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Bug: webrtc:9147 Change-Id: I61ec7bc5299201e25e1efc503b73b84d5be3ebbf Reviewed-on: https://webrtc-review.googlesource.com/71740 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23151}
921 lines
32 KiB
C++
921 lines
32 KiB
C++
/*
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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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#ifndef LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_NEW_H_
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#define LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_NEW_H_
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#include <iterator>
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#include <map>
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#include <set>
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#include <string>
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#include <utility> // pair
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#include <vector>
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#include "call/video_receive_stream.h"
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#include "call/video_send_stream.h"
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#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
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#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
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#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
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#include "logging/rtc_event_log/rtc_event_log.h"
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#include "logging/rtc_event_log/rtc_stream_config.h"
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#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
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#include "rtc_base/ignore_wundef.h"
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// Files generated at build-time by the protobuf compiler.
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RTC_PUSH_IGNORING_WUNDEF()
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#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
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#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
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#else
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#include "logging/rtc_event_log/rtc_event_log.pb.h"
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#endif
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RTC_POP_IGNORING_WUNDEF()
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namespace webrtc {
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enum class BandwidthUsage;
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struct AudioEncoderRuntimeConfig;
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struct LoggedAlrStateEvent {
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int64_t timestamp_us;
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bool in_alr;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedAudioPlayoutEvent {
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int64_t timestamp_us;
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uint32_t ssrc;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedAudioNetworkAdaptationEvent {
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int64_t timestamp_us;
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AudioEncoderRuntimeConfig config;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedBweDelayBasedUpdate {
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int64_t timestamp_us;
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int32_t bitrate_bps;
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BandwidthUsage detector_state;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedBweLossBasedUpdate {
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int64_t timestamp_us;
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int32_t bitrate_bps;
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uint8_t fraction_lost;
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int32_t expected_packets;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedBweProbeClusterCreatedEvent {
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int64_t timestamp_us;
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int32_t id;
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int32_t bitrate_bps;
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uint32_t min_packets;
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uint32_t min_bytes;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedBweProbeResultEvent {
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int64_t timestamp_us;
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int32_t id;
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rtc::Optional<int32_t> bitrate_bps;
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rtc::Optional<ProbeFailureReason> failure_reason;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedIceCandidatePairConfig {
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int64_t timestamp_us;
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IceCandidatePairEventType type;
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uint32_t candidate_pair_id;
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IceCandidateType local_candidate_type;
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IceCandidatePairProtocol local_relay_protocol;
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IceCandidateNetworkType local_network_type;
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IceCandidatePairAddressFamily local_address_family;
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IceCandidateType remote_candidate_type;
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IceCandidatePairAddressFamily remote_address_family;
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IceCandidatePairProtocol candidate_pair_protocol;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedIceCandidatePairEvent {
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int64_t timestamp_us;
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IceCandidatePairEventType type;
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uint32_t candidate_pair_id;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedRtpPacket {
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LoggedRtpPacket(uint64_t timestamp_us,
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RTPHeader header,
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size_t header_length,
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size_t total_length)
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: timestamp_us(timestamp_us),
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header(header),
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header_length(header_length),
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total_length(total_length) {}
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int64_t timestamp_us;
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// TODO(terelius): This allocates space for 15 CSRCs even if none are used.
