mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
Remove the old RTC event log parser.
The new parser provides the same functionality (with a slightly different API) and is backwards compatible with the legacy wire format. Downstream projects seem to have transitioned to the new parser API. Bug: webrtc:8111 Change-Id: Icb458f0d55e0a4566c4b7b4a53cab48f0b9d6fd8 Reviewed-on: https://webrtc-review.googlesource.com/c/110782 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25635}
This commit is contained in:
Bjorn Terelius
Commit Bot
parent
38578ca9a0
commit
76f9954b17
3 changed files with 0 additions and 1112 deletions
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@ -286,8 +286,6 @@ if (rtc_enable_protobuf) {
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rtc_static_library("rtc_event_log_parser") {
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visibility = [ "*" ]
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sources = [
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"rtc_event_log/rtc_event_log_parser.cc",
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"rtc_event_log/rtc_event_log_parser.h",
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"rtc_event_log/rtc_event_log_parser_new.cc",
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"rtc_event_log/rtc_event_log_parser_new.h",
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"rtc_event_log/rtc_event_processor.h",
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@ -1,851 +0,0 @@
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/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "logging/rtc_event_log/rtc_event_log_parser.h"
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#include <stdint.h>
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#include <string.h>
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#include <algorithm>
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#include <fstream>
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#include <istream>
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#include <map>
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#include <utility>
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#include "logging/rtc_event_log/rtc_event_log.h"
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#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
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#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/protobuf_utils.h"
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namespace webrtc {
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namespace {
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RtcpMode GetRuntimeRtcpMode(rtclog::VideoReceiveConfig::RtcpMode rtcp_mode) {
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switch (rtcp_mode) {
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case rtclog::VideoReceiveConfig::RTCP_COMPOUND:
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return RtcpMode::kCompound;
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case rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE:
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return RtcpMode::kReducedSize;
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}
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RTC_NOTREACHED();
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return RtcpMode::kOff;
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}
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ParsedRtcEventLog::EventType GetRuntimeEventType(
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rtclog::Event::EventType event_type) {
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switch (event_type) {
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case rtclog::Event::UNKNOWN_EVENT:
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return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
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case rtclog::Event::LOG_START:
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return ParsedRtcEventLog::EventType::LOG_START;
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case rtclog::Event::LOG_END:
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return ParsedRtcEventLog::EventType::LOG_END;
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case rtclog::Event::RTP_EVENT:
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return ParsedRtcEventLog::EventType::RTP_EVENT;
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case rtclog::Event::RTCP_EVENT:
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return ParsedRtcEventLog::EventType::RTCP_EVENT;
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case rtclog::Event::AUDIO_PLAYOUT_EVENT:
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return ParsedRtcEventLog::EventType::AUDIO_PLAYOUT_EVENT;
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case rtclog::Event::LOSS_BASED_BWE_UPDATE:
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return ParsedRtcEventLog::EventType::LOSS_BASED_BWE_UPDATE;
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case rtclog::Event::DELAY_BASED_BWE_UPDATE:
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return ParsedRtcEventLog::EventType::DELAY_BASED_BWE_UPDATE;
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case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT:
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return ParsedRtcEventLog::EventType::VIDEO_RECEIVER_CONFIG_EVENT;
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case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT:
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return ParsedRtcEventLog::EventType::VIDEO_SENDER_CONFIG_EVENT;
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case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT:
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return ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT;
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case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
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return ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT;
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case rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT:
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return ParsedRtcEventLog::EventType::AUDIO_NETWORK_ADAPTATION_EVENT;
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case rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT:
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return ParsedRtcEventLog::EventType::BWE_PROBE_CLUSTER_CREATED_EVENT;
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case rtclog::Event::BWE_PROBE_RESULT_EVENT:
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return ParsedRtcEventLog::EventType::BWE_PROBE_RESULT_EVENT;
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case rtclog::Event::ALR_STATE_EVENT:
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return ParsedRtcEventLog::EventType::ALR_STATE_EVENT;
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case rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG:
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return ParsedRtcEventLog::EventType::ICE_CANDIDATE_PAIR_CONFIG;
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case rtclog::Event::ICE_CANDIDATE_PAIR_EVENT:
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return ParsedRtcEventLog::EventType::ICE_CANDIDATE_PAIR_EVENT;
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}
