Add NetEq delay plotting to event_log_visualizer

This CL adds the capability to analyze and plot how NetEq behaves in
response to a network trace.

BUG=webrtc:7467

Review-Url: https://codereview.webrtc.org/2876423002
Cr-Commit-Position: refs/heads/master@{#18590}

webrtc/modules/audio_coding/BUILD.gn

@@ -1160,6 +1160,8 @@ rtc_source_set("neteq_tools") {
     "neteq/tools/fake_decode_from_file.h",
     "neteq/tools/input_audio_file.cc",
     "neteq/tools/input_audio_file.h",
+    "neteq/tools/neteq_delay_analyzer.cc",
+    "neteq/tools/neteq_delay_analyzer.h",
     "neteq/tools/neteq_replacement_input.cc",
     "neteq/tools/neteq_replacement_input.h",
     "neteq/tools/resample_input_audio_file.cc",

webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc

@@ -0,0 +1,173 @@
+/*
+ *  Copyright (c) 2017 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.
+ */
+
+#include "webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
+
+#include <algorithm>
+#include <limits>
+#include <utility>
+
+namespace webrtc {
+namespace test {
+namespace {
+// Helper function for NetEqDelayAnalyzer::CreateGraphs. Returns the
+// interpolated value of a function at the point x. Vector x_vec contains the
+// sample points, and y_vec contains the function values at these points. The
+// return value is a linear interpolation between y_vec values.
+double LinearInterpolate(double x,
+                         const std::vector<int64_t>& x_vec,
+                         const std::vector<int64_t>& y_vec) {
+  // Find first element which is larger than x.
+  auto it = std::upper_bound(x_vec.begin(), x_vec.end(), x);
+  if (it == x_vec.end()) {
+    --it;
+  }
+  const size_t upper_ix = it - x_vec.begin();
+
+  size_t lower_ix;
+  if (upper_ix == 0 || x_vec[upper_ix] <= x) {
+    lower_ix = upper_ix;
+  } else {
+    lower_ix = upper_ix - 1;
+  }
+  double y;
+  if (lower_ix == upper_ix) {
+    y = y_vec[lower_ix];
+  } else {
+    RTC_DCHECK_NE(x_vec[lower_ix], x_vec[upper_ix]);
+    y = (x - x_vec[lower_ix]) * (y_vec[upper_ix] - y_vec[lower_ix]) /
+            (x_vec[upper_ix] - x_vec[lower_ix]) +
+        y_vec[lower_ix];
+  }
+  return y;
+}
+}  // namespace
+
+void NetEqDelayAnalyzer::AfterInsertPacket(
+    const test::NetEqInput::PacketData& packet,
+    NetEq* neteq) {
+  data_.insert(
+      std::make_pair(packet.header.timestamp, TimingData(packet.time_ms)));
+}
+
+void NetEqDelayAnalyzer::BeforeGetAudio(NetEq* neteq) {
+  last_sync_buffer_ms_ = neteq->SyncBufferSizeMs();
+}
+
+void NetEqDelayAnalyzer::AfterGetAudio(int64_t time_now_ms,
+                                       const AudioFrame& audio_frame,
+                                       bool /*muted*/,
+                                       NetEq* neteq) {
+  get_audio_time_ms_.push_back(time_now_ms);
+  // Check what timestamps were decoded in the last GetAudio call.
+  std::vector<uint32_t> dec_ts = neteq->LastDecodedTimestamps();
+  // Find those timestamps in data_, insert their decoding time and sync
+  // delay.
+  for (uint32_t ts : dec_ts) {
+    auto it = data_.find(ts);
+    if (it == data_.end()) {
+      // This is a packet that was split out from another packet. Skip it.
+      continue;
+    }
+    auto& it_timing = it->second;
+    RTC_CHECK(!it_timing.decode_get_audio_count)
+        << "Decode time already written";
+    it_timing.decode_get_audio_count = rtc::Optional<int64_t>(get_audio_count_);
+    RTC_CHECK(!it_timing.sync_delay_ms) << "Decode time already written";
+    it_timing.sync_delay_ms = rtc::Optional<int64_t>(last_sync_buffer_ms_);
+    it_timing.target_delay_ms = rtc::Optional<int>(neteq->TargetDelayMs());
+    it_timing.current_delay_ms =
+        rtc::Optional<int>(neteq->FilteredCurrentDelayMs());
+  }
+  last_sample_rate_hz_ = audio_frame.sample_rate_hz_;
+  ++get_audio_count_;
+}
+
+void NetEqDelayAnalyzer::CreateGraphs(
+    std::vector<float>* send_time_s,
+    std::vector<float>* arrival_delay_ms,
+    std::vector<float>* corrected_arrival_delay_ms,
+    std::vector<rtc::Optional<float>>* playout_delay_ms,
+    std::vector<rtc::Optional<float>>* target_delay_ms) const {
+  if (get_audio_time_ms_.empty()) {
+    return;
+  }
+  // Create nominal_get_audio_time_ms, a vector starting at
+  // get_audio_time_ms_[0] and increasing by 10 for each element.
+  std::vector<int64_t> nominal_get_audio_time_ms(get_audio_time_ms_.size());
+  nominal_get_audio_time_ms[0] = get_audio_time_ms_[0];
+  std::transform(
+      nominal_get_audio_time_ms.begin(), nominal_get_audio_time_ms.end() - 1,
+      nominal_get_audio_time_ms.begin() + 1, [](int64_t& x) { return x + 10; });
+  RTC_DCHECK(
+      std::is_sorted(get_audio_time_ms_.begin(), get_audio_time_ms_.end()));
+
+  std::vector<double> rtp_timestamps_ms;
+  double offset = std::numeric_limits<double>::max();
+  TimestampUnwrapper unwrapper;
+  // This loop traverses data_ and populates rtp_timestamps_ms as well as
+  // calculates the base offset.
+  for (auto& d : data_) {
+    rtp_timestamps_ms.push_back(unwrapper.Unwrap(d.first) /
+                                (last_sample_rate_hz_ / 1000.f));
+    offset =
+        std::min(offset, d.second.arrival_time_ms - rtp_timestamps_ms.back());
+  }
+
+  // Calculate send times in seconds for each packet. This is the (unwrapped)
+  // RTP timestamp in ms divided by 1000.
+  send_time_s->resize(rtp_timestamps_ms.size());
+  std::transform(rtp_timestamps_ms.begin(), rtp_timestamps_ms.end(),
+                 send_time_s->begin(), [rtp_timestamps_ms](double x) {
+                   return (x - rtp_timestamps_ms[0]) / 1000.f;
+                 });
+  RTC_DCHECK_EQ(send_time_s->size(), rtp_timestamps_ms.size());
+
+  // This loop traverses the data again and populates the graph vectors. The
+  // reason to have two loops and traverse twice is that the offset cannot be
+  // known until the first traversal is done. Meanwhile, the final offset must
+  // be known already at the start of this second loop.
+  auto data_it = data_.cbegin();
+  for (size_t i = 0; i < send_time_s->size(); ++i, ++data_it) {
+    RTC_DCHECK(data_it != data_.end());
+    const double offset_send_time_ms = rtp_timestamps_ms[i] + offset;
+    const auto& timing = data_it->second;
+    corrected_arrival_delay_ms->push_back(
+        LinearInterpolate(timing.arrival_time_ms, get_audio_time_ms_,
+                          nominal_get_audio_time_ms) -
+        offset_send_time_ms);
+    arrival_delay_ms->push_back(timing.arrival_time_ms - offset_send_time_ms);
+
+    if (timing.decode_get_audio_count) {
+      // This packet was decoded.
+      RTC_DCHECK(timing.sync_delay_ms);
+      const float playout_ms = *timing.decode_get_audio_count * 10 +
+                               get_audio_time_ms_[0] + *timing.sync_delay_ms -
+                               offset_send_time_ms;
+      playout_delay_ms->push_back(rtc::Optional<float>(playout_ms));
+      RTC_DCHECK(timing.target_delay_ms);
+      RTC_DCHECK(timing.current_delay_ms);
+      const float target =
+          playout_ms - *timing.current_delay_ms + *timing.target_delay_ms;
+      target_delay_ms->push_back(rtc::Optional<float>(target));
+    } else {
+      // This packet was never decoded. Mark target and playout delays as empty.
+      playout_delay_ms->push_back(rtc::Optional<float>());
+      target_delay_ms->push_back(rtc::Optional<float>());
+    }
+  }
+  RTC_DCHECK(data_it == data_.end());
+  RTC_DCHECK_EQ(send_time_s->size(), corrected_arrival_delay_ms->size());
+  RTC_DCHECK_EQ(send_time_s->size(), playout_delay_ms->size());
+  RTC_DCHECK_EQ(send_time_s->size(), target_delay_ms->size());
+}
+
+}  // namespace test
+}  // namespace webrtc

webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h

@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2017 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
+
+#include <map>
+#include <vector>
+
+#include "webrtc/base/optional.h"
+#include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h"
+#include "webrtc/modules/audio_coding/neteq/tools/neteq_test.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+class NetEqDelayAnalyzer : public test::NetEqPostInsertPacket,
+                           public test::NetEqGetAudioCallback {
+ public:
+  void AfterInsertPacket(const test::NetEqInput::PacketData& packet,
+                         NetEq* neteq) override;
+
+  void BeforeGetAudio(NetEq* neteq) override;
+
+  void AfterGetAudio(int64_t time_now_ms,
+                     const AudioFrame& audio_frame,
+                     bool muted,
+                     NetEq* neteq) override;
+
+  void CreateGraphs(std::vector<float>* send_times_s,
+                    std::vector<float>* arrival_delay_ms,
+                    std::vector<float>* corrected_arrival_delay_ms,
+                    std::vector<rtc::Optional<float>>* playout_delay_ms,
+                    std::vector<rtc::Optional<float>>* target_delay_ms) const;
+
+ private:
+  struct TimingData {
+    explicit TimingData(double at) : arrival_time_ms(at) {}
+    double arrival_time_ms;
+    rtc::Optional<int64_t> decode_get_audio_count;
+    rtc::Optional<int64_t> sync_delay_ms;
+    rtc::Optional<int> target_delay_ms;
+    rtc::Optional<int> current_delay_ms;
+  };
+  std::map<uint32_t, TimingData> data_;
+  std::vector<int64_t> get_audio_time_ms_;
+  size_t get_audio_count_ = 0;
+  size_t last_sync_buffer_ms_ = 0;
+  int last_sample_rate_hz_ = 0;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_

webrtc/tools/BUILD.gn

@@ -212,6 +212,7 @@ if (rtc_enable_protobuf) {
       "../logging:rtc_event_log_parser",
       "../modules:module_api",
       "../modules/audio_coding:ana_debug_dump_proto",
+      "../modules/audio_coding:neteq_tools",
 
       # TODO(kwiberg): Remove this dependency.
       "../api/audio_codecs:audio_codecs_api",
@@ -246,6 +247,7 @@ if (rtc_include_tests) {
         ":event_log_visualizer_utils",
         "../base:rtc_base_approved",
         "../test:field_trial",
+        "../test:test_support",
       ]
     }
   }

webrtc/tools/DEPS

@@ -5,6 +5,7 @@ include_rules = [
   "+webrtc/logging/rtc_event_log",
   "+webrtc/modules/audio_device",
   "+webrtc/modules/audio_coding/audio_network_adaptor",
+  "+webrtc/modules/audio_coding/neteq/tools",
   "+webrtc/modules/audio_processing",
   "+webrtc/modules/bitrate_controller",
   "+webrtc/modules/congestion_controller",

webrtc/tools/event_log_visualizer/analyzer.cc

@@ -18,6 +18,7 @@
 #include <utility>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/ptr_util.h"
 #include "webrtc/base/rate_statistics.h"
@@ -25,6 +26,12 @@
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/call.h"
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
+#include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h"
+#include "webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
+#include "webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h"
+#include "webrtc/modules/audio_coding/neteq/tools/neteq_test.h"
+#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
@@ -302,6 +309,8 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
   //             this can be removed. Tracking bug: webrtc:6399
   RtpHeaderExtensionMap default_extension_map = GetDefaultHeaderExtensionMap();
 
