Migrate rtc event log away from legacy rtp header parser
Legacy parser is unsupported,
e.g. it doesn't support parsing two-byte header rtp header extension.
Bug: None
Change-Id: I83d683a7fcb9b147dbfe06bc8808fb5731e0e8f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226560
Reviewed-by: Björn Terelius <terelius@google.com>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34572}
diff --git a/logging/BUILD.gn b/logging/BUILD.gn
index 90a05f7..0700403 100644
--- a/logging/BUILD.gn
+++ b/logging/BUILD.gn
@@ -345,7 +345,6 @@
"../call:video_stream_api",
"../modules:module_api_public",
"../modules/audio_coding:audio_network_adaptor",
- "../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:ignore_wundef",
diff --git a/logging/rtc_event_log/rtc_event_log_parser.cc b/logging/rtc_event_log/rtc_event_log_parser.cc
index 91df18b..cb4be8f 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -35,7 +35,7 @@
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
-#include "modules/rtp_rtcp/source/rtp_utility.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
@@ -1368,7 +1368,6 @@
ParsedRtcEventLog::ParseStatus ParsedRtcEventLog::StoreParsedLegacyEvent(
const rtclog::Event& event) {
RTC_PARSE_CHECK_OR_RETURN(event.has_type());
- RTC_PARSE_CHECK_OR_RETURN(event.has_type());
switch (event.type()) {
case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT: {
auto config = GetVideoReceiveConfig(event);
@@ -1426,40 +1425,45 @@
break;
}
case rtclog::Event::RTP_EVENT: {
- PacketDirection direction;
- uint8_t header[IP_PACKET_SIZE];
- size_t header_length;
- size_t total_length;
- ParseStatus status = GetRtpHeader(event, &direction, header,
- &header_length, &total_length, nullptr);
- RTC_RETURN_IF_ERROR(status);
+ RTC_PARSE_CHECK_OR_RETURN(event.has_rtp_packet());
+ const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
+ RTC_PARSE_CHECK_OR_RETURN(rtp_packet.has_header());
+ RTC_PARSE_CHECK_OR_RETURN(rtp_packet.has_incoming());
+ RTC_PARSE_CHECK_OR_RETURN(rtp_packet.has_packet_length());
+ size_t total_length = rtp_packet.packet_length();
- uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(header + 8);
- const RtpHeaderExtensionMap* extension_map =
- GetRtpHeaderExtensionMap(direction, ssrc);
- RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
+ // Use RtpPacketReceived instead of more generic RtpPacket because former
+ // has a buildin convertion to RTPHeader.
+ RtpPacketReceived rtp_header;
+ RTC_PARSE_CHECK_OR_RETURN(rtp_header.Parse(rtp_packet.header()));
+
+ if (const RtpHeaderExtensionMap* extension_map = GetRtpHeaderExtensionMap(
+ rtp_packet.incoming(), rtp_header.Ssrc())) {
+ rtp_header.IdentifyExtensions(*extension_map);
+ }
+
RTPHeader parsed_header;
- rtp_parser.Parse(&parsed_header, extension_map, /*header_only*/ true);
+ rtp_header.GetHeader(&parsed_header);
// Since we give the parser only a header, there is no way for it to know
// the padding length. The best solution would be to log the padding
// length in RTC event log. In absence of it, we assume the RTP packet to
// contain only padding, if the padding bit is set.
// TODO(webrtc:9730): Use a generic way to obtain padding length.
- if ((header[0] & 0x20) != 0)
- parsed_header.paddingLength = total_length - header_length;
+ if (rtp_header.has_padding())
+ parsed_header.paddingLength = total_length - rtp_header.size();
RTC_PARSE_CHECK_OR_RETURN(event.has_timestamp_us());
int64_t timestamp_us = event.timestamp_us();
- if (direction == kIncomingPacket) {
+ if (rtp_packet.incoming()) {
incoming_rtp_packets_map_[parsed_header.ssrc].push_back(
LoggedRtpPacketIncoming(Timestamp::Micros(timestamp_us),
- parsed_header, header_length,
+ parsed_header, rtp_header.size(),
total_length));
} else {
outgoing_rtp_packets_map_[parsed_header.ssrc].push_back(
LoggedRtpPacketOutgoing(Timestamp::Micros(timestamp_us),
- parsed_header, header_length,
+ parsed_header, rtp_header.size(),
total_length));
}
break;
@@ -1574,59 +1578,11 @@
return ParseStatus::Success();
}
-// The header must have space for at least IP_PACKET_SIZE bytes.