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return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
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}
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BandwidthUsage GetRuntimeDetectorState(
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rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
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switch (detector_state) {
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case rtclog::DelayBasedBweUpdate::BWE_NORMAL:
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return BandwidthUsage::kBwNormal;
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case rtclog::DelayBasedBweUpdate::BWE_UNDERUSING:
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return BandwidthUsage::kBwUnderusing;
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case rtclog::DelayBasedBweUpdate::BWE_OVERUSING:
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return BandwidthUsage::kBwOverusing;
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}
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RTC_NOTREACHED();
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return BandwidthUsage::kBwNormal;
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}
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IceCandidatePairConfigType GetRuntimeIceCandidatePairConfigType(
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rtclog::IceCandidatePairConfig::IceCandidatePairConfigType type) {
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switch (type) {
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case rtclog::IceCandidatePairConfig::ADDED:
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return IceCandidatePairConfigType::kAdded;
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case rtclog::IceCandidatePairConfig::UPDATED:
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return IceCandidatePairConfigType::kUpdated;
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case rtclog::IceCandidatePairConfig::DESTROYED:
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return IceCandidatePairConfigType::kDestroyed;
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case rtclog::IceCandidatePairConfig::SELECTED:
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return IceCandidatePairConfigType::kSelected;
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}
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RTC_NOTREACHED();
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return IceCandidatePairConfigType::kAdded;
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}
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IceCandidateType GetRuntimeIceCandidateType(
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rtclog::IceCandidatePairConfig::IceCandidateType type) {
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switch (type) {
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case rtclog::IceCandidatePairConfig::LOCAL:
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return IceCandidateType::kLocal;
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case rtclog::IceCandidatePairConfig::STUN:
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return IceCandidateType::kStun;
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case rtclog::IceCandidatePairConfig::PRFLX:
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return IceCandidateType::kPrflx;
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case rtclog::IceCandidatePairConfig::RELAY:
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return IceCandidateType::kRelay;
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case rtclog::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE:
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return IceCandidateType::kUnknown;
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}
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RTC_NOTREACHED();
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return IceCandidateType::kUnknown;
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}
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IceCandidatePairProtocol GetRuntimeIceCandidatePairProtocol(
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rtclog::IceCandidatePairConfig::Protocol protocol) {
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switch (protocol) {
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case rtclog::IceCandidatePairConfig::UDP:
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return IceCandidatePairProtocol::kUdp;
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case rtclog::IceCandidatePairConfig::TCP:
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return IceCandidatePairProtocol::kTcp;
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case rtclog::IceCandidatePairConfig::SSLTCP:
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return IceCandidatePairProtocol::kSsltcp;
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case rtclog::IceCandidatePairConfig::TLS:
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return IceCandidatePairProtocol::kTls;
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case rtclog::IceCandidatePairConfig::UNKNOWN_PROTOCOL:
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return IceCandidatePairProtocol::kUnknown;
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}
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RTC_NOTREACHED();
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return IceCandidatePairProtocol::kUnknown;
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}
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IceCandidatePairAddressFamily GetRuntimeIceCandidatePairAddressFamily(
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rtclog::IceCandidatePairConfig::AddressFamily address_family) {
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switch (address_family) {
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case rtclog::IceCandidatePairConfig::IPV4:
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return IceCandidatePairAddressFamily::kIpv4;
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case rtclog::IceCandidatePairConfig::IPV6:
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return IceCandidatePairAddressFamily::kIpv6;
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case rtclog::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY:
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return IceCandidatePairAddressFamily::kUnknown;
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}
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RTC_NOTREACHED();
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return IceCandidatePairAddressFamily::kUnknown;
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}
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IceCandidateNetworkType GetRuntimeIceCandidateNetworkType(
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rtclog::IceCandidatePairConfig::NetworkType network_type) {
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switch (network_type) {
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case rtclog::IceCandidatePairConfig::ETHERNET:
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return IceCandidateNetworkType::kEthernet;
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case rtclog::IceCandidatePairConfig::LOOPBACK:
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return IceCandidateNetworkType::kLoopback;
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case rtclog::IceCandidatePairConfig::WIFI:
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return IceCandidateNetworkType::kWifi;
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case rtclog::IceCandidatePairConfig::VPN:
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return IceCandidateNetworkType::kVpn;
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case rtclog::IceCandidatePairConfig::CELLULAR:
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return IceCandidateNetworkType::kCellular;
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case rtclog::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE:
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return IceCandidateNetworkType::kUnknown;
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}
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RTC_NOTREACHED();
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return IceCandidateNetworkType::kUnknown;
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}
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IceCandidatePairEventType GetRuntimeIceCandidatePairEventType(
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rtclog::IceCandidatePairEvent::IceCandidatePairEventType type) {
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switch (type) {
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case rtclog::IceCandidatePairEvent::CHECK_SENT:
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return IceCandidatePairEventType::kCheckSent;
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case rtclog::IceCandidatePairEvent::CHECK_RECEIVED:
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return IceCandidatePairEventType::kCheckReceived;
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case rtclog::IceCandidatePairEvent::CHECK_RESPONSE_SENT:
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return IceCandidatePairEventType::kCheckResponseSent;
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case rtclog::IceCandidatePairEvent::CHECK_RESPONSE_RECEIVED:
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return IceCandidatePairEventType::kCheckResponseReceived;
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}
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RTC_NOTREACHED();
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return IceCandidatePairEventType::kCheckSent;
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}
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std::pair<uint64_t, bool> ParseVarInt(std::istream& stream) {
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uint64_t varint = 0;
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for (size_t bytes_read = 0; bytes_read < 10; ++bytes_read) {
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// The most significant bit of each byte is 0 if it is the last byte in
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// the varint and 1 otherwise. Thus, we take the 7 least significant bits
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// of each byte and shift them 7 bits for each byte read previously to get
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// the (unsigned) integer.
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int byte = stream.get();
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if (stream.eof()) {
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return std::make_pair(varint, false);
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}
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RTC_DCHECK_GE(byte, 0);
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RTC_DCHECK_LE(byte, 255);
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varint |= static_cast<uint64_t>(byte & 0x7F) << (7 * bytes_read);
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if ((byte & 0x80) == 0) {
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return std::make_pair(varint, true);
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}
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}
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return std::make_pair(varint, false);
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}
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void GetHeaderExtensions(std::vector<RtpExtension>* header_extensions,
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const RepeatedPtrField<rtclog::RtpHeaderExtension>&
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proto_header_extensions) {
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header_extensions->clear();
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for (auto& p : proto_header_extensions) {
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RTC_CHECK(p.has_name());
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RTC_CHECK(p.has_id());
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const std::string& name = p.name();
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int id = p.id();
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header_extensions->push_back(RtpExtension(name, id));
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}
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}
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} // namespace
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ParsedRtcEventLog::ParsedRtcEventLog() = default;
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ParsedRtcEventLog::~ParsedRtcEventLog() = default;
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ParsedRtcEventLog::BweProbeResultEvent::BweProbeResultEvent() = default;
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ParsedRtcEventLog::BweProbeResultEvent::BweProbeResultEvent(
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const BweProbeResultEvent&) = default;
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bool ParsedRtcEventLog::ParseFile(const std::string& filename) {
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std::ifstream file(filename, std::ios_base::in | std::ios_base::binary);
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if (!file.good() || !file.is_open()) {
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RTC_LOG(LS_WARNING) << "Could not open file for reading.";
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return false;
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}
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return ParseStream(file);
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}
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bool ParsedRtcEventLog::ParseString(const std::string& s) {
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std::istringstream stream(s, std::ios_base::in | std::ios_base::binary);
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return ParseStream(stream);
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}
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bool ParsedRtcEventLog::ParseStream(std::istream& stream) {
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events_.clear();
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const size_t kMaxEventSize = (1u << 16) - 1;
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std::vector<char> tmp_buffer(kMaxEventSize);
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uint64_t tag;
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uint64_t message_length;
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bool success;
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RTC_DCHECK(stream.good());
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while (1) {
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// Check whether we have reached end of file.
