Delete legacy rtp header parser as no longer used

Bug: None
Change-Id: I3c532eee7f2d9e5295874dd538730625c8d423ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227086
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34676}
This commit is contained in:
Danil Chapovalov 2021-07-28 14:09:48 +02:00 committed by WebRTC LUCI CQ
parent bbdecaff89
commit 5ce7d14f81
11 changed files with 0 additions and 923 deletions

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@ -1106,7 +1106,6 @@ rtc_library("neteq_test_tools") {
"../../rtc_base:rtc_base_approved", "../../rtc_base:rtc_base_approved",
"../../rtc_base/system:arch", "../../rtc_base/system:arch",
"../../test:rtp_test_utils", "../../test:rtp_test_utils",
"../rtp_rtcp",
"../rtp_rtcp:rtp_rtcp_format", "../rtp_rtcp:rtp_rtcp_format",
] ]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]

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@ -19,7 +19,6 @@
#include "absl/types/optional.h" #include "absl/types/optional.h"
#include "modules/audio_coding/neteq/tools/packet_source.h" #include "modules/audio_coding/neteq/tools/packet_source.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "rtc_base/constructor_magic.h" #include "rtc_base/constructor_magic.h"
namespace webrtc { namespace webrtc {

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@ -211,8 +211,6 @@ rtc_library("rtp_rtcp") {
"source/rtp_sender_video_frame_transformer_delegate.h", "source/rtp_sender_video_frame_transformer_delegate.h",
"source/rtp_sequence_number_map.cc", "source/rtp_sequence_number_map.cc",
"source/rtp_sequence_number_map.h", "source/rtp_sequence_number_map.h",
"source/rtp_utility.cc",
"source/rtp_utility.h",
"source/source_tracker.cc", "source/source_tracker.cc",
"source/source_tracker.h", "source/source_tracker.h",
"source/time_util.cc", "source/time_util.cc",
@ -553,7 +551,6 @@ if (rtc_include_tests) {
"source/rtp_sender_video_unittest.cc", "source/rtp_sender_video_unittest.cc",
"source/rtp_sequence_number_map_unittest.cc", "source/rtp_sequence_number_map_unittest.cc",
"source/rtp_util_unittest.cc", "source/rtp_util_unittest.cc",
"source/rtp_utility_unittest.cc",
"source/rtp_video_layers_allocation_extension_unittest.cc", "source/rtp_video_layers_allocation_extension_unittest.cc",
"source/source_tracker_unittest.cc", "source/source_tracker_unittest.cc",
"source/time_util_unittest.cc", "source/time_util_unittest.cc",

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@ -30,7 +30,6 @@
#include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h" #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "modules/rtp_rtcp/source/rtp_sender_video.h" #include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "modules/rtp_rtcp/source/video_fec_generator.h" #include "modules/rtp_rtcp/source/video_fec_generator.h"
#include "rtc_base/arraysize.h" #include "rtc_base/arraysize.h"
#include "rtc_base/logging.h" #include "rtc_base/logging.h"

