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webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
Elad Alon 157540ac05 Stop hard-coding default IDs for RTP extensions 
Hard-coding default values forces IDs over 14 to be used even
when we offer less than 15 different extensions.

Note that the code relies on MergeRtpHdrExts for making sure
that extension IDs are kept consistent and non-colliding between
different streams (audio/video).

Bug: webrtc:10288
Change-Id: I3e59f7ddc8ca43cea91084a6b7f36df70fb6be4a
Reviewed-on: https://webrtc-review.googlesource.com/c/121646
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26622}
2019-02-09 01:04:35 +00:00

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/*
* 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() {
// Values from before the default RTP header extension IDs were removed.
constexpr int kAudioLevelDefaultId = 1;
constexpr int kTimestampOffsetDefaultId = 2;
constexpr int kAbsSendTimeDefaultId = 3;
constexpr int kVideoRotationDefaultId = 4;
constexpr int kTransportSequenceNumberDefaultId = 5;
constexpr int kPlayoutDelayDefaultId = 6;
constexpr int kVideoContentTypeDefaultId = 7;
constexpr int kVideoTimingDefaultId = 8;
webrtc::RtpHeaderExtensionMap default_map;
default_map.Register<AudioLevel>(kAudioLevelDefaultId);
default_map.Register<TransmissionOffset>(kTimestampOffsetDefaultId);
default_map.Register<AbsoluteSendTime>(kAbsSendTimeDefaultId);
default_map.Register<VideoOrientation>(kVideoRotationDefaultId);
default_map.Register<TransportSequenceNumber>(
kTransportSequenceNumberDefaultId);
default_map.Register<PlayoutDelayLimits>(kPlayoutDelayDefaultId);
default_map.Register<VideoContentTypeExtension>(kVideoContentTypeDefaultId);
default_map.Register<VideoTimingExtension>(kVideoTimingDefaultId);
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);
process.ProcessEventsInOrder();
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
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