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stream.peek();
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if (stream.eof()) {
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// Process all extensions maps for faster look-up later.
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for (auto& event_stream : streams_) {
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rtp_extensions_maps_[StreamId(event_stream.ssrc,
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event_stream.direction)] =
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&event_stream.rtp_extensions_map;
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}
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return true;
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}
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// Read the next message tag. The tag number is defined as
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// (fieldnumber << 3) | wire_type. In our case, the field number is
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// supposed to be 1 and the wire type for an
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// length-delimited field is 2.
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const uint64_t kExpectedTag = (1 << 3) | 2;
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std::tie(tag, success) = ParseVarInt(stream);
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if (!success) {
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RTC_LOG(LS_WARNING)
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<< "Missing field tag from beginning of protobuf event.";
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return false;
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} else if (tag != kExpectedTag) {
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RTC_LOG(LS_WARNING)
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<< "Unexpected field tag at beginning of protobuf event.";
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return false;
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}
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// Read the length field.
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std::tie(message_length, success) = ParseVarInt(stream);
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if (!success) {
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RTC_LOG(LS_WARNING) << "Missing message length after protobuf field tag.";
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return false;
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} else if (message_length > kMaxEventSize) {
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RTC_LOG(LS_WARNING) << "Protobuf message length is too large.";
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return false;
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}
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// Read the next protobuf event to a temporary char buffer.
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stream.read(tmp_buffer.data(), message_length);
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if (stream.gcount() != static_cast<int>(message_length)) {
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RTC_LOG(LS_WARNING) << "Failed to read protobuf message from file.";
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return false;
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}
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// Parse the protobuf event from the buffer.
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rtclog::Event event;
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if (!event.ParseFromArray(tmp_buffer.data(), message_length)) {
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RTC_LOG(LS_WARNING) << "Failed to parse protobuf message.";
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return false;
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}
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EventType type = GetRuntimeEventType(event.type());
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switch (type) {
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case VIDEO_RECEIVER_CONFIG_EVENT: {
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rtclog::StreamConfig config = GetVideoReceiveConfig(event);
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streams_.emplace_back(config.remote_ssrc, MediaType::VIDEO,
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kIncomingPacket,
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RtpHeaderExtensionMap(config.rtp_extensions));
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streams_.emplace_back(config.local_ssrc, MediaType::VIDEO,
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kOutgoingPacket,
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RtpHeaderExtensionMap(config.rtp_extensions));
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break;
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}
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case VIDEO_SENDER_CONFIG_EVENT: {
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std::vector<rtclog::StreamConfig> configs = GetVideoSendConfig(event);
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for (size_t i = 0; i < configs.size(); i++) {
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streams_.emplace_back(
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configs[i].local_ssrc, MediaType::VIDEO, kOutgoingPacket,
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RtpHeaderExtensionMap(configs[i].rtp_extensions));
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streams_.emplace_back(
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configs[i].rtx_ssrc, MediaType::VIDEO, kOutgoingPacket,
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RtpHeaderExtensionMap(configs[i].rtp_extensions));
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}
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break;
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}
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case AUDIO_RECEIVER_CONFIG_EVENT: {
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rtclog::StreamConfig config = GetAudioReceiveConfig(event);
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streams_.emplace_back(config.remote_ssrc, MediaType::AUDIO,
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kIncomingPacket,
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RtpHeaderExtensionMap(config.rtp_extensions));
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streams_.emplace_back(config.local_ssrc, MediaType::AUDIO,
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kOutgoingPacket,
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RtpHeaderExtensionMap(config.rtp_extensions));
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break;
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}
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case AUDIO_SENDER_CONFIG_EVENT: {
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rtclog::StreamConfig config = GetAudioSendConfig(event);
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streams_.emplace_back(config.local_ssrc, MediaType::AUDIO,
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kOutgoingPacket,
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RtpHeaderExtensionMap(config.rtp_extensions));
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break;
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}
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default:
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break;
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}
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events_.push_back(event);
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}
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}
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size_t ParsedRtcEventLog::GetNumberOfEvents() const {
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return events_.size();
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}
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int64_t ParsedRtcEventLog::GetTimestamp(size_t index) const {
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RTC_CHECK_LT(index, GetNumberOfEvents());
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const rtclog::Event& event = events_[index];
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RTC_CHECK(event.has_timestamp_us());
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return event.timestamp_us();
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}
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ParsedRtcEventLog::EventType ParsedRtcEventLog::GetEventType(
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size_t index) const {
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RTC_CHECK_LT(index, GetNumberOfEvents());
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const rtclog::Event& event = events_[index];
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RTC_CHECK(event.has_type());
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return GetRuntimeEventType(event.type());
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}
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// The header must have space for at least IP_PACKET_SIZE bytes.