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@ -1,520 +0,0 @@
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include <stddef.h>
#include <string>
#include "api/array_view.h"
#include "api/video/video_content_type.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/video_coding/codecs/interface/common_constants.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace RtpUtility {
enum {
kRtcpExpectedVersion = 2,
kRtcpMinHeaderLength = 4,
kRtcpMinParseLength = 8,
kRtpExpectedVersion = 2,
kRtpMinParseLength = 12
};
RtpHeaderParser::RtpHeaderParser(const uint8_t* rtpData,
const size_t rtpDataLength)
: _ptrRTPDataBegin(rtpData),
_ptrRTPDataEnd(rtpData ? (rtpData + rtpDataLength) : NULL) {}
RtpHeaderParser::~RtpHeaderParser() {}
bool RtpHeaderParser::RTCP() const {
// 72 to 76 is reserved for RTP
// 77 to 79 is not reserver but they are not assigned we will block them
// for RTCP 200 SR == marker bit + 72
// for RTCP 204 APP == marker bit + 76
/*
* RTCP
*
* FIR full INTRA-frame request 192 [RFC2032] supported
* NACK negative acknowledgement 193 [RFC2032]
* IJ Extended inter-arrival jitter report 195 [RFC-ietf-avt-rtp-toff
* set-07.txt] http://tools.ietf.org/html/draft-ietf-avt-rtp-toffset-07
* SR sender report 200 [RFC3551] supported
* RR receiver report 201 [RFC3551] supported
* SDES source description 202 [RFC3551] supported
* BYE goodbye 203 [RFC3551] supported
* APP application-defined 204 [RFC3551] ignored
* RTPFB Transport layer FB message 205 [RFC4585] supported
* PSFB Payload-specific FB message 206 [RFC4585] supported
* XR extended report 207 [RFC3611] supported
*/
/* 205 RFC 5104
* FMT 1 NACK supported
* FMT 2 reserved
* FMT 3 TMMBR supported
* FMT 4 TMMBN supported
*/
/* 206 RFC 5104
* FMT 1: Picture Loss Indication (PLI) supported
* FMT 2: Slice Lost Indication (SLI)
* FMT 3: Reference Picture Selection Indication (RPSI)
* FMT 4: Full Intra Request (FIR) Command supported
* FMT 5: Temporal-Spatial Trade-off Request (TSTR)
* FMT 6: Temporal-Spatial Trade-off Notification (TSTN)
* FMT 7: Video Back Channel Message (VBCM)
* FMT 15: Application layer FB message
*/
const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
if (length < kRtcpMinHeaderLength) {
return false;
}
const uint8_t V = _ptrRTPDataBegin[0] >> 6;
if (V != kRtcpExpectedVersion) {
return false;
}
const uint8_t payloadType = _ptrRTPDataBegin[1];
switch (payloadType) {
case 192:
return true;
case 193:
// not supported
// pass through and check for a potential RTP packet
return false;
case 195:
case 200:
case 201:
case 202:
case 203:
case 204:
case 205:
case 206:
case 207:
return true;
default:
return false;
}
}
bool RtpHeaderParser::Parse(RTPHeader* header,
const RtpHeaderExtensionMap* ptrExtensionMap,
bool header_only) const {
const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
if (length < kRtpMinParseLength) {
return false;
}
// Version
const uint8_t V = _ptrRTPDataBegin[0] >> 6;
// Padding
const bool P = ((_ptrRTPDataBegin[0] & 0x20) == 0) ? false : true;
// eXtension
const bool X = ((_ptrRTPDataBegin[0] & 0x10) == 0) ? false : true;
const uint8_t CC = _ptrRTPDataBegin[0] & 0x0f;
const bool M = ((_ptrRTPDataBegin[1] & 0x80) == 0) ? false : true;
const uint8_t PT = _ptrRTPDataBegin[1] & 0x7f;
const uint16_t sequenceNumber =
(_ptrRTPDataBegin[2] << 8) + _ptrRTPDataBegin[3];
const uint8_t* ptr = &_ptrRTPDataBegin[4];
uint32_t RTPTimestamp = ByteReader<uint32_t>::ReadBigEndian(ptr);
ptr += 4;
uint32_t SSRC = ByteReader<uint32_t>::ReadBigEndian(ptr);
ptr += 4;
if (V != kRtpExpectedVersion) {
return false;
}
const size_t CSRCocts = CC * 4;
if ((ptr + CSRCocts) > _ptrRTPDataEnd) {
return false;
}
header->markerBit = M;
header->payloadType = PT;
header->sequenceNumber = sequenceNumber;
header->timestamp = RTPTimestamp;
header->ssrc = SSRC;
header->numCSRCs = CC;
if (!P || header_only) {
header->paddingLength = 0;
}
for (uint8_t i = 0; i < CC; ++i) {
uint32_t CSRC = ByteReader<uint32_t>::ReadBigEndian(ptr);
ptr += 4;
header->arrOfCSRCs[i] = CSRC;
}
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;
// May not be present in packet.
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;
// May not be present in packet.
header->extension.hasVideoRotation = false;
header->extension.videoRotation = kVideoRotation_0;
