Plot NetEq stats in RTC event log visualizer.
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}diff --git a/logging/rtc_event_log/rtc_event_log_parser_new.cc b/logging/rtc_event_log/rtc_event_log_parser_new.cc
index c52e474..47c635e 100644
--- a/logging/rtc_event_log/rtc_event_log_parser_new.cc
+++ b/logging/rtc_event_log/rtc_event_log_parser_new.cc
@@ -633,8 +633,7 @@
}
case ParsedRtcEventLogNew::AUDIO_PLAYOUT_EVENT: {
LoggedAudioPlayoutEvent playout_event = GetAudioPlayout(event);
- audio_playout_events_[playout_event.ssrc].push_back(
- playout_event.timestamp_us);
+ audio_playout_events_[playout_event.ssrc].push_back(playout_event);
break;
}
case ParsedRtcEventLogNew::LOSS_BASED_BWE_UPDATE: {
diff --git a/logging/rtc_event_log/rtc_event_log_parser_new.h b/logging/rtc_event_log/rtc_event_log_parser_new.h
index 19cfcbb..98d2a52 100644
--- a/logging/rtc_event_log/rtc_event_log_parser_new.h
+++ b/logging/rtc_event_log/rtc_event_log_parser_new.h
@@ -636,7 +636,8 @@
const std::vector<LoggedStopEvent>& stop_log_events() const {
return stop_log_events_;
}
- const std::map<uint32_t, std::vector<int64_t>>& audio_playout_events() const {
+ const std::map<uint32_t, std::vector<LoggedAudioPlayoutEvent>>&
+ audio_playout_events() const {
return audio_playout_events_;
}
const std::vector<LoggedAudioNetworkAdaptationEvent>&
@@ -874,8 +875,8 @@
std::vector<LoggedStartEvent> start_log_events_;
std::vector<LoggedStopEvent> stop_log_events_;
- // Maps an SSRC to the timestamps of parsed audio playout events.
- std::map<uint32_t, std::vector<int64_t>> audio_playout_events_;
+ std::map<uint32_t, std::vector<LoggedAudioPlayoutEvent>>
+ audio_playout_events_;
std::vector<LoggedAudioNetworkAdaptationEvent>
audio_network_adaptation_events_;
diff --git a/modules/audio_coding/neteq/tools/neteq_input.h b/modules/audio_coding/neteq/tools/neteq_input.h
index 88d9eb9..68c076e 100644
--- a/modules/audio_coding/neteq/tools/neteq_input.h
+++ b/modules/audio_coding/neteq/tools/neteq_input.h
@@ -32,7 +32,7 @@
RTPHeader header;
rtc::Buffer payload;
- double time_ms;
+ int64_t time_ms;
};
virtual ~NetEqInput() = default;
diff --git a/rtc_tools/event_log_visualizer/analyzer.cc b/rtc_tools/event_log_visualizer/analyzer.cc
index efb14eb..145b2ac 100644
--- a/rtc_tools/event_log_visualizer/analyzer.cc
+++ b/rtc_tools/event_log_visualizer/analyzer.cc
@@ -604,16 +604,15 @@
uint32_t ssrc = playout_stream.first;
if (!MatchingSsrc(ssrc, desired_ssrc_))
continue;
- rtc::Optional<int64_t> last_playout;
+ rtc::Optional<int64_t> last_playout_ms;
TimeSeries time_series(SsrcToString(ssrc), LineStyle::kBar);
- for (const auto& playout_time : playout_stream.second) {
- float x = ToCallTime(playout_time);
+ for (const auto& playout_event : playout_stream.second) {
+ float x = ToCallTime(playout_event.log_time_us());
+ int64_t playout_time_ms = playout_event.log_time_ms();
// If there were no previous playouts, place the point on the x-axis.
- float y = static_cast<float>(playout_time -
- last_playout.value_or(playout_time)) /
- 1000;
+ float y = playout_time_ms - last_playout_ms.value_or(playout_time_ms);
time_series.points.push_back(TimeSeriesPoint(x, y));
- last_playout.emplace(playout_time);
+ last_playout_ms.emplace(playout_time_ms);
}
plot->AppendTimeSeries(std::move(time_series));
}
@@ -1561,37 +1560,35 @@
// Does not take any ownership, and all pointers must refer to valid objects
// that outlive the one constructed.