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webrtc::RtpHeaderExtensionMap* ParsedRtcEventLog::GetRtpHeader(
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size_t index,
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PacketDirection* incoming,
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uint8_t* header,
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size_t* header_length,
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size_t* total_length,
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int* probe_cluster_id) 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;
|
||||
}
|
||||
}
|
||||
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;
|
||||
}
|
||||
}
|
||||
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()) {
|
||||
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;
|
||||
}
|
||||
|
||||
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()) {
|
||||
RTC_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 = ana_event.bitrate_bps();
|
||||
if (ana_event.has_enable_fec())
|
||||
config->enable_fec = ana_event.enable_fec();
|
||||
if (ana_event.has_enable_dtx())
|
||||
config->enable_dtx = ana_event.enable_dtx();
|
||||
if (ana_event.has_frame_length_ms())
|
||||
config->frame_length_ms = ana_event.frame_length_ms();
|
||||
if (ana_event.has_num_channels())
|
||||
config->num_channels = ana_event.num_channels();
|
||||
if (ana_event.has_uplink_packet_loss_fraction())
|
||||
config->uplink_packet_loss_fraction =
|
||||
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 = pr_event.bitrate_bps();
|
||||
} else 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();
|
||||
}
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
ParsedRtcEventLog::AlrStateEvent ParsedRtcEventLog::GetAlrState(
|
||||
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::ALR_STATE_EVENT);
|
||||
RTC_CHECK(event.has_alr_state());
|
||||
const rtclog::AlrState& alr_event = event.alr_state();
|
||||
AlrStateEvent res;
|
||||
res.timestamp = GetTimestamp(index);
|
||||
RTC_CHECK(alr_event.has_in_alr());
|
||||
res.in_alr = alr_event.in_alr();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
ParsedRtcEventLog::IceCandidatePairConfig
|
||||
ParsedRtcEventLog::GetIceCandidatePairConfig(size_t index) const {
|
||||
RTC_CHECK_LT(index, GetNumberOfEvents());
|
||||
const rtclog::Event& rtc_event = events_[index];
|
||||
RTC_CHECK(rtc_event.has_type());
|
||||
RTC_CHECK_EQ(rtc_event.type(), rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG);
|
||||
IceCandidatePairConfig res;
|
||||
const rtclog::IceCandidatePairConfig& config =
|
||||
rtc_event.ice_candidate_pair_config();
|
||||
res.timestamp = GetTimestamp(index);
|
||||
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;
|
||||
}
|
||||
|
||||
ParsedRtcEventLog::IceCandidatePairEvent
|
||||
ParsedRtcEventLog::GetIceCandidatePairEvent(size_t index) const {
|
||||
RTC_CHECK_LT(index, GetNumberOfEvents());
|
||||
const rtclog::Event& rtc_event = events_[index];
|
||||
RTC_CHECK(rtc_event.has_type());
|
||||
RTC_CHECK_EQ(rtc_event.type(), rtclog::Event::ICE_CANDIDATE_PAIR_EVENT);
|
||||
IceCandidatePairEvent res;
|
||||
const rtclog::IceCandidatePairEvent& event =
|
||||
rtc_event.ice_candidate_pair_event();
|
||||
res.timestamp = GetTimestamp(index);
|
||||
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();
|
||||
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
|
||||
|
|
@ -1,259 +0,0 @@
|
|||
/*
|
||||
* 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.
