commit 01d2a67a70adf52a8304929d948aa2a8f5aeadcf
author Minyue Li <minyue@webrtc.org> Thu Jun 21 21:17:19 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Mon Jun 25 12:17:39 2018 +0000
tree 29120b3a05b663b7500bb588f762b1a57b943cb1
parent 91280e4d0402478f0d8b89a6a59071309bf03320

Adding jitter buffer plots for all SSRCs in event log visualizer.

Bug: webrtc:9147
Change-Id: I64291666d329c026f35ecf1c4245b192794441fe
Reviewed-on: https://webrtc-review.googlesource.com/84745
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23726}

rtc_tools/event_log_visualizer/analyzer.cc

diff --git a/rtc_tools/event_log_visualizer/analyzer.cc b/rtc_tools/event_log_visualizer/analyzer.cc
index 9e86eb0..99235e0 100644
--- a/rtc_tools/event_log_visualizer/analyzer.cc
+++ b/rtc_tools/event_log_visualizer/analyzer.cc
@@ -1775,17 +1775,15 @@
 // incoming audio SSRC. If the stream contains more than one incoming audio
 // SSRC, all but the first will be ignored.
 void EventLogAnalyzer::CreateAudioJitterBufferGraph(
-    const NetEqStatsGetterMap& neteq_stats,
+    uint32_t ssrc,
+    const test::NetEqStatsGetter* stats_getter,
     Plot* plot) const {
-  RTC_CHECK(!neteq_stats.empty());
-  const uint32_t ssrc = neteq_stats.begin()->first;
-
   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(
+  stats_getter->delay_analyzer()->CreateGraphs(
       &arrival_delay_ms, &corrected_arrival_delay_ms, &playout_delay_ms,
       &target_delay_ms);
 

rtc_tools/event_log_visualizer/analyzer.h

diff --git a/rtc_tools/event_log_visualizer/analyzer.h b/rtc_tools/event_log_visualizer/analyzer.h
index d4a7fc3..6c7e056 100644
--- a/rtc_tools/event_log_visualizer/analyzer.h
+++ b/rtc_tools/event_log_visualizer/analyzer.h
@@ -76,9 +76,9 @@
       std::map<uint32_t, std::unique_ptr<test::NetEqStatsGetter>>;
   NetEqStatsGetterMap SimulateNetEq(const std::string& replacement_file_name,
                                     int file_sample_rate_hz) const;
-  void CreateAudioJitterBufferGraph(
-      const NetEqStatsGetterMap& neteq_stats_getters,
-      Plot* plot) const;
+  void CreateAudioJitterBufferGraph(uint32_t ssrc,
+                                    const test::NetEqStatsGetter* stats_getter,
+                                    Plot* plot) const;
   void CreateNetEqStatsGraph(
       const NetEqStatsGetterMap& neteq_stats_getters,
       rtc::FunctionView<float(const NetEqNetworkStatistics&)> stats_extractor,

rtc_tools/event_log_visualizer/main.cc

diff --git a/rtc_tools/event_log_visualizer/main.cc b/rtc_tools/event_log_visualizer/main.cc
index 7d9d45e..8a303b2 100644
--- a/rtc_tools/event_log_visualizer/main.cc
+++ b/rtc_tools/event_log_visualizer/main.cc
@@ -335,12 +335,12 @@
           "audio_processing/conversational_speech/EN_script2_F_sp2_B1", "wav");
     }
     auto neteq_stats = analyzer.SimulateNetEq(wav_path, 48000);
-
-    if (!neteq_stats.empty()) {
-      analyzer.CreateAudioJitterBufferGraph(neteq_stats,
+    for (webrtc::EventLogAnalyzer::NetEqStatsGetterMap::const_iterator it =
+             neteq_stats.cbegin();
+         it != neteq_stats.cend(); ++it) {
+      analyzer.CreateAudioJitterBufferGraph(it->first, it->second.get(),
                                             collection->AppendNewPlot());
     }
-
     analyzer.CreateNetEqStatsGraph(
         neteq_stats,
         [](const webrtc::NetEqNetworkStatistics& stats) {