[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}
diff --git a/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg b/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg
index 7edbb96..c318452 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 82c813f..d4cbe54 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 @@
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 940d12b..6ad666b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -579,7 +579,7 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
}
// 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 86407f9..d361443 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 @@
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 1ca7831..6bc1222 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 @@
} 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 @@
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 @@
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 @@
// 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 @@
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 @@
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 @@
// 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 @@
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 @@
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 @@
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 @@
// 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 @@
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 @@
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 @@
// 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 e4604f9..5a565df 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 @@
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 bd7df42..43433b9 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 @@
#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 @@
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 @@
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 @@
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 @@
return false;
}
- header.headerLength += 4;
+ header->headerLength += 4;
uint16_t definedByProfile = ByteReader<uint16_t>::ReadBigEndian(ptr);
ptr += 2;
@@ -320,15 +288,16 @@
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 @@
// | 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 @@
// | 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 @@
// | 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 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | 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 @@
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 @@
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 bdcb11c..57f54c1 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 @@
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;
+};
- struct Payload
- {
- char name[RTP_PAYLOAD_NAME_SIZE];
- bool audio;
- PayloadUnion typeSpecific;
- };
+typedef std::map<int8_t, Payload*> PayloadTypeMap;
- typedef std::map<int8_t, Payload*> PayloadTypeMap;
+bool StringCompare(const char* str1, const char* str2, const uint32_t length);
- uint32_t pow2(uint8_t exp);
+// Round up to the nearest size that is a multiple of 4.
+size_t Word32Align(size_t size);
- // 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);
+class RtpHeaderParser {
+ public:
+ RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength);
+ ~RtpHeaderParser();
- bool StringCompare(const char* str1,
- const char* str2,
- const uint32_t length);
+ bool RTCP() const;
+ bool ParseRtcp(RTPHeader* header) const;
+ bool Parse(RTPHeader* parsedPacket,
+ RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const;
- // Round up to the nearest size that is a multiple of 4.
- size_t Word32Align(size_t size);
+ private:
+ void ParseOneByteExtensionHeader(RTPHeader* parsedPacket,
+ const RtpHeaderExtensionMap* ptrExtensionMap,
+ const uint8_t* ptrRTPDataExtensionEnd,
+ const uint8_t* ptr) const;
- class RtpHeaderParser {
- public:
- RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength);
- ~RtpHeaderParser();
+ uint8_t ParsePaddingBytes(const uint8_t* ptrRTPDataExtensionEnd,
+ const uint8_t* ptr) const;
- 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;
- };
+ 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_
diff --git a/webrtc/test/layer_filtering_transport.cc b/webrtc/test/layer_filtering_transport.cc
index 9cf02ed..41d63ad 100644
--- a/webrtc/test/layer_filtering_transport.cc
+++ b/webrtc/test/layer_filtering_transport.cc
@@ -47,7 +47,7 @@
bool set_marker_bit = false;
RtpUtility::RtpHeaderParser parser(packet, length);
RTPHeader header;
- parser.Parse(header);
+ parser.Parse(&header);
RTC_DCHECK_LE(length, static_cast<size_t>(IP_PACKET_SIZE));
uint8_t temp_buffer[IP_PACKET_SIZE];
diff --git a/webrtc/test/rtp_file_reader.cc b/webrtc/test/rtp_file_reader.cc
index cb0e407..1413f00 100644
--- a/webrtc/test/rtp_file_reader.cc
+++ b/webrtc/test/rtp_file_reader.cc
@@ -458,7 +458,7 @@
rtp_parser.ParseRtcp(&marker.rtp_header);
packets_.push_back(marker);
} else {
- if (!rtp_parser.Parse(marker.rtp_header, NULL)) {
+ if (!rtp_parser.Parse(&marker.rtp_header, nullptr)) {
DEBUG_LOG("Not recognized as RTP/RTCP");
return kResultSkip;
}
diff --git a/webrtc/test/rtp_file_reader_unittest.cc b/webrtc/test/rtp_file_reader_unittest.cc
index 929813f..15a456c 100644
--- a/webrtc/test/rtp_file_reader_unittest.cc
+++ b/webrtc/test/rtp_file_reader_unittest.cc
@@ -85,7 +85,8 @@
while (rtp_packet_source_->NextPacket(&packet)) {
RtpUtility::RtpHeaderParser rtp_header_parser(packet.data, packet.length);
webrtc::RTPHeader header;
- if (!rtp_header_parser.RTCP() && rtp_header_parser.Parse(header, NULL)) {
+ if (!rtp_header_parser.RTCP() &&
+ rtp_header_parser.Parse(&header, nullptr)) {
pps[header.ssrc]++;
}
}
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index d09d2a2..5b23643 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -116,7 +116,7 @@
const PacketTime& packet_time) override {
RtpUtility::RtpHeaderParser parser(packet, length);
RTPHeader header;
- parser.Parse(header);
+ parser.Parse(&header);
{
rtc::CritScope lock(&crit_);
recv_times_[header.timestamp - rtp_timestamp_delta_] =
@@ -152,7 +152,7 @@
const PacketOptions& options) override {
RtpUtility::RtpHeaderParser parser(packet, length);
RTPHeader header;
- parser.Parse(header);
+ parser.Parse(&header);
int64_t current_time =
Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
