[rtp_rtcp] Lint errors cleaned from rtp_utility

R=åsapersson
BUG=webrtc:5277

Review URL: https://codereview.webrtc.org/1539423003

Cr-Commit-Position: refs/heads/master@{#11131}

8 files changed, 109 insertions, 165 deletions

diff --git a/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg b/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg
index 7edbb96e9b..c318452482 100644
--- a/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg
+++ b/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg
@@ -1,7 +1,5 @@
 #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
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
index 82c813fd76..d4cbe544cc 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
@@ -58,7 +58,7 @@ bool RtpHeaderParserImpl::Parse(const uint8_t* packet,
     rtp_header_extension_map_.GetCopy(&map);
   }
 
-  const bool valid_rtpheader = rtp_parser.Parse(*header, &map);
+  const bool valid_rtpheader = rtp_parser.Parse(header, &map);
   if (!valid_rtpheader) {
     return false;
   }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 940d12b621..6ad666b01a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -579,7 +579,7 @@ size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send) {
       break;
     RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
     RTPHeader rtp_header;
-    rtp_parser.Parse(rtp_header);
+    rtp_parser.Parse(&rtp_header);
     bytes_left -= static_cast<int>(length - rtp_header.headerLength);
   }
   return bytes_to_send - bytes_left;
@@ -589,8 +589,7 @@ void RTPSender::BuildPaddingPacket(uint8_t* packet,
                                    size_t header_length,
                                    size_t padding_length) {
   packet[0] |= 0x20;  // Set padding bit.
-  int32_t *data =
-      reinterpret_cast<int32_t *>(&(packet[header_length]));
+  int32_t* data = reinterpret_cast<int32_t*>(&(packet[header_length]));
 
   // Fill data buffer with random data.
   for (size_t j = 0; j < (padding_length >> 2); ++j) {
@@ -671,7 +670,7 @@ size_t RTPSender::SendPadData(size_t bytes,
 
     RtpUtility::RtpHeaderParser rtp_parser(padding_packet, length);
     RTPHeader rtp_header;
-    rtp_parser.Parse(rtp_header);
+    rtp_parser.Parse(&rtp_header);
 
     if (capture_time_ms > 0) {
       UpdateTransmissionTimeOffset(
@@ -723,7 +722,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
   if (paced_sender_) {
     RtpUtility::RtpHeaderParser rtp_parser(data_buffer, length);
     RTPHeader header;
-    if (!rtp_parser.Parse(header)) {
+    if (!rtp_parser.Parse(&header)) {
       assert(false);
       return -1;
     }
@@ -909,11 +908,11 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
                                      int64_t capture_time_ms,
                                      bool send_over_rtx,
                                      bool is_retransmit) {
-  uint8_t *buffer_to_send_ptr = buffer;
+  uint8_t* buffer_to_send_ptr = buffer;
 
   RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
   RTPHeader rtp_header;
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
   if (!is_retransmit && rtp_header.markerBit) {
     TRACE_EVENT_ASYNC_END0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PacedSend",
                            capture_time_ms);
@@ -1032,7 +1031,7 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
   RtpUtility::RtpHeaderParser rtp_parser(buffer,
                                          payload_length + rtp_header_length);
   RTPHeader rtp_header;
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
 
   int64_t now_ms = clock_->TimeInMilliseconds();
 
@@ -1175,7 +1174,7 @@ size_t RTPSender::CreateRtpHeader(uint8_t* header,
   int32_t rtp_header_length = kRtpHeaderLength;
 
   if (csrcs.size() > 0) {
-    uint8_t *ptr = &header[rtp_header_length];
+    uint8_t* ptr = &header[rtp_header_length];
     for (size_t i = 0; i < csrcs.size(); ++i) {
       ByteWriter<uint32_t>::WriteBigEndian(ptr, csrcs[i]);
       ptr += 4;
@@ -1827,7 +1826,7 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
       reinterpret_cast<const uint8_t*>(buffer), *length);
 
   RTPHeader rtp_header;
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
 
   // Add original RTP header.
   memcpy(data_buffer_rtx, buffer, rtp_header.headerLength);
@@ -1840,7 +1839,7 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
   }
 