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RTPHeader header;
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size_t header_length;
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size_t total_length;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedRtpPacketIncoming {
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LoggedRtpPacketIncoming(uint64_t timestamp_us,
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RTPHeader header,
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size_t header_length,
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size_t total_length)
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: rtp(timestamp_us, header, header_length, total_length) {}
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LoggedRtpPacket rtp;
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int64_t log_time_us() const { return rtp.timestamp_us; }
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int64_t log_time_ms() const { return rtp.timestamp_us / 1000; }
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};
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struct LoggedRtpPacketOutgoing {
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LoggedRtpPacketOutgoing(uint64_t timestamp_us,
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RTPHeader header,
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size_t header_length,
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size_t total_length)
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: rtp(timestamp_us, header, header_length, total_length) {}
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LoggedRtpPacket rtp;
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int64_t log_time_us() const { return rtp.timestamp_us; }
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int64_t log_time_ms() const { return rtp.timestamp_us / 1000; }
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};
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struct LoggedRtcpPacket {
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LoggedRtcpPacket(uint64_t timestamp_us,
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const uint8_t* packet,
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size_t total_length)
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: timestamp_us(timestamp_us), raw_data(packet, packet + total_length) {}
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int64_t timestamp_us;
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std::vector<uint8_t> raw_data;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedRtcpPacketIncoming {
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LoggedRtcpPacketIncoming(uint64_t timestamp_us,
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const uint8_t* packet,
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size_t total_length)
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: rtcp(timestamp_us, packet, total_length) {}
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LoggedRtcpPacket rtcp;
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int64_t log_time_us() const { return rtcp.timestamp_us; }
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int64_t log_time_ms() const { return rtcp.timestamp_us / 1000; }
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};
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struct LoggedRtcpPacketOutgoing {
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LoggedRtcpPacketOutgoing(uint64_t timestamp_us,
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const uint8_t* packet,
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size_t total_length)
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: rtcp(timestamp_us, packet, total_length) {}
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LoggedRtcpPacket rtcp;
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int64_t log_time_us() const { return rtcp.timestamp_us; }
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int64_t log_time_ms() const { return rtcp.timestamp_us / 1000; }
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};
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struct LoggedRtcpPacketReceiverReport {
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int64_t timestamp_us;
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rtcp::ReceiverReport rr;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedRtcpPacketSenderReport {
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int64_t timestamp_us;
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rtcp::SenderReport sr;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedRtcpPacketRemb {
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int64_t timestamp_us;
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rtcp::Remb remb;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedRtcpPacketNack {
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int64_t timestamp_us;
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rtcp::Nack nack;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedRtcpPacketTransportFeedback {
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int64_t timestamp_us;
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rtcp::TransportFeedback transport_feedback;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedStartEvent {
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explicit LoggedStartEvent(uint64_t timestamp_us)
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: timestamp_us(timestamp_us) {}
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int64_t timestamp_us;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedStopEvent {
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explicit LoggedStopEvent(uint64_t timestamp_us)
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: timestamp_us(timestamp_us) {}
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int64_t timestamp_us;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedAudioRecvConfig {
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LoggedAudioRecvConfig(int64_t timestamp_us, const rtclog::StreamConfig config)
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: timestamp_us(timestamp_us), config(config) {}
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int64_t timestamp_us;
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rtclog::StreamConfig config;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedAudioSendConfig {
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LoggedAudioSendConfig(int64_t timestamp_us, const rtclog::StreamConfig config)
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: timestamp_us(timestamp_us), config(config) {}
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int64_t timestamp_us;
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rtclog::StreamConfig config;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedVideoRecvConfig {
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LoggedVideoRecvConfig(int64_t timestamp_us, const rtclog::StreamConfig config)
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: timestamp_us(timestamp_us), config(config) {}
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int64_t timestamp_us;
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rtclog::StreamConfig config;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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struct LoggedVideoSendConfig {
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LoggedVideoSendConfig(int64_t timestamp_us,
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const std::vector<rtclog::StreamConfig> configs)
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: timestamp_us(timestamp_us), configs(configs) {}
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int64_t timestamp_us;
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std::vector<rtclog::StreamConfig> configs;
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int64_t log_time_us() const { return timestamp_us; }
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int64_t log_time_ms() const { return timestamp_us / 1000; }
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};
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template <typename T>
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class PacketView;
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template <typename T>
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class PacketIterator {
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friend class PacketView<T>;
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public:
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// Standard iterator traits.
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using difference_type = std::ptrdiff_t;
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using value_type = T;
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using pointer = T*;
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using reference = T&;
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using iterator_category = std::bidirectional_iterator_tag;
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// The default-contructed iterator is meaningless, but is required by the
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// ForwardIterator concept.
