Add ana config to event log visualiser

webrtc/tools/event_log_visualizer/analyzer.cc

 /*
  *  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.
  */
 
@@ skipping 90 matching lines @@
 // TODO(ivoc): Remove this once this mapping is stored in the event log for
 //             audio streams. Tracking bug: webrtc:6399
 webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap() {
   webrtc::RtpHeaderExtensionMap default_map;
   default_map.Register<AudioLevel>(webrtc::RtpExtension::kAudioLevelDefaultId);
   default_map.Register<AbsoluteSendTime>(
       webrtc::RtpExtension::kAbsSendTimeDefaultId);
   return default_map;
 }
 
+void FillAudioEncoderTimeSeries(

[minyue-webrtc, 2017/02/16 09:54:56]

I don't mind making this a member method to avoid passing
audio_network_adaptation_events_ and begin_time_.

[terelius, 2017/02/16 10:02:27]

The function "Pointwise" is quite similar to this except that it takes a
function object rather than a lambda. (Name comes from processing the points
one-by-one.) Maybe it would be useful to merge them?

[michaelt, 2017/02/16 10:24:22]

The rtc::Optional in "FillAudioEncoderTimeSeries" is different than in
"Pointwise".
But we still should be able to merge them. The questions is just which approach
we will use. lambda or function object.
I would prefer lambda's since otherwise I would have to create a object for each
CreateAudioEncoder function. But I'm open for your ideas.

[michaelt, 2017/02/16 10:29:42]

On the other hand it would be quite a work, to change the current impl. to
lambdas, as well.

+    Plot* plot,
+    const std::vector<AudioNetworkAdaptationEvent>&
+        audio_network_adaptation_events,
+    uint64_t begin_time,
+    std::function<rtc::Optional<float>(
+        const AudioNetworkAdaptationEvent& ana_event)> get_y) {
+  plot->series_list_.push_back(TimeSeries());
+  plot->series_list_.back().style = LINE_DOT_GRAPH;
+  for (auto& ana_event : audio_network_adaptation_events) {
+    rtc::Optional<float> y = get_y(ana_event);
+    if (y) {
+      float x = static_cast<float>(ana_event.timestamp - begin_time) / 1000000;
+      plot->series_list_.back().points.emplace_back(x, *y);
+    }
+  }
+}
+
 constexpr float kLeftMargin = 0.01f;
 constexpr float kRightMargin = 0.02f;
 constexpr float kBottomMargin = 0.02f;
 constexpr float kTopMargin = 0.05f;
 
 class PacketSizeBytes {
  public:
   using DataType = LoggedRtpPacket;
   using ResultType = size_t;
   size_t operator()(const LoggedRtpPacket& packet) {
@@ skipping 317 matching lines @@
       case ParsedRtcEventLog::BWE_PACKET_LOSS_EVENT: {
         BwePacketLossEvent bwe_update;
         bwe_update.timestamp = parsed_log_.GetTimestamp(i);
         parsed_log_.GetBwePacketLossEvent(i, &bwe_update.new_bitrate,
                                              &bwe_update.fraction_loss,
                                              &bwe_update.expected_packets);
         bwe_loss_updates_.push_back(bwe_update);
         break;
       }
       case ParsedRtcEventLog::AUDIO_NETWORK_ADAPTATION_EVENT: {
+        AudioNetworkAdaptationEvent ana_event;
+        ana_event.timestamp = parsed_log_.GetTimestamp(i);
+        parsed_log_.GetAudioNetworkAdaptation(i, &ana_event.config);
+        audio_network_adaptation_events_.push_back(ana_event);
         break;
       }
       case ParsedRtcEventLog::BWE_PACKET_DELAY_EVENT: {
         break;
       }
       case ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT: {
         break;
       }
       case ParsedRtcEventLog::UNKNOWN_EVENT: {
         break;
@@ skipping 809 matching lines @@
         }
         plot->series_list_.push_back(std::move(timestamp_data));
       }
     }
   }
 
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetSuggestedYAxis(0, 1, "Timestamp (90khz)", kBottomMargin, kTopMargin);
   plot->SetTitle("Timestamps");
 }
+
+void EventLogAnalyzer::CreateAudioEncoderTargetBitrateGraph(Plot* plot) {
+  FillAudioEncoderTimeSeries(
+      plot, audio_network_adaptation_events_, begin_time_,
+      [](const AudioNetworkAdaptationEvent& ana_event) {
+        if (ana_event.config.bitrate_bps)
+          return rtc::Optional<float>(
+              static_cast<float>(*ana_event.config.bitrate_bps));
+        return rtc::Optional<float>();
+      });
+  plot->series_list_.back().label = "Audio encoder target bitrate";
+  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Bitrate (bps)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Reported audio encoder target bitrate");
+}
+
+void EventLogAnalyzer::CreateAudioEncoderFrameLengthGraph(Plot* plot) {
+  FillAudioEncoderTimeSeries(
+      plot, audio_network_adaptation_events_, begin_time_,
+      [](const AudioNetworkAdaptationEvent& ana_event) {
+        if (ana_event.config.frame_length_ms)
+          return rtc::Optional<float>(
+              static_cast<float>(*ana_event.config.frame_length_ms));
+        return rtc::Optional<float>();
+      });
+  plot->series_list_.back().label = "Audio encoder frame length";
+  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Frame length (ms)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Reported audio encoder frame length");
+}
+
+void EventLogAnalyzer::CreateAudioEncoderUplinkPacketLossFractionGraph(
+    Plot* plot) {
+  FillAudioEncoderTimeSeries(
+      plot, audio_network_adaptation_events_, begin_time_,
+      [&](const AudioNetworkAdaptationEvent& ana_event) {
+        if (ana_event.config.uplink_packet_loss_fraction)
+          return rtc::Optional<float>(static_cast<float>(
+              *ana_event.config.uplink_packet_loss_fraction));
+        return rtc::Optional<float>();
+      });
+  plot->series_list_.back().label = "Audio encoder uplink packet loss fraction";
+  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 10, "Percent lost packets", kBottomMargin,
+                          kTopMargin);
+  plot->SetTitle("Reported audio encoder lost packets");
+}
+
+void EventLogAnalyzer::CreateAudioEncoderEnableFecGraph(Plot* plot) {
+  FillAudioEncoderTimeSeries(
+      plot, audio_network_adaptation_events_, begin_time_,
+      [&](const AudioNetworkAdaptationEvent& ana_event) {
+        if (ana_event.config.enable_fec)
+          return rtc::Optional<float>(
+              static_cast<float>(*ana_event.config.enable_fec));
+        return rtc::Optional<float>();
+      });
+  plot->series_list_.back().label = "Audio encoder FEC";
+  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "FEC (false/true)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Reported audio encoder FEC");
+}
+
+void EventLogAnalyzer::CreateAudioEncoderEnableDtxGraph(Plot* plot) {
+  FillAudioEncoderTimeSeries(
+      plot, audio_network_adaptation_events_, begin_time_,
+      [&](const AudioNetworkAdaptationEvent& ana_event) {
+        if (ana_event.config.enable_dtx)
+          return rtc::Optional<float>(
+              static_cast<float>(*ana_event.config.enable_dtx));
+        return rtc::Optional<float>();
+      });
+  plot->series_list_.back().label = "Audio encoder DTX";
+  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "DTX (false/true)", kBottomMargin, kTopMargin);
+  plot->SetTitle("Reported audio encoder DTX");
+}
+
+void EventLogAnalyzer::CreateAudioEncoderNumChannelsGraph(Plot* plot) {
+  FillAudioEncoderTimeSeries(
+      plot, audio_network_adaptation_events_, begin_time_,
+      [&](const AudioNetworkAdaptationEvent& ana_event) {
+        if (ana_event.config.num_channels)
+          return rtc::Optional<float>(
+              static_cast<float>(*ana_event.config.num_channels));
+        return rtc::Optional<float>();
+      });
+  plot->series_list_.back().label = "Audio encoder number of channels";
+  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
+  plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))",
+                          kBottomMargin, kTopMargin);
+  plot->SetTitle("Reported audio encoder number of channels");
+}
 }  // namespace plotting
 }  // namespace webrtc