   // Replace sequence number.
-  uint8_t *ptr = data_buffer_rtx + 2;
+  uint8_t* ptr = data_buffer_rtx + 2;
   ByteWriter<uint16_t>::WriteBigEndian(ptr, sequence_number_rtx_++);
 
   // Replace SSRC.
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 86407f9f40..d361443cee 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -350,7 +350,7 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
   size_t packetSize = payloadSize + rtpHeaderLength;
   RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
   RTPHeader rtp_header;
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
   _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
                                (frameType == kAudioFrameSpeech),
                                audio_level_dbov);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 1ca7831ab2..6bc122201a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -208,7 +208,7 @@ class RtpSenderVideoTest : public RtpSenderTest {
     } else {
       ASSERT_EQ(kRtpHeaderSize, length);
     }
-    ASSERT_TRUE(rtp_parser.Parse(rtp_header, map));
+    ASSERT_TRUE(rtp_parser.Parse(&rtp_header, map));
     ASSERT_FALSE(rtp_parser.RTCP());
     EXPECT_EQ(payload_, rtp_header.payloadType);
     EXPECT_EQ(seq_num, rtp_header.sequenceNumber);
@@ -335,7 +335,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacket) {
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, nullptr);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, nullptr);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -370,7 +370,7 @@ TEST_F(RtpSenderTestWithoutPacer,
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionTransmissionTimeOffset,
                kTransmissionTimeOffsetExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -381,7 +381,7 @@ TEST_F(RtpSenderTestWithoutPacer,
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
-  const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
+  const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
@@ -410,7 +410,7 @@ TEST_F(RtpSenderTestWithoutPacer,
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionTransmissionTimeOffset,
                kTransmissionTimeOffsetExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -437,7 +437,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) {
 
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -448,7 +448,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) {
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
-  const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
+  const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
@@ -476,7 +476,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithVideoRotation_MarkerBit) {
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header, &map));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header, &map));
   ASSERT_FALSE(rtp_parser.RTCP());
   VerifyRTPHeaderCommon(rtp_header);
   EXPECT_EQ(length, rtp_header.headerLength);
@@ -504,7 +504,7 @@ TEST_F(RtpSenderTestWithoutPacer,
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header, &map));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header, &map));
   ASSERT_FALSE(rtp_parser.RTCP());
   VerifyRTPHeaderCommon(rtp_header, false);
   EXPECT_EQ(length, rtp_header.headerLength);
@@ -525,12 +525,12 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
   webrtc::RTPHeader rtp_header;
 
   // Updating audio level is done in RTPSenderAudio, so simulate it here.
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
   rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel);
 
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -542,7 +542,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
-  const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
+  const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
@@ -579,7 +579,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
   webrtc::RTPHeader rtp_header;
 
   // Updating audio level is done in RTPSenderAudio, so simulate it here.
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
   rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel);
 
   RtpHeaderExtensionMap map;
@@ -589,7 +589,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
   map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
   map.Register(kRtpExtensionTransportSequenceNumber,
                kTransportSequenceNumberExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -608,7 +608,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
-  const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
+  const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
@@ -667,7 +667,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
   map.Register(kRtpExtensionTransmissionTimeOffset,
                kTransmissionTimeOffsetExtensionId);
   map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
   ASSERT_TRUE(valid_rtp_header);
 
   // Verify transmission time offset.
@@ -727,7 +727,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
   map.Register(kRtpExtensionTransmissionTimeOffset,
                kTransmissionTimeOffsetExtensionId);
   map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
   ASSERT_TRUE(valid_rtp_header);
 
   // Verify transmission time offset.
@@ -934,7 +934,7 @@ TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
   const uint8_t* payload_data =
       GetPayloadData(rtp_header, transport_.last_sent_packet_);
@@ -959,7 +959,7 @@ TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
 