NetEqStreamInput(const std::vector<LoggedRtpPacketIncoming>* packet_stream,
- const std::vector<int64_t>* output_events_us,
- rtc::Optional<int64_t> end_time_us)
+ const std::vector<LoggedAudioPlayoutEvent>* output_events,
+ rtc::Optional<int64_t> end_time_ms)
: packet_stream_(*packet_stream),
packet_stream_it_(packet_stream_.begin()),
- output_events_us_it_(output_events_us->begin()),
- output_events_us_end_(output_events_us->end()),
- end_time_us_(end_time_us) {
+ output_events_it_(output_events->begin()),
+ output_events_end_(output_events->end()),
+ end_time_ms_(end_time_ms) {
RTC_DCHECK(packet_stream);
- RTC_DCHECK(output_events_us);
+ RTC_DCHECK(output_events);
}
rtc::Optional<int64_t> NextPacketTime() const override {
if (packet_stream_it_ == packet_stream_.end()) {
return rtc::nullopt;
}
- if (end_time_us_ && packet_stream_it_->rtp.log_time_us() > *end_time_us_) {
+ if (end_time_ms_ && packet_stream_it_->rtp.log_time_ms() > *end_time_ms_) {
return rtc::nullopt;
}
- // Convert from us to ms.
- return packet_stream_it_->rtp.log_time_us() / 1000;
+ return packet_stream_it_->rtp.log_time_ms();
}
rtc::Optional<int64_t> NextOutputEventTime() const override {
- if (output_events_us_it_ == output_events_us_end_) {
+ if (output_events_it_ == output_events_end_) {
return rtc::nullopt;
}
- if (end_time_us_ && *output_events_us_it_ > *end_time_us_) {
+ if (end_time_ms_ && output_events_it_->log_time_ms() > *end_time_ms_) {
return rtc::nullopt;
}
- // Convert from us to ms.
- return rtc::checked_cast<int64_t>(*output_events_us_it_ / 1000);
+ return output_events_it_->log_time_ms();
}
std::unique_ptr<PacketData> PopPacket() override {
@@ -1600,8 +1597,7 @@
}
std::unique_ptr<PacketData> packet_data(new PacketData());
packet_data->header = packet_stream_it_->rtp.header;
- // Convert from us to ms.
- packet_data->time_ms = packet_stream_it_->rtp.log_time_us() / 1000.0;
+ packet_data->time_ms = packet_stream_it_->rtp.log_time_ms();
// This is a header-only "dummy" packet. Set the payload to all zeros, with
// length according to the virtual length.
@@ -1614,8 +1610,8 @@
}
void AdvanceOutputEvent() override {
- if (output_events_us_it_ != output_events_us_end_) {
- ++output_events_us_it_;
+ if (output_events_it_ != output_events_end_) {
+ ++output_events_it_;
}
}
@@ -1631,9 +1627,9 @@
private:
const std::vector<LoggedRtpPacketIncoming>& packet_stream_;
std::vector<LoggedRtpPacketIncoming>::const_iterator packet_stream_it_;
- std::vector<int64_t>::const_iterator output_events_us_it_;
- const std::vector<int64_t>::const_iterator output_events_us_end_;
- const rtc::Optional<int64_t> end_time_us_;
+ std::vector<LoggedAudioPlayoutEvent>::const_iterator output_events_it_;
+ const std::vector<LoggedAudioPlayoutEvent>::const_iterator output_events_end_;
+ const rtc::Optional<int64_t> end_time_ms_;
};
namespace {
@@ -1642,12 +1638,12 @@
// instrument the test.
