13181035bc
The plot is constructed by actually running the congestion controller with the logged rtp headers and rtcp feedback messages to reproduce the same behavior as in the real call. R=phoglund@webrtc.org, terelius@webrtc.org Review URL: https://codereview.webrtc.org/2193763002 . Cr-Commit-Position: refs/heads/master@{#13574}
124 lines
3.9 KiB
C++
124 lines
3.9 KiB
C++
/*
|
|
* 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 WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
|
|
#define WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
|
|
|
|
#include <vector>
|
|
#include <map>
|
|
#include <memory>
|
|
#include <utility>
|
|
|
|
#include "webrtc/call/rtc_event_log_parser.h"
|
|
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
|
|
#include "webrtc/tools/event_log_visualizer/plot_base.h"
|
|
|
|
namespace webrtc {
|
|
namespace plotting {
|
|
|
|
class EventLogAnalyzer {
|
|
public:
|
|
// 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 CreatePlayoutGraph(Plot* plot);
|
|
|
|
void CreateSequenceNumberGraph(Plot* plot);
|
|
|
|
void CreateDelayChangeGraph(Plot* plot);
|
|
|
|
void CreateAccumulatedDelayChangeGraph(Plot* plot);
|
|
|
|
void CreateTotalBitrateGraph(PacketDirection desired_direction, Plot* plot);
|
|
|
|
void CreateStreamBitrateGraph(PacketDirection desired_direction, Plot* plot);
|
|
|
|
void CreateBweGraph(Plot* plot);
|
|
|
|
private:
|
|
class StreamId {
|
|
public:
|
|
StreamId(uint32_t ssrc, webrtc::PacketDirection direction)
|
|
: ssrc_(ssrc), direction_(direction) {}
|
|
bool operator<(const StreamId& other) const;
|
|
bool operator==(const StreamId& other) const;
|
|
uint32_t GetSsrc() const { return ssrc_; }
|
|
webrtc::PacketDirection GetDirection() const { return direction_; }
|
|
|
|
private:
|
|
uint32_t ssrc_;
|
|
webrtc::PacketDirection direction_;
|
|
};
|
|
|
|
struct LoggedRtpPacket {
|
|
LoggedRtpPacket(uint64_t timestamp, RTPHeader header, size_t total_length)
|
|
: timestamp(timestamp), header(header), total_length(total_length) {}
|
|
uint64_t timestamp;
|
|
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 BwePacketLossEvent {
|
|
uint64_t timestamp;
|
|
int32_t new_bitrate;
|
|
uint8_t fraction_loss;
|
|
int32_t expected_packets;
|
|
};
|
|
|
|
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_;
|
|
|
|
// Maps a stream identifier consisting of ssrc, direction and MediaType
|
|
// 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_;
|
|
|
|
// A list of all updates from the send-side loss-based bandwidth estimator.
|
|
std::vector<BwePacketLossEvent> bwe_loss_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_;
|
|
};
|
|
|
|
} // namespace plotting
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
|