blob: a7237ee8511507228ee026a08e7440f522138976 [file] [log] [blame]
/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include <stdint.h>
#include <string.h>
#include <algorithm>
#include <fstream>
#include <istream>
#include <map>
#include <utility>
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/protobuf_utils.h"
namespace webrtc {
namespace {
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;
}
ParsedRtcEventLog::EventType GetRuntimeEventType(
rtclog::Event::EventType event_type) {
switch (event_type) {
case rtclog::Event::UNKNOWN_EVENT:
return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
case rtclog::Event::LOG_START:
return ParsedRtcEventLog::EventType::LOG_START;
case rtclog::Event::LOG_END:
return ParsedRtcEventLog::EventType::LOG_END;
case rtclog::Event::RTP_EVENT:
return ParsedRtcEventLog::EventType::RTP_EVENT;
case rtclog::Event::RTCP_EVENT:
return ParsedRtcEventLog::EventType::RTCP_EVENT;
case rtclog::Event::AUDIO_PLAYOUT_EVENT:
return ParsedRtcEventLog::EventType::AUDIO_PLAYOUT_EVENT;
case rtclog::Event::LOSS_BASED_BWE_UPDATE:
return ParsedRtcEventLog::EventType::LOSS_BASED_BWE_UPDATE;
case rtclog::Event::DELAY_BASED_BWE_UPDATE:
return ParsedRtcEventLog::EventType::DELAY_BASED_BWE_UPDATE;
case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT:
return ParsedRtcEventLog::EventType::VIDEO_RECEIVER_CONFIG_EVENT;
case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT:
return ParsedRtcEventLog::EventType::VIDEO_SENDER_CONFIG_EVENT;
case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT:
return ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT;
case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
return ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT;
case rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT:
return ParsedRtcEventLog::EventType::AUDIO_NETWORK_ADAPTATION_EVENT;
case rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT:
return ParsedRtcEventLog::EventType::BWE_PROBE_CLUSTER_CREATED_EVENT;
case rtclog::Event::BWE_PROBE_RESULT_EVENT:
return ParsedRtcEventLog::EventType::BWE_PROBE_RESULT_EVENT;
}
RTC_NOTREACHED();
return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
}
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;
}
std::pair<uint64_t, bool> ParseVarInt(std::istream& stream) {
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 std::make_pair(varint, false);
}
RTC_DCHECK_GE(byte, 0);
RTC_DCHECK_LE(byte, 255);
varint |= static_cast<uint64_t>(byte & 0x7F) << (7 * bytes_read);
if ((byte & 0x80) == 0) {
return std::make_pair(varint, true);
}
}
return std::make_pair(varint, false);
}
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));
}
}
} // namespace
bool ParsedRtcEventLog::ParseFile(const std::string& filename) {
std::ifstream file(filename, std::ios_base::in | std::ios_base::binary);
if (!file.good() || !file.is_open()) {
LOG(LS_WARNING) << "Could not open file for reading.";
return false;
}
return ParseStream(file);
}
bool ParsedRtcEventLog::ParseString(const std::string& s) {
std::istringstream stream(s, std::ios_base::in | std::ios_base::binary);
return ParseStream(stream);
}
bool ParsedRtcEventLog::ParseStream(std::istream& stream) {
events_.clear();
const size_t kMaxEventSize = (1u << 16) - 1;
std::vector<char> tmp_buffer(kMaxEventSize);
uint64_t tag;
uint64_t message_length;
bool success;
RTC_DCHECK(stream.good());
while (1) {
// Check whether we have reached end of file.
stream.peek();
if (stream.eof()) {
// Process all extensions maps for faster look-up later.
for (auto& event_stream : streams_) {
rtp_extensions_maps_[StreamId(event_stream.ssrc,
event_stream.direction)] =
&event_stream.rtp_extensions_map;
}
return true;
}
// Read the next message tag. The tag number is defined as
// (fieldnumber << 3) | wire_type. In our case, the field number is
// supposed to be 1 and the wire type for an
// length-delimited field is 2.
const uint64_t kExpectedTag = (1 << 3) | 2;
std::tie(tag, success) = ParseVarInt(stream);
if (!success) {
LOG(LS_WARNING) << "Missing field tag from beginning of protobuf event.";
return false;
} else if (tag != kExpectedTag) {
LOG(LS_WARNING) << "Unexpected field tag at beginning of protobuf event.";
return false;
}
// Read the length field.
std::tie(message_length, success) = ParseVarInt(stream);
if (!success) {
LOG(LS_WARNING) << "Missing message length after protobuf field tag.";
return false;
} else if (message_length > kMaxEventSize) {
LOG(LS_WARNING) << "Protobuf message length is too large.";
return false;
}
// Read the next protobuf event to a temporary char buffer.
