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webrtc / src / 9ce800d6d1635ba3a471ab105ad92cd03c0b5a02 / . / logging / rtc_event_log / rtc_event_log_parser.cc
blob: 1a0c8eb01454918b314c38eb581156dcccf633df [file] [log] [blame]
/*
* 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 "logging/rtc_event_log/rtc_event_log_parser.h"
#include <stdint.h>
#include <string.h>
#include <algorithm>
#include <fstream>
#include <istream> // no-presubmit-check TODO(webrtc:8982)
#include <limits>
#include <map>
#include <utility>
#include "absl/memory/memory.h"
#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "logging/rtc_event_log/encoder/blob_encoding.h"
#include "logging/rtc_event_log/encoder/delta_encoding.h"
#include "logging/rtc_event_log/encoder/rtc_event_log_encoder_common.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_event_processor.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
#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 "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/sequence_number_util.h"
#include "rtc_base/protobuf_utils.h"
using webrtc_event_logging::ToSigned;
using webrtc_event_logging::ToUnsigned;
namespace webrtc {
namespace {
constexpr size_t kIpv4Overhead = 20;
constexpr size_t kIpv6Overhead = 40;
constexpr size_t kUdpOverhead = 8;
constexpr size_t kSrtpOverhead = 10;
constexpr size_t kStunOverhead = 4;
constexpr uint16_t kDefaultOverhead =
kUdpOverhead + kSrtpOverhead + kIpv4Overhead;
// Starting at a multiple of common audio sample rate (48000) and video tick
// rate (90000) to make a tick count of 0 to correspond to something without
// decimals in base 10. Starting at 0 is not safe as it would cause negative
// wraparound if the first timestamps are out of order.
constexpr uint64_t kStartingCaptureTimeTicks = 90 * 48 * 1000;
struct MediaStreamInfo {
MediaStreamInfo() : unwrap_capture_ticks(kStartingCaptureTimeTicks) {}
MediaStreamInfo(LoggedMediaType media_type, bool rtx)
: media_type(media_type),
rtx(rtx),
unwrap_capture_ticks(kStartingCaptureTimeTicks) {}
LoggedMediaType media_type = LoggedMediaType::kUnknown;
bool rtx = false;
SeqNumUnwrapper<uint32_t> unwrap_capture_ticks;
};
template <typename Iterable>
void AddRecvStreamInfos(std::map<uint32_t, MediaStreamInfo>* streams,
const Iterable configs,
LoggedMediaType media_type) {
for (auto& conf : configs) {
streams->insert({conf.config.remote_ssrc, {media_type, false}});
if (conf.config.rtx_ssrc != 0)
streams->insert({conf.config.rtx_ssrc, {media_type, true}});
}
}
template <typename Iterable>
void AddSendStreamInfos(std::map<uint32_t, MediaStreamInfo>* streams,
const Iterable configs,
LoggedMediaType media_type) {
for (auto& conf : configs) {
streams->insert({conf.config.local_ssrc, {media_type, false}});
if (conf.config.rtx_ssrc != 0)
streams->insert({conf.config.rtx_ssrc, {media_type, true}});
}
}
struct OverheadChangeEvent {
Timestamp timestamp;
uint16_t overhead;
};
std::vector<OverheadChangeEvent> GetOverheadChangingEvents(
const std::vector<LoggedRouteChangeEvent>& route_changes,
PacketDirection direction) {
std::vector<OverheadChangeEvent> overheads;
for (auto& event : route_changes) {
uint16_t new_overhead = direction == PacketDirection::kIncomingPacket
? event.return_overhead
: event.send_overhead;
if (overheads.empty() || new_overhead != overheads.back().overhead) {
overheads.push_back({event.log_time, new_overhead});
}
}
return overheads;
}
bool IdenticalRtcpContents(const std::vector<uint8_t>& last_rtcp,
absl::string_view new_rtcp) {
if (last_rtcp.size() != new_rtcp.size())
return false;
return memcmp(last_rtcp.data(), new_rtcp.data(), new_rtcp.size()) == 0;
}
// Conversion functions for legacy wire format.
RtcpMode GetRuntimeRtcpMode(rtclog::VideoReceiveConfig::RtcpMode rtcp_mode) {
switch (rtcp_mode) {
case rtclog::VideoReceiveConfig::RTCP_COMPOUND:
return RtcpMode::kCompound;
case rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE:
return RtcpMode::kReducedSize;
}
RTC_NOTREACHED();
return RtcpMode::kOff;
}
BandwidthUsage GetRuntimeDetectorState(
rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
switch (detector_state) {
case rtclog::DelayBasedBweUpdate::BWE_NORMAL:
return BandwidthUsage::kBwNormal;
case rtclog::DelayBasedBweUpdate::BWE_UNDERUSING:
return BandwidthUsage::kBwUnderusing;
case rtclog::DelayBasedBweUpdate::BWE_OVERUSING:
return BandwidthUsage::kBwOverusing;
}
RTC_NOTREACHED();
return BandwidthUsage::kBwNormal;
}
IceCandidatePairConfigType GetRuntimeIceCandidatePairConfigType(
rtclog::IceCandidatePairConfig::IceCandidatePairConfigType type) {
switch (type) {
case rtclog::IceCandidatePairConfig::ADDED:
return IceCandidatePairConfigType::kAdded;
case rtclog::IceCandidatePairConfig::UPDATED:
return IceCandidatePairConfigType::kUpdated;
case rtclog::IceCandidatePairConfig::DESTROYED:
return IceCandidatePairConfigType::kDestroyed;
case rtclog::IceCandidatePairConfig::SELECTED:
return IceCandidatePairConfigType::kSelected;
}
RTC_NOTREACHED();
return IceCandidatePairConfigType::kAdded;
}
IceCandidateType GetRuntimeIceCandidateType(
rtclog::IceCandidatePairConfig::IceCandidateType type) {
switch (type) {
case rtclog::IceCandidatePairConfig::LOCAL:
return IceCandidateType::kLocal;
case rtclog::IceCandidatePairConfig::STUN:
return IceCandidateType::kStun;
case rtclog::IceCandidatePairConfig::PRFLX:
return IceCandidateType::kPrflx;
case rtclog::IceCandidatePairConfig::RELAY:
return IceCandidateType::kRelay;
case rtclog::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE:
return IceCandidateType::kUnknown;
}
RTC_NOTREACHED();
return IceCandidateType::kUnknown;
}
IceCandidatePairProtocol GetRuntimeIceCandidatePairProtocol(
rtclog::IceCandidatePairConfig::Protocol protocol) {
switch (protocol) {
case rtclog::IceCandidatePairConfig::UDP:
return IceCandidatePairProtocol::kUdp;
case rtclog::IceCandidatePairConfig::TCP:
return IceCandidatePairProtocol::kTcp;
case rtclog::IceCandidatePairConfig::SSLTCP:
return IceCandidatePairProtocol::kSsltcp;
case rtclog::IceCandidatePairConfig::TLS:
return IceCandidatePairProtocol::kTls;
case rtclog::IceCandidatePairConfig::UNKNOWN_PROTOCOL:
return IceCandidatePairProtocol::kUnknown;
}
RTC_NOTREACHED();
return IceCandidatePairProtocol::kUnknown;
}
IceCandidatePairAddressFamily GetRuntimeIceCandidatePairAddressFamily(
rtclog::IceCandidatePairConfig::AddressFamily address_family) {
switch (address_family) {
case rtclog::IceCandidatePairConfig::IPV4:
return IceCandidatePairAddressFamily::kIpv4;
case rtclog::IceCandidatePairConfig::IPV6:
return IceCandidatePairAddressFamily::kIpv6;
case rtclog::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY:
return IceCandidatePairAddressFamily::kUnknown;
}
RTC_NOTREACHED();
return IceCandidatePairAddressFamily::kUnknown;
}
IceCandidateNetworkType GetRuntimeIceCandidateNetworkType(
rtclog::IceCandidatePairConfig::NetworkType network_type) {
switch (network_type) {
case rtclog::IceCandidatePairConfig::ETHERNET:
return IceCandidateNetworkType::kEthernet;
case rtclog::IceCandidatePairConfig::LOOPBACK:
return IceCandidateNetworkType::kLoopback;
case rtclog::IceCandidatePairConfig::WIFI:
return IceCandidateNetworkType::kWifi;
case rtclog::IceCandidatePairConfig::VPN:
return IceCandidateNetworkType::kVpn;
case rtclog::IceCandidatePairConfig::CELLULAR:
return IceCandidateNetworkType::kCellular;
case rtclog::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE:
return IceCandidateNetworkType::kUnknown;
}
RTC_NOTREACHED();
return IceCandidateNetworkType::kUnknown;
}
IceCandidatePairEventType GetRuntimeIceCandidatePairEventType(
rtclog::IceCandidatePairEvent::IceCandidatePairEventType type) {
switch (type) {
case rtclog::IceCandidatePairEvent::CHECK_SENT:
return IceCandidatePairEventType::kCheckSent;
case rtclog::IceCandidatePairEvent::CHECK_RECEIVED:
return IceCandidatePairEventType::kCheckReceived;
case rtclog::IceCandidatePairEvent::CHECK_RESPONSE_SENT:
return IceCandidatePairEventType::kCheckResponseSent;
case rtclog::IceCandidatePairEvent::CHECK_RESPONSE_RECEIVED:
return IceCandidatePairEventType::kCheckResponseReceived;
}
RTC_NOTREACHED();
return IceCandidatePairEventType::kCheckSent;
}
// Conversion functions for version 2 of the wire format.
BandwidthUsage GetRuntimeDetectorState(
rtclog2::DelayBasedBweUpdates::DetectorState detector_state) {
switch (detector_state) {
case rtclog2::DelayBasedBweUpdates::BWE_NORMAL:
return BandwidthUsage::kBwNormal;
case rtclog2::DelayBasedBweUpdates::BWE_UNDERUSING:
return BandwidthUsage::kBwUnderusing;
case rtclog2::DelayBasedBweUpdates::BWE_OVERUSING:
return BandwidthUsage::kBwOverusing;
case rtclog2::DelayBasedBweUpdates::BWE_UNKNOWN_STATE:
break;
}
RTC_NOTREACHED();
return BandwidthUsage::kBwNormal;
}
ProbeFailureReason GetRuntimeProbeFailureReason(
rtclog2::BweProbeResultFailure::FailureReason failure) {
switch (failure) {
case rtclog2::BweProbeResultFailure::INVALID_SEND_RECEIVE_INTERVAL:
return ProbeFailureReason::kInvalidSendReceiveInterval;
case rtclog2::BweProbeResultFailure::INVALID_SEND_RECEIVE_RATIO:
return ProbeFailureReason::kInvalidSendReceiveRatio;
case rtclog2::BweProbeResultFailure::TIMEOUT:
return ProbeFailureReason::kTimeout;
case rtclog2::BweProbeResultFailure::UNKNOWN:
break;
}
RTC_NOTREACHED();
return ProbeFailureReason::kTimeout;
}
DtlsTransportState GetRuntimeDtlsTransportState(
rtclog2::DtlsTransportStateEvent::DtlsTransportState state) {
switch (state) {
case rtclog2::DtlsTransportStateEvent::DTLS_TRANSPORT_NEW:
return DtlsTransportState::kNew;
case rtclog2::DtlsTransportStateEvent::DTLS_TRANSPORT_CONNECTING:
return DtlsTransportState::kConnecting;
case rtclog2::DtlsTransportStateEvent::DTLS_TRANSPORT_CONNECTED:
return DtlsTransportState::kConnected;
case rtclog2::DtlsTransportStateEvent::DTLS_TRANSPORT_CLOSED:
return DtlsTransportState::kClosed;
case rtclog2::DtlsTransportStateEvent::DTLS_TRANSPORT_FAILED:
return DtlsTransportState::kFailed;
case rtclog2::DtlsTransportStateEvent::UNKNOWN_DTLS_TRANSPORT_STATE:
RTC_NOTREACHED();
return DtlsTransportState::kNumValues;
}
RTC_NOTREACHED();
return DtlsTransportState::kNumValues;
}
IceCandidatePairConfigType GetRuntimeIceCandidatePairConfigType(
rtclog2::IceCandidatePairConfig::IceCandidatePairConfigType type) {
switch (type) {
case rtclog2::IceCandidatePairConfig::ADDED:
return IceCandidatePairConfigType::kAdded;
case rtclog2::IceCandidatePairConfig::UPDATED:
return IceCandidatePairConfigType::kUpdated;
case rtclog2::IceCandidatePairConfig::DESTROYED:
return IceCandidatePairConfigType::kDestroyed;
case rtclog2::IceCandidatePairConfig::SELECTED:
return IceCandidatePairConfigType::kSelected;
case rtclog2::IceCandidatePairConfig::UNKNOWN_CONFIG_TYPE:
break;
}
RTC_NOTREACHED();
return IceCandidatePairConfigType::kAdded;
}
IceCandidateType GetRuntimeIceCandidateType(
rtclog2::IceCandidatePairConfig::IceCandidateType type) {
switch (type) {
case rtclog2::IceCandidatePairConfig::LOCAL:
return IceCandidateType::kLocal;
case rtclog2::IceCandidatePairConfig::STUN:
return IceCandidateType::kStun;
case rtclog2::IceCandidatePairConfig::PRFLX:
return IceCandidateType::kPrflx;
case rtclog2::IceCandidatePairConfig::RELAY:
return IceCandidateType::kRelay;
case rtclog2::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE:
return IceCandidateType::kUnknown;
}
RTC_NOTREACHED();
return IceCandidateType::kUnknown;
}
IceCandidatePairProtocol GetRuntimeIceCandidatePairProtocol(
rtclog2::IceCandidatePairConfig::Protocol protocol) {
switch (protocol) {
case rtclog2::IceCandidatePairConfig::UDP:
return IceCandidatePairProtocol::kUdp;
case rtclog2::IceCandidatePairConfig::TCP:
return IceCandidatePairProtocol::kTcp;
case rtclog2::IceCandidatePairConfig::SSLTCP:
return IceCandidatePairProtocol::kSsltcp;
case rtclog2::IceCandidatePairConfig::TLS:
return IceCandidatePairProtocol::kTls;
case rtclog2::IceCandidatePairConfig::UNKNOWN_PROTOCOL:
return IceCandidatePairProtocol::kUnknown;
}
RTC_NOTREACHED();
return IceCandidatePairProtocol::kUnknown;
}
IceCandidatePairAddressFamily GetRuntimeIceCandidatePairAddressFamily(
rtclog2::IceCandidatePairConfig::AddressFamily address_family) {
switch (address_family) {
case rtclog2::IceCandidatePairConfig::IPV4:
return IceCandidatePairAddressFamily::kIpv4;
case rtclog2::IceCandidatePairConfig::IPV6:
return IceCandidatePairAddressFamily::kIpv6;
case rtclog2::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY:
return IceCandidatePairAddressFamily::kUnknown;
}
RTC_NOTREACHED();
return IceCandidatePairAddressFamily::kUnknown;
}
IceCandidateNetworkType GetRuntimeIceCandidateNetworkType(
rtclog2::IceCandidatePairConfig::NetworkType network_type) {
switch (network_type) {
case rtclog2::IceCandidatePairConfig::ETHERNET:
return IceCandidateNetworkType::kEthernet;
case rtclog2::IceCandidatePairConfig::LOOPBACK:
return IceCandidateNetworkType::kLoopback;
case rtclog2::IceCandidatePairConfig::WIFI:
return IceCandidateNetworkType::kWifi;
case rtclog2::IceCandidatePairConfig::VPN:
return IceCandidateNetworkType::kVpn;
case rtclog2::IceCandidatePairConfig::CELLULAR:
return IceCandidateNetworkType::kCellular;
case rtclog2::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE:
return IceCandidateNetworkType::kUnknown;
}
RTC_NOTREACHED();
return IceCandidateNetworkType::kUnknown;
}
IceCandidatePairEventType GetRuntimeIceCandidatePairEventType(
rtclog2::IceCandidatePairEvent::IceCandidatePairEventType type) {
switch (type) {
case rtclog2::IceCandidatePairEvent::CHECK_SENT:
return IceCandidatePairEventType::kCheckSent;
case rtclog2::IceCandidatePairEvent::CHECK_RECEIVED:
return IceCandidatePairEventType::kCheckReceived;
case rtclog2::IceCandidatePairEvent::CHECK_RESPONSE_SENT:
return IceCandidatePairEventType::kCheckResponseSent;
case rtclog2::IceCandidatePairEvent::CHECK_RESPONSE_RECEIVED:
return IceCandidatePairEventType::kCheckResponseReceived;
case rtclog2::IceCandidatePairEvent::UNKNOWN_CHECK_TYPE:
break;
}
RTC_NOTREACHED();
return IceCandidatePairEventType::kCheckSent;
}
std::vector<RtpExtension> GetRuntimeRtpHeaderExtensionConfig(
const rtclog2::RtpHeaderExtensionConfig& proto_header_extensions) {
std::vector<RtpExtension> rtp_extensions;
if (proto_header_extensions.has_transmission_time_offset_id()) {
rtp_extensions.emplace_back(
RtpExtension::kTimestampOffsetUri,
proto_header_extensions.transmission_time_offset_id());
}
if (proto_header_extensions.has_absolute_send_time_id()) {
rtp_extensions.emplace_back(
RtpExtension::kAbsSendTimeUri,
proto_header_extensions.absolute_send_time_id());
}
if (proto_header_extensions.has_transport_sequence_number_id()) {
rtp_extensions.emplace_back(
RtpExtension::kTransportSequenceNumberUri,
proto_header_extensions.transport_sequence_number_id());
}
if (proto_header_extensions.has_audio_level_id()) {
rtp_extensions.emplace_back(RtpExtension::kAudioLevelUri,
proto_header_extensions.audio_level_id());
}
if (proto_header_extensions.has_video_rotation_id()) {
rtp_extensions.emplace_back(RtpExtension::kVideoRotationUri,
proto_header_extensions.video_rotation_id());
}
return rtp_extensions;
}
// End of conversion functions.