// May not be present in packet.
header->extension.playout_delay.min_ms = -1;
header->extension.playout_delay.max_ms = -1;
// May not be present in packet.
header->extension.hasVideoContentType = false;
header->extension.videoContentType = VideoContentType::UNSPECIFIED;
header->extension.has_video_timing = false;
header->extension.video_timing = {0u, 0u, 0u, 0u, 0u, 0u, false};
if (X) {
/* RTP header extension, RFC 3550.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| defined by profile | length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| header extension |
| .... |
*/
const ptrdiff_t remain = _ptrRTPDataEnd - ptr;
if (remain < 4) {
return false;
}
header->headerLength += 4;
uint16_t definedByProfile = ByteReader<uint16_t>::ReadBigEndian(ptr);
ptr += 2;
// in 32 bit words
size_t XLen = ByteReader<uint16_t>::ReadBigEndian(ptr);
ptr += 2;
XLen *= 4; // in bytes
if (static_cast<size_t>(remain) < (4 + XLen)) {
return false;
}
static constexpr uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
if (definedByProfile == kRtpOneByteHeaderExtensionId) {
const uint8_t* ptrRTPDataExtensionEnd = ptr + XLen;
ParseOneByteExtensionHeader(header, ptrExtensionMap,
ptrRTPDataExtensionEnd, ptr);
}
header->headerLength += XLen;
}
if (header->headerLength > static_cast<size_t>(length))
return false;
if (P && !header_only) {
// Packet has padding.
if (header->headerLength != static_cast<size_t>(length)) {
// Packet is not header only. We can parse padding length now.
header->paddingLength = *(_ptrRTPDataEnd - 1);
} else {
RTC_LOG(LS_WARNING) << "Cannot parse padding length.";
// Packet is header only. We have no clue of the padding length.
return false;
}
}
if (header->headerLength + header->paddingLength >
static_cast<size_t>(length))
return false;
return true;
}
void RtpHeaderParser::ParseOneByteExtensionHeader(
RTPHeader* header,
const RtpHeaderExtensionMap* ptrExtensionMap,
const uint8_t* ptrRTPDataExtensionEnd,
const uint8_t* ptr) const {
if (!ptrExtensionMap) {
return;
}
while (ptrRTPDataExtensionEnd - ptr > 0) {
// 0
// 0 1 2 3 4 5 6 7
// +-+-+-+-+-+-+-+-+
// | ID | len |
// +-+-+-+-+-+-+-+-+
// Note that 'len' is the header extension element length, which is the
// number of bytes - 1.
const int id = (*ptr & 0xf0) >> 4;
const int len = (*ptr & 0x0f);
ptr++;
if (id == 0) {
// Padding byte, skip ignoring len.
continue;
}
if (id == 15) {
RTC_LOG(LS_VERBOSE)
<< "RTP extension header 15 encountered. Terminate parsing.";
return;
}
if (ptrRTPDataExtensionEnd - ptr < (len + 1)) {
RTC_LOG(LS_WARNING) << "Incorrect one-byte extension len: " << (len + 1)
<< ", bytes left in buffer: "
<< (ptrRTPDataExtensionEnd - ptr);
return;
}
RTPExtensionType type = ptrExtensionMap->GetType(id);
if (type == RtpHeaderExtensionMap::kInvalidType) {
// If we encounter an unknown extension, just skip over it.
RTC_LOG(LS_WARNING) << "Failed to find extension id: " << id;
} else {
switch (type) {
case kRtpExtensionTransmissionTimeOffset: {
if (len != 2) {
RTC_LOG(LS_WARNING)
<< "Incorrect transmission time offset len: " << len;
return;
}
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=2 | transmission offset |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
header->extension.transmissionTimeOffset =
ByteReader<int32_t, 3>::ReadBigEndian(ptr);
header->extension.hasTransmissionTimeOffset = true;
break;
}
case kRtpExtensionCsrcAudioLevel: {
RTC_LOG(LS_WARNING) << "Csrc audio level extension not supported";
return;
}
case kRtpExtensionAudioLevel: {
if (len != 0) {
RTC_LOG(LS_WARNING) << "Incorrect audio level len: " << len;
return;
}
// 0 1
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=0 |V| level |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//
header->extension.audioLevel = ptr[0] & 0x7f;
header->extension.voiceActivity = (ptr[0] & 0x80) != 0;
header->extension.hasAudioLevel = true;
break;
}
case kRtpExtensionAbsoluteSendTime: {
if (len != 2) {
RTC_LOG(LS_WARNING) << "Incorrect absolute send time len: " << len;
return;
}
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=2 | absolute send time |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
header->extension.absoluteSendTime =
ByteReader<uint32_t, 3>::ReadBigEndian(ptr);
header->extension.hasAbsoluteSendTime = true;
break;
}
