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Aligning time in audio jitter buffer plot to other plots in rtc event log visualizer.

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Bug: webrtc:9147
Change-Id: I4ddb3e93ea04a11a68e097ecad731d6d9d6842a9
Reviewed-on: https://webrtc-review.googlesource.com/75322
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23712}
This commit is contained in:
Minyue Li
2018-06-21 11:47:14 +02:00
committed by Commit Bot
parent 1ec04f19c6
commit 45fc6dfaaa
4 changed files with 133 additions and 172 deletions
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75
rtc_tools/event_log_visualizer/analyzer.cc
View File

@ -1777,49 +1777,43 @@ EventLogAnalyzer::NetEqStatsGetterMap EventLogAnalyzer::SimulateNetEq(
void EventLogAnalyzer::CreateAudioJitterBufferGraph(
const NetEqStatsGetterMap& neteq_stats,
Plot* plot) const {
if (neteq_stats.size() < 1)
return;
RTC_CHECK(!neteq_stats.empty());
const uint32_t ssrc = neteq_stats.begin()->first;
std::vector<float> send_times_s;
std::vector<float> arrival_delay_ms;
std::vector<float> corrected_arrival_delay_ms;
std::vector<absl::optional<float>> playout_delay_ms;
std::vector<absl::optional<float>> target_delay_ms;
test::NetEqDelayAnalyzer::Delays arrival_delay_ms;
test::NetEqDelayAnalyzer::Delays corrected_arrival_delay_ms;
test::NetEqDelayAnalyzer::Delays playout_delay_ms;
test::NetEqDelayAnalyzer::Delays target_delay_ms;
neteq_stats.at(ssrc)->delay_analyzer()->CreateGraphs(
&send_times_s, &arrival_delay_ms, &corrected_arrival_delay_ms,
&playout_delay_ms, &target_delay_ms);
RTC_DCHECK_EQ(send_times_s.size(), arrival_delay_ms.size());
RTC_DCHECK_EQ(send_times_s.size(), corrected_arrival_delay_ms.size());
RTC_DCHECK_EQ(send_times_s.size(), playout_delay_ms.size());
RTC_DCHECK_EQ(send_times_s.size(), target_delay_ms.size());
&arrival_delay_ms, &corrected_arrival_delay_ms, &playout_delay_ms,
&target_delay_ms);
std::map<uint32_t, TimeSeries> time_series_packet_arrival;
std::map<uint32_t, TimeSeries> time_series_relative_packet_arrival;
std::map<uint32_t, TimeSeries> time_series_play_time;
std::map<uint32_t, TimeSeries> time_series_target_time;
float min_y_axis = 0.f;
float max_y_axis = 0.f;
for (size_t i = 0; i < send_times_s.size(); ++i) {
time_series_packet_arrival[ssrc].points.emplace_back(
TimeSeriesPoint(send_times_s[i], arrival_delay_ms[i]));
for (const auto& data : arrival_delay_ms) {
const float x = ToCallTimeSec(data.first * 1000); // ms to us.
const float y = data.second;
time_series_packet_arrival[ssrc].points.emplace_back(TimeSeriesPoint(x, y));
}
for (const auto& data : corrected_arrival_delay_ms) {
const float x = ToCallTimeSec(data.first * 1000); // ms to us.
const float y = data.second;
time_series_relative_packet_arrival[ssrc].points.emplace_back(
TimeSeriesPoint(send_times_s[i], corrected_arrival_delay_ms[i]));
min_y_axis = std::min(min_y_axis, corrected_arrival_delay_ms[i]);
max_y_axis = std::max(max_y_axis, corrected_arrival_delay_ms[i]);
if (playout_delay_ms[i]) {
time_series_play_time[ssrc].points.emplace_back(
TimeSeriesPoint(send_times_s[i], *playout_delay_ms[i]));
min_y_axis = std::min(min_y_axis, *playout_delay_ms[i]);
max_y_axis = std::max(max_y_axis, *playout_delay_ms[i]);
}
if (target_delay_ms[i]) {
time_series_target_time[ssrc].points.emplace_back(
TimeSeriesPoint(send_times_s[i], *target_delay_ms[i]));
min_y_axis = std::min(min_y_axis, *target_delay_ms[i]);
max_y_axis = std::max(max_y_axis, *target_delay_ms[i]);
}
TimeSeriesPoint(x, y));
}
for (const auto& data : playout_delay_ms) {
const float x = ToCallTimeSec(data.first * 1000); // ms to us.
const float y = data.second;
time_series_play_time[ssrc].points.emplace_back(TimeSeriesPoint(x, y));
}
for (const auto& data : target_delay_ms) {
const float x = ToCallTimeSec(data.first * 1000); // ms to us.
const float y = data.second;
time_series_target_time[ssrc].points.emplace_back(TimeSeriesPoint(x, y));
}
// This code is adapted for a single stream. The creation of the streams above
@ -1847,8 +1841,8 @@ void EventLogAnalyzer::CreateAudioJitterBufferGraph(
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetYAxis(min_y_axis, max_y_axis, "Relative delay (ms)", kBottomMargin,
kTopMargin);
plot->SetSuggestedYAxis(0, 1, "Relative delay (ms)", kBottomMargin,
kTopMargin);
plot->SetTitle("NetEq timing for " + GetStreamName(kIncomingPacket, ssrc));
}
@ -1857,12 +1851,7 @@ void EventLogAnalyzer::CreateNetEqStatsGraph(
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;
@ -1872,8 +1861,6 @@ void EventLogAnalyzer::CreateNetEqStatsGraph(
const float time = ToCallTimeSec(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);
}
}
@ -1885,7 +1872,7 @@ void EventLogAnalyzer::CreateNetEqStatsGraph(
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetYAxis(min_y_axis, max_y_axis, plot_name, kBottomMargin, kTopMargin);
plot->SetSuggestedYAxis(0, 1, plot_name, kBottomMargin, kTopMargin);
plot->SetTitle(plot_name);
}

8
rtc_tools/event_log_visualizer/main.cc
View File

@ -335,8 +335,12 @@ int main(int argc, char* argv[]) {
"audio_processing/conversational_speech/EN_script2_F_sp2_B1", "wav");
}
auto neteq_stats = analyzer.SimulateNetEq(wav_path, 48000);
analyzer.CreateAudioJitterBufferGraph(neteq_stats,
collection->AppendNewPlot());
if (!neteq_stats.empty()) {
analyzer.CreateAudioJitterBufferGraph(neteq_stats,
collection->AppendNewPlot());
}
analyzer.CreateNetEqStatsGraph(
neteq_stats,
[](const webrtc::NetEqNetworkStatistics& stats) {