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PacketIterator() : ptr_(nullptr), element_size_(0) {}
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PacketIterator(const PacketIterator& other)
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: ptr_(other.ptr_), element_size_(other.element_size_) {}
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PacketIterator(const PacketIterator&& other)
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: ptr_(other.ptr_), element_size_(other.element_size_) {}
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~PacketIterator() = default;
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PacketIterator& operator=(const PacketIterator& other) {
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ptr_ = other.ptr_;
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element_size_ = other.element_size_;
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return *this;
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}
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PacketIterator& operator=(const PacketIterator&& other) {
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ptr_ = other.ptr_;
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element_size_ = other.element_size_;
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return *this;
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}
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bool operator==(const PacketIterator<T>& other) const {
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RTC_DCHECK_EQ(element_size_, other.element_size_);
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return ptr_ == other.ptr_;
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}
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bool operator!=(const PacketIterator<T>& other) const {
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RTC_DCHECK_EQ(element_size_, other.element_size_);
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return ptr_ != other.ptr_;
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}
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PacketIterator& operator++() {
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ptr_ += element_size_;
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return *this;
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}
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PacketIterator& operator--() {
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ptr_ -= element_size_;
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return *this;
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}
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PacketIterator operator++(int) {
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PacketIterator iter_copy(ptr_, element_size_);
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ptr_ += element_size_;
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return iter_copy;
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}
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PacketIterator operator--(int) {
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PacketIterator iter_copy(ptr_, element_size_);
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ptr_ -= element_size_;
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return iter_copy;
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}
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T& operator*() { return *reinterpret_cast<T*>(ptr_); }
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const T& operator*() const { return *reinterpret_cast<const T*>(ptr_); }
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private:
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PacketIterator(typename std::conditional<std::is_const<T>::value,
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const void*,
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void*>::type p,
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size_t s)
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: ptr_(reinterpret_cast<decltype(ptr_)>(p)), element_size_(s) {}
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typename std::conditional<std::is_const<T>::value, const char*, char*>::type
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ptr_;
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size_t element_size_;
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};
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// Suppose that we have a struct S where we are only interested in a specific
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// member M. Given an array of S, PacketView can be used to treat the array
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// as an array of M, without exposing the type S to surrounding code and without
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// accessing the member through a virtual function. In this case, we want to
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// have a common view for incoming and outgoing RtpPackets, hence the PacketView
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// name.
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// Note that constructing a PacketView bypasses the typesystem, so the caller
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// has to take extra care when constructing these objects. The implementation
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// also requires that the containing struct is standard-layout (e.g. POD).
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//
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// Usage example:
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// struct A {...};
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// struct B { A a; ...};
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// struct C { A a; ...};
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// size_t len = 10;
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// B* array1 = new B[len];
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// C* array2 = new C[len];
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//
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// PacketView<A> view1 = PacketView<A>::Create<B>(array1, len, offsetof(B, a));
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// PacketView<A> view2 = PacketView<A>::Create<C>(array2, len, offsetof(C, a));
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//
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// The following code works with either view1 or view2.
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// void f(PacketView<A> view)
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// for (A& a : view) {
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// DoSomething(a);
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// }
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template <typename T>
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class PacketView {
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public:
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template <typename U>
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static PacketView Create(U* ptr, size_t num_elements, size_t offset) {
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static_assert(std::is_standard_layout<U>::value,
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"PacketView can only be created for standard layout types.");
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static_assert(std::is_standard_layout<T>::value,
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"PacketView can only be created for standard layout types.");
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return PacketView(ptr, num_elements, offset, sizeof(U));
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}
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using iterator = PacketIterator<T>;
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using const_iterator = PacketIterator<const T>;
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using reverse_iterator = std::reverse_iterator<iterator>;
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using const_reverse_iterator = std::reverse_iterator<const_iterator>;
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iterator begin() { return iterator(data_, element_size_); }
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iterator end() {
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auto end_ptr = data_ + num_elements_ * element_size_;
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return iterator(end_ptr, element_size_);
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}
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const_iterator begin() const { return const_iterator(data_, element_size_); }
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const_iterator end() const {
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auto end_ptr = data_ + num_elements_ * element_size_;
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return const_iterator(end_ptr, element_size_);
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}
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reverse_iterator rbegin() { return reverse_iterator(end()); }
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reverse_iterator rend() { return reverse_iterator(begin()); }
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const_reverse_iterator rbegin() const {
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return const_reverse_iterator(end());
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}
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const_reverse_iterator rend() const {
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return const_reverse_iterator(begin());
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}
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size_t size() const { return num_elements_; }
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T& operator[](size_t i) {
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auto elem_ptr = data_ + i * element_size_;
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return *reinterpret_cast<T*>(elem_ptr);
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}
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const T& operator[](size_t i) const {
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auto elem_ptr = data_ + i * element_size_;
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return *reinterpret_cast<const T*>(elem_ptr);
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}
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private:
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PacketView(typename std::conditional<std::is_const<T>::value,
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const void*,
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void*>::type data,
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size_t num_elements,
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size_t offset,
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size_t element_size)
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: data_(reinterpret_cast<decltype(data_)>(data) + offset),
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num_elements_(num_elements),
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element_size_(element_size) {}
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typename std::conditional<std::is_const<T>::value, const char*, char*>::type
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data_;
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size_t num_elements_;
|
|
size_t element_size_;
|
|
};
|
|
|
|
class ParsedRtcEventLogNew {
|
|
friend class RtcEventLogTestHelper;
|
|
|
|
public:
|
|
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 };
|
|
enum class UnconfiguredHeaderExtensions {
|
|
kDontParse,
|
|
kAttemptWebrtcDefaultConfig
|
|
};
|
|
|
|
explicit ParsedRtcEventLogNew(
|
|
UnconfiguredHeaderExtensions parse_unconfigured_header_extensions =
|
|
UnconfiguredHeaderExtensions::kDontParse);
|
|
|
|
// Clears previously parsed events and resets the ParsedRtcEventLogNew to an
|
|
// empty state.