// Reads a VarInt from |stream| and returns it. Also writes the read bytes to
// |buffer| starting |bytes_written| bytes into the buffer. |bytes_written| is
// incremented for each written byte.
absl::optional<uint64_t> ParseVarInt(
std::istream& stream, // no-presubmit-check TODO(webrtc:8982)
char* buffer,
size_t* bytes_written) {
uint64_t varint = 0;
for (size_t bytes_read = 0; bytes_read < 10; ++bytes_read) {
// The most significant bit of each byte is 0 if it is the last byte in
// the varint and 1 otherwise. Thus, we take the 7 least significant bits
// of each byte and shift them 7 bits for each byte read previously to get
// the (unsigned) integer.
int byte = stream.get();
if (stream.eof()) {
return absl::nullopt;
}
RTC_DCHECK_GE(byte, 0);
RTC_DCHECK_LE(byte, 255);
varint |= static_cast<uint64_t>(byte & 0x7F) << (7 * bytes_read);
buffer[*bytes_written] = byte;
*bytes_written += 1;
if ((byte & 0x80) == 0) {
return varint;
}
}
return absl::nullopt;
}
void GetHeaderExtensions(std::vector<RtpExtension>* header_extensions,
const RepeatedPtrField<rtclog::RtpHeaderExtension>&
proto_header_extensions) {
header_extensions->clear();
for (auto& p : proto_header_extensions) {
RTC_CHECK(p.has_name());
RTC_CHECK(p.has_id());
const std::string& name = p.name();
int id = p.id();
header_extensions->push_back(RtpExtension(name, id));
}
}
template <typename ProtoType, typename LoggedType>
void StoreRtpPackets(
const ProtoType& proto,
std::map<uint32_t, std::vector<LoggedType>>* rtp_packets_map) {
RTC_CHECK(proto.has_timestamp_ms());
RTC_CHECK(proto.has_marker());
RTC_CHECK(proto.has_payload_type());
RTC_CHECK(proto.has_sequence_number());
RTC_CHECK(proto.has_rtp_timestamp());
RTC_CHECK(proto.has_ssrc());
RTC_CHECK(proto.has_payload_size());
RTC_CHECK(proto.has_header_size());
RTC_CHECK(proto.has_padding_size());
// Base event
{
RTPHeader header;
header.markerBit = rtc::checked_cast<bool>(proto.marker());
header.payloadType = rtc::checked_cast<uint8_t>(proto.payload_type());
header.sequenceNumber =
rtc::checked_cast<uint16_t>(proto.sequence_number());
header.timestamp = rtc::checked_cast<uint32_t>(proto.rtp_timestamp());
header.ssrc = rtc::checked_cast<uint32_t>(proto.ssrc());
header.numCSRCs = 0; // TODO(terelius): Implement CSRC.
header.paddingLength = rtc::checked_cast<size_t>(proto.padding_size());
header.headerLength = rtc::checked_cast<size_t>(proto.header_size());
// TODO(terelius): Should we implement payload_type_frequency?
if (proto.has_transport_sequence_number()) {
header.extension.hasTransportSequenceNumber = true;
header.extension.transportSequenceNumber =
rtc::checked_cast<uint16_t>(proto.transport_sequence_number());
}
if (proto.has_transmission_time_offset()) {
header.extension.hasTransmissionTimeOffset = true;
header.extension.transmissionTimeOffset =
rtc::checked_cast<int32_t>(proto.transmission_time_offset());
}
if (proto.has_absolute_send_time()) {
header.extension.hasAbsoluteSendTime = true;
header.extension.absoluteSendTime =
rtc::checked_cast<uint32_t>(proto.absolute_send_time());
}
if (proto.has_video_rotation()) {
header.extension.hasVideoRotation = true;
header.extension.videoRotation = ConvertCVOByteToVideoRotation(
rtc::checked_cast<uint8_t>(proto.video_rotation()));
}
if (proto.has_audio_level()) {
RTC_CHECK(proto.has_voice_activity());
header.extension.hasAudioLevel = true;
header.extension.voiceActivity =
rtc::checked_cast<bool>(proto.voice_activity());
const uint8_t audio_level =
rtc::checked_cast<uint8_t>(proto.audio_level());
RTC_CHECK_LE(audio_level, 0x7Fu);
header.extension.audioLevel = audio_level;
} else {
RTC_CHECK(!proto.has_voice_activity());
}
(*rtp_packets_map)[header.ssrc].emplace_back(
proto.timestamp_ms() * 1000, header, proto.header_size(),
proto.payload_size() + header.headerLength + header.paddingLength);
}
const size_t number_of_deltas =
proto.has_number_of_deltas() ? proto.number_of_deltas() : 0u;
if (number_of_deltas == 0) {
return;
}
// timestamp_ms (event)
std::vector<absl::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_CHECK_EQ(timestamp_ms_values.size(), number_of_deltas);
// marker (RTP base)
std::vector<absl::optional<uint64_t>> marker_values =
DecodeDeltas(proto.marker_deltas(), proto.marker(), number_of_deltas);
RTC_CHECK_EQ(marker_values.size(), number_of_deltas);
// payload_type (RTP base)
std::vector<absl::optional<uint64_t>> payload_type_values = DecodeDeltas(
proto.payload_type_deltas(), proto.payload_type(), number_of_deltas);
RTC_CHECK_EQ(payload_type_values.size(), number_of_deltas);
// sequence_number (RTP base)
std::vector<absl::optional<uint64_t>> sequence_number_values =
DecodeDeltas(proto.sequence_number_deltas(), proto.sequence_number(),
number_of_deltas);
RTC_CHECK_EQ(sequence_number_values.size(), number_of_deltas);
// rtp_timestamp (RTP base)
std::vector<absl::optional<uint64_t>> rtp_timestamp_values = DecodeDeltas(
proto.rtp_timestamp_deltas(), proto.rtp_timestamp(), number_of_deltas);
RTC_CHECK_EQ(rtp_timestamp_values.size(), number_of_deltas);
// ssrc (RTP base)
std::vector<absl::optional<uint64_t>> ssrc_values =
DecodeDeltas(proto.ssrc_deltas(), proto.ssrc(), number_of_deltas);
RTC_CHECK_EQ(ssrc_values.size(), number_of_deltas);
// payload_size (RTP base)
std::vector<absl::optional<uint64_t>> payload_size_values = DecodeDeltas(
proto.payload_size_deltas(), proto.payload_size(), number_of_deltas);
RTC_CHECK_EQ(payload_size_values.size(), number_of_deltas);
// header_size (RTP base)
std::vector<absl::optional<uint64_t>> header_size_values = DecodeDeltas(
proto.header_size_deltas(), proto.header_size(), number_of_deltas);
RTC_CHECK_EQ(header_size_values.size(), number_of_deltas);
// padding_size (RTP base)
std::vector<absl::optional<uint64_t>> padding_size_values = DecodeDeltas(
proto.padding_size_deltas(), proto.padding_size(), number_of_deltas);
RTC_CHECK_EQ(padding_size_values.size(), number_of_deltas);
// transport_sequence_number (RTP extension)
std::vector<absl::optional<uint64_t>> transport_sequence_number_values;
{
const absl::optional<uint64_t> base_transport_sequence_number =
proto.has_transport_sequence_number()
? proto.transport_sequence_number()
: absl::optional<uint64_t>();
transport_sequence_number_values =
DecodeDeltas(proto.transport_sequence_number_deltas(),
base_transport_sequence_number, number_of_deltas);
RTC_CHECK_EQ(transport_sequence_number_values.size(), number_of_deltas);
}
// transmission_time_offset (RTP extension)
std::vector<absl::optional<uint64_t>> transmission_time_offset_values;
{
const absl::optional<uint64_t> unsigned_base_transmission_time_offset =
proto.has_transmission_time_offset()
? ToUnsigned(proto.transmission_time_offset())
: absl::optional<uint64_t>();
transmission_time_offset_values =
DecodeDeltas(proto.transmission_time_offset_deltas(),
unsigned_base_transmission_time_offset, number_of_deltas);
RTC_CHECK_EQ(transmission_time_offset_values.size(), number_of_deltas);
}
// absolute_send_time (RTP extension)
std::vector<absl::optional<uint64_t>> absolute_send_time_values;
{
const absl::optional<uint64_t> base_absolute_send_time =
proto.has_absolute_send_time() ? proto.absolute_send_time()
: absl::optional<uint64_t>();
absolute_send_time_values =
DecodeDeltas(proto.absolute_send_time_deltas(), base_absolute_send_time,
number_of_deltas);
RTC_CHECK_EQ(absolute_send_time_values.size(), number_of_deltas);
}
// video_rotation (RTP extension)
std::vector<absl::optional<uint64_t>> video_rotation_values;
{
const absl::optional<uint64_t> base_video_rotation =
proto.has_video_rotation() ? proto.video_rotation()
: absl::optional<uint64_t>();
video_rotation_values = DecodeDeltas(proto.video_rotation_deltas(),
base_video_rotation, number_of_deltas);
RTC_CHECK_EQ(video_rotation_values.size(), number_of_deltas);
}
// audio_level (RTP extension)
std::vector<absl::optional<uint64_t>> audio_level_values;
{
const absl::optional<uint64_t> base_audio_level =
proto.has_audio_level() ? proto.audio_level()
: absl::optional<uint64_t>();
audio_level_values = DecodeDeltas(proto.audio_level_deltas(),
base_audio_level, number_of_deltas);
RTC_CHECK_EQ(audio_level_values.size(), number_of_deltas);
}
// voice_activity (RTP extension)
std::vector<absl::optional<uint64_t>> voice_activity_values;
{
const absl::optional<uint64_t> base_voice_activity =
proto.has_voice_activity() ? proto.voice_activity()
: absl::optional<uint64_t>();
voice_activity_values = DecodeDeltas(proto.voice_activity_deltas(),
base_voice_activity, number_of_deltas);
RTC_CHECK_EQ(voice_activity_values.size(), number_of_deltas);
}
// Delta decoding
for (size_t i = 0; i < number_of_deltas; ++i) {
RTC_CHECK(timestamp_ms_values[i].has_value());
RTC_CHECK(marker_values[i].has_value());
RTC_CHECK(payload_type_values[i].has_value());
RTC_CHECK(sequence_number_values[i].has_value());
RTC_CHECK(rtp_timestamp_values[i].has_value());
RTC_CHECK(ssrc_values[i].has_value());
RTC_CHECK(payload_size_values[i].has_value());
RTC_CHECK(header_size_values[i].has_value());
RTC_CHECK(padding_size_values[i].has_value());
int64_t timestamp_ms;
RTC_CHECK(ToSigned(timestamp_ms_values[i].value(), &timestamp_ms));
RTPHeader header;
header.markerBit = rtc::checked_cast<bool>(*marker_values[i]);
header.payloadType = rtc::checked_cast<uint8_t>(*payload_type_values[i]);
header.sequenceNumber =
rtc::checked_cast<uint16_t>(*sequence_number_values[i]);
header.timestamp = rtc::checked_cast<uint32_t>(*rtp_timestamp_values[i]);
header.ssrc = rtc::checked_cast<uint32_t>(*ssrc_values[i]);
header.numCSRCs = 0; // TODO(terelius): Implement CSRC.