case kRtpExtensionAbsoluteCaptureTime: {
AbsoluteCaptureTime extension;
if (!AbsoluteCaptureTimeExtension::Parse(
rtc::MakeArrayView(ptr, len + 1), &extension)) {
RTC_LOG(LS_WARNING)
<< "Incorrect absolute capture time len: " << len;
return;
}
header->extension.absolute_capture_time = extension;
break;
}
case kRtpExtensionVideoRotation: {
if (len != 0) {
RTC_LOG(LS_WARNING)
<< "Incorrect coordination of video coordination len: " << len;
return;
}
// 0 1
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=0 |0 0 0 0 C F R R|
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
header->extension.hasVideoRotation = true;
header->extension.videoRotation =
ConvertCVOByteToVideoRotation(ptr[0]);
break;
}
case kRtpExtensionTransportSequenceNumber: {
if (len != 1) {
RTC_LOG(LS_WARNING)
<< "Incorrect transport sequence number len: " << len;
return;
}
// 0 1 2
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | L=1 |transport wide sequence number |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
uint16_t sequence_number = ptr[0] << 8;
sequence_number += ptr[1];
header->extension.transportSequenceNumber = sequence_number;
header->extension.hasTransportSequenceNumber = true;
break;
}
case kRtpExtensionTransportSequenceNumber02:
RTC_LOG(WARNING) << "TransportSequenceNumberV2 unsupported by rtp "
"header parser.";
break;
case kRtpExtensionPlayoutDelay: {
if (len != 2) {
RTC_LOG(LS_WARNING) << "Incorrect playout delay len: " << len;
return;
}
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=2 | MIN delay | MAX delay |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
int min_playout_delay = (ptr[0] << 4) | ((ptr[1] >> 4) & 0xf);
int max_playout_delay = ((ptr[1] & 0xf) << 8) | ptr[2];
header->extension.playout_delay.min_ms =
min_playout_delay * PlayoutDelayLimits::kGranularityMs;
header->extension.playout_delay.max_ms =
max_playout_delay * PlayoutDelayLimits::kGranularityMs;
break;
}
case kRtpExtensionVideoContentType: {
if (len != 0) {
RTC_LOG(LS_WARNING) << "Incorrect video content type len: " << len;
return;
}
// 0 1
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=0 | Content type |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
if (videocontenttypehelpers::IsValidContentType(ptr[0])) {
header->extension.hasVideoContentType = true;
header->extension.videoContentType =
static_cast<VideoContentType>(ptr[0]);
}
break;
}
case kRtpExtensionVideoTiming: {
if (len != VideoTimingExtension::kValueSizeBytes - 1) {
RTC_LOG(LS_WARNING) << "Incorrect video timing len: " << len;
return;
}
header->extension.has_video_timing = true;
VideoTimingExtension::Parse(rtc::MakeArrayView(ptr, len + 1),
&header->extension.video_timing);
break;
}
case kRtpExtensionVideoLayersAllocation:
RTC_LOG(WARNING) << "VideoLayersAllocation extension unsupported by "
"rtp header parser.";
break;
case kRtpExtensionRtpStreamId: {
std::string name(reinterpret_cast<const char*>(ptr), len + 1);
if (IsLegalRsidName(name)) {
header->extension.stream_id = name;
} else {
RTC_LOG(LS_WARNING) << "Incorrect RtpStreamId";
}
break;
}
case kRtpExtensionRepairedRtpStreamId: {
std::string name(reinterpret_cast<const char*>(ptr), len + 1);
if (IsLegalRsidName(name)) {
header->extension.repaired_stream_id = name;
} else {
RTC_LOG(LS_WARNING) << "Incorrect RepairedRtpStreamId";
}
break;
}
case kRtpExtensionMid: {
std::string name(reinterpret_cast<const char*>(ptr), len + 1);
if (IsLegalMidName(name)) {
header->extension.mid = name;
} else {
RTC_LOG(LS_WARNING) << "Incorrect Mid";
}
break;
}
case kRtpExtensionGenericFrameDescriptor00:
case kRtpExtensionGenericFrameDescriptor02:
RTC_LOG(WARNING)
<< "RtpGenericFrameDescriptor unsupported by rtp header parser.";
break;
case kRtpExtensionColorSpace:
RTC_LOG(WARNING)
<< "RtpExtensionColorSpace unsupported by rtp header parser.";
break;
case kRtpExtensionInbandComfortNoise:
RTC_LOG(WARNING) << "Inband comfort noise extension unsupported by "
"rtp header parser.";
break;
case kRtpExtensionVideoFrameTrackingId:
RTC_LOG(WARNING)
<< "VideoFrameTrackingId unsupported by rtp header parser.";
break;
case kRtpExtensionNone:
case kRtpExtensionNumberOfExtensions: {
RTC_NOTREACHED() << "Invalid extension type: " << type;
return;
}
}
}
ptr += (len + 1);
}
}
} // namespace RtpUtility
} // namespace webrtc