std::unique_ptr<test::NetEqStatsGetter> CreateNetEqTestAndRun(
const std::vector<LoggedRtpPacketIncoming>* packet_stream,
- const std::vector<int64_t>* output_events_us,
- rtc::Optional<int64_t> end_time_us,
+ const std::vector<LoggedAudioPlayoutEvent>* output_events,
+ rtc::Optional<int64_t> end_time_ms,
const std::string& replacement_file_name,
int file_sample_rate_hz) {
std::unique_ptr<test::NetEqInput> input(
- new NetEqStreamInput(packet_stream, output_events_us, end_time_us));
+ new NetEqStreamInput(packet_stream, output_events, end_time_ms));
constexpr int kReplacementPt = 127;
std::set<uint8_t> cn_types;
@@ -1698,37 +1694,37 @@
int file_sample_rate_hz) const {
NetEqStatsGetterMap neteq_stats;
- const std::vector<LoggedRtpPacketIncoming>* audio_packets = nullptr;
- uint32_t ssrc;
for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
- if (IsAudioSsrc(kIncomingPacket, stream.ssrc)) {
- audio_packets = &stream.incoming_packets;
- ssrc = stream.ssrc;
- break;
+ const uint32_t ssrc = stream.ssrc;
+ if (!IsAudioSsrc(kIncomingPacket, ssrc))
+ continue;
+ const std::vector<LoggedRtpPacketIncoming>* audio_packets =
+ &stream.incoming_packets;
+ if (audio_packets == nullptr) {
+ // No incoming audio stream found.
+ continue;
}
- }
- if (audio_packets == nullptr) {
- // No incoming audio stream found.
- return neteq_stats;
- }
- std::map<uint32_t, std::vector<int64_t>>::const_iterator output_events_it =
- parsed_log_.audio_playout_events().find(ssrc);
- if (output_events_it == parsed_log_.audio_playout_events().end()) {
- // Could not find output events with SSRC matching the input audio stream.
- // Using the first available stream of output events.
- output_events_it = parsed_log_.audio_playout_events().cbegin();
+ RTC_DCHECK(neteq_stats.find(ssrc) == neteq_stats.end());
+
+ std::map<uint32_t, std::vector<LoggedAudioPlayoutEvent>>::const_iterator
+ output_events_it = parsed_log_.audio_playout_events().find(ssrc);
+ if (output_events_it == parsed_log_.audio_playout_events().end()) {
+ // Could not find output events with SSRC matching the input audio stream.
+ // Using the first available stream of output events.
+ output_events_it = parsed_log_.audio_playout_events().cbegin();
+ }
+
+ rtc::Optional<int64_t> end_time_ms =
+ log_segments_.empty()
+ ? rtc::nullopt
+ : rtc::Optional<int64_t>(log_segments_.front().second / 1000);
+
+ neteq_stats[ssrc] = CreateNetEqTestAndRun(
+ audio_packets, &output_events_it->second, end_time_ms,
+ replacement_file_name, file_sample_rate_hz);
}
- rtc::Optional<int64_t> end_time_us =
- log_segments_.empty()
- ? rtc::nullopt
- : rtc::Optional<int64_t>(log_segments_.front().second);
-
- neteq_stats[ssrc] = CreateNetEqTestAndRun(
- audio_packets, &output_events_it->second, end_time_us,
- replacement_file_name, file_sample_rate_hz);
-
return neteq_stats;
}
@@ -1809,7 +1805,43 @@
plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetYAxis(min_y_axis, max_y_axis, "Relative delay (ms)", kBottomMargin,
kTopMargin);
- plot->SetTitle("NetEq timing");
+ plot->SetTitle("NetEq timing for " + GetStreamName(kIncomingPacket, ssrc));
+}
+
+void EventLogAnalyzer::CreateNetEqStatsGraph(
+ const NetEqStatsGetterMap& neteq_stats,
+ rtc::FunctionView<float(const NetEqNetworkStatistics&)> stats_extractor,
+ const std::string& plot_name,
+ Plot* plot) const {
+ if (neteq_stats.size() < 1)
+ return;
+
+ std::map<uint32_t, TimeSeries> time_series;
+ float min_y_axis = std::numeric_limits<float>::max();
+ float max_y_axis = std::numeric_limits<float>::min();
+
+ for (const auto& st : neteq_stats) {
+ const uint32_t ssrc = st.first;
+ const auto& stats = st.second->stats();
+
+ for (size_t i = 0; i < stats.size(); ++i) {
+ const float time = ToCallTime(stats[i].first * 1000); // ms to us.