header.paddingLength = rtc::checked_cast<size_t>(*padding_size_values[i]);
header.headerLength = rtc::checked_cast<size_t>(*header_size_values[i]);
// TODO(terelius): Should we implement payload_type_frequency?
if (transport_sequence_number_values.size() > i &&
transport_sequence_number_values[i].has_value()) {
header.extension.hasTransportSequenceNumber = true;
header.extension.transportSequenceNumber = rtc::checked_cast<uint16_t>(
transport_sequence_number_values[i].value());
}
if (transmission_time_offset_values.size() > i &&
transmission_time_offset_values[i].has_value()) {
header.extension.hasTransmissionTimeOffset = true;
int32_t transmission_time_offset;
RTC_CHECK(ToSigned(transmission_time_offset_values[i].value(),
&transmission_time_offset));
header.extension.transmissionTimeOffset = transmission_time_offset;
}
if (absolute_send_time_values.size() > i &&
absolute_send_time_values[i].has_value()) {
header.extension.hasAbsoluteSendTime = true;
header.extension.absoluteSendTime =
rtc::checked_cast<uint32_t>(absolute_send_time_values[i].value());
}
if (video_rotation_values.size() > i &&
video_rotation_values[i].has_value()) {
header.extension.hasVideoRotation = true;
header.extension.videoRotation = ConvertCVOByteToVideoRotation(
rtc::checked_cast<uint8_t>(video_rotation_values[i].value()));
}
if (audio_level_values.size() > i && audio_level_values[i].has_value()) {
RTC_CHECK(voice_activity_values.size() > i &&
voice_activity_values[i].has_value());
header.extension.hasAudioLevel = true;
header.extension.voiceActivity =
rtc::checked_cast<bool>(voice_activity_values[i].value());
const uint8_t audio_level =
rtc::checked_cast<uint8_t>(audio_level_values[i].value());
RTC_CHECK_LE(audio_level, 0x7Fu);
header.extension.audioLevel = audio_level;
} else {
RTC_CHECK(voice_activity_values.size() <= i ||
!voice_activity_values[i].has_value());
}
(*rtp_packets_map)[header.ssrc].emplace_back(
1000 * timestamp_ms, header, header.headerLength,
payload_size_values[i].value() + header.headerLength +
header.paddingLength);
}
}
template <typename ProtoType, typename LoggedType>
void StoreRtcpPackets(const ProtoType& proto,
std::vector<LoggedType>* rtcp_packets,
bool remove_duplicates) {
RTC_CHECK(proto.has_timestamp_ms());
RTC_CHECK(proto.has_raw_packet());
// TODO(terelius): Incoming RTCP may be delivered once for audio and once
// for video. As a work around, we remove the duplicated packets since they
// cause problems when analyzing the log or feeding it into the transport
// feedback adapter.
if (!remove_duplicates || rtcp_packets->empty() ||
!IdenticalRtcpContents(rtcp_packets->back().rtcp.raw_data,
proto.raw_packet())) {
// Base event
rtcp_packets->emplace_back(proto.timestamp_ms() * 1000, proto.raw_packet());
}
const size_t number_of_deltas =
proto.has_number_of_deltas() ? proto.number_of_deltas() : 0u;
if (number_of_deltas == 0) {
return;
}
// timestamp_ms
std::vector<absl::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_CHECK_EQ(timestamp_ms_values.size(), number_of_deltas);
// raw_packet
RTC_CHECK(proto.has_raw_packet_blobs());
std::vector<absl::string_view> raw_packet_values =
DecodeBlobs(proto.raw_packet_blobs(), number_of_deltas);
RTC_CHECK_EQ(raw_packet_values.size(), number_of_deltas);
// Delta decoding
for (size_t i = 0; i < number_of_deltas; ++i) {
RTC_CHECK(timestamp_ms_values[i].has_value());
int64_t timestamp_ms;
RTC_CHECK(ToSigned(timestamp_ms_values[i].value(), &timestamp_ms));
// TODO(terelius): Incoming RTCP may be delivered once for audio and once
// for video. As a work around, we remove the duplicated packets since they
// cause problems when analyzing the log or feeding it into the transport
// feedback adapter.
if (remove_duplicates && !rtcp_packets->empty() &&
IdenticalRtcpContents(rtcp_packets->back().rtcp.raw_data,
raw_packet_values[i])) {
continue;
}
const size_t data_size = raw_packet_values[i].size();
const uint8_t* data =
reinterpret_cast<const uint8_t*>(raw_packet_values[i].data());
rtcp_packets->emplace_back(1000 * timestamp_ms, data, data_size);
}
}
void StoreRtcpBlocks(
int64_t timestamp_us,
const uint8_t* packet_begin,
const uint8_t* packet_end,
std::vector<LoggedRtcpPacketTransportFeedback>* transport_feedback_list,
std::vector<LoggedRtcpPacketSenderReport>* sr_list,
std::vector<LoggedRtcpPacketReceiverReport>* rr_list,
std::vector<LoggedRtcpPacketRemb>* remb_list,
std::vector<LoggedRtcpPacketNack>* nack_list,
std::vector<LoggedRtcpPacketLossNotification>* loss_notification_list) {
rtcp::CommonHeader header;
for (const uint8_t* block = packet_begin; block < packet_end;
block = header.NextPacket()) {
RTC_CHECK(header.Parse(block, packet_end - block));
if (header.type() == rtcp::TransportFeedback::kPacketType &&
header.fmt() == rtcp::TransportFeedback::kFeedbackMessageType) {
LoggedRtcpPacketTransportFeedback parsed_block;
parsed_block.timestamp_us = timestamp_us;
if (parsed_block.transport_feedback.Parse(header))
transport_feedback_list->push_back(std::move(parsed_block));
} else if (header.type() == rtcp::SenderReport::kPacketType) {
LoggedRtcpPacketSenderReport parsed_block;
parsed_block.timestamp_us = timestamp_us;
if (parsed_block.sr.Parse(header)) {
sr_list->push_back(std::move(parsed_block));
}
} else if (header.type() == rtcp::ReceiverReport::kPacketType) {
LoggedRtcpPacketReceiverReport parsed_block;
parsed_block.timestamp_us = timestamp_us;
if (parsed_block.rr.Parse(header)) {
rr_list->push_back(std::move(parsed_block));
}
} else if (header.type() == rtcp::Remb::kPacketType &&
header.fmt() == rtcp::Remb::kFeedbackMessageType) {
bool type_found = false;
if (!type_found) {
LoggedRtcpPacketRemb parsed_block;
parsed_block.timestamp_us = timestamp_us;
if (parsed_block.remb.Parse(header)) {
remb_list->push_back(std::move(parsed_block));
type_found = true;
}
}
if (!type_found) {
LoggedRtcpPacketLossNotification parsed_block;
parsed_block.timestamp_us = timestamp_us;
if (parsed_block.loss_notification.Parse(header)) {
loss_notification_list->push_back(std::move(parsed_block));
type_found = true;
}
}
} else if (header.type() == rtcp::Nack::kPacketType &&
header.fmt() == rtcp::Nack::kFeedbackMessageType) {
LoggedRtcpPacketNack parsed_block;
parsed_block.timestamp_us = timestamp_us;
if (parsed_block.nack.Parse(header)) {
nack_list->push_back(std::move(parsed_block));
}
}
}
}
} // namespace
LoggedRtcpPacket::LoggedRtcpPacket(uint64_t timestamp_us,
const uint8_t* packet,
size_t total_length)
: timestamp_us(timestamp_us), raw_data(packet, packet + total_length) {}
LoggedRtcpPacket::LoggedRtcpPacket(uint64_t timestamp_us,
const std::string& packet)
: timestamp_us(timestamp_us), raw_data(packet.size()) {
memcpy(raw_data.data(), packet.data(), packet.size());
}
LoggedRtcpPacket::LoggedRtcpPacket(const LoggedRtcpPacket& rhs) = default;
LoggedRtcpPacket::~LoggedRtcpPacket() = default;
ParsedRtcEventLog::~ParsedRtcEventLog() = default;
ParsedRtcEventLog::LoggedRtpStreamIncoming::LoggedRtpStreamIncoming() = default;
ParsedRtcEventLog::LoggedRtpStreamIncoming::LoggedRtpStreamIncoming(
const LoggedRtpStreamIncoming& rhs) = default;
ParsedRtcEventLog::LoggedRtpStreamIncoming::~LoggedRtpStreamIncoming() =
default;
ParsedRtcEventLog::LoggedRtpStreamOutgoing::LoggedRtpStreamOutgoing() = default;
ParsedRtcEventLog::LoggedRtpStreamOutgoing::LoggedRtpStreamOutgoing(
const LoggedRtpStreamOutgoing& rhs) = default;
ParsedRtcEventLog::LoggedRtpStreamOutgoing::~LoggedRtpStreamOutgoing() =
default;
ParsedRtcEventLog::LoggedRtpStreamView::LoggedRtpStreamView(
uint32_t ssrc,
const LoggedRtpPacketIncoming* ptr,
size_t num_elements)
: ssrc(ssrc),
packet_view(PacketView<const LoggedRtpPacket>::Create(
ptr,
num_elements,
offsetof(LoggedRtpPacketIncoming, rtp))) {}
ParsedRtcEventLog::LoggedRtpStreamView::LoggedRtpStreamView(
uint32_t ssrc,
const LoggedRtpPacketOutgoing* ptr,
size_t num_elements)
: ssrc(ssrc),
packet_view(PacketView<const LoggedRtpPacket>::Create(
ptr,
num_elements,
offsetof(LoggedRtpPacketOutgoing, rtp))) {}
ParsedRtcEventLog::LoggedRtpStreamView::LoggedRtpStreamView(
const LoggedRtpStreamView&) = default;
// Return default values for header extensions, to use on streams without stored
// mapping data. Currently this only applies to audio streams, since the mapping
// is not stored in the event log.
// TODO(ivoc): Remove this once this mapping is stored in the event log for
// audio streams. Tracking bug: webrtc:6399
webrtc::RtpHeaderExtensionMap
ParsedRtcEventLog::GetDefaultHeaderExtensionMap() {
webrtc::RtpHeaderExtensionMap default_map;
default_map.Register<AudioLevel>(webrtc::RtpExtension::kAudioLevelDefaultId);
default_map.Register<TransmissionOffset>(
webrtc::RtpExtension::kTimestampOffsetDefaultId);
default_map.Register<AbsoluteSendTime>(
webrtc::RtpExtension::kAbsSendTimeDefaultId);
default_map.Register<VideoOrientation>(
webrtc::RtpExtension::kVideoRotationDefaultId);
default_map.Register<VideoContentTypeExtension>(
webrtc::RtpExtension::kVideoContentTypeDefaultId);
default_map.Register<VideoTimingExtension>(
webrtc::RtpExtension::kVideoTimingDefaultId);
default_map.Register<TransportSequenceNumber>(
webrtc::RtpExtension::kTransportSequenceNumberDefaultId);
default_map.Register<PlayoutDelayLimits>(
webrtc::RtpExtension::kPlayoutDelayDefaultId);
return default_map;
}
ParsedRtcEventLog::ParsedRtcEventLog(
UnconfiguredHeaderExtensions parse_unconfigured_header_extensions)
: parse_unconfigured_header_extensions_(
parse_unconfigured_header_extensions) {
Clear();
}
void ParsedRtcEventLog::Clear() {
default_extension_map_ = GetDefaultHeaderExtensionMap();
incoming_rtx_ssrcs_.clear();
incoming_video_ssrcs_.clear();
incoming_audio_ssrcs_.clear();
outgoing_rtx_ssrcs_.clear();
outgoing_video_ssrcs_.clear();
outgoing_audio_ssrcs_.clear();
incoming_rtp_packets_map_.clear();
outgoing_rtp_packets_map_.clear();
incoming_rtp_packets_by_ssrc_.clear();
outgoing_rtp_packets_by_ssrc_.clear();
incoming_rtp_packet_views_by_ssrc_.clear();
outgoing_rtp_packet_views_by_ssrc_.clear();
incoming_rtcp_packets_.clear();
outgoing_rtcp_packets_.clear();
incoming_rr_.clear();
outgoing_rr_.clear();
incoming_sr_.clear();
outgoing_sr_.clear();
incoming_nack_.clear();
outgoing_nack_.clear();
incoming_remb_.clear();
outgoing_remb_.clear();
incoming_transport_feedback_.clear();
outgoing_transport_feedback_.clear();
incoming_loss_notification_.clear();
outgoing_loss_notification_.clear();
start_log_events_.clear();
stop_log_events_.clear();
audio_playout_events_.clear();
audio_network_adaptation_events_.clear();
bwe_probe_cluster_created_events_.clear();
bwe_probe_failure_events_.clear();
bwe_probe_success_events_.clear();
bwe_delay_updates_.clear();
bwe_loss_updates_.clear();
dtls_transport_states_.clear();
dtls_writable_states_.clear();
alr_state_events_.clear();
ice_candidate_pair_configs_.clear();
ice_candidate_pair_events_.clear();
audio_recv_configs_.clear();
audio_send_configs_.clear();
video_recv_configs_.clear();
video_send_configs_.clear();
memset(last_incoming_rtcp_packet_, 0, IP_PACKET_SIZE);
last_incoming_rtcp_packet_length_ = 0;
first_timestamp_ = std::numeric_limits<int64_t>::max();
last_timestamp_ = std::numeric_limits<int64_t>::min();
incoming_rtp_extensions_maps_.clear();
outgoing_rtp_extensions_maps_.clear();
}
bool ParsedRtcEventLog::ParseFile(const std::string& filename) {
std::ifstream file( // no-presubmit-check TODO(webrtc:8982)
filename, std::ios_base::in | std::ios_base::binary);
if (!file.good() || !file.is_open()) {
RTC_LOG(LS_WARNING) << "Could not open file for reading.";
return false;
}
return ParseStream(file);
}
bool ParsedRtcEventLog::ParseString(const std::string& s) {
std::istringstream stream( // no-presubmit-check TODO(webrtc:8982)
s, std::ios_base::in | std::ios_base::binary);
return ParseStream(stream);
}
bool ParsedRtcEventLog::ParseStream(
std::istream& stream) { // no-presubmit-check TODO(webrtc:8982)
Clear();
bool success = ParseStreamInternal(stream);
// Cache the configured SSRCs.
for (const auto& video_recv_config : video_recv_configs()) {
incoming_video_ssrcs_.insert(video_recv_config.config.remote_ssrc);
incoming_video_ssrcs_.insert(video_recv_config.config.rtx_ssrc);
incoming_rtx_ssrcs_.insert(video_recv_config.config.rtx_ssrc);
}
for (const auto& video_send_config : video_send_configs()) {
outgoing_video_ssrcs_.insert(video_send_config.config.local_ssrc);
outgoing_video_ssrcs_.insert(video_send_config.config.rtx_ssrc);
outgoing_rtx_ssrcs_.insert(video_send_config.config.rtx_ssrc);
}
for (const auto& audio_recv_config : audio_recv_configs()) {
incoming_audio_ssrcs_.insert(audio_recv_config.config.remote_ssrc);
}
for (const auto& audio_send_config : audio_send_configs()) {
outgoing_audio_ssrcs_.insert(audio_send_config.config.local_ssrc);
}
// ParseStreamInternal stores the RTP packets in a map indexed by SSRC.