|
|
void Clear();
|
|
|
|
// 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); // no-presubmit-check TODO(webrtc:8982)
|
|
|
|
// 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;
|
|
int64_t GetTimestamp(const rtclog::Event& event) 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).
|
|
const webrtc::RtpHeaderExtensionMap* GetRtpHeader(
|
|
size_t index,
|
|
PacketDirection* incoming,
|
|
uint8_t* header,
|
|
size_t* header_length,
|
|
size_t* total_length,
|
|
int* probe_cluster_id) const;
|
|
const webrtc::RtpHeaderExtensionMap* GetRtpHeader(
|
|
const rtclog::Event& event,
|
|
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;
|
|
void GetRtcpPacket(const rtclog::Event& event,
|
|
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.
|
|
LoggedAudioPlayoutEvent GetAudioPlayout(size_t index) 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.
|
|
LoggedBweLossBasedUpdate GetLossBasedBweUpdate(size_t index) 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.
|
|
LoggedBweDelayBasedUpdate 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.
|
|
LoggedAudioNetworkAdaptationEvent GetAudioNetworkAdaptation(
|
|
size_t index) const;
|
|
|
|
LoggedBweProbeClusterCreatedEvent GetBweProbeClusterCreated(
|
|
size_t index) const;
|
|
|
|
LoggedBweProbeResultEvent GetBweProbeResult(size_t index) const;
|
|
|
|
MediaType GetMediaType(uint32_t ssrc, PacketDirection direction) const;
|
|
|
|
LoggedAlrStateEvent GetAlrState(size_t index) const;
|
|
|
|
LoggedIceCandidatePairConfig GetIceCandidatePairConfig(size_t index) const;
|
|
|
|
LoggedIceCandidatePairEvent GetIceCandidatePairEvent(size_t index) const;
|
|
|
|
const std::set<uint32_t>& incoming_rtx_ssrcs() const {
|
|
return incoming_rtx_ssrcs_;
|
|
}
|
|
const std::set<uint32_t>& incoming_video_ssrcs() const {
|
|
return incoming_video_ssrcs_;
|
|
}
|
|
const std::set<uint32_t>& incoming_audio_ssrcs() const {
|
|
return incoming_audio_ssrcs_;
|
|
}
|
|
const std::set<uint32_t>& outgoing_rtx_ssrcs() const {
|
|
return outgoing_rtx_ssrcs_;
|
|
}
|
|
const std::set<uint32_t>& outgoing_video_ssrcs() const {
|
|
return outgoing_video_ssrcs_;
|
|
}
|
|
const std::set<uint32_t>& outgoing_audio_ssrcs() const {
|
|
return outgoing_audio_ssrcs_;
|
|
}
|
|
|
|
const std::vector<LoggedStartEvent>& start_log_events() const {
|
|
return start_log_events_;
|
|
}
|
|
const std::vector<LoggedStopEvent>& stop_log_events() const {
|
|
return stop_log_events_;
|
|
}
|
|
const std::map<uint32_t, std::vector<LoggedAudioPlayoutEvent>>&
|
|
audio_playout_events() const {
|
|
return audio_playout_events_;
|
|
}
|
|
const std::vector<LoggedAudioNetworkAdaptationEvent>&
|
|
audio_network_adaptation_events() const {
|
|
return audio_network_adaptation_events_;
|
|
}
|
|
const std::vector<LoggedBweProbeClusterCreatedEvent>&
|
|
bwe_probe_cluster_created_events() const {
|
|
return bwe_probe_cluster_created_events_;
|
|
}
|
|
const std::vector<LoggedBweProbeResultEvent>& bwe_probe_result_events()
|
|
const {
|
|
return bwe_probe_result_events_;
|
|
}
|
|
const std::vector<LoggedBweDelayBasedUpdate>& bwe_delay_updates() const {
