blob: 328fc531babc068ef22cc1a129a6ed7b7995b70a [file] [log] [blame]
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <map>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/rtc_base/function_view.h"
#include "webrtc/rtc_tools/event_log_visualizer/plot_base.h"
namespace webrtc {
namespace plotting {
struct LoggedRtpPacket {
LoggedRtpPacket(uint64_t timestamp, RTPHeader header, size_t total_length)
: timestamp(timestamp), header(header), total_length(total_length) {}
uint64_t timestamp;
// TODO(terelius): This allocates space for 15 CSRCs even if none are used.
RTPHeader header;
size_t total_length;
struct LoggedRtcpPacket {
LoggedRtcpPacket(uint64_t timestamp,
RTCPPacketType rtcp_type,
std::unique_ptr<rtcp::RtcpPacket> rtcp_packet)
: timestamp(timestamp), type(rtcp_type), packet(std::move(rtcp_packet)) {}
uint64_t timestamp;
RTCPPacketType type;
std::unique_ptr<rtcp::RtcpPacket> packet;
struct LossBasedBweUpdate {
uint64_t timestamp;
int32_t new_bitrate;
uint8_t fraction_loss;
int32_t expected_packets;
struct AudioNetworkAdaptationEvent {
uint64_t timestamp;
AudioEncoderRuntimeConfig config;
class EventLogAnalyzer {
// The EventLogAnalyzer keeps a reference to the ParsedRtcEventLog for the
// duration of its lifetime. The ParsedRtcEventLog must not be destroyed or
// modified while the EventLogAnalyzer is being used.
explicit EventLogAnalyzer(const ParsedRtcEventLog& log);
void CreatePacketGraph(PacketDirection desired_direction, Plot* plot);
void CreateAccumulatedPacketsGraph(PacketDirection desired_direction,
Plot* plot);
void CreatePlayoutGraph(Plot* plot);
void CreateAudioLevelGraph(Plot* plot);
void CreateSequenceNumberGraph(Plot* plot);
void CreateIncomingPacketLossGraph(Plot* plot);
void CreateIncomingDelayDeltaGraph(Plot* plot);
void CreateIncomingDelayGraph(Plot* plot);
void CreateFractionLossGraph(Plot* plot);
void CreateTotalBitrateGraph(PacketDirection desired_direction,
Plot* plot,
bool show_detector_state = false);
void CreateStreamBitrateGraph(PacketDirection desired_direction, Plot* plot);
void CreateBweSimulationGraph(Plot* plot);
void CreateNetworkDelayFeedbackGraph(Plot* plot);
void CreateTimestampGraph(Plot* plot);
void CreateAudioEncoderTargetBitrateGraph(Plot* plot);
void CreateAudioEncoderFrameLengthGraph(Plot* plot);
void CreateAudioEncoderPacketLossGraph(Plot* plot);
void CreateAudioEncoderEnableFecGraph(Plot* plot);
void CreateAudioEncoderEnableDtxGraph(Plot* plot);
void CreateAudioEncoderNumChannelsGraph(Plot* plot);
void CreateAudioJitterBufferGraph(const std::string& replacement_file_name,
int file_sample_rate_hz,
Plot* plot);
// Returns a vector of capture and arrival timestamps for the video frames
// of the stream with the most number of frames.
std::vector<std::pair<int64_t, int64_t>> GetFrameTimestamps() const;
class StreamId {
StreamId(uint32_t ssrc, webrtc::PacketDirection direction)
: ssrc_(ssrc), direction_(direction) {}
bool operator<(const StreamId& other) const {
return std::tie(ssrc_, direction_) <
std::tie(other.ssrc_, other.direction_);
bool operator==(const StreamId& other) const {
return std::tie(ssrc_, direction_) ==
std::tie(other.ssrc_, other.direction_);
uint32_t GetSsrc() const { return ssrc_; }
webrtc::PacketDirection GetDirection() const { return direction_; }
uint32_t ssrc_;
webrtc::PacketDirection direction_;
template <typename T>
void CreateAccumulatedPacketsTimeSeries(
PacketDirection desired_direction,
Plot* plot,
const std::map<StreamId, std::vector<T>>& packets,
const std::string& label_prefix);
bool IsRtxSsrc(StreamId stream_id) const;
bool IsVideoSsrc(StreamId stream_id) const;
bool IsAudioSsrc(StreamId stream_id) const;
std::string GetStreamName(StreamId) const;
const ParsedRtcEventLog& parsed_log_;
// A list of SSRCs we are interested in analysing.
// If left empty, all SSRCs will be considered relevant.
std::vector<uint32_t> desired_ssrc_;
// 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<StreamId> rtx_ssrcs_;
std::set<StreamId> video_ssrcs_;
std::set<StreamId> audio_ssrcs_;
// Maps a stream identifier consisting of ssrc and direction to the parsed
// RTP headers in that stream. Header extensions are parsed if the stream
// has been configured.
std::map<StreamId, std::vector<LoggedRtpPacket>> rtp_packets_;
std::map<StreamId, std::vector<LoggedRtcpPacket>> rtcp_packets_;
// Maps an SSRC to the timestamps of parsed audio playout events.
std::map<uint32_t, std::vector<uint64_t>> audio_playout_events_;
// Stores the timestamps for all log segments, in the form of associated start
// and end events.
std::vector<std::pair<uint64_t, uint64_t>> log_segments_;
// A list of all updates from the send-side loss-based bandwidth estimator.
std::vector<LossBasedBweUpdate> bwe_loss_updates_;
std::vector<AudioNetworkAdaptationEvent> audio_network_adaptation_events_;
std::vector<ParsedRtcEventLog::BweProbeResultEvent> bwe_probe_result_events_;
std::vector<ParsedRtcEventLog::BweDelayBasedUpdate> bwe_delay_updates_;
// Window and step size used for calculating moving averages, e.g. bitrate.
// The generated data points will be |step_| microseconds apart.
// Only events occuring at most |window_duration_| microseconds before the
// current data point will be part of the average.
uint64_t window_duration_;
uint64_t step_;
// First and last events of the log.
uint64_t begin_time_;
uint64_t end_time_;
// Duration (in seconds) of log file.
float call_duration_s_;
} // namespace plotting
} // namespace webrtc