// Since we dont need rapid lookup based on SSRC after parsing, we move the
// packets_streams from map to vector.
incoming_rtp_packets_by_ssrc_.reserve(incoming_rtp_packets_map_.size());
for (auto& kv : incoming_rtp_packets_map_) {
incoming_rtp_packets_by_ssrc_.emplace_back(LoggedRtpStreamIncoming());
incoming_rtp_packets_by_ssrc_.back().ssrc = kv.first;
incoming_rtp_packets_by_ssrc_.back().incoming_packets =
std::move(kv.second);
}
incoming_rtp_packets_map_.clear();
outgoing_rtp_packets_by_ssrc_.reserve(outgoing_rtp_packets_map_.size());
for (auto& kv : outgoing_rtp_packets_map_) {
outgoing_rtp_packets_by_ssrc_.emplace_back(LoggedRtpStreamOutgoing());
outgoing_rtp_packets_by_ssrc_.back().ssrc = kv.first;
outgoing_rtp_packets_by_ssrc_.back().outgoing_packets =
std::move(kv.second);
}
outgoing_rtp_packets_map_.clear();
// Build PacketViews for easier iteration over RTP packets.
for (const auto& stream : incoming_rtp_packets_by_ssrc_) {
incoming_rtp_packet_views_by_ssrc_.emplace_back(
LoggedRtpStreamView(stream.ssrc, stream.incoming_packets.data(),
stream.incoming_packets.size()));
}
for (const auto& stream : outgoing_rtp_packets_by_ssrc_) {
outgoing_rtp_packet_views_by_ssrc_.emplace_back(
LoggedRtpStreamView(stream.ssrc, stream.outgoing_packets.data(),
stream.outgoing_packets.size()));
}
// Set up convenience wrappers around the most commonly used RTCP types.
for (const auto& incoming : incoming_rtcp_packets_) {
const int64_t timestamp_us = incoming.rtcp.timestamp_us;
const uint8_t* packet_begin = incoming.rtcp.raw_data.data();
const uint8_t* packet_end = packet_begin + incoming.rtcp.raw_data.size();
StoreRtcpBlocks(timestamp_us, packet_begin, packet_end,
&incoming_transport_feedback_, &incoming_sr_, &incoming_rr_,
&incoming_remb_, &incoming_nack_,
&incoming_loss_notification_);
}
for (const auto& outgoing : outgoing_rtcp_packets_) {
const int64_t timestamp_us = outgoing.rtcp.timestamp_us;
const uint8_t* packet_begin = outgoing.rtcp.raw_data.data();
const uint8_t* packet_end = packet_begin + outgoing.rtcp.raw_data.size();
StoreRtcpBlocks(timestamp_us, packet_begin, packet_end,
&outgoing_transport_feedback_, &outgoing_sr_, &outgoing_rr_,
&outgoing_remb_, &outgoing_nack_,
&outgoing_loss_notification_);
}
// Store first and last timestamp events that might happen before the call is
// connected or after the call is disconnected. Typical examples are
// stream configurations and starting/stopping the log.
// TODO(terelius): Figure out if we actually need to find the first and last
// timestamp in the parser. It seems like this could be done by the caller.
first_timestamp_ = std::numeric_limits<int64_t>::max();
last_timestamp_ = std::numeric_limits<int64_t>::min();
StoreFirstAndLastTimestamp(alr_state_events());
for (const auto& audio_stream : audio_playout_events()) {
// Audio playout events are grouped by SSRC.
StoreFirstAndLastTimestamp(audio_stream.second);
}
StoreFirstAndLastTimestamp(audio_network_adaptation_events());
StoreFirstAndLastTimestamp(bwe_probe_cluster_created_events());
StoreFirstAndLastTimestamp(bwe_probe_failure_events());
StoreFirstAndLastTimestamp(bwe_probe_success_events());
StoreFirstAndLastTimestamp(bwe_delay_updates());
StoreFirstAndLastTimestamp(bwe_loss_updates());
StoreFirstAndLastTimestamp(dtls_transport_states());
StoreFirstAndLastTimestamp(dtls_writable_states());
StoreFirstAndLastTimestamp(ice_candidate_pair_configs());
StoreFirstAndLastTimestamp(ice_candidate_pair_events());
for (const auto& rtp_stream : incoming_rtp_packets_by_ssrc()) {
StoreFirstAndLastTimestamp(rtp_stream.incoming_packets);
}
for (const auto& rtp_stream : outgoing_rtp_packets_by_ssrc()) {
StoreFirstAndLastTimestamp(rtp_stream.outgoing_packets);
}
StoreFirstAndLastTimestamp(incoming_rtcp_packets());
StoreFirstAndLastTimestamp(outgoing_rtcp_packets());
return success;
}
bool ParsedRtcEventLog::ParseStreamInternal(
std::istream& stream) { // no-presubmit-check TODO(webrtc:8982)
constexpr uint64_t kMaxEventSize = 10000000; // Sanity check.
std::vector<char> buffer(0xFFFF);
RTC_DCHECK(stream.good());
while (1) {
// Check whether we have reached end of file.
stream.peek();
if (stream.eof()) {
break;
}
// Read the next message tag. Protobuf defines the message tag as
// (field_number << 3) | wire_type. In the legacy encoding, the field number
// is supposed to be 1 and the wire type for a length-delimited field is 2.
// In the new encoding we still expect the wire type to be 2, but the field
// number will be greater than 1.
constexpr uint64_t kExpectedV1Tag = (1 << 3) | 2;
size_t bytes_written = 0;
absl::optional<uint64_t> tag =
ParseVarInt(stream, buffer.data(), &bytes_written);
if (!tag) {
RTC_LOG(LS_WARNING)
<< "Missing field tag from beginning of protobuf event.";
return false;
}
constexpr uint64_t kWireTypeMask = 0x07;
const uint64_t wire_type = *tag & kWireTypeMask;
if (wire_type != 2) {
RTC_LOG(LS_WARNING) << "Expected field tag with wire type 2 (length "
"delimited message). Found wire type "
<< wire_type;
return false;
}
// Read the length field.
absl::optional<uint64_t> message_length =
ParseVarInt(stream, buffer.data(), &bytes_written);
if (!message_length) {
RTC_LOG(LS_WARNING) << "Missing message length after protobuf field tag.";
return false;
} else if (*message_length > kMaxEventSize) {
RTC_LOG(LS_WARNING) << "Protobuf message length is too large.";
return false;
}
// Read the next protobuf event to a temporary char buffer.
if (buffer.size() < bytes_written + *message_length)
buffer.resize(bytes_written + *message_length);
stream.read(buffer.data() + bytes_written, *message_length);
if (stream.gcount() != static_cast<int>(*message_length)) {
RTC_LOG(LS_WARNING) << "Failed to read protobuf message from file.";
return false;
}
size_t buffer_size = bytes_written + *message_length;
if (*tag == kExpectedV1Tag) {
// Parse the protobuf event from the buffer.
rtclog::EventStream event_stream;
if (!event_stream.ParseFromArray(buffer.data(), buffer_size)) {
RTC_LOG(LS_WARNING)
<< "Failed to parse legacy-format protobuf message.";
return false;
}
RTC_CHECK_EQ(event_stream.stream_size(), 1);
StoreParsedLegacyEvent(event_stream.stream(0));
} else {
// Parse the protobuf event from the buffer.
rtclog2::EventStream event_stream;
if (!event_stream.ParseFromArray(buffer.data(), buffer_size)) {
RTC_LOG(LS_WARNING) << "Failed to parse new-format protobuf message.";
return false;
}
StoreParsedNewFormatEvent(event_stream);
}
}
return true;
}
template <typename T>
void ParsedRtcEventLog::StoreFirstAndLastTimestamp(const std::vector<T>& v) {
if (v.empty())
return;
first_timestamp_ = std::min(first_timestamp_, v.front().log_time_us());
last_timestamp_ = std::max(last_timestamp_, v.back().log_time_us());
}
void ParsedRtcEventLog::StoreParsedLegacyEvent(const rtclog::Event& event) {
RTC_CHECK(event.has_type());
switch (event.type()) {
case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT: {
rtclog::StreamConfig config = GetVideoReceiveConfig(event);
video_recv_configs_.emplace_back(GetTimestamp(event), config);
if (!config.rtp_extensions.empty()) {
incoming_rtp_extensions_maps_[config.remote_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
incoming_rtp_extensions_maps_[config.rtx_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
}
break;
}
case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT: {
rtclog::StreamConfig config = GetVideoSendConfig(event);
video_send_configs_.emplace_back(GetTimestamp(event), config);
if (!config.rtp_extensions.empty()) {
outgoing_rtp_extensions_maps_[config.local_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
outgoing_rtp_extensions_maps_[config.rtx_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
}
break;
}
case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT: {
rtclog::StreamConfig config = GetAudioReceiveConfig(event);
audio_recv_configs_.emplace_back(GetTimestamp(event), config);
if (!config.rtp_extensions.empty()) {
incoming_rtp_extensions_maps_[config.remote_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
}
break;
}
case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT: {
rtclog::StreamConfig config = GetAudioSendConfig(event);
audio_send_configs_.emplace_back(GetTimestamp(event), config);
if (!config.rtp_extensions.empty()) {
outgoing_rtp_extensions_maps_[config.local_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
}
break;
}
case rtclog::Event::RTP_EVENT: {
PacketDirection direction;
uint8_t header[IP_PACKET_SIZE];
size_t header_length;
size_t total_length;
const RtpHeaderExtensionMap* extension_map = GetRtpHeader(
event, &direction, header, &header_length, &total_length, nullptr);
RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
RTPHeader parsed_header;
if (extension_map != nullptr) {
rtp_parser.Parse(&parsed_header, extension_map);
} else {
// Use the default extension map.
// TODO(terelius): This should be removed. GetRtpHeader will return the
// default map if the parser is configured for it.
// TODO(ivoc): Once configuration of audio streams is stored in the
// event log, this can be removed.
// Tracking bug: webrtc:6399
rtp_parser.Parse(&parsed_header, &default_extension_map_);
}
// 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;
RTC_CHECK(event.has_timestamp_us());
uint64_t timestamp_us = event.timestamp_us();
if (direction == kIncomingPacket) {
incoming_rtp_packets_map_[parsed_header.ssrc].push_back(
LoggedRtpPacketIncoming(timestamp_us, parsed_header, header_length,
total_length));
} else {
outgoing_rtp_packets_map_[parsed_header.ssrc].push_back(
LoggedRtpPacketOutgoing(timestamp_us, parsed_header, header_length,
total_length));
}
break;
}
case rtclog::Event::RTCP_EVENT: {
PacketDirection direction;
uint8_t packet[IP_PACKET_SIZE];
size_t total_length;
GetRtcpPacket(event, &direction, packet, &total_length);
uint64_t timestamp_us = GetTimestamp(event);
RTC_CHECK_LE(total_length, IP_PACKET_SIZE);
if (direction == kIncomingPacket) {
// Currently incoming RTCP packets are logged twice, both for audio and
// video. Only act on one of them. Compare against the previous parsed
// incoming RTCP packet.
if (total_length == last_incoming_rtcp_packet_length_ &&
memcmp(last_incoming_rtcp_packet_, packet, total_length) == 0)
break;
incoming_rtcp_packets_.push_back(
LoggedRtcpPacketIncoming(timestamp_us, packet, total_length));
last_incoming_rtcp_packet_length_ = total_length;
memcpy(last_incoming_rtcp_packet_, packet, total_length);
} else {
outgoing_rtcp_packets_.push_back(
LoggedRtcpPacketOutgoing(timestamp_us, packet, total_length));
}
break;
}
case rtclog::Event::LOG_START: {
start_log_events_.push_back(LoggedStartEvent(GetTimestamp(event)));
break;
}
case rtclog::Event::LOG_END: {
stop_log_events_.push_back(LoggedStopEvent(GetTimestamp(event)));
break;
}
case rtclog::Event::AUDIO_PLAYOUT_EVENT: {
LoggedAudioPlayoutEvent playout_event = GetAudioPlayout(event);
audio_playout_events_[playout_event.ssrc].push_back(playout_event);
break;
}
case rtclog::Event::LOSS_BASED_BWE_UPDATE: {
bwe_loss_updates_.push_back(GetLossBasedBweUpdate(event));
break;
}
case rtclog::Event::DELAY_BASED_BWE_UPDATE: {
bwe_delay_updates_.push_back(GetDelayBasedBweUpdate(event));
break;
}
case rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT: {
LoggedAudioNetworkAdaptationEvent ana_event =
GetAudioNetworkAdaptation(event);
audio_network_adaptation_events_.push_back(ana_event);
break;
}
case rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT: {
bwe_probe_cluster_created_events_.push_back(
GetBweProbeClusterCreated(event));
break;
}
case rtclog::Event::BWE_PROBE_RESULT_EVENT: {
// Probe successes and failures are currently stored in the same proto
// message, we are moving towards separate messages. Probe results
// therefore need special treatment in the parser.