+  rtc::Optional<uint64_t> last_log_start;
+
   for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
     ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
     if (event_type != ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT &&
@@ -437,12 +446,26 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
         break;
       }
       case ParsedRtcEventLog::LOG_START: {
+        if (last_log_start) {
+          // A LOG_END event was missing. Use last_timestamp.
+          RTC_DCHECK_GE(last_timestamp, *last_log_start);
+          log_segments_.push_back(
+            std::make_pair(*last_log_start, last_timestamp));
+        }
+        last_log_start = rtc::Optional<uint64_t>(parsed_log_.GetTimestamp(i));
         break;
       }
       case ParsedRtcEventLog::LOG_END: {
+        RTC_DCHECK(last_log_start);
+        log_segments_.push_back(
+            std::make_pair(*last_log_start, parsed_log_.GetTimestamp(i)));
+        last_log_start.reset();
         break;
       }
       case ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT: {
+        uint32_t this_ssrc;
+        parsed_log_.GetAudioPlayout(i, &this_ssrc);
+        audio_playout_events_[this_ssrc].push_back(parsed_log_.GetTimestamp(i));
         break;
       }
       case ParsedRtcEventLog::LOSS_BASED_BWE_UPDATE: {
@@ -487,6 +510,10 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
   begin_time_ = first_timestamp;
   end_time_ = last_timestamp;
   call_duration_s_ = static_cast<float>(end_time_ - begin_time_) / 1000000;
+  if (last_log_start) {
+    // The log was missing the last LOG_END event. Fake it.
+    log_segments_.push_back(std::make_pair(*last_log_start, end_time_));
+  }
 }
 
