_coding/test/Channel.h b/modules/audio_coding/test/Channel.h
index e428a71..4d7f0b7 100644
--- a/modules/audio_coding/test/Channel.h
+++ b/modules/audio_coding/test/Channel.h
@@ -81,7 +81,7 @@
   }
 
  private:
-  void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize);
+  void CalcStatistics(const RTPHeader& rtp_header, size_t payloadSize);
 
   AudioCodingModule* _receiverACM;
   uint16_t _seqNo;
@@ -94,7 +94,7 @@
   int16_t _lastPayloadType;
   ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
   bool _isStereo;
-  WebRtcRTPHeader _rtpInfo;
+  RTPHeader _rtp_header;
   bool _leftChannel;
   uint32_t _lastInTimestamp;
   bool _useLastFrameSize;
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index cd57ecd..28ee8aa 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -155,7 +155,7 @@
   if (!_rtpStream->EndOfFile()) {
     if (_firstTime) {
       _firstTime = false;
-      _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+      _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
                                                _payloadSizeBytes, &_nextTime);
       if (_realPayloadSizeBytes == 0) {
         if (_rtpStream->EndOfFile()) {
@@ -168,8 +168,8 @@
     }
 
     EXPECT_EQ(0, _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
-                                      _rtpInfo.header));
-    _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+                                      _rtpHeader));
+    _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
                                              _payloadSizeBytes, &_nextTime);
     if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
       _firstTime = true;
diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h
index df6ee5f..d9c22d7 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -84,7 +84,7 @@
   AudioCodingModule* _acm;
   uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
   RTPStream* _rtpStream;
-  WebRtcRTPHeader _rtpInfo;
+  RTPHeader _rtpHeader;
   size_t _realPayloadSizeBytes;
   size_t _payloadSizeBytes;
   uint32_t _nextTime;
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index fd76224..cbe066f 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -44,7 +44,7 @@
 bool ReceiverWithPacketLoss::IncomingPacket() {
   if (!_rtpStream->EndOfFile()) {
     if (packet_counter_ == 0) {
-      _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+      _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
                                                _payloadSizeBytes, &_nextTime);
       if (_realPayloadSizeBytes == 0) {
         if (_rtpStream->EndOfFile()) {
@@ -57,11 +57,10 @@
     }
 
     if (!PacketLost()) {
-      _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
-                           _rtpInfo.header);
+      _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpHeader);
     }
     packet_counter_++;
-    _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+    _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
                                              _payloadSizeBytes, &_nextTime);
     if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
       packet_counter_ = 0;
diff --git a/modules/audio_coding/test/RTPFile.cc b/modules/audio_coding/test/RTPFile.cc
index cfe93fa..1273fa8 100644
--- a/modules/audio_coding/test/RTPFile.cc
+++ b/modules/audio_coding/test/RTPFile.cc
@@ -25,19 +25,19 @@
 
 namespace webrtc {
 
-void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo,
+void RTPStream::ParseRTPHeader(RTPHeader* rtp_header,
                                const uint8_t* rtpHeader) {
-  rtpInfo->header.payloadType = rtpHeader[1];
-  rtpInfo->header.sequenceNumber =
+  rtp_header->payloadType = rtpHeader[1];
+  rtp_header->sequenceNumber =
       (static_cast<uint16_t>(rtpHeader[2]) << 8) | rtpHeader[3];
-  rtpInfo->header.timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) |
-                              (static_cast<uint32_t>(rtpHeader[5]) << 16) |
-                              (static_cast<uint32_t>(rtpHeader[6]) << 8) |
-                              rtpHeader[7];
-  rtpInfo->header.ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) |
-                         (static_cast<uint32_t>(rtpHeader[9]) << 16) |
-                         (static_cast<uint32_t>(rtpHeader[10]) << 8) |
-                         rtpHeader[11];
+  rtp_header->timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) |
+                          (static_cast<uint32_t>(rtpHeader[5]) << 16) |
+                          (static_cast<uint32_t>(rtpHeader[6]) << 8) |
+                          rtpHeader[7];
+  rtp_header->ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) |
+                     (static_cast<uint32_t>(rtpHeader[9]) << 16) |
+                     (static_cast<uint32_t>(rtpHeader[10]) << 8) |
+                     rtpHeader[11];
 }
 
 void RTPStream::MakeRTPheader(uint8_t* rtpHeader,
@@ -101,7 +101,7 @@
   _queueRWLock->ReleaseLockExclusive();
 }
 