RTC_CHECK(event.has_probe_result());
RTC_CHECK(event.probe_result().has_result());
if (event.probe_result().result() == rtclog::BweProbeResult::SUCCESS) {
bwe_probe_success_events_.push_back(GetBweProbeSuccess(event));
} else {
bwe_probe_failure_events_.push_back(GetBweProbeFailure(event));
}
break;
}
case rtclog::Event::ALR_STATE_EVENT: {
alr_state_events_.push_back(GetAlrState(event));
break;
}
case rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG: {
ice_candidate_pair_configs_.push_back(GetIceCandidatePairConfig(event));
break;
}
case rtclog::Event::ICE_CANDIDATE_PAIR_EVENT: {
ice_candidate_pair_events_.push_back(GetIceCandidatePairEvent(event));
break;
}
case rtclog::Event::UNKNOWN_EVENT: {
break;
}
}
}
int64_t ParsedRtcEventLog::GetTimestamp(const rtclog::Event& event) const {
RTC_CHECK(event.has_timestamp_us());
return event.timestamp_us();
}
// The header must have space for at least IP_PACKET_SIZE bytes.
const webrtc::RtpHeaderExtensionMap* 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_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::RTP_EVENT);
RTC_CHECK(event.has_rtp_packet());
const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
// Get direction of packet.
RTC_CHECK(rtp_packet.has_incoming());
if (incoming != nullptr) {
*incoming = rtp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
}
// Get packet length.
RTC_CHECK(rtp_packet.has_packet_length());
if (total_length != nullptr) {
*total_length = rtp_packet.packet_length();
}
// Get header length.
RTC_CHECK(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_CHECK_NE(*probe_cluster_id, PacedPacketInfo::kNotAProbe);
} else {
*probe_cluster_id = PacedPacketInfo::kNotAProbe;
}
}
// Get header contents.
if (header != nullptr) {
const size_t kMinRtpHeaderSize = 12;
RTC_CHECK_GE(rtp_packet.header().size(), kMinRtpHeaderSize);
RTC_CHECK_LE(rtp_packet.header().size(),
static_cast<size_t>(IP_PACKET_SIZE));
memcpy(header, rtp_packet.header().data(), rtp_packet.header().size());
uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(header + 8);
auto& extensions_maps = rtp_packet.incoming()
? incoming_rtp_extensions_maps_
: outgoing_rtp_extensions_maps_;
auto it = extensions_maps.find(ssrc);
if (it != extensions_maps.end()) {
return &(it->second);
}
if (parse_unconfigured_header_extensions_ ==
UnconfiguredHeaderExtensions::kAttemptWebrtcDefaultConfig) {
RTC_LOG(LS_WARNING) << "Using default header extension map for SSRC "
<< ssrc;
extensions_maps.insert(std::make_pair(ssrc, default_extension_map_));
return &default_extension_map_;
}
}
return nullptr;
}
// The packet must have space for at least IP_PACKET_SIZE bytes.
void ParsedRtcEventLog::GetRtcpPacket(const rtclog::Event& event,
PacketDirection* incoming,
uint8_t* packet,
size_t* length) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::RTCP_EVENT);
RTC_CHECK(event.has_rtcp_packet());
const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
// Get direction of packet.
RTC_CHECK(rtcp_packet.has_incoming());
if (incoming != nullptr) {
*incoming = rtcp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
}
// Get packet length.
RTC_CHECK(rtcp_packet.has_packet_data());
if (length != nullptr) {
*length = rtcp_packet.packet_data().size();
}
// Get packet contents.
if (packet != nullptr) {
RTC_CHECK_LE(rtcp_packet.packet_data().size(),
static_cast<unsigned>(IP_PACKET_SIZE));
memcpy(packet, rtcp_packet.packet_data().data(),
rtcp_packet.packet_data().size());
}
}
rtclog::StreamConfig ParsedRtcEventLog::GetVideoReceiveConfig(
const rtclog::Event& event) const {
rtclog::StreamConfig config;
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
RTC_CHECK(event.has_video_receiver_config());
const rtclog::VideoReceiveConfig& receiver_config =
event.video_receiver_config();
// Get SSRCs.
RTC_CHECK(receiver_config.has_remote_ssrc());
config.remote_ssrc = receiver_config.remote_ssrc();
RTC_CHECK(receiver_config.has_local_ssrc());
config.local_ssrc = receiver_config.local_ssrc();
config.rtx_ssrc = 0;
// Get RTCP settings.
RTC_CHECK(receiver_config.has_rtcp_mode());
config.rtcp_mode = GetRuntimeRtcpMode(receiver_config.rtcp_mode());
RTC_CHECK(receiver_config.has_remb());
config.remb = receiver_config.remb();
// Get RTX map.
std::map<uint32_t, const rtclog::RtxConfig> rtx_map;
for (int i = 0; i < receiver_config.rtx_map_size(); i++) {
const rtclog::RtxMap& map = receiver_config.rtx_map(i);
RTC_CHECK(map.has_payload_type());
RTC_CHECK(map.has_config());
RTC_CHECK(map.config().has_rtx_ssrc());
RTC_CHECK(map.config().has_rtx_payload_type());
rtx_map.insert(std::make_pair(map.payload_type(), map.config()));
}
// Get header extensions.
GetHeaderExtensions(&config.rtp_extensions,
receiver_config.header_extensions());
// Get decoders.
config.codecs.clear();
for (int i = 0; i < receiver_config.decoders_size(); i++) {
RTC_CHECK(receiver_config.decoders(i).has_name());
RTC_CHECK(receiver_config.decoders(i).has_payload_type());
int rtx_payload_type = 0;
auto rtx_it = rtx_map.find(receiver_config.decoders(i).payload_type());
if (rtx_it != rtx_map.end()) {
rtx_payload_type = rtx_it->second.rtx_payload_type();
if (config.rtx_ssrc != 0 &&
config.rtx_ssrc != rtx_it->second.rtx_ssrc()) {
RTC_LOG(LS_WARNING)
<< "RtcEventLog protobuf contained different SSRCs for "
"different received RTX payload types. Will only use "
"rtx_ssrc = "
<< config.rtx_ssrc << ".";
} else {
config.rtx_ssrc = rtx_it->second.rtx_ssrc();
}
}
config.codecs.emplace_back(receiver_config.decoders(i).name(),
receiver_config.decoders(i).payload_type(),
rtx_payload_type);
}
return config;
}
rtclog::StreamConfig ParsedRtcEventLog::GetVideoSendConfig(
const rtclog::Event& event) const {
rtclog::StreamConfig config;
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
RTC_CHECK(event.has_video_sender_config());
const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
// Get SSRCs.
RTC_CHECK_EQ(sender_config.ssrcs_size(), 1)
<< "VideoSendStreamConfig no longer stores multiple SSRCs. If you are "
"analyzing a very old log, try building the parser from the same "
"WebRTC version.";
config.local_ssrc = sender_config.ssrcs(0);
RTC_CHECK_LE(sender_config.rtx_ssrcs_size(), 1);
if (sender_config.rtx_ssrcs_size() == 1) {
config.rtx_ssrc = sender_config.rtx_ssrcs(0);
}
// Get header extensions.
GetHeaderExtensions(&config.rtp_extensions,
sender_config.header_extensions());
// Get the codec.
RTC_CHECK(sender_config.has_encoder());
RTC_CHECK(sender_config.encoder().has_name());
RTC_CHECK(sender_config.encoder().has_payload_type());
config.codecs.emplace_back(
sender_config.encoder().name(), sender_config.encoder().payload_type(),
sender_config.has_rtx_payload_type() ? sender_config.rtx_payload_type()
: 0);
return config;
}
rtclog::StreamConfig ParsedRtcEventLog::GetAudioReceiveConfig(
const rtclog::Event& event) const {
rtclog::StreamConfig config;
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT);
RTC_CHECK(event.has_audio_receiver_config());
const rtclog::AudioReceiveConfig& receiver_config =
event.audio_receiver_config();
// Get SSRCs.
RTC_CHECK(receiver_config.has_remote_ssrc());
config.remote_ssrc = receiver_config.remote_ssrc();
RTC_CHECK(receiver_config.has_local_ssrc());
config.local_ssrc = receiver_config.local_ssrc();
// Get header extensions.
GetHeaderExtensions(&config.rtp_extensions,
receiver_config.header_extensions());
return config;
}
rtclog::StreamConfig ParsedRtcEventLog::GetAudioSendConfig(
const rtclog::Event& event) const {
rtclog::StreamConfig config;
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_SENDER_CONFIG_EVENT);
RTC_CHECK(event.has_audio_sender_config());
const rtclog::AudioSendConfig& sender_config = event.audio_sender_config();
// Get SSRCs.
RTC_CHECK(sender_config.has_ssrc());
config.local_ssrc = sender_config.ssrc();
// Get header extensions.
GetHeaderExtensions(&config.rtp_extensions,
sender_config.header_extensions());
return config;
}
LoggedAudioPlayoutEvent ParsedRtcEventLog::GetAudioPlayout(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_PLAYOUT_EVENT);
RTC_CHECK(event.has_audio_playout_event());
const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
LoggedAudioPlayoutEvent res;
res.timestamp_us = GetTimestamp(event);
RTC_CHECK(playout_event.has_local_ssrc());
res.ssrc = playout_event.local_ssrc();
return res;
}
LoggedBweLossBasedUpdate ParsedRtcEventLog::GetLossBasedBweUpdate(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::LOSS_BASED_BWE_UPDATE);
RTC_CHECK(event.has_loss_based_bwe_update());
const rtclog::LossBasedBweUpdate& loss_event = event.loss_based_bwe_update();
LoggedBweLossBasedUpdate bwe_update;
bwe_update.timestamp_us = GetTimestamp(event);
RTC_CHECK(loss_event.has_bitrate_bps());
bwe_update.bitrate_bps = loss_event.bitrate_bps();
RTC_CHECK(loss_event.has_fraction_loss());
bwe_update.fraction_lost = loss_event.fraction_loss();
RTC_CHECK(loss_event.has_total_packets());
bwe_update.expected_packets = loss_event.total_packets();
return bwe_update;
}
LoggedBweDelayBasedUpdate ParsedRtcEventLog::GetDelayBasedBweUpdate(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::DELAY_BASED_BWE_UPDATE);
RTC_CHECK(event.has_delay_based_bwe_update());
const rtclog::DelayBasedBweUpdate& delay_event =
event.delay_based_bwe_update();
LoggedBweDelayBasedUpdate res;
res.timestamp_us = GetTimestamp(event);
RTC_CHECK(delay_event.has_bitrate_bps());
res.bitrate_bps = delay_event.bitrate_bps();
RTC_CHECK(delay_event.has_detector_state());
res.detector_state = GetRuntimeDetectorState(delay_event.detector_state());
return res;
}
LoggedAudioNetworkAdaptationEvent ParsedRtcEventLog::GetAudioNetworkAdaptation(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
RTC_CHECK(event.has_audio_network_adaptation());
const rtclog::AudioNetworkAdaptation& ana_event =
event.audio_network_adaptation();
LoggedAudioNetworkAdaptationEvent res;
res.timestamp_us = GetTimestamp(event);
if (ana_event.has_bitrate_bps())
res.config.bitrate_bps = ana_event.bitrate_bps();
if (ana_event.has_enable_fec())
res.config.enable_fec = ana_event.enable_fec();
if (ana_event.has_enable_dtx())
res.config.enable_dtx = ana_event.enable_dtx();
if (ana_event.has_frame_length_ms())
res.config.frame_length_ms = ana_event.frame_length_ms();
if (ana_event.has_num_channels())
res.config.num_channels = ana_event.num_channels();
if (ana_event.has_uplink_packet_loss_fraction())