stream.read(tmp_buffer.data(), message_length);
if (stream.gcount() != static_cast<int>(message_length)) {
LOG(LS_WARNING) << "Failed to read protobuf message from file.";
return false;
}
// Parse the protobuf event from the buffer.
rtclog::Event event;
if (!event.ParseFromArray(tmp_buffer.data(), message_length)) {
LOG(LS_WARNING) << "Failed to parse protobuf message.";
return false;
}
EventType type = GetRuntimeEventType(event.type());
switch (type) {
case VIDEO_RECEIVER_CONFIG_EVENT: {
rtclog::StreamConfig config = GetVideoReceiveConfig(event);
streams_.emplace_back(config.remote_ssrc, MediaType::VIDEO,
kIncomingPacket,
RtpHeaderExtensionMap(config.rtp_extensions));
streams_.emplace_back(config.local_ssrc, MediaType::VIDEO,
kOutgoingPacket,
RtpHeaderExtensionMap(config.rtp_extensions));
break;
}
case VIDEO_SENDER_CONFIG_EVENT: {
std::vector<rtclog::StreamConfig> configs = GetVideoSendConfig(event);
for (size_t i = 0; i < configs.size(); i++) {
streams_.emplace_back(
configs[i].local_ssrc, MediaType::VIDEO, kOutgoingPacket,
RtpHeaderExtensionMap(configs[i].rtp_extensions));
streams_.emplace_back(
configs[i].rtx_ssrc, MediaType::VIDEO, kOutgoingPacket,
RtpHeaderExtensionMap(configs[i].rtp_extensions));
}
break;
}
case AUDIO_RECEIVER_CONFIG_EVENT: {
rtclog::StreamConfig config = GetAudioReceiveConfig(event);
streams_.emplace_back(config.remote_ssrc, MediaType::AUDIO,
kIncomingPacket,
RtpHeaderExtensionMap(config.rtp_extensions));
streams_.emplace_back(config.local_ssrc, MediaType::AUDIO,
kOutgoingPacket,
RtpHeaderExtensionMap(config.rtp_extensions));
break;
}
case AUDIO_SENDER_CONFIG_EVENT: {
rtclog::StreamConfig config = GetAudioSendConfig(event);
streams_.emplace_back(config.local_ssrc, MediaType::AUDIO,
kOutgoingPacket,
RtpHeaderExtensionMap(config.rtp_extensions));
break;
}
default:
break;
}
events_.push_back(event);
}
}
size_t ParsedRtcEventLog::GetNumberOfEvents() const {
return events_.size();
}
int64_t ParsedRtcEventLog::GetTimestamp(size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
RTC_CHECK(event.has_timestamp_us());
return event.timestamp_us();
}
ParsedRtcEventLog::EventType ParsedRtcEventLog::GetEventType(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
RTC_CHECK(event.has_type());
return GetRuntimeEventType(event.type());
}
// The header must have space for at least IP_PACKET_SIZE bytes.
webrtc::RtpHeaderExtensionMap* ParsedRtcEventLog::GetRtpHeader(
size_t index,
PacketDirection* incoming,
uint8_t* header,
size_t* header_length,
size_t* total_length) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
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();
}
// 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);
StreamId stream_id(
ssrc, rtp_packet.incoming() ? kIncomingPacket : kOutgoingPacket);
auto it = rtp_extensions_maps_.find(stream_id);
if (it != rtp_extensions_maps_.end()) {
return it->second;
}
}
return nullptr;
}
// The packet must have space for at least IP_PACKET_SIZE bytes.
void ParsedRtcEventLog::GetRtcpPacket(size_t index,
PacketDirection* incoming,
uint8_t* packet,
size_t* length) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
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(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
return GetVideoReceiveConfig(events_[index]);
}
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()) {
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;
}
std::vector<rtclog::StreamConfig> ParsedRtcEventLog::GetVideoSendConfig(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
return GetVideoSendConfig(events_[index]);
}
std::vector<rtclog::StreamConfig> ParsedRtcEventLog::GetVideoSendConfig(
const rtclog::Event& event) const {
std::vector<rtclog::StreamConfig> configs;
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();
if (sender_config.rtx_ssrcs_size() > 0 &&
sender_config.ssrcs_size() != sender_config.rtx_ssrcs_size()) {
LOG(WARNING) << "VideoSendConfig is configured for RTX but the number of "
"SSRCs doesn't match the number of RTX SSRCs.";
}
configs.resize(sender_config.ssrcs_size());
for (int i = 0; i < sender_config.ssrcs_size(); i++) {
// Get SSRCs.
configs[i].local_ssrc = sender_config.ssrcs(i);
if (sender_config.rtx_ssrcs_size() > 0 &&
i < sender_config.rtx_ssrcs_size()) {
RTC_CHECK(sender_config.has_rtx_payload_type());
configs[i].rtx_ssrc = sender_config.rtx_ssrcs(i);
}
// Get header extensions.
