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Revert "Create new API for RtcEventLogParser."

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This reverts commit 9e336ec0b8a77c3461d13677cff3563c11c88daa.

Reason for revert: Code can accidentally include the deprecated parser but link with the new one, or vice versa. Reverting to fix naming.

Original change's description:
> Create new API for RtcEventLogParser.
> 
> The new API stores events gathered by event type. For example, it is
> possible to ask fo a list of all incoming RTCP messages or all audio
> playout events.
> 
> The new API is experimental and may change over next few weeks. Once
> it has stabilized and all unit tests and existing tools have been
> ported to the new API, the old one will be removed.
> 
> This CL also updates the event_log_visualizer tool to use the new
> parser API. This is not a funcional change except for:
> - Incoming and outgoing audio level are now drawn in two separate plots.
> - Incoming and outgoing timstamps are now drawn in two separate plots.
> - RTCP count is no longer split into Video and Audio. It also counts
>   all RTCP packets rather than only specific message types.
> - Slight timing difference in sendside BWE simulation due to only
>   iterating over transport feedbacks and not over all RTCP packets.
>   This timing changes are not visible in the plots.
> 
> 
> Media type for RTCP messages might not be identified correctly by
> rtc_event_log2text anymore. On the other hand, assigning a specific
> media type to an RTCP packet was a bit hacky to begin with.
> 
> Bug: webrtc:8111
> Change-Id: I8e7168302beb69b2e163a097a2a142b86dd4a26b
> Reviewed-on: https://webrtc-review.googlesource.com/60865
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23015}

TBR=terelius@webrtc.org,srte@webrtc.org,minyue@webrtc.org

Change-Id: Ib4bbcf0563423675a3cc1dce59ebf665e0c5dae9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8111
Reviewed-on: https://webrtc-review.googlesource.com/72500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23026}
This commit is contained in:
Björn Terelius
2018-04-25 14:23:01 +00:00
committed by Commit Bot
parent 65fb4049c1
commit ff61273c01
23 changed files with 1422 additions and 3279 deletions
Download Patch File Download Diff File Expand all files Collapse all files