res.config.uplink_packet_loss_fraction =
ana_event.uplink_packet_loss_fraction();
return res;
}
LoggedBweProbeClusterCreatedEvent ParsedRtcEventLog::GetBweProbeClusterCreated(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT);
RTC_CHECK(event.has_probe_cluster());
const rtclog::BweProbeCluster& pcc_event = event.probe_cluster();
LoggedBweProbeClusterCreatedEvent res;
res.timestamp_us = GetTimestamp(event);
RTC_CHECK(pcc_event.has_id());
res.id = pcc_event.id();
RTC_CHECK(pcc_event.has_bitrate_bps());
res.bitrate_bps = pcc_event.bitrate_bps();
RTC_CHECK(pcc_event.has_min_packets());
res.min_packets = pcc_event.min_packets();
RTC_CHECK(pcc_event.has_min_bytes());
res.min_bytes = pcc_event.min_bytes();
return res;
}
LoggedBweProbeFailureEvent ParsedRtcEventLog::GetBweProbeFailure(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_PROBE_RESULT_EVENT);
RTC_CHECK(event.has_probe_result());
const rtclog::BweProbeResult& pr_event = event.probe_result();
RTC_CHECK(pr_event.has_result());
RTC_CHECK_NE(pr_event.result(), rtclog::BweProbeResult::SUCCESS);
LoggedBweProbeFailureEvent res;
res.timestamp_us = GetTimestamp(event);
RTC_CHECK(pr_event.has_id());
res.id = pr_event.id();
RTC_CHECK(pr_event.has_result());
if (pr_event.result() ==
rtclog::BweProbeResult::INVALID_SEND_RECEIVE_INTERVAL) {
res.failure_reason = ProbeFailureReason::kInvalidSendReceiveInterval;
} else if (pr_event.result() ==
rtclog::BweProbeResult::INVALID_SEND_RECEIVE_RATIO) {
res.failure_reason = ProbeFailureReason::kInvalidSendReceiveRatio;
} else if (pr_event.result() == rtclog::BweProbeResult::TIMEOUT) {
res.failure_reason = ProbeFailureReason::kTimeout;
} else {
RTC_NOTREACHED();
}
RTC_CHECK(!pr_event.has_bitrate_bps());
return res;
}
LoggedBweProbeSuccessEvent ParsedRtcEventLog::GetBweProbeSuccess(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_PROBE_RESULT_EVENT);
RTC_CHECK(event.has_probe_result());
const rtclog::BweProbeResult& pr_event = event.probe_result();
RTC_CHECK(pr_event.has_result());
RTC_CHECK_EQ(pr_event.result(), rtclog::BweProbeResult::SUCCESS);
LoggedBweProbeSuccessEvent res;
res.timestamp_us = GetTimestamp(event);
RTC_CHECK(pr_event.has_id());
res.id = pr_event.id();
RTC_CHECK(pr_event.has_bitrate_bps());
res.bitrate_bps = pr_event.bitrate_bps();
return res;
}
LoggedAlrStateEvent ParsedRtcEventLog::GetAlrState(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::ALR_STATE_EVENT);
RTC_CHECK(event.has_alr_state());
const rtclog::AlrState& alr_event = event.alr_state();
LoggedAlrStateEvent res;
res.timestamp_us = GetTimestamp(event);
RTC_CHECK(alr_event.has_in_alr());
res.in_alr = alr_event.in_alr();
return res;
}
LoggedIceCandidatePairConfig ParsedRtcEventLog::GetIceCandidatePairConfig(
const rtclog::Event& rtc_event) const {
RTC_CHECK(rtc_event.has_type());
RTC_CHECK_EQ(rtc_event.type(), rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG);
LoggedIceCandidatePairConfig res;
const rtclog::IceCandidatePairConfig& config =
rtc_event.ice_candidate_pair_config();
res.timestamp_us = GetTimestamp(rtc_event);
RTC_CHECK(config.has_config_type());
res.type = GetRuntimeIceCandidatePairConfigType(config.config_type());
RTC_CHECK(config.has_candidate_pair_id());
res.candidate_pair_id = config.candidate_pair_id();
RTC_CHECK(config.has_local_candidate_type());
res.local_candidate_type =
GetRuntimeIceCandidateType(config.local_candidate_type());
RTC_CHECK(config.has_local_relay_protocol());
res.local_relay_protocol =
GetRuntimeIceCandidatePairProtocol(config.local_relay_protocol());
RTC_CHECK(config.has_local_network_type());
res.local_network_type =
GetRuntimeIceCandidateNetworkType(config.local_network_type());
RTC_CHECK(config.has_local_address_family());
res.local_address_family =
GetRuntimeIceCandidatePairAddressFamily(config.local_address_family());
RTC_CHECK(config.has_remote_candidate_type());
res.remote_candidate_type =
GetRuntimeIceCandidateType(config.remote_candidate_type());
RTC_CHECK(config.has_remote_address_family());
res.remote_address_family =
GetRuntimeIceCandidatePairAddressFamily(config.remote_address_family());
RTC_CHECK(config.has_candidate_pair_protocol());
res.candidate_pair_protocol =
GetRuntimeIceCandidatePairProtocol(config.candidate_pair_protocol());
return res;
}
LoggedIceCandidatePairEvent ParsedRtcEventLog::GetIceCandidatePairEvent(
const rtclog::Event& rtc_event) const {
RTC_CHECK(rtc_event.has_type());
RTC_CHECK_EQ(rtc_event.type(), rtclog::Event::ICE_CANDIDATE_PAIR_EVENT);
LoggedIceCandidatePairEvent res;
const rtclog::IceCandidatePairEvent& event =
rtc_event.ice_candidate_pair_event();
res.timestamp_us = GetTimestamp(rtc_event);
RTC_CHECK(event.has_event_type());
res.type = GetRuntimeIceCandidatePairEventType(event.event_type());
RTC_CHECK(event.has_candidate_pair_id());
res.candidate_pair_id = event.candidate_pair_id();
// transaction_id is not supported by rtclog::Event
res.transaction_id = 0;
return res;
}
// Returns the MediaType for registered SSRCs. Search from the end to use last
// registered types first.
ParsedRtcEventLog::MediaType ParsedRtcEventLog::GetMediaType(
uint32_t ssrc,
PacketDirection direction) const {
if (direction == kIncomingPacket) {
if (std::find(incoming_video_ssrcs_.begin(), incoming_video_ssrcs_.end(),
ssrc) != incoming_video_ssrcs_.end()) {
return MediaType::VIDEO;
}
if (std::find(incoming_audio_ssrcs_.begin(), incoming_audio_ssrcs_.end(),
ssrc) != incoming_audio_ssrcs_.end()) {
return MediaType::AUDIO;
}
} else {
if (std::find(outgoing_video_ssrcs_.begin(), outgoing_video_ssrcs_.end(),
ssrc) != outgoing_video_ssrcs_.end()) {
return MediaType::VIDEO;
}
if (std::find(outgoing_audio_ssrcs_.begin(), outgoing_audio_ssrcs_.end(),
ssrc) != outgoing_audio_ssrcs_.end()) {
return MediaType::AUDIO;
}
}
return MediaType::ANY;
}
std::vector<LoggedRouteChangeEvent> ParsedRtcEventLog::GetRouteChanges() const {
std::vector<LoggedRouteChangeEvent> route_changes;
for (auto& candidate : ice_candidate_pair_configs()) {
if (candidate.type == IceCandidatePairConfigType::kSelected) {
LoggedRouteChangeEvent route;
route.route_id = candidate.candidate_pair_id;
route.log_time = Timestamp::ms(candidate.log_time_ms());
route.send_overhead = kUdpOverhead + kSrtpOverhead + kIpv4Overhead;
if (candidate.remote_address_family ==
IceCandidatePairAddressFamily::kIpv6)
route.send_overhead += kIpv6Overhead - kIpv4Overhead;
if (candidate.remote_candidate_type != IceCandidateType::kLocal)
route.send_overhead += kStunOverhead;
route.return_overhead = kUdpOverhead + kSrtpOverhead + kIpv4Overhead;
if (candidate.remote_address_family ==
IceCandidatePairAddressFamily::kIpv6)
route.return_overhead += kIpv6Overhead - kIpv4Overhead;
if (candidate.remote_candidate_type != IceCandidateType::kLocal)
route.return_overhead += kStunOverhead;
route_changes.push_back(route);
}
}
return route_changes;
}
std::vector<LoggedPacketInfo> ParsedRtcEventLog::GetPacketInfos(
PacketDirection direction) const {
std::map<uint32_t, MediaStreamInfo> streams;
if (direction == PacketDirection::kIncomingPacket) {
AddRecvStreamInfos(&streams, audio_recv_configs(), LoggedMediaType::kAudio);
AddRecvStreamInfos(&streams, video_recv_configs(), LoggedMediaType::kVideo);
} else if (direction == PacketDirection::kOutgoingPacket) {
AddSendStreamInfos(&streams, audio_send_configs(), LoggedMediaType::kAudio);
AddSendStreamInfos(&streams, video_send_configs(), LoggedMediaType::kVideo);
}
TransportFeedbackAdapter feedback_adapter;
std::vector<OverheadChangeEvent> overheads =
GetOverheadChangingEvents(GetRouteChanges(), direction);
auto overhead_iter = overheads.begin();
std::vector<LoggedPacketInfo> packets;
std::map<int64_t, size_t> indices;
uint16_t current_overhead = kDefaultOverhead;
Timestamp last_log_time = Timestamp::Zero();
auto advance_time = [&](Timestamp new_log_time) {
if (overhead_iter != overheads.end() &&
new_log_time >= overhead_iter->timestamp) {
current_overhead = overhead_iter->overhead;
++overhead_iter;
}
RTC_DCHECK(new_log_time >= last_log_time);
last_log_time = new_log_time;
};
auto rtp_handler = [&](const LoggedRtpPacket& rtp) {
advance_time(Timestamp::ms(rtp.log_time_ms()));
MediaStreamInfo* stream = &streams[rtp.header.ssrc];
Timestamp capture_time = Timestamp::MinusInfinity();
if (!stream->rtx) {
// RTX copy the timestamp of the retransmitted packets. This means that
// RTX streams don't have a unique clock offset and frequency, so
// the RTP timstamps can't be unwrapped.
uint64_t capture_ticks =
stream->unwrap_capture_ticks.Unwrap(rtp.header.timestamp);
// TODO(srte): Use logged sample rate when it is added to the format.
capture_time = Timestamp::seconds(
capture_ticks /
(stream->media_type == LoggedMediaType::kAudio ? 48000.0 : 90000.0));
}
LoggedPacketInfo logged(rtp, stream->media_type, stream->rtx, capture_time);
logged.overhead = current_overhead;
if (rtp.header.extension.hasTransportSequenceNumber) {
logged.log_feedback_time = Timestamp::PlusInfinity();
rtc::SentPacket sent_packet;
sent_packet.send_time_ms = rtp.log_time_ms();
sent_packet.info.packet_size_bytes = rtp.total_length;
sent_packet.info.included_in_feedback = true;
sent_packet.packet_id = rtp.header.extension.transportSequenceNumber;
feedback_adapter.AddPacket(rtp.header.ssrc, sent_packet.packet_id,
rtp.total_length, PacedPacketInfo(),
Timestamp::ms(rtp.log_time_ms()));
auto sent_packet_msg = feedback_adapter.ProcessSentPacket(sent_packet);
RTC_CHECK(sent_packet_msg);
indices[sent_packet_msg->sequence_number] = packets.size();
}
packets.push_back(logged);
};
auto feedback_handler = [&](const LoggedRtcpPacketTransportFeedback& logged) {
advance_time(Timestamp::ms(logged.log_time_ms()));
auto msg = feedback_adapter.ProcessTransportFeedback(
logged.transport_feedback, Timestamp::ms(logged.log_time_ms()));
if (!msg.has_value() || msg->packet_feedbacks.empty())
return;
auto& last_fb = msg->packet_feedbacks.back();
Timestamp last_recv_time = last_fb.receive_time;
for (auto& fb : msg->packet_feedbacks) {
if (indices.find(fb.sent_packet.sequence_number) == indices.end()) {
RTC_LOG(LS_ERROR) << "Received feedback for unknown packet: "
<< fb.sent_packet.sequence_number;
continue;
}
LoggedPacketInfo* sent =
&packets[indices[fb.sent_packet.sequence_number]];
sent->reported_recv_time = fb.receive_time;
// Is we have received feedback with a valid receive time for this packet
// before, we keep the previous values.
