| /* |
| * 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 |