257
rtc_tools/event_log_visualizer/analyzer.h
View File

@ -18,12 +18,54 @@
#include <utility>
#include <vector>
#include "logging/rtc_event_log/rtc_event_log_parser2.h"
#include "rtc_base/strings/string_builder.h"
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet.h"
#include "rtc_base/function_view.h"
#include "rtc_tools/event_log_visualizer/plot_base.h"
#include "rtc_tools/event_log_visualizer/triage_notifications.h"
namespace webrtc {
namespace plotting {
struct LoggedRtpPacket {
LoggedRtpPacket(uint64_t timestamp,
RTPHeader header,
size_t header_length,
size_t total_length)
: timestamp(timestamp),
header(header),
header_length(header_length),
total_length(total_length) {}
uint64_t timestamp;
// TODO(terelius): This allocates space for 15 CSRCs even if none are used.
RTPHeader header;
size_t header_length;
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 {
public:
@ -32,13 +74,14 @@ class EventLogAnalyzer {
// modified while the EventLogAnalyzer is being used.
explicit EventLogAnalyzer(const ParsedRtcEventLog& log);
void CreatePacketGraph(PacketDirection direction, Plot* plot);
void CreatePacketGraph(PacketDirection desired_direction, Plot* plot);
void CreateAccumulatedPacketsGraph(PacketDirection direction, Plot* plot);
void CreateAccumulatedPacketsGraph(PacketDirection desired_direction,
Plot* plot);
void CreatePlayoutGraph(Plot* plot);
void CreateAudioLevelGraph(PacketDirection direction, Plot* plot);
void CreateAudioLevelGraph(Plot* plot);
void CreateSequenceNumberGraph(Plot* plot);
@ -49,20 +92,19 @@ class EventLogAnalyzer {
void CreateFractionLossGraph(Plot* plot);
void CreateTotalIncomingBitrateGraph(Plot* plot);
void CreateTotalOutgoingBitrateGraph(Plot* plot,
bool show_detector_state = false,
bool show_alr_state = false);
void CreateTotalBitrateGraph(PacketDirection desired_direction,
Plot* plot,
bool show_detector_state = false,
bool show_alr_state = false);
void CreateStreamBitrateGraph(PacketDirection direction, Plot* plot);
void CreateStreamBitrateGraph(PacketDirection desired_direction, Plot* plot);
void CreateSendSideBweSimulationGraph(Plot* plot);
void CreateReceiveSideBweSimulationGraph(Plot* plot);
void CreateNetworkDelayFeedbackGraph(Plot* plot);
void CreatePacerDelayGraph(Plot* plot);
void CreateTimestampGraph(PacketDirection direction, Plot* plot);
void CreateTimestampGraph(Plot* plot);
void CreateAudioEncoderTargetBitrateGraph(Plot* plot);
void CreateAudioEncoderFrameLengthGraph(Plot* plot);
@ -77,114 +119,55 @@ class EventLogAnalyzer {
void CreateIceCandidatePairConfigGraph(Plot* plot);
void CreateIceConnectivityCheckGraph(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;
void CreateTriageNotifications();
void PrintNotifications(FILE* file);
private:
bool IsRtxSsrc(PacketDirection direction, uint32_t ssrc) const {
if (direction == kIncomingPacket) {
return parsed_log_.incoming_rtx_ssrcs().find(ssrc) !=
parsed_log_.incoming_rtx_ssrcs().end();
} else {
return parsed_log_.outgoing_rtx_ssrcs().find(ssrc) !=
parsed_log_.outgoing_rtx_ssrcs().end();
class StreamId {
public:
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 IsVideoSsrc(PacketDirection direction, uint32_t ssrc) const {
if (direction == kIncomingPacket) {
return parsed_log_.incoming_video_ssrcs().find(ssrc) !=
parsed_log_.incoming_video_ssrcs().end();
} else {
return parsed_log_.outgoing_video_ssrcs().find(ssrc) !=
parsed_log_.outgoing_video_ssrcs().end();
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_; }
bool IsAudioSsrc(PacketDirection direction, uint32_t ssrc) const {
if (direction == kIncomingPacket) {
return parsed_log_.incoming_audio_ssrcs().find(ssrc) !=
parsed_log_.incoming_audio_ssrcs().end();
} else {
return parsed_log_.outgoing_audio_ssrcs().find(ssrc) !=
parsed_log_.outgoing_audio_ssrcs().end();
}
}
private:
uint32_t ssrc_;
webrtc::PacketDirection direction_;
};
template <typename IterableType>
void CreateAccumulatedPacketsTimeSeries(Plot* plot,
const IterableType& packets,
const std::string& label);
template <typename T>
void CreateAccumulatedPacketsTimeSeries(
PacketDirection desired_direction,
Plot* plot,
const std::map<StreamId, std::vector<T>>& packets,