   RtpUtility::RtpHeaderParser rtp_parser2(transport_.last_sent_packet_,
                                           transport_.last_sent_packet_len_);
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
   payload_data = GetPayloadData(rtp_header, transport_.last_sent_packet_);
   generic_header = *payload_data++;
@@ -1217,7 +1217,7 @@ TEST_F(RtpSenderAudioTest, SendAudio) {
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
   const uint8_t* payload_data =
       GetPayloadData(rtp_header, transport_.last_sent_packet_);
@@ -1246,7 +1246,7 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
   const uint8_t* payload_data =
       GetPayloadData(rtp_header, transport_.last_sent_packet_);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index e4604f97c9..5a565dfa99 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -304,7 +304,7 @@ int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
       size_t packetSize = payloadSize + rtp_header_length;
       RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
       RTPHeader rtp_header;
-      rtp_parser.Parse(rtp_header);
+      rtp_parser.Parse(&rtp_header);
       _rtpSender.UpdateVideoRotation(dataBuffer, packetSize, rtp_header,
                                      rtpHdr->rotation);
     }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
index bd7df42288..43433b94d9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
@@ -10,38 +10,10 @@
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
-#include <assert.h>
-#include <math.h>  // ceil
-#include <string.h>  // memcpy
-
-#if defined(_WIN32)
-// Order for these headers are important
-#include <winsock2.h>  // timeval
-#include <windows.h>  // FILETIME NOLINT(build/include_alpha)
-#include <MMSystem.h>  // timeGetTime
-#elif ((defined WEBRTC_LINUX) || (defined WEBRTC_MAC))
-#include <sys/time.h>  // gettimeofday
-#include <time.h>
-#endif
-#if (!defined(NDEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
-#include <stdio.h>
-#endif
+#include <string.h>
 
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-
-#if (!defined(NDEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
-#define DEBUG_PRINT(...)           \
-  {                                \
-    char msg[256];                 \
-    sprintf(msg, __VA_ARGS__);     \
-    OutputDebugString(msg);        \
-  }
-#else
-// special fix for visual 2003
-#define DEBUG_PRINT(exp)        ((void)0)
-#endif  // !defined(NDEBUG) && defined(_WIN32)
 
 namespace webrtc {
 
@@ -83,12 +55,12 @@ enum {
 #if defined(_WIN32)
 bool StringCompare(const char* str1, const char* str2,
                    const uint32_t length) {
-  return (_strnicmp(str1, str2, length) == 0) ? true : false;
+  return _strnicmp(str1, str2, length) == 0;
 }
 #elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
 bool StringCompare(const char* str1, const char* str2,
                    const uint32_t length) {
-  return (strncasecmp(str1, str2, length) == 0) ? true : false;
+  return strncasecmp(str1, str2, length) == 0;
 }
 #endif
 
@@ -99,10 +71,6 @@ size_t Word32Align(size_t size) {
   return size;
 }
 
-uint32_t pow2(uint8_t exp) {
-  return 1 << exp;
-}
-
 RtpHeaderParser::RtpHeaderParser(const uint8_t* rtpData,
                                  const size_t rtpDataLength)
     : _ptrRTPDataBegin(rtpData),
@@ -212,7 +180,7 @@ bool RtpHeaderParser::ParseRtcp(RTPHeader* header) const {
   return true;
 }
 
-bool RtpHeaderParser::Parse(RTPHeader& header,
+bool RtpHeaderParser::Parse(RTPHeader* header,
                             RtpHeaderExtensionMap* ptrExtensionMap) const {
   const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
   if (length < kRtpMinParseLength) {
@@ -251,39 +219,39 @@ bool RtpHeaderParser::Parse(RTPHeader& header,
     return false;
   }
 
-  header.markerBit      = M;
-  header.payloadType    = PT;
-  header.sequenceNumber = sequenceNumber;
-  header.timestamp      = RTPTimestamp;
-  header.ssrc           = SSRC;
-  header.numCSRCs       = CC;
-  header.paddingLength  = P ? *(_ptrRTPDataEnd - 1) : 0;
+  header->markerBit      = M;
+  header->payloadType    = PT;
+  header->sequenceNumber = sequenceNumber;
+  header->timestamp      = RTPTimestamp;
+  header->ssrc           = SSRC;
+  header->numCSRCs       = CC;
+  header->paddingLength  = P ? *(_ptrRTPDataEnd - 1) : 0;
 
   for (uint8_t i = 0; i < CC; ++i) {
     uint32_t CSRC = ByteReader<uint32_t>::ReadBigEndian(ptr);
     ptr += 4;
-    header.arrOfCSRCs[i] = CSRC;
+    header->arrOfCSRCs[i] = CSRC;
   }
 