if (sent->log_feedback_time.IsFinite() &&
sent->reported_recv_time.IsFinite())
continue;
sent->log_feedback_time = msg->feedback_time;
if (direction == PacketDirection::kOutgoingPacket) {
sent->feedback_hold_duration = last_recv_time - fb.receive_time;
} else {
sent->feedback_hold_duration =
Timestamp::ms(logged.log_time_ms()) - sent->log_packet_time;
}
sent->last_in_feedback = (&fb == &last_fb);
}
};
RtcEventProcessor process;
for (const auto& rtp_packets : rtp_packets_by_ssrc(direction)) {
process.AddEvents(rtp_packets.packet_view, rtp_handler);
}
if (direction == PacketDirection::kOutgoingPacket) {
process.AddEvents(incoming_transport_feedback_, feedback_handler);
} else {
process.AddEvents(outgoing_transport_feedback_, feedback_handler);
}
process.ProcessEventsInOrder();
return packets;
}
std::vector<LoggedIceCandidatePairConfig> ParsedRtcEventLog::GetIceCandidates()
const {
std::vector<LoggedIceCandidatePairConfig> candidates;
std::set<uint32_t> added;
for (auto& candidate : ice_candidate_pair_configs()) {
if (added.find(candidate.candidate_pair_id) == added.end()) {
candidates.push_back(candidate);
added.insert(candidate.candidate_pair_id);
}
}
return candidates;
}
std::vector<LoggedIceEvent> ParsedRtcEventLog::GetIceEvents() const {
using CheckType = IceCandidatePairEventType;
using ConfigType = IceCandidatePairConfigType;
using Combined = LoggedIceEventType;
std::map<CheckType, Combined> check_map(
{{CheckType::kCheckSent, Combined::kCheckSent},
{CheckType::kCheckReceived, Combined::kCheckReceived},
{CheckType::kCheckResponseSent, Combined::kCheckResponseSent},
{CheckType::kCheckResponseReceived, Combined::kCheckResponseReceived}});
std::map<ConfigType, Combined> config_map(
{{ConfigType::kAdded, Combined::kAdded},
{ConfigType::kUpdated, Combined::kUpdated},
{ConfigType::kDestroyed, Combined::kDestroyed},
{ConfigType::kSelected, Combined::kSelected}});
std::vector<LoggedIceEvent> log_events;
auto handle_check = [&](const LoggedIceCandidatePairEvent& check) {
log_events.push_back(LoggedIceEvent{check.candidate_pair_id,
Timestamp::ms(check.log_time_ms()),
check_map[check.type]});
};
auto handle_config = [&](const LoggedIceCandidatePairConfig& conf) {
log_events.push_back(LoggedIceEvent{conf.candidate_pair_id,
Timestamp::ms(conf.log_time_ms()),
config_map[conf.type]});
};
RtcEventProcessor process;
process.AddEvents(ice_candidate_pair_events(), handle_check);
process.AddEvents(ice_candidate_pair_configs(), handle_config);
return log_events;
}
const std::vector<MatchedSendArrivalTimes> GetNetworkTrace(
const ParsedRtcEventLog& parsed_log) {
std::vector<MatchedSendArrivalTimes> rtp_rtcp_matched;
for (auto& packet :
parsed_log.GetPacketInfos(PacketDirection::kOutgoingPacket)) {
if (packet.log_feedback_time.IsFinite()) {
rtp_rtcp_matched.emplace_back(
packet.log_feedback_time.ms(), packet.log_packet_time.ms(),
packet.reported_recv_time.ms_or(-1), packet.size);
}
}
return rtp_rtcp_matched;
}
// Helper functions for new format start here
void ParsedRtcEventLog::StoreParsedNewFormatEvent(
const rtclog2::EventStream& stream) {
RTC_DCHECK_EQ(stream.stream_size(), 0);
RTC_DCHECK_EQ(
stream.incoming_rtp_packets_size() + stream.outgoing_rtp_packets_size() +
stream.incoming_rtcp_packets_size() +
stream.outgoing_rtcp_packets_size() +
stream.audio_playout_events_size() + stream.begin_log_events_size() +
stream.end_log_events_size() + stream.loss_based_bwe_updates_size() +
stream.delay_based_bwe_updates_size() +
stream.dtls_transport_state_events_size() +
stream.dtls_writable_states_size() +
stream.audio_network_adaptations_size() +
stream.probe_clusters_size() + stream.probe_success_size() +
stream.probe_failure_size() + stream.alr_states_size() +
stream.ice_candidate_configs_size() +
stream.ice_candidate_events_size() +
stream.audio_recv_stream_configs_size() +
stream.audio_send_stream_configs_size() +
stream.video_recv_stream_configs_size() +
stream.video_send_stream_configs_size(),
1u);
if (stream.incoming_rtp_packets_size() == 1) {
StoreIncomingRtpPackets(stream.incoming_rtp_packets(0));
} else if (stream.outgoing_rtp_packets_size() == 1) {
StoreOutgoingRtpPackets(stream.outgoing_rtp_packets(0));
} else if (stream.incoming_rtcp_packets_size() == 1) {
StoreIncomingRtcpPackets(stream.incoming_rtcp_packets(0));
} else if (stream.outgoing_rtcp_packets_size() == 1) {
StoreOutgoingRtcpPackets(stream.outgoing_rtcp_packets(0));
} else if (stream.audio_playout_events_size() == 1) {
StoreAudioPlayoutEvent(stream.audio_playout_events(0));
} else if (stream.begin_log_events_size() == 1) {
StoreStartEvent(stream.begin_log_events(0));
} else if (stream.end_log_events_size() == 1) {
StoreStopEvent(stream.end_log_events(0));
} else if (stream.loss_based_bwe_updates_size() == 1) {
StoreBweLossBasedUpdate(stream.loss_based_bwe_updates(0));
} else if (stream.delay_based_bwe_updates_size() == 1) {
StoreBweDelayBasedUpdate(stream.delay_based_bwe_updates(0));
} else if (stream.dtls_transport_state_events_size() == 1) {
StoreDtlsTransportState(stream.dtls_transport_state_events(0));
} else if (stream.dtls_writable_states_size() == 1) {
StoreDtlsWritableState(stream.dtls_writable_states(0));
} else if (stream.audio_network_adaptations_size() == 1) {
StoreAudioNetworkAdaptationEvent(stream.audio_network_adaptations(0));
} else if (stream.probe_clusters_size() == 1) {
StoreBweProbeClusterCreated(stream.probe_clusters(0));
} else if (stream.probe_success_size() == 1) {
StoreBweProbeSuccessEvent(stream.probe_success(0));
} else if (stream.probe_failure_size() == 1) {
StoreBweProbeFailureEvent(stream.probe_failure(0));
} else if (stream.alr_states_size() == 1) {
StoreAlrStateEvent(stream.alr_states(0));
} else if (stream.ice_candidate_configs_size() == 1) {
StoreIceCandidatePairConfig(stream.ice_candidate_configs(0));
} else if (stream.ice_candidate_events_size() == 1) {
StoreIceCandidateEvent(stream.ice_candidate_events(0));
} else if (stream.audio_recv_stream_configs_size() == 1) {
StoreAudioRecvConfig(stream.audio_recv_stream_configs(0));
} else if (stream.audio_send_stream_configs_size() == 1) {
StoreAudioSendConfig(stream.audio_send_stream_configs(0));
} else if (stream.video_recv_stream_configs_size() == 1) {
StoreVideoRecvConfig(stream.video_recv_stream_configs(0));
} else if (stream.video_send_stream_configs_size() == 1) {
StoreVideoSendConfig(stream.video_send_stream_configs(0));
} else {
RTC_NOTREACHED();
}
}
void ParsedRtcEventLog::StoreAlrStateEvent(const rtclog2::AlrState& proto) {
RTC_CHECK(proto.has_timestamp_ms());
RTC_CHECK(proto.has_in_alr());
LoggedAlrStateEvent alr_event;
alr_event.timestamp_us = proto.timestamp_ms() * 1000;
alr_event.in_alr = proto.in_alr();
alr_state_events_.push_back(alr_event);
// TODO(terelius): Should we delta encode this event type?
}
void ParsedRtcEventLog::StoreAudioPlayoutEvent(
const rtclog2::AudioPlayoutEvents& proto) {
RTC_CHECK(proto.has_timestamp_ms());
RTC_CHECK(proto.has_local_ssrc());
// Base event
auto map_it = audio_playout_events_[proto.local_ssrc()];
audio_playout_events_[proto.local_ssrc()].emplace_back(
1000 * proto.timestamp_ms(), proto.local_ssrc());
const size_t number_of_deltas =
proto.has_number_of_deltas() ? proto.number_of_deltas() : 0u;
if (number_of_deltas == 0) {
return;
}
// timestamp_ms
std::vector<absl::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_CHECK_EQ(timestamp_ms_values.size(), number_of_deltas);
// local_ssrc
std::vector<absl::optional<uint64_t>> local_ssrc_values = DecodeDeltas(
proto.local_ssrc_deltas(), proto.local_ssrc(), number_of_deltas);
RTC_CHECK_EQ(local_ssrc_values.size(), number_of_deltas);
// Delta decoding
for (size_t i = 0; i < number_of_deltas; ++i) {
RTC_CHECK(timestamp_ms_values[i].has_value());
RTC_CHECK(local_ssrc_values[i].has_value());
RTC_CHECK_LE(local_ssrc_values[i].value(),
std::numeric_limits<uint32_t>::max());
int64_t timestamp_ms;
RTC_CHECK(ToSigned(timestamp_ms_values[i].value(), &timestamp_ms));
const uint32_t local_ssrc =
static_cast<uint32_t>(local_ssrc_values[i].value());
audio_playout_events_[local_ssrc].emplace_back(1000 * timestamp_ms,
local_ssrc);
}
}
void ParsedRtcEventLog::StoreIncomingRtpPackets(
const rtclog2::IncomingRtpPackets& proto) {
StoreRtpPackets(proto, &incoming_rtp_packets_map_);
}
void ParsedRtcEventLog::StoreOutgoingRtpPackets(
const rtclog2::OutgoingRtpPackets& proto) {
StoreRtpPackets(proto, &outgoing_rtp_packets_map_);
}
void ParsedRtcEventLog::StoreIncomingRtcpPackets(
const rtclog2::IncomingRtcpPackets& proto) {
StoreRtcpPackets(proto, &incoming_rtcp_packets_, /*remove_duplicates=*/true);
}
void ParsedRtcEventLog::StoreOutgoingRtcpPackets(
const rtclog2::OutgoingRtcpPackets& proto) {
StoreRtcpPackets(proto, &outgoing_rtcp_packets_, /*remove_duplicates=*/false);
}
void ParsedRtcEventLog::StoreStartEvent(const rtclog2::BeginLogEvent& proto) {
RTC_CHECK(proto.has_timestamp_ms());
RTC_CHECK(proto.has_version());
RTC_CHECK(proto.has_utc_time_ms());
RTC_CHECK_EQ(proto.version(), 2);
LoggedStartEvent start_event(proto.timestamp_ms() * 1000,
proto.utc_time_ms());
start_log_events_.push_back(start_event);
}
void ParsedRtcEventLog::StoreStopEvent(const rtclog2::EndLogEvent& proto) {
RTC_CHECK(proto.has_timestamp_ms());
LoggedStopEvent stop_event(proto.timestamp_ms() * 1000);
stop_log_events_.push_back(stop_event);
}
void ParsedRtcEventLog::StoreBweLossBasedUpdate(
const rtclog2::LossBasedBweUpdates& proto) {
RTC_CHECK(proto.has_timestamp_ms());
RTC_CHECK(proto.has_bitrate_bps());
RTC_CHECK(proto.has_fraction_loss());
RTC_CHECK(proto.has_total_packets());
// Base event
bwe_loss_updates_.emplace_back(1000 * proto.timestamp_ms(),
proto.bitrate_bps(), proto.fraction_loss(),
proto.total_packets());
const size_t number_of_deltas =
proto.has_number_of_deltas() ? proto.number_of_deltas() : 0u;
if (number_of_deltas == 0) {
return;
}
// timestamp_ms
std::vector<absl::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_CHECK_EQ(timestamp_ms_values.size(), number_of_deltas);
// bitrate_bps
std::vector<absl::optional<uint64_t>> bitrate_bps_values = DecodeDeltas(
proto.bitrate_bps_deltas(), proto.bitrate_bps(), number_of_deltas);
RTC_CHECK_EQ(bitrate_bps_values.size(), number_of_deltas);
// fraction_loss
std::vector<absl::optional<uint64_t>> fraction_loss_values = DecodeDeltas(
proto.fraction_loss_deltas(), proto.fraction_loss(), number_of_deltas);
RTC_CHECK_EQ(fraction_loss_values.size(), number_of_deltas);
// total_packets
std::vector<absl::optional<uint64_t>> total_packets_values = DecodeDeltas(
proto.total_packets_deltas(), proto.total_packets(), number_of_deltas);
RTC_CHECK_EQ(total_packets_values.size(), number_of_deltas);
// Delta decoding
for (size_t i = 0; i < number_of_deltas; ++i) {
RTC_CHECK(timestamp_ms_values[i].has_value());
int64_t timestamp_ms;
RTC_CHECK(ToSigned(timestamp_ms_values[i].value(), &timestamp_ms));
RTC_CHECK(bitrate_bps_values[i].has_value());
RTC_CHECK_LE(bitrate_bps_values[i].value(),
std::numeric_limits<uint32_t>::max());
const uint32_t bitrate_bps =
static_cast<uint32_t>(bitrate_bps_values[i].value());
RTC_CHECK(fraction_loss_values[i].has_value());
RTC_CHECK_LE(fraction_loss_values[i].value(),
std::numeric_limits<uint32_t>::max());
const uint32_t fraction_loss =
static_cast<uint32_t>(fraction_loss_values[i].value());
RTC_CHECK(total_packets_values[i].has_value());
RTC_CHECK_LE(total_packets_values[i].value(),
std::numeric_limits<uint32_t>::max());
const uint32_t total_packets =
static_cast<uint32_t>(total_packets_values[i].value());
bwe_loss_updates_.emplace_back(1000 * timestamp_ms, bitrate_bps,
fraction_loss, total_packets);
}
}
void ParsedRtcEventLog::StoreBweDelayBasedUpdate(
const rtclog2::DelayBasedBweUpdates& proto) {
RTC_CHECK(proto.has_timestamp_ms());
RTC_CHECK(proto.has_bitrate_bps());
RTC_CHECK(proto.has_detector_state());
// Base event
const BandwidthUsage base_detector_state =
GetRuntimeDetectorState(proto.detector_state());
bwe_delay_updates_.emplace_back(1000 * proto.timestamp_ms(),
proto.bitrate_bps(), base_detector_state);
const size_t number_of_deltas =
proto.has_number_of_deltas() ? proto.number_of_deltas() : 0u;
if (number_of_deltas == 0) {
return;
}
// timestamp_ms
std::vector<absl::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_CHECK_EQ(timestamp_ms_values.size(), number_of_deltas);
// bitrate_bps
std::vector<absl::optional<uint64_t>> bitrate_bps_values = DecodeDeltas(
proto.bitrate_bps_deltas(), proto.bitrate_bps(), number_of_deltas);
RTC_CHECK_EQ(bitrate_bps_values.size(), number_of_deltas);
// detector_state
std::vector<absl::optional<uint64_t>> detector_state_values = DecodeDeltas(
proto.detector_state_deltas(),
static_cast<uint64_t>(proto.detector_state()), number_of_deltas);
RTC_CHECK_EQ(detector_state_values.size(), number_of_deltas);
// Delta decoding
for (size_t i = 0; i < number_of_deltas; ++i) {
RTC_CHECK(timestamp_ms_values[i].has_value());
int64_t timestamp_ms;
RTC_CHECK(ToSigned(timestamp_ms_values[i].value(), &timestamp_ms));
RTC_CHECK(bitrate_bps_values[i].has_value());
RTC_CHECK_LE(bitrate_bps_values[i].value(),
std::numeric_limits<uint32_t>::max());
const uint32_t bitrate_bps =
static_cast<uint32_t>(bitrate_bps_values[i].value());
RTC_CHECK(detector_state_values[i].has_value());
const auto detector_state =
static_cast<rtclog2::DelayBasedBweUpdates::DetectorState>(
detector_state_values[i].value());
bwe_delay_updates_.emplace_back(1000 * timestamp_ms, bitrate_bps,
GetRuntimeDetectorState(detector_state));
}
}
void ParsedRtcEventLog::StoreBweProbeClusterCreated(
const rtclog2::BweProbeCluster& proto) {
LoggedBweProbeClusterCreatedEvent probe_cluster;
RTC_CHECK(proto.has_timestamp_ms());
probe_cluster.timestamp_us = proto.timestamp_ms() * 1000;
RTC_CHECK(proto.has_id());
probe_cluster.id = proto.id();
RTC_CHECK(proto.has_bitrate_bps());
probe_cluster.bitrate_bps = proto.bitrate_bps();
RTC_CHECK(proto.has_min_packets());
probe_cluster.min_packets = proto.min_packets();
RTC_CHECK(proto.has_min_bytes());
probe_cluster.min_bytes = proto.min_bytes();
bwe_probe_cluster_created_events_.push_back(probe_cluster);
// TODO(terelius): Should we delta encode this event type?
