Add NetEq delay plotting to event_log_visualizer

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

Index: webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h b/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
new file mode 100644
index 0000000000000000000000000000000000000000..ef5e743f283de79c28d46934fcda41ec282dd4b5
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
@@ -0,0 +1,63 @@
+/*
+ *  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 {
+    TimingData(uint16_t sn, double at) : rtp_sn(sn), arrival_time_ms(at) {}
+    uint16_t rtp_sn;

[ivoc, 2017/05/16 13:25:50]

I think it would be nicer to spell out the abbreviation here, i.e.
rtp_sequence_number.

[hlundin-webrtc, 2017/05/30 14:56:06]

Turns out I didn't even use it...

+    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_