|
||||
*/
|
||||
#ifndef LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
|
||||
#define LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
|
||||
|
||||
#include <map>
|
||||
#include <sstream> // no-presubmit-check TODO(webrtc:8982)
|
||||
#include <string>
|
||||
#include <utility> // pair
|
||||
#include <vector>
|
||||
|
||||
#include "call/video_receive_stream.h"
|
||||
#include "call/video_send_stream.h"
|
||||
#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
|
||||
#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
|
||||
#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
|
||||
#include "logging/rtc_event_log/rtc_event_log.h"
|
||||
#include "logging/rtc_event_log/rtc_stream_config.h"
|
||||
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
|
||||
#include "modules/rtp_rtcp/source/byte_io.h"
|
||||
#include "rtc_base/ignore_wundef.h"
|
||||
|
||||
// Files generated at build-time by the protobuf compiler.
|
||||
RTC_PUSH_IGNORING_WUNDEF()
|
||||
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
|
||||
#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
|
||||
#else
|
||||
#include "logging/rtc_event_log/rtc_event_log.pb.h"
|
||||
#endif
|
||||
RTC_POP_IGNORING_WUNDEF()
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
enum class BandwidthUsage;
|
||||
enum class MediaType;
|
||||
|
||||
struct AudioEncoderRuntimeConfig;
|
||||
|
||||
class ParsedRtcEventLog {
|
||||
friend class RtcEventLogTestHelper;
|
||||
|
||||
public:
|
||||
ParsedRtcEventLog();
|
||||
~ParsedRtcEventLog();
|
||||
|
||||
struct BweProbeClusterCreatedEvent {
|
||||
uint64_t timestamp;
|
||||
uint32_t id;
|
||||
uint64_t bitrate_bps;
|
||||
uint32_t min_packets;
|
||||
uint32_t min_bytes;
|
||||
};
|
||||
|
||||
struct BweProbeResultEvent {
|
||||
BweProbeResultEvent();
|
||||
BweProbeResultEvent(const BweProbeResultEvent&);
|
||||
|
||||
uint64_t timestamp;
|
||||
uint32_t id;
|
||||
absl::optional<uint64_t> bitrate_bps;
|
||||
absl::optional<ProbeFailureReason> failure_reason;
|
||||
};
|
||||
|
||||
struct BweDelayBasedUpdate {
|
||||
uint64_t timestamp;
|
||||
int32_t bitrate_bps;
|
||||
BandwidthUsage detector_state;
|
||||
};
|
||||
|
||||
struct AlrStateEvent {
|
||||
uint64_t timestamp;
|
||||
bool in_alr;
|
||||
};
|
||||
|
||||
struct IceCandidatePairConfig {
|
||||
uint64_t timestamp;
|
||||
IceCandidatePairConfigType type;
|
||||
uint32_t candidate_pair_id;
|
||||
IceCandidateType local_candidate_type;
|
||||
IceCandidatePairProtocol local_relay_protocol;
|
||||
IceCandidateNetworkType local_network_type;
|
||||
IceCandidatePairAddressFamily local_address_family;
|
||||
IceCandidateType remote_candidate_type;
|
||||
IceCandidatePairAddressFamily remote_address_family;
|
||||
IceCandidatePairProtocol candidate_pair_protocol;
|
||||
};
|
||||
|
||||
struct IceCandidatePairEvent {
|
||||
uint64_t timestamp;
|
||||
IceCandidatePairEventType type;
|
||||
uint32_t candidate_pair_id;
|
||||
};
|
||||
|
||||
enum EventType {
|
||||
UNKNOWN_EVENT = 0,
|
||||
LOG_START = 1,
|
||||
LOG_END = 2,
|
||||
RTP_EVENT = 3,
|
||||
RTCP_EVENT = 4,
|
||||
AUDIO_PLAYOUT_EVENT = 5,
|
||||
LOSS_BASED_BWE_UPDATE = 6,
|
||||
DELAY_BASED_BWE_UPDATE = 7,
|
||||
VIDEO_RECEIVER_CONFIG_EVENT = 8,
|
||||
VIDEO_SENDER_CONFIG_EVENT = 9,
|
||||
AUDIO_RECEIVER_CONFIG_EVENT = 10,
|
||||
AUDIO_SENDER_CONFIG_EVENT = 11,
|
||||
AUDIO_NETWORK_ADAPTATION_EVENT = 16,
|
||||
BWE_PROBE_CLUSTER_CREATED_EVENT = 17,
|
||||
BWE_PROBE_RESULT_EVENT = 18,
|
||||
ALR_STATE_EVENT = 19,
|
||||
ICE_CANDIDATE_PAIR_CONFIG = 20,
|
||||
ICE_CANDIDATE_PAIR_EVENT = 21,
|
||||
};
|
||||
|
||||
enum class MediaType { ANY, AUDIO, VIDEO, DATA };
|
||||
|
||||
// Reads an RtcEventLog file and returns true if parsing was successful.