}
void ParsedRtcEventLog::StoreBweProbeSuccessEvent(
const rtclog2::BweProbeResultSuccess& proto) {
LoggedBweProbeSuccessEvent probe_result;
RTC_CHECK(proto.has_timestamp_ms());
probe_result.timestamp_us = proto.timestamp_ms() * 1000;
RTC_CHECK(proto.has_id());
probe_result.id = proto.id();
RTC_CHECK(proto.has_bitrate_bps());
probe_result.bitrate_bps = proto.bitrate_bps();
bwe_probe_success_events_.push_back(probe_result);
// TODO(terelius): Should we delta encode this event type?
}
void ParsedRtcEventLog::StoreBweProbeFailureEvent(
const rtclog2::BweProbeResultFailure& proto) {
LoggedBweProbeFailureEvent probe_result;
RTC_CHECK(proto.has_timestamp_ms());
probe_result.timestamp_us = proto.timestamp_ms() * 1000;
RTC_CHECK(proto.has_id());
probe_result.id = proto.id();
RTC_CHECK(proto.has_failure());
probe_result.failure_reason = GetRuntimeProbeFailureReason(proto.failure());
bwe_probe_failure_events_.push_back(probe_result);
// TODO(terelius): Should we delta encode this event type?
}
void ParsedRtcEventLog::StoreAudioNetworkAdaptationEvent(
const rtclog2::AudioNetworkAdaptations& proto) {
RTC_CHECK(proto.has_timestamp_ms());
// Base event
{
AudioEncoderRuntimeConfig runtime_config;
if (proto.has_bitrate_bps()) {
runtime_config.bitrate_bps = proto.bitrate_bps();
}
if (proto.has_frame_length_ms()) {
runtime_config.frame_length_ms = proto.frame_length_ms();
}
if (proto.has_uplink_packet_loss_fraction()) {
float uplink_packet_loss_fraction;
RTC_CHECK(ParsePacketLossFractionFromProtoFormat(
proto.uplink_packet_loss_fraction(), &uplink_packet_loss_fraction));
runtime_config.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
}
if (proto.has_enable_fec()) {
runtime_config.enable_fec = proto.enable_fec();
}
if (proto.has_enable_dtx()) {
runtime_config.enable_dtx = proto.enable_dtx();
}
if (proto.has_num_channels()) {
// Note: Encoding N as N-1 only done for |num_channels_deltas|.
runtime_config.num_channels = proto.num_channels();
}
audio_network_adaptation_events_.emplace_back(1000 * proto.timestamp_ms(),
runtime_config);
}
const size_t number_of_deltas =
proto.has_number_of_deltas() ? proto.number_of_deltas() : 0u;
if (number_of_deltas == 0) {
return;
}
// timestamp_ms
std::vector<absl::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_CHECK_EQ(timestamp_ms_values.size(), number_of_deltas);
// bitrate_bps
const absl::optional<uint64_t> unsigned_base_bitrate_bps =
proto.has_bitrate_bps()
? absl::optional<uint64_t>(ToUnsigned(proto.bitrate_bps()))
: absl::optional<uint64_t>();
std::vector<absl::optional<uint64_t>> bitrate_bps_values = DecodeDeltas(
proto.bitrate_bps_deltas(), unsigned_base_bitrate_bps, number_of_deltas);
RTC_CHECK_EQ(bitrate_bps_values.size(), number_of_deltas);
// frame_length_ms
const absl::optional<uint64_t> unsigned_base_frame_length_ms =
proto.has_frame_length_ms()
? absl::optional<uint64_t>(ToUnsigned(proto.frame_length_ms()))
: absl::optional<uint64_t>();
std::vector<absl::optional<uint64_t>> frame_length_ms_values =
DecodeDeltas(proto.frame_length_ms_deltas(),
unsigned_base_frame_length_ms, number_of_deltas);
RTC_CHECK_EQ(frame_length_ms_values.size(), number_of_deltas);
// uplink_packet_loss_fraction
const absl::optional<uint64_t> uplink_packet_loss_fraction =
proto.has_uplink_packet_loss_fraction()
? absl::optional<uint64_t>(proto.uplink_packet_loss_fraction())
: absl::optional<uint64_t>();
std::vector<absl::optional<uint64_t>> uplink_packet_loss_fraction_values =
DecodeDeltas(proto.uplink_packet_loss_fraction_deltas(),
uplink_packet_loss_fraction, number_of_deltas);
RTC_CHECK_EQ(uplink_packet_loss_fraction_values.size(), number_of_deltas);
// enable_fec
const absl::optional<uint64_t> enable_fec =
proto.has_enable_fec() ? absl::optional<uint64_t>(proto.enable_fec())
: absl::optional<uint64_t>();
std::vector<absl::optional<uint64_t>> enable_fec_values =
DecodeDeltas(proto.enable_fec_deltas(), enable_fec, number_of_deltas);
RTC_CHECK_EQ(enable_fec_values.size(), number_of_deltas);
// enable_dtx
const absl::optional<uint64_t> enable_dtx =
proto.has_enable_dtx() ? absl::optional<uint64_t>(proto.enable_dtx())
: absl::optional<uint64_t>();
std::vector<absl::optional<uint64_t>> enable_dtx_values =
DecodeDeltas(proto.enable_dtx_deltas(), enable_dtx, number_of_deltas);
RTC_CHECK_EQ(enable_dtx_values.size(), number_of_deltas);
// num_channels
// Note: For delta encoding, all num_channel values, including the base,
// were shifted down by one, but in the base event, they were not.
// We likewise shift the base event down by one, to get the same base as
// encoding had, but then shift all of the values (except the base) back up
// to their original value.
absl::optional<uint64_t> shifted_base_num_channels;
if (proto.has_num_channels()) {
shifted_base_num_channels =
absl::optional<uint64_t>(proto.num_channels() - 1);
}
std::vector<absl::optional<uint64_t>> num_channels_values = DecodeDeltas(
proto.num_channels_deltas(), shifted_base_num_channels, number_of_deltas);
for (size_t i = 0; i < num_channels_values.size(); ++i) {
if (num_channels_values[i].has_value()) {
num_channels_values[i] = num_channels_values[i].value() + 1;
}
}
RTC_CHECK_EQ(num_channels_values.size(), number_of_deltas);
// Delta decoding
for (size_t i = 0; i < number_of_deltas; ++i) {
RTC_CHECK(timestamp_ms_values[i].has_value());
int64_t timestamp_ms;
RTC_CHECK(ToSigned(timestamp_ms_values[i].value(), &timestamp_ms));
AudioEncoderRuntimeConfig runtime_config;
if (bitrate_bps_values[i].has_value()) {
int signed_bitrate_bps;
RTC_CHECK(ToSigned(bitrate_bps_values[i].value(), &signed_bitrate_bps));
runtime_config.bitrate_bps = signed_bitrate_bps;
}
if (frame_length_ms_values[i].has_value()) {
int signed_frame_length_ms;
RTC_CHECK(
ToSigned(frame_length_ms_values[i].value(), &signed_frame_length_ms));
runtime_config.frame_length_ms = signed_frame_length_ms;
}
if (uplink_packet_loss_fraction_values[i].has_value()) {
float uplink_packet_loss_fraction;
RTC_CHECK(ParsePacketLossFractionFromProtoFormat(
rtc::checked_cast<uint32_t>(
uplink_packet_loss_fraction_values[i].value()),
&uplink_packet_loss_fraction));
runtime_config.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
}
if (enable_fec_values[i].has_value()) {
runtime_config.enable_fec =
rtc::checked_cast<bool>(enable_fec_values[i].value());
}
if (enable_dtx_values[i].has_value()) {
runtime_config.enable_dtx =
rtc::checked_cast<bool>(enable_dtx_values[i].value());
}
if (num_channels_values[i].has_value()) {
runtime_config.num_channels =
rtc::checked_cast<size_t>(num_channels_values[i].value());
}
audio_network_adaptation_events_.emplace_back(1000 * timestamp_ms,
runtime_config);
}
}
void ParsedRtcEventLog::StoreDtlsTransportState(
const rtclog2::DtlsTransportStateEvent& proto) {
LoggedDtlsTransportState dtls_state;
RTC_CHECK(proto.has_timestamp_ms());
dtls_state.timestamp_us = proto.timestamp_ms() * 1000;
RTC_CHECK(proto.has_dtls_transport_state());
dtls_state.dtls_transport_state =
GetRuntimeDtlsTransportState(proto.dtls_transport_state());
dtls_transport_states_.push_back(dtls_state);
}
void ParsedRtcEventLog::StoreDtlsWritableState(
const rtclog2::DtlsWritableState& proto) {
LoggedDtlsWritableState dtls_writable_state;
RTC_CHECK(proto.has_timestamp_ms());
dtls_writable_state.timestamp_us = proto.timestamp_ms() * 1000;
RTC_CHECK(proto.has_writable());
dtls_writable_state.writable = proto.writable();
dtls_writable_states_.push_back(dtls_writable_state);
}
void ParsedRtcEventLog::StoreIceCandidatePairConfig(
const rtclog2::IceCandidatePairConfig& proto) {
LoggedIceCandidatePairConfig ice_config;
RTC_CHECK(proto.has_timestamp_ms());
ice_config.timestamp_us = proto.timestamp_ms() * 1000;
RTC_CHECK(proto.has_config_type());
ice_config.type = GetRuntimeIceCandidatePairConfigType(proto.config_type());
RTC_CHECK(proto.has_candidate_pair_id());
ice_config.candidate_pair_id = proto.candidate_pair_id();
RTC_CHECK(proto.has_local_candidate_type());
ice_config.local_candidate_type =
GetRuntimeIceCandidateType(proto.local_candidate_type());
RTC_CHECK(proto.has_local_relay_protocol());
ice_config.local_relay_protocol =
GetRuntimeIceCandidatePairProtocol(proto.local_relay_protocol());
RTC_CHECK(proto.has_local_network_type());
ice_config.local_network_type =
GetRuntimeIceCandidateNetworkType(proto.local_network_type());
RTC_CHECK(proto.has_local_address_family());
ice_config.local_address_family =
GetRuntimeIceCandidatePairAddressFamily(proto.local_address_family());
RTC_CHECK(proto.has_remote_candidate_type());
ice_config.remote_candidate_type =
GetRuntimeIceCandidateType(proto.remote_candidate_type());
RTC_CHECK(proto.has_remote_address_family());
ice_config.remote_address_family =
GetRuntimeIceCandidatePairAddressFamily(proto.remote_address_family());
RTC_CHECK(proto.has_candidate_pair_protocol());
ice_config.candidate_pair_protocol =
GetRuntimeIceCandidatePairProtocol(proto.candidate_pair_protocol());
ice_candidate_pair_configs_.push_back(ice_config);
// TODO(terelius): Should we delta encode this event type?
}
void ParsedRtcEventLog::StoreIceCandidateEvent(
const rtclog2::IceCandidatePairEvent& proto) {
LoggedIceCandidatePairEvent ice_event;
RTC_CHECK(proto.has_timestamp_ms());
ice_event.timestamp_us = proto.timestamp_ms() * 1000;
RTC_CHECK(proto.has_event_type());
ice_event.type = GetRuntimeIceCandidatePairEventType(proto.event_type());
RTC_CHECK(proto.has_candidate_pair_id());
ice_event.candidate_pair_id = proto.candidate_pair_id();
// TODO(zstein): Make the transaction_id field required once all old versions
// of the log (which don't have the field) are obsolete.
ice_event.transaction_id =
proto.has_transaction_id() ? proto.transaction_id() : 0;
ice_candidate_pair_events_.push_back(ice_event);
// TODO(terelius): Should we delta encode this event type?
}
void ParsedRtcEventLog::StoreVideoRecvConfig(
const rtclog2::VideoRecvStreamConfig& proto) {
LoggedVideoRecvConfig stream;
RTC_CHECK(proto.has_timestamp_ms());
stream.timestamp_us = proto.timestamp_ms() * 1000;
RTC_CHECK(proto.has_remote_ssrc());
stream.config.remote_ssrc = proto.remote_ssrc();
RTC_CHECK(proto.has_local_ssrc());
stream.config.local_ssrc = proto.local_ssrc();
if (proto.has_rtx_ssrc()) {
stream.config.rtx_ssrc = proto.rtx_ssrc();
}
if (proto.has_header_extensions()) {
stream.config.rtp_extensions =
GetRuntimeRtpHeaderExtensionConfig(proto.header_extensions());
}
video_recv_configs_.push_back(stream);
}
void ParsedRtcEventLog::StoreVideoSendConfig(
const rtclog2::VideoSendStreamConfig& proto) {
LoggedVideoSendConfig stream;
RTC_CHECK(proto.has_timestamp_ms());
stream.timestamp_us = proto.timestamp_ms() * 1000;
RTC_CHECK(proto.has_ssrc());
stream.config.local_ssrc = proto.ssrc();
if (proto.has_rtx_ssrc()) {
stream.config.rtx_ssrc = proto.rtx_ssrc();
}
if (proto.has_header_extensions()) {
stream.config.rtp_extensions =
GetRuntimeRtpHeaderExtensionConfig(proto.header_extensions());
}
video_send_configs_.push_back(stream);
}
void ParsedRtcEventLog::StoreAudioRecvConfig(
const rtclog2::AudioRecvStreamConfig& proto) {
LoggedAudioRecvConfig stream;
RTC_CHECK(proto.has_timestamp_ms());
stream.timestamp_us = proto.timestamp_ms() * 1000;
RTC_CHECK(proto.has_remote_ssrc());
stream.config.remote_ssrc = proto.remote_ssrc();
RTC_CHECK(proto.has_local_ssrc());
stream.config.local_ssrc = proto.local_ssrc();
if (proto.has_header_extensions()) {
stream.config.rtp_extensions =
GetRuntimeRtpHeaderExtensionConfig(proto.header_extensions());
}
audio_recv_configs_.push_back(stream);
}
void ParsedRtcEventLog::StoreAudioSendConfig(
const rtclog2::AudioSendStreamConfig& proto) {
LoggedAudioSendConfig stream;
RTC_CHECK(proto.has_timestamp_ms());
stream.timestamp_us = proto.timestamp_ms() * 1000;
RTC_CHECK(proto.has_ssrc());
stream.config.local_ssrc = proto.ssrc();
if (proto.has_header_extensions()) {
stream.config.rtp_extensions =
GetRuntimeRtpHeaderExtensionConfig(proto.header_extensions());
}
audio_send_configs_.push_back(stream);
}
} // namespace webrtc