-  header.headerLength   = 12 + CSRCocts;
+  header->headerLength   = 12 + CSRCocts;
 
   // If in effect, MAY be omitted for those packets for which the offset
   // is zero.
-  header.extension.hasTransmissionTimeOffset = false;
-  header.extension.transmissionTimeOffset = 0;
+  header->extension.hasTransmissionTimeOffset = false;
+  header->extension.transmissionTimeOffset = 0;
 
   // May not be present in packet.
-  header.extension.hasAbsoluteSendTime = false;
-  header.extension.absoluteSendTime = 0;
+  header->extension.hasAbsoluteSendTime = false;
+  header->extension.absoluteSendTime = 0;
 
   // May not be present in packet.
-  header.extension.hasAudioLevel = false;
-  header.extension.voiceActivity = false;
-  header.extension.audioLevel = 0;
+  header->extension.hasAudioLevel = false;
+  header->extension.voiceActivity = false;
+  header->extension.audioLevel = 0;
 
   // May not be present in packet.
-  header.extension.hasVideoRotation = false;
-  header.extension.videoRotation = 0;
+  header->extension.hasVideoRotation = false;
+  header->extension.videoRotation = 0;
 
   if (X) {
     /* RTP header extension, RFC 3550.
@@ -300,7 +268,7 @@ bool RtpHeaderParser::Parse(RTPHeader& header,
       return false;
     }
 
-    header.headerLength += 4;
+    header->headerLength += 4;
 
     uint16_t definedByProfile = ByteReader<uint16_t>::ReadBigEndian(ptr);
     ptr += 2;
@@ -320,15 +288,16 @@ bool RtpHeaderParser::Parse(RTPHeader& header,
                                   ptrRTPDataExtensionEnd,
                                   ptr);
     }
-    header.headerLength += XLen;
+    header->headerLength += XLen;
   }
-  if (header.headerLength + header.paddingLength > static_cast<size_t>(length))
+  if (header->headerLength + header->paddingLength >
+      static_cast<size_t>(length))
     return false;
   return true;
 }
 
 void RtpHeaderParser::ParseOneByteExtensionHeader(
-    RTPHeader& header,
+    RTPHeader* header,
     const RtpHeaderExtensionMap* ptrExtensionMap,
     const uint8_t* ptrRTPDataExtensionEnd,
     const uint8_t* ptr) const {
@@ -374,9 +343,9 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
           // |  ID   | len=2 |              transmission offset              |
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
-          header.extension.transmissionTimeOffset =
+          header->extension.transmissionTimeOffset =
               ByteReader<int32_t, 3>::ReadBigEndian(ptr);
-          header.extension.hasTransmissionTimeOffset = true;
+          header->extension.hasTransmissionTimeOffset = true;
           break;
         }
         case kRtpExtensionAudioLevel: {
@@ -390,9 +359,9 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
           // |  ID   | len=0 |V|   level     |
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
           //
-          header.extension.audioLevel = ptr[0] & 0x7f;
-          header.extension.voiceActivity = (ptr[0] & 0x80) != 0;
-          header.extension.hasAudioLevel = true;
+          header->extension.audioLevel = ptr[0] & 0x7f;
+          header->extension.voiceActivity = (ptr[0] & 0x80) != 0;
+          header->extension.hasAudioLevel = true;
           break;
         }
         case kRtpExtensionAbsoluteSendTime: {
@@ -406,9 +375,9 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
           // |  ID   | len=2 |              absolute send time               |
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
-          header.extension.absoluteSendTime =
+          header->extension.absoluteSendTime =
               ByteReader<uint32_t, 3>::ReadBigEndian(ptr);
-          header.extension.hasAbsoluteSendTime = true;
+          header->extension.hasAbsoluteSendTime = true;
           break;
         }
         case kRtpExtensionVideoRotation: {
@@ -422,8 +391,8 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
           // |  ID   | len=0 |0 0 0 0 C F R R|
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-          header.extension.hasVideoRotation = true;
-          header.extension.videoRotation = ptr[0];
+          header->extension.hasVideoRotation = true;
+          header->extension.videoRotation = ptr[0];
           break;
         }
         case kRtpExtensionTransportSequenceNumber: {
@@ -440,8 +409,8 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
 