|
||||
bool ParseFile(const std::string& file_name);
|
||||
|
||||
// Reads an RtcEventLog from a string and returns true if successful.
|
||||
bool ParseString(const std::string& s);
|
||||
|
||||
// Reads an RtcEventLog from an istream and returns true if successful.
|
||||
bool ParseStream(std::istream& stream);
|
||||
|
||||
// Returns the number of events in an EventStream.
|
||||
size_t GetNumberOfEvents() const;
|
||||
|
||||
// Reads the arrival timestamp (in microseconds) from a rtclog::Event.
|
||||
int64_t GetTimestamp(size_t index) const;
|
||||
|
||||
// Reads the event type of the rtclog::Event at |index|.
|
||||
EventType GetEventType(size_t index) const;
|
||||
|
||||
// Reads the header, direction, header length and packet length from the RTP
|
||||
// event at |index|, and stores the values in the corresponding output
|
||||
// parameters. Each output parameter can be set to nullptr if that value
|
||||
// isn't needed.
|
||||
// NB: The header must have space for at least IP_PACKET_SIZE bytes.
|
||||
// Returns: a pointer to a header extensions map acquired from parsing
|
||||
// corresponding Audio/Video Sender/Receiver config events.
|
||||
// Warning: if the same SSRC is reused by both video and audio streams during
|
||||
// call, extensions maps may be incorrect (the last one would be returned).
|
||||
webrtc::RtpHeaderExtensionMap* GetRtpHeader(size_t index,
|
||||
PacketDirection* incoming,
|
||||
uint8_t* header,
|
||||
size_t* header_length,
|
||||
size_t* total_length,
|
||||
int* probe_cluster_id) const;
|
||||
|
||||
// Reads packet, direction and packet length from the RTCP event at |index|,
|
||||
// and stores the values in the corresponding output parameters.
|
||||
// Each output parameter can be set to nullptr if that value isn't needed.
|
||||
// NB: The packet must have space for at least IP_PACKET_SIZE bytes.
|
||||
void GetRtcpPacket(size_t index,
|
||||
PacketDirection* incoming,
|
||||
uint8_t* packet,
|
||||
size_t* length) const;
|
||||
|
||||
// Reads a video receive config event to a StreamConfig struct.
|
||||
// Only the fields that are stored in the protobuf will be written.
|
||||
rtclog::StreamConfig GetVideoReceiveConfig(size_t index) const;
|
||||
|
||||
// Reads a video send config event to a StreamConfig struct. If the proto
|
||||
// contains multiple SSRCs and RTX SSRCs (this used to be the case for
|
||||
// simulcast streams) then we return one StreamConfig per SSRC,RTX_SSRC pair.
|
||||
// Only the fields that are stored in the protobuf will be written.
|
||||
std::vector<rtclog::StreamConfig> GetVideoSendConfig(size_t index) const;
|
||||
|
||||
// Reads a audio receive config event to a StreamConfig struct.
|
||||
// Only the fields that are stored in the protobuf will be written.