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@ -1,50 +0,0 @@
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
#include <stdint.h>
#include <algorithm>
#include "absl/base/attributes.h"
#include "absl/strings/string_view.h"
#include "api/rtp_headers.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
namespace RtpUtility {
class RtpHeaderParser {
public:
RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength);
~RtpHeaderParser();
ABSL_DEPRECATED("Use IsRtpPacket or IsRtcpPacket")
bool RTCP() const;
bool Parse(RTPHeader* parsedPacket,
const RtpHeaderExtensionMap* ptrExtensionMap = nullptr,
bool header_only = false) const;
private:
void ParseOneByteExtensionHeader(RTPHeader* parsedPacket,
const RtpHeaderExtensionMap* ptrExtensionMap,
const uint8_t* ptrRTPDataExtensionEnd,
const uint8_t* ptr) const;
const uint8_t* const _ptrRTPDataBegin;
const uint8_t* const _ptrRTPDataEnd;
};
} // namespace RtpUtility
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_

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@ -1,285 +0,0 @@
/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "test/gmock.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
using ::testing::ElementsAreArray;
using ::testing::make_tuple;
const int8_t kPayloadType = 100;
const uint32_t kSsrc = 0x12345678;
const uint16_t kSeqNum = 88;
const uint32_t kTimestamp = 0x65431278;
} // namespace
TEST(RtpHeaderParser, ParseMinimum) {
// clang-format off
const uint8_t kPacket[] = {
0x80, kPayloadType, 0x00, kSeqNum,
0x65, 0x43, 0x12, 0x78, // kTimestamp.
0x12, 0x34, 0x56, 0x78}; // kSsrc.
// clang-format on
RtpUtility::RtpHeaderParser parser(kPacket, sizeof(kPacket));
RTPHeader header;
EXPECT_TRUE(parser.Parse(&header, nullptr));
EXPECT_EQ(kPayloadType, header.payloadType);
EXPECT_EQ(kSeqNum, header.sequenceNumber);
EXPECT_EQ(kTimestamp, header.timestamp);
EXPECT_EQ(kSsrc, header.ssrc);
EXPECT_EQ(0u, header.paddingLength);
EXPECT_EQ(sizeof(kPacket), header.headerLength);
}
TEST(RtpHeaderParser, ParseWithExtension) {
// clang-format off
const uint8_t kPacket[] = {
0x90, kPayloadType, 0x00, kSeqNum,
0x65, 0x43, 0x12, 0x78, // kTimestamp.
0x12, 0x34, 0x56, 0x78, // kSsrc.
0xbe, 0xde, 0x00, 0x01, // Extension block of size 1 x 32bit words.
0x12, 0x01, 0x56, 0xce};
// clang-format on
RtpHeaderExtensionMap extensions;
extensions.Register<TransmissionOffset>(1);
RtpUtility::RtpHeaderParser parser(kPacket, sizeof(kPacket));
RTPHeader header;
EXPECT_TRUE(parser.Parse(&header, &extensions));
EXPECT_EQ(kPayloadType, header.payloadType);
EXPECT_EQ(kSeqNum, header.sequenceNumber);
EXPECT_EQ(kTimestamp, header.timestamp);
EXPECT_EQ(kSsrc, header.ssrc);
ASSERT_TRUE(header.extension.hasTransmissionTimeOffset);
EXPECT_EQ(0x156ce, header.extension.transmissionTimeOffset);
}
TEST(RtpHeaderParser, ParseWithInvalidSizedExtension) {
const size_t kPayloadSize = 7;
// clang-format off
const uint8_t kPacket[] = {
0x90, kPayloadType, 0x00, kSeqNum,
0x65, 0x43, 0x12, 0x78, // kTimestamp.
0x12, 0x34, 0x56, 0x78, // kSsrc.
0xbe, 0xde, 0x00, 0x02, // Extension block of size 2 x 32bit words.
0x16, // (6+1)-bytes, but Transmission Offset expected to be 3-bytes.
'e', 'x', 't',
'd', 'a', 't', 'a',
'p', 'a', 'y', 'l', 'o', 'a', 'd'
};
// clang-format on
RtpHeaderExtensionMap extensions;
extensions.Register<TransmissionOffset>(1);
RtpUtility::RtpHeaderParser parser(kPacket, sizeof(kPacket));
RTPHeader header;