|
|
return bwe_delay_updates_;
|
|
}
|
|
const std::vector<LoggedBweLossBasedUpdate>& bwe_loss_updates() const {
|
|
return bwe_loss_updates_;
|
|
}
|
|
const std::vector<LoggedAlrStateEvent>& alr_state_events() const {
|
|
return alr_state_events_;
|
|
}
|
|
const std::vector<LoggedIceCandidatePairConfig>& ice_candidate_pair_configs()
|
|
const {
|
|
return ice_candidate_pair_configs_;
|
|
}
|
|
const std::vector<LoggedIceCandidatePairEvent>& ice_candidate_pair_events()
|
|
const {
|
|
return ice_candidate_pair_events_;
|
|
}
|
|
|
|
struct LoggedRtpStreamIncoming {
|
|
uint32_t ssrc;
|
|
std::vector<LoggedRtpPacketIncoming> incoming_packets;
|
|
};
|
|
|
|
struct LoggedRtpStreamOutgoing {
|
|
uint32_t ssrc;
|
|
std::vector<LoggedRtpPacketOutgoing> outgoing_packets;
|
|
};
|
|
|
|
struct LoggedRtpStreamView {
|
|
LoggedRtpStreamView(uint32_t ssrc,
|
|
const LoggedRtpPacketIncoming* ptr,
|
|
size_t num_elements)
|
|
: ssrc(ssrc),
|
|
packet_view(PacketView<const LoggedRtpPacket>::Create(
|
|
ptr,
|
|
num_elements,
|
|
offsetof(LoggedRtpPacketIncoming, rtp))) {}
|
|
LoggedRtpStreamView(uint32_t ssrc,
|
|
const LoggedRtpPacketOutgoing* ptr,
|
|
size_t num_elements)
|
|
: ssrc(ssrc),
|
|
packet_view(PacketView<const LoggedRtpPacket>::Create(
|
|
ptr,
|
|
num_elements,
|
|
offsetof(LoggedRtpPacketOutgoing, rtp))) {}
|
|
uint32_t ssrc;
|
|
PacketView<const LoggedRtpPacket> packet_view;
|
|
};
|
|
|
|
const std::vector<LoggedRtpStreamIncoming>& incoming_rtp_packets_by_ssrc()
|
|
const {
|
|
return incoming_rtp_packets_by_ssrc_;
|
|
}
|
|
|
|
const std::vector<LoggedRtpStreamOutgoing>& outgoing_rtp_packets_by_ssrc()
|
|
const {
|
|
return outgoing_rtp_packets_by_ssrc_;
|
|
}
|
|
|
|
const std::vector<LoggedRtcpPacketIncoming>& incoming_rtcp_packets() const {
|
|
return incoming_rtcp_packets_;
|
|
}
|
|
|
|
const std::vector<LoggedRtcpPacketOutgoing>& outgoing_rtcp_packets() const {
|
|
return outgoing_rtcp_packets_;
|
|
}
|
|
|
|
const std::vector<LoggedRtpStreamView>& rtp_packets_by_ssrc(
|
|
PacketDirection direction) const {
|
|
if (direction == kIncomingPacket)
|
|
return incoming_rtp_packet_views_by_ssrc_;
|
|
else
|
|
return outgoing_rtp_packet_views_by_ssrc_;
|
|
}
|
|
|
|
const std::vector<LoggedRtcpPacketReceiverReport>& receiver_reports(
|
|
PacketDirection direction) const {
|
|
if (direction == kIncomingPacket) {
|
|
return incoming_rr_;
|
|
} else {
|
|
return outgoing_rr_;
|
|
}
|
|
}
|
|
|
|
const std::vector<LoggedRtcpPacketSenderReport>& sender_reports(
|
|
PacketDirection direction) const {
|
|
if (direction == kIncomingPacket) {
|
|
return incoming_sr_;
|
|
} else {
|
|
return outgoing_sr_;
|
|
}
|
|
}
|
|
|
|
const std::vector<LoggedRtcpPacketNack>& nacks(
|
|
PacketDirection direction) const {
|
|
if (direction == kIncomingPacket) {
|
|
return incoming_nack_;
|
|
} else {
|
|
return outgoing_nack_;
|
|
}
|
|
}
|
|
|
|
const std::vector<LoggedRtcpPacketRemb>& rembs(
|
|
PacketDirection direction) const {
|
|
if (direction == kIncomingPacket) {
|
|
return incoming_remb_;
|
|
} else {
|
|
return outgoing_remb_;
|