 class BitrateObserver : public CongestionController::Observer,
@@ -1406,5 +1433,246 @@ void EventLogAnalyzer::CreateAudioEncoderNumChannelsGraph(Plot* plot) {
                           kBottomMargin, kTopMargin);
   plot->SetTitle("Reported audio encoder number of channels");
 }
+
+class NetEqStreamInput : public test::NetEqInput {
+ public:
+  // Does not take any ownership, and all pointers must refer to valid objects
+  // that outlive the one constructed.
+  NetEqStreamInput(const std::vector<LoggedRtpPacket>* packet_stream,
+                   const std::vector<uint64_t>* output_events_us,
+                   rtc::Optional<uint64_t> end_time_us)
+      : 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) {
+    RTC_DCHECK(packet_stream);
+    RTC_DCHECK(output_events_us);
+  }
+
+  rtc::Optional<int64_t> NextPacketTime() const override {
+    if (packet_stream_it_ == packet_stream_.end()) {
+      return rtc::Optional<int64_t>();
+    }
+    if (end_time_us_ && packet_stream_it_->timestamp > *end_time_us_) {
+      return rtc::Optional<int64_t>();
+    }
+    // Convert from us to ms.
+    return rtc::Optional<int64_t>(packet_stream_it_->timestamp / 1000);
+  }
+
+  rtc::Optional<int64_t> NextOutputEventTime() const override {
+    if (output_events_us_it_ == output_events_us_end_) {
+      return rtc::Optional<int64_t>();
+    }
+    if (end_time_us_ && *output_events_us_it_ > *end_time_us_) {
+      return rtc::Optional<int64_t>();
+    }
+    // Convert from us to ms.
+    return rtc::Optional<int64_t>(
+        rtc::checked_cast<int64_t>(*output_events_us_it_ / 1000));
+  }
+
+  std::unique_ptr<PacketData> PopPacket() override {
+    if (packet_stream_it_ == packet_stream_.end()) {
+      return std::unique_ptr<PacketData>();
+    }
+    std::unique_ptr<PacketData> packet_data(new PacketData());
+    packet_data->header = packet_stream_it_->header;
+    // Convert from us to ms.
+    packet_data->time_ms = packet_stream_it_->timestamp / 1000.0;
+
+    // This is a header-only "dummy" packet. Set the payload to all zeros, with
+    // length according to the virtual length.
+    packet_data->payload.SetSize(packet_stream_it_->total_length);
+    std::fill_n(packet_data->payload.data(), packet_data->payload.size(), 0);
+
+    ++packet_stream_it_;
+    return packet_data;
+  }
+
+  void AdvanceOutputEvent() override {
+    if (output_events_us_it_ != output_events_us_end_) {
+      ++output_events_us_it_;
+    }
+  }
+
+  bool ended() const override { return !NextEventTime(); }
+
+  rtc::Optional<RTPHeader> NextHeader() const override {
+    if (packet_stream_it_ == packet_stream_.end()) {
+      return rtc::Optional<RTPHeader>();
+    }
+    return rtc::Optional<RTPHeader>(packet_stream_it_->header);
+  }
+
+ private:
+  const std::vector<LoggedRtpPacket>& packet_stream_;
+  std::vector<LoggedRtpPacket>::const_iterator packet_stream_it_;
+  std::vector<uint64_t>::const_iterator output_events_us_it_;
+  const std::vector<uint64_t>::const_iterator output_events_us_end_;
+  const rtc::Optional<uint64_t> end_time_us_;
+};
+
+namespace {
+// Creates a NetEq test object and all necessary input and output helpers. Runs
+// the test and returns the NetEqDelayAnalyzer object that was used to
+// instrument the test.
+std::unique_ptr<test::NetEqDelayAnalyzer> CreateNetEqTestAndRun(
+    const std::vector<LoggedRtpPacket>* packet_stream,
+    const std::vector<uint64_t>* output_events_us,
+    rtc::Optional<uint64_t> end_time_us,
+    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));
+
+  constexpr int kReplacementPt = 127;
+  std::set<uint8_t> cn_types;
+  std::set<uint8_t> forbidden_types;
+  input.reset(new test::NetEqReplacementInput(std::move(input), kReplacementPt,
+                                              cn_types, forbidden_types));
+
+  NetEq::Config config;
+  config.max_packets_in_buffer = 200;
+  config.enable_fast_accelerate = true;
+
+  std::unique_ptr<test::VoidAudioSink> output(new test::VoidAudioSink());
+
+  test::NetEqTest::DecoderMap codecs;
+
+  // Create a "replacement decoder" that produces the decoded audio by reading
+  // from a file rather than from the encoded payloads.
+  std::unique_ptr<test::ResampleInputAudioFile> replacement_file(
+      new test::ResampleInputAudioFile(replacement_file_name,
+                                       file_sample_rate_hz));
+  replacement_file->set_output_rate_hz(48000);
+  std::unique_ptr<AudioDecoder> replacement_decoder(