EXPECT_TRUE(parser.Parse(&header, &extensions));
// Extension should be ignored.
EXPECT_FALSE(header.extension.hasTransmissionTimeOffset);
// But shouldn't prevent reading payload.
EXPECT_THAT(sizeof(kPacket) - kPayloadSize, header.headerLength);
}
TEST(RtpHeaderParser, ParseWithExtensionPadding) {
// clang-format off
const uint8_t kPacket[] = {
0x90, kPayloadType, 0x00, kSeqNum,
0x65, 0x43, 0x12, 0x78, // kTimestamp.
0x12, 0x34, 0x56, 0x78, // kSsrc.
0xbe, 0xde, 0x00, 0x02, // Extension of size 1x32bit word.
0x02, // A byte of (invalid) padding.
0x12, 0x1a, 0xda, 0x03, // TransmissionOffset extension.
0x0f, 0x00, 0x03, // More invalid padding bytes: id=0, but len > 0.
};
// clang-format on
RtpHeaderExtensionMap extensions;
extensions.Register<TransmissionOffset>(1);
RtpUtility::RtpHeaderParser parser(kPacket, sizeof(kPacket));
RTPHeader header;
EXPECT_TRUE(parser.Parse(&header, &extensions));
// Parse should skip padding and read extension.
EXPECT_TRUE(header.extension.hasTransmissionTimeOffset);
EXPECT_EQ(0x1ada03, header.extension.transmissionTimeOffset);
EXPECT_EQ(sizeof(kPacket), header.headerLength);
}
TEST(RtpHeaderParser, ParseWithOverSizedExtension) {
// clang-format off
const uint8_t kPacket[] = {
0x90, kPayloadType, 0x00, kSeqNum,
0x65, 0x43, 0x12, 0x78, // kTimestamp.
0x12, 0x34, 0x56, 0x78, // kSsrc.
0xbe, 0xde, 0x00, 0x01, // Extension of size 1x32bit word.
0x00, // Add a byte of padding.
0x12, // Extension id 1 size (2+1).
0xda, 0x1a // Only 2 bytes of extension payload.
};
// clang-format on
RtpHeaderExtensionMap extensions;
extensions.Register<TransmissionOffset>(1);
RtpUtility::RtpHeaderParser parser(kPacket, sizeof(kPacket));
RTPHeader header;
EXPECT_TRUE(parser.Parse(&header, &extensions));
// Parse should ignore extension.
EXPECT_FALSE(header.extension.hasTransmissionTimeOffset);
EXPECT_EQ(sizeof(kPacket), header.headerLength);
}
TEST(RtpHeaderParser, ParseAll9Extensions) {
const uint8_t kAudioLevel = 0x5a;
// clang-format off
const uint8_t kPacket[] = {
0x90, kPayloadType, 0x00, kSeqNum,
0x65, 0x43, 0x12, 0x78, // kTimestamp.
0x12, 0x34, 0x56, 0x78, // kSsrc.
0xbe, 0xde, 0x00, 0x0c, // Extension of size 12x32bit words.
0x40, 0x80|kAudioLevel, // AudioLevel.
0x22, 0x01, 0x56, 0xce, // TransmissionOffset.
0x62, 0x12, 0x34, 0x56, // AbsoluteSendTime.
0x7f, 0x12, 0x34, 0x56, 0x78, // AbsoluteCaptureTime.
0x90, 0xab, 0xcd, 0xef, // AbsoluteCaptureTime. (cont.)
0x11, 0x22, 0x33, 0x44, // AbsoluteCaptureTime. (cont.)
0x55, 0x66, 0x77, 0x88, // AbsoluteCaptureTime. (cont.)
0x81, 0xce, 0xab, // TransportSequenceNumber.
0xa0, 0x03, // VideoRotation.
0xb2, 0x12, 0x48, 0x76, // PlayoutDelayLimits.
0xc2, 'r', 't', 'x', // RtpStreamId
0xd5, 's', 't', 'r', 'e', 'a', 'm', // RepairedRtpStreamId
0x00, // Padding to 32bit boundary.
};
// clang-format on
ASSERT_EQ(sizeof(kPacket) % 4, 0u);
RtpHeaderExtensionMap extensions;
extensions.Register<TransmissionOffset>(2);
extensions.Register<AudioLevel>(4);
extensions.Register<AbsoluteSendTime>(6);
extensions.Register<AbsoluteCaptureTimeExtension>(7);
extensions.Register<TransportSequenceNumber>(8);
extensions.Register<VideoOrientation>(0xa);
extensions.Register<PlayoutDelayLimits>(0xb);
extensions.Register<RtpStreamId>(0xc);
extensions.Register<RepairedRtpStreamId>(0xd);
RtpUtility::RtpHeaderParser parser(kPacket, sizeof(kPacket));
RTPHeader header;
EXPECT_TRUE(parser.Parse(&header, &extensions));
EXPECT_TRUE(header.extension.hasTransmissionTimeOffset);