|
}
|
|
}
|
|
|
|
const std::vector<LoggedRtcpPacketTransportFeedback>& transport_feedbacks(
|
|
PacketDirection direction) const {
|
|
if (direction == kIncomingPacket) {
|
|
return incoming_transport_feedback_;
|
|
} else {
|
|
return outgoing_transport_feedback_;
|
|
}
|
|
}
|
|
|
|
int64_t first_timestamp() const { return first_timestamp_; }
|
|
int64_t last_timestamp() const { return last_timestamp_; }
|
|
|
|
private:
|
|
void StoreParsedEvent(const rtclog::Event& event);
|
|
|
|
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;
|
|
|
|
LoggedAudioPlayoutEvent GetAudioPlayout(const rtclog::Event& event) const;
|
|
|
|
LoggedBweLossBasedUpdate GetLossBasedBweUpdate(
|
|
const rtclog::Event& event) const;
|
|
LoggedBweDelayBasedUpdate GetDelayBasedBweUpdate(
|
|
const rtclog::Event& event) const;
|
|
|
|
LoggedAudioNetworkAdaptationEvent GetAudioNetworkAdaptation(
|
|
const rtclog::Event& event) const;
|
|
|
|
LoggedBweProbeClusterCreatedEvent GetBweProbeClusterCreated(
|
|
const rtclog::Event& event) const;
|
|
LoggedBweProbeResultEvent GetBweProbeResult(const rtclog::Event& event) const;
|
|
|
|
LoggedAlrStateEvent GetAlrState(const rtclog::Event& event) const;
|
|
|
|
LoggedIceCandidatePairConfig GetIceCandidatePairConfig(
|
|
const rtclog::Event& event) const;
|
|
LoggedIceCandidatePairEvent GetIceCandidatePairEvent(
|
|
const rtclog::Event& event) const;
|
|
|
|
std::vector<rtclog::Event> events_;
|
|
|
|
struct Stream {
|
|
Stream(uint32_t ssrc,
|
|
MediaType media_type,
|
|
PacketDirection direction,
|
|
webrtc::RtpHeaderExtensionMap map)
|
|
: ssrc(ssrc),
|
|
media_type(media_type),
|
|
direction(direction),
|
|
rtp_extensions_map(map) {}
|
|
uint32_t ssrc;
|
|
MediaType media_type;
|
|
PacketDirection direction;
|
|
webrtc::RtpHeaderExtensionMap rtp_extensions_map;
|
|
};
|
|
|
|
const UnconfiguredHeaderExtensions parse_unconfigured_header_extensions_;
|
|
|
|
// Make a default extension map for streams without configuration information.
|
|
// TODO(ivoc): Once configuration of audio streams is stored in the event log,
|
|
// this can be removed. Tracking bug: webrtc:6399
|
|
RtpHeaderExtensionMap default_extension_map_;
|
|
|
|
// Tracks what each stream is configured for. Note that a single SSRC can be
|
|
// in several sets. For example, the SSRC used for sending video over RTX
|
|
// will appear in both video_ssrcs_ and rtx_ssrcs_. In the unlikely case that
|
|
// an SSRC is reconfigured to a different media type mid-call, it will also
|
|
// appear in multiple sets.
|
|
std::set<uint32_t> incoming_rtx_ssrcs_;
|
|
std::set<uint32_t> incoming_video_ssrcs_;
|
|
std::set<uint32_t> incoming_audio_ssrcs_;
|
|
std::set<uint32_t> outgoing_rtx_ssrcs_;
|
|
std::set<uint32_t> outgoing_video_ssrcs_;
|
|
std::set<uint32_t> outgoing_audio_ssrcs_;
|
|
|
|
// Maps an SSRC to the parsed RTP headers in that stream. Header extensions
|
|
// are parsed if the stream has been configured. This is only used for
|
|
// grouping the events by SSRC during parsing; the events are moved to
|
|
// incoming_rtp_packets_by_ssrc_ once the parsing is done.