-size_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo,
+size_t RTPBuffer::Read(RTPHeader* rtp_header,
                        uint8_t* payloadData,
                        size_t payloadSize,
                        uint32_t* offset) {
@@ -109,11 +109,11 @@
   RTPPacket* packet = _rtpQueue.front();
   _rtpQueue.pop();
   _queueRWLock->ReleaseLockShared();
-  rtpInfo->header.markerBit = 1;
-  rtpInfo->header.payloadType = packet->payloadType;
-  rtpInfo->header.sequenceNumber = packet->seqNo;
-  rtpInfo->header.ssrc = 0;
-  rtpInfo->header.timestamp = packet->timeStamp;
+  rtp_header->markerBit = 1;
+  rtp_header->payloadType = packet->payloadType;
+  rtp_header->sequenceNumber = packet->seqNo;
+  rtp_header->ssrc = 0;
+  rtp_header->timestamp = packet->timeStamp;
   if (packet->payloadSize > 0 && payloadSize >= packet->payloadSize) {
     memcpy(payloadData, packet->payloadData, packet->payloadSize);
   } else {
@@ -199,7 +199,7 @@
   EXPECT_EQ(payloadSize, fwrite(payloadData, 1, payloadSize, _rtpFile));
 }
 
-size_t RTPFile::Read(WebRtcRTPHeader* rtpInfo,
+size_t RTPFile::Read(RTPHeader* rtp_header,
                      uint8_t* payloadData,
                      size_t payloadSize,
                      uint32_t* offset) {
@@ -220,7 +220,7 @@
   EXPECT_GT(plen, 11);
 
   EXPECT_EQ(1u, fread(rtpHeader, 12, 1, _rtpFile));
-  ParseRTPHeader(rtpInfo, rtpHeader);
+  ParseRTPHeader(rtp_header, rtpHeader);
   EXPECT_EQ(lengthBytes, plen + 8);
 
   if (plen == 0) {
diff --git a/modules/audio_coding/test/RTPFile.h b/modules/audio_coding/test/RTPFile.h
index 141075b..1c555ed 100644
--- a/modules/audio_coding/test/RTPFile.h
+++ b/modules/audio_coding/test/RTPFile.h
@@ -33,7 +33,7 @@
 
   // Returns the packet's payload size. Zero should be treated as an
   // end-of-stream (in the case that EndOfFile() is true) or an error.
-  virtual size_t Read(WebRtcRTPHeader* rtpInfo,
+  virtual size_t Read(RTPHeader* rtp_Header,
                       uint8_t* payloadData,
                       size_t payloadSize,
                       uint32_t* offset) = 0;
@@ -46,7 +46,7 @@
                      uint32_t timeStamp,
                      uint32_t ssrc);
 
-  void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
+  void ParseRTPHeader(RTPHeader* rtp_header, const uint8_t* rtpHeader);
 };
 
 class RTPPacket {
@@ -81,7 +81,7 @@
              const size_t payloadSize,
              uint32_t frequency) override;
 
-  size_t Read(WebRtcRTPHeader* rtpInfo,
+  size_t Read(RTPHeader* rtp_header,
               uint8_t* payloadData,
               size_t payloadSize,
               uint32_t* offset) override;
@@ -114,7 +114,7 @@
              const size_t payloadSize,
              uint32_t frequency) override;
 
-  size_t Read(WebRtcRTPHeader* rtpInfo,
+  size_t Read(RTPHeader* rtp_header,
               uint8_t* payloadData,
               size_t payloadSize,
               uint32_t* offset) override;