+      new test::FakeDecodeFromFile(std::move(replacement_file), 48000, false));
+  test::NetEqTest::ExtDecoderMap ext_codecs;
+  ext_codecs[kReplacementPt] = {replacement_decoder.get(),
+                                NetEqDecoder::kDecoderArbitrary,
+                                "replacement codec"};
+
+  std::unique_ptr<test::NetEqDelayAnalyzer> delay_cb(
+      new test::NetEqDelayAnalyzer);
+  test::DefaultNetEqTestErrorCallback error_cb;
+  test::NetEqTest::Callbacks callbacks;
+  callbacks.error_callback = &error_cb;
+  callbacks.post_insert_packet = delay_cb.get();
+  callbacks.get_audio_callback = delay_cb.get();
+
+  test::NetEqTest test(config, codecs, ext_codecs, std::move(input),
+                       std::move(output), callbacks);
+  test.Run();
+  return delay_cb;
+}
+}  // namespace
+
+// Plots the jitter buffer delay profile. This will plot only for the first
+// incoming audio SSRC. If the stream contains more than one incoming audio
+// SSRC, all but the first will be ignored.
+void EventLogAnalyzer::CreateAudioJitterBufferGraph(
+    const std::string& replacement_file_name,
+    int file_sample_rate_hz,
+    Plot* plot) {
+  const auto& incoming_audio_kv = std::find_if(
+      rtp_packets_.begin(), rtp_packets_.end(),
+      [this](std::pair<StreamId, std::vector<LoggedRtpPacket>> kv) {
+        return kv.first.GetDirection() == kIncomingPacket &&
+               this->IsAudioSsrc(kv.first);
+      });
+  if (incoming_audio_kv == rtp_packets_.end()) {
+    // No incoming audio stream found.
+    return;
+  }
+
+  const uint32_t ssrc = incoming_audio_kv->first.GetSsrc();
+
+  std::map<uint32_t, std::vector<uint64_t>>::const_iterator output_events_it =
+      audio_playout_events_.find(ssrc);
+  if (output_events_it == 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 = audio_playout_events_.cbegin();
+  }
+
+  rtc::Optional<uint64_t> end_time_us =
+      log_segments_.empty()
+          ? rtc::Optional<uint64_t>()
+          : rtc::Optional<uint64_t>(log_segments_.front().second);
+
+  auto delay_cb = CreateNetEqTestAndRun(
+      &incoming_audio_kv->second, &output_events_it->second, end_time_us,
+      replacement_file_name, file_sample_rate_hz);
+
+  std::vector<float> send_times_s;
+  std::vector<float> arrival_delay_ms;
+  std::vector<float> corrected_arrival_delay_ms;
+  std::vector<rtc::Optional<float>> playout_delay_ms;
+  std::vector<rtc::Optional<float>> target_delay_ms;
+  delay_cb->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());
+
+  std::map<StreamId, TimeSeries> time_series_packet_arrival;
+  std::map<StreamId, TimeSeries> time_series_relative_packet_arrival;
+  std::map<StreamId, TimeSeries> time_series_play_time;
+  std::map<StreamId, TimeSeries> time_series_target_time;
+  float min_y_axis = 0.f;
+  float max_y_axis = 0.f;
+  const StreamId stream_id = incoming_audio_kv->first;
+  for (size_t i = 0; i < send_times_s.size(); ++i) {
+    time_series_packet_arrival[stream_id].points.emplace_back(
+        TimeSeriesPoint(send_times_s[i], arrival_delay_ms[i]));
+    time_series_relative_packet_arrival[stream_id].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[stream_id].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[stream_id].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]);
+    }
+  }
+
+  // This code is adapted for a single stream. The creation of the streams above
+  // guarantee that no more than one steam is included. If multiple streams are
+  // to be plotted, they should likely be given distinct labels below.
+  RTC_DCHECK_EQ(time_series_relative_packet_arrival.size(), 1);
+  for (auto& series : time_series_relative_packet_arrival) {
+    series.second.label = "Relative packet arrival delay";
+    series.second.style = LINE_GRAPH;
+    plot->AppendTimeSeries(std::move(series.second));
+  }
+  RTC_DCHECK_EQ(time_series_play_time.size(), 1);
+  for (auto& series : time_series_play_time) {
+    series.second.label = "Playout delay";
+    series.second.style = LINE_GRAPH;
+    plot->AppendTimeSeries(std::move(series.second));
+  }
+  RTC_DCHECK_EQ(time_series_target_time.size(), 1);
+  for (auto& series : time_series_target_time) {
+    series.second.label = "Target delay";
+    series.second.style = LINE_DOT_GRAPH;
+    plot->AppendTimeSeries(std::move(series.second));
+  }
+
+  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");
+}
 }  // namespace plotting
 }  // namespace webrtc