           uint16_t sequence_number = ptr[0] << 8;
           sequence_number += ptr[1];
-          header.extension.transportSequenceNumber = sequence_number;
-          header.extension.hasTransportSequenceNumber = true;
+          header->extension.transportSequenceNumber = sequence_number;
+          header->extension.hasTransportSequenceNumber = true;
           break;
         }
         default: {
@@ -470,5 +439,4 @@ uint8_t RtpHeaderParser::ParsePaddingBytes(
   return num_zero_bytes;
 }
 }  // namespace RtpUtility
-
 }  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
index bdcb11ccc2..57f54c1afc 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
@@ -11,8 +11,6 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
 
-#include <stddef.h> // size_t, ptrdiff_t
-
 #include <map>
 
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
@@ -31,62 +29,43 @@ RtpAudioFeedback* NullObjectRtpAudioFeedback();
 ReceiveStatistics* NullObjectReceiveStatistics();
 
 namespace RtpUtility {
-    // January 1970, in NTP seconds.
-    const uint32_t NTP_JAN_1970 = 2208988800UL;
-
-    // Magic NTP fractional unit.
-    const double NTP_FRAC = 4.294967296E+9;
-
-    struct Payload
-    {
-        char name[RTP_PAYLOAD_NAME_SIZE];
-        bool audio;
-        PayloadUnion typeSpecific;
-    };
-
-    typedef std::map<int8_t, Payload*> PayloadTypeMap;
-
-    uint32_t pow2(uint8_t exp);
-
-    // Returns true if |newTimestamp| is older than |existingTimestamp|.
-    // |wrapped| will be set to true if there has been a wraparound between the
-    // two timestamps.
-    bool OldTimestamp(uint32_t newTimestamp,
-                      uint32_t existingTimestamp,
-                      bool* wrapped);
-
-    bool StringCompare(const char* str1,
-                       const char* str2,
-                       const uint32_t length);
-
-    // Round up to the nearest size that is a multiple of 4.
-    size_t Word32Align(size_t size);
-
-    class RtpHeaderParser {
-    public:
-     RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength);
-     ~RtpHeaderParser();
-
-        bool RTCP() const;
-        bool ParseRtcp(RTPHeader* header) const;
-        bool Parse(RTPHeader& parsedPacket,
-                   RtpHeaderExtensionMap* ptrExtensionMap = NULL) const;
-
-    private:
-        void ParseOneByteExtensionHeader(
-            RTPHeader& parsedPacket,
-            const RtpHeaderExtensionMap* ptrExtensionMap,
-            const uint8_t* ptrRTPDataExtensionEnd,
-            const uint8_t* ptr) const;
-
-        uint8_t ParsePaddingBytes(
-            const uint8_t* ptrRTPDataExtensionEnd,
-            const uint8_t* ptr) const;
-
-        const uint8_t* const _ptrRTPDataBegin;
-        const uint8_t* const _ptrRTPDataEnd;
-    };
+
+struct Payload {
+  char name[RTP_PAYLOAD_NAME_SIZE];
+  bool audio;
+  PayloadUnion typeSpecific;
+};
+
+typedef std::map<int8_t, Payload*> PayloadTypeMap;
+
+bool StringCompare(const char* str1, const char* str2, const uint32_t length);
+
+// Round up to the nearest size that is a multiple of 4.
+size_t Word32Align(size_t size);
+
+class RtpHeaderParser {
+ public:
+  RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength);
+  ~RtpHeaderParser();
+
+  bool RTCP() const;
+  bool ParseRtcp(RTPHeader* header) const;
+  bool Parse(RTPHeader* parsedPacket,
+             RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const;
+
+ private:
+  void ParseOneByteExtensionHeader(RTPHeader* parsedPacket,
+                                   const RtpHeaderExtensionMap* ptrExtensionMap,
+                                   const uint8_t* ptrRTPDataExtensionEnd,
+                                   const uint8_t* ptr) const;
+
+  uint8_t ParsePaddingBytes(const uint8_t* ptrRTPDataExtensionEnd,
+                            const uint8_t* ptr) const;
+
+  const uint8_t* const _ptrRTPDataBegin;
+  const uint8_t* const _ptrRTPDataEnd;
+};
 }  // namespace RtpUtility
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_