-ParsedRtcEventLog::ParseStatus ParsedRtcEventLog::GetRtpHeader(
- const rtclog::Event& event,
- PacketDirection* incoming,
- uint8_t* header,
- size_t* header_length,
- size_t* total_length,
- int* probe_cluster_id) const {
- RTC_PARSE_CHECK_OR_RETURN(event.has_type());
- RTC_PARSE_CHECK_OR_RETURN_EQ(event.type(), rtclog::Event::RTP_EVENT);
- RTC_PARSE_CHECK_OR_RETURN(event.has_rtp_packet());
- const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
- // Get direction of packet.
- RTC_PARSE_CHECK_OR_RETURN(rtp_packet.has_incoming());
- if (incoming != nullptr) {
- *incoming = rtp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
- }
- // Get packet length.
- RTC_PARSE_CHECK_OR_RETURN(rtp_packet.has_packet_length());
- if (total_length != nullptr) {
- *total_length = rtp_packet.packet_length();
- }
- // Get header length.
- RTC_PARSE_CHECK_OR_RETURN(rtp_packet.has_header());
- if (header_length != nullptr) {
- *header_length = rtp_packet.header().size();
- }
- if (probe_cluster_id != nullptr) {
- if (rtp_packet.has_probe_cluster_id()) {
- *probe_cluster_id = rtp_packet.probe_cluster_id();
- RTC_PARSE_CHECK_OR_RETURN_NE(*probe_cluster_id,
- PacedPacketInfo::kNotAProbe);
- } else {
- *probe_cluster_id = PacedPacketInfo::kNotAProbe;
- }
- }
- // Get header contents.
- if (header != nullptr) {
- const size_t kMinRtpHeaderSize = 12;
- RTC_PARSE_CHECK_OR_RETURN_GE(rtp_packet.header().size(), kMinRtpHeaderSize);
- RTC_PARSE_CHECK_OR_RETURN_LE(rtp_packet.header().size(),
- static_cast<size_t>(IP_PACKET_SIZE));
- memcpy(header, rtp_packet.header().data(), rtp_packet.header().size());
- }
- return ParseStatus::Success();
-}
-
const RtpHeaderExtensionMap* ParsedRtcEventLog::GetRtpHeaderExtensionMap(
- PacketDirection direction,
+ bool incoming,
uint32_t ssrc) {
- auto& extensions_maps = direction == PacketDirection::kIncomingPacket
- ? incoming_rtp_extensions_maps_
- : outgoing_rtp_extensions_maps_;
+ auto& extensions_maps =
+ incoming ? incoming_rtp_extensions_maps_ : outgoing_rtp_extensions_maps_;
auto it = extensions_maps.find(ssrc);
if (it != extensions_maps.end()) {
return &(it->second);
diff --git a/logging/rtc_event_log/rtc_event_log_parser.h b/logging/rtc_event_log/rtc_event_log_parser.h
index 4898022..d18a238 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.h
+++ b/logging/rtc_event_log/rtc_event_log_parser.h
@@ -627,25 +627,12 @@
template <typename T>
void StoreFirstAndLastTimestamp(const std::vector<T>& v);
- // Reads the header, direction, header length and packet length from the RTP
- // event at |index|, and stores the values in the corresponding output
- // parameters. Each output parameter can be set to nullptr if that value
- // isn't needed.
- // NB: The header must have space for at least IP_PACKET_SIZE bytes.
- ParseStatus GetRtpHeader(const rtclog::Event& event,
- PacketDirection* incoming,
- uint8_t* header,
- size_t* header_length,
- size_t* total_length,
- int* probe_cluster_id) const;
-
// Returns: a pointer to a header extensions map acquired from parsing
// corresponding Audio/Video Sender/Receiver config events.
// Warning: if the same SSRC is reused by both video and audio streams during
// call, extensions maps may be incorrect (the last one would be returned).
- const RtpHeaderExtensionMap* GetRtpHeaderExtensionMap(
- PacketDirection direction,
- uint32_t ssrc);
+ const RtpHeaderExtensionMap* GetRtpHeaderExtensionMap(bool incoming,
+ uint32_t ssrc);
// Reads packet, direction and packet length from the RTCP event at |index|,
// and stores the values in the corresponding output parameters.