+ const float value = stats_extractor(stats[i].second);
+ time_series[ssrc].points.emplace_back(TimeSeriesPoint(time, value));
+ min_y_axis = std::min(min_y_axis, value);
+ max_y_axis = std::max(max_y_axis, value);
+ }
+ }
+
+ for (auto& series : time_series) {
+ series.second.label = GetStreamName(kIncomingPacket, series.first);
+ series.second.line_style = LineStyle::kLine;
+ plot->AppendTimeSeries(std::move(series.second));
+ }
+
+ plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
+ plot->SetYAxis(min_y_axis, max_y_axis, plot_name, kBottomMargin, kTopMargin);
+ plot->SetTitle(plot_name);
}
void EventLogAnalyzer::CreateIceCandidatePairConfigGraph(Plot* plot) {
diff --git a/rtc_tools/event_log_visualizer/analyzer.h b/rtc_tools/event_log_visualizer/analyzer.h
index 572884c..51ae015 100644
--- a/rtc_tools/event_log_visualizer/analyzer.h
+++ b/rtc_tools/event_log_visualizer/analyzer.h
@@ -79,6 +79,11 @@
void CreateAudioJitterBufferGraph(
const NetEqStatsGetterMap& neteq_stats_getters,
Plot* plot) const;
+ void CreateNetEqStatsGraph(
+ const NetEqStatsGetterMap& neteq_stats_getters,
+ rtc::FunctionView<float(const NetEqNetworkStatistics&)> stats_extractor,
+ const std::string& plot_name,
+ Plot* plot) const;
void CreateIceCandidatePairConfigGraph(Plot* plot);
void CreateIceConnectivityCheckGraph(Plot* plot);
diff --git a/rtc_tools/event_log_visualizer/main.cc b/rtc_tools/event_log_visualizer/main.cc
index d12935d..221b040 100644
--- a/rtc_tools/event_log_visualizer/main.cc
+++ b/rtc_tools/event_log_visualizer/main.cc
@@ -112,9 +112,7 @@
DEFINE_bool(plot_audio_encoder_num_channels,
false,
"Plot the audio encoder number of channels.");
-DEFINE_bool(plot_audio_jitter_buffer,
- false,
- "Plot the audio jitter buffer delay profile.");
+DEFINE_bool(plot_neteq_stats, false, "Plot the NetEq statistics.");
DEFINE_bool(plot_ice_candidate_pair_config,
false,
"Plot the ICE candidate pair config events.");
@@ -325,7 +323,7 @@
if (FLAG_plot_audio_encoder_num_channels) {
analyzer.CreateAudioEncoderNumChannelsGraph(collection->AppendNewPlot());
}
- if (FLAG_plot_audio_jitter_buffer) {
+ if (FLAG_plot_neteq_stats) {
std::string wav_path;
if (FLAG_wav_filename[0] != '\0') {
wav_path = FLAG_wav_filename;
@@ -336,6 +334,30 @@
auto neteq_stats = analyzer.SimulateNetEq(wav_path, 48000);
analyzer.CreateAudioJitterBufferGraph(neteq_stats,
collection->AppendNewPlot());
+ analyzer.CreateNetEqStatsGraph(
+ neteq_stats,
+ [](const webrtc::NetEqNetworkStatistics& stats) {
+ return stats.expand_rate / 16384.f;
+ },
+ "Expand rate", collection->AppendNewPlot());
+ analyzer.CreateNetEqStatsGraph(
+ neteq_stats,
+ [](const webrtc::NetEqNetworkStatistics& stats) {
+ return stats.speech_expand_rate / 16384.f;
+ },
+ "Speech expand rate", collection->AppendNewPlot());
+ analyzer.CreateNetEqStatsGraph(
+ neteq_stats,
+ [](const webrtc::NetEqNetworkStatistics& stats) {
+ return stats.accelerate_rate / 16384.f;
+ },
+ "Accelerate rate", collection->AppendNewPlot());
+ analyzer.CreateNetEqStatsGraph(
+ neteq_stats,
+ [](const webrtc::NetEqNetworkStatistics& stats) {
+ return stats.packet_loss_rate / 16384.f;
+ },
+ "Packet loss rate", collection->AppendNewPlot());
}
if (FLAG_plot_ice_candidate_pair_config) {
@@ -382,7 +404,7 @@
FLAG_plot_audio_encoder_fec = setting;
FLAG_plot_audio_encoder_dtx = setting;
FLAG_plot_audio_encoder_num_channels = setting;
- FLAG_plot_audio_jitter_buffer = setting;
+ FLAG_plot_neteq_stats = setting;
FLAG_plot_ice_candidate_pair_config = setting;
FLAG_plot_ice_connectivity_check = setting;
}