|
||||
rtclog::StreamConfig GetAudioReceiveConfig(size_t index) const;
|
||||
|
||||
// Reads a config event to a StreamConfig struct.
|
||||
// Only the fields that are stored in the protobuf will be written.
|
||||
rtclog::StreamConfig GetAudioSendConfig(size_t index) const;
|
||||
|
||||
// Reads the SSRC from the audio playout event at |index|. The SSRC is stored
|
||||
// in the output parameter ssrc. The output parameter can be set to nullptr
|
||||
// and in that case the function only asserts that the event is well formed.
|
||||
void GetAudioPlayout(size_t index, uint32_t* ssrc) const;
|
||||
|
||||
// Reads bitrate, fraction loss (as defined in RFC 1889) and total number of
|
||||
// expected packets from the loss based BWE event at |index| and stores the
|
||||
// values in
|
||||
// the corresponding output parameters. Each output parameter can be set to
|
||||
// nullptr if that
|
||||
// value isn't needed.
|
||||
void GetLossBasedBweUpdate(size_t index,
|
||||
int32_t* bitrate_bps,
|
||||
uint8_t* fraction_loss,
|
||||
int32_t* total_packets) const;
|
||||
|
||||
// Reads bitrate and detector_state from the delay based BWE event at |index|
|
||||
// and stores the values in the corresponding output parameters. Each output
|
||||
// parameter can be set to nullptr if that
|
||||
// value isn't needed.
|
||||
BweDelayBasedUpdate GetDelayBasedBweUpdate(size_t index) const;
|
||||
|
||||
// Reads a audio network adaptation event to a (non-NULL)
|
||||
// AudioEncoderRuntimeConfig struct. Only the fields that are
|
||||
// stored in the protobuf will be written.
|
||||
void GetAudioNetworkAdaptation(size_t index,
|
||||
AudioEncoderRuntimeConfig* config) const;
|
||||
|
||||
BweProbeClusterCreatedEvent GetBweProbeClusterCreated(size_t index) const;
|
||||
|
||||
BweProbeResultEvent GetBweProbeResult(size_t index) const;
|
||||
|
||||
MediaType GetMediaType(uint32_t ssrc, PacketDirection direction) const;
|
||||
|
||||
AlrStateEvent GetAlrState(size_t index) const;
|
||||
|
||||
IceCandidatePairConfig GetIceCandidatePairConfig(size_t index) const;
|
||||
IceCandidatePairEvent GetIceCandidatePairEvent(size_t index) const;
|
||||
|
||||
private:
|
||||
rtclog::StreamConfig GetVideoReceiveConfig(const rtclog::Event& event) const;
|
||||
std::vector<rtclog::StreamConfig> GetVideoSendConfig(
|
||||
const rtclog::Event& event) const;
|
||||
rtclog::StreamConfig GetAudioReceiveConfig(const rtclog::Event& event) const;
|
||||
rtclog::StreamConfig GetAudioSendConfig(const rtclog::Event& event) const;
|
||||
|
||||
std::vector<rtclog::Event> events_;
|
||||
|
||||
struct Stream {
|
||||
Stream(uint32_t ssrc,
|
||||
MediaType media_type,
|
||||
webrtc::PacketDirection direction,
|
||||
webrtc::RtpHeaderExtensionMap map)
|
||||
: ssrc(ssrc),
|
||||
media_type(media_type),
|
||||
direction(direction),
|
||||
rtp_extensions_map(map) {}
|
||||
uint32_t ssrc;
|
||||
MediaType media_type;
|
||||
webrtc::PacketDirection direction;
|
||||
webrtc::RtpHeaderExtensionMap rtp_extensions_map;
|
||||
};
|
||||
|
||||
// All configured streams found in the event log.
|
||||
std::vector<Stream> streams_;
|
||||
|
||||
// To find configured extensions map for given stream, what are needed to
|
||||
// parse a header.
|
||||
typedef std::pair<uint32_t, webrtc::PacketDirection> StreamId;
|
||||
std::map<StreamId, webrtc::RtpHeaderExtensionMap*> rtp_extensions_maps_;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
|
||||
Loading…
Reference in a new issue