EXPECT_EQ(0x156ce, header.extension.transmissionTimeOffset);
EXPECT_TRUE(header.extension.hasAudioLevel);
EXPECT_TRUE(header.extension.voiceActivity);
EXPECT_EQ(kAudioLevel, header.extension.audioLevel);
EXPECT_TRUE(header.extension.hasAbsoluteSendTime);
EXPECT_EQ(0x123456U, header.extension.absoluteSendTime);
ASSERT_TRUE(header.extension.absolute_capture_time.has_value());
EXPECT_EQ(0x1234567890abcdefULL,
header.extension.absolute_capture_time->absolute_capture_timestamp);
ASSERT_TRUE(header.extension.absolute_capture_time
->estimated_capture_clock_offset.has_value());
EXPECT_EQ(0x1122334455667788LL, header.extension.absolute_capture_time
->estimated_capture_clock_offset.value());
EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
EXPECT_EQ(0xceab, header.extension.transportSequenceNumber);
EXPECT_TRUE(header.extension.hasVideoRotation);
EXPECT_EQ(kVideoRotation_270, header.extension.videoRotation);
EXPECT_EQ(0x124 * PlayoutDelayLimits::kGranularityMs,
header.extension.playout_delay.min_ms);
EXPECT_EQ(0x876 * PlayoutDelayLimits::kGranularityMs,
header.extension.playout_delay.max_ms);
EXPECT_EQ(header.extension.stream_id, "rtx");
EXPECT_EQ(header.extension.repaired_stream_id, "stream");
}
TEST(RtpHeaderParser, ParseMalformedRsidExtensions) {
// clang-format off
const uint8_t kPacket[] = {
0x90, kPayloadType, 0x00, kSeqNum,
0x65, 0x43, 0x12, 0x78, // kTimestamp.
0x12, 0x34, 0x56, 0x78, // kSsrc.
0xbe, 0xde, 0x00, 0x03, // Extension of size 3x32bit words.
0xc2, '\0', 't', 'x', // empty RtpStreamId
0xd5, 's', 't', 'r', '\0', 'a', 'm', // RepairedRtpStreamId
0x00, // Padding to 32bit boundary.
};
// clang-format on
ASSERT_EQ(sizeof(kPacket) % 4, 0u);
RtpHeaderExtensionMap extensions;
extensions.Register<RtpStreamId>(0xc);
extensions.Register<RepairedRtpStreamId>(0xd);
RtpUtility::RtpHeaderParser parser(kPacket, sizeof(kPacket));
RTPHeader header;
EXPECT_TRUE(parser.Parse(&header, &extensions));
EXPECT_TRUE(header.extension.stream_id.empty());
EXPECT_TRUE(header.extension.repaired_stream_id.empty());
}
TEST(RtpHeaderParser, ParseWithCsrcsExtensionAndPadding) {
const uint8_t kPacketPaddingSize = 8;
const uint32_t kCsrcs[] = {0x34567890, 0x32435465};
const size_t kPayloadSize = 7;
// clang-format off
const uint8_t kPacket[] = {
0xb2, kPayloadType, 0x00, kSeqNum,
0x65, 0x43, 0x12, 0x78, // kTimestamp.
0x12, 0x34, 0x56, 0x78, // kSsrc.
0x34, 0x56, 0x78, 0x90, // kCsrcs[0].
0x32, 0x43, 0x54, 0x65, // kCsrcs[1].
0xbe, 0xde, 0x00, 0x01, // Extension.
0x12, 0x00, 0x56, 0xce, // TransmissionTimeOffset with id = 1.
'p', 'a', 'y', 'l', 'o', 'a', 'd',
'p', 'a', 'd', 'd', 'i', 'n', 'g', kPacketPaddingSize};
// clang-format on
RtpHeaderExtensionMap extensions;
extensions.Register<TransmissionOffset>(1);
RtpUtility::RtpHeaderParser parser(kPacket, sizeof(kPacket));
RTPHeader header;
EXPECT_TRUE(parser.Parse(&header, &extensions));
EXPECT_EQ(kPayloadType, header.payloadType);
EXPECT_EQ(kSeqNum, header.sequenceNumber);
EXPECT_EQ(kTimestamp, header.timestamp);
EXPECT_EQ(kSsrc, header.ssrc);
EXPECT_THAT(make_tuple(header.arrOfCSRCs, header.numCSRCs),
ElementsAreArray(kCsrcs));
EXPECT_EQ(kPacketPaddingSize, header.paddingLength);
EXPECT_THAT(sizeof(kPacket) - kPayloadSize - kPacketPaddingSize,
header.headerLength);
EXPECT_TRUE(header.extension.hasTransmissionTimeOffset);
EXPECT_EQ(0x56ce, header.extension.transmissionTimeOffset);
}
} // namespace webrtc