const std::string& label_prefix);
void CreateStreamGapAlerts(PacketDirection direction);
void CreateTransmissionGapAlerts(PacketDirection direction);
bool IsRtxSsrc(StreamId stream_id) const;
std::string GetStreamName(PacketDirection direction, uint32_t ssrc) const {
char buffer[200];
rtc::SimpleStringBuilder name(buffer);
if (IsAudioSsrc(direction, ssrc)) {
name << "Audio ";
} else if (IsVideoSsrc(direction, ssrc)) {
name << "Video ";
} else {
name << "Unknown ";
}
if (IsRtxSsrc(direction, ssrc)) {
name << "RTX ";
}
if (direction == kIncomingPacket)
name << "(In) ";
else
name << "(Out) ";
name << "SSRC " << ssrc;
return name.str();
}
bool IsVideoSsrc(StreamId stream_id) const;
bool IsAudioSsrc(StreamId stream_id) const;
std::string GetStreamName(StreamId stream_id) const;
rtc::Optional<uint32_t> EstimateRtpClockFrequency(
const std::vector<LoggedRtpPacket>& packets) const;
float ToCallTime(int64_t timestamp) const;
void Alert_RtpLogTimeGap(PacketDirection direction,
float time_seconds,
int64_t duration) {
if (direction == kIncomingPacket) {
incoming_rtp_recv_time_gaps_.emplace_back(time_seconds, duration);
} else {
outgoing_rtp_send_time_gaps_.emplace_back(time_seconds, duration);
}
}
void Alert_RtcpLogTimeGap(PacketDirection direction,
float time_seconds,
int64_t duration) {
if (direction == kIncomingPacket) {
incoming_rtcp_recv_time_gaps_.emplace_back(time_seconds, duration);
} else {
outgoing_rtcp_send_time_gaps_.emplace_back(time_seconds, duration);
}
}
void Alert_SeqNumJump(PacketDirection direction,
float time_seconds,
uint32_t ssrc) {
if (direction == kIncomingPacket) {
incoming_seq_num_jumps_.emplace_back(time_seconds, ssrc);
} else {
outgoing_seq_num_jumps_.emplace_back(time_seconds, ssrc);
}
}
void Alert_CaptureTimeJump(PacketDirection direction,
float time_seconds,
uint32_t ssrc) {
if (direction == kIncomingPacket) {
incoming_capture_time_jumps_.emplace_back(time_seconds, ssrc);
} else {
outgoing_capture_time_jumps_.emplace_back(time_seconds, ssrc);
}
}
void Alert_OutgoingHighLoss(double avg_loss_fraction) {
outgoing_high_loss_alerts_.emplace_back(avg_loss_fraction);
}
void Notification(std::unique_ptr<TriageNotification> notification);
std::string GetCandidatePairLogDescriptionFromId(uint32_t candidate_pair_id);
@ -194,19 +177,50 @@ class EventLogAnalyzer {
// 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<int64_t, int64_t>> log_segments_;
std::vector<std::pair<uint64_t, uint64_t>> log_segments_;
std::vector<IncomingRtpReceiveTimeGap> incoming_rtp_recv_time_gaps_;
std::vector<IncomingRtcpReceiveTimeGap> incoming_rtcp_recv_time_gaps_;
std::vector<OutgoingRtpSendTimeGap> outgoing_rtp_send_time_gaps_;
std::vector<OutgoingRtcpSendTimeGap> outgoing_rtcp_send_time_gaps_;
std::vector<IncomingSeqNumJump> incoming_seq_num_jumps_;
std::vector<IncomingCaptureTimeJump> incoming_capture_time_jumps_;
std::vector<OutgoingSeqNoJump> outgoing_seq_num_jumps_;
std::vector<OutgoingCaptureTimeJump> outgoing_capture_time_jumps_;
std::vector<OutgoingHighLoss> outgoing_high_loss_alerts_;
// 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::BweProbeClusterCreatedEvent>
bwe_probe_cluster_created_events_;
std::vector<ParsedRtcEventLog::BweProbeResultEvent> bwe_probe_result_events_;
std::vector<ParsedRtcEventLog::BweDelayBasedUpdate> bwe_delay_updates_;
std::vector<std::unique_ptr<TriageNotification>> notifications_;
std::vector<ParsedRtcEventLog::AlrStateEvent> alr_state_events_;
std::vector<ParsedRtcEventLog::IceCandidatePairConfig>
ice_candidate_pair_configs_;
std::vector<ParsedRtcEventLog::IceCandidatePairEvent>
ice_candidate_pair_events_;
std::map<uint32_t, std::string> candidate_pair_desc_by_id_;
@ -214,17 +228,18 @@ class EventLogAnalyzer {
// 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.
int64_t window_duration_;
int64_t step_;
uint64_t window_duration_;
uint64_t step_;
// First and last events of the log.
int64_t begin_time_;
int64_t end_time_;
uint64_t begin_time_;
uint64_t end_time_;
// Duration (in seconds) of log file.
float call_duration_s_;
};
} // namespace plotting
} // namespace webrtc
#endif // RTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_