Add NetEq delay plotting to event_log_visualizer

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

+/*
+ *  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 {
+
+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_;
+}
+
+namespace {

[ivoc, 2017/05/30 16:29:45]

Shouldn't this block be at the top of the file?

+// 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.
+// TODO(henrik.lundin) Make this faster. This is the main bottleneck.
+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::find_if(x_vec.begin(), x_vec.end(),

[ivoc, 2017/05/30 16:29:45]

If we know that x_vec is sorted (is it?), this can be done more efficiently with
std::upper_bound, which uses a binary search.

[hlundin-webrtc, 2017/05/31 06:38:33]

Awesome! Great improvement.

+                         [x](int64_t v) -> bool { return v > 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::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