View File

@ -20,7 +20,6 @@
#include "modules/rtp_rtcp/source/byte_io.h" #include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/forward_error_correction.h" #include "modules/rtp_rtcp/source/forward_error_correction.h"
#include "modules/rtp_rtcp/source/forward_error_correction_internal.h" #include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "rtc_base/checks.h" #include "rtc_base/checks.h"
#include "rtc_base/synchronization/mutex.h" #include "rtc_base/synchronization/mutex.h"

View File

@ -49,7 +49,6 @@
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h" #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "rtc_base/checks.h" #include "rtc_base/checks.h"
#include "rtc_base/format_macros.h" #include "rtc_base/format_macros.h"
#include "rtc_base/logging.h" #include "rtc_base/logging.h"

View File

@ -231,14 +231,6 @@ webrtc_fuzzer_test("rtp_packetizer_av1_fuzzer") {
] ]
} }
webrtc_fuzzer_test("rtp_header_fuzzer") {
sources = [ "rtp_header_fuzzer.cc" ]
deps = [
"../../modules/rtp_rtcp",
"../../modules/rtp_rtcp:rtp_rtcp_format",
]
}
webrtc_fuzzer_test("congestion_controller_feedback_fuzzer") { webrtc_fuzzer_test("congestion_controller_feedback_fuzzer") {
sources = [ "congestion_controller_feedback_fuzzer.cc" ] sources = [ "congestion_controller_feedback_fuzzer.cc" ]
deps = [ deps = [

View File

@ -1,52 +0,0 @@
/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <bitset>
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
namespace webrtc {
// We decide which header extensions to register by reading four bytes
// from the beginning of `data` and interpreting it as a bitmask over
// the RTPExtensionType enum. This assert ensures four bytes are enough.
static_assert(kRtpExtensionNumberOfExtensions <= 32,
"Insufficient bits read to configure all header extensions. Add "
"an extra byte and update the switches.");
void FuzzOneInput(const uint8_t* data, size_t size) {
if (size <= 4)
return;
// Don't use the configuration byte as part of the packet.
std::bitset<32> extensionMask(*reinterpret_cast<const uint32_t*>(data));
data += 4;
size -= 4;
RtpPacketReceived::ExtensionManager extensions(/*extmap_allow_mixed=*/true);
// Start at local_id = 1 since 0 is an invalid extension id.
int local_id = 1;
// Skip i = 0 since it maps to kRtpExtensionNone.
for (int i = 1; i < kRtpExtensionNumberOfExtensions; i++) {
RTPExtensionType extension_type = static_cast<RTPExtensionType>(i);
if (extensionMask[i]) {
// Extensions are registered with an ID, which you signal to the
// peer so they know what to expect. This code only cares about
// parsing so the value of the ID isn't relevant.
extensions.RegisterByType(local_id++, extension_type);
}
}
RTPHeader rtp_header;
RtpUtility::RtpHeaderParser rtp_parser(data, size);
rtp_parser.Parse(&rtp_header, &extensions);
}
} // namespace webrtc