GetHeaderExtensions(&configs[i].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());
configs[i].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 configs;
}
rtclog::StreamConfig ParsedRtcEventLog::GetAudioReceiveConfig(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
return GetAudioReceiveConfig(events_[index]);
}
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(size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
return GetAudioSendConfig(events_[index]);
}
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;
}
void ParsedRtcEventLog::GetAudioPlayout(size_t index, uint32_t* ssrc) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
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& loss_event = event.audio_playout_event();
RTC_CHECK(loss_event.has_local_ssrc());
if (ssrc != nullptr) {
*ssrc = loss_event.local_ssrc();
}
}
void ParsedRtcEventLog::GetLossBasedBweUpdate(size_t index,
int32_t* bitrate_bps,
uint8_t* fraction_loss,
int32_t* total_packets) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
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();
RTC_CHECK(loss_event.has_bitrate_bps());
if (bitrate_bps != nullptr) {
*bitrate_bps = loss_event.bitrate_bps();
}
RTC_CHECK(loss_event.has_fraction_loss());
if (fraction_loss != nullptr) {
*fraction_loss = loss_event.fraction_loss();
}
RTC_CHECK(loss_event.has_total_packets());
if (total_packets != nullptr) {
*total_packets = loss_event.total_packets();
}
}
ParsedRtcEventLog::BweDelayBasedUpdate
ParsedRtcEventLog::GetDelayBasedBweUpdate(size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
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();
BweDelayBasedUpdate res;
res.timestamp = GetTimestamp(index);
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;
}
void ParsedRtcEventLog::GetAudioNetworkAdaptation(
size_t index,
AudioEncoderRuntimeConfig* config) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
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();
if (ana_event.has_bitrate_bps())
config->bitrate_bps = rtc::Optional<int>(ana_event.bitrate_bps());
if (ana_event.has_enable_fec())
config->enable_fec = rtc::Optional<bool>(ana_event.enable_fec());
if (ana_event.has_enable_dtx())
config->enable_dtx = rtc::Optional<bool>(ana_event.enable_dtx());
if (ana_event.has_frame_length_ms())
config->frame_length_ms = rtc::Optional<int>(ana_event.frame_length_ms());
if (ana_event.has_num_channels())
config->num_channels = rtc::Optional<size_t>(ana_event.num_channels());
if (ana_event.has_uplink_packet_loss_fraction())
config->uplink_packet_loss_fraction =
rtc::Optional<float>(ana_event.uplink_packet_loss_fraction());
}
ParsedRtcEventLog::BweProbeClusterCreatedEvent
ParsedRtcEventLog::GetBweProbeClusterCreated(size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
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();
BweProbeClusterCreatedEvent res;
res.timestamp = GetTimestamp(index);
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;
}
ParsedRtcEventLog::BweProbeResultEvent ParsedRtcEventLog::GetBweProbeResult(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
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();
BweProbeResultEvent res;
res.timestamp = GetTimestamp(index);
RTC_CHECK(pr_event.has_id());
res.id = pr_event.id();
RTC_CHECK(pr_event.has_result());
if (pr_event.result() == rtclog::BweProbeResult::SUCCESS) {
RTC_CHECK(pr_event.has_bitrate_bps());
res.bitrate_bps = rtc::Optional<uint64_t>(pr_event.bitrate_bps());
} else if (pr_event.result() ==
rtclog::BweProbeResult::INVALID_SEND_RECEIVE_INTERVAL) {
res.failure_reason =
rtc::Optional<ProbeFailureReason>(kInvalidSendReceiveInterval);
} else if (pr_event.result() ==
rtclog::BweProbeResult::INVALID_SEND_RECEIVE_RATIO) {
res.failure_reason =
rtc::Optional<ProbeFailureReason>(kInvalidSendReceiveRatio);
} else if (pr_event.result() == rtclog::BweProbeResult::TIMEOUT) {
res.failure_reason = rtc::Optional<ProbeFailureReason>(kTimeout);
} else {
RTC_NOTREACHED();
}
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 {
for (auto rit = streams_.rbegin(); rit != streams_.rend(); ++rit) {
if (rit->ssrc == ssrc && rit->direction == direction)
return rit->media_type;
}
return MediaType::ANY;
}
} // namespace webrtc