Adds delta-time logging for audio playout

webrtc/modules/audio_device/audio_device_buffer.cc

 /*
  *  Copyright (c) 2012 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 <algorithm>
+
 #include "webrtc/modules/audio_device/audio_device_buffer.h"
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_device/audio_device_config.h"
 
 namespace webrtc {
 
 static const int kHighDelayThresholdMs = 300;
@@ skipping 42 matching lines @@
       last_play_samples_(0),
       last_log_stat_time_(0) {
   LOG(INFO) << "AudioDeviceBuffer::ctor";
   memset(_recBuffer, 0, kMaxBufferSizeBytes);
   memset(_playBuffer, 0, kMaxBufferSizeBytes);
 }
 
 AudioDeviceBuffer::~AudioDeviceBuffer() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(INFO) << "AudioDeviceBuffer::~dtor";
+
+  int total_diff_time = 0;
+  int num_measurements = 0;
+  LOG(INFO) << "[playout diff time => #measurements]";
+  for (size_t diff = 0; diff < arraysize(playout_diff_times_); ++diff) {
+    int num_elements = playout_diff_times_[diff];
+    if (num_elements > 0) {
+      total_diff_time += num_elements * diff;
+      num_measurements += num_elements;
+      LOG(INFO) << "[" << diff << " => " << num_elements << "]";
+    }
+  }
+  LOG(INFO) << "total_diff_time: " << total_diff_time;
+  LOG(INFO) << "num_measurements: " << num_measurements;
+  LOG(INFO) << "average: "
+            << static_cast<float>(total_diff_time) / num_measurements;
+
   _recFile.Flush();
   _recFile.CloseFile();
   delete &_recFile;
 
   _playFile.Flush();
   _playFile.CloseFile();
   delete &_playFile;
 }
 
 int32_t AudioDeviceBuffer::RegisterAudioCallback(
     AudioTransport* audioCallback) {
   LOG(INFO) << __FUNCTION__;
   rtc::CritScope lock(&_critSectCb);
   _ptrCbAudioTransport = audioCallback;
   return 0;
 }
 
 int32_t AudioDeviceBuffer::InitPlayout() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(INFO) << __FUNCTION__;
+  last_playout_time_ = rtc::TimeMillis();
   if (!timer_has_started_) {
     StartTimer();
     timer_has_started_ = true;
   }
   return 0;
 }
 
 int32_t AudioDeviceBuffer::InitRecording() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(INFO) << __FUNCTION__;
@@ skipping 221 matching lines @@
   }
 
   return 0;
 }
 
 int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples) {
   uint32_t playSampleRate = 0;
   size_t playBytesPerSample = 0;
   size_t playChannels = 0;
 
+  // Measure time since last function call and update an array where the
+  // position/index corresponds to time differences (in milliseconds) between
+  // two successive playout callbacks, and the stored value is the number of
+  // times a given time difference was found.
+  int64_t now_time = rtc::TimeMillis();
+  size_t diff_time = rtc::TimeDiff(now_time, last_playout_time_);
+  // Truncate at 500ms to limit the size of the array.
+  diff_time = std::min(kMaxDeltaTimeInMs, diff_time);
+  last_playout_time_ = now_time;
+  playout_diff_times_[diff_time]++;
+
   // TOOD(henrika): improve bad locking model and make it more clear that only
   // 10ms buffer sizes is supported in WebRTC.
   {
     rtc::CritScope lock(&_critSect);
 
     // Store copies under lock and use copies hereafter to avoid race with
     // setter methods.
     playSampleRate = _playSampleRate;
     playBytesPerSample = _playBytesPerSample;
     playChannels = _playChannels;
@@ skipping 111 matching lines @@
   rec_samples_ += num_samples;
 }
 
 void AudioDeviceBuffer::UpdatePlayStats(size_t num_samples) {
   RTC_DCHECK(task_queue_.IsCurrent());
   ++play_callbacks_;
   play_samples_ += num_samples;
 }
 
 }  // namespace webrtc