|
|
std::map<uint32_t, std::vector<LoggedRtpPacketIncoming>>
|
|
incoming_rtp_packets_map_;
|
|
std::map<uint32_t, std::vector<LoggedRtpPacketOutgoing>>
|
|
outgoing_rtp_packets_map_;
|
|
|
|
// RTP headers.
|
|
std::vector<LoggedRtpStreamIncoming> incoming_rtp_packets_by_ssrc_;
|
|
std::vector<LoggedRtpStreamOutgoing> outgoing_rtp_packets_by_ssrc_;
|
|
std::vector<LoggedRtpStreamView> incoming_rtp_packet_views_by_ssrc_;
|
|
std::vector<LoggedRtpStreamView> outgoing_rtp_packet_views_by_ssrc_;
|
|
|
|
// Raw RTCP packets.
|
|
std::vector<LoggedRtcpPacketIncoming> incoming_rtcp_packets_;
|
|
std::vector<LoggedRtcpPacketOutgoing> outgoing_rtcp_packets_;
|
|
|
|
// Parsed RTCP messages. Currently not separated based on SSRC.
|
|
std::vector<LoggedRtcpPacketReceiverReport> incoming_rr_;
|
|
std::vector<LoggedRtcpPacketReceiverReport> outgoing_rr_;
|
|
std::vector<LoggedRtcpPacketSenderReport> incoming_sr_;
|
|
std::vector<LoggedRtcpPacketSenderReport> outgoing_sr_;
|
|
std::vector<LoggedRtcpPacketNack> incoming_nack_;
|
|
std::vector<LoggedRtcpPacketNack> outgoing_nack_;
|
|
std::vector<LoggedRtcpPacketRemb> incoming_remb_;
|
|
std::vector<LoggedRtcpPacketRemb> outgoing_remb_;
|
|
std::vector<LoggedRtcpPacketTransportFeedback> incoming_transport_feedback_;
|
|
std::vector<LoggedRtcpPacketTransportFeedback> outgoing_transport_feedback_;
|
|
|
|
std::vector<LoggedStartEvent> start_log_events_;
|
|
std::vector<LoggedStopEvent> stop_log_events_;
|
|
|
|
std::map<uint32_t, std::vector<LoggedAudioPlayoutEvent>>
|
|
audio_playout_events_;
|
|
|
|
std::vector<LoggedAudioNetworkAdaptationEvent>
|
|
audio_network_adaptation_events_;
|
|
|
|
std::vector<LoggedBweProbeClusterCreatedEvent>
|
|
bwe_probe_cluster_created_events_;
|
|
|
|
std::vector<LoggedBweProbeResultEvent> bwe_probe_result_events_;
|
|
|
|
std::vector<LoggedBweDelayBasedUpdate> bwe_delay_updates_;
|
|
|
|
// A list of all updates from the send-side loss-based bandwidth estimator.
|
|
std::vector<LoggedBweLossBasedUpdate> bwe_loss_updates_;
|
|
|
|
std::vector<LoggedAlrStateEvent> alr_state_events_;
|
|
|
|
std::vector<LoggedIceCandidatePairConfig> ice_candidate_pair_configs_;
|
|
|
|
std::vector<LoggedIceCandidatePairEvent> ice_candidate_pair_events_;
|
|
|
|
std::vector<LoggedAudioRecvConfig> audio_recv_configs_;
|
|
std::vector<LoggedAudioSendConfig> audio_send_configs_;
|
|
std::vector<LoggedVideoRecvConfig> video_recv_configs_;
|
|
std::vector<LoggedVideoSendConfig> video_send_configs_;
|
|
|
|
uint8_t last_incoming_rtcp_packet_[IP_PACKET_SIZE];
|
|
uint8_t last_incoming_rtcp_packet_length_;
|
|
|
|
int64_t first_timestamp_;
|
|
int64_t last_timestamp_;
|
|
|
|
// The extension maps are mutable to allow us to insert the default
|
|
// configuration when parsing an RTP header for an unconfigured stream.
|
|
mutable std::map<uint32_t, webrtc::RtpHeaderExtensionMap>
|
|
incoming_rtp_extensions_maps_;
|
|
mutable std::map<uint32_t, webrtc::RtpHeaderExtensionMap>
|
|
outgoing_rtp_extensions_maps_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_NEW_H_
|