webrtc/tools/event_log_visualizer/analyzer.h

@@ -100,6 +100,9 @@ class EventLogAnalyzer {
   void CreateAudioEncoderEnableFecGraph(Plot* plot);
   void CreateAudioEncoderEnableDtxGraph(Plot* plot);
   void CreateAudioEncoderNumChannelsGraph(Plot* plot);
+  void CreateAudioJitterBufferGraph(const std::string& replacement_file_name,
+                                    int file_sample_rate_hz,
+                                    Plot* plot);
 
   // Returns a vector of capture and arrival timestamps for the video frames
   // of the stream with the most number of frames.
@@ -163,6 +166,13 @@ class EventLogAnalyzer {
 
   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<uint64_t, uint64_t>> log_segments_;
+
   // A list of all updates from the send-side loss-based bandwidth estimator.
   std::vector<LossBasedBweUpdate> bwe_loss_updates_;
 

webrtc/tools/event_log_visualizer/main.cc

@@ -13,6 +13,7 @@
 #include "webrtc/base/flags.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 #include "webrtc/test/field_trial.h"
+#include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/tools/event_log_visualizer/analyzer.h"
 #include "webrtc/tools/event_log_visualizer/plot_base.h"
 #include "webrtc/tools/event_log_visualizer/plot_python.h"
@@ -77,6 +78,9 @@ DEFINE_bool(audio_encoder_dtx, false, "Plot the audio encoder DTX.");
 DEFINE_bool(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_string(
     force_fieldtrials,
     "",
@@ -105,6 +109,7 @@ int main(int argc, char* argv[]) {
     return 0;
   }
 
+  webrtc::test::SetExecutablePath(argv[0]);
   webrtc::test::InitFieldTrialsFromString(FLAG_force_fieldtrials);
 
   std::string filename = argv[1];
@@ -231,6 +236,14 @@ int main(int argc, char* argv[]) {
     analyzer.CreateAudioEncoderNumChannelsGraph(collection->AppendNewPlot());
   }
 
+  if (FLAG_plot_all || FLAG_plot_audio_jitter_buffer) {
+    analyzer.CreateAudioJitterBufferGraph(
+        webrtc::test::ResourcePath(
+            "audio_processing/conversational_speech/EN_script2_F_sp2_B1",
+            "wav"),
+        48000, collection->AppendNewPlot());
+  }
+
   